From mailinglist at fribert.dk  Mon Feb  1 00:28:51 2010
From: mailinglist at fribert.dk (mailinglist)
Date: Mon, 01 Feb 2010 09:28:51 +0100
Subject: [Freeswitch-users] Svar: Re: Somebody help me understand the	
	'features' set up please :-)
In-Reply-To: <b39273531001311645u618ea87am8a3d84ee6b1bcfcb@mail.gmail.com>
References: <4B66226C020000E10000043C@mail.fribert.dk>
	<b39273531001311645u618ea87am8a3d84ee6b1bcfcb@mail.gmail.com>
Message-ID: <4B669ED3020000E100000447@mail.fribert.dk>

Hi Rupa
 
Ahh, I see.
Looking at example 2, and http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bind_meta_app
I still need to get where I enter this.
Is it in the call handling in the dialplan?
So I've gotten my current entry to handle incoming call from the outside:
 
<extension name="PrivateIn">
   <condition field="destination_number" expression="^(8202)$">
       <action application="set" data="call_timeout=45"/>
       <action application="set" data="domain_name=10.11.12.25"/>
       <action application="set" data="continue_on_fail=true"/>
       <action application="set" data="hangup_after_bridge=true"/>
       <action application="set" data="transfer_ringback=$${hold_music}"/>
       <action application="bridge" data="group/private@${domain_name}"/ ( 'mailto:group/private@$domain_name"/' )>
       <action application="answer" data=""/>
       <action application="sleep" data="1000"/>
       <action application="voicemail" data="default 10.11.12.25 45"/>
   </condition>
</extension>
Do I stick in there, or do I enter it in features.xml, or?
 
 
Best regards
Fribse
 
/Running Freeswitch on pfSense running on a VMWare ESXi 4.0


>>> 01-02-2010 kl. 01:45 skrev Rupa Schomaker <rupa at rupa.com> i meddelelsen <b39273531001311645u618ea87am8a3d84ee6b1bcfcb at mail.gmail.com>:

Look at bind_meta_app in the default dialplan.  It binds the dtmf to
the features context.

On Sun, Jan 31, 2010 at 5:38 PM, mailinglist <mailinglist at fribert.dk> wrote:
> Ok, I've gotten the Freeswitch to register to my VoIP provider.
> I've gotten my phones to register to Freeswitch, and I can receive and make
> calls, all very nice.
>
> I've gotten a Sipura SPA901 and a Siemens Gigaset S68IP registered to the
> Freeswitch.
>
> When I receive a call, I would like to be able to transfer the call to
> another phone, or change the call to a conference call with two local
> phones.
>
> So I've been looking at the examples in the wiki, and I can't make them
> work, not as I understand them anyways. Especially the att_xfer seems to be
> able to do what I need.
>
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_att_xfer
>
> As I understand Example1, I should answer the call, and then press *3 during
> the call, and either transfer it or change it to a threeway call.
>
> I get the first part, create an extension in the dialplan called att_xfer.
> But what is meant by the second par 'then bind this feature to DTMF 3', how
> do I enter that, and where?
>
> I hope somebody can help me with this (again)?
>
>
>
> Best regards
> Fribse
>
> /Running Freeswitch on pfSense running on a VMWare ESXi 4.0
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
-Rupa

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100201/fbf23b78/attachment.html 

From codecomplete at free.fr  Mon Feb  1 01:22:28 2010
From: codecomplete at free.fr (Fred-145)
Date: Mon, 01 Feb 2010 10:22:28 +0100
Subject: [Freeswitch-users] Equivalent to Asterisk's"directrtpsetup=yes"?
References: <20100129164359.9B3C22B10D@cuneorg-email.cune.pri><j4e8m5tqfl3fo8ug7d3sdgdiu7gofidbj8@4ax.com><j4e8m5tqfl3fo8ug7d3sdgdiu7gofidbj8-e09XROE/p8c@public.gmane.org><4468a6771001310247n4bf5d1a4rf4aea9f2d6c35a1f@mail.gmail.com>
	<qiqbm55lueuqup1m6r6d0mff36b4l4fb9l@4ax.com>
	<15D48404014D48D19F85CFFFC4BBC76F@cune.pri>
	<15D48404014D48D19F85CFFFC4BBC76F-u4Jt3PGDs+M@public.gmane.org>
	<7FA9CDA6-EAB3-4C1D-9CD3-8D4B06B0F335@freeswitch.org>
Message-ID: <b87dm5dn1qg1pkk8480064qjvc9j62jvj1@4ax.com>

On Sun, 31 Jan 2010 22:36:25 -0600, Brian West
<brian at freeswitch.org> wrote:
>Going back no_media after hold isn't supported yet.. Anthony said he would add it if someone really really wanted it and posted a bounty of $500 to cover his time to implement it.

Thanks all for the info. The bottom line seems to be that it's really
not a good idea to remove Freeswitch from the media path, and it's a
better idea to look elsewhere if scalability is an issue.



From mailinglist at fribert.dk  Mon Feb  1 00:57:45 2010
From: mailinglist at fribert.dk (mailinglist)
Date: Mon, 01 Feb 2010 09:57:45 +0100
Subject: [Freeswitch-users] Svar: Re: Somebody help me understand the		
	'features' set up please :-)
In-Reply-To: <4B669ED3020000E100000447@mail.fribert.dk>
References: <4B66226C020000E10000043C@mail.fribert.dk>
	<b39273531001311645u618ea87am8a3d84ee6b1bcfcb@mail.gmail.com>
	<4B669ED3020000E100000447@mail.fribert.dk>
Message-ID: <4B66A599020000E10000044C@mail.fribert.dk>

Sorry, I think I'm being unclear here.
 
I should add 
    <extension name="att_xfer">
       <condition field="destination_number" expression="^att_xfer$">
           <action application="read" data="3 4 sounds/getdigits.wav attxfer_callthis 30000 #"/>
           <action application="att_xfer" data="sofia/default/${attxfer_callthis}"/>
       </condition>
    </extension>
 
to the features.xml
 
Then add 
 
<action application="bind_meta_app" data="3 a a execute_extension::att_xfer XML features"/>
 
To the handling of the incoming call, right?
 
Something like
 
<extension name="PrivateIn">
   <condition field="destination_number" expression="^(8202)$">
       <action application="set" data="call_timeout=45"/>
       <action application="set" data="domain_name=10.11.12.25"/>
       <action application="set" data="continue_on_fail=true"/>
       <action application="set" data="hangup_after_bridge=true"/>
       <action application="set" data="transfer_ringback=$${hold_music}"/>
       <action application="bind_meta_app" data="3 a a execute_extension::att_xfer XML features"/>
       <action application="bridge" data="group/private@${domain_name}"/ ( mailto:group/private@$%7Bdomain_name%7D%22/ )>
       <action application="answer" data=""/>
       <action application="sleep" data="1000"/>
       <action application="voicemail" data="default 10.11.12.25 45"/>
   </condition>
</extension>
 
or?
 
 
 
Best regards
Fribse
 
/Running Freeswitch on pfSense running on a VMWare ESXi 4.0

>>> 01-02-2010 kl. 09:28 skrev "mailinglist" <mailinglist at fribert.dk> i meddelelsen <4B669ED3020000E100000447 at mail.fribert.dk>:

Hi Rupa
 
Ahh, I see.
Looking at example 2, and http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bind_meta_app
I still need to get where I enter this.
Is it in the call handling in the dialplan?
So I've gotten my current entry to handle incoming call from the outside:
 
<extension name="PrivateIn">
   <condition field="destination_number" expression="^(8202)$">
       <action application="set" data="call_timeout=45"/>
       <action application="set" data="domain_name=10.11.12.25"/>
       <action application="set" data="continue_on_fail=true"/>
       <action application="set" data="hangup_after_bridge=true"/>
       <action application="set" data="transfer_ringback=$${hold_music}"/>
       <action application="bridge" data="group/private@${domain_name}"/ ( 'mailto:group/private@$domain_name"/' )>
       <action application="answer" data=""/>
       <action application="sleep" data="1000"/>
       <action application="voicemail" data="default 10.11.12.25 45"/>
   </condition>
</extension>
Do I stick in there, or do I enter it in features.xml, or?
 
 
Best regards
Fribse
 
/Running Freeswitch on pfSense running on a VMWare ESXi 4.0


>>> 01-02-2010 kl. 01:45 skrev Rupa Schomaker <rupa at rupa.com> i meddelelsen <b39273531001311645u618ea87am8a3d84ee6b1bcfcb at mail.gmail.com>:

Look at bind_meta_app in the default dialplan.  It binds the dtmf to
the features context.

On Sun, Jan 31, 2010 at 5:38 PM, mailinglist <mailinglist at fribert.dk> wrote:
> Ok, I've gotten the Freeswitch to register to my VoIP provider.
> I've gotten my phones to register to Freeswitch, and I can receive and make
> calls, all very nice.
>
> I've gotten a Sipura SPA901 and a Siemens Gigaset S68IP registered to the
> Freeswitch.
>
> When I receive a call, I would like to be able to transfer the call to
> another phone, or change the call to a conference call with two local
> phones.
>
> So I've been looking at the examples in the wiki, and I can't make them
> work, not as I understand them anyways. Especially the att_xfer seems to be
> able to do what I need.
>
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_att_xfer
>
> As I understand Example1, I should answer the call, and then press *3 during
> the call, and either transfer it or change it to a threeway call.
>
> I get the first part, create an extension in the dialplan called att_xfer.
> But what is meant by the second par 'then bind this feature to DTMF 3', how
> do I enter that, and where?
>
> I hope somebody can help me with this (again)?
>
>
>
> Best regards
> Fribse
>
> /Running Freeswitch on pfSense running on a VMWare ESXi 4.0
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
-Rupa

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100201/58efb0ae/attachment-0001.html 

From jason at jasonjgw.net  Mon Feb  1 02:43:37 2010
From: jason at jasonjgw.net (Jason White)
Date: Mon, 1 Feb 2010 21:43:37 +1100
Subject: [Freeswitch-users] Equivalent to Asterisk's"directrtpsetup=yes"?
In-Reply-To: <b87dm5dn1qg1pkk8480064qjvc9j62jvj1@4ax.com>
References: <20100129164359.9B3C22B10D@cuneorg-email.cune.pri>
	<j4e8m5tqfl3fo8ug7d3sdgdiu7gofidbj8@4ax.com>
	<j4e8m5tqfl3fo8ug7d3sdgdiu7gofidbj8-e09XROE/p8c@public.gmane.org>
	<4468a6771001310247n4bf5d1a4rf4aea9f2d6c35a1f@mail.gmail.com>
	<qiqbm55lueuqup1m6r6d0mff36b4l4fb9l@4ax.com>
	<15D48404014D48D19F85CFFFC4BBC76F@cune.pri>
	<15D48404014D48D19F85CFFFC4BBC76F-u4Jt3PGDs+M@public.gmane.org>
	<7FA9CDA6-EAB3-4C1D-9CD3-8D4B06B0F335@freeswitch.org>
	<b87dm5dn1qg1pkk8480064qjvc9j62jvj1@4ax.com>
Message-ID: <20100201104337.GA985@jdc.jasonjgw.net>

Fred-145 <codecomplete at free.fr> wrote:
> On Sun, 31 Jan 2010 22:36:25 -0600, Brian West
> <brian at freeswitch.org> wrote:
> >Going back no_media after hold isn't supported yet.. Anthony said he would add it if someone really really wanted it and posted a bounty of $500 to cover his time to implement it.
> 
> Thanks all for the info. The bottom line seems to be that it's really
> not a good idea to remove Freeswitch from the media path, and it's a
> better idea to look elsewhere if scalability is an issue.

I don't think that's a good summary. What contributors to this thread are
saying is that there are trade-offs: you can keep FreeSWITCH in the media
path, with its consequent load on the server, or you can configure it for
bypass media or proxy media modes to reduce the load. In the latter case, nat
traversal issues may have to be dealt with, if relevant to your situation, and
if anyone wants to return to bypass media after hold, they're welcome to pay
the bounty to have this functionality implemented.




From lakindia89 at gmail.com  Mon Feb  1 02:57:25 2010
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Mon, 1 Feb 2010 16:27:25 +0530
Subject: [Freeswitch-users] nixevent behavior
In-Reply-To: <191c3a031001300848h65d65c9cg9b355cd07e922@mail.gmail.com>
References: <7d79b3931001300648j6aa55258yfc496d9cea5c4b8b@mail.gmail.com>
	<191c3a031001300848h65d65c9cg9b355cd07e922@mail.gmail.com>
Message-ID: <7d79b3931002010257hc84a997k26bdc172f27315bf@mail.gmail.com>

Thanks antony.. That work's well. I also understood the functionality of
send and sendRecv. Thanks..



On Sat, Jan 30, 2010 at 10:18 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> use $e = $con->sendRecv("command");
>
> every time
> for each send you do you must do a recv so this does both.
>
>
> On Sat, Jan 30, 2010 at 8:48 AM, lakshmanan ganapathy <
> lakindia89 at gmail.com> wrote:
>
>> Dear all
>>
>> I've done the following sample script to experiment the nixevent. I found
>> some difference in behavior because of nixevent. Let me explain my question
>> down the script.
>>
>> require ESL;
>> use IO::Socket::INET;
>>
>> my $ip = "192.168.1.222";
>> my $sock = new IO::Socket::INET ( LocalHost => $ip,  LocalPort => '8447',
>> Proto => 'tcp',  Listen => 1,  Reuse => 1 );
>> die "Could not create socket: $!\n" unless $sock;
>> for(;;) {
>>     my $new_sock = $sock->accept();
>>     next if (not defined ($new_sock));
>>     my $pid = fork();
>>     if ($pid) {
>>         close($new_sock);
>>         next;
>>     }
>>     print "CHILD PID: $$\n";
>>     my $host = $new_sock->sockhost();
>>     my $fd = fileno($new_sock);
>>
>>     my $con = new ESL::ESLconnection($fd);
>>     my $info = $con->getInfo();
>>
>>     my $uuid = $info->getHeader("unique-id");
>>
>>     printf "Connected call %s, from %s\n", $uuid,
>> $info->getHeader("caller-caller-id-number");
>>     my $r=$con->execute("answer");
>>     $con->events("plain","all");
>> ##########################
>>     $con->send("nixevent DTMF");
>>     my $val=$con->api("create_uuid");
>>     $val = $val->getBody();            # LINE 1
>>     chomp($val);
>>     print "UUID is $val\n";
>>     my $e = $con->recvEvent();
>>     $val = $e->getBody();               # LINE 2
>>     chomp($val);
>>     print "UUID is $val\n";
>>     close($new_sock);
>> }
>>
>> # If the line ($con->send("nixevent DTMF");) is commented, then the result
>> of create_uuid is obtained in LINE 1.
>> # else, the result isn't obtained in the LINE 1 and it has nothing. The
>> result is obtained only when I do a recvEvent,
>> # followed by a getBody (LINE 2)
>>
>> Just want to know why the behavior differs when nixevent is present???
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100201/13ce2f90/attachment.html 

From rupa at rupa.com  Mon Feb  1 03:03:51 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Mon, 1 Feb 2010 05:03:51 -0600
Subject: [Freeswitch-users] Svar: Re: Somebody help me understand the
	'features' set up please :-)
In-Reply-To: <4B669ED3020000E100000447@mail.fribert.dk>
References: <4B66226C020000E10000043C@mail.fribert.dk>
	<b39273531001311645u618ea87am8a3d84ee6b1bcfcb@mail.gmail.com>
	<4B669ED3020000E100000447@mail.fribert.dk>
Message-ID: <b39273531002010303h6e66baf7tf77be8d67449122b@mail.gmail.com>

Look at line 758 in the default dialplan.  It shows how it is used.

On Mon, Feb 1, 2010 at 2:28 AM, mailinglist <mailinglist at fribert.dk> wrote:

>  Hi Rupa
>
> Ahh, I see.
> Looking at example 2, and
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bind_meta_app
> I still need to get where I enter this.
> Is it in the call handling in the dialplan?
> So I've gotten my current entry to handle incoming call from the outside:
>
> <extension name="PrivateIn">
>    <condition field="destination_number" expression="^(8202)$">
>        <action application="set" data="call_timeout=45"/>
>        <action application="set" data="domain_name=10.11.12.25"/>
>        <action application="set" data="continue_on_fail=true"/>
>        <action application="set" data="hangup_after_bridge=true"/>
>        <action application="set" data="transfer_ringback=$${hold_music}"/>
>        <action application="bridge" data="group/private@${domain_name}"/<group/private@$%7Bdomain_name%7D%22/>
> >
>        <action application="answer" data=""/>
>        <action application="sleep" data="1000"/>
>        <action application="voicemail" data="default 10.11.12.25 45"/>
>    </condition>
> </extension>
> Do I stick in there, or do I enter it in features.xml, or?
>
>
>  Best regards
> Fribse
>
> /Running Freeswitch on pfSense running on a VMWare ESXi 4.0
>
>
> >>> 01-02-2010 kl. 01:45 skrev Rupa Schomaker <rupa at rupa.com> i
> meddelelsen <b39273531001311645u618ea87am8a3d84ee6b1bcfcb at mail.gmail.com>:
>   Look at bind_meta_app in the default dialplan.  It binds the dtmf to
> the features context.
>
> On Sun, Jan 31, 2010 at 5:38 PM, mailinglist <mailinglist at fribert.dk>
> wrote:
> > Ok, I've gotten the Freeswitch to register to my VoIP provider.
> > I've gotten my phones to register to Freeswitch, and I can receive and
> make
> > calls, all very nice.
> >
> > I've gotten a Sipura SPA901 and a Siemens Gigaset S68IP registered to the
> > Freeswitch.
> >
> > When I receive a call, I would like to be able to transfer the call to
> > another phone, or change the call to a conference call with two local
> > phones.
> >
> > So I've been looking at the examples in the wiki, and I can't make them
> > work, not as I understand them anyways. Especially the att_xfer seems to
> be
> > able to do what I need.
> >
> > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_att_xfer
> >
> > As I understand Example1, I should answer the call, and then press *3
> during
> > the call, and either transfer it or change it to a threeway call.
> >
> > I get the first part, create an extension in the dialplan called
> att_xfer.
> > But what is meant by the second par 'then bind this feature to DTMF 3',
> how
> > do I enter that, and where?
> >
> > I hope somebody can help me with this (again)?
> >
> >
> >
> > Best regards
> > Fribse
> >
> > /Running Freeswitch on pfSense running on a VMWare ESXi 4.0
>
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> -Rupa
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100201/6030a10f/attachment.html 

From moizchinoy at gmail.com  Mon Feb  1 03:08:29 2010
From: moizchinoy at gmail.com (Moiz Chinoy)
Date: Mon, 1 Feb 2010 15:08:29 +0400
Subject: [Freeswitch-users] Dialogic Board (JCT & DM3) Supported?
Message-ID: <29b888f81002010308uf875b53h838453b6e77e9a9e@mail.gmail.com>

Dear All,

Can anyone please advise that whether Dialogic boards (JCT and DM3)
are supported by FS.

-- 
Regards,
Moiz Chinoy.


From david.villasmil.work at gmail.com  Mon Feb  1 03:41:14 2010
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Mon, 1 Feb 2010 12:41:14 +0100
Subject: [Freeswitch-users] error loading module 'luasql.mysql'
In-Reply-To: <191c3a031001310621s20264d79u27afbf8ff0ba1a64@mail.gmail.com>
References: <eb7797d71001271615n2b86c197wc943d533fc0ffbe1@mail.gmail.com>
	<9853f4ff1001271709w60445c0ar7d3f4cb5fee36d4@mail.gmail.com>
	<eb7797d71001272047l48614934v3a35567f9fd0aa25@mail.gmail.com>
	<536A30FA-3494-40C2-9B9D-D8F63CA6BCC6@jerris.com>
	<4B657FDC.5080109@puzzled.xs4all.nl>
	<191c3a031001310621s20264d79u27afbf8ff0ba1a64@mail.gmail.com>
Message-ID: <9853f4ff1002010341w1aad5980h2cef90b541c21770@mail.gmail.com>

Hello Anthony,

Is this in planning?

David

On Sun, Jan 31, 2010 at 3:21 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> Be careful with lua and sql
> I have heard countless reports of the luasql leaking memory like a fire
> hydrant.....
>
> We may need to make our own odbc obj so every embedded lang can share it.
> But it takes time and resources.
>
> On Jan 31, 2010 7:11 AM, "Patrick" <freeswitch-list at puzzled.xs4all.nl>
> wrote:
>
> To fix a similar error message this is what I had in an old spec file:
> /sbin/restorecon -v /usr/lib64/somelib.so
>
> Iirc this is not the proper way to fix this and one should use the chcon
> command (chcon -t ...) or create an selinux policy. man chcon and google
> has more info.
>
> Regards,
> Patrick
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at l...
>
> On 01/31/2010 06:58 AM, Michael Jerris wrote:
>> http://www.google.com/search?q=cannot+restore+segmen...
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From sos at sokhapkin.dyndns.org  Mon Feb  1 03:58:21 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Mon, 1 Feb 2010 06:58:21 -0500
Subject: [Freeswitch-users] Equivalent to Asterisk's"directrtpsetup=yes"?
In-Reply-To: <b87dm5dn1qg1pkk8480064qjvc9j62jvj1@4ax.com>
References: <20100129164359.9B3C22B10D@cuneorg-email.cune.pri>
	<7FA9CDA6-EAB3-4C1D-9CD3-8D4B06B0F335@freeswitch.org>
	<b87dm5dn1qg1pkk8480064qjvc9j62jvj1@4ax.com>
Message-ID: <201002010658.21644.sos@sokhapkin.dyndns.org>

I'd say, you need bypass_media if provide SIP wholesale kind of service, and 
stay in audio path if you provide PBX-like kind of service.

On Monday 01 February 2010, Fred-145 wrote:
> On Sun, 31 Jan 2010 22:36:25 -0600, Brian West
>
> <brian at freeswitch.org> wrote:
> >Going back no_media after hold isn't supported yet.. Anthony said he would
> > add it if someone really really wanted it and posted a bounty of $500 to
> > cover his time to implement it.
>
> Thanks all for the info. The bottom line seems to be that it's really
> not a good idea to remove Freeswitch from the media path, and it's a
> better idea to look elsewhere if scalability is an issue.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From codecomplete at free.fr  Mon Feb  1 04:07:03 2010
From: codecomplete at free.fr (Fred-145)
Date: Mon, 01 Feb 2010 13:07:03 +0100
Subject: [Freeswitch-users] Equivalent to Asterisk's"directrtpsetup=yes"?
References: <20100129164359.9B3C22B10D@cuneorg-email.cune.pri>
	<7FA9CDA6-EAB3-4C1D-9CD3-8D4B06B0F335@freeswitch.org>
	<b87dm5dn1qg1pkk8480064qjvc9j62jvj1@4ax.com>
	<b87dm5dn1qg1pkk8480064qjvc9j62jvj1-e09XROE/p8c@public.gmane.org>
	<201002010658.21644.sos@sokhapkin.dyndns.org>
Message-ID: <ntgdm5lpgjal4psaq7dg5p9f6udqqsleg4@4ax.com>

On Mon, 1 Feb 2010 06:58:21 -0500, Sergey Okhapkin
<sos at sokhapkin.dyndns.org> wrote:
>I'd say, you need bypass_media if provide SIP wholesale kind of service, and 
>stay in audio path if you provide PBX-like kind of service.

Yup, that's a better summary. Thanks all for the information. Things
make a lot more sense now :-)



From stevendt at primrosebank.net  Mon Feb  1 05:27:13 2010
From: stevendt at primrosebank.net (Dave Stevenson)
Date: Mon, 1 Feb 2010 13:27:13 -0000
Subject: [Freeswitch-users] Trunk Version Number
References: <2360B060EDF44D368707DD12064BD416@bp1.ad.bp.com>
	<8976E2C4-CF8C-433A-AC79-B58F65105D08@jerris.com>
Message-ID: <12301A128C654024832FC6AA8CE43F31@bp1.ad.bp.com>

Hi Michael,

thanks for the reply. Yes, I have built from an SVN checkout (using Tortoise SVN).

I did not quite understand what is required to fix things though ?

regards
Dave
  ----- Original Message ----- 
  From: Michael Jerris 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Monday, February 01, 2010 6:11 AM
  Subject: Re: [Freeswitch-users] Trunk Version Number


  it should.  This can happen if you build from an svn checkout and the svn client your using is newer than our static linked svnversion.exe.  If anyone can make me a newer stripped down version like that I would appreciate it I have not had the time.


  On Jan 31, 2010, at 9:30 AM, Dave Stevenson wrote:


    Hi,

    Running the latest SVN (16453) under Windows, the console "Version" command displays :-

    "FreeSWITCH Version 1.0.trunk (UNKNOWN)"

    Should the version number not include a meaningful build version in the brackets ?

    regards
    Dave
    _______________________________________________
    FreeSWITCH-users mailing list
    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    http://www.freeswitch.org





------------------------------------------------------------------------------


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100201/02c440d3/attachment.html 

From mailinglist at fribert.dk  Mon Feb  1 05:37:40 2010
From: mailinglist at fribert.dk (mailinglist)
Date: Mon, 01 Feb 2010 14:37:40 +0100
Subject: [Freeswitch-users] Svar: Re: Somebody help me understand	 the
	'features' set up please :-)
In-Reply-To: <b39273531002010303h6e66baf7tf77be8d67449122b@mail.gmail.com>
References: <4B66226C020000E10000043C@mail.fribert.dk>
	<b39273531001311645u618ea87am8a3d84ee6b1bcfcb@mail.gmail.com>
	<4B669ED3020000E100000447@mail.fribert.dk>
	<b39273531002010303h6e66baf7tf77be8d67449122b@mail.gmail.com>
Message-ID: <4B66E734020000E100000451@mail.fribert.dk>

Hmm, I've just downloaded the default.xml under conf/dialplan from the SVN just to be on the safe side.
Line 758 is the last </include>, but I did find some examples on line 249-251.
 
So I've changed my dialplan entry handling calls from the outside to this:

<extension name="PrivateIn">
   <condition field="destination_number" expression="^(8202)$">
       <action application="bind_meta_app" data="1 b s execute_extension::att_xfer XML features"/>
       <action application="bind_meta_app" data="2 b s record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
       <action application="set" data="call_timeout=45"/>
       <action application="set" data="domain_name=10.11.12.25"/>
       <action application="set" data="continue_on_fail=true"/>
       <action application="set" data="hangup_after_bridge=true"/>
       <action application="set" data="transfer_ringback=$${hold_music}"/>
       <action application="bridge" data="group/private@${domain_name}"/ ( 'mailto:group/private@$domain_name"/' )>
       <action application="answer" data=""/>
       <action application="sleep" data="1000"/>
       <action application="voicemail" data="default 10.11.12.25 45"/>
   </condition>
</extension>
As I understand the bind_meta_app it listens for *1 and then it runs the att_xfer, *2 to record the call.
I've included the att_xfer in the XML features.
 
Question is, will it work at all when I bridge to a group?
 
Nothing happens when I press *1 and an extension.
 
 
 
 
Best regards
Fribse
 
/Running Freeswitch on pfSense running on a VMWare ESXi 4.0

>>> 01-02-2010 kl. 12:03 skrev Rupa Schomaker <rupa at rupa.com> i meddelelsen <b39273531002010303h6e66baf7tf77be8d67449122b at mail.gmail.com>:

Look at line 758 in the default dialplan. It shows how it is used.

On Mon, Feb 1, 2010 at 2:28 AM, mailinglist <mailinglist at fribert.dk> wrote:


Hi Rupa
Ahh, I see.
Looking at example 2, and http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bind_meta_app
I still need to get where I enter this.
Is it in the call handling in the dialplan?
So I've gotten my current entry to handle incoming call from the outside:
<extension name="PrivateIn">
<condition field="destination_number" expression="^(8202)$">
<action application="set" data="call_timeout=45"/>
<action application="set" data="domain_name=10.11.12.25"/>
<action application="set" data="continue_on_fail=true"/>
<action application="set" data="hangup_after_bridge=true"/>
<action application="set" data="transfer_ringback=$${hold_music}"/>
<action application="bridge" data="group/private@${domain_name}"/ ( mailto:group/private@$%7Bdomain_name%7D%22/ )>
<action application="answer" data=""/>
<action application="sleep" data="1000"/>
<action application="voicemail" data="default 10.11.12.25 45"/>
</condition>
</extension>
Do I stick in there, or do I enter it in features.xml, or?
Best regards
Fribse
/Running Freeswitch on pfSense running on a VMWare ESXi 4.0


>>> 01-02-2010 kl. 01:45 skrev Rupa Schomaker <rupa at rupa.com> i meddelelsen <b39273531001311645u618ea87am8a3d84ee6b1bcfcb at mail.gmail.com>:

Look at bind_meta_app in the default dialplan. It binds the dtmf to
the features context.

On Sun, Jan 31, 2010 at 5:38 PM, mailinglist <mailinglist at fribert.dk> wrote:
> Ok, I've gotten the Freeswitch to register to my VoIP provider.
> I've gotten my phones to register to Freeswitch, and I can receive and make
> calls, all very nice.
>
> I've gotten a Sipura SPA901 and a Siemens Gigaset S68IP registered to the
> Freeswitch.
>
> When I receive a call, I would like to be able to transfer the call to
> another phone, or change the call to a conference call with two local
> phones.
>
> So I've been looking at the examples in the wiki, and I can't make them
> work, not as I understand them anyways. Especially the att_xfer seems to be
> able to do what I need.
>
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_att_xfer
>
> As I understand Example1, I should answer the call, and then press *3 during
> the call, and either transfer it or change it to a threeway call.
>
> I get the first part, create an extension in the dialplan called att_xfer.
> But what is meant by the second par 'then bind this feature to DTMF 3', how
> do I enter that, and where?
>
> I hope somebody can help me with this (again)?
>
>
>
> Best regards
> Fribse
>
> /Running Freeswitch on pfSense running on a VMWare ESXi 4.0
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
-Rupa

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org





-- 
-Rupa
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100201/d8c1bc6a/attachment-0001.html 

From frank at carmickle.com  Mon Feb  1 07:29:16 2010
From: frank at carmickle.com (Frank Carmickle)
Date: Mon, 1 Feb 2010 10:29:16 -0500
Subject: [Freeswitch-users] latency growing with conference
Message-ID: <20100201152916.GD27405@base.carmickle.com>

Hello

I am noticing that when short phrases are spoken with cepstral in a conference it pauses all the audio before speaking it.  It seems as though the pauses are of different lengths and over time this adds up to a great amount of latency for the conference members to each other.  Rejoining the conference fixes this.  I already have 

      <param name="member-flags" value="waste"/>

I also notice that one member, who has lots of dropped packets, becomes very latent also.  I have

        <action application="set" data="rtp_autoflush=true"/>

Any thing else I can do about the two similar yet different issues?  I am on 16431 at the moment.

Thanks
--FC



From brian at freeswitch.org  Mon Feb  1 07:30:32 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 1 Feb 2010 09:30:32 -0600
Subject: [Freeswitch-users] Wrong RTP port submitted?
In-Reply-To: <4B6011EF.6090706@gmx.net>
References: <4B6011EF.6090706@gmx.net>
Message-ID: <2EBC2DDA-1799-44B5-9D1B-EE6EC1618482@freeswitch.org>

You have proxy media on don't you? From what this looks like you have an inbound invite from a Snom and we have an outbound invite.  If you're doing something such as proxy media it will try to pass it thru as is...  Giving you little or NO control over the port.  Unless you fix your endpoint to also use the same restrictive port range.  Because by default we will not use anything in the 48000 range .. the giveaway here is the fact I see P-Key-Flags: header which is indication you're using a snom with proxy media.  Am I correct?

/b

On Jan 27, 2010, at 4:14 AM, Peter P GMX wrote:

> I have defined the rtp port range for 12000-12100 in switch.conf.xml.
> However Freeswitch is offering a port 48320 in the invite message. The
> result is, that the incoming RTP stream is blocked by the firewall (I
> can see a reject for UDP 48320).
> Any hint how to solve this?
> 
> Best regards Peter
> 
> See config and invite message:
>    <!--RTP port range -->
>    <param name="rtp-start-port" value="12000"/>-->
>    <param name="rtp-end-port" value="12100"/>-->
> 
> Invite:
> INVITE sip:027xxxxxxxx at sip.itsp.de SIP/2.0.
> Via: SIP/2.0/UDP 217.24.xx.xxx:5080;rport;branch=z9hG4bKjD923NvctXaFm.
> Max-Forwards: 69.
> From: "0608xxxxxxx"
> <sip:4918xxxxxxxxxxxx at sip.itsp.de;transport=udp>;tag=0Kp4tvU44UmXp.
> To: <sip:027xxxxxxxx at sip.itsp.de>.
> Call-ID: 30c86b94-85ca-122d-f88e-080027e51f59.
> CSeq: 126174137 INVITE.
> Contact: <sip:gw+tgnet3 at 217.24.xx.xxx:5080;transport=udp;gw=tgnet3>.
> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16032.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY.
> Supported: timer, precondition, path, replaces.
> Allow-Events: talk, refer.
> Content-Type: application/sdp.
> Content-Disposition: session.
> Content-Length: 320.
> P-Key-Flags: keys="3".
> X-FS-Support: update_display.
> Remote-Party-ID: "0608xxxxxxx"
> <sip:0608xxxxxxx at sip.itsp.de>;party=calling;screen=yes;privacy=off.
> .
> v=0.
> o=FreeSWITCH 1264536651 1264536652 IN IP4 217.24.xx.xxx.
> s=FreeSWITCH.
> c=IN IP4 217.24.xx.xxx.
> t=0 0.
> m=audio 48320 RTP/AVP 8 0 98 3 101 13.
> a=rtpmap:8 PCMA/8000.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:98 SPEEX/8000.
> a=rtpmap:3 GSM/8000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-16.
> a=rtpmap:13 CN/8000.
> a=ptime:20.
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From pjintheusa at gmail.com  Mon Feb  1 07:37:52 2010
From: pjintheusa at gmail.com (Phillip Jones)
Date: Mon, 1 Feb 2010 10:37:52 -0500
Subject: [Freeswitch-users] Establishing a Call from .Net based
	application
In-Reply-To: <df97b9c51001310545n54195cadr9faa3db46917ca9e@mail.gmail.com>
References: <874941.17255.qm@web33502.mail.mud.yahoo.com>
	<df97b9c51001310545n54195cadr9faa3db46917ca9e@mail.gmail.com>
Message-ID: <367751821002010737o4ca83476j62c96cb60c8a456a@mail.gmail.com>

As mentioned http://wiki.freeswitch.org/wiki/Mod_managed should give you
every thing you need to get mod_managed set up.

Then in the source take a look at demo.csx and particularity AppDemo class.

That should get you started.


On Sun, Jan 31, 2010 at 8:45 AM, Scott Fernandez <scottferri09 at gmail.com>wrote:

> Hi,
>
> Thx for the information. Can I have some detailed steps to configure
> mod_managed class call control and how do we write the API commands in .Net
> applications?
>
> In addition, how do we get the current STATE of the call when I use
> webapi?. Because it is required for me to route the call to the user upon it
> is answered or disconnect  it.
>
> Thanks,
> Scott
>
> On Wed, Jan 20, 2010 at 8:47 PM, Diego Toro <dftoro at yahoo.com> wrote:
>
>> Hi, the answer is yes, you can to use mod_managed wich offer C# managed
>> class to call control http://wiki.freeswitch.org/wiki/Mod_managed. Or
>> using managed ESL (libs/esl/managed) which offer C# managed class to receive
>> and send events and commands to FreeSwitch.
>>
>> Diego Toro
>> http://lacarretade.blogspot.com/
>>
>>
>> --- On Wed, 1/20/10, Scott Fernandez <scottferri09 at gmail.com> wrote:
>>
>> > From: Scott Fernandez <scottferri09 at gmail.com>
>> > Subject: Re: [Freeswitch-users] Establishing a Call from .Net based
>> application
>> > To: freeswitch-users at lists.freeswitch.org
>> > Date: Wednesday, January 20, 2010, 2:17 AM
>>  > Thanks Dome. Will try it out and get back to
>> > you if I come across any issues.
>> >
>> > Regards,
>> > Scott.
>> >
>> > On Wed, Jan 20, 2010 at 11:02 AM,
>> > Dome Charoenyost <dome at tel.co.th>
>> > wrote:
>> >
>> > Please try http://wiki.freeswitch.org/wiki/Webapi
>> >
>> >
>> > you can create class and map to webapi.
>> >
>> >
>> >
>> > Dome C.
>> >
>> >
>> >
>> > 2010/1/19 Scott Fernandez <scottferri09 at gmail.com>:
>> >
>> > > Hi,
>> >
>> > >
>> >
>> > > Is there any API modules available for me to initiate
>> > a call from .Net based
>> >
>> > > application?.
>> >
>> > >
>> >
>> > > The idea is to include the API modules if any with the
>> > .NET base classes so
>> >
>> > > that the API commands will be made available on it. I
>> > know it is doable when
>> >
>> > > I use socket programming in .NET in which Telnet
>> > session is created.
>> >
>> > > However, this would potentially hamper the performance
>> > of the application
>> >
>> > > because of multiple sessions that will be created for
>> > each call.
>> >
>> > >
>> >
>> > > Other than that, Is there any Freeswitch API modules
>> > (like plug-ins)
>> >
>> > > available in order to include it into the .Net classes
>> > and start building
>> >
>> > > the customized application?
>> >
>> > >
>> >
>> > > Any help from any one is highly appreciated.
>> >
>> > >
>> >
>> > > Thanks,
>> >
>> > > Scott
>> >
>> > >
>> >
>> > >
>> > _______________________________________________
>> >
>> > > FreeSWITCH-users mailing list
>> >
>> > > FreeSWITCH-users at lists.freeswitch.org
>> >
>> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >
>> > > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >
>> > > http://www.freeswitch.org
>> >
>> > >
>> >
>> > >
>> >
>> >
>> >
>> > _______________________________________________
>> >
>> > FreeSWITCH-users mailing list
>> >
>> > FreeSWITCH-users at lists.freeswitch.org
>> >
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >
>> > http://www.freeswitch.org
>> >
>> >
>> >
>> >
>> > -----Inline Attachment Follows-----
>>  >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>>
>>
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100201/fba1c5e5/attachment.html 

From frank at carmickle.com  Mon Feb  1 07:44:22 2010
From: frank at carmickle.com (Frank Carmickle)
Date: Mon, 1 Feb 2010 10:44:22 -0500
Subject: [Freeswitch-users] conference and xmpp
Message-ID: <20100201154421.GE27405@base.carmickle.com>

Hello

I am trying to interact with conferences over xmpp.  I want to be able to see status changes and mute/unmute etc...  I can only list the members if I ask for a list.  I am having trouble understanding the wiki entree on this topic.  Profile names are confusing because all the examples show default as a name so it is hard to see what each profile name corresponds to.  My config looks like this currently.

   <profile name="ourchatperms">
    <!-- both of these users have a functionally equivalent capability set -->
    <user name="shawn at djernes.net" commands="all"/>
    <!-- individually specified commands must be book-end'ed with the | charachter -->
    <user name="frankiec at djernes.net" commands="|deaf|dial|energy|kick|list|lock|mute|norecord|play|record|relate|say|saymember|stop|transfer|undeaf|unlock|unmute|volume_in|volume_out|"/>
   </profile>
  </chat-permissions>
 
    <profile name="cdquality">
<param name="caller-controls" value="custom-ewr"/>
      <param name="chat-permissions" value="ourchatperms"/>
      <param name="domain" value="$${domain}"/>

Also we can only talk to the conference by conf+confnum-fqdn at fqdn.  Is there a way to make it conf+confnum at fqdn?

Any help would be greatly appreciated.  Thank you.
--FC


From mouncifbb at gmail.com  Mon Feb  1 07:47:22 2010
From: mouncifbb at gmail.com (Mouncif Benniane)
Date: Mon, 1 Feb 2010 10:47:22 -0500
Subject: [Freeswitch-users] mod_lcr problem
In-Reply-To: <b39273531001311507hb1bfe02m8d370443b882ef81@mail.gmail.com>
References: <ade741d61001291443l6134a3achf28be64fc7746c8e@mail.gmail.com>
	<b39273531001300702r1a81f094pac0bd20aeb5ceed6@mail.gmail.com>
	<ade741d61001301338x8014e86t9c9094403f3361f@mail.gmail.com>
	<b39273531001301559h60668746kd5c63ada240513f1@mail.gmail.com>
	<ade741d61001301623w70d7e449i44a9f1e9fb7beae4@mail.gmail.com>
	<b39273531001301645w10446308o3578479dcbf2da2@mail.gmail.com>
	<ade741d61001302057w6ce72355iee6e8b4a2872fc81@mail.gmail.com>
	<b39273531001310532y1de53609rf474755cec04f218@mail.gmail.com>
	<9AB8B620-5A69-4F08-B62C-FFEB03FC6762@gmail.com>
	<b39273531001311507hb1bfe02m8d370443b882ef81@mail.gmail.com>
Message-ID: <ade741d61002010747v6da3d2e1x554b46aa26c27b18@mail.gmail.com>

any example on how to use: set_profile_var?

thanks


On Sun, Jan 31, 2010 at 6:07 PM, Rupa Schomaker <rupa at rupa.com> wrote:

> Yes, you need to normalize the values passed to lcr.  Otherwise, how could
> it work?
>
> You can normalize the CID by matching and adding a 1 for 10 digit #s, or
> removing the leading + or other things you might need then setting it back
> to the profile using the set_profile_var app (
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_set_profile_var).
>  (mod_cidlookup will set it after doing a #->name/area lookup - but for now
> you can set it yourself)
>
> You can normalize the DID by doing similar matching rules as above and then
> transfering to that normalized DID for the rest of your call plan
> processing.
>
> I'm pretty sure mod_cidlookup has an example of normalizing... yeah:
> http://wiki.freeswitch.org/wiki/Mod_cidlookup#Dialplan_Application
>
> On Sun, Jan 31, 2010 at 4:18 PM, Mouncifbb <mouncifbb at gmail.com> wrote:
>
>> So the CID must have 1 at front also? Usually people
>> Send only npa and nxx ex 6176427788 7817612233
>> Do I need to alter it?
>>
>> Sent from my iPhone
>>
>> On Jan 31, 2010, at 8:32 AM, Rupa Schomaker <rupa at rupa.com> wrote:
>>
>>
>>
>> On Sat, Jan 30, 2010 at 10:57 PM, Mouncif Benniane <<mouncifbb at gmail.com>
>> mouncifbb at gmail.com> wrote:
>>
>>> OK going back to use default profile to keep things simple below 2
>>> results
>>>
>>> Using:
>>>
>>> lcr 16179470890  default  19785223241  ( this one consult
>>> npa_nxx_company_ocn)
>>>
>>> lcr 6179470890  default  9785223241 ( this one don't!! )
>>>
>>>
>>>
>> Oh, right!  mod_lcr really expects you to normalize your prefix to e164
>> format.  I thought there was discussion about this in the wiki, but maybe
>> not.  For simple prefix matching it doesn't matter, but for things that make
>> decisions based on the # (like the lata/state stuff) it does.
>>
>> npanxx lookup only makes sense for NANPA numbers.  NANPA numbers have a
>> country code of "1" and a total length of 11 (including the 1).
>>
>> This is the only rational way to do it when you have a rate table with
>> both domestic (NANPA) and international prefixes.
>>
>>
>>> freeswitch> lcr 16179470890  default  19785223241
>>> 2010-01-30 23:53:34.681842 [DEBUG] mod_lcr.c:1329 data passed to lcr is
>>> [16179470890  default  19785223241]
>>> 2010-01-30 23:53:34.681842 [DEBUG] mod_lcr.c:1365 Set Caller ID to
>>> [19785223241]
>>> 2010-01-30 23:53:34.681842 [DEBUG] mod_lcr.c:736 SQL: SELECT 'state',
>>> count(DISTINCT state) FROM npa_nxx_company_ocn WHERE (npa=617 AND nxx=947)
>>> OR (npa=978 AND nxx=522) UNION SELECT 'lata', count(DISTINCT lata) FROM
>>> npa_nxx_company_ocn WHERE (npa=617 AND nxx=947) OR (npa=978 AND nxx=522)
>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:696 Type: state, Count: 1
>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:696 Type: lata, Count: 1
>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:786 intra routing [state:1
>>> lata:1] so rate field is [intralata_rate]
>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:802 we have an event
>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:826 SQL: SELECT l.digits,
>>> c.carrier_name, l.intralata_rate, cg.prefix AS gw_prefix, cg.suffix AS
>>> gw_suffix, l.lead_strip, l.trail_strip, l.prefix, l.suffix , cg.codec ,
>>> l.cid FROM lcr l JOIN carriers c ON l.carrier_id=c.id JOIN
>>> carrier_gateway cg ON c.id=cg.carrier_id WHERE c.enabled = '1' AND
>>> cg.enabled = '1' AND l.enabled = '1' AND digits IN (16179470890, 1617947089,
>>> 161794708, 16179470, 1617947, 161794, 16179, 1617, 161, 16, 1) AND
>>> CURRENT_TIMESTAMP BETWEEN date_start AND date_end ORDER BY digits DESC,
>>> intralata_rate, rand();
>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:307 Returning Dialstring
>>> [lcr_carrier=carrier1,lcr_rate=0.00000]sofia/gateway/carrier1/16179470890
>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:615 Adding carrier1 to head
>>> of list
>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:307 Returning Dialstring
>>> [lcr_carrier=carrier2,lcr_rate=0.00000]sofia/external/
>>> 06179470890 at proxy.carrier2.net:5060
>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:667 adding carrier2 to end
>>> of list after carrier1
>>>
>>>
>>>  | Digit Match | Carrier  | Rate     | Codec | CID Regexp |
>>> Dialstring
>>> |
>>>  | 1           | carrier1 | 0.00000  |       |            |
>>> [lcr_carrier=carrier1,lcr_rate=0.00000]sofia/gateway/carrier1/16179470890
>>> |
>>>  | 1           | carrier2 | 0.00000  |       |            |
>>> [lcr_carrier=carrier2,lcr_rate=0.00000]sofia/external/
>>> 06179470890 at proxy.carrier2.net:5060 |
>>>
>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:307 Returning Dialstring
>>> [lcr_carrier=carrier1,lcr_rate=0.00000]sofia/gateway/carrier1/16179470890
>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:307 Returning Dialstring
>>> [lcr_carrier=carrier2,lcr_rate=0.00000]sofia/external/
>>> 06179470890 at proxy.carrier2.net:5060
>>>
>>>
>>>
>>>
>>>
>>> freeswitch> lcr 6179470890  default  9785223241
>>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:1329 data passed to lcr is
>>> [6179470890  default  9785223241]
>>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:1365 Set Caller ID to
>>> [9785223241]
>>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:786 intra routing [state:0
>>> lata:0] so rate field is [rate]
>>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:802 we have an event
>>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:826 SQL: SELECT l.digits,
>>> c.carrier_name, l.rate, cg.prefix AS gw_prefix, cg.suffix AS gw_suffix,
>>> l.lead_strip, l.trail_strip, l.prefix, l.suffix , cg.codec , l.cid FROM lcr
>>> l JOIN carriers c ON l.carrier_id=c.id JOIN carrier_gateway cg ON c.id=cg.carrier_id
>>> WHERE c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1' AND digits IN
>>> (6179470890, 617947089, 61794708, 6179470, 617947, 61794, 6179, 617, 61, 6)
>>> AND CURRENT_TIMESTAMP BETWEEN date_start AND date_end ORDER BY digits DESC,
>>> rate, rand();
>>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:307 Returning Dialstring
>>> [lcr_carrier=carrier1,lcr_rate=0.09000]sofia/gateway/carrier1/16179470890
>>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:615 Adding carrier1 to head
>>> of list
>>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:307 Returning Dialstring
>>> [lcr_carrier=carrier1,lcr_rate=0.09000]sofia/gateway/carrier1/16179470890
>>>
>>>
>>>  | Digit Match | Carrier  | Rate     | Codec | CID Regexp |
>>> Dialstring                                                                |
>>>   | 617947      | carrier1 | 0.09000  |       |            |
>>> [lcr_carrier=carrier1,lcr_rate=0.09000]sofia/gateway/carrier1/16179470890 |
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>> On Sat, Jan 30, 2010 at 7:45 PM, Rupa Schomaker < <rupa at rupa.com>
>>> rupa at rupa.com> wrote:
>>>
>>>> turn up logging to debug again, and then reload mod_lcr.  It'll spit out
>>>> a bunch of crap when it tests out each profile you have defined.  Give me
>>>> the full log (here or in <http://pastebin.freeswitch.org>
>>>> pastebin.freeswitch.org).  That may show more useful info as to why
>>>> things are mucked up?
>>>>
>>>>
>>>> On Sat, Jan 30, 2010 at 6:23 PM, Mouncif Benniane <<mouncifbb at gmail.com>
>>>> mouncifbb at gmail.com> wrote:
>>>>
>>>>> yes I use NANPA_STD profile instead of default cause I thought the
>>>>> custom profile was causing issues, but looks like it's returning same
>>>>> results.
>>>>>
>>>>> There is this line in thw wiki:
>>>>> intra lata/state selection is done manually by setting the channel
>>>>> variables *intrastate* or *intralata* to the value *true*.
>>>>>
>>>>> do I have to set these ? if yes how?
>>>>>
>>>>> Thanks
>>>>>
>>>>>
>>>>> On Sat, Jan 30, 2010 at 6:59 PM, Rupa Schomaker < <rupa at rupa.com>
>>>>> rupa at rupa.com> wrote:
>>>>>
>>>>>> Stuff inline.
>>>>>>
>>>>>> On Sat, Jan 30, 2010 at 3:38 PM, Mouncif Benniane <<mouncifbb at gmail.com>
>>>>>> mouncifbb at gmail.com> wrote:
>>>>>>
>>>>>>>  NPANXX,"LATA","OCN","NTER","INTRA"
>>>>>>> 201007,"224","7229","0.0059","0.0127"
>>>>>>> 201040,"224","9206","0.0036","0.0036"
>>>>>>>
>>>>>>
>>>>>> Looks like they give you the LATA and OCN values with the prefix.  We
>>>>>> (should) look that up ourselves.
>>>>>>
>>>>>>
>>>>>>> FreeSWITCH Version 1.0.trunk (16540)
>>>>>>>
>>>>>>>
>>>>>>> Also I noticed the *npa_nxx_ocn* table never get consulted.
>>>>>>>
>>>>>>> I also see this now when making a real call instead of running
>>>>>>> thorugh CLI
>>>>>>>
>>>>>>> EXECUTE sofia/external/6179472456 at 174.x.x.x lcr(16179470890
>>>>>>> NANPA_STD)
>>>>>>> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:1230 *intrastate
>>>>>>> channel var is [undef]*
>>>>>>
>>>>>>
>>>>>> This is fine.  it is a leftover from when you would tell mod_lcr via a
>>>>>> channel var that it should do intrastate.  I later had mod_lcr do the lookup
>>>>>> itself, but we still honor the old var.  There are no channel vars
>>>>>> associated with the cli, so you wouldn't see that msg.
>>>>>>
>>>>>>
>>>>>>>
>>>>>>> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:1233 Select routes based
>>>>>>> on interstate rates
>>>>>>> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:1252 LCR Lookup on
>>>>>>> 16179470893 using profile NANPA_STD
>>>>>>> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:786 intra routing
>>>>>>> [state:0 lata:0] so rate field is [rate]
>>>>>>>
>>>>>>> called number 6179470890  caller ID: 6179472456
>>>>>>>
>>>>>>> any ideas??
>>>>>>>
>>>>>>>
>>>>>> Only thing that jumps out at me.
>>>>>>
>>>>>> The output from lcr_admin show profiles showed only the default one.
>>>>>>  On the dialplan you use the NANPA_STD profile.  Can you check lcr_admin
>>>>>> list and see if that profile is defined and if so if it says it is using the
>>>>>> npanxx table?
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> On Sat, Jan 30, 2010 at 10:02 AM, Rupa Schomaker < <rupa at rupa.com>
>>>>>>> rupa at rupa.com> wrote:
>>>>>>>
>>>>>>>> Something is still missing from the logs. Note the query of the
>>>>>>>> npanxx
>>>>>>>> table, the flags being set, and the rate field being chosen.  Umm..
>>>>>>>> oh, what version of fs are you running?
>>>>>>>>
>>>>>>>> Yes, the npa_nxx_ocn table needs to be loaded up as described in:
>>>>>>>> <http://wiki.freeswitch.org/wiki/Mod_cidlookup#Falling_back_to_.22City_State.22_in_the_absense_of_a_name>
>>>>>>>> http://wiki.freeswitch.org/wiki/Mod_cidlookup#Falling_back_to_.22City_State.22_in_the_absense_of_a_name
>>>>>>>> (there is a link to that from mod_lcr's wiki page).
>>>>>>>>
>>>>>>>> An example from my own setup:
>>>>>>>>
>>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:1384 data passed to lcr
>>>>>>>> is [12148267711 default 12148267712]
>>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:1420 Set Caller ID to
>>>>>>>> [12148267712]
>>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:759 SQL: SELECT
>>>>>>>> 'state',
>>>>>>>> count(DISTINCT state) FROM npa_nxx_company_ocn WHERE (npa=214 AND
>>>>>>>> nxx=826) OR (npa=214 AND nxx=826) UNION SELECT 'lata',
>>>>>>>> count(DISTINCT
>>>>>>>> lata) FROM npa_nxx_company_ocn WHERE (npa=214 AND nxx=826) OR
>>>>>>>> (npa=214
>>>>>>>> AND nxx=826)
>>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:719 Type: lata, Count:
>>>>>>>> 1
>>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:719 Type: state, Count:
>>>>>>>> 1
>>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:809 intra routing
>>>>>>>> [state:1 lata:1] so rate field is [intralata_rate]
>>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:825 we have an event
>>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:849 SQL:  SELECT
>>>>>>>> l.digits
>>>>>>>> AS lcr_digits,         c.carrier_name AS lcr_carrier_name,
>>>>>>>> l.intralata_rate as lcr_rate_field,        cg.prefix AS
>>>>>>>> lcr_gw_prefix,
>>>>>>>> cg.suffix AS lcr_gw_suffix,        l.lead_strip AS lcr_lead_strip,
>>>>>>>> l.trail_strip AS lcr_trail_strip,        l.prefix AS lcr_prefix,
>>>>>>>> l.suffix AS lcr_suffix,        cg.codec AS lcr_codec, l.cid AS
>>>>>>>> lcr_cid,        'carriers' AS lcr_limit_realm, c.carrier_name AS
>>>>>>>> lcr_limit_id, 5 AS lcr_limit_max   FROM lcr l      JOIN carriers c
>>>>>>>> ON
>>>>>>>> l.carrier_id=c.id      JOIN carrier_gateway cg ON c.id
>>>>>>>> =cg.carrier_id
>>>>>>>>  WHERE c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1'
>>>>>>>> AND digits_prefix @> '12148267711'      AND CURRENT_TIMESTAMP
>>>>>>>> BETWEEN
>>>>>>>> date_start AND date_end   ORDER BY digits DESC, intralata_rate,
>>>>>>>> random();
>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning
>>>>>>>> Dialstring
>>>>>>>>
>>>>>>>> [lcr_carrier=grnvoip_us,lcr_rate=0.00591]sofia/gateway/grnvoip/XXXX12148267711
>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:638 Adding grnvoip_us
>>>>>>>> to
>>>>>>>> head of list
>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning
>>>>>>>> Dialstring
>>>>>>>>
>>>>>>>> [lcr_carrier=teliax,lcr_rate=0.01000]sofia/gateway/teliax_atl/12148267711
>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:660 Adding teliax to
>>>>>>>> end of list
>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning
>>>>>>>> Dialstring
>>>>>>>>
>>>>>>>> [lcr_carrier=teliax,lcr_rate=0.01000]sofia/gateway/teliax/12148267711
>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:660 Adding teliax to
>>>>>>>> end of list
>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning
>>>>>>>> Dialstring
>>>>>>>>
>>>>>>>> [lcr_carrier=vitelity,lcr_rate=0.01440]sofia/gateway/vitelity/12148267711
>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:660 Adding vitelity to
>>>>>>>> end of list
>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning
>>>>>>>> Dialstring
>>>>>>>>
>>>>>>>> [lcr_carrier=grnvoip_std,lcr_rate=0.01500]sofia/gateway/grnvoip/YYYY12148267711
>>>>>>>> [...]
>>>>>>>>
>>>>>>>> On Fri, Jan 29, 2010 at 10:42 PM, Mouncif Benniane <<mouncifbb at gmail.com>
>>>>>>>> mouncifbb at gmail.com> wrote:
>>>>>>>> > Also the Provider has presented the rates in this format?
>>>>>>>> > NPANXXLATA OCN INTER INTRA
>>>>>>>> >
>>>>>>>> >
>>>>>>>> > On Fri, Jan 29, 2010 at 11:30 PM, Mouncif Benniane <<mouncifbb at gmail.com>
>>>>>>>> mouncifbb at gmail.com>
>>>>>>>> > wrote:
>>>>>>>> >>
>>>>>>>> >> Tried it and it's not giving me intralata  instead I get
>>>>>>>> interstate, does
>>>>>>>> >> the npa_nxx_company_ocn table needs to be used in this case?,
>>>>>>>> also do I have
>>>>>>>> >> to have the rate field in lcr table?
>>>>>>>> >>
>>>>>>>> >> lcr 617642  default 6176421212
>>>>>>>> >>  | Digit Match | Carrier  | Rate     | Codec | CID Regexp |
>>>>>>>> Dialstring
>>>>>>>> >>                                                        |
>>>>>>>> >>  | 617642      | carrier1 | 0.00500  |       |            |
>>>>>>>> >>
>>>>>>>> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642 |
>>>>>>>> >>
>>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:1329 data passed to
>>>>>>>> lcr is
>>>>>>>> >> [617642  default 6176421212]
>>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:1365 Set Caller ID
>>>>>>>> to
>>>>>>>> >> [6176421212]
>>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:786 intra routing
>>>>>>>> [state:0
>>>>>>>> >> lata:0] so rate field is [rate]
>>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:802 we have an event
>>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:826 SQL: SELECT
>>>>>>>> l.digits,
>>>>>>>> >> c.carrier_name, l.rate, cg.prefix AS gw_prefix, cg.suffix AS
>>>>>>>> gw_suffix,
>>>>>>>> >> l.lead_strip, l.trail_strip, l.prefix, l.suffix , cg.codec ,
>>>>>>>> l.cid FROM lcr
>>>>>>>> >> l JOIN carriers c ON l.carrier_id=c.id JOIN carrier_gateway cg
>>>>>>>> ON
>>>>>>>> >> c.id=cg.carrier_id WHERE c.enabled = '1' AND cg.enabled = '1'
>>>>>>>> AND l.enabled
>>>>>>>> >> = '1' AND digits IN (617642, 61764, 6176, 617, 61, 6) AND
>>>>>>>> CURRENT_TIMESTAMP
>>>>>>>> >> BETWEEN date_start AND date_end ORDER BY digits DESC,  rate,
>>>>>>>> rand();
>>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:307 Returning
>>>>>>>> Dialstring
>>>>>>>> >>
>>>>>>>> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642
>>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:615 Adding carrier1
>>>>>>>> to head
>>>>>>>> >> of list
>>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:307 Returning
>>>>>>>> Dialstring
>>>>>>>> >>
>>>>>>>> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642
>>>>>>>> >>
>>>>>>>> >> Thank you Rupa!
>>>>>>>> >>
>>>>>>>> >> On Fri, Jan 29, 2010 at 7:37 PM, Rupa Schomaker <<rupa at rupa.com>
>>>>>>>> rupa at rupa.com> wrote:
>>>>>>>> >>>
>>>>>>>> >>> turn console logging up to debug and redo the lcr lookup.  The
>>>>>>>> sql
>>>>>>>> >>> statements along with status info will show up.  This should
>>>>>>>> give
>>>>>>>> >>> enough information to debug what is happening.
>>>>>>>> >>>
>>>>>>>> >>> I'm assuming the npanxx table is actually populated and not just
>>>>>>>> >>> existing?
>>>>>>>> >>>
>>>>>>>> >>> When doing the lookup from the cli you have to tell lcr what CID
>>>>>>>> to
>>>>>>>> >>> use (remember, it is relative to the src/dest number).  I'm
>>>>>>>> pretty
>>>>>>>> >>> sure you get something on the console log when you don't specify
>>>>>>>> a CID
>>>>>>>> >>> when using the commandline.  Anyway:
>>>>>>>> >>>
>>>>>>>> >>> lcr 617642  default 6176421212
>>>>>>>> >>>
>>>>>>>> >>> should give you intralata.
>>>>>>>> >>>
>>>>>>>> >>> Note that the definition of intralata doesn't mean "local" for
>>>>>>>> some
>>>>>>>> >>> providers.  Some providers define local to "same ratecenter"
>>>>>>>> which is
>>>>>>>> >>> even more restrictive.
>>>>>>>> >>>
>>>>>>>> >>> On Fri, Jan 29, 2010 at 4:43 PM, Mouncif Benniane <<mouncifbb at gmail.com>
>>>>>>>> mouncifbb at gmail.com>
>>>>>>>> >>> wrote:
>>>>>>>> >>> > i can't make use of mod_lcr using Intra/Interstate rating, I
>>>>>>>> am using
>>>>>>>> >>> > svn: FreeSWITCH Version 1.0.trunk (16517)
>>>>>>>> >>> >
>>>>>>>> >>> > lcr mysql table structure:
>>>>>>>> >>> > CREATE TABLE `lcr` (
>>>>>>>> >>> >   `id` INT(11) NOT NULL AUTO_INCREMENT,
>>>>>>>> >>> >   `digits` VARCHAR(15) DEFAULT NULL,
>>>>>>>> >>> >   `rate` FLOAT(11,5) DEFAULT NULL,
>>>>>>>> >>> >   `intrastate_rate` FLOAT(11,5) DEFAULT NULL,
>>>>>>>> >>> >   `intralata_rate` FLOAT(11,5) DEFAULT NULL,
>>>>>>>> >>> >   `carrier_id` INT(11) NOT NULL,
>>>>>>>> >>> >   `lead_strip` INT(11) NOT NULL,
>>>>>>>> >>> >   `trail_strip` INT(11) NOT NULL,
>>>>>>>> >>> >   `prefix` VARCHAR(16) NOT NULL,
>>>>>>>> >>> >   `suffix` VARCHAR(16) NOT NULL,
>>>>>>>> >>> >   `lcr_profile` VARCHAR(32) DEFAULT NULL,
>>>>>>>> >>> >   `date_start` DATETIME NOT NULL DEFAULT '1970-01-01
>>>>>>>> 00:00:00',
>>>>>>>> >>> >   `date_end` DATETIME NOT NULL DEFAULT '2030-12-31 00:00:00',
>>>>>>>> >>> >   `quality` FLOAT(10,6) NOT NULL,
>>>>>>>> >>> >   `reliability` FLOAT(10,6) NOT NULL,
>>>>>>>> >>> >   `cid` VARCHAR(32) NOT NULL DEFAULT '',
>>>>>>>> >>> >   `enabled` TINYINT(1) NOT NULL DEFAULT '1',
>>>>>>>> >>> >   PRIMARY KEY  (`id`),
>>>>>>>> >>> >   KEY `carrier_id` (`carrier_id`),
>>>>>>>> >>> >   KEY `digits` (`digits`),
>>>>>>>> >>> >   KEY `lcr_profile` (`lcr_profile`),
>>>>>>>> >>> >   KEY `digits_profile_cid_rate` USING BTREE (`digits`),
>>>>>>>> >>> >   CONSTRAINT `carrier_id` FOREIGN KEY (`carrier_id`)
>>>>>>>> REFERENCES
>>>>>>>> >>> > `carriers`
>>>>>>>> >>> > (`id`) ON DELETE CASCADE ON UPDATE CASCADE
>>>>>>>> >>> > ) ENGINE=INNODB AUTO_INCREMENT=6 DEFAULT CHARSET=latin1
>>>>>>>> >>> >
>>>>>>>> >>> >
>>>>>>>> >>> > lcr_admin show profiles
>>>>>>>> >>> > Name:           default
>>>>>>>> >>> >  custom sql:    SELECT l.digits, c.carrier_name,
>>>>>>>> l.${lcr_rate_field},
>>>>>>>> >>> > cg.prefix AS gw_prefix, cg.suffix AS gw_suffix, l.lead_strip,
>>>>>>>> >>> > l.trail_strip,
>>>>>>>> >>> > l.prefix, l.suffix , cg.codec , l.cid FROM lcr l JOIN carriers
>>>>>>>> c ON
>>>>>>>> >>> > l.carrier_id=c.id JOIN carrier_gateway cg ON c.id=cg.carrier_id
>>>>>>>> WHERE
>>>>>>>> >>> > c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1' AND
>>>>>>>> digits IN
>>>>>>>> >>> > (${lcr_query_expanded_digits}) AND CURRENT_TIMESTAMP BETWEEN
>>>>>>>> date_start
>>>>>>>> >>> > AND
>>>>>>>> >>> > date_end ORDER BY digits DESC,  ${lcr_rate_field},  quality
>>>>>>>> DESC,
>>>>>>>> >>> >  reliability DESC, rand();
>>>>>>>> >>> >  has %:         false
>>>>>>>> >>> >  has vars:      true
>>>>>>>> >>> >  has intrastate:        true
>>>>>>>> >>> >  has intralata: true
>>>>>>>> >>> >  has npanxx:    true
>>>>>>>> >>> >  Reorder rate:  enabled
>>>>>>>> >>> >  Info in headers:       disabled
>>>>>>>> >>> >  Quote IN() List:       disabled
>>>>>>>> >>> >
>>>>>>>> >>> >
>>>>>>>> >>> >
>>>>>>>> >>> > lcr 617642  default  returns rate from the rate field table
>>>>>>>> and not
>>>>>>>> >>> > intra/inter state fields rates.
>>>>>>>> >>> >
>>>>>>>> >>> > Any ideas? thanks!
>>>>>>>> >>> >
>>>>>>>> >>> >
>>>>>>>> >>> >
>>>>>>>> >>> >
>>>>>>>> >>> > _______________________________________________
>>>>>>>> >>> > FreeSWITCH-users mailing list
>>>>>>>> >>> > <FreeSWITCH-users at lists.freeswitch.org>
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> >>> >
>>>>>>>> <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> >>> >
>>>>>>>> >>> > UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> >>> > <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>>> >>> >
>>>>>>>> >>> >
>>>>>>>> >>>
>>>>>>>> >>>
>>>>>>>> >>>
>>>>>>>> >>> --
>>>>>>>> >>> -Rupa
>>>>>>>> >>>
>>>>>>>> >>> _______________________________________________
>>>>>>>> >>> FreeSWITCH-users mailing list
>>>>>>>> >>> <FreeSWITCH-users at lists.freeswitch.org>
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> >>> <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> >>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> >>> <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>>> >>
>>>>>>>> >
>>>>>>>> >
>>>>>>>> > _______________________________________________
>>>>>>>> > FreeSWITCH-users mailing list
>>>>>>>> > <FreeSWITCH-users at lists.freeswitch.org>
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> > <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> > UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> > <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>>> >
>>>>>>>> >
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> --
>>>>>>>> -Rupa
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>> -Rupa
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>>
>>>> --
>>>> -Rupa
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>> FreeSWITCH-users at lists.freeswitch.org
>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> -Rupa
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100201/342b0520/attachment-0001.html 

From mouncifbb at gmail.com  Mon Feb  1 08:16:29 2010
From: mouncifbb at gmail.com (Mouncif Benniane)
Date: Mon, 1 Feb 2010 11:16:29 -0500
Subject: [Freeswitch-users] mod_lcr problem
In-Reply-To: <ade741d61002010747v6da3d2e1x554b46aa26c27b18@mail.gmail.com>
References: <ade741d61001291443l6134a3achf28be64fc7746c8e@mail.gmail.com>
	<ade741d61001301338x8014e86t9c9094403f3361f@mail.gmail.com>
	<b39273531001301559h60668746kd5c63ada240513f1@mail.gmail.com>
	<ade741d61001301623w70d7e449i44a9f1e9fb7beae4@mail.gmail.com>
	<b39273531001301645w10446308o3578479dcbf2da2@mail.gmail.com>
	<ade741d61001302057w6ce72355iee6e8b4a2872fc81@mail.gmail.com>
	<b39273531001310532y1de53609rf474755cec04f218@mail.gmail.com>
	<9AB8B620-5A69-4F08-B62C-FFEB03FC6762@gmail.com>
	<b39273531001311507hb1bfe02m8d370443b882ef81@mail.gmail.com>
	<ade741d61002010747v6da3d2e1x554b46aa26c27b18@mail.gmail.com>
Message-ID: <ade741d61002010816x4e42ab0eqbfdb00089883654a@mail.gmail.com>

I got it! nevermind.

session.execute("set_profile_var","caller_id_number=1617947XXXX"); ( since I
am using js)

Thanks

On Mon, Feb 1, 2010 at 10:47 AM, Mouncif Benniane <mouncifbb at gmail.com>wrote:

> any example on how to use: set_profile_var?
>
> thanks
>
>
>
> On Sun, Jan 31, 2010 at 6:07 PM, Rupa Schomaker <rupa at rupa.com> wrote:
>
>> Yes, you need to normalize the values passed to lcr.  Otherwise, how could
>> it work?
>>
>> You can normalize the CID by matching and adding a 1 for 10 digit #s, or
>> removing the leading + or other things you might need then setting it back
>> to the profile using the set_profile_var app (
>> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_set_profile_var).
>>  (mod_cidlookup will set it after doing a #->name/area lookup - but for now
>> you can set it yourself)
>>
>> You can normalize the DID by doing similar matching rules as above and
>> then transfering to that normalized DID for the rest of your call plan
>> processing.
>>
>> I'm pretty sure mod_cidlookup has an example of normalizing... yeah:
>> http://wiki.freeswitch.org/wiki/Mod_cidlookup#Dialplan_Application
>>
>> On Sun, Jan 31, 2010 at 4:18 PM, Mouncifbb <mouncifbb at gmail.com> wrote:
>>
>>> So the CID must have 1 at front also? Usually people
>>> Send only npa and nxx ex 6176427788 7817612233
>>> Do I need to alter it?
>>>
>>> Sent from my iPhone
>>>
>>> On Jan 31, 2010, at 8:32 AM, Rupa Schomaker <rupa at rupa.com> wrote:
>>>
>>>
>>>
>>> On Sat, Jan 30, 2010 at 10:57 PM, Mouncif Benniane <<mouncifbb at gmail.com>
>>> mouncifbb at gmail.com> wrote:
>>>
>>>> OK going back to use default profile to keep things simple below 2
>>>> results
>>>>
>>>> Using:
>>>>
>>>> lcr 16179470890  default  19785223241  ( this one consult
>>>> npa_nxx_company_ocn)
>>>>
>>>> lcr 6179470890  default  9785223241 ( this one don't!! )
>>>>
>>>>
>>>>
>>> Oh, right!  mod_lcr really expects you to normalize your prefix to e164
>>> format.  I thought there was discussion about this in the wiki, but maybe
>>> not.  For simple prefix matching it doesn't matter, but for things that make
>>> decisions based on the # (like the lata/state stuff) it does.
>>>
>>> npanxx lookup only makes sense for NANPA numbers.  NANPA numbers have a
>>> country code of "1" and a total length of 11 (including the 1).
>>>
>>> This is the only rational way to do it when you have a rate table with
>>> both domestic (NANPA) and international prefixes.
>>>
>>>
>>>> freeswitch> lcr 16179470890  default  19785223241
>>>> 2010-01-30 23:53:34.681842 [DEBUG] mod_lcr.c:1329 data passed to lcr is
>>>> [16179470890  default  19785223241]
>>>> 2010-01-30 23:53:34.681842 [DEBUG] mod_lcr.c:1365 Set Caller ID to
>>>> [19785223241]
>>>> 2010-01-30 23:53:34.681842 [DEBUG] mod_lcr.c:736 SQL: SELECT 'state',
>>>> count(DISTINCT state) FROM npa_nxx_company_ocn WHERE (npa=617 AND nxx=947)
>>>> OR (npa=978 AND nxx=522) UNION SELECT 'lata', count(DISTINCT lata) FROM
>>>> npa_nxx_company_ocn WHERE (npa=617 AND nxx=947) OR (npa=978 AND nxx=522)
>>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:696 Type: state, Count: 1
>>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:696 Type: lata, Count: 1
>>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:786 intra routing [state:1
>>>> lata:1] so rate field is [intralata_rate]
>>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:802 we have an event
>>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:826 SQL: SELECT l.digits,
>>>> c.carrier_name, l.intralata_rate, cg.prefix AS gw_prefix, cg.suffix AS
>>>> gw_suffix, l.lead_strip, l.trail_strip, l.prefix, l.suffix , cg.codec ,
>>>> l.cid FROM lcr l JOIN carriers c ON l.carrier_id=c.id JOIN
>>>> carrier_gateway cg ON c.id=cg.carrier_id WHERE c.enabled = '1' AND
>>>> cg.enabled = '1' AND l.enabled = '1' AND digits IN (16179470890, 1617947089,
>>>> 161794708, 16179470, 1617947, 161794, 16179, 1617, 161, 16, 1) AND
>>>> CURRENT_TIMESTAMP BETWEEN date_start AND date_end ORDER BY digits DESC,
>>>> intralata_rate, rand();
>>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:307 Returning Dialstring
>>>> [lcr_carrier=carrier1,lcr_rate=0.00000]sofia/gateway/carrier1/16179470890
>>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:615 Adding carrier1 to head
>>>> of list
>>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:307 Returning Dialstring
>>>> [lcr_carrier=carrier2,lcr_rate=0.00000]sofia/external/
>>>> 06179470890 at proxy.carrier2.net:5060
>>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:667 adding carrier2 to end
>>>> of list after carrier1
>>>>
>>>>
>>>>  | Digit Match | Carrier  | Rate     | Codec | CID Regexp |
>>>> Dialstring
>>>> |
>>>>  | 1           | carrier1 | 0.00000  |       |            |
>>>> [lcr_carrier=carrier1,lcr_rate=0.00000]sofia/gateway/carrier1/16179470890
>>>> |
>>>>  | 1           | carrier2 | 0.00000  |       |            |
>>>> [lcr_carrier=carrier2,lcr_rate=0.00000]sofia/external/
>>>> 06179470890 at proxy.carrier2.net:5060 |
>>>>
>>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:307 Returning Dialstring
>>>> [lcr_carrier=carrier1,lcr_rate=0.00000]sofia/gateway/carrier1/16179470890
>>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:307 Returning Dialstring
>>>> [lcr_carrier=carrier2,lcr_rate=0.00000]sofia/external/
>>>> 06179470890 at proxy.carrier2.net:5060
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> freeswitch> lcr 6179470890  default  9785223241
>>>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:1329 data passed to lcr is
>>>> [6179470890  default  9785223241]
>>>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:1365 Set Caller ID to
>>>> [9785223241]
>>>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:786 intra routing [state:0
>>>> lata:0] so rate field is [rate]
>>>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:802 we have an event
>>>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:826 SQL: SELECT l.digits,
>>>> c.carrier_name, l.rate, cg.prefix AS gw_prefix, cg.suffix AS gw_suffix,
>>>> l.lead_strip, l.trail_strip, l.prefix, l.suffix , cg.codec , l.cid FROM lcr
>>>> l JOIN carriers c ON l.carrier_id=c.id JOIN carrier_gateway cg ON c.id=cg.carrier_id
>>>> WHERE c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1' AND digits IN
>>>> (6179470890, 617947089, 61794708, 6179470, 617947, 61794, 6179, 617, 61, 6)
>>>> AND CURRENT_TIMESTAMP BETWEEN date_start AND date_end ORDER BY digits DESC,
>>>> rate, rand();
>>>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:307 Returning Dialstring
>>>> [lcr_carrier=carrier1,lcr_rate=0.09000]sofia/gateway/carrier1/16179470890
>>>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:615 Adding carrier1 to head
>>>> of list
>>>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:307 Returning Dialstring
>>>> [lcr_carrier=carrier1,lcr_rate=0.09000]sofia/gateway/carrier1/16179470890
>>>>
>>>>
>>>>  | Digit Match | Carrier  | Rate     | Codec | CID Regexp |
>>>> Dialstring                                                                |
>>>>   | 617947      | carrier1 | 0.09000  |       |            |
>>>> [lcr_carrier=carrier1,lcr_rate=0.09000]sofia/gateway/carrier1/16179470890 |
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> On Sat, Jan 30, 2010 at 7:45 PM, Rupa Schomaker < <rupa at rupa.com>
>>>> rupa at rupa.com> wrote:
>>>>
>>>>> turn up logging to debug again, and then reload mod_lcr.  It'll spit
>>>>> out a bunch of crap when it tests out each profile you have defined.  Give
>>>>> me the full log (here or in <http://pastebin.freeswitch.org>
>>>>> pastebin.freeswitch.org).  That may show more useful info as to why
>>>>> things are mucked up?
>>>>>
>>>>>
>>>>> On Sat, Jan 30, 2010 at 6:23 PM, Mouncif Benniane <<mouncifbb at gmail.com>
>>>>> mouncifbb at gmail.com> wrote:
>>>>>
>>>>>> yes I use NANPA_STD profile instead of default cause I thought the
>>>>>> custom profile was causing issues, but looks like it's returning same
>>>>>> results.
>>>>>>
>>>>>> There is this line in thw wiki:
>>>>>> intra lata/state selection is done manually by setting the channel
>>>>>> variables *intrastate* or *intralata* to the value *true*.
>>>>>>
>>>>>> do I have to set these ? if yes how?
>>>>>>
>>>>>> Thanks
>>>>>>
>>>>>>
>>>>>> On Sat, Jan 30, 2010 at 6:59 PM, Rupa Schomaker < <rupa at rupa.com>
>>>>>> rupa at rupa.com> wrote:
>>>>>>
>>>>>>> Stuff inline.
>>>>>>>
>>>>>>> On Sat, Jan 30, 2010 at 3:38 PM, Mouncif Benniane <<mouncifbb at gmail.com>
>>>>>>> mouncifbb at gmail.com> wrote:
>>>>>>>
>>>>>>>>  NPANXX,"LATA","OCN","NTER","INTRA"
>>>>>>>> 201007,"224","7229","0.0059","0.0127"
>>>>>>>> 201040,"224","9206","0.0036","0.0036"
>>>>>>>>
>>>>>>>
>>>>>>> Looks like they give you the LATA and OCN values with the prefix.  We
>>>>>>> (should) look that up ourselves.
>>>>>>>
>>>>>>>
>>>>>>>> FreeSWITCH Version 1.0.trunk (16540)
>>>>>>>>
>>>>>>>>
>>>>>>>> Also I noticed the *npa_nxx_ocn* table never get consulted.
>>>>>>>>
>>>>>>>> I also see this now when making a real call instead of running
>>>>>>>> thorugh CLI
>>>>>>>>
>>>>>>>> EXECUTE sofia/external/6179472456 at 174.x.x.x lcr(16179470890
>>>>>>>> NANPA_STD)
>>>>>>>> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:1230 *intrastate
>>>>>>>> channel var is [undef]*
>>>>>>>
>>>>>>>
>>>>>>> This is fine.  it is a leftover from when you would tell mod_lcr via
>>>>>>> a channel var that it should do intrastate.  I later had mod_lcr do the
>>>>>>> lookup itself, but we still honor the old var.  There are no channel vars
>>>>>>> associated with the cli, so you wouldn't see that msg.
>>>>>>>
>>>>>>>
>>>>>>>>
>>>>>>>> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:1233 Select routes
>>>>>>>> based on interstate rates
>>>>>>>> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:1252 LCR Lookup on
>>>>>>>> 16179470893 using profile NANPA_STD
>>>>>>>> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:786 intra routing
>>>>>>>> [state:0 lata:0] so rate field is [rate]
>>>>>>>>
>>>>>>>> called number 6179470890  caller ID: 6179472456
>>>>>>>>
>>>>>>>> any ideas??
>>>>>>>>
>>>>>>>>
>>>>>>> Only thing that jumps out at me.
>>>>>>>
>>>>>>> The output from lcr_admin show profiles showed only the default one.
>>>>>>>  On the dialplan you use the NANPA_STD profile.  Can you check lcr_admin
>>>>>>> list and see if that profile is defined and if so if it says it is using the
>>>>>>> npanxx table?
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> On Sat, Jan 30, 2010 at 10:02 AM, Rupa Schomaker < <rupa at rupa.com>
>>>>>>>> rupa at rupa.com> wrote:
>>>>>>>>
>>>>>>>>> Something is still missing from the logs. Note the query of the
>>>>>>>>> npanxx
>>>>>>>>> table, the flags being set, and the rate field being chosen.  Umm..
>>>>>>>>> oh, what version of fs are you running?
>>>>>>>>>
>>>>>>>>> Yes, the npa_nxx_ocn table needs to be loaded up as described in:
>>>>>>>>> <http://wiki.freeswitch.org/wiki/Mod_cidlookup#Falling_back_to_.22City_State.22_in_the_absense_of_a_name>
>>>>>>>>> http://wiki.freeswitch.org/wiki/Mod_cidlookup#Falling_back_to_.22City_State.22_in_the_absense_of_a_name
>>>>>>>>> (there is a link to that from mod_lcr's wiki page).
>>>>>>>>>
>>>>>>>>> An example from my own setup:
>>>>>>>>>
>>>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:1384 data passed to
>>>>>>>>> lcr
>>>>>>>>> is [12148267711 default 12148267712]
>>>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:1420 Set Caller ID to
>>>>>>>>> [12148267712]
>>>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:759 SQL: SELECT
>>>>>>>>> 'state',
>>>>>>>>> count(DISTINCT state) FROM npa_nxx_company_ocn WHERE (npa=214 AND
>>>>>>>>> nxx=826) OR (npa=214 AND nxx=826) UNION SELECT 'lata',
>>>>>>>>> count(DISTINCT
>>>>>>>>> lata) FROM npa_nxx_company_ocn WHERE (npa=214 AND nxx=826) OR
>>>>>>>>> (npa=214
>>>>>>>>> AND nxx=826)
>>>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:719 Type: lata, Count:
>>>>>>>>> 1
>>>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:719 Type: state,
>>>>>>>>> Count: 1
>>>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:809 intra routing
>>>>>>>>> [state:1 lata:1] so rate field is [intralata_rate]
>>>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:825 we have an event
>>>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:849 SQL:  SELECT
>>>>>>>>> l.digits
>>>>>>>>> AS lcr_digits,         c.carrier_name AS lcr_carrier_name,
>>>>>>>>> l.intralata_rate as lcr_rate_field,        cg.prefix AS
>>>>>>>>> lcr_gw_prefix,
>>>>>>>>> cg.suffix AS lcr_gw_suffix,        l.lead_strip AS lcr_lead_strip,
>>>>>>>>> l.trail_strip AS lcr_trail_strip,        l.prefix AS lcr_prefix,
>>>>>>>>> l.suffix AS lcr_suffix,        cg.codec AS lcr_codec, l.cid AS
>>>>>>>>> lcr_cid,        'carriers' AS lcr_limit_realm, c.carrier_name AS
>>>>>>>>> lcr_limit_id, 5 AS lcr_limit_max   FROM lcr l      JOIN carriers c
>>>>>>>>> ON
>>>>>>>>> l.carrier_id=c.id      JOIN carrier_gateway cg ON c.id
>>>>>>>>> =cg.carrier_id
>>>>>>>>>  WHERE c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1'
>>>>>>>>> AND digits_prefix @> '12148267711'      AND CURRENT_TIMESTAMP
>>>>>>>>> BETWEEN
>>>>>>>>> date_start AND date_end   ORDER BY digits DESC, intralata_rate,
>>>>>>>>> random();
>>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning
>>>>>>>>> Dialstring
>>>>>>>>>
>>>>>>>>> [lcr_carrier=grnvoip_us,lcr_rate=0.00591]sofia/gateway/grnvoip/XXXX12148267711
>>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:638 Adding grnvoip_us
>>>>>>>>> to
>>>>>>>>> head of list
>>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning
>>>>>>>>> Dialstring
>>>>>>>>>
>>>>>>>>> [lcr_carrier=teliax,lcr_rate=0.01000]sofia/gateway/teliax_atl/12148267711
>>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:660 Adding teliax to
>>>>>>>>> end of list
>>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning
>>>>>>>>> Dialstring
>>>>>>>>>
>>>>>>>>> [lcr_carrier=teliax,lcr_rate=0.01000]sofia/gateway/teliax/12148267711
>>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:660 Adding teliax to
>>>>>>>>> end of list
>>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning
>>>>>>>>> Dialstring
>>>>>>>>>
>>>>>>>>> [lcr_carrier=vitelity,lcr_rate=0.01440]sofia/gateway/vitelity/12148267711
>>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:660 Adding vitelity to
>>>>>>>>> end of list
>>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning
>>>>>>>>> Dialstring
>>>>>>>>>
>>>>>>>>> [lcr_carrier=grnvoip_std,lcr_rate=0.01500]sofia/gateway/grnvoip/YYYY12148267711
>>>>>>>>> [...]
>>>>>>>>>
>>>>>>>>> On Fri, Jan 29, 2010 at 10:42 PM, Mouncif Benniane <<mouncifbb at gmail.com>
>>>>>>>>> mouncifbb at gmail.com> wrote:
>>>>>>>>> > Also the Provider has presented the rates in this format?
>>>>>>>>> > NPANXXLATA OCN INTER INTRA
>>>>>>>>> >
>>>>>>>>> >
>>>>>>>>> > On Fri, Jan 29, 2010 at 11:30 PM, Mouncif Benniane <<mouncifbb at gmail.com>
>>>>>>>>> mouncifbb at gmail.com>
>>>>>>>>> > wrote:
>>>>>>>>> >>
>>>>>>>>> >> Tried it and it's not giving me intralata  instead I get
>>>>>>>>> interstate, does
>>>>>>>>> >> the npa_nxx_company_ocn table needs to be used in this case?,
>>>>>>>>> also do I have
>>>>>>>>> >> to have the rate field in lcr table?
>>>>>>>>> >>
>>>>>>>>> >> lcr 617642  default 6176421212
>>>>>>>>> >>  | Digit Match | Carrier  | Rate     | Codec | CID Regexp |
>>>>>>>>> Dialstring
>>>>>>>>> >>                                                        |
>>>>>>>>> >>  | 617642      | carrier1 | 0.00500  |       |            |
>>>>>>>>> >>
>>>>>>>>> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642 |
>>>>>>>>> >>
>>>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:1329 data passed to
>>>>>>>>> lcr is
>>>>>>>>> >> [617642  default 6176421212]
>>>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:1365 Set Caller ID
>>>>>>>>> to
>>>>>>>>> >> [6176421212]
>>>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:786 intra routing
>>>>>>>>> [state:0
>>>>>>>>> >> lata:0] so rate field is [rate]
>>>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:802 we have an
>>>>>>>>> event
>>>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:826 SQL: SELECT
>>>>>>>>> l.digits,
>>>>>>>>> >> c.carrier_name, l.rate, cg.prefix AS gw_prefix, cg.suffix AS
>>>>>>>>> gw_suffix,
>>>>>>>>> >> l.lead_strip, l.trail_strip, l.prefix, l.suffix , cg.codec ,
>>>>>>>>> l.cid FROM lcr
>>>>>>>>> >> l JOIN carriers c ON l.carrier_id=c.id JOIN carrier_gateway cg
>>>>>>>>> ON
>>>>>>>>> >> c.id=cg.carrier_id WHERE c.enabled = '1' AND cg.enabled = '1'
>>>>>>>>> AND l.enabled
>>>>>>>>> >> = '1' AND digits IN (617642, 61764, 6176, 617, 61, 6) AND
>>>>>>>>> CURRENT_TIMESTAMP
>>>>>>>>> >> BETWEEN date_start AND date_end ORDER BY digits DESC,  rate,
>>>>>>>>> rand();
>>>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:307 Returning
>>>>>>>>> Dialstring
>>>>>>>>> >>
>>>>>>>>> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642
>>>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:615 Adding carrier1
>>>>>>>>> to head
>>>>>>>>> >> of list
>>>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:307 Returning
>>>>>>>>> Dialstring
>>>>>>>>> >>
>>>>>>>>> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642
>>>>>>>>> >>
>>>>>>>>> >> Thank you Rupa!
>>>>>>>>> >>
>>>>>>>>> >> On Fri, Jan 29, 2010 at 7:37 PM, Rupa Schomaker <<rupa at rupa.com>
>>>>>>>>> rupa at rupa.com> wrote:
>>>>>>>>> >>>
>>>>>>>>> >>> turn console logging up to debug and redo the lcr lookup.  The
>>>>>>>>> sql
>>>>>>>>> >>> statements along with status info will show up.  This should
>>>>>>>>> give
>>>>>>>>> >>> enough information to debug what is happening.
>>>>>>>>> >>>
>>>>>>>>> >>> I'm assuming the npanxx table is actually populated and not
>>>>>>>>> just
>>>>>>>>> >>> existing?
>>>>>>>>> >>>
>>>>>>>>> >>> When doing the lookup from the cli you have to tell lcr what
>>>>>>>>> CID to
>>>>>>>>> >>> use (remember, it is relative to the src/dest number).  I'm
>>>>>>>>> pretty
>>>>>>>>> >>> sure you get something on the console log when you don't
>>>>>>>>> specify a CID
>>>>>>>>> >>> when using the commandline.  Anyway:
>>>>>>>>> >>>
>>>>>>>>> >>> lcr 617642  default 6176421212
>>>>>>>>> >>>
>>>>>>>>> >>> should give you intralata.
>>>>>>>>> >>>
>>>>>>>>> >>> Note that the definition of intralata doesn't mean "local" for
>>>>>>>>> some
>>>>>>>>> >>> providers.  Some providers define local to "same ratecenter"
>>>>>>>>> which is
>>>>>>>>> >>> even more restrictive.
>>>>>>>>> >>>
>>>>>>>>> >>> On Fri, Jan 29, 2010 at 4:43 PM, Mouncif Benniane <<mouncifbb at gmail.com>
>>>>>>>>> mouncifbb at gmail.com>
>>>>>>>>> >>> wrote:
>>>>>>>>> >>> > i can't make use of mod_lcr using Intra/Interstate rating, I
>>>>>>>>> am using
>>>>>>>>> >>> > svn: FreeSWITCH Version 1.0.trunk (16517)
>>>>>>>>> >>> >
>>>>>>>>> >>> > lcr mysql table structure:
>>>>>>>>> >>> > CREATE TABLE `lcr` (
>>>>>>>>> >>> >   `id` INT(11) NOT NULL AUTO_INCREMENT,
>>>>>>>>> >>> >   `digits` VARCHAR(15) DEFAULT NULL,
>>>>>>>>> >>> >   `rate` FLOAT(11,5) DEFAULT NULL,
>>>>>>>>> >>> >   `intrastate_rate` FLOAT(11,5) DEFAULT NULL,
>>>>>>>>> >>> >   `intralata_rate` FLOAT(11,5) DEFAULT NULL,
>>>>>>>>> >>> >   `carrier_id` INT(11) NOT NULL,
>>>>>>>>> >>> >   `lead_strip` INT(11) NOT NULL,
>>>>>>>>> >>> >   `trail_strip` INT(11) NOT NULL,
>>>>>>>>> >>> >   `prefix` VARCHAR(16) NOT NULL,
>>>>>>>>> >>> >   `suffix` VARCHAR(16) NOT NULL,
>>>>>>>>> >>> >   `lcr_profile` VARCHAR(32) DEFAULT NULL,
>>>>>>>>> >>> >   `date_start` DATETIME NOT NULL DEFAULT '1970-01-01
>>>>>>>>> 00:00:00',
>>>>>>>>> >>> >   `date_end` DATETIME NOT NULL DEFAULT '2030-12-31 00:00:00',
>>>>>>>>> >>> >   `quality` FLOAT(10,6) NOT NULL,
>>>>>>>>> >>> >   `reliability` FLOAT(10,6) NOT NULL,
>>>>>>>>> >>> >   `cid` VARCHAR(32) NOT NULL DEFAULT '',
>>>>>>>>> >>> >   `enabled` TINYINT(1) NOT NULL DEFAULT '1',
>>>>>>>>> >>> >   PRIMARY KEY  (`id`),
>>>>>>>>> >>> >   KEY `carrier_id` (`carrier_id`),
>>>>>>>>> >>> >   KEY `digits` (`digits`),
>>>>>>>>> >>> >   KEY `lcr_profile` (`lcr_profile`),
>>>>>>>>> >>> >   KEY `digits_profile_cid_rate` USING BTREE (`digits`),
>>>>>>>>> >>> >   CONSTRAINT `carrier_id` FOREIGN KEY (`carrier_id`)
>>>>>>>>> REFERENCES
>>>>>>>>> >>> > `carriers`
>>>>>>>>> >>> > (`id`) ON DELETE CASCADE ON UPDATE CASCADE
>>>>>>>>> >>> > ) ENGINE=INNODB AUTO_INCREMENT=6 DEFAULT CHARSET=latin1
>>>>>>>>> >>> >
>>>>>>>>> >>> >
>>>>>>>>> >>> > lcr_admin show profiles
>>>>>>>>> >>> > Name:           default
>>>>>>>>> >>> >  custom sql:    SELECT l.digits, c.carrier_name,
>>>>>>>>> l.${lcr_rate_field},
>>>>>>>>> >>> > cg.prefix AS gw_prefix, cg.suffix AS gw_suffix, l.lead_strip,
>>>>>>>>> >>> > l.trail_strip,
>>>>>>>>> >>> > l.prefix, l.suffix , cg.codec , l.cid FROM lcr l JOIN
>>>>>>>>> carriers c ON
>>>>>>>>> >>> > l.carrier_id=c.id JOIN carrier_gateway cg ON c.id=cg.carrier_id
>>>>>>>>> WHERE
>>>>>>>>> >>> > c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1' AND
>>>>>>>>> digits IN
>>>>>>>>> >>> > (${lcr_query_expanded_digits}) AND CURRENT_TIMESTAMP BETWEEN
>>>>>>>>> date_start
>>>>>>>>> >>> > AND
>>>>>>>>> >>> > date_end ORDER BY digits DESC,  ${lcr_rate_field},  quality
>>>>>>>>> DESC,
>>>>>>>>> >>> >  reliability DESC, rand();
>>>>>>>>> >>> >  has %:         false
>>>>>>>>> >>> >  has vars:      true
>>>>>>>>> >>> >  has intrastate:        true
>>>>>>>>> >>> >  has intralata: true
>>>>>>>>> >>> >  has npanxx:    true
>>>>>>>>> >>> >  Reorder rate:  enabled
>>>>>>>>> >>> >  Info in headers:       disabled
>>>>>>>>> >>> >  Quote IN() List:       disabled
>>>>>>>>> >>> >
>>>>>>>>> >>> >
>>>>>>>>> >>> >
>>>>>>>>> >>> > lcr 617642  default  returns rate from the rate field table
>>>>>>>>> and not
>>>>>>>>> >>> > intra/inter state fields rates.
>>>>>>>>> >>> >
>>>>>>>>> >>> > Any ideas? thanks!
>>>>>>>>> >>> >
>>>>>>>>> >>> >
>>>>>>>>> >>> >
>>>>>>>>> >>> >
>>>>>>>>> >>> > _______________________________________________
>>>>>>>>> >>> > FreeSWITCH-users mailing list
>>>>>>>>> >>> > <FreeSWITCH-users at lists.freeswitch.org>
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> >>> >
>>>>>>>>> <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> >>> >
>>>>>>>>> >>> > UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> >>> > <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>>>> >>> >
>>>>>>>>> >>> >
>>>>>>>>> >>>
>>>>>>>>> >>>
>>>>>>>>> >>>
>>>>>>>>> >>> --
>>>>>>>>> >>> -Rupa
>>>>>>>>> >>>
>>>>>>>>> >>> _______________________________________________
>>>>>>>>> >>> FreeSWITCH-users mailing list
>>>>>>>>> >>> <FreeSWITCH-users at lists.freeswitch.org>
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> >>>
>>>>>>>>> <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> >>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> >>> <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>>>> >>
>>>>>>>>> >
>>>>>>>>> >
>>>>>>>>> > _______________________________________________
>>>>>>>>> > FreeSWITCH-users mailing list
>>>>>>>>> > <FreeSWITCH-users at lists.freeswitch.org>
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> > <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> > UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> > <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>>>> >
>>>>>>>>> >
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> --
>>>>>>>>> -Rupa
>>>>>>>>>
>>>>>>>>> _______________________________________________
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> --
>>>>>>> -Rupa
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> -Rupa
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> -Rupa
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> -Rupa
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100201/eb884ed3/attachment-0001.html 

From shouldbeq931 at googlemail.com  Mon Feb  1 08:54:33 2010
From: shouldbeq931 at googlemail.com (shouldbe q931)
Date: Mon, 1 Feb 2010 16:54:33 +0000
Subject: [Freeswitch-users] PRI Simulator
In-Reply-To: <1AF7532D-2B17-4D6E-97A0-421C595FF674@freeswitch.org>
References: <mailman.5734.1264787871.2495.freeswitch-users@lists.freeswitch.org>
	<AF84ED7F907F493DAC0A29C8FBC54E44@greyhawk.tonecommander.com>
	<1AF7532D-2B17-4D6E-97A0-421C595FF674@freeswitch.org>
Message-ID: <649eaa471002010854s2052d14dr32d5679b97f65c86@mail.gmail.com>

I use a 2nd user Avaya Prologix (v8, so it has teh trace functions),
its not exactly a simulator, but it suffices for most things :-)

Cheers

On Sat, Jan 30, 2010 at 12:21 AM, Michael S  Collins <msc at freeswitch.org> wrote:
> What's your budget?
>
> Sent from my iPhone
>
> On Jan 29, 2010, at 1:14 PM, "Jerry Richards" <jerry.richards at teotech.com
> ?> wrote:
>
>>
>> Can anyone recommend a good PRI simulator? ?Sorry this is off topic
>> a bit.
>>
>> Thanks,
>> Jerry
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>> users
>> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From mbsip at gazeta.pl  Mon Feb  1 09:30:36 2010
From: mbsip at gazeta.pl (mbsip)
Date: Mon, 1 Feb 2010 18:30:36 +0100
Subject: [Freeswitch-users] vm-disk-quota
In-Reply-To: <7DFBD163-A72A-4186-BB89-6D468FD9ABA2@jerris.com>
References: <28f27f5d1001311246h17b426a4x39e0d48d3d305342@mail.gmail.com>
	<7DFBD163-A72A-4186-BB89-6D468FD9ABA2@jerris.com>
Message-ID: <28f27f5d1002010930x61bcef3v54e22621761c8bbd@mail.gmail.com>

Probably You are right Mike.
I am about to do some tests and give you feadback here.

Thx,
Maciej.

2010/2/1 Michael Jerris <mike at jerris.com>:
> If I read the code right (mod_voicemail.c:3051) it looks like it is measuring in seconds of vocicemail, but the wiki indicates number of voicemails, neither seems to match the name of the param.
>
> Tamas- ?Can you comment on how this was intended to work?
>
> Mike
>
> On Jan 31, 2010, at 3:46 PM, mbsip wrote:
>
>> Hi ALL,
>>
>> Maybe this question will be piece of cake for most of you, but it
>> makes me think.
>>
>> I would like to configure "vm-disk-quota" for all users i have.
>> I followed the wiki page and provided:
>> <param name="vm-disk-quota" value="10"/>
>> to /conf/directory/default/1000.xml
>>
>> After reloadxml, incoming call give me "mod_voicemail.c:3057 Voicemail
>> disk quota is exceeded" feedback
>> No surprise for me because i had more less 10 voice mails already
>> recorded (before the vm-disk-quota was set up).
>> Strange is that increasing value even to 100 does not change anything.
>> The same thing with deleting recordings from user directory.
>> The only wayout is to set it to default value=0 (even FS shutdown
>> doesn't change anything)
>>
>> I am wondering why vm-disk-quota produces "Voicemail disk quota is
>> exceeded" all the time
>> Where the module is looking for stored voicemail recordings.
>>
>> Below is part of my configuration.
>> 1) /conf/autoload_configs/voicemail.conf.xml
>> <param name="storage-dir" value="/usr/local/freeswitch/vm"/>
>> 2) /conf/directory/default/1000.xml
>> <param name="vm-disk-quota" value="100"/>
>> 3) /vm/FS_ip_address/1000 is empty
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From jerry.richards at teotech.com  Mon Feb  1 09:30:57 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Mon, 1 Feb 2010 09:30:57 -0800
Subject: [Freeswitch-users] Presence Change Distribution
In-Reply-To: <191c3a031001271308l5c0c4eedw925e7660fbc2069d@mail.gmail.com>
References: <96888A19920E403880AAA9F6EE061BB6@greyhawk.tonecommander.com><2160023e0912290002q3d0f3fden5adee6d87d4bde25@mail.gmail.com><26B8578C14BA4BE18F2D1278B0C9561B@greyhawk.tonecommander.com><191c3a031001251104p55ba7009g9381841f7de56d65@mail.gmail.com><BE6F5600B8E54EE297963A8B08F70E5E@greyhawk.tonecommander.com><191c3a031001261321v2e8ea21cm7da19e01a11b59f9@mail.gmail.com><591B9C113F064880993543272B16ADF3@greyhawk.tonecommander.com>
	<191c3a031001271308l5c0c4eedw925e7660fbc2069d@mail.gmail.com>
Message-ID: <0616A915D48F4D439FA49390E881B2B6@greyhawk.tonecommander.com>

Cool.  It appears to be working now.  I see instantaneous changes in
presence between two Bria softphones.
 
Thanks and Best Regards,
Jerry


  _____  

From: Anthony Minessale [mailto:anthony.minessale at gmail.com] 
Sent: Wednesday, January 27, 2010 1:09 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Presence Change Distribution


Try latest trunk.
I tried forcing the db update in real-time to avoid a race on the event.



 


On Wed, Jan 27, 2010 at 1:56 PM, Jerry Richards <jerry.richards at teotech.com>
wrote:


There are two places in the XML body that are diffierent:
 
FS Rcvd PUBLISH has:  <rpid:away/> and <dm:note>Away</dm:note>
FS Sent NOTIFY   has:  <rpid:busy/> and <dm:note>Busy</dm:note>
 
This behavior (above) is why I'm not seeing the published presence at the
subscribing softphone.  FS should be sending the new Away status in the
NOTIFY message.
 
I did notice there is an "[ERR] sofia_presnece.c:674" FS log between the
PUBLISH and the NOTIFY (please see Line 89 of
http://pastebin.freeswitch.org/11953).  Line 674 is in the
sofia_presence_event_thread_run() function where it calls
switch_mutex_unlock(mod_sofia_globals.mutex).  Do you think this [ERR] is
related to why FS sends the previous status and not updated status?
 
Thanks And Best Regards,
Jerry
 

  _____  


From: Anthony Minessale [mailto:anthony.minessale at gmail.com] 

Sent: Tuesday, January 26, 2010 1:22 PM 

To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Presence Change Distribution


its sending a notify to them right away (line 174 of your PB)
the xml in the notify we send looks the same as what they sent except one
thing 

They send: <rpid:away/>
We send: <rpid:busy/>

everybody who implements this seems to have their own idea of what to say
here.

This crazy xml presence crap is pure garbage so maybe that's it.

 


On Tue, Jan 26, 2010 at 3:02 PM, Jerry Richards <jerry.richards at teotech.com>
wrote:


Okay, I setup my FS to force SUBSCRIBE Expires to 3600 seconds.  Then I
captured a FS console trace of a Bria softphone changing it's presence state
from 'Busy' to 'Away' (see http://pastebin.freeswitch.org/11953) and
observed that the subscribing Bria softphone did not update to 'Away'.  At
the same time, I executed the sqlite3 app and pasted each of the 3 SQL
select statements I saw in the FS console log, and pasted them below.  I'm
new to sqlite3.  Do you see what my issue is?
 
sqlite>
sqlite> select
sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_hos
t,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscripti
ons.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscripti
ons.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subsc
riptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,
'Away','away','192.168.72.79',sip_presence.status,sip_presence.rpid from
sip_subscriptions left join sip_presence on
(sip_subscriptions.sub_to_user=sip_presence.sip_user and
sip_subscriptions.sub_to_host=sip_presence.sip_host and
sip_subscriptions.profile_name=sip_presence.profile_name) where
sip_subscriptions.expires > -1 and (event='presence' or event='presence')
and sub_to_user='5401' and (sub_to_host='192.168.72.79' or presence_hosts
like '%192.168.72.79%') and (sip_subscriptions.profile_name = 'internal' or
sip_subscriptions.presence_hosts != sip_subscriptions.sub_to_host);
sip|5382|192.168.72.79|5401|192.168.72.79|presence|"5382 on 79"
<sip:5382 at 192.168.72.150:34672>|ZTQ2ZWQwZGRlZjRiNTdkYTJjNGM5NTgzOWIyNmIwZmU.
|"5382 on 79" <sip:5382 at 192.168.72.79 <mailto:sip%3A5382 at 192.168.72.79>
>;tag=68bb4eb6|SIP/2.0/UDP
192.168.72.150:34672;branch=z9hG4bK-d8754z-eafc60166305eaef-1---d8754z-;rpor
t=34672|1264546204|Teo Softphone release 2.5.4 stamp
55958||internal|Away|away|192.168.72.79|Away|away
sqlite>
sqlite>
sqlite>
sqlite>
sqlite>
sqlite>
sqlite>
sqlite>
sqlite> select
sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_hos
t,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscripti
ons.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscripti
ons.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subsc
riptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,
'Away','away','192.168.72.79',sip_presence.status,sip_presence.rpid from
sip_subscriptions left join sip_presence on
(sip_subscriptions.sub_to_user=sip_presence.sip_user and
sip_subscriptions.sub_to_host=sip_presence.sip_host and
sip_subscriptions.profile_name=sip_presence.profile_name) where
sip_subscriptions.expires > -1 and (event='presence' or event='presence')
and sub_to_user='5401' and (sub_to_host='192.168.72.79' or presence_hosts
like '%192.168.72.79%') and (sip_subscriptions.profile_name =
'internal-ipv6' or sip_subscriptions.presence_hosts !=
sip_subscriptions.sub_to_host);
sqlite>
sqlite>
sqlite>
sqlite>
sqlite>
sqlite>
sqlite>
sqlite> select
sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_hos
t,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscripti
ons.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscripti
ons.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subsc
riptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,
'Away','away','192.168.72.79',sip_presence.status,sip_presence.rpid from
sip_subscriptions left join sip_presence on
(sip_subscriptions.sub_to_user=sip_presence.sip_user and
sip_subscriptions.sub_to_host=sip_presence.sip_host and
sip_subscriptions.profile_name=sip_presence.profile_name) where
sip_subscriptions.expires > -1 and (event='presence' or event='presence')
and sub_to_user='5401' and (sub_to_host='192.168.72.79' or presence_hosts
like '%192.168.72.79%') and (sip_subscriptions.profile_name = 'external' or
sip_subscriptions.presence_hosts != sip_subscriptions.sub_to_host);
sqlite>

Thanks and Best Regards,
Jerry


  _____  

From: Anthony Minessale [mailto:anthony.minessale at gmail.com] 
Sent: Monday, January 25, 2010 11:05 AM 

To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Presence Change Distribution


the notify will be instant after the publish
the notify you see are not triggered by the publish or they would be
instant.

Same drill, turn on presence debugging in sofia.conf.xml 
and look at the sql stmts and see why



On Mon, Jan 25, 2010 at 12:30 PM, Jerry Richards
<jerry.richards at teotech.com> wrote:


Okay, I notice that if I reduce the Presence SUBSCRIBE Expires duration
(from 3600 seconds to 60 seconds), then the delay between PUBLISH's and
NOTIFY's is reduced, but FS still waits nearly 45 seconds to send the
NOTIFY's after it receives a PUBLISH.
 
Can a change be made in FS so that NOTIFYs are sent as a direct result of
receipt of the PUBLISH message?  And not tied to the SUBSCRIBE expiration?
I really don't want to configure all my phones to re-subscribe every 30 or
15 seconds.
 
Thanks and Best Regards,
Jerry
 


  _____  

From: RobertT [mailto:siniypin at gmail.com] 
Sent: Tuesday, December 29, 2009 12:02 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Presence Change Distribution


You can try to reduce your registration time. 
I for one made my client apps send PUBLISH message every minute in addition
to reduced registration time.

Regards, Robert.


2009/12/28 Jerry Richards <jerry.richards at teotech.com>


Is there a setting to control how fast FS distributes presence changes to
subscribers?  Currently, it appears to take several minutes before I see
presence changes.  I would like to see them almost instantaneously, if
possible.

Thanks and Best Regards,
Jerry


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org






-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
<mailto:MSN%3Aanthony_minessale at hotmail.com> 
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
<mailto:PAYPAL%3Aanthony.minessale at gmail.com> 
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:+19193869900



_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org






-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
<mailto:MSN%3Aanthony_minessale at hotmail.com> 
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
<mailto:PAYPAL%3Aanthony.minessale at gmail.com> 
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:+19193869900



_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org






-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
<mailto:MSN%3Aanthony_minessale at hotmail.com> 
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
<mailto:PAYPAL%3Aanthony.minessale at gmail.com> 
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:+19193869900


-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100201/04eb7edc/attachment-0001.html 

From rupa at rupa.com  Mon Feb  1 09:32:06 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Mon, 1 Feb 2010 11:32:06 -0600
Subject: [Freeswitch-users] mod_lcr problem
In-Reply-To: <ade741d61002010816x4e42ab0eqbfdb00089883654a@mail.gmail.com>
References: <ade741d61001291443l6134a3achf28be64fc7746c8e@mail.gmail.com>
	<b39273531001301559h60668746kd5c63ada240513f1@mail.gmail.com>
	<ade741d61001301623w70d7e449i44a9f1e9fb7beae4@mail.gmail.com>
	<b39273531001301645w10446308o3578479dcbf2da2@mail.gmail.com>
	<ade741d61001302057w6ce72355iee6e8b4a2872fc81@mail.gmail.com>
	<b39273531001310532y1de53609rf474755cec04f218@mail.gmail.com>
	<9AB8B620-5A69-4F08-B62C-FFEB03FC6762@gmail.com>
	<b39273531001311507hb1bfe02m8d370443b882ef81@mail.gmail.com>
	<ade741d61002010747v6da3d2e1x554b46aa26c27b18@mail.gmail.com>
	<ade741d61002010816x4e42ab0eqbfdb00089883654a@mail.gmail.com>
Message-ID: <b39273531002010932h387914dq8836aa26fbb2cda9@mail.gmail.com>

Thanks.  It is a bit of a foot-gun.  This example might be worth adding to
the wiki for that app, would you mind doing it?

On Mon, Feb 1, 2010 at 10:16 AM, Mouncif Benniane <mouncifbb at gmail.com>wrote:

> I got it! nevermind.
>
> session.execute("set_profile_var","caller_id_number=1617947XXXX"); ( since
> I am using js)
>
> Thanks
>
>
> On Mon, Feb 1, 2010 at 10:47 AM, Mouncif Benniane <mouncifbb at gmail.com>wrote:
>
>> any example on how to use: set_profile_var?
>>
>> thanks
>>
>>
>>
>> On Sun, Jan 31, 2010 at 6:07 PM, Rupa Schomaker <rupa at rupa.com> wrote:
>>
>>> Yes, you need to normalize the values passed to lcr.  Otherwise, how
>>> could it work?
>>>
>>> You can normalize the CID by matching and adding a 1 for 10 digit #s, or
>>> removing the leading + or other things you might need then setting it back
>>> to the profile using the set_profile_var app (
>>> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_set_profile_var).
>>>  (mod_cidlookup will set it after doing a #->name/area lookup - but for now
>>> you can set it yourself)
>>>
>>> You can normalize the DID by doing similar matching rules as above and
>>> then transfering to that normalized DID for the rest of your call plan
>>> processing.
>>>
>>> I'm pretty sure mod_cidlookup has an example of normalizing... yeah:
>>> http://wiki.freeswitch.org/wiki/Mod_cidlookup#Dialplan_Application
>>>
>>> On Sun, Jan 31, 2010 at 4:18 PM, Mouncifbb <mouncifbb at gmail.com> wrote:
>>>
>>>> So the CID must have 1 at front also? Usually people
>>>> Send only npa and nxx ex 6176427788 7817612233
>>>> Do I need to alter it?
>>>>
>>>> Sent from my iPhone
>>>>
>>>> On Jan 31, 2010, at 8:32 AM, Rupa Schomaker <rupa at rupa.com> wrote:
>>>>
>>>>
>>>>
>>>> On Sat, Jan 30, 2010 at 10:57 PM, Mouncif Benniane <<mouncifbb at gmail.com>
>>>> mouncifbb at gmail.com> wrote:
>>>>
>>>>> OK going back to use default profile to keep things simple below 2
>>>>> results
>>>>>
>>>>> Using:
>>>>>
>>>>> lcr 16179470890  default  19785223241  ( this one consult
>>>>> npa_nxx_company_ocn)
>>>>>
>>>>> lcr 6179470890  default  9785223241 ( this one don't!! )
>>>>>
>>>>>
>>>>>
>>>> Oh, right!  mod_lcr really expects you to normalize your prefix to e164
>>>> format.  I thought there was discussion about this in the wiki, but maybe
>>>> not.  For simple prefix matching it doesn't matter, but for things that make
>>>> decisions based on the # (like the lata/state stuff) it does.
>>>>
>>>> npanxx lookup only makes sense for NANPA numbers.  NANPA numbers have a
>>>> country code of "1" and a total length of 11 (including the 1).
>>>>
>>>> This is the only rational way to do it when you have a rate table with
>>>> both domestic (NANPA) and international prefixes.
>>>>
>>>>
>>>>> freeswitch> lcr 16179470890  default  19785223241
>>>>> 2010-01-30 23:53:34.681842 [DEBUG] mod_lcr.c:1329 data passed to lcr is
>>>>> [16179470890  default  19785223241]
>>>>> 2010-01-30 23:53:34.681842 [DEBUG] mod_lcr.c:1365 Set Caller ID to
>>>>> [19785223241]
>>>>> 2010-01-30 23:53:34.681842 [DEBUG] mod_lcr.c:736 SQL: SELECT 'state',
>>>>> count(DISTINCT state) FROM npa_nxx_company_ocn WHERE (npa=617 AND nxx=947)
>>>>> OR (npa=978 AND nxx=522) UNION SELECT 'lata', count(DISTINCT lata) FROM
>>>>> npa_nxx_company_ocn WHERE (npa=617 AND nxx=947) OR (npa=978 AND nxx=522)
>>>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:696 Type: state, Count: 1
>>>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:696 Type: lata, Count: 1
>>>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:786 intra routing [state:1
>>>>> lata:1] so rate field is [intralata_rate]
>>>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:802 we have an event
>>>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:826 SQL: SELECT l.digits,
>>>>> c.carrier_name, l.intralata_rate, cg.prefix AS gw_prefix, cg.suffix AS
>>>>> gw_suffix, l.lead_strip, l.trail_strip, l.prefix, l.suffix , cg.codec ,
>>>>> l.cid FROM lcr l JOIN carriers c ON l.carrier_id=c.id JOIN
>>>>> carrier_gateway cg ON c.id=cg.carrier_id WHERE c.enabled = '1' AND
>>>>> cg.enabled = '1' AND l.enabled = '1' AND digits IN (16179470890, 1617947089,
>>>>> 161794708, 16179470, 1617947, 161794, 16179, 1617, 161, 16, 1) AND
>>>>> CURRENT_TIMESTAMP BETWEEN date_start AND date_end ORDER BY digits DESC,
>>>>> intralata_rate, rand();
>>>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:307 Returning Dialstring
>>>>> [lcr_carrier=carrier1,lcr_rate=0.00000]sofia/gateway/carrier1/16179470890
>>>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:615 Adding carrier1 to
>>>>> head of list
>>>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:307 Returning Dialstring
>>>>> [lcr_carrier=carrier2,lcr_rate=0.00000]sofia/external/
>>>>> 06179470890 at proxy.carrier2.net:5060
>>>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:667 adding carrier2 to end
>>>>> of list after carrier1
>>>>>
>>>>>
>>>>>  | Digit Match | Carrier  | Rate     | Codec | CID Regexp |
>>>>> Dialstring
>>>>> |
>>>>>  | 1           | carrier1 | 0.00000  |       |            |
>>>>> [lcr_carrier=carrier1,lcr_rate=0.00000]sofia/gateway/carrier1/16179470890
>>>>> |
>>>>>  | 1           | carrier2 | 0.00000  |       |            |
>>>>> [lcr_carrier=carrier2,lcr_rate=0.00000]sofia/external/
>>>>> 06179470890 at proxy.carrier2.net:5060 |
>>>>>
>>>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:307 Returning Dialstring
>>>>> [lcr_carrier=carrier1,lcr_rate=0.00000]sofia/gateway/carrier1/16179470890
>>>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:307 Returning Dialstring
>>>>> [lcr_carrier=carrier2,lcr_rate=0.00000]sofia/external/
>>>>> 06179470890 at proxy.carrier2.net:5060
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> freeswitch> lcr 6179470890  default  9785223241
>>>>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:1329 data passed to lcr is
>>>>> [6179470890  default  9785223241]
>>>>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:1365 Set Caller ID to
>>>>> [9785223241]
>>>>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:786 intra routing [state:0
>>>>> lata:0] so rate field is [rate]
>>>>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:802 we have an event
>>>>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:826 SQL: SELECT l.digits,
>>>>> c.carrier_name, l.rate, cg.prefix AS gw_prefix, cg.suffix AS gw_suffix,
>>>>> l.lead_strip, l.trail_strip, l.prefix, l.suffix , cg.codec , l.cid FROM lcr
>>>>> l JOIN carriers c ON l.carrier_id=c.id JOIN carrier_gateway cg ON c.id=cg.carrier_id
>>>>> WHERE c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1' AND digits IN
>>>>> (6179470890, 617947089, 61794708, 6179470, 617947, 61794, 6179, 617, 61, 6)
>>>>> AND CURRENT_TIMESTAMP BETWEEN date_start AND date_end ORDER BY digits DESC,
>>>>> rate, rand();
>>>>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:307 Returning Dialstring
>>>>> [lcr_carrier=carrier1,lcr_rate=0.09000]sofia/gateway/carrier1/16179470890
>>>>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:615 Adding carrier1 to
>>>>> head of list
>>>>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:307 Returning Dialstring
>>>>> [lcr_carrier=carrier1,lcr_rate=0.09000]sofia/gateway/carrier1/16179470890
>>>>>
>>>>>
>>>>>  | Digit Match | Carrier  | Rate     | Codec | CID Regexp |
>>>>> Dialstring                                                                |
>>>>>   | 617947      | carrier1 | 0.09000  |       |            |
>>>>> [lcr_carrier=carrier1,lcr_rate=0.09000]sofia/gateway/carrier1/16179470890 |
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> On Sat, Jan 30, 2010 at 7:45 PM, Rupa Schomaker < <rupa at rupa.com>
>>>>> rupa at rupa.com> wrote:
>>>>>
>>>>>> turn up logging to debug again, and then reload mod_lcr.  It'll spit
>>>>>> out a bunch of crap when it tests out each profile you have defined.  Give
>>>>>> me the full log (here or in <http://pastebin.freeswitch.org>
>>>>>> pastebin.freeswitch.org).  That may show more useful info as to why
>>>>>> things are mucked up?
>>>>>>
>>>>>>
>>>>>> On Sat, Jan 30, 2010 at 6:23 PM, Mouncif Benniane <<mouncifbb at gmail.com>
>>>>>> mouncifbb at gmail.com> wrote:
>>>>>>
>>>>>>> yes I use NANPA_STD profile instead of default cause I thought the
>>>>>>> custom profile was causing issues, but looks like it's returning same
>>>>>>> results.
>>>>>>>
>>>>>>> There is this line in thw wiki:
>>>>>>> intra lata/state selection is done manually by setting the channel
>>>>>>> variables *intrastate* or *intralata* to the value *true*.
>>>>>>>
>>>>>>> do I have to set these ? if yes how?
>>>>>>>
>>>>>>> Thanks
>>>>>>>
>>>>>>>
>>>>>>> On Sat, Jan 30, 2010 at 6:59 PM, Rupa Schomaker < <rupa at rupa.com>
>>>>>>> rupa at rupa.com> wrote:
>>>>>>>
>>>>>>>> Stuff inline.
>>>>>>>>
>>>>>>>> On Sat, Jan 30, 2010 at 3:38 PM, Mouncif Benniane <<mouncifbb at gmail.com>
>>>>>>>> mouncifbb at gmail.com> wrote:
>>>>>>>>
>>>>>>>>>  NPANXX,"LATA","OCN","NTER","INTRA"
>>>>>>>>> 201007,"224","7229","0.0059","0.0127"
>>>>>>>>> 201040,"224","9206","0.0036","0.0036"
>>>>>>>>>
>>>>>>>>
>>>>>>>> Looks like they give you the LATA and OCN values with the prefix.
>>>>>>>>  We (should) look that up ourselves.
>>>>>>>>
>>>>>>>>
>>>>>>>>> FreeSWITCH Version 1.0.trunk (16540)
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> Also I noticed the *npa_nxx_ocn* table never get consulted.
>>>>>>>>>
>>>>>>>>> I also see this now when making a real call instead of running
>>>>>>>>> thorugh CLI
>>>>>>>>>
>>>>>>>>> EXECUTE sofia/external/6179472456 at 174.x.x.x lcr(16179470890
>>>>>>>>> NANPA_STD)
>>>>>>>>> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:1230 *intrastate
>>>>>>>>> channel var is [undef]*
>>>>>>>>
>>>>>>>>
>>>>>>>> This is fine.  it is a leftover from when you would tell mod_lcr via
>>>>>>>> a channel var that it should do intrastate.  I later had mod_lcr do the
>>>>>>>> lookup itself, but we still honor the old var.  There are no channel vars
>>>>>>>> associated with the cli, so you wouldn't see that msg.
>>>>>>>>
>>>>>>>>
>>>>>>>>>
>>>>>>>>> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:1233 Select routes
>>>>>>>>> based on interstate rates
>>>>>>>>> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:1252 LCR Lookup on
>>>>>>>>> 16179470893 using profile NANPA_STD
>>>>>>>>> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:786 intra routing
>>>>>>>>> [state:0 lata:0] so rate field is [rate]
>>>>>>>>>
>>>>>>>>> called number 6179470890  caller ID: 6179472456
>>>>>>>>>
>>>>>>>>> any ideas??
>>>>>>>>>
>>>>>>>>>
>>>>>>>> Only thing that jumps out at me.
>>>>>>>>
>>>>>>>> The output from lcr_admin show profiles showed only the default one.
>>>>>>>>  On the dialplan you use the NANPA_STD profile.  Can you check lcr_admin
>>>>>>>> list and see if that profile is defined and if so if it says it is using the
>>>>>>>> npanxx table?
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> On Sat, Jan 30, 2010 at 10:02 AM, Rupa Schomaker < <rupa at rupa.com>
>>>>>>>>> rupa at rupa.com> wrote:
>>>>>>>>>
>>>>>>>>>> Something is still missing from the logs. Note the query of the
>>>>>>>>>> npanxx
>>>>>>>>>> table, the flags being set, and the rate field being chosen.
>>>>>>>>>>  Umm..
>>>>>>>>>> oh, what version of fs are you running?
>>>>>>>>>>
>>>>>>>>>> Yes, the npa_nxx_ocn table needs to be loaded up as described in:
>>>>>>>>>> <http://wiki.freeswitch.org/wiki/Mod_cidlookup#Falling_back_to_.22City_State.22_in_the_absense_of_a_name>
>>>>>>>>>> http://wiki.freeswitch.org/wiki/Mod_cidlookup#Falling_back_to_.22City_State.22_in_the_absense_of_a_name
>>>>>>>>>> (there is a link to that from mod_lcr's wiki page).
>>>>>>>>>>
>>>>>>>>>> An example from my own setup:
>>>>>>>>>>
>>>>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:1384 data passed to
>>>>>>>>>> lcr
>>>>>>>>>> is [12148267711 default 12148267712]
>>>>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:1420 Set Caller ID to
>>>>>>>>>> [12148267712]
>>>>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:759 SQL: SELECT
>>>>>>>>>> 'state',
>>>>>>>>>> count(DISTINCT state) FROM npa_nxx_company_ocn WHERE (npa=214 AND
>>>>>>>>>> nxx=826) OR (npa=214 AND nxx=826) UNION SELECT 'lata',
>>>>>>>>>> count(DISTINCT
>>>>>>>>>> lata) FROM npa_nxx_company_ocn WHERE (npa=214 AND nxx=826) OR
>>>>>>>>>> (npa=214
>>>>>>>>>> AND nxx=826)
>>>>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:719 Type: lata,
>>>>>>>>>> Count: 1
>>>>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:719 Type: state,
>>>>>>>>>> Count: 1
>>>>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:809 intra routing
>>>>>>>>>> [state:1 lata:1] so rate field is [intralata_rate]
>>>>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:825 we have an event
>>>>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:849 SQL:  SELECT
>>>>>>>>>> l.digits
>>>>>>>>>> AS lcr_digits,         c.carrier_name AS lcr_carrier_name,
>>>>>>>>>> l.intralata_rate as lcr_rate_field,        cg.prefix AS
>>>>>>>>>> lcr_gw_prefix,
>>>>>>>>>> cg.suffix AS lcr_gw_suffix,        l.lead_strip AS lcr_lead_strip,
>>>>>>>>>> l.trail_strip AS lcr_trail_strip,        l.prefix AS lcr_prefix,
>>>>>>>>>> l.suffix AS lcr_suffix,        cg.codec AS lcr_codec, l.cid AS
>>>>>>>>>> lcr_cid,        'carriers' AS lcr_limit_realm, c.carrier_name AS
>>>>>>>>>> lcr_limit_id, 5 AS lcr_limit_max   FROM lcr l      JOIN carriers c
>>>>>>>>>> ON
>>>>>>>>>> l.carrier_id=c.id      JOIN carrier_gateway cg ON c.id
>>>>>>>>>> =cg.carrier_id
>>>>>>>>>>  WHERE c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1'
>>>>>>>>>> AND digits_prefix @> '12148267711'      AND CURRENT_TIMESTAMP
>>>>>>>>>> BETWEEN
>>>>>>>>>> date_start AND date_end   ORDER BY digits DESC, intralata_rate,
>>>>>>>>>> random();
>>>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning
>>>>>>>>>> Dialstring
>>>>>>>>>>
>>>>>>>>>> [lcr_carrier=grnvoip_us,lcr_rate=0.00591]sofia/gateway/grnvoip/XXXX12148267711
>>>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:638 Adding grnvoip_us
>>>>>>>>>> to
>>>>>>>>>> head of list
>>>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning
>>>>>>>>>> Dialstring
>>>>>>>>>>
>>>>>>>>>> [lcr_carrier=teliax,lcr_rate=0.01000]sofia/gateway/teliax_atl/12148267711
>>>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:660 Adding teliax to
>>>>>>>>>> end of list
>>>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning
>>>>>>>>>> Dialstring
>>>>>>>>>>
>>>>>>>>>> [lcr_carrier=teliax,lcr_rate=0.01000]sofia/gateway/teliax/12148267711
>>>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:660 Adding teliax to
>>>>>>>>>> end of list
>>>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning
>>>>>>>>>> Dialstring
>>>>>>>>>>
>>>>>>>>>> [lcr_carrier=vitelity,lcr_rate=0.01440]sofia/gateway/vitelity/12148267711
>>>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:660 Adding vitelity
>>>>>>>>>> to end of list
>>>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning
>>>>>>>>>> Dialstring
>>>>>>>>>>
>>>>>>>>>> [lcr_carrier=grnvoip_std,lcr_rate=0.01500]sofia/gateway/grnvoip/YYYY12148267711
>>>>>>>>>> [...]
>>>>>>>>>>
>>>>>>>>>> On Fri, Jan 29, 2010 at 10:42 PM, Mouncif Benniane <<mouncifbb at gmail.com>
>>>>>>>>>> mouncifbb at gmail.com> wrote:
>>>>>>>>>> > Also the Provider has presented the rates in this format?
>>>>>>>>>> > NPANXXLATA OCN INTER INTRA
>>>>>>>>>> >
>>>>>>>>>> >
>>>>>>>>>> > On Fri, Jan 29, 2010 at 11:30 PM, Mouncif Benniane <<mouncifbb at gmail.com>
>>>>>>>>>> mouncifbb at gmail.com>
>>>>>>>>>> > wrote:
>>>>>>>>>> >>
>>>>>>>>>> >> Tried it and it's not giving me intralata  instead I get
>>>>>>>>>> interstate, does
>>>>>>>>>> >> the npa_nxx_company_ocn table needs to be used in this case?,
>>>>>>>>>> also do I have
>>>>>>>>>> >> to have the rate field in lcr table?
>>>>>>>>>> >>
>>>>>>>>>> >> lcr 617642  default 6176421212
>>>>>>>>>> >>  | Digit Match | Carrier  | Rate     | Codec | CID Regexp |
>>>>>>>>>> Dialstring
>>>>>>>>>> >>                                                        |
>>>>>>>>>> >>  | 617642      | carrier1 | 0.00500  |       |            |
>>>>>>>>>> >>
>>>>>>>>>> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642 |
>>>>>>>>>> >>
>>>>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:1329 data passed
>>>>>>>>>> to lcr is
>>>>>>>>>> >> [617642  default 6176421212]
>>>>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:1365 Set Caller ID
>>>>>>>>>> to
>>>>>>>>>> >> [6176421212]
>>>>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:786 intra routing
>>>>>>>>>> [state:0
>>>>>>>>>> >> lata:0] so rate field is [rate]
>>>>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:802 we have an
>>>>>>>>>> event
>>>>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:826 SQL: SELECT
>>>>>>>>>> l.digits,
>>>>>>>>>> >> c.carrier_name, l.rate, cg.prefix AS gw_prefix, cg.suffix AS
>>>>>>>>>> gw_suffix,
>>>>>>>>>> >> l.lead_strip, l.trail_strip, l.prefix, l.suffix , cg.codec ,
>>>>>>>>>> l.cid FROM lcr
>>>>>>>>>> >> l JOIN carriers c ON l.carrier_id=c.id JOIN carrier_gateway cg
>>>>>>>>>> ON
>>>>>>>>>> >> c.id=cg.carrier_id WHERE c.enabled = '1' AND cg.enabled = '1'
>>>>>>>>>> AND l.enabled
>>>>>>>>>> >> = '1' AND digits IN (617642, 61764, 6176, 617, 61, 6) AND
>>>>>>>>>> CURRENT_TIMESTAMP
>>>>>>>>>> >> BETWEEN date_start AND date_end ORDER BY digits DESC,  rate,
>>>>>>>>>> rand();
>>>>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:307 Returning
>>>>>>>>>> Dialstring
>>>>>>>>>> >>
>>>>>>>>>> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642
>>>>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:615 Adding
>>>>>>>>>> carrier1 to head
>>>>>>>>>> >> of list
>>>>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:307 Returning
>>>>>>>>>> Dialstring
>>>>>>>>>> >>
>>>>>>>>>> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642
>>>>>>>>>> >>
>>>>>>>>>> >> Thank you Rupa!
>>>>>>>>>> >>
>>>>>>>>>> >> On Fri, Jan 29, 2010 at 7:37 PM, Rupa Schomaker <<rupa at rupa.com>
>>>>>>>>>> rupa at rupa.com> wrote:
>>>>>>>>>> >>>
>>>>>>>>>> >>> turn console logging up to debug and redo the lcr lookup.  The
>>>>>>>>>> sql
>>>>>>>>>> >>> statements along with status info will show up.  This should
>>>>>>>>>> give
>>>>>>>>>> >>> enough information to debug what is happening.
>>>>>>>>>> >>>
>>>>>>>>>> >>> I'm assuming the npanxx table is actually populated and not
>>>>>>>>>> just
>>>>>>>>>> >>> existing?
>>>>>>>>>> >>>
>>>>>>>>>> >>> When doing the lookup from the cli you have to tell lcr what
>>>>>>>>>> CID to
>>>>>>>>>> >>> use (remember, it is relative to the src/dest number).  I'm
>>>>>>>>>> pretty
>>>>>>>>>> >>> sure you get something on the console log when you don't
>>>>>>>>>> specify a CID
>>>>>>>>>> >>> when using the commandline.  Anyway:
>>>>>>>>>> >>>
>>>>>>>>>> >>> lcr 617642  default 6176421212
>>>>>>>>>> >>>
>>>>>>>>>> >>> should give you intralata.
>>>>>>>>>> >>>
>>>>>>>>>> >>> Note that the definition of intralata doesn't mean "local" for
>>>>>>>>>> some
>>>>>>>>>> >>> providers.  Some providers define local to "same ratecenter"
>>>>>>>>>> which is
>>>>>>>>>> >>> even more restrictive.
>>>>>>>>>> >>>
>>>>>>>>>> >>> On Fri, Jan 29, 2010 at 4:43 PM, Mouncif Benniane <<mouncifbb at gmail.com>
>>>>>>>>>> mouncifbb at gmail.com>
>>>>>>>>>> >>> wrote:
>>>>>>>>>> >>> > i can't make use of mod_lcr using Intra/Interstate rating, I
>>>>>>>>>> am using
>>>>>>>>>> >>> > svn: FreeSWITCH Version 1.0.trunk (16517)
>>>>>>>>>> >>> >
>>>>>>>>>> >>> > lcr mysql table structure:
>>>>>>>>>> >>> > CREATE TABLE `lcr` (
>>>>>>>>>> >>> >   `id` INT(11) NOT NULL AUTO_INCREMENT,
>>>>>>>>>> >>> >   `digits` VARCHAR(15) DEFAULT NULL,
>>>>>>>>>> >>> >   `rate` FLOAT(11,5) DEFAULT NULL,
>>>>>>>>>> >>> >   `intrastate_rate` FLOAT(11,5) DEFAULT NULL,
>>>>>>>>>> >>> >   `intralata_rate` FLOAT(11,5) DEFAULT NULL,
>>>>>>>>>> >>> >   `carrier_id` INT(11) NOT NULL,
>>>>>>>>>> >>> >   `lead_strip` INT(11) NOT NULL,
>>>>>>>>>> >>> >   `trail_strip` INT(11) NOT NULL,
>>>>>>>>>> >>> >   `prefix` VARCHAR(16) NOT NULL,
>>>>>>>>>> >>> >   `suffix` VARCHAR(16) NOT NULL,
>>>>>>>>>> >>> >   `lcr_profile` VARCHAR(32) DEFAULT NULL,
>>>>>>>>>> >>> >   `date_start` DATETIME NOT NULL DEFAULT '1970-01-01
>>>>>>>>>> 00:00:00',
>>>>>>>>>> >>> >   `date_end` DATETIME NOT NULL DEFAULT '2030-12-31
>>>>>>>>>> 00:00:00',
>>>>>>>>>> >>> >   `quality` FLOAT(10,6) NOT NULL,
>>>>>>>>>> >>> >   `reliability` FLOAT(10,6) NOT NULL,
>>>>>>>>>> >>> >   `cid` VARCHAR(32) NOT NULL DEFAULT '',
>>>>>>>>>> >>> >   `enabled` TINYINT(1) NOT NULL DEFAULT '1',
>>>>>>>>>> >>> >   PRIMARY KEY  (`id`),
>>>>>>>>>> >>> >   KEY `carrier_id` (`carrier_id`),
>>>>>>>>>> >>> >   KEY `digits` (`digits`),
>>>>>>>>>> >>> >   KEY `lcr_profile` (`lcr_profile`),
>>>>>>>>>> >>> >   KEY `digits_profile_cid_rate` USING BTREE (`digits`),
>>>>>>>>>> >>> >   CONSTRAINT `carrier_id` FOREIGN KEY (`carrier_id`)
>>>>>>>>>> REFERENCES
>>>>>>>>>> >>> > `carriers`
>>>>>>>>>> >>> > (`id`) ON DELETE CASCADE ON UPDATE CASCADE
>>>>>>>>>> >>> > ) ENGINE=INNODB AUTO_INCREMENT=6 DEFAULT CHARSET=latin1
>>>>>>>>>> >>> >
>>>>>>>>>> >>> >
>>>>>>>>>> >>> > lcr_admin show profiles
>>>>>>>>>> >>> > Name:           default
>>>>>>>>>> >>> >  custom sql:    SELECT l.digits, c.carrier_name,
>>>>>>>>>> l.${lcr_rate_field},
>>>>>>>>>> >>> > cg.prefix AS gw_prefix, cg.suffix AS gw_suffix,
>>>>>>>>>> l.lead_strip,
>>>>>>>>>> >>> > l.trail_strip,
>>>>>>>>>> >>> > l.prefix, l.suffix , cg.codec , l.cid FROM lcr l JOIN
>>>>>>>>>> carriers c ON
>>>>>>>>>> >>> > l.carrier_id=c.id JOIN carrier_gateway cg ON c.id=cg.carrier_id
>>>>>>>>>> WHERE
>>>>>>>>>> >>> > c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1' AND
>>>>>>>>>> digits IN
>>>>>>>>>> >>> > (${lcr_query_expanded_digits}) AND CURRENT_TIMESTAMP BETWEEN
>>>>>>>>>> date_start
>>>>>>>>>> >>> > AND
>>>>>>>>>> >>> > date_end ORDER BY digits DESC,  ${lcr_rate_field},  quality
>>>>>>>>>> DESC,
>>>>>>>>>> >>> >  reliability DESC, rand();
>>>>>>>>>> >>> >  has %:         false
>>>>>>>>>> >>> >  has vars:      true
>>>>>>>>>> >>> >  has intrastate:        true
>>>>>>>>>> >>> >  has intralata: true
>>>>>>>>>> >>> >  has npanxx:    true
>>>>>>>>>> >>> >  Reorder rate:  enabled
>>>>>>>>>> >>> >  Info in headers:       disabled
>>>>>>>>>> >>> >  Quote IN() List:       disabled
>>>>>>>>>> >>> >
>>>>>>>>>> >>> >
>>>>>>>>>> >>> >
>>>>>>>>>> >>> > lcr 617642  default  returns rate from the rate field table
>>>>>>>>>> and not
>>>>>>>>>> >>> > intra/inter state fields rates.
>>>>>>>>>> >>> >
>>>>>>>>>> >>> > Any ideas? thanks!
>>>>>>>>>> >>> >
>>>>>>>>>> >>> >
>>>>>>>>>> >>> >
>>>>>>>>>> >>> >
>>>>>>>>>> >>> > _______________________________________________
>>>>>>>>>> >>> > FreeSWITCH-users mailing list
>>>>>>>>>> >>> > <FreeSWITCH-users at lists.freeswitch.org>
>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>> >>> >
>>>>>>>>>> <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>> >>> >
>>>>>>>>>> >>> > UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>> >>> > <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>>>>> >>> >
>>>>>>>>>> >>> >
>>>>>>>>>> >>>
>>>>>>>>>> >>>
>>>>>>>>>> >>>
>>>>>>>>>> >>> --
>>>>>>>>>> >>> -Rupa
>>>>>>>>>> >>>
>>>>>>>>>> >>> _______________________________________________
>>>>>>>>>> >>> FreeSWITCH-users mailing list
>>>>>>>>>> >>> <FreeSWITCH-users at lists.freeswitch.org>
>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>> >>>
>>>>>>>>>> <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>> >>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>> >>> <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>>>>> >>
>>>>>>>>>> >
>>>>>>>>>> >
>>>>>>>>>> > _______________________________________________
>>>>>>>>>> > FreeSWITCH-users mailing list
>>>>>>>>>> > <FreeSWITCH-users at lists.freeswitch.org>
>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>> > <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>> > UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>> > <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>>>>> >
>>>>>>>>>> >
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> --
>>>>>>>>>> -Rupa
>>>>>>>>>>
>>>>>>>>>> _______________________________________________
>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> _______________________________________________
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>>>>
>>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> --
>>>>>>>> -Rupa
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>> -Rupa
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>>
>>>> --
>>>> -Rupa
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> -Rupa
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100201/72f5cf14/attachment-0001.html 

From ranjtech at gmail.com  Mon Feb  1 09:51:31 2010
From: ranjtech at gmail.com (RR)
Date: Mon, 1 Feb 2010 12:51:31 -0500
Subject: [Freeswitch-users] Outbound Call (No Registration)
In-Reply-To: <02dd01caa127$428aa760$c79ff620$@com>
References: <020c01ca9fe9$1d5952f0$580bf8d0$@com>
	<1254B3C5-D7D3-413D-BA82-54FE4789B360@freeswitch.org>
	<022701caa04a$44f60b80$cee22280$@com>
	<697C7F93-3737-4ABD-8934-6A48DC09C088@freeswitch.org>
	<025701caa0a7$e1ca6200$a55f2600$@com>
	<02b001caa10c$913b9100$b3b2b300$@com>
	<F03237B7-89EE-4CE4-A67F-FCB8E8D26700@freeswitch.org>
	<02c501caa113$105532b0$30ff9810$@com>
	<87f2f3b91001291110l226ce6d8u6e806d0a1b782c5f@mail.gmail.com>
	<02dd01caa127$428aa760$c79ff620$@com>
Message-ID: <032301caa367$307c6c60$91754520$@com>

Hi Folks,

 

As I continue to learn configuring FS, I am trying to use FS as a peering
switch sending a call to our SBC on the other side of the pond (across the
pacific) where the IP address of the FS (in the US) is configured to receive
traffic from. In this case no registration is required by the sending
gateway to make calls through the system overseas as its IP address is in
the "known" gateways (ACL) list. 

 

Two questions:

a)      Where must the configuration of this overseas gateway be? Currently
I have it in $FREESWITCH_HOME/conf/sip_profiles/external. Is that the right
location for it?

b)      Looking at the gateway configuration that was present in that
directory for the sample gateway, I didn't see any of the params that were
relevant other than the "realm" and maybe the "proxy". Hence my
configuration is extremely simple (2 Lines). As I mentioned, I don't need to
register with the overseas SBC. The problem with this now is that this
gateway is not being picked up by FS when I do a reloadxml OR restart FS. On
typing "sofia status gateway MyGW", I get "Invalid Gateway". Following from
this then, when the dialplan is configured to send the call using this
gateway, I see the messages like

 

2010-02-01 12:42:50.138444 [ERR] mod_sofia.c:3108 Invalid Gateway

2010-02-01 12:42:50.138444 [ERR] switch_ivr_originate.c:2411 Cannot create
outgoing channel of type [sofia] cause: [INVALID_NUMBER_FORMAT]

2010-02-01 12:42:50.138444 [DEBUG] switch_ivr_originate.c:3209 Originate
Resulted in Error Cause: 28 [INVALID_NUMBER_FORMAT]

2010-02-01 12:42:50.138444 [INFO] mod_dptools.c:2346 Originate Failed.
Cause: INVALID_NUMBER_FORMAT

2010-02-01 12:42:50.138444 [NOTICE] mod_dptools.c:2409 Hangup
sofia/internal/1000 at 10.1.2.110 [CS_EXECUTE] [INVALID_NUMBER_FORMAT]

 

c)       Please note that the call has NOT even left FS so this message is
not coming from the remote gateway/SBC as I have a trace running there as
well and it never sees the call. Also, when I start FS, I see the error
"username is a required param". Upon entering a bogus username and password
field, I am able to load the gateway and the prfile status looks good.
However, when I try and make a call I get the following:

d)       

2010-02-01 12:49:03.118479 [DEBUG] sofia.c:4011 Channel
sofia/external/0061434144942 entering state [terminated][403]

2010-02-01 12:49:03.118479 [NOTICE] sofia.c:4655 Hangup
sofia/external/0061434144942 [CS_CONSUME_MEDIA] [CALL_REJECTED]

2010-02-01 12:49:03.118479 [DEBUG] switch_channel.c:1947 Send signal
sofia/external/0061434144942 [KILL]

2010-02-01 12:49:03.118479 [DEBUG] switch_core_session.c:1019 Send signal
sofia/external/0061434144942 [BREAK]

2010-02-01 12:49:03.118479 [DEBUG] switch_ivr_originate.c:3209 Originate
Resulted in Error Cause: 21 [CALL_REJECTED]

2010-02-01 12:49:03.118479 [INFO] mod_dptools.c:2346 Originate Failed.
Cause: CALL_REJECTED

2010-02-01 12:49:03.118479 [NOTICE] mod_dptools.c:2409 Hangup
sofia/internal/1000 at 10.1.2.110 [CS_EXECUTE] [CALL_REJECTED]

 

Any ideas?

 

Thanks

\R

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100201/0138de52/attachment.html 

From msc at freeswitch.org  Mon Feb  1 09:58:10 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 1 Feb 2010 09:58:10 -0800
Subject: [Freeswitch-users] PRI Simulator
In-Reply-To: <649eaa471002010854s2052d14dr32d5679b97f65c86@mail.gmail.com>
References: <mailman.5734.1264787871.2495.freeswitch-users@lists.freeswitch.org>
	<AF84ED7F907F493DAC0A29C8FBC54E44@greyhawk.tonecommander.com>
	<1AF7532D-2B17-4D6E-97A0-421C595FF674@freeswitch.org>
	<649eaa471002010854s2052d14dr32d5679b97f65c86@mail.gmail.com>
Message-ID: <87f2f3b91002010958t1e0d9188v659a237bade6e7d5@mail.gmail.com>

On Mon, Feb 1, 2010 at 8:54 AM, shouldbe q931
<shouldbeq931 at googlemail.com>wrote:

> I use a 2nd user Avaya Prologix (v8, so it has teh trace functions),
> its not exactly a simulator, but it suffices for most things :-)
>
> Cheers
>

Heck, in a pinch I've used an Asterisk box. Sure it's unpredictable, crashes
all the time and you never know if it's doing what it's supposed to be
doing. Hmm, on second thought, I guess that makes it a great PRI simulator
because it's just like all the lame telcos...

-MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100201/84a23366/attachment.html 

From shouldbeq931 at googlemail.com  Mon Feb  1 10:06:40 2010
From: shouldbeq931 at googlemail.com (shouldbe q931)
Date: Mon, 1 Feb 2010 18:06:40 +0000
Subject: [Freeswitch-users] PRI Simulator
In-Reply-To: <87f2f3b91002010958t1e0d9188v659a237bade6e7d5@mail.gmail.com>
References: <mailman.5734.1264787871.2495.freeswitch-users@lists.freeswitch.org>
	<AF84ED7F907F493DAC0A29C8FBC54E44@greyhawk.tonecommander.com>
	<1AF7532D-2B17-4D6E-97A0-421C595FF674@freeswitch.org>
	<649eaa471002010854s2052d14dr32d5679b97f65c86@mail.gmail.com>
	<87f2f3b91002010958t1e0d9188v659a237bade6e7d5@mail.gmail.com>
Message-ID: <649eaa471002011006o397574f0nc7e239acfbba229d@mail.gmail.com>

The Prologix might be a little more stable :-)

Eicon (now Dialogic) Diva Server cards running under Windows or Linux
are quite good as well...

On Mon, Feb 1, 2010 at 5:58 PM, Michael Collins <msc at freeswitch.org> wrote:
>
>
> On Mon, Feb 1, 2010 at 8:54 AM, shouldbe q931 <shouldbeq931 at googlemail.com>
> wrote:
>>
>> I use a 2nd user Avaya Prologix (v8, so it has teh trace functions),
>> its not exactly a simulator, but it suffices for most things :-)
>>
>> Cheers
>
> Heck, in a pinch I've used an Asterisk box. Sure it's unpredictable, crashes
> all the time and you never know if it's doing what it's supposed to be
> doing. Hmm, on second thought, I guess that makes it a great PRI simulator
> because it's just like all the lame telcos...
>
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From paul.gore.j at gmail.com  Mon Feb  1 11:18:52 2010
From: paul.gore.j at gmail.com (paul gore)
Date: Mon, 1 Feb 2010 14:18:52 -0500
Subject: [Freeswitch-users] Logging question
In-Reply-To: <8BEB200D-32AC-4D80-B59D-07C8228D7380@jerris.com>
References: <da4ab26e1001292020o280f330ft62a278b93c7fb1fd@mail.gmail.com>
	<8BEB200D-32AC-4D80-B59D-07C8228D7380@jerris.com>
Message-ID: <da4ab26e1002011118i2a5dc8a0yc836b799662adfab@mail.gmail.com>

I did "sofia profile internal siptrace on" - and that gave me the trace on
the console, but still nothing in log files.
This command seem to have same effect as changing the parameter in the
profile xml file and then do profile rescan.
So there is no way to get the trace in logs?

On Mon, Feb 1, 2010 at 1:04 AM, Michael Jerris <mike at jerris.com> wrote:

> sofia profile <profile_name> siptrace on
>
> There is also a config param, it should be documented int he current
> default configs.
>
> Mike
>
>
> On Jan 29, 2010, at 11:20 PM, paul gore wrote:
>
> > Hi there,
> > I am running FS 1.0.trunk (14501) (I know it's old but we serve a small
> community and don't have time to upgrade/test the latest/greatest). I am
> having troubles understanding how to switch SIP trace in log files, I tried
> >
> > fsctl loglevel debug
> > sofia tracelevel debug
> >
> > but it seem to have no effect, I only get sofia debug messages but no
> detailed SIP info.
> > What also puzzling me is if I do
> >
> > console loglevel 0
> >
> > I still get debug information on console.
> > What am I doing wrong?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100201/b5e769a5/attachment-0001.html 

From scottferri09 at gmail.com  Mon Feb  1 11:22:52 2010
From: scottferri09 at gmail.com (Scott Fernandez)
Date: Tue, 2 Feb 2010 00:52:52 +0530
Subject: [Freeswitch-users] Limit the extension creation
Message-ID: <df97b9c51002011122t29c505adu60c4ccb73746432c@mail.gmail.com>

Hi,

Is there a way to restrict the number of extension that FS supports/serves?.
The idea is to limit the concurrent usage of the system for which we need to
restrict the FS to support upto a predefined no. of users/extensions.

Can anyone assist please?

Thanks,
Scott.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100202/2b0c01de/attachment.html 

From frank at carmickle.com  Mon Feb  1 11:34:59 2010
From: frank at carmickle.com (Frank Carmickle)
Date: Mon, 1 Feb 2010 14:34:59 -0500
Subject: [Freeswitch-users] Outbound Call (No Registration)
In-Reply-To: <032301caa367$307c6c60$91754520$@com>
References: <1254B3C5-D7D3-413D-BA82-54FE4789B360@freeswitch.org>
	<022701caa04a$44f60b80$cee22280$@com>
	<697C7F93-3737-4ABD-8934-6A48DC09C088@freeswitch.org>
	<025701caa0a7$e1ca6200$a55f2600$@com>
	<02b001caa10c$913b9100$b3b2b300$@com>
	<F03237B7-89EE-4CE4-A67F-FCB8E8D26700@freeswitch.org>
	<02c501caa113$105532b0$30ff9810$@com>
	<87f2f3b91001291110l226ce6d8u6e806d0a1b782c5f@mail.gmail.com>
	<02dd01caa127$428aa760$c79ff620$@com>
	<032301caa367$307c6c60$91754520$@com>
Message-ID: <20100201193459.GI27405@base.carmickle.com>

On Mon, Feb 01, RR wrote:
> Hi Folks,
> 
>  
> 
> As I continue to learn configuring FS, I am trying to use FS as a peering
> switch sending a call to our SBC on the other side of the pond (across the
> pacific) where the IP address of the FS (in the US) is configured to receive
> traffic from. In this case no registration is required by the sending
> gateway to make calls through the system overseas as its IP address is in
> the "known" gateways (ACL) list. 
> 

I allowed calls in to my pbx from a specific ip in the dialplan with out an acl or a gateway by putting a statement like

  <extension name="fjc-pbx-inbound">
  <condition field="network_addr" expression="^2001\:470\:XXXX\:XXXX\:X:X\:X\:X$"/>
    <condition field="destination_number" expression="^(.*)$">
      <action application="transfer" data="$1 xml default"/>
    </condition>
  </extension>

in the public section of the dialplan.  In this example you see that it is a ipv6 address.  This could easily be replaced with a ipv4 address like the level3 address 4.3.2.1.

There are other ways to do this with a gateway if you so choose.

--FC


From ranjtech at gmail.com  Mon Feb  1 12:24:17 2010
From: ranjtech at gmail.com (RR)
Date: Mon, 1 Feb 2010 15:24:17 -0500
Subject: [Freeswitch-users] Outbound Call (No Registration)
In-Reply-To: <20100201193459.GI27405@base.carmickle.com>
References: <1254B3C5-D7D3-413D-BA82-54FE4789B360@freeswitch.org>
	<022701caa04a$44f60b80$cee22280$@com>
	<697C7F93-3737-4ABD-8934-6A48DC09C088@freeswitch.org>
	<025701caa0a7$e1ca6200$a55f2600$@com>
	<02b001caa10c$913b9100$b3b2b300$@com>
	<F03237B7-89EE-4CE4-A67F-FCB8E8D26700@freeswitch.org>
	<02c501caa113$105532b0$30ff9810$@com>
	<87f2f3b91001291110l226ce6d8u6e806d0a1b782c5f@mail.gmail.com>
	<02dd01caa127$428aa760$c79ff620$@com>
	<032301caa367$307c6c60$91754520$@com>
	<20100201193459.GI27405@base.carmickle.com>
Message-ID: <033e01caa37c$87cd2080$97676180$@com>

Hi Frank,

Thanks for the response. The remote gateway is not running FreeSWITCH. But
that's ok, I figured out the problem. I was capturing packets from the wrong
IP address so after adding the username/password, the call did manage to get
to the other end (after I realized the source IP was wrong) and realized
that my dialplan at the other side wasn't setup correctly. Should be able to
fix that. 

BTW, before I go hunting in the Wiki, can you off the top of your head tell
me how I can manipulate numbers in the dialplan?

If my destination number = 00614xxxxxxxx but I want to send 120#614xxxxxxx
instead, how would I do that?

Thanks
\R

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Frank
Carmickle
Sent: Monday, February 01, 2010 2:35 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Outbound Call (No Registration)

On Mon, Feb 01, RR wrote:
> Hi Folks,
> 
>  
> 
> As I continue to learn configuring FS, I am trying to use FS as a peering
> switch sending a call to our SBC on the other side of the pond (across the
> pacific) where the IP address of the FS (in the US) is configured to
receive
> traffic from. In this case no registration is required by the sending
> gateway to make calls through the system overseas as its IP address is in
> the "known" gateways (ACL) list. 
> 

I allowed calls in to my pbx from a specific ip in the dialplan with out an
acl or a gateway by putting a statement like

  <extension name="fjc-pbx-inbound">
  <condition field="network_addr"
expression="^2001\:470\:XXXX\:XXXX\:X:X\:X\:X$"/>
    <condition field="destination_number" expression="^(.*)$">
      <action application="transfer" data="$1 xml default"/>
    </condition>
  </extension>

in the public section of the dialplan.  In this example you see that it is a
ipv6 address.  This could easily be replaced with a ipv4 address like the
level3 address 4.3.2.1.

There are other ways to do this with a gateway if you so choose.

--FC

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
 

__________ Information from ESET NOD32 Antivirus, version of virus signature
database 4824 (20100201) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com
 
 

__________ Information from ESET NOD32 Antivirus, version of virus signature
database 4825 (20100201) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com
 



From mouncifbb at gmail.com  Mon Feb  1 12:25:57 2010
From: mouncifbb at gmail.com (Mouncif Benniane)
Date: Mon, 1 Feb 2010 15:25:57 -0500
Subject: [Freeswitch-users] mod_lcr problem
In-Reply-To: <b39273531001311507hb1bfe02m8d370443b882ef81@mail.gmail.com>
References: <ade741d61001291443l6134a3achf28be64fc7746c8e@mail.gmail.com>
	<b39273531001300702r1a81f094pac0bd20aeb5ceed6@mail.gmail.com>
	<ade741d61001301338x8014e86t9c9094403f3361f@mail.gmail.com>
	<b39273531001301559h60668746kd5c63ada240513f1@mail.gmail.com>
	<ade741d61001301623w70d7e449i44a9f1e9fb7beae4@mail.gmail.com>
	<b39273531001301645w10446308o3578479dcbf2da2@mail.gmail.com>
	<ade741d61001302057w6ce72355iee6e8b4a2872fc81@mail.gmail.com>
	<b39273531001310532y1de53609rf474755cec04f218@mail.gmail.com>
	<9AB8B620-5A69-4F08-B62C-FFEB03FC6762@gmail.com>
	<b39273531001311507hb1bfe02m8d370443b882ef81@mail.gmail.com>
Message-ID: <ade741d61002011225t15d66333v69f31b0572880de9@mail.gmail.com>

So I have to alter my LCR table to look like:

NPANXX,"LATA","OCN","NTER","INTRA"  1201007,"224","7229","0.0059","0.0127"
1201040,"224","9206","0.0036","0.0036"


instead of:


NPANXX,"LATA","OCN","NTER","INTRA"  201007,"224","7229","0.0059","0.0127"
201040,"224","9206","0.0036","0.0036"


On Sun, Jan 31, 2010 at 6:07 PM, Rupa Schomaker <rupa at rupa.com> wrote:

> Yes, you need to normalize the values passed to lcr.  Otherwise, how could
> it work?
>
> You can normalize the CID by matching and adding a 1 for 10 digit #s, or
> removing the leading + or other things you might need then setting it back
> to the profile using the set_profile_var app (
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_set_profile_var).
>  (mod_cidlookup will set it after doing a #->name/area lookup - but for now
> you can set it yourself)
>
> You can normalize the DID by doing similar matching rules as above and then
> transfering to that normalized DID for the rest of your call plan
> processing.
>
> I'm pretty sure mod_cidlookup has an example of normalizing... yeah:
> http://wiki.freeswitch.org/wiki/Mod_cidlookup#Dialplan_Application
>
> On Sun, Jan 31, 2010 at 4:18 PM, Mouncifbb <mouncifbb at gmail.com> wrote:
>
>> So the CID must have 1 at front also? Usually people
>> Send only npa and nxx ex 6176427788 7817612233
>> Do I need to alter it?
>>
>> Sent from my iPhone
>>
>> On Jan 31, 2010, at 8:32 AM, Rupa Schomaker <rupa at rupa.com> wrote:
>>
>>
>>
>> On Sat, Jan 30, 2010 at 10:57 PM, Mouncif Benniane <<mouncifbb at gmail.com>
>> mouncifbb at gmail.com> wrote:
>>
>>> OK going back to use default profile to keep things simple below 2
>>> results
>>>
>>> Using:
>>>
>>> lcr 16179470890  default  19785223241  ( this one consult
>>> npa_nxx_company_ocn)
>>>
>>> lcr 6179470890  default  9785223241 ( this one don't!! )
>>>
>>>
>>>
>> Oh, right!  mod_lcr really expects you to normalize your prefix to e164
>> format.  I thought there was discussion about this in the wiki, but maybe
>> not.  For simple prefix matching it doesn't matter, but for things that make
>> decisions based on the # (like the lata/state stuff) it does.
>>
>> npanxx lookup only makes sense for NANPA numbers.  NANPA numbers have a
>> country code of "1" and a total length of 11 (including the 1).
>>
>> This is the only rational way to do it when you have a rate table with
>> both domestic (NANPA) and international prefixes.
>>
>>
>>> freeswitch> lcr 16179470890  default  19785223241
>>> 2010-01-30 23:53:34.681842 [DEBUG] mod_lcr.c:1329 data passed to lcr is
>>> [16179470890  default  19785223241]
>>> 2010-01-30 23:53:34.681842 [DEBUG] mod_lcr.c:1365 Set Caller ID to
>>> [19785223241]
>>> 2010-01-30 23:53:34.681842 [DEBUG] mod_lcr.c:736 SQL: SELECT 'state',
>>> count(DISTINCT state) FROM npa_nxx_company_ocn WHERE (npa=617 AND nxx=947)
>>> OR (npa=978 AND nxx=522) UNION SELECT 'lata', count(DISTINCT lata) FROM
>>> npa_nxx_company_ocn WHERE (npa=617 AND nxx=947) OR (npa=978 AND nxx=522)
>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:696 Type: state, Count: 1
>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:696 Type: lata, Count: 1
>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:786 intra routing [state:1
>>> lata:1] so rate field is [intralata_rate]
>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:802 we have an event
>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:826 SQL: SELECT l.digits,
>>> c.carrier_name, l.intralata_rate, cg.prefix AS gw_prefix, cg.suffix AS
>>> gw_suffix, l.lead_strip, l.trail_strip, l.prefix, l.suffix , cg.codec ,
>>> l.cid FROM lcr l JOIN carriers c ON l.carrier_id=c.id JOIN
>>> carrier_gateway cg ON c.id=cg.carrier_id WHERE c.enabled = '1' AND
>>> cg.enabled = '1' AND l.enabled = '1' AND digits IN (16179470890, 1617947089,
>>> 161794708, 16179470, 1617947, 161794, 16179, 1617, 161, 16, 1) AND
>>> CURRENT_TIMESTAMP BETWEEN date_start AND date_end ORDER BY digits DESC,
>>> intralata_rate, rand();
>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:307 Returning Dialstring
>>> [lcr_carrier=carrier1,lcr_rate=0.00000]sofia/gateway/carrier1/16179470890
>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:615 Adding carrier1 to head
>>> of list
>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:307 Returning Dialstring
>>> [lcr_carrier=carrier2,lcr_rate=0.00000]sofia/external/
>>> 06179470890 at proxy.carrier2.net:5060
>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:667 adding carrier2 to end
>>> of list after carrier1
>>>
>>>
>>>  | Digit Match | Carrier  | Rate     | Codec | CID Regexp |
>>> Dialstring
>>> |
>>>  | 1           | carrier1 | 0.00000  |       |            |
>>> [lcr_carrier=carrier1,lcr_rate=0.00000]sofia/gateway/carrier1/16179470890
>>> |
>>>  | 1           | carrier2 | 0.00000  |       |            |
>>> [lcr_carrier=carrier2,lcr_rate=0.00000]sofia/external/
>>> 06179470890 at proxy.carrier2.net:5060 |
>>>
>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:307 Returning Dialstring
>>> [lcr_carrier=carrier1,lcr_rate=0.00000]sofia/gateway/carrier1/16179470890
>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:307 Returning Dialstring
>>> [lcr_carrier=carrier2,lcr_rate=0.00000]sofia/external/
>>> 06179470890 at proxy.carrier2.net:5060
>>>
>>>
>>>
>>>
>>>
>>> freeswitch> lcr 6179470890  default  9785223241
>>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:1329 data passed to lcr is
>>> [6179470890  default  9785223241]
>>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:1365 Set Caller ID to
>>> [9785223241]
>>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:786 intra routing [state:0
>>> lata:0] so rate field is [rate]
>>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:802 we have an event
>>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:826 SQL: SELECT l.digits,
>>> c.carrier_name, l.rate, cg.prefix AS gw_prefix, cg.suffix AS gw_suffix,
>>> l.lead_strip, l.trail_strip, l.prefix, l.suffix , cg.codec , l.cid FROM lcr
>>> l JOIN carriers c ON l.carrier_id=c.id JOIN carrier_gateway cg ON c.id=cg.carrier_id
>>> WHERE c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1' AND digits IN
>>> (6179470890, 617947089, 61794708, 6179470, 617947, 61794, 6179, 617, 61, 6)
>>> AND CURRENT_TIMESTAMP BETWEEN date_start AND date_end ORDER BY digits DESC,
>>> rate, rand();
>>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:307 Returning Dialstring
>>> [lcr_carrier=carrier1,lcr_rate=0.09000]sofia/gateway/carrier1/16179470890
>>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:615 Adding carrier1 to head
>>> of list
>>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:307 Returning Dialstring
>>> [lcr_carrier=carrier1,lcr_rate=0.09000]sofia/gateway/carrier1/16179470890
>>>
>>>
>>>  | Digit Match | Carrier  | Rate     | Codec | CID Regexp |
>>> Dialstring                                                                |
>>>   | 617947      | carrier1 | 0.09000  |       |            |
>>> [lcr_carrier=carrier1,lcr_rate=0.09000]sofia/gateway/carrier1/16179470890 |
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>> On Sat, Jan 30, 2010 at 7:45 PM, Rupa Schomaker < <rupa at rupa.com>
>>> rupa at rupa.com> wrote:
>>>
>>>> turn up logging to debug again, and then reload mod_lcr.  It'll spit out
>>>> a bunch of crap when it tests out each profile you have defined.  Give me
>>>> the full log (here or in <http://pastebin.freeswitch.org>
>>>> pastebin.freeswitch.org).  That may show more useful info as to why
>>>> things are mucked up?
>>>>
>>>>
>>>> On Sat, Jan 30, 2010 at 6:23 PM, Mouncif Benniane <<mouncifbb at gmail.com>
>>>> mouncifbb at gmail.com> wrote:
>>>>
>>>>> yes I use NANPA_STD profile instead of default cause I thought the
>>>>> custom profile was causing issues, but looks like it's returning same
>>>>> results.
>>>>>
>>>>> There is this line in thw wiki:
>>>>> intra lata/state selection is done manually by setting the channel
>>>>> variables *intrastate* or *intralata* to the value *true*.
>>>>>
>>>>> do I have to set these ? if yes how?
>>>>>
>>>>> Thanks
>>>>>
>>>>>
>>>>> On Sat, Jan 30, 2010 at 6:59 PM, Rupa Schomaker < <rupa at rupa.com>
>>>>> rupa at rupa.com> wrote:
>>>>>
>>>>>> Stuff inline.
>>>>>>
>>>>>> On Sat, Jan 30, 2010 at 3:38 PM, Mouncif Benniane <<mouncifbb at gmail.com>
>>>>>> mouncifbb at gmail.com> wrote:
>>>>>>
>>>>>>>  NPANXX,"LATA","OCN","NTER","INTRA"
>>>>>>> 201007,"224","7229","0.0059","0.0127"
>>>>>>> 201040,"224","9206","0.0036","0.0036"
>>>>>>>
>>>>>>
>>>>>> Looks like they give you the LATA and OCN values with the prefix.  We
>>>>>> (should) look that up ourselves.
>>>>>>
>>>>>>
>>>>>>> FreeSWITCH Version 1.0.trunk (16540)
>>>>>>>
>>>>>>>
>>>>>>> Also I noticed the *npa_nxx_ocn* table never get consulted.
>>>>>>>
>>>>>>> I also see this now when making a real call instead of running
>>>>>>> thorugh CLI
>>>>>>>
>>>>>>> EXECUTE sofia/external/6179472456 at 174.x.x.x lcr(16179470890
>>>>>>> NANPA_STD)
>>>>>>> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:1230 *intrastate
>>>>>>> channel var is [undef]*
>>>>>>
>>>>>>
>>>>>> This is fine.  it is a leftover from when you would tell mod_lcr via a
>>>>>> channel var that it should do intrastate.  I later had mod_lcr do the lookup
>>>>>> itself, but we still honor the old var.  There are no channel vars
>>>>>> associated with the cli, so you wouldn't see that msg.
>>>>>>
>>>>>>
>>>>>>>
>>>>>>> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:1233 Select routes based
>>>>>>> on interstate rates
>>>>>>> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:1252 LCR Lookup on
>>>>>>> 16179470893 using profile NANPA_STD
>>>>>>> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:786 intra routing
>>>>>>> [state:0 lata:0] so rate field is [rate]
>>>>>>>
>>>>>>> called number 6179470890  caller ID: 6179472456
>>>>>>>
>>>>>>> any ideas??
>>>>>>>
>>>>>>>
>>>>>> Only thing that jumps out at me.
>>>>>>
>>>>>> The output from lcr_admin show profiles showed only the default one.
>>>>>>  On the dialplan you use the NANPA_STD profile.  Can you check lcr_admin
>>>>>> list and see if that profile is defined and if so if it says it is using the
>>>>>> npanxx table?
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> On Sat, Jan 30, 2010 at 10:02 AM, Rupa Schomaker < <rupa at rupa.com>
>>>>>>> rupa at rupa.com> wrote:
>>>>>>>
>>>>>>>> Something is still missing from the logs. Note the query of the
>>>>>>>> npanxx
>>>>>>>> table, the flags being set, and the rate field being chosen.  Umm..
>>>>>>>> oh, what version of fs are you running?
>>>>>>>>
>>>>>>>> Yes, the npa_nxx_ocn table needs to be loaded up as described in:
>>>>>>>> <http://wiki.freeswitch.org/wiki/Mod_cidlookup#Falling_back_to_.22City_State.22_in_the_absense_of_a_name>
>>>>>>>> http://wiki.freeswitch.org/wiki/Mod_cidlookup#Falling_back_to_.22City_State.22_in_the_absense_of_a_name
>>>>>>>> (there is a link to that from mod_lcr's wiki page).
>>>>>>>>
>>>>>>>> An example from my own setup:
>>>>>>>>
>>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:1384 data passed to lcr
>>>>>>>> is [12148267711 default 12148267712]
>>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:1420 Set Caller ID to
>>>>>>>> [12148267712]
>>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:759 SQL: SELECT
>>>>>>>> 'state',
>>>>>>>> count(DISTINCT state) FROM npa_nxx_company_ocn WHERE (npa=214 AND
>>>>>>>> nxx=826) OR (npa=214 AND nxx=826) UNION SELECT 'lata',
>>>>>>>> count(DISTINCT
>>>>>>>> lata) FROM npa_nxx_company_ocn WHERE (npa=214 AND nxx=826) OR
>>>>>>>> (npa=214
>>>>>>>> AND nxx=826)
>>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:719 Type: lata, Count:
>>>>>>>> 1
>>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:719 Type: state, Count:
>>>>>>>> 1
>>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:809 intra routing
>>>>>>>> [state:1 lata:1] so rate field is [intralata_rate]
>>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:825 we have an event
>>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:849 SQL:  SELECT
>>>>>>>> l.digits
>>>>>>>> AS lcr_digits,         c.carrier_name AS lcr_carrier_name,
>>>>>>>> l.intralata_rate as lcr_rate_field,        cg.prefix AS
>>>>>>>> lcr_gw_prefix,
>>>>>>>> cg.suffix AS lcr_gw_suffix,        l.lead_strip AS lcr_lead_strip,
>>>>>>>> l.trail_strip AS lcr_trail_strip,        l.prefix AS lcr_prefix,
>>>>>>>> l.suffix AS lcr_suffix,        cg.codec AS lcr_codec, l.cid AS
>>>>>>>> lcr_cid,        'carriers' AS lcr_limit_realm, c.carrier_name AS
>>>>>>>> lcr_limit_id, 5 AS lcr_limit_max   FROM lcr l      JOIN carriers c
>>>>>>>> ON
>>>>>>>> l.carrier_id=c.id      JOIN carrier_gateway cg ON c.id
>>>>>>>> =cg.carrier_id
>>>>>>>>  WHERE c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1'
>>>>>>>> AND digits_prefix @> '12148267711'      AND CURRENT_TIMESTAMP
>>>>>>>> BETWEEN
>>>>>>>> date_start AND date_end   ORDER BY digits DESC, intralata_rate,
>>>>>>>> random();
>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning
>>>>>>>> Dialstring
>>>>>>>>
>>>>>>>> [lcr_carrier=grnvoip_us,lcr_rate=0.00591]sofia/gateway/grnvoip/XXXX12148267711
>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:638 Adding grnvoip_us
>>>>>>>> to
>>>>>>>> head of list
>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning
>>>>>>>> Dialstring
>>>>>>>>
>>>>>>>> [lcr_carrier=teliax,lcr_rate=0.01000]sofia/gateway/teliax_atl/12148267711
>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:660 Adding teliax to
>>>>>>>> end of list
>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning
>>>>>>>> Dialstring
>>>>>>>>
>>>>>>>> [lcr_carrier=teliax,lcr_rate=0.01000]sofia/gateway/teliax/12148267711
>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:660 Adding teliax to
>>>>>>>> end of list
>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning
>>>>>>>> Dialstring
>>>>>>>>
>>>>>>>> [lcr_carrier=vitelity,lcr_rate=0.01440]sofia/gateway/vitelity/12148267711
>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:660 Adding vitelity to
>>>>>>>> end of list
>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning
>>>>>>>> Dialstring
>>>>>>>>
>>>>>>>> [lcr_carrier=grnvoip_std,lcr_rate=0.01500]sofia/gateway/grnvoip/YYYY12148267711
>>>>>>>> [...]
>>>>>>>>
>>>>>>>> On Fri, Jan 29, 2010 at 10:42 PM, Mouncif Benniane <<mouncifbb at gmail.com>
>>>>>>>> mouncifbb at gmail.com> wrote:
>>>>>>>> > Also the Provider has presented the rates in this format?
>>>>>>>> > NPANXXLATA OCN INTER INTRA
>>>>>>>> >
>>>>>>>> >
>>>>>>>> > On Fri, Jan 29, 2010 at 11:30 PM, Mouncif Benniane <<mouncifbb at gmail.com>
>>>>>>>> mouncifbb at gmail.com>
>>>>>>>> > wrote:
>>>>>>>> >>
>>>>>>>> >> Tried it and it's not giving me intralata  instead I get
>>>>>>>> interstate, does
>>>>>>>> >> the npa_nxx_company_ocn table needs to be used in this case?,
>>>>>>>> also do I have
>>>>>>>> >> to have the rate field in lcr table?
>>>>>>>> >>
>>>>>>>> >> lcr 617642  default 6176421212
>>>>>>>> >>  | Digit Match | Carrier  | Rate     | Codec | CID Regexp |
>>>>>>>> Dialstring
>>>>>>>> >>                                                        |
>>>>>>>> >>  | 617642      | carrier1 | 0.00500  |       |            |
>>>>>>>> >>
>>>>>>>> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642 |
>>>>>>>> >>
>>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:1329 data passed to
>>>>>>>> lcr is
>>>>>>>> >> [617642  default 6176421212]
>>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:1365 Set Caller ID
>>>>>>>> to
>>>>>>>> >> [6176421212]
>>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:786 intra routing
>>>>>>>> [state:0
>>>>>>>> >> lata:0] so rate field is [rate]
>>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:802 we have an event
>>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:826 SQL: SELECT
>>>>>>>> l.digits,
>>>>>>>> >> c.carrier_name, l.rate, cg.prefix AS gw_prefix, cg.suffix AS
>>>>>>>> gw_suffix,
>>>>>>>> >> l.lead_strip, l.trail_strip, l.prefix, l.suffix , cg.codec ,
>>>>>>>> l.cid FROM lcr
>>>>>>>> >> l JOIN carriers c ON l.carrier_id=c.id JOIN carrier_gateway cg
>>>>>>>> ON
>>>>>>>> >> c.id=cg.carrier_id WHERE c.enabled = '1' AND cg.enabled = '1'
>>>>>>>> AND l.enabled
>>>>>>>> >> = '1' AND digits IN (617642, 61764, 6176, 617, 61, 6) AND
>>>>>>>> CURRENT_TIMESTAMP
>>>>>>>> >> BETWEEN date_start AND date_end ORDER BY digits DESC,  rate,
>>>>>>>> rand();
>>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:307 Returning
>>>>>>>> Dialstring
>>>>>>>> >>
>>>>>>>> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642
>>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:615 Adding carrier1
>>>>>>>> to head
>>>>>>>> >> of list
>>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:307 Returning
>>>>>>>> Dialstring
>>>>>>>> >>
>>>>>>>> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642
>>>>>>>> >>
>>>>>>>> >> Thank you Rupa!
>>>>>>>> >>
>>>>>>>> >> On Fri, Jan 29, 2010 at 7:37 PM, Rupa Schomaker <<rupa at rupa.com>
>>>>>>>> rupa at rupa.com> wrote:
>>>>>>>> >>>
>>>>>>>> >>> turn console logging up to debug and redo the lcr lookup.  The
>>>>>>>> sql
>>>>>>>> >>> statements along with status info will show up.  This should
>>>>>>>> give
>>>>>>>> >>> enough information to debug what is happening.
>>>>>>>> >>>
>>>>>>>> >>> I'm assuming the npanxx table is actually populated and not just
>>>>>>>> >>> existing?
>>>>>>>> >>>
>>>>>>>> >>> When doing the lookup from the cli you have to tell lcr what CID
>>>>>>>> to
>>>>>>>> >>> use (remember, it is relative to the src/dest number).  I'm
>>>>>>>> pretty
>>>>>>>> >>> sure you get something on the console log when you don't specify
>>>>>>>> a CID
>>>>>>>> >>> when using the commandline.  Anyway:
>>>>>>>> >>>
>>>>>>>> >>> lcr 617642  default 6176421212
>>>>>>>> >>>
>>>>>>>> >>> should give you intralata.
>>>>>>>> >>>
>>>>>>>> >>> Note that the definition of intralata doesn't mean "local" for
>>>>>>>> some
>>>>>>>> >>> providers.  Some providers define local to "same ratecenter"
>>>>>>>> which is
>>>>>>>> >>> even more restrictive.
>>>>>>>> >>>
>>>>>>>> >>> On Fri, Jan 29, 2010 at 4:43 PM, Mouncif Benniane <<mouncifbb at gmail.com>
>>>>>>>> mouncifbb at gmail.com>
>>>>>>>> >>> wrote:
>>>>>>>> >>> > i can't make use of mod_lcr using Intra/Interstate rating, I
>>>>>>>> am using
>>>>>>>> >>> > svn: FreeSWITCH Version 1.0.trunk (16517)
>>>>>>>> >>> >
>>>>>>>> >>> > lcr mysql table structure:
>>>>>>>> >>> > CREATE TABLE `lcr` (
>>>>>>>> >>> >   `id` INT(11) NOT NULL AUTO_INCREMENT,
>>>>>>>> >>> >   `digits` VARCHAR(15) DEFAULT NULL,
>>>>>>>> >>> >   `rate` FLOAT(11,5) DEFAULT NULL,
>>>>>>>> >>> >   `intrastate_rate` FLOAT(11,5) DEFAULT NULL,
>>>>>>>> >>> >   `intralata_rate` FLOAT(11,5) DEFAULT NULL,
>>>>>>>> >>> >   `carrier_id` INT(11) NOT NULL,
>>>>>>>> >>> >   `lead_strip` INT(11) NOT NULL,
>>>>>>>> >>> >   `trail_strip` INT(11) NOT NULL,
>>>>>>>> >>> >   `prefix` VARCHAR(16) NOT NULL,
>>>>>>>> >>> >   `suffix` VARCHAR(16) NOT NULL,
>>>>>>>> >>> >   `lcr_profile` VARCHAR(32) DEFAULT NULL,
>>>>>>>> >>> >   `date_start` DATETIME NOT NULL DEFAULT '1970-01-01
>>>>>>>> 00:00:00',
>>>>>>>> >>> >   `date_end` DATETIME NOT NULL DEFAULT '2030-12-31 00:00:00',
>>>>>>>> >>> >   `quality` FLOAT(10,6) NOT NULL,
>>>>>>>> >>> >   `reliability` FLOAT(10,6) NOT NULL,
>>>>>>>> >>> >   `cid` VARCHAR(32) NOT NULL DEFAULT '',
>>>>>>>> >>> >   `enabled` TINYINT(1) NOT NULL DEFAULT '1',
>>>>>>>> >>> >   PRIMARY KEY  (`id`),
>>>>>>>> >>> >   KEY `carrier_id` (`carrier_id`),
>>>>>>>> >>> >   KEY `digits` (`digits`),
>>>>>>>> >>> >   KEY `lcr_profile` (`lcr_profile`),
>>>>>>>> >>> >   KEY `digits_profile_cid_rate` USING BTREE (`digits`),
>>>>>>>> >>> >   CONSTRAINT `carrier_id` FOREIGN KEY (`carrier_id`)
>>>>>>>> REFERENCES
>>>>>>>> >>> > `carriers`
>>>>>>>> >>> > (`id`) ON DELETE CASCADE ON UPDATE CASCADE
>>>>>>>> >>> > ) ENGINE=INNODB AUTO_INCREMENT=6 DEFAULT CHARSET=latin1
>>>>>>>> >>> >
>>>>>>>> >>> >
>>>>>>>> >>> > lcr_admin show profiles
>>>>>>>> >>> > Name:           default
>>>>>>>> >>> >  custom sql:    SELECT l.digits, c.carrier_name,
>>>>>>>> l.${lcr_rate_field},
>>>>>>>> >>> > cg.prefix AS gw_prefix, cg.suffix AS gw_suffix, l.lead_strip,
>>>>>>>> >>> > l.trail_strip,
>>>>>>>> >>> > l.prefix, l.suffix , cg.codec , l.cid FROM lcr l JOIN carriers
>>>>>>>> c ON
>>>>>>>> >>> > l.carrier_id=c.id JOIN carrier_gateway cg ON c.id=cg.carrier_id
>>>>>>>> WHERE
>>>>>>>> >>> > c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1' AND
>>>>>>>> digits IN
>>>>>>>> >>> > (${lcr_query_expanded_digits}) AND CURRENT_TIMESTAMP BETWEEN
>>>>>>>> date_start
>>>>>>>> >>> > AND
>>>>>>>> >>> > date_end ORDER BY digits DESC,  ${lcr_rate_field},  quality
>>>>>>>> DESC,
>>>>>>>> >>> >  reliability DESC, rand();
>>>>>>>> >>> >  has %:         false
>>>>>>>> >>> >  has vars:      true
>>>>>>>> >>> >  has intrastate:        true
>>>>>>>> >>> >  has intralata: true
>>>>>>>> >>> >  has npanxx:    true
>>>>>>>> >>> >  Reorder rate:  enabled
>>>>>>>> >>> >  Info in headers:       disabled
>>>>>>>> >>> >  Quote IN() List:       disabled
>>>>>>>> >>> >
>>>>>>>> >>> >
>>>>>>>> >>> >
>>>>>>>> >>> > lcr 617642  default  returns rate from the rate field table
>>>>>>>> and not
>>>>>>>> >>> > intra/inter state fields rates.
>>>>>>>> >>> >
>>>>>>>> >>> > Any ideas? thanks!
>>>>>>>> >>> >
>>>>>>>> >>> >
>>>>>>>> >>> >
>>>>>>>> >>> >
>>>>>>>> >>> > _______________________________________________
>>>>>>>> >>> > FreeSWITCH-users mailing list
>>>>>>>> >>> > <FreeSWITCH-users at lists.freeswitch.org>
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> >>> >
>>>>>>>> <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> >>> >
>>>>>>>> >>> > UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> >>> > <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>>> >>> >
>>>>>>>> >>> >
>>>>>>>> >>>
>>>>>>>> >>>
>>>>>>>> >>>
>>>>>>>> >>> --
>>>>>>>> >>> -Rupa
>>>>>>>> >>>
>>>>>>>> >>> _______________________________________________
>>>>>>>> >>> FreeSWITCH-users mailing list
>>>>>>>> >>> <FreeSWITCH-users at lists.freeswitch.org>
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> >>> <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> >>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> >>> <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>>> >>
>>>>>>>> >
>>>>>>>> >
>>>>>>>> > _______________________________________________
>>>>>>>> > FreeSWITCH-users mailing list
>>>>>>>> > <FreeSWITCH-users at lists.freeswitch.org>
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> > <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> > UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> > <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>>> >
>>>>>>>> >
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> --
>>>>>>>> -Rupa
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>> -Rupa
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>>
>>>> --
>>>> -Rupa
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>> FreeSWITCH-users at lists.freeswitch.org
>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> -Rupa
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100201/773e798f/attachment-0001.html 

From paul.gore.j at gmail.com  Mon Feb  1 13:07:26 2010
From: paul.gore.j at gmail.com (paul gore)
Date: Mon, 1 Feb 2010 16:07:26 -0500
Subject: [Freeswitch-users] Logging question
Message-ID: <da4ab26e1002011307n6b10e97tea3e281cbff90e13@mail.gmail.com>

I tried "sofia profile internal siptrace on" from console and got SIP trace
on the console,
but nothing in logs still.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100201/47e22d6f/attachment.html 

From rupa at rupa.com  Mon Feb  1 14:39:03 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Mon, 1 Feb 2010 16:39:03 -0600
Subject: [Freeswitch-users] mod_lcr problem
In-Reply-To: <ade741d61002011225t15d66333v69f31b0572880de9@mail.gmail.com>
References: <ade741d61001291443l6134a3achf28be64fc7746c8e@mail.gmail.com>
	<ade741d61001301338x8014e86t9c9094403f3361f@mail.gmail.com>
	<b39273531001301559h60668746kd5c63ada240513f1@mail.gmail.com>
	<ade741d61001301623w70d7e449i44a9f1e9fb7beae4@mail.gmail.com>
	<b39273531001301645w10446308o3578479dcbf2da2@mail.gmail.com>
	<ade741d61001302057w6ce72355iee6e8b4a2872fc81@mail.gmail.com>
	<b39273531001310532y1de53609rf474755cec04f218@mail.gmail.com>
	<9AB8B620-5A69-4F08-B62C-FFEB03FC6762@gmail.com>
	<b39273531001311507hb1bfe02m8d370443b882ef81@mail.gmail.com>
	<ade741d61002011225t15d66333v69f31b0572880de9@mail.gmail.com>
Message-ID: <b39273531002011439h2a2ce74ahca048432579129c1@mail.gmail.com>

yes, otherwise you'll have issues when you load your international rates in
the same table.

On Mon, Feb 1, 2010 at 2:25 PM, Mouncif Benniane <mouncifbb at gmail.com>wrote:

> So I have to alter my LCR table to look like:
>
> NPANXX,"LATA","OCN","NTER","INTRA"  1201007,"224","7229","0.0059","0.0127"
> 1201040,"224","9206","0.0036","0.0036"
>
>
> instead of:
>
>
>
> NPANXX,"LATA","OCN","NTER","INTRA"  201007,"224","7229","0.0059","0.0127"
> 201040,"224","9206","0.0036","0.0036"
>
>
> On Sun, Jan 31, 2010 at 6:07 PM, Rupa Schomaker <rupa at rupa.com> wrote:
>
>> Yes, you need to normalize the values passed to lcr.  Otherwise, how could
>> it work?
>>
>> You can normalize the CID by matching and adding a 1 for 10 digit #s, or
>> removing the leading + or other things you might need then setting it back
>> to the profile using the set_profile_var app (
>> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_set_profile_var).
>>  (mod_cidlookup will set it after doing a #->name/area lookup - but for now
>> you can set it yourself)
>>
>> You can normalize the DID by doing similar matching rules as above and
>> then transfering to that normalized DID for the rest of your call plan
>> processing.
>>
>> I'm pretty sure mod_cidlookup has an example of normalizing... yeah:
>> http://wiki.freeswitch.org/wiki/Mod_cidlookup#Dialplan_Application
>>
>> On Sun, Jan 31, 2010 at 4:18 PM, Mouncifbb <mouncifbb at gmail.com> wrote:
>>
>>> So the CID must have 1 at front also? Usually people
>>> Send only npa and nxx ex 6176427788 7817612233
>>> Do I need to alter it?
>>>
>>> Sent from my iPhone
>>>
>>> On Jan 31, 2010, at 8:32 AM, Rupa Schomaker <rupa at rupa.com> wrote:
>>>
>>>
>>>
>>> On Sat, Jan 30, 2010 at 10:57 PM, Mouncif Benniane <<mouncifbb at gmail.com>
>>> mouncifbb at gmail.com> wrote:
>>>
>>>> OK going back to use default profile to keep things simple below 2
>>>> results
>>>>
>>>> Using:
>>>>
>>>> lcr 16179470890  default  19785223241  ( this one consult
>>>> npa_nxx_company_ocn)
>>>>
>>>> lcr 6179470890  default  9785223241 ( this one don't!! )
>>>>
>>>>
>>>>
>>> Oh, right!  mod_lcr really expects you to normalize your prefix to e164
>>> format.  I thought there was discussion about this in the wiki, but maybe
>>> not.  For simple prefix matching it doesn't matter, but for things that make
>>> decisions based on the # (like the lata/state stuff) it does.
>>>
>>> npanxx lookup only makes sense for NANPA numbers.  NANPA numbers have a
>>> country code of "1" and a total length of 11 (including the 1).
>>>
>>> This is the only rational way to do it when you have a rate table with
>>> both domestic (NANPA) and international prefixes.
>>>
>>>
>>>> freeswitch> lcr 16179470890  default  19785223241
>>>> 2010-01-30 23:53:34.681842 [DEBUG] mod_lcr.c:1329 data passed to lcr is
>>>> [16179470890  default  19785223241]
>>>> 2010-01-30 23:53:34.681842 [DEBUG] mod_lcr.c:1365 Set Caller ID to
>>>> [19785223241]
>>>> 2010-01-30 23:53:34.681842 [DEBUG] mod_lcr.c:736 SQL: SELECT 'state',
>>>> count(DISTINCT state) FROM npa_nxx_company_ocn WHERE (npa=617 AND nxx=947)
>>>> OR (npa=978 AND nxx=522) UNION SELECT 'lata', count(DISTINCT lata) FROM
>>>> npa_nxx_company_ocn WHERE (npa=617 AND nxx=947) OR (npa=978 AND nxx=522)
>>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:696 Type: state, Count: 1
>>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:696 Type: lata, Count: 1
>>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:786 intra routing [state:1
>>>> lata:1] so rate field is [intralata_rate]
>>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:802 we have an event
>>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:826 SQL: SELECT l.digits,
>>>> c.carrier_name, l.intralata_rate, cg.prefix AS gw_prefix, cg.suffix AS
>>>> gw_suffix, l.lead_strip, l.trail_strip, l.prefix, l.suffix , cg.codec ,
>>>> l.cid FROM lcr l JOIN carriers c ON l.carrier_id=c.id JOIN
>>>> carrier_gateway cg ON c.id=cg.carrier_id WHERE c.enabled = '1' AND
>>>> cg.enabled = '1' AND l.enabled = '1' AND digits IN (16179470890, 1617947089,
>>>> 161794708, 16179470, 1617947, 161794, 16179, 1617, 161, 16, 1) AND
>>>> CURRENT_TIMESTAMP BETWEEN date_start AND date_end ORDER BY digits DESC,
>>>> intralata_rate, rand();
>>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:307 Returning Dialstring
>>>> [lcr_carrier=carrier1,lcr_rate=0.00000]sofia/gateway/carrier1/16179470890
>>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:615 Adding carrier1 to head
>>>> of list
>>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:307 Returning Dialstring
>>>> [lcr_carrier=carrier2,lcr_rate=0.00000]sofia/external/
>>>> 06179470890 at proxy.carrier2.net:5060
>>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:667 adding carrier2 to end
>>>> of list after carrier1
>>>>
>>>>
>>>>  | Digit Match | Carrier  | Rate     | Codec | CID Regexp |
>>>> Dialstring
>>>> |
>>>>  | 1           | carrier1 | 0.00000  |       |            |
>>>> [lcr_carrier=carrier1,lcr_rate=0.00000]sofia/gateway/carrier1/16179470890
>>>> |
>>>>  | 1           | carrier2 | 0.00000  |       |            |
>>>> [lcr_carrier=carrier2,lcr_rate=0.00000]sofia/external/
>>>> 06179470890 at proxy.carrier2.net:5060 |
>>>>
>>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:307 Returning Dialstring
>>>> [lcr_carrier=carrier1,lcr_rate=0.00000]sofia/gateway/carrier1/16179470890
>>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:307 Returning Dialstring
>>>> [lcr_carrier=carrier2,lcr_rate=0.00000]sofia/external/
>>>> 06179470890 at proxy.carrier2.net:5060
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> freeswitch> lcr 6179470890  default  9785223241
>>>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:1329 data passed to lcr is
>>>> [6179470890  default  9785223241]
>>>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:1365 Set Caller ID to
>>>> [9785223241]
>>>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:786 intra routing [state:0
>>>> lata:0] so rate field is [rate]
>>>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:802 we have an event
>>>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:826 SQL: SELECT l.digits,
>>>> c.carrier_name, l.rate, cg.prefix AS gw_prefix, cg.suffix AS gw_suffix,
>>>> l.lead_strip, l.trail_strip, l.prefix, l.suffix , cg.codec , l.cid FROM lcr
>>>> l JOIN carriers c ON l.carrier_id=c.id JOIN carrier_gateway cg ON c.id=cg.carrier_id
>>>> WHERE c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1' AND digits IN
>>>> (6179470890, 617947089, 61794708, 6179470, 617947, 61794, 6179, 617, 61, 6)
>>>> AND CURRENT_TIMESTAMP BETWEEN date_start AND date_end ORDER BY digits DESC,
>>>> rate, rand();
>>>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:307 Returning Dialstring
>>>> [lcr_carrier=carrier1,lcr_rate=0.09000]sofia/gateway/carrier1/16179470890
>>>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:615 Adding carrier1 to head
>>>> of list
>>>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:307 Returning Dialstring
>>>> [lcr_carrier=carrier1,lcr_rate=0.09000]sofia/gateway/carrier1/16179470890
>>>>
>>>>
>>>>  | Digit Match | Carrier  | Rate     | Codec | CID Regexp |
>>>> Dialstring                                                                |
>>>>   | 617947      | carrier1 | 0.09000  |       |            |
>>>> [lcr_carrier=carrier1,lcr_rate=0.09000]sofia/gateway/carrier1/16179470890 |
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> On Sat, Jan 30, 2010 at 7:45 PM, Rupa Schomaker < <rupa at rupa.com>
>>>> rupa at rupa.com> wrote:
>>>>
>>>>> turn up logging to debug again, and then reload mod_lcr.  It'll spit
>>>>> out a bunch of crap when it tests out each profile you have defined.  Give
>>>>> me the full log (here or in <http://pastebin.freeswitch.org>
>>>>> pastebin.freeswitch.org).  That may show more useful info as to why
>>>>> things are mucked up?
>>>>>
>>>>>
>>>>> On Sat, Jan 30, 2010 at 6:23 PM, Mouncif Benniane <<mouncifbb at gmail.com>
>>>>> mouncifbb at gmail.com> wrote:
>>>>>
>>>>>> yes I use NANPA_STD profile instead of default cause I thought the
>>>>>> custom profile was causing issues, but looks like it's returning same
>>>>>> results.
>>>>>>
>>>>>> There is this line in thw wiki:
>>>>>> intra lata/state selection is done manually by setting the channel
>>>>>> variables *intrastate* or *intralata* to the value *true*.
>>>>>>
>>>>>> do I have to set these ? if yes how?
>>>>>>
>>>>>> Thanks
>>>>>>
>>>>>>
>>>>>> On Sat, Jan 30, 2010 at 6:59 PM, Rupa Schomaker < <rupa at rupa.com>
>>>>>> rupa at rupa.com> wrote:
>>>>>>
>>>>>>> Stuff inline.
>>>>>>>
>>>>>>> On Sat, Jan 30, 2010 at 3:38 PM, Mouncif Benniane <<mouncifbb at gmail.com>
>>>>>>> mouncifbb at gmail.com> wrote:
>>>>>>>
>>>>>>>>  NPANXX,"LATA","OCN","NTER","INTRA"
>>>>>>>> 201007,"224","7229","0.0059","0.0127"
>>>>>>>> 201040,"224","9206","0.0036","0.0036"
>>>>>>>>
>>>>>>>
>>>>>>> Looks like they give you the LATA and OCN values with the prefix.  We
>>>>>>> (should) look that up ourselves.
>>>>>>>
>>>>>>>
>>>>>>>> FreeSWITCH Version 1.0.trunk (16540)
>>>>>>>>
>>>>>>>>
>>>>>>>> Also I noticed the *npa_nxx_ocn* table never get consulted.
>>>>>>>>
>>>>>>>> I also see this now when making a real call instead of running
>>>>>>>> thorugh CLI
>>>>>>>>
>>>>>>>> EXECUTE sofia/external/6179472456 at 174.x.x.x lcr(16179470890
>>>>>>>> NANPA_STD)
>>>>>>>> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:1230 *intrastate
>>>>>>>> channel var is [undef]*
>>>>>>>
>>>>>>>
>>>>>>> This is fine.  it is a leftover from when you would tell mod_lcr via
>>>>>>> a channel var that it should do intrastate.  I later had mod_lcr do the
>>>>>>> lookup itself, but we still honor the old var.  There are no channel vars
>>>>>>> associated with the cli, so you wouldn't see that msg.
>>>>>>>
>>>>>>>
>>>>>>>>
>>>>>>>> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:1233 Select routes
>>>>>>>> based on interstate rates
>>>>>>>> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:1252 LCR Lookup on
>>>>>>>> 16179470893 using profile NANPA_STD
>>>>>>>> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:786 intra routing
>>>>>>>> [state:0 lata:0] so rate field is [rate]
>>>>>>>>
>>>>>>>> called number 6179470890  caller ID: 6179472456
>>>>>>>>
>>>>>>>> any ideas??
>>>>>>>>
>>>>>>>>
>>>>>>> Only thing that jumps out at me.
>>>>>>>
>>>>>>> The output from lcr_admin show profiles showed only the default one.
>>>>>>>  On the dialplan you use the NANPA_STD profile.  Can you check lcr_admin
>>>>>>> list and see if that profile is defined and if so if it says it is using the
>>>>>>> npanxx table?
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> On Sat, Jan 30, 2010 at 10:02 AM, Rupa Schomaker < <rupa at rupa.com>
>>>>>>>> rupa at rupa.com> wrote:
>>>>>>>>
>>>>>>>>> Something is still missing from the logs. Note the query of the
>>>>>>>>> npanxx
>>>>>>>>> table, the flags being set, and the rate field being chosen.  Umm..
>>>>>>>>> oh, what version of fs are you running?
>>>>>>>>>
>>>>>>>>> Yes, the npa_nxx_ocn table needs to be loaded up as described in:
>>>>>>>>> <http://wiki.freeswitch.org/wiki/Mod_cidlookup#Falling_back_to_.22City_State.22_in_the_absense_of_a_name>
>>>>>>>>> http://wiki.freeswitch.org/wiki/Mod_cidlookup#Falling_back_to_.22City_State.22_in_the_absense_of_a_name
>>>>>>>>> (there is a link to that from mod_lcr's wiki page).
>>>>>>>>>
>>>>>>>>> An example from my own setup:
>>>>>>>>>
>>>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:1384 data passed to
>>>>>>>>> lcr
>>>>>>>>> is [12148267711 default 12148267712]
>>>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:1420 Set Caller ID to
>>>>>>>>> [12148267712]
>>>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:759 SQL: SELECT
>>>>>>>>> 'state',
>>>>>>>>> count(DISTINCT state) FROM npa_nxx_company_ocn WHERE (npa=214 AND
>>>>>>>>> nxx=826) OR (npa=214 AND nxx=826) UNION SELECT 'lata',
>>>>>>>>> count(DISTINCT
>>>>>>>>> lata) FROM npa_nxx_company_ocn WHERE (npa=214 AND nxx=826) OR
>>>>>>>>> (npa=214
>>>>>>>>> AND nxx=826)
>>>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:719 Type: lata, Count:
>>>>>>>>> 1
>>>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:719 Type: state,
>>>>>>>>> Count: 1
>>>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:809 intra routing
>>>>>>>>> [state:1 lata:1] so rate field is [intralata_rate]
>>>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:825 we have an event
>>>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:849 SQL:  SELECT
>>>>>>>>> l.digits
>>>>>>>>> AS lcr_digits,         c.carrier_name AS lcr_carrier_name,
>>>>>>>>> l.intralata_rate as lcr_rate_field,        cg.prefix AS
>>>>>>>>> lcr_gw_prefix,
>>>>>>>>> cg.suffix AS lcr_gw_suffix,        l.lead_strip AS lcr_lead_strip,
>>>>>>>>> l.trail_strip AS lcr_trail_strip,        l.prefix AS lcr_prefix,
>>>>>>>>> l.suffix AS lcr_suffix,        cg.codec AS lcr_codec, l.cid AS
>>>>>>>>> lcr_cid,        'carriers' AS lcr_limit_realm, c.carrier_name AS
>>>>>>>>> lcr_limit_id, 5 AS lcr_limit_max   FROM lcr l      JOIN carriers c
>>>>>>>>> ON
>>>>>>>>> l.carrier_id=c.id      JOIN carrier_gateway cg ON c.id
>>>>>>>>> =cg.carrier_id
>>>>>>>>>  WHERE c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1'
>>>>>>>>> AND digits_prefix @> '12148267711'      AND CURRENT_TIMESTAMP
>>>>>>>>> BETWEEN
>>>>>>>>> date_start AND date_end   ORDER BY digits DESC, intralata_rate,
>>>>>>>>> random();
>>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning
>>>>>>>>> Dialstring
>>>>>>>>>
>>>>>>>>> [lcr_carrier=grnvoip_us,lcr_rate=0.00591]sofia/gateway/grnvoip/XXXX12148267711
>>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:638 Adding grnvoip_us
>>>>>>>>> to
>>>>>>>>> head of list
>>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning
>>>>>>>>> Dialstring
>>>>>>>>>
>>>>>>>>> [lcr_carrier=teliax,lcr_rate=0.01000]sofia/gateway/teliax_atl/12148267711
>>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:660 Adding teliax to
>>>>>>>>> end of list
>>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning
>>>>>>>>> Dialstring
>>>>>>>>>
>>>>>>>>> [lcr_carrier=teliax,lcr_rate=0.01000]sofia/gateway/teliax/12148267711
>>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:660 Adding teliax to
>>>>>>>>> end of list
>>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning
>>>>>>>>> Dialstring
>>>>>>>>>
>>>>>>>>> [lcr_carrier=vitelity,lcr_rate=0.01440]sofia/gateway/vitelity/12148267711
>>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:660 Adding vitelity to
>>>>>>>>> end of list
>>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning
>>>>>>>>> Dialstring
>>>>>>>>>
>>>>>>>>> [lcr_carrier=grnvoip_std,lcr_rate=0.01500]sofia/gateway/grnvoip/YYYY12148267711
>>>>>>>>> [...]
>>>>>>>>>
>>>>>>>>> On Fri, Jan 29, 2010 at 10:42 PM, Mouncif Benniane <<mouncifbb at gmail.com>
>>>>>>>>> mouncifbb at gmail.com> wrote:
>>>>>>>>> > Also the Provider has presented the rates in this format?
>>>>>>>>> > NPANXXLATA OCN INTER INTRA
>>>>>>>>> >
>>>>>>>>> >
>>>>>>>>> > On Fri, Jan 29, 2010 at 11:30 PM, Mouncif Benniane <<mouncifbb at gmail.com>
>>>>>>>>> mouncifbb at gmail.com>
>>>>>>>>> > wrote:
>>>>>>>>> >>
>>>>>>>>> >> Tried it and it's not giving me intralata  instead I get
>>>>>>>>> interstate, does
>>>>>>>>> >> the npa_nxx_company_ocn table needs to be used in this case?,
>>>>>>>>> also do I have
>>>>>>>>> >> to have the rate field in lcr table?
>>>>>>>>> >>
>>>>>>>>> >> lcr 617642  default 6176421212
>>>>>>>>> >>  | Digit Match | Carrier  | Rate     | Codec | CID Regexp |
>>>>>>>>> Dialstring
>>>>>>>>> >>                                                        |
>>>>>>>>> >>  | 617642      | carrier1 | 0.00500  |       |            |
>>>>>>>>> >>
>>>>>>>>> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642 |
>>>>>>>>> >>
>>>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:1329 data passed to
>>>>>>>>> lcr is
>>>>>>>>> >> [617642  default 6176421212]
>>>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:1365 Set Caller ID
>>>>>>>>> to
>>>>>>>>> >> [6176421212]
>>>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:786 intra routing
>>>>>>>>> [state:0
>>>>>>>>> >> lata:0] so rate field is [rate]
>>>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:802 we have an
>>>>>>>>> event
>>>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:826 SQL: SELECT
>>>>>>>>> l.digits,
>>>>>>>>> >> c.carrier_name, l.rate, cg.prefix AS gw_prefix, cg.suffix AS
>>>>>>>>> gw_suffix,
>>>>>>>>> >> l.lead_strip, l.trail_strip, l.prefix, l.suffix , cg.codec ,
>>>>>>>>> l.cid FROM lcr
>>>>>>>>> >> l JOIN carriers c ON l.carrier_id=c.id JOIN carrier_gateway cg
>>>>>>>>> ON
>>>>>>>>> >> c.id=cg.carrier_id WHERE c.enabled = '1' AND cg.enabled = '1'
>>>>>>>>> AND l.enabled
>>>>>>>>> >> = '1' AND digits IN (617642, 61764, 6176, 617, 61, 6) AND
>>>>>>>>> CURRENT_TIMESTAMP
>>>>>>>>> >> BETWEEN date_start AND date_end ORDER BY digits DESC,  rate,
>>>>>>>>> rand();
>>>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:307 Returning
>>>>>>>>> Dialstring
>>>>>>>>> >>
>>>>>>>>> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642
>>>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:615 Adding carrier1
>>>>>>>>> to head
>>>>>>>>> >> of list
>>>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:307 Returning
>>>>>>>>> Dialstring
>>>>>>>>> >>
>>>>>>>>> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642
>>>>>>>>> >>
>>>>>>>>> >> Thank you Rupa!
>>>>>>>>> >>
>>>>>>>>> >> On Fri, Jan 29, 2010 at 7:37 PM, Rupa Schomaker <<rupa at rupa.com>
>>>>>>>>> rupa at rupa.com> wrote:
>>>>>>>>> >>>
>>>>>>>>> >>> turn console logging up to debug and redo the lcr lookup.  The
>>>>>>>>> sql
>>>>>>>>> >>> statements along with status info will show up.  This should
>>>>>>>>> give
>>>>>>>>> >>> enough information to debug what is happening.
>>>>>>>>> >>>
>>>>>>>>> >>> I'm assuming the npanxx table is actually populated and not
>>>>>>>>> just
>>>>>>>>> >>> existing?
>>>>>>>>> >>>
>>>>>>>>> >>> When doing the lookup from the cli you have to tell lcr what
>>>>>>>>> CID to
>>>>>>>>> >>> use (remember, it is relative to the src/dest number).  I'm
>>>>>>>>> pretty
>>>>>>>>> >>> sure you get something on the console log when you don't
>>>>>>>>> specify a CID
>>>>>>>>> >>> when using the commandline.  Anyway:
>>>>>>>>> >>>
>>>>>>>>> >>> lcr 617642  default 6176421212
>>>>>>>>> >>>
>>>>>>>>> >>> should give you intralata.
>>>>>>>>> >>>
>>>>>>>>> >>> Note that the definition of intralata doesn't mean "local" for
>>>>>>>>> some
>>>>>>>>> >>> providers.  Some providers define local to "same ratecenter"
>>>>>>>>> which is
>>>>>>>>> >>> even more restrictive.
>>>>>>>>> >>>
>>>>>>>>> >>> On Fri, Jan 29, 2010 at 4:43 PM, Mouncif Benniane <<mouncifbb at gmail.com>
>>>>>>>>> mouncifbb at gmail.com>
>>>>>>>>> >>> wrote:
>>>>>>>>> >>> > i can't make use of mod_lcr using Intra/Interstate rating, I
>>>>>>>>> am using
>>>>>>>>> >>> > svn: FreeSWITCH Version 1.0.trunk (16517)
>>>>>>>>> >>> >
>>>>>>>>> >>> > lcr mysql table structure:
>>>>>>>>> >>> > CREATE TABLE `lcr` (
>>>>>>>>> >>> >   `id` INT(11) NOT NULL AUTO_INCREMENT,
>>>>>>>>> >>> >   `digits` VARCHAR(15) DEFAULT NULL,
>>>>>>>>> >>> >   `rate` FLOAT(11,5) DEFAULT NULL,
>>>>>>>>> >>> >   `intrastate_rate` FLOAT(11,5) DEFAULT NULL,
>>>>>>>>> >>> >   `intralata_rate` FLOAT(11,5) DEFAULT NULL,
>>>>>>>>> >>> >   `carrier_id` INT(11) NOT NULL,
>>>>>>>>> >>> >   `lead_strip` INT(11) NOT NULL,
>>>>>>>>> >>> >   `trail_strip` INT(11) NOT NULL,
>>>>>>>>> >>> >   `prefix` VARCHAR(16) NOT NULL,
>>>>>>>>> >>> >   `suffix` VARCHAR(16) NOT NULL,
>>>>>>>>> >>> >   `lcr_profile` VARCHAR(32) DEFAULT NULL,
>>>>>>>>> >>> >   `date_start` DATETIME NOT NULL DEFAULT '1970-01-01
>>>>>>>>> 00:00:00',
>>>>>>>>> >>> >   `date_end` DATETIME NOT NULL DEFAULT '2030-12-31 00:00:00',
>>>>>>>>> >>> >   `quality` FLOAT(10,6) NOT NULL,
>>>>>>>>> >>> >   `reliability` FLOAT(10,6) NOT NULL,
>>>>>>>>> >>> >   `cid` VARCHAR(32) NOT NULL DEFAULT '',
>>>>>>>>> >>> >   `enabled` TINYINT(1) NOT NULL DEFAULT '1',
>>>>>>>>> >>> >   PRIMARY KEY  (`id`),
>>>>>>>>> >>> >   KEY `carrier_id` (`carrier_id`),
>>>>>>>>> >>> >   KEY `digits` (`digits`),
>>>>>>>>> >>> >   KEY `lcr_profile` (`lcr_profile`),
>>>>>>>>> >>> >   KEY `digits_profile_cid_rate` USING BTREE (`digits`),
>>>>>>>>> >>> >   CONSTRAINT `carrier_id` FOREIGN KEY (`carrier_id`)
>>>>>>>>> REFERENCES
>>>>>>>>> >>> > `carriers`
>>>>>>>>> >>> > (`id`) ON DELETE CASCADE ON UPDATE CASCADE
>>>>>>>>> >>> > ) ENGINE=INNODB AUTO_INCREMENT=6 DEFAULT CHARSET=latin1
>>>>>>>>> >>> >
>>>>>>>>> >>> >
>>>>>>>>> >>> > lcr_admin show profiles
>>>>>>>>> >>> > Name:           default
>>>>>>>>> >>> >  custom sql:    SELECT l.digits, c.carrier_name,
>>>>>>>>> l.${lcr_rate_field},
>>>>>>>>> >>> > cg.prefix AS gw_prefix, cg.suffix AS gw_suffix, l.lead_strip,
>>>>>>>>> >>> > l.trail_strip,
>>>>>>>>> >>> > l.prefix, l.suffix , cg.codec , l.cid FROM lcr l JOIN
>>>>>>>>> carriers c ON
>>>>>>>>> >>> > l.carrier_id=c.id JOIN carrier_gateway cg ON c.id=cg.carrier_id
>>>>>>>>> WHERE
>>>>>>>>> >>> > c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1' AND
>>>>>>>>> digits IN
>>>>>>>>> >>> > (${lcr_query_expanded_digits}) AND CURRENT_TIMESTAMP BETWEEN
>>>>>>>>> date_start
>>>>>>>>> >>> > AND
>>>>>>>>> >>> > date_end ORDER BY digits DESC,  ${lcr_rate_field},  quality
>>>>>>>>> DESC,
>>>>>>>>> >>> >  reliability DESC, rand();
>>>>>>>>> >>> >  has %:         false
>>>>>>>>> >>> >  has vars:      true
>>>>>>>>> >>> >  has intrastate:        true
>>>>>>>>> >>> >  has intralata: true
>>>>>>>>> >>> >  has npanxx:    true
>>>>>>>>> >>> >  Reorder rate:  enabled
>>>>>>>>> >>> >  Info in headers:       disabled
>>>>>>>>> >>> >  Quote IN() List:       disabled
>>>>>>>>> >>> >
>>>>>>>>> >>> >
>>>>>>>>> >>> >
>>>>>>>>> >>> > lcr 617642  default  returns rate from the rate field table
>>>>>>>>> and not
>>>>>>>>> >>> > intra/inter state fields rates.
>>>>>>>>> >>> >
>>>>>>>>> >>> > Any ideas? thanks!
>>>>>>>>> >>> >
>>>>>>>>> >>> >
>>>>>>>>> >>> >
>>>>>>>>> >>> >
>>>>>>>>> >>> > _______________________________________________
>>>>>>>>> >>> > FreeSWITCH-users mailing list
>>>>>>>>> >>> > <FreeSWITCH-users at lists.freeswitch.org>
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> >>> >
>>>>>>>>> <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> >>> >
>>>>>>>>> >>> > UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> >>> > <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>>>> >>> >
>>>>>>>>> >>> >
>>>>>>>>> >>>
>>>>>>>>> >>>
>>>>>>>>> >>>
>>>>>>>>> >>> --
>>>>>>>>> >>> -Rupa
>>>>>>>>> >>>
>>>>>>>>> >>> _______________________________________________
>>>>>>>>> >>> FreeSWITCH-users mailing list
>>>>>>>>> >>> <FreeSWITCH-users at lists.freeswitch.org>
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> >>>
>>>>>>>>> <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> >>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> >>> <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>>>> >>
>>>>>>>>> >
>>>>>>>>> >
>>>>>>>>> > _______________________________________________
>>>>>>>>> > FreeSWITCH-users mailing list
>>>>>>>>> > <FreeSWITCH-users at lists.freeswitch.org>
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> > <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> > UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> > <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>>>> >
>>>>>>>>> >
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> --
>>>>>>>>> -Rupa
>>>>>>>>>
>>>>>>>>> _______________________________________________
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> --
>>>>>>> -Rupa
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> -Rupa
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> -Rupa
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> -Rupa
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100201/6ef88a40/attachment-0001.html 

From mbsip at gazeta.pl  Mon Feb  1 15:15:28 2010
From: mbsip at gazeta.pl (mbsip)
Date: Tue, 2 Feb 2010 00:15:28 +0100
Subject: [Freeswitch-users] vm-disk-quota
In-Reply-To: <28f27f5d1002010930x61bcef3v54e22621761c8bbd@mail.gmail.com>
References: <28f27f5d1001311246h17b426a4x39e0d48d3d305342@mail.gmail.com>
	<7DFBD163-A72A-4186-BB89-6D468FD9ABA2@jerris.com>
	<28f27f5d1002010930x61bcef3v54e22621761c8bbd@mail.gmail.com>
Message-ID: <28f27f5d1002011515u2c270c3cp4b38450e56a0bbf8@mail.gmail.com>

>> If I read the code right (mod_voicemail.c:3051) it looks like it is measuring in seconds of vocicemail, but the wiki indicates number of voicemails, neither seems to match the name of the param.

I double checked - as Mike stated "vm-disk-quota" limits seconds of
voicemail messages.

As an example let's provide FS with <param name="vm-disk-quota" value="10"/>.
It is then possible to store let say 4sek + 5sek + 6sek of vm messages.

Thx for clearing this up.
Maciej.


From mbsip at gazeta.pl  Mon Feb  1 15:22:38 2010
From: mbsip at gazeta.pl (mbsip)
Date: Tue, 2 Feb 2010 00:22:38 +0100
Subject: [Freeswitch-users] voicemail_greeting_number - question
In-Reply-To: <28f27f5d1001310905r41b16ca7r5ef1f236f76a070c@mail.gmail.com>
References: <28f27f5d1001310905r41b16ca7r5ef1f236f76a070c@mail.gmail.com>
Message-ID: <28f27f5d1002011522h31e03c0aoa71f26dabfa1d174@mail.gmail.com>

> Hi ALL,
>
> I am playing around with VM and want to play user recorded greeting
> instead of default one.
> I've scaned wiki Mod_Voicemail and found proper parameter
> "voicemail_greeting_number".
> Unfortunately there is a lack of example hence i dont know if it is
> already working.
>
> Aforementioned param was placed in /conf/directory/default/1000.xml
> file (param name="voicemail_greeting_number", i tried many values)
> The effect is that the default greeting is played.
>
> Is this param embeeded into FS right now?
> How to use it?
> Is there any other place I should do the changes?
>
> I am running ?FreeSWITCH Version 1.0.trunk (16456).
>
> Thx in advance.
> Maciej
>

Anyone knows how to use this param?

Of course i may provide voicemail_default.db with proper greeting_path
manually but i am not sure if "voicemail_greeting_number" works the
same way and is somehow correlated?

Thx,
Maciej.


From msc at freeswitch.org  Mon Feb  1 16:47:47 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 1 Feb 2010 16:47:47 -0800
Subject: [Freeswitch-users] Svar: Re: Somebody help me understand the
	'features' set up please :-)
In-Reply-To: <4B66E734020000E100000451@mail.fribert.dk>
References: <4B66226C020000E10000043C@mail.fribert.dk>
	<b39273531001311645u618ea87am8a3d84ee6b1bcfcb@mail.gmail.com>
	<4B669ED3020000E100000447@mail.fribert.dk>
	<b39273531002010303h6e66baf7tf77be8d67449122b@mail.gmail.com>
	<4B66E734020000E100000451@mail.fribert.dk>
Message-ID: <87f2f3b91002011647k792d2fdbyf4eaa7c2a41d9ab5@mail.gmail.com>

On Mon, Feb 1, 2010 at 5:37 AM, mailinglist <mailinglist at fribert.dk> wrote:

>  Hmm, I've just downloaded the default.xml under conf/dialplan from the
> SVN just to be on the safe side.
> Line 758 is the last </include>, but I did find some examples on line
> 249-251.
>
> So I've changed my dialplan entry handling calls from the outside to this:
>
> <extension name="PrivateIn">
>    <condition field="destination_number" expression="^(8202)$">
>        <action application="bind_meta_app" data="1 b s
> execute_extension::att_xfer XML features"/>
>        <action application="bind_meta_app" data="2 b s
> record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
>
>        <action application="set" data="call_timeout=45"/>
>        <action application="set" data="domain_name=10.11.12.25"/>
>        <action application="set" data="continue_on_fail=true"/>
>        <action application="set" data="hangup_after_bridge=true"/>
>        <action application="set" data="transfer_ringback=$${hold_music}"/>
>        <action application="bridge" data="group/private@${domain_name}"/<group/private@$%7Bdomain_name%7D%22/>
> >
>        <action application="answer" data=""/>
>        <action application="sleep" data="1000"/>
>        <action application="voicemail" data="default 10.11.12.25 45"/>
>    </condition>
> </extension>
> As I understand the bind_meta_app it listens for *1 and then it runs the
> att_xfer, *2 to record the call.
> I've included the att_xfer in the XML features.
>
> Question is, will it work at all when I bridge to a group?
>
> Nothing happens when I press *1 and an extension.
>

Fribse,

I think your confusion might be from the purpose of the bind_meta_app
application. The *1 or *2, etc. must be dialed after the call has been
established. In other words, bind_meta_app sits there on an existing call,
listening for *1 or *2 (etc.) and if the person on the appropriate call leg
dials it then the application in bind_meta_app gets executed. Example:

<action application="bind_meta_app" data="2 b s
record_session::$${recordings_
dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>

The "2 b s" part of that means: Listen for *2 on the b leg and execute the
app on the s leg. So, the b leg could dial *2 and it would initiate call
recording.

Let's take a step back... are you sure you need bind_meta_app? What exactly
is your use case here? In general terms, what are you trying to accomplish
with your dialplan extension?

-MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100201/2cb0cded/attachment.html 

From msc at freeswitch.org  Mon Feb  1 16:49:07 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 1 Feb 2010 16:49:07 -0800
Subject: [Freeswitch-users] voicemail_greeting_number - question
In-Reply-To: <28f27f5d1002011522h31e03c0aoa71f26dabfa1d174@mail.gmail.com>
References: <28f27f5d1001310905r41b16ca7r5ef1f236f76a070c@mail.gmail.com>
	<28f27f5d1002011522h31e03c0aoa71f26dabfa1d174@mail.gmail.com>
Message-ID: <87f2f3b91002011649p62ddff50o3f47bbb2b0be538a@mail.gmail.com>

On Mon, Feb 1, 2010 at 3:22 PM, mbsip <mbsip at gazeta.pl> wrote:

> > Hi ALL,
> >
> > I am playing around with VM and want to play user recorded greeting
> > instead of default one.
> > I've scaned wiki Mod_Voicemail and found proper parameter
> > "voicemail_greeting_number".
> > Unfortunately there is a lack of example hence i dont know if it is
> > already working.
> >
> > Aforementioned param was placed in /conf/directory/default/1000.xml
> > file (param name="voicemail_greeting_number", i tried many values)
> > The effect is that the default greeting is played.
> >
> > Is this param embeeded into FS right now?
> > How to use it?
> > Is there any other place I should do the changes?
> >
> > I am running  FreeSWITCH Version 1.0.trunk (16456).
> >
> > Thx in advance.
> > Maciej
> >
>
> Anyone knows how to use this param?
>
> Of course i may provide voicemail_default.db with proper greeting_path
> manually but i am not sure if "voicemail_greeting_number" works the
> same way and is somehow correlated?
>
> Have you recorded vm greeting one, vm greeting two, etc. before changing
the param?
-MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100201/d69c62b7/attachment.html 

From msc at freeswitch.org  Mon Feb  1 16:55:47 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 1 Feb 2010 16:55:47 -0800
Subject: [Freeswitch-users] Limit the extension creation
In-Reply-To: <df97b9c51002011122t29c505adu60c4ccb73746432c@mail.gmail.com>
References: <df97b9c51002011122t29c505adu60c4ccb73746432c@mail.gmail.com>
Message-ID: <87f2f3b91002011655q186d29eif5a7124d256dcb54@mail.gmail.com>

On Mon, Feb 1, 2010 at 11:22 AM, Scott Fernandez <scottferri09 at gmail.com>wrote:

> Hi,
>
> Is there a way to restrict the number of extension that FS
> supports/serves?. The idea is to limit the concurrent usage of the system
> for which we need to restrict the FS to support upto a predefined no. of
> users/extensions.
>
> Can anyone assist please?
>
>
Do you mean limiting the number of extensions defined in the XML configs, or
do you mean the number of concurrent calls? If the former you'll probably
need to use mod_xml_curl and have your backend db & logic enforce the max
number of users/extensions. If the latter then you'll want to read up on
mod_limit: http://wiki.freeswitch.org/wiki/Mod_limit

-MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100201/5dddbc9d/attachment.html 

From msc at freeswitch.org  Mon Feb  1 16:59:54 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 1 Feb 2010 16:59:54 -0800
Subject: [Freeswitch-users] Logging question
In-Reply-To: <da4ab26e1002011307n6b10e97tea3e281cbff90e13@mail.gmail.com>
References: <da4ab26e1002011307n6b10e97tea3e281cbff90e13@mail.gmail.com>
Message-ID: <87f2f3b91002011659i25f9c3f5la3f2f6706fd259d8@mail.gmail.com>

On Mon, Feb 1, 2010 at 1:07 PM, paul gore <paul.gore.j at gmail.com> wrote:

> I tried "sofia profile internal siptrace on" from console and got SIP
> trace on the console,
> but nothing in logs still.
>
> AFAIK the SIP logs will only go to the console. Is there something in
particular that you are trying to capture? SIP is quite the chatty protocol
and will happily fill up your entire disk with requests and responses. In
many cases the FreeSWITCH log has the basic information from the SIP
message, like why a call failed (i.e. which 4xx message was received). Are
you tracking a particular problem?

-MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100201/45256d86/attachment.html 

From sos at sokhapkin.dyndns.org  Mon Feb  1 17:08:21 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Mon, 1 Feb 2010 20:08:21 -0500
Subject: [Freeswitch-users] Logging question
In-Reply-To: <87f2f3b91002011659i25f9c3f5la3f2f6706fd259d8@mail.gmail.com>
References: <da4ab26e1002011307n6b10e97tea3e281cbff90e13@mail.gmail.com>
	<87f2f3b91002011659i25f9c3f5la3f2f6706fd259d8@mail.gmail.com>
Message-ID: <201002012008.21189.sos@sokhapkin.dyndns.org>

"sofia tracelevel info" sends log to log file.

On Monday 01 February 2010, Michael Collins wrote:
> On Mon, Feb 1, 2010 at 1:07 PM, paul gore <paul.gore.j at gmail.com> wrote:
> > I tried "sofia profile internal siptrace on" from console and got SIP
> > trace on the console,
> > but nothing in logs still.
> >
> > AFAIK the SIP logs will only go to the console. Is there something in
>
> particular that you are trying to capture? SIP is quite the chatty protocol
> and will happily fill up your entire disk with requests and responses. In
> many cases the FreeSWITCH log has the basic information from the SIP
> message, like why a call failed (i.e. which 4xx message was received). Are
> you tracking a particular problem?
>
> -MC




From paul.gore.j at gmail.com  Mon Feb  1 17:31:18 2010
From: paul.gore.j at gmail.com (paul gore)
Date: Mon, 1 Feb 2010 20:31:18 -0500
Subject: [Freeswitch-users] Logging question
In-Reply-To: <87f2f3b91002011659i25f9c3f5la3f2f6706fd259d8@mail.gmail.com>
References: <da4ab26e1002011307n6b10e97tea3e281cbff90e13@mail.gmail.com>
	<87f2f3b91002011659i25f9c3f5la3f2f6706fd259d8@mail.gmail.com>
Message-ID: <da4ab26e1002011731s4b5f8906se23de33a713e0396@mail.gmail.com>

Yes, I am trying to understand why Grandstream GXP2000 times out when
connected to FS voice mail at exactly 60 sec. Can't seem to figure it out
from regular debug log, it just shows "normal clearing". I already used
"record_waste.." param but it does not seem to have any effect.
I guess I have to do tcp dump.


On Mon, Feb 1, 2010 at 7:59 PM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Mon, Feb 1, 2010 at 1:07 PM, paul gore <paul.gore.j at gmail.com> wrote:
>
>> I tried "sofia profile internal siptrace on" from console and got SIP
>> trace on the console,
>> but nothing in logs still.
>>
>> AFAIK the SIP logs will only go to the console. Is there something in
> particular that you are trying to capture? SIP is quite the chatty protocol
> and will happily fill up your entire disk with requests and responses. In
> many cases the FreeSWITCH log has the basic information from the SIP
> message, like why a call failed (i.e. which 4xx message was received). Are
> you tracking a particular problem?
>
> -MC
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100201/c4feb2ad/attachment-0001.html 

From brian at freeswitch.org  Mon Feb  1 17:36:33 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 1 Feb 2010 19:36:33 -0600
Subject: [Freeswitch-users] Logging question
In-Reply-To: <da4ab26e1002011731s4b5f8906se23de33a713e0396@mail.gmail.com>
References: <da4ab26e1002011307n6b10e97tea3e281cbff90e13@mail.gmail.com>
	<87f2f3b91002011659i25f9c3f5la3f2f6706fd259d8@mail.gmail.com>
	<da4ab26e1002011731s4b5f8906se23de33a713e0396@mail.gmail.com>
Message-ID: <50A3745B-369A-40B9-9484-40E3FD41AFF9@freeswitch.org>

Well if you do "sofia profile internal siptrace on" and then press F8, make the call I'm going to guess its the session timers and its freaking out and not answering the reinvite.

/b

On Feb 1, 2010, at 7:31 PM, paul gore wrote:

> Yes, I am trying to understand why Grandstream GXP2000 times out when connected to FS voice mail at exactly 60 sec. Can't seem to figure it out from regular debug log, it just shows "normal clearing". I already used "record_waste.." param but it does not seem to have any effect. 
> I guess I have to do tcp dump.
> 



From emptysands at gmail.com  Mon Feb  1 19:52:29 2010
From: emptysands at gmail.com (Nicholas Lee)
Date: Tue, 2 Feb 2010 16:52:29 +1300
Subject: [Freeswitch-users] Hybrid Encryption?
In-Reply-To: <B2BE2748-ADA4-4D1C-8004-99AD72FB7806@jerris.com>
References: <2b6116b31001272154l3c0bbe80y8bf3db94961e8e1d@mail.gmail.com>
	<8A9EDC4E-C49B-488D-9DBF-169A185462AB@freeswitch.org>
	<2b6116b31001281631u7ada7876wc419bb7afadd7ef7@mail.gmail.com>
	<46C06209-9515-4B1B-B449-F55A51FF548B@freeswitch.org>
	<2b6116b31001281808x1a004cd2ne7e8dcb9f16fec3e@mail.gmail.com>
	<B2BE2748-ADA4-4D1C-8004-99AD72FB7806@jerris.com>
Message-ID: <2b6116b31002011952w71108bf7sfd10c9e29ad2af7c@mail.gmail.com>

Ok, so when [1] talks about Freeswitch acting as intermediary in order to
encrypt further communications between an end node SIP device and the main
PBX, it means that the FS intermediary node is actually a full SIP node. The
phones will need to auth to this and any calls will be routed to trunks via
the dial plan.


[1] http://wiki.freeswitch.org/wiki/SIP_TLS#Hybrid_Encryption


On Sat, Jan 30, 2010 at 8:50 PM, Michael Jerris <mike at jerris.com> wrote:

> Freeswitch isn't a proxy, and no, we don't provide support for passthrough
> auth like this.  A proxy would, but not sure of any proxy based solution
> that would do the srtp work for you.
>
> Mike
>
> On Jan 28, 2010, at 9:08 PM, Nicholas Lee wrote:
>
> Is there a way to do it transparently?  The FS proxies will past though the
> extension creds.
>
> On Fri, Jan 29, 2010 at 1:52 PM, Brian West <brian at freeswitch.org> wrote:
>
>> Then yes you could use FreeSWITCH to augment your Asterisk install and
>> enable encryption from site to site.
>>
>> /b
>>
>>
>> > Unfortunately it's not going to cover every situation.
>> >
>> >
>> > Nicholas
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100202/f6472db4/attachment.html 

From mike at jerris.com  Mon Feb  1 22:57:26 2010
From: mike at jerris.com (Michael Jerris)
Date: Tue, 2 Feb 2010 01:57:26 -0500
Subject: [Freeswitch-users] vm-disk-quota
In-Reply-To: <28f27f5d1002011515u2c270c3cp4b38450e56a0bbf8@mail.gmail.com>
References: <28f27f5d1001311246h17b426a4x39e0d48d3d305342@mail.gmail.com>
	<7DFBD163-A72A-4186-BB89-6D468FD9ABA2@jerris.com>
	<28f27f5d1002010930x61bcef3v54e22621761c8bbd@mail.gmail.com>
	<28f27f5d1002011515u2c270c3cp4b38450e56a0bbf8@mail.gmail.com>
Message-ID: <1BBB9F7F-47B0-4280-B3B3-315282D150F0@jerris.com>

Don't depend on this behavior, really, open a bug on jira for us to figure this one out.  We either need to rename this or change behavior

Mike

On Feb 1, 2010, at 6:15 PM, mbsip wrote:

>>> If I read the code right (mod_voicemail.c:3051) it looks like it is measuring in seconds of vocicemail, but the wiki indicates number of voicemails, neither seems to match the name of the param.
> 
> I double checked - as Mike stated "vm-disk-quota" limits seconds of
> voicemail messages.
> 
> As an example let's provide FS with <param name="vm-disk-quota" value="10"/>.
> It is then possible to store let say 4sek + 5sek + 6sek of vm messages.



From mike at jerris.com  Tue Feb  2 00:14:35 2010
From: mike at jerris.com (Michael Jerris)
Date: Tue, 2 Feb 2010 03:14:35 -0500
Subject: [Freeswitch-users] Dialogic Board (JCT & DM3) Supported?
In-Reply-To: <29b888f81002010308uf875b53h838453b6e77e9a9e@mail.gmail.com>
References: <29b888f81002010308uf875b53h838453b6e77e9a9e@mail.gmail.com>
Message-ID: <D7BC4112-CA0B-4825-8ADA-A8F228C2F201@jerris.com>

Nope

On Feb 1, 2010, at 6:08 AM, Moiz Chinoy wrote:
> Can anyone please advise that whether Dialogic boards (JCT and DM3)
> are supported by FS.



From mike at jerris.com  Tue Feb  2 00:19:00 2010
From: mike at jerris.com (Michael Jerris)
Date: Tue, 2 Feb 2010 03:19:00 -0500
Subject: [Freeswitch-users] Outbound Call (No Registration)
In-Reply-To: <033e01caa37c$87cd2080$97676180$@com>
References: <1254B3C5-D7D3-413D-BA82-54FE4789B360@freeswitch.org>
	<022701caa04a$44f60b80$cee22280$@com>
	<697C7F93-3737-4ABD-8934-6A48DC09C088@freeswitch.org>
	<025701caa0a7$e1ca6200$a55f2600$@com>
	<02b001caa10c$913b9100$b3b2b300$@com>
	<F03237B7-89EE-4CE4-A67F-FCB8E8D26700@freeswitch.org>
	<02c501caa113$105532b0$30ff9810$@com>
	<87f2f3b91001291110l226ce6d8u6e806d0a1b782c5f@mail.gmail.com>
	<02dd01caa127$428aa760$c79ff620$@com>
	<032301caa367$307c6c60$91754520$@com>
	<20100201193459.GI27405@base.carmickle.com>
	<033e01caa37c$87cd2080$97676180$@com>
Message-ID: <21BCBF5F-F609-474C-A295-0B4FFAE75144@jerris.com>

you use normal regex replacement


^00614(\d{8})

in your dialing rule you would use 120#614$1

On Feb 1, 2010, at 3:24 PM, RR wrote:

> Hi Frank,
> 
> Thanks for the response. The remote gateway is not running FreeSWITCH. But
> that's ok, I figured out the problem. I was capturing packets from the wrong
> IP address so after adding the username/password, the call did manage to get
> to the other end (after I realized the source IP was wrong) and realized
> that my dialplan at the other side wasn't setup correctly. Should be able to
> fix that. 
> 
> BTW, before I go hunting in the Wiki, can you off the top of your head tell
> me how I can manipulate numbers in the dialplan?
> 
> If my destination number = 00614xxxxxxxx but I want to send 120#614xxxxxxx
> instead, how would I do that?
> 



From mailinglist at fribert.dk  Tue Feb  2 00:12:11 2010
From: mailinglist at fribert.dk (mailinglist)
Date: Tue, 02 Feb 2010 09:12:11 +0100
Subject: [Freeswitch-users] Svar: Re: Somebody help me understand	 the
	'features' set up please :-)
In-Reply-To: <87f2f3b91002011647k792d2fdbyf4eaa7c2a41d9ab5@mail.gmail.com>
References: <4B66226C020000E10000043C@mail.fribert.dk>
	<b39273531001311645u618ea87am8a3d84ee6b1bcfcb@mail.gmail.com>
	<4B669ED3020000E100000447@mail.fribert.dk>
	<b39273531002010303h6e66baf7tf77be8d67449122b@mail.gmail.com>
	<4B66E734020000E100000451@mail.fribert.dk>
	<87f2f3b91002011647k792d2fdbyf4eaa7c2a41d9ab5@mail.gmail.com>
Message-ID: <4B67EC6B020000E100000456@mail.fribert.dk>

Hi Michael
 
I'm trying to get the possibility of transfering an incoming call from one extension to another, and give the possibility of turning it into a conference.
I don't have a 'transfer' button.
I do have an 'R' button on the Siemens handsets, and a 'Flash' button on the Sipura. The 'Flash' button gives me a new dialtone, gives the caller MOH, and then I can dial the new extension, and transfer the call, but not create a conference.
But the Siemens handset does not have a 'flash', and pressing the R doesn't do anything.
 
It might be two different features 'transfer' and 'conference'...

But I thought that using the bind_meta_app would accomplish both.
 
It's on an incoming call from the outside.
So the situation:
The Public folder has an entry that matches the dialed number, and does a transfer to 8202.
Then the dialplan matches the 8202 with a group, and the phone rings.
Somebody picks it up, finds out that it needs to be transferred to another extension, or transferred to a conference with a second extension.
 
 
Best regards
Fribse
 
/Running Freeswitch on pfSense running on a VMWare ESXi 4.0

>>> 02-02-2010 kl. 01:47 skrev Michael Collins <msc at freeswitch.org> i meddelelsen <87f2f3b91002011647k792d2fdbyf4eaa7c2a41d9ab5 at mail.gmail.com>:



On Mon, Feb 1, 2010 at 5:37 AM, mailinglist <mailinglist at fribert.dk> wrote:


Hmm, I've just downloaded the default.xml under conf/dialplan from the SVN just to be on the safe side.
Line 758 is the last </include>, but I did find some examples on line 249-251.
So I've changed my dialplan entry handling calls from the outside to this:

<extension name="PrivateIn">
<condition field="destination_number" expression="^(8202)$">
<action application="bind_meta_app" data="1 b s execute_extension::att_xfer XML features"/>
<action application="bind_meta_app" data="2 b s record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
<action application="set" data="call_timeout=45"/>
<action application="set" data="domain_name=10.11.12.25"/>
<action application="set" data="continue_on_fail=true"/>
<action application="set" data="hangup_after_bridge=true"/>
<action application="set" data="transfer_ringback=$${hold_music}"/>
<action application="bridge" data="group/private@${domain_name}"/ ( mailto:group/private@$%7Bdomain_name%7D%22/ )>
<action application="answer" data=""/>
<action application="sleep" data="1000"/>
<action application="voicemail" data="default 10.11.12.25 45"/>
</condition>
</extension>
As I understand the bind_meta_app it listens for *1 and then it runs the att_xfer, *2 to record the call.
I've included the att_xfer in the XML features.
Question is, will it work at all when I bridge to a group?
Nothing happens when I press *1 and an extension.


Fribse,

I think your confusion might be from the purpose of the bind_meta_app application. The *1 or *2, etc. must be dialed after the call has been established. In other words, bind_meta_app sits there on an existing call, listening for *1 or *2 (etc.) and if the person on the appropriate call leg dials it then the application in bind_meta_app gets executed. Example:

<action application="bind_meta_app" data="2 b s record_session::$${recordings_
dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>

The "2 b s" part of that means: Listen for *2 on the b leg and execute the app on the s leg. So, the b leg could dial *2 and it would initiate call recording. 

Let's take a step back... are you sure you need bind_meta_app? What exactly is your use case here? In general terms, what are you trying to accomplish with your dialplan extension?

-MC


-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100202/22f4bb1a/attachment-0001.html 

From d at d-man.org  Tue Feb  2 07:26:03 2010
From: d at d-man.org (Darren Schreiber)
Date: Tue, 2 Feb 2010 07:26:03 -0800
Subject: [Freeswitch-users] Announcing... the FreeSWITCH Users Group! COME
	JOIN US!
Message-ID: <8A034A3098ED3C4990F7D9DE40F5585F177DEE2F7C@EXVMBX020-3.exch020.serverdata.net>

Hey everyone!
    I'm writing to invite you all to sign up for a new set of events - FreeSWITCH Users Group Meetups. These in-person gatherings across the country exist to encourage you in creating awesome telephony and general communications software, hardware and services. Anyone interested in, working on or otherwise wanting to learn about FreeSWITCH and general telecommunications services should attend our meetings. Note that topics are not necessarily limited to just FreeSWITCH!

     Meetings will involve formal trainings, informal install fests, lots of Q&A time and general time to just share ideas and meet people.

     Each meeting has a formal topic to help kick things off, so don't worry if you are not sure what you can offer to the group - just showing up and asking questions is a great help! If you do have a specific topic you'd like to learn more about,  email the group's organizer and we'll see what we can do.

We'll kick off with three group locations across the country. You can sign-up to learn more about the scheduled meet-ups by using the links below:
    * San Francisco, California: http://www.meetup.com/fsusers/
    * Manhattan, New York: http://www.meetup.com/fsusers-ny/
    * Orlando, Florida: http://www.meetup.com/fsusers-orlando/

PLEASE SIGN-UP ONLINE SO WE CAN GAUGE INTEREST - You are just showing you might come - not committing to a date!

     Dates for each meet-up will be announced on the relevant websites in the coming months. We'll also have a VoIP dial-in for locations that will allow it.

---> If you'd like to start a meet-up in your area, please let me know. I'll help you arrange the meet-up spot, topics and advertising of the event.

     These events are currently free. The events are run by an independent FreeSWITCH enthusiast and are not associated with any corporation or formal group.

     Enjoy and we look forward to seeing you at our first meet-up!

- Darren Schreiber
  the FreeSWITCH Users Group


From lawwton at gmail.com  Tue Feb  2 09:05:36 2010
From: lawwton at gmail.com (Alfredo Quiroga-Villamil)
Date: Tue, 2 Feb 2010 12:05:36 -0500
Subject: [Freeswitch-users] Announcing... the FreeSWITCH Users Group!
	COME JOIN US!
In-Reply-To: <8A034A3098ED3C4990F7D9DE40F5585F177DEE2F7C@EXVMBX020-3.exch020.serverdata.net>
References: <8A034A3098ED3C4990F7D9DE40F5585F177DEE2F7C@EXVMBX020-3.exch020.serverdata.net>
Message-ID: <5fe6fa8f1002020905s4ebba5d3hc6c59cfa4fa2b27d@mail.gmail.com>

Any freeswitch users in the RTP area in NC?

On Tue, Feb 2, 2010 at 10:26 AM, Darren Schreiber <d at d-man.org> wrote:
> Hey everyone!
> ? ?I'm writing to invite you all to sign up for a new set of events - FreeSWITCH Users Group Meetups. These in-person gatherings across the country exist to encourage you in creating awesome telephony and general communications software, hardware and services. Anyone interested in, working on or otherwise wanting to learn about FreeSWITCH and general telecommunications services should attend our meetings. Note that topics are not necessarily limited to just FreeSWITCH!
>
> ? ? Meetings will involve formal trainings, informal install fests, lots of Q&A time and general time to just share ideas and meet people.
>
> ? ? Each meeting has a formal topic to help kick things off, so don't worry if you are not sure what you can offer to the group - just showing up and asking questions is a great help! If you do have a specific topic you'd like to learn more about, ?email the group's organizer and we'll see what we can do.
>
> We'll kick off with three group locations across the country. You can sign-up to learn more about the scheduled meet-ups by using the links below:
> ? ?* San Francisco, California: http://www.meetup.com/fsusers/
> ? ?* Manhattan, New York: http://www.meetup.com/fsusers-ny/
> ? ?* Orlando, Florida: http://www.meetup.com/fsusers-orlando/
>
> PLEASE SIGN-UP ONLINE SO WE CAN GAUGE INTEREST - You are just showing you might come - not committing to a date!
>
> ? ? Dates for each meet-up will be announced on the relevant websites in the coming months. We'll also have a VoIP dial-in for locations that will allow it.
>
> ---> If you'd like to start a meet-up in your area, please let me know. I'll help you arrange the meet-up spot, topics and advertising of the event.
>
> ? ? These events are currently free. The events are run by an independent FreeSWITCH enthusiast and are not associated with any corporation or formal group.
>
> ? ? Enjoy and we look forward to seeing you at our first meet-up!
>
> - Darren Schreiber
> ?the FreeSWITCH Users Group
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From max at ramax.it  Tue Feb  2 07:03:05 2010
From: max at ramax.it (Massimiliano Ravelli)
Date: Tue, 2 Feb 2010 16:03:05 +0100
Subject: [Freeswitch-users] Fifo: ring agents without answering to caller
Message-ID: <302375DF-6EC3-492D-A82A-7FC8344B01D6@ramax.it>

Hi everybody.

I am using asterisk for a call center pbx and evaluating if I can replace it with FS.
With asterisk queue I can ring agents without actually answer to the caller so he won't pay till the agent pickup.
Can I do the same with FS ?

Moreover I'd really like to fullfill this scenario: I managed to solve it in asterisk only with an ugly workaround.
Customer calls the queue and he doesn't get answered for 20 seconds and then he hears welcome message, music and whatever is configured in the fifo.
Obviously the agents should ring even in the starting 20 seconds.

Any hint or pointer ?

Thanks in advance,
Massimiliano



From stevendt at primrosebank.net  Tue Feb  2 09:29:34 2010
From: stevendt at primrosebank.net (Dave Stevenson)
Date: Tue, 2 Feb 2010 17:29:34 -0000
Subject: [Freeswitch-users] Phones losing registration
Message-ID: <0C95C94A5E2B42449ECF9244C9760D51@bp1.ad.bp.com>

Hi,

I have a mixture of phones and am having a problem with some of them periodically losing registration with FreeSwitch.

All the phones start off working after provisioning and have registration expiry times configured of 3600 seconds.

The Cisco and Thomson phones are OK, staying registered without problems but other phones, including SwissVoice IP10Ss and a cheap wireless VOIP phone lose registration after a random time, sometimes hours, sometimes days.

Is this a common issue with known work-arounds or can someone point me in the right direction for how to see if any errors are being generated or if I need to turn on any additional FreeSWITCH logging options to trap the errors please ?

regards
Dave
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100202/ddd7d241/attachment.html 

From msc at freeswitch.org  Tue Feb  2 09:31:10 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 2 Feb 2010 09:31:10 -0800
Subject: [Freeswitch-users] Fifo: ring agents without answering to caller
In-Reply-To: <302375DF-6EC3-492D-A82A-7FC8344B01D6@ramax.it>
References: <302375DF-6EC3-492D-A82A-7FC8344B01D6@ramax.it>
Message-ID: <87f2f3b91002020931r424629fcqee1eae8a2875be7c@mail.gmail.com>

On Tue, Feb 2, 2010 at 7:03 AM, Massimiliano Ravelli <max at ramax.it> wrote:

> Hi everybody.
>
> I am using asterisk for a call center pbx and evaluating if I can replace
> it with FS.
> With asterisk queue I can ring agents without actually answer to the caller
> so he won't pay till the agent pickup.
> Can I do the same with FS ?
>
> Yes. However you will need to investigate how to handle your billing and
accounting. FreeSWITCH has many hooks for this and there are some 3rd party
projects that use these. You can even use mod_nibblebill for pre-paid
billing.


> Moreover I'd really like to fullfill this scenario: I managed to solve it
> in asterisk only with an ugly workaround.
> Customer calls the queue and he doesn't get answered for 20 seconds and
> then he hears welcome message, music and whatever is configured in the fifo.
> Obviously the agents should ring even in the starting 20 seconds.
>
>
What happens in the first 20 seconds? Do any phones ring? Just curious
what's happening in that limbo period. In any case, I highly recommend that
you grab a spare Linux box and load up FreeSWITCH and play around. The wiki
has tons of information and you can get good real-time help in #freeswitch
in irc.freenode.net.

-MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100202/f41cd995/attachment.html 

From edpimentl at gmail.com  Tue Feb  2 09:33:52 2010
From: edpimentl at gmail.com (EdPimentl)
Date: Tue, 2 Feb 2010 12:33:52 -0500
Subject: [Freeswitch-users] Announcing... the FreeSWITCH Users Group!
	COME JOIN US!
In-Reply-To: <5fe6fa8f1002020905s4ebba5d3hc6c59cfa4fa2b27d@mail.gmail.com>
References: <8A034A3098ED3C4990F7D9DE40F5585F177DEE2F7C@EXVMBX020-3.exch020.serverdata.net>
	<5fe6fa8f1002020905s4ebba5d3hc6c59cfa4fa2b27d@mail.gmail.com>
Message-ID: <9dc4a1671002020933p1c05b00at188274ab564e7aa6@mail.gmail.com>

Any FS users  in ATlanta?

-E
http://vCardCloud.com
http://JustGoogl.Me
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100202/05894bb5/attachment.html 

From Suneel.Papineni at mettoni.com  Tue Feb  2 10:09:50 2010
From: Suneel.Papineni at mettoni.com (Suneel Papineni)
Date: Tue, 2 Feb 2010 18:09:50 -0000
Subject: [Freeswitch-users] Attendant call transfer
Message-ID: <3181A30B8C35AB4AA8577B78DDF461380668B5D3@nickel.mettonigroup.com>

Hi,

 

I am trying to establish attendant call transfer using event sockets. 

1.       A call has come into Freeswitch from an external Gateway  and this call is parked (it is configured to park all calls coming to freeswitch) {Caller A ? FS}

2.       Once the call is parked, I am sending a command to originate a call to another number connected to external gateway. {FS ? Caller B}. Call is established between FS and caller B. ("api originate sofia/external/<another number>@<gateway IP address> 9999")

3.       On receiving event message as "Application: Answer", I am sending another command to bridge call between A & B. ("api uuid_bridge <callerA UUID> <caller UUID>")

4.       With this call is established between A & B, but there is a huge delay (appox 30 secs).

 

I believe that FS is still in the call and might be this is creating  delay (not sure).

 

Could you please tell me if I am doing something wrong or process to achieve this scenario working.

 

I tried in to transfer the call instead of bridging  using the command ("uuid_transfer <callerA UUID> intercept: <caller UUID> inline"), but the response is same as above with huge delay.

 

Thanks & Regards

Suneel

 


*************************************************************************
Please consider the environment before printing this e-mail
*************************************************************************
This email and any files transmitted with it are confidential and
intended solely for the use of the individual or entity to whom they
are addressed. If you have received this email in error please notify
the system manager.  http://www.mettoni.com

Mettoni Ltd
Registered in England and Wales: 4485956
9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN
*************************************************************************

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100202/de05f44a/attachment-0001.html 

From brian at freeswitch.org  Tue Feb  2 10:17:14 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 2 Feb 2010 12:17:14 -0600
Subject: [Freeswitch-users] Attendant call transfer
In-Reply-To: <3181A30B8C35AB4AA8577B78DDF461380668B5D3@nickel.mettonigroup.com>
References: <3181A30B8C35AB4AA8577B78DDF461380668B5D3@nickel.mettonigroup.com>
Message-ID: <07247F72-DE53-4028-AFBF-BB5EB23FEA7B@freeswitch.org>

Suneel,
	After printing 100 copies of this email It dawned on me that you failed to include any details about what SVN revision you're using.  If you can reply with that info I can promptly print out 100 more copies and see if we can find your problem.

Thanks,
Brian
PS: just kidding about the printing part, but the svn rev would be helpful.

On Feb 2, 2010, at 12:09 PM, Suneel Papineni wrote:

> Hi,
>  
> I am trying to establish attendant call transfer using event sockets.
> 1.       A call has come into Freeswitch from an external Gateway  and this call is parked (it is configured to park all calls coming to freeswitch) {Caller A ? FS}
> 2.       Once the call is parked, I am sending a command to originate a call to another number connected to external gateway. {FS ? Caller B}. Call is established between FS and caller B. (?api originate sofia/external/<another number>@<gateway IP address> 9999?)
> 3.       On receiving event message as ?Application: Answer?, I am sending another command to bridge call between A & B. (?api uuid_bridge <callerA UUID> <caller UUID>?)
> 4.       With this call is established between A & B, but there is a huge delay (appox 30 secs).
>  
> I believe that FS is still in the call and might be this is creating  delay (not sure).
>  
> Could you please tell me if I am doing something wrong or process to achieve this scenario working.
>  
> I tried in to transfer the call instead of bridging  using the command (?uuid_transfer <callerA UUID> intercept: <caller UUID> inline?), but the response is same as above with huge delay.
>  
> Thanks & Regards
> Suneel
>  
> *************************************************************************
> Please consider the environment before printing this e-mail
> *************************************************************************
> This email and any files transmitted with it are confidential and
> intended solely for the use of the individual or entity to whom they
> are addressed. If you have received this email in error please notify
> the system manager.  http://www.mettoni.com
> 
> Mettoni Ltd
> Registered in England and Wales: 4485956
> 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN
> *************************************************************************
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100202/6083a1ae/attachment.html 

From tim at communicatefreely.net  Tue Feb  2 10:32:06 2010
From: tim at communicatefreely.net (Tim St. Pierre)
Date: Tue, 02 Feb 2010 13:32:06 -0500
Subject: [Freeswitch-users] Subdomain directory
Message-ID: <4B686FA6.7070400@communicatefreely.net>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hi folks,

I'm building a multi-tenant system, and I need to separate customers from each other, but only in
certain situations.

I want to apply a subdomain to directory entries, so that a search for a user within a subdomain
will only return a user in that domain, but a search in the parent domain will match all users in
that domain AND all subdomains.

Is there some sort of domain aliasing mechanism, or should I do this in the external scripting?

I am doing everything with xml_curl.  I could write the script so that it will look at the domain
requested and decide if a user should be returned or not.

Does anyone see any problems with this setup, or know of an easier way?

The goal is so that things like SIP subscription and registrations are all done in the parent domain
(only one domain system wide), but voice mail, directory application (dial by name), and some other
features can be restricted to the same customer.

All our user ID's (extension numbers for us) are unique.

Thanks!

- -Tim
- --
Tim St. Pierre
IP Voice technician
Communicate Freely
1-877-291-8647 x5101
sip:5101 at communicatefreely.net
tim at communicatefreely.net
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v2.0.4 (FreeBSD)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org

iQCVAwUBS2hvpoqVcvNCnHOrAQK9sgP/eiEtcEx1+OnMZPw0ZQ4gQUG+v/ddewa7
M8c5At4MoqjqNum5ruHtVA22UQt9U3gCG4gsXwaIXmDEDiMm6CTnjYYAgv5q4DvQ
0CmgHo/e1tgTly65GWIys2kwwUqyFsktZ0AeC7FFTyHLfr8l3uE4FJEY5lMRRdrC
0Eceg7MVoY8=
=e45O
-----END PGP SIGNATURE-----


From jerry.richards at teotech.com  Tue Feb  2 12:30:06 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Tue, 2 Feb 2010 12:30:06 -0800
Subject: [Freeswitch-users] Sangoma_boost_client.c:356 Invalid Boost Version
	100 Expecting 101
Message-ID: <0713E04C73F84E47842F6AE17699609C@greyhawk.tonecommander.com>


I upgraded my FS version to 20100201 (i.e. Feb 1st 2010) and the console is
endlessly logging the following errors:

2010-02-02 12:18:10.125121 [CRIT] sangoma_boost_client.c:356 Invalid Boost
Version 100  Expecting 101
2010-02-02 12:18:10.125121 [CRIT] sangoma_boost_client.c:370 Critical Error:
PQ Invalid Event lenght from boost rxlen=23 evsz=1031
2010-02-02 12:18:12.134803 [CRIT] sangoma_boost_client.c:356 Invalid Boost
Version 100  Expecting 101
2010-02-02 12:18:12.134803 [CRIT] sangoma_boost_client.c:370 Critical Error:
PQ Invalid Event lenght from boost rxlen=23 evsz=1031
2010-02-02 12:18:14.134440 [CRIT] sangoma_boost_client.c:356 Invalid Boost
Version 100  Expecting 101
2010-02-02 12:18:14.134440 [CRIT] sangoma_boost_client.c:370 Critical Error:
PQ Invalid Event lenght from boost rxlen=23 evsz=1031

Does anyone know what is causing this?  I am using Wanpipe Driver
wanpipe-3.5.8.6.  I reinstalled the driver and rebuilt FS a couple of times.

Thanks and Best Regards,
Jerry



From moises.silva at gmail.com  Tue Feb  2 13:11:09 2010
From: moises.silva at gmail.com (Moises Silva)
Date: Tue, 2 Feb 2010 16:11:09 -0500
Subject: [Freeswitch-users] Sangoma_boost_client.c:356 Invalid Boost
	Version 100 Expecting 101
In-Reply-To: <0713E04C73F84E47842F6AE17699609C@greyhawk.tonecommander.com>
References: <0713E04C73F84E47842F6AE17699609C@greyhawk.tonecommander.com>
Message-ID: <c4d05cbe1002021311y3d015c3avcab6bcbe4e851c15@mail.gmail.com>

Hello Jerry,

Please download the wanpipe driver version at
ftp://ftp.sangoma.com/linux/custom/DavidYS/wanpipe-3.5.10.smg_pri.1.tgz

As per instructions found at
http://wiki.sangoma.com/wanpipe-SmgPriInstallation

The problem is that at some point FreeSWITCH started requiring a very recent
Sangoma boost version, which has not been released formally yet.

If you run into any other issue let me know,

-- 
Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com

On Tue, Feb 2, 2010 at 3:30 PM, Jerry Richards
<jerry.richards at teotech.com>wrote:

>
> I upgraded my FS version to 20100201 (i.e. Feb 1st 2010) and the console is
> endlessly logging the following errors:
>
> 2010-02-02 12:18:10.125121 [CRIT] sangoma_boost_client.c:356 Invalid Boost
> Version 100  Expecting 101
> 2010-02-02 12:18:10.125121 [CRIT] sangoma_boost_client.c:370 Critical
> Error:
> PQ Invalid Event lenght from boost rxlen=23 evsz=1031
> 2010-02-02 12:18:12.134803 [CRIT] sangoma_boost_client.c:356 Invalid Boost
> Version 100  Expecting 101
> 2010-02-02 12:18:12.134803 [CRIT] sangoma_boost_client.c:370 Critical
> Error:
> PQ Invalid Event lenght from boost rxlen=23 evsz=1031
> 2010-02-02 12:18:14.134440 [CRIT] sangoma_boost_client.c:356 Invalid Boost
> Version 100  Expecting 101
> 2010-02-02 12:18:14.134440 [CRIT] sangoma_boost_client.c:370 Critical
> Error:
> PQ Invalid Event lenght from boost rxlen=23 evsz=1031
>
> Does anyone know what is causing this?  I am using Wanpipe Driver
> wanpipe-3.5.8.6.  I reinstalled the driver and rebuilt FS a couple of
> times.
>
> Thanks and Best Regards,
> Jerry
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100202/0e4fa933/attachment.html 

From jerry.richards at teotech.com  Tue Feb  2 13:13:21 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Tue, 2 Feb 2010 13:13:21 -0800
Subject: [Freeswitch-users] Sangoma_boost_client.c:356 Invalid Boost
	Version 100 Expecting 101
In-Reply-To: <4B688D0D.4080407@sangoma.com>
References: <0713E04C73F84E47842F6AE17699609C@greyhawk.tonecommander.com>
	<4B688D0D.4080407@sangoma.com>
Message-ID: <5C5862A8AFBB4A6785D9EA001E2F21DE@greyhawk.tonecommander.com>

Yannick,
 
That procedure had no effect on the problem.  Any other ideas?
 
Thanks and Best Regards,
Jerry


  _____  

From: Yannick Lam [mailto:yannick at sangoma.com] 
Sent: Tuesday, February 02, 2010 12:38 PM
To: Jerry Richards
Subject: Re: Sangoma_boost_client.c:356 Invalid Boost Version 100 Expecting
101


Hi Jerry,

Can you please stop freeswitch, smg and the wanpipe driver.
Then shutdown your computer and then re insert the card in the systema nd
then boot again and see if you are still getting the issue.



Thank-you, 



Yannick Lam Hang, B.Eng, Tech Support Engineer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
e. yannick at sangoma.com  |   <skype:yannick.sangoma?add> Skype  |
<mailto:yannick_sangoma at hotmail.com> msn 

 
<mailbox://C%7C/Documents%20and%20Settings/Marc/Application%20Data/Thunderbi
rd/Profiles/6u5xr3yx.default/Mail/Local%20Folders/www.sangoma.com>
www.sangoma.com  |  <http://wiki.sangoma.com/> wiki.sangoma.com


Lifetime Warranty. Because it must work!







 





Jerry Richards wrote: 

I upgraded my FS version to 20100201 (i.e. Feb 1st 2010) and the console is

endlessly logging the following errors:



2010-02-02 12:18:10.125121 [CRIT] sangoma_boost_client.c:356 Invalid Boost

Version 100  Expecting 101

2010-02-02 12:18:10.125121 [CRIT] sangoma_boost_client.c:370 Critical Error:

PQ Invalid Event lenght from boost rxlen=23 evsz=1031

2010-02-02 12:18:12.134803 [CRIT] sangoma_boost_client.c:356 Invalid Boost

Version 100  Expecting 101

2010-02-02 12:18:12.134803 [CRIT] sangoma_boost_client.c:370 Critical Error:

PQ Invalid Event lenght from boost rxlen=23 evsz=1031

2010-02-02 12:18:14.134440 [CRIT] sangoma_boost_client.c:356 Invalid Boost

Version 100  Expecting 101

2010-02-02 12:18:14.134440 [CRIT] sangoma_boost_client.c:370 Critical Error:

PQ Invalid Event lenght from boost rxlen=23 evsz=1031



Does anyone know what is causing this?  I am using Wanpipe Driver

wanpipe-3.5.8.6.  I reinstalled the driver and rebuilt FS a couple of times.



Thanks and Best Regards,

Jerry





  

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100202/86746748/attachment-0001.html 

From robert.hadley at teotech.com  Tue Feb  2 14:54:27 2010
From: robert.hadley at teotech.com (Robert Hadley)
Date: Tue, 2 Feb 2010 14:54:27 -0800
Subject: [Freeswitch-users] Sangoma A200 FXS callwaiting and dialplan
	statements
Message-ID: <56F86E3E1D2F45ACB4AEE1FE448E68E9@greyhawk.tonecommander.com>

I have a server that has Freeswitch (1.0.5.pre9) a Sangoma A101 (E1/T1) and
Analog A200 cards using OpenZap.  The cards are connected.  The wanrouter
version is 3.5.8.6 as the newer version didn't work for us yet.  Also trying
to test with the FS trunk but having other wanpipe driver issues.

 

I have a question about using the A200 card.  The intent is to hook up FAX
machines to the two FXS ports.  The desired behavior is the first call goes
to the first FXS port one, and while port one is busy a second call goes to
the second port.

 

In the Freeswitch dialplan there is a statement to bridge to FXS port 1 and
if that fails then bridge to FXS port 2.  However, the first bridge never
fails even when it's busy.  I notice the A200 card supports call waiting.
The FS debug messages show that is what is happening.

 

Question: How do I disable call waiting on the A200 card when used with
OpenZap?

 

I am not sure about whether the hangup_after_bridge=true and
continue_on_fail=true are necessary or in the correct spots.  I've tried
calling with these statements present or not and moved after the first
bridge but doesn't appear to affect the status of the first bridge
statement.

 

Question: Are the hangup_after_bridge=true and continue_on_fail=true
statements necessary and in the right spot?

 

Dialplan statements:

    <extension name="fxsfax">

      <!-- This routes destinations 1010 and 1011 calls to FXS card -->

      <condition field="destination_number" expression="^(101[0-1])$">

        <action application="answer"/>

        <action application="set" data="hangup_after_bridge=true"/>

        <action application="set" data="continue_on_fail=true"/>

        <action application="bridge" data="openzap/FXS/1"/>

        <action application="bridge" data="openzap/FXS/2"/>

 

Debug statements for second call to FXS/1 extension:

(Notice that several of statements mention CALLWAITING.)

2010-02-02 14:00:55.193948 [DEBUG] switch_core_session.c:639 Send signal
sofia/internal/1045 at 192.168.72.141:5060 [BREAK]

2010-02-02 14:00:55.193948 [NOTICE] mod_dptools.c:658 Channel
[sofia/internal/1045 at 192.168.72.141:5060] has been answered

2010-02-02 14:00:55.193948 [DEBUG] switch_channel.c:182
sofia/internal/1045 at 192.168.72.141:5060 receive message [AUDIO_SYNC]

EXECUTE sofia/internal/1045 at 192.168.72.141:5060 set(continue_on_fail=true)

2010-02-02 14:00:55.193948 [DEBUG] sofia.c:3787 Channel
sofia/internal/1045 at 192.168.72.141:5060 entering state [completed][200]

2010-02-02 14:00:55.193948 [DEBUG] mod_dptools.c:768
sofia/internal/1045 at 192.168.72.141:5060 SET [continue_on_fail]=[true]

EXECUTE sofia/internal/1045 at 192.168.72.141:5060 bridge(openzap/FXS/1)

2010-02-02 14:00:55.194948 [DEBUG] mod_openzap.c:366 Set codec PCMU 20ms

2010-02-02 14:00:55.194948 [DEBUG] mod_openzap.c:1191 Connect outbound
channel OpenZAP/2:1/

2010-02-02 14:00:55.194948 [NOTICE] switch_channel.c:613 New Channel
OpenZAP/2:1/ [bc08ce84-eff2-491c-b601-14b7329c71ea]

2010-02-02 14:00:55.194948 [DEBUG] mod_openzap.c:1203 (OpenZAP/2:1/) State
Change CS_NEW -> CS_INIT

2010-02-02 14:00:55.194948 [DEBUG] switch_core_session.c:1018 Send signal
OpenZAP/2:1/ [BREAK]

2010-02-02 14:00:55.194948 [DEBUG] ozmod_analog.c:78 Changing state on 2:1
from UP to CALLWAITING

2010-02-02 14:00:55.194948 [DEBUG] switch_core_state_machine.c:314
(OpenZAP/2:1/) Running State Change CS_INIT

2010-02-02 14:00:55.194948 [DEBUG] switch_core_state_machine.c:338
(OpenZAP/2:1/) State INIT

2010-02-02 14:00:55.194948 [DEBUG] mod_openzap.c:390 (OpenZAP/2:1/) State
Change CS_INIT -> CS_ROUTING

2010-02-02 14:00:55.194948 [DEBUG] switch_core_session.c:1018 Send signal
OpenZAP/2:1/ [BREAK]

2010-02-02 14:00:55.194948 [DEBUG] switch_core_state_machine.c:338
(OpenZAP/2:1/) State INIT going to sleep

2010-02-02 14:00:55.194948 [DEBUG] switch_core_state_machine.c:314
(OpenZAP/2:1/) Running State Change CS_ROUTING

2010-02-02 14:00:55.194948 [DEBUG] switch_core_state_machine.c:341
(OpenZAP/2:1/) State ROUTING

2010-02-02 14:00:55.194948 [DEBUG] mod_openzap.c:413 OpenZAP/2:1/ CHANNEL
ROUTING

2010-02-02 14:00:55.194948 [DEBUG] switch_ivr_originate.c:66 (OpenZAP/2:1/)
State Change CS_ROUTING -> CS_CONSUME_MEDIA

2010-02-02 14:00:55.194948 [DEBUG] switch_core_session.c:1018 Send signal
OpenZAP/2:1/ [BREAK]

2010-02-02 14:00:55.194948 [DEBUG] switch_core_state_machine.c:341
(OpenZAP/2:1/) State ROUTING going to sleep

2010-02-02 14:00:55.194948 [DEBUG] switch_core_state_machine.c:314
(OpenZAP/2:1/) Running State Change CS_CONSUME_MEDIA

2010-02-02 14:00:55.194948 [DEBUG] switch_core_state_machine.c:360
(OpenZAP/2:1/) State CONSUME_MEDIA

2010-02-02 14:00:55.194948 [DEBUG] switch_core_state_machine.c:360
(OpenZAP/2:1/) State CONSUME_MEDIA going to sleep

2010-02-02 14:00:55.210047 [DEBUG] sofia.c:3787 Channel
sofia/internal/1045 at 192.168.72.141:5060 entering state [ready][200]

2010-02-02 14:00:55.213006 [DEBUG] ozmod_analog.c:450 Executing state
handler on 2:1 for CALLWAITING

2010-02-02 14:00:55.657988 [DEBUG] switch_rtp.c:2004 Correct ip/port
confirmed.

2010-02-02 14:01:05.723334 [DEBUG] ozmod_analog.c:422 Changing state on 2:1
from CALLWAITING to UP

 

 

Thanks,

Robert

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100202/cc56a777/attachment.html 

From mbsip at gazeta.pl  Tue Feb  2 14:57:05 2010
From: mbsip at gazeta.pl (mbsip)
Date: Tue, 2 Feb 2010 23:57:05 +0100
Subject: [Freeswitch-users] vm-disk-quota
In-Reply-To: <1BBB9F7F-47B0-4280-B3B3-315282D150F0@jerris.com>
References: <28f27f5d1001311246h17b426a4x39e0d48d3d305342@mail.gmail.com>
	<7DFBD163-A72A-4186-BB89-6D468FD9ABA2@jerris.com>
	<28f27f5d1002010930x61bcef3v54e22621761c8bbd@mail.gmail.com>
	<28f27f5d1002011515u2c270c3cp4b38450e56a0bbf8@mail.gmail.com>
	<1BBB9F7F-47B0-4280-B3B3-315282D150F0@jerris.com>
Message-ID: <28f27f5d1002021457i39e03944j230ee3e9fda6e622@mail.gmail.com>

Done, MODAPP-173.

Thx,
Maciej.

2010/2/2 Michael Jerris <mike at jerris.com>:
> Don't depend on this behavior, really, open a bug on jira for us to figure this one out. ?We either need to rename this or change behavior
>


From robert.hadley at teotech.com  Tue Feb  2 15:02:22 2010
From: robert.hadley at teotech.com (Robert Hadley)
Date: Tue, 2 Feb 2010 15:02:22 -0800
Subject: [Freeswitch-users] Announcing... the FreeSWITCH Users Group!
	COMEJOIN US!
In-Reply-To: <8A034A3098ED3C4990F7D9DE40F5585F177DEE2F7C@EXVMBX020-3.exch020.serverdata.net>
References: <8A034A3098ED3C4990F7D9DE40F5585F177DEE2F7C@EXVMBX020-3.exch020.serverdata.net>
Message-ID: <2974D37F499B455FBC036B02A4DC6107@greyhawk.tonecommander.com>


Any interested FreeSWITCH users in the Seattle area?


-----Original Message-----
From: Darren Schreiber [mailto:d at d-man.org] 
Sent: Tuesday, February 02, 2010 7:26 AM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Announcing... the FreeSWITCH Users Group!
COMEJOIN US!

Hey everyone!
    I'm writing to invite you all to sign up for a new set of events -
FreeSWITCH Users Group Meetups. These in-person gatherings across the
country exist to encourage you in creating awesome telephony and general
communications software, hardware and services. Anyone interested in,
working on or otherwise wanting to learn about FreeSWITCH and general
telecommunications services should attend our meetings. Note that topics are
not necessarily limited to just FreeSWITCH!

     Meetings will involve formal trainings, informal install fests, lots of
Q&A time and general time to just share ideas and meet people.

     Each meeting has a formal topic to help kick things off, so don't worry
if you are not sure what you can offer to the group - just showing up and
asking questions is a great help! If you do have a specific topic you'd like
to learn more about,  email the group's organizer and we'll see what we can
do.

We'll kick off with three group locations across the country. You can
sign-up to learn more about the scheduled meet-ups by using the links below:
    * San Francisco, California: http://www.meetup.com/fsusers/
    * Manhattan, New York: http://www.meetup.com/fsusers-ny/
    * Orlando, Florida: http://www.meetup.com/fsusers-orlando/

PLEASE SIGN-UP ONLINE SO WE CAN GAUGE INTEREST - You are just showing you
might come - not committing to a date!

     Dates for each meet-up will be announced on the relevant websites in
the coming months. We'll also have a VoIP dial-in for locations that will
allow it.

---> If you'd like to start a meet-up in your area, please let me know. I'll
help you arrange the meet-up spot, topics and advertising of the event.

     These events are currently free. The events are run by an independent
FreeSWITCH enthusiast and are not associated with any corporation or formal
group.

     Enjoy and we look forward to seeing you at our first meet-up!

- Darren Schreiber
  the FreeSWITCH Users Group





From jerry.richards at teotech.com  Tue Feb  2 15:38:30 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Tue, 2 Feb 2010 15:38:30 -0800
Subject: [Freeswitch-users] Sangoma_boost_client.c:356 Invalid
	BoostVersion 100 Expecting 101
In-Reply-To: <c4d05cbe1002021311y3d015c3avcab6bcbe4e851c15@mail.gmail.com>
References: <0713E04C73F84E47842F6AE17699609C@greyhawk.tonecommander.com>
	<c4d05cbe1002021311y3d015c3avcab6bcbe4e851c15@mail.gmail.com>
Message-ID: <CC572C08D8F342EBA856DDACAF354C19@greyhawk.tonecommander.com>

Thank You Moises.  The
<ftp://ftp.sangoma.com/linux/custom/DavidYS/wanpipe-3.5.10.smg_pri.1.tgz>
ftp://ftp.sangoma.com/linux/custom/DavidYS/wanpipe-3.5.10.smg_pri.1.tgz
driver works.
 
Best Regards,
Jerry
 


  _____  

From: Moises Silva [mailto:moises.silva at gmail.com] 
Sent: Tuesday, February 02, 2010 1:11 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Sangoma_boost_client.c:356 Invalid
BoostVersion 100 Expecting 101


Hello Jerry,

Please download the wanpipe driver version at
ftp://ftp.sangoma.com/linux/custom/DavidYS/wanpipe-3.5.10.smg_pri.1.tgz

As per instructions found at
http://wiki.sangoma.com/wanpipe-SmgPriInstallation

The problem is that at some point FreeSWITCH started requiring a very recent
Sangoma boost version, which has not been released formally yet.

If you run into any other issue let me know,

-- 
Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com


On Tue, Feb 2, 2010 at 3:30 PM, Jerry Richards <jerry.richards at teotech.com>
wrote:



I upgraded my FS version to 20100201 (i.e. Feb 1st 2010) and the console is
endlessly logging the following errors:

2010-02-02 12:18:10.125121 [CRIT] sangoma_boost_client.c:356 Invalid Boost
Version 100  Expecting 101
2010-02-02 12:18:10.125121 [CRIT] sangoma_boost_client.c:370 Critical Error:
PQ Invalid Event lenght from boost rxlen=23 evsz=1031
2010-02-02 12:18:12.134803 [CRIT] sangoma_boost_client.c:356 Invalid Boost
Version 100  Expecting 101
2010-02-02 12:18:12.134803 [CRIT] sangoma_boost_client.c:370 Critical Error:
PQ Invalid Event lenght from boost rxlen=23 evsz=1031
2010-02-02 12:18:14.134440 [CRIT] sangoma_boost_client.c:356 Invalid Boost
Version 100  Expecting 101
2010-02-02 12:18:14.134440 [CRIT] sangoma_boost_client.c:370 Critical Error:
PQ Invalid Event lenght from boost rxlen=23 evsz=1031

Does anyone know what is causing this?  I am using Wanpipe Driver
wanpipe-3.5.8.6.  I reinstalled the driver and rebuilt FS a couple of times.

Thanks and Best Regards,
Jerry


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100202/ec0787c5/attachment-0001.html 

From anthony.minessale at gmail.com  Tue Feb  2 15:38:41 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 2 Feb 2010 17:38:41 -0600
Subject: [Freeswitch-users] Attendant call transfer
In-Reply-To: <07247F72-DE53-4028-AFBF-BB5EB23FEA7B@freeswitch.org>
References: <3181A30B8C35AB4AA8577B78DDF461380668B5D3@nickel.mettonigroup.com>
	<07247F72-DE53-4028-AFBF-BB5EB23FEA7B@freeswitch.org>
Message-ID: <191c3a031002021538j3d7ab405w3cf727bf98c04b03@mail.gmail.com>

better still,
just download the latest code because we can only help you with the very
latest release.


On Tue, Feb 2, 2010 at 12:17 PM, Brian West <brian at freeswitch.org> wrote:

> Suneel,
> After printing 100 copies of this email It dawned on me that you failed to
> include any details about what SVN revision you're using.  If you can reply
> with that info I can promptly print out 100 more copies and see if we can
> find your problem.
>
> Thanks,
> Brian
> PS: just kidding about the printing part, but the svn rev would be helpful.
>
> On Feb 2, 2010, at 12:09 PM, Suneel Papineni wrote:
>
> Hi,
>
> I am trying to establish attendant call transfer using event sockets.
> 1.       A call has come into Freeswitch from an external Gateway  and
> this call is parked (it is configured to park all calls coming to
> freeswitch) {Caller A ? FS}
> 2.       Once the call is parked, I am sending a command to originate a
> call to another number connected to external gateway. {FS ? Caller B}.
> Call is established between FS and caller B. (?api originate
> sofia/external/<another number>@<gateway IP address> 9999?)
> 3.       On receiving event message as ?Application: Answer?, I am sending
> another command to bridge call between A & B. (?api uuid_bridge <callerA
> UUID> <caller UUID>?)
> 4.       With this call is established between A & B, but there is a huge
> delay (appox 30 secs).
>
> I believe that FS is still in the call and might be this is creating  delay
> (not sure).
>
> Could you please tell me if I am doing something wrong or process to
> achieve this scenario working.
>
> I tried in to transfer the call instead of bridging  using the command
> (?uuid_transfer <callerA UUID> intercept: <caller UUID> inline?), but the
> response is same as above with huge delay.
>
> Thanks & Regards
> Suneel
>
>
> *************************************************************************
> Please consider the environment before printing this e-mail
> *************************************************************************
> This email and any files transmitted with it are confidential and
> intended solely for the use of the individual or entity to whom they
> are addressed. If you have received this email in error please notify
> the system manager.  http://www.mettoni.com
>
> Mettoni Ltd
> Registered in England and Wales: 4485956
> 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN
> *************************************************************************
>
> _______________________________________________
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100202/d7e6417b/attachment.html 

From mbsip at gazeta.pl  Tue Feb  2 15:39:54 2010
From: mbsip at gazeta.pl (mbsip)
Date: Wed, 3 Feb 2010 00:39:54 +0100
Subject: [Freeswitch-users] voicemail_greeting_number - question
In-Reply-To: <87f2f3b91002011649p62ddff50o3f47bbb2b0be538a@mail.gmail.com>
References: <28f27f5d1001310905r41b16ca7r5ef1f236f76a070c@mail.gmail.com>
	<28f27f5d1002011522h31e03c0aoa71f26dabfa1d174@mail.gmail.com>
	<87f2f3b91002011649p62ddff50o3f47bbb2b0be538a@mail.gmail.com>
Message-ID: <28f27f5d1002021539l6a20be3bu774c5d7b32c791e1@mail.gmail.com>

>> > Hi ALL,
>> >
>> > I am playing around with VM and want to play user recorded greeting
>> > instead of default one.
>> > I've scaned wiki Mod_Voicemail and found proper parameter
>> > "voicemail_greeting_number".
>> > Unfortunately there is a lack of example hence i dont know if it is
>> > already working.
>> >
>> > Aforementioned param was placed in /conf/directory/default/1000.xml
>> > file (param name="voicemail_greeting_number", i tried many values)
>> > The effect is that the default greeting is played.
>> >
>> > Is this param embeeded into FS right now?
>> > How to use it?
>> > Is there any other place I should do the changes?
>> >
>> > I am running ?FreeSWITCH Version 1.0.trunk (16456).
>> >
>> > Thx in advance.
>> > Maciej
>> >
>>
>> Anyone knows how to use this param?
>>
>> Of course i may provide voicemail_default.db with proper greeting_path
>> manually but i am not sure if "voicemail_greeting_number" works the
>> same way and is somehow correlated?
>>
> Have you recorded vm greeting one, vm greeting two, etc. before changing the
> param?
> -MC

Here is what i did:
- recorded two files /tmp/vm1.wav and /tmp/vm2.wav
- inserted two entries into voicemail_default.db/voicemail_prefs
1000|XX.xx.XX.xx|1|/tmp/vm1.wav|
1000|XX.xx.XX.xx|2|/tmp/vm2.wav|
- inserted param
<param name="voicemail_greeting_number" value="1"/> after tests exchanged with
<param name="voicemail_greeting_number" value="2"/>

The result is that every call vm2.wav is played - so it only depends
on db entries (param is omitted)

Could sb tell me how to configure this or point me to detailed
description of this param.

Thx,
Maciej.


From jerry.richards at teotech.com  Tue Feb  2 15:55:07 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Tue, 2 Feb 2010 15:55:07 -0800
Subject: [Freeswitch-users] FS Core Dump
Message-ID: <40B8BCC90CB0486EA7730EB86BA1D1DB@greyhawk.tonecommander.com>

I spoke too soon.  Using Freeswitch version Feb 01, 2010 and Wanpipe Driver
ftp://ftp.sangoma.com/linux/custom/DavidYS/wanpipe-3.5.10.smg_pri.1.tgz, the
FS core will dump if I call my external cell phone, answer and then hangup.
See the pastebin: http://pastebin.freeswitch.org/12035.

This happens every time.
 
Best Regards,
Jerry
 



From max.bridgewater at gmail.com  Tue Feb  2 17:27:51 2010
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Tue, 2 Feb 2010 20:27:51 -0500
Subject: [Freeswitch-users] Ringback after before` Bridge
Message-ID: <c55d46951002021727v76e79adcg2281347ff8a0458f@mail.gmail.com>

Hi,

I'm trying to place a call to A and then bridge it to B. The problem I'm
having right now is that after A answers and while dialing B is being dialed
or rining, I want to send A a ringing tone. I don't succeed in doing this.
No tone/ringback is being sent to A.  Here is what i did using ESL:

 api originate
{ringback=\'%(400,200,400,450);%(400,2200,400,450)\',transfer_ringback=\'%(400,200,400,450);%(400,2200,400,450)\',origination_caller_id_number=4156781020}sofia/gateway/voipms/4152309090
&park()

To bridge, I then send the message:

sendmsg e9dae14c-e473-466e-9d65-704e36a82e5f
call-command: execute
execute-app-name: bridge
execute-app-arg:
{{ringback=\'%(400,200,400,450);%(400,2200,400,450\'},origination_caller_id_number=4152309090
}sofia/gateway/voipms/4156781020

any idea?

Max.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100202/a6da8143/attachment.html 

From moises.silva at gmail.com  Tue Feb  2 17:42:20 2010
From: moises.silva at gmail.com (Moises Silva)
Date: Tue, 2 Feb 2010 20:42:20 -0500
Subject: [Freeswitch-users] FS Core Dump
In-Reply-To: <40B8BCC90CB0486EA7730EB86BA1D1DB@greyhawk.tonecommander.com>
References: <40B8BCC90CB0486EA7730EB86BA1D1DB@greyhawk.tonecommander.com>
Message-ID: <c4d05cbe1002021742m546286bcxc96930990e57d9e6@mail.gmail.com>

On Tue, Feb 2, 2010 at 6:55 PM, Jerry Richards
<jerry.richards at teotech.com>wrote:

> I spoke too soon.  Using Freeswitch version Feb 01, 2010 and Wanpipe Driver
> ftp://ftp.sangoma.com/linux/custom/DavidYS/wanpipe-3.5.10.smg_pri.1.tgz,
> the
> FS core will dump if I call my external cell phone, answer and then hangup.
> See the pastebin: http://pastebin.freeswitch.org/12035.
>


Make sure you have latest revision of openzap, a few revisions ago I saw a
commit that set the zchan to null on hangup and you have a core dump on
hangup where the zchan is null, however on the latest trunk zchan is no
longer set to null on SIGEVENT_STOP ... suspicious ... if you can reproduce
with latest openzap trunk (at least revision 1021) then let me know and I
will take a look.

 --
Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100202/d6c7e18c/attachment.html 

From msc at freeswitch.org  Tue Feb  2 19:19:24 2010
From: msc at freeswitch.org (Michael S Collins)
Date: Tue, 2 Feb 2010 19:19:24 -0800
Subject: [Freeswitch-users] Ringback after before` Bridge
In-Reply-To: <c55d46951002021727v76e79adcg2281347ff8a0458f@mail.gmail.com>
References: <c55d46951002021727v76e79adcg2281347ff8a0458f@mail.gmail.com>
Message-ID: <91DD5271-AE00-454A-A5E5-9AA933E0B459@freeswitch.org>

Try the transfer_ringback var. Check the wiki for details.
-MC

Sent from my iPhone

On Feb 2, 2010, at 5:27 PM, Max Bridgewater  
<max.bridgewater at gmail.com> wrote:

> Hi,
>
> I'm trying to place a call to A and then bridge it to B. The problem  
> I'm having right now is that after A answers and while dialing B is  
> being dialed or rining, I want to send A a ringing tone. I don't  
> succeed in doing this. No tone/ringback is being sent to A.  Here is  
> what i did using ESL:
>
>  api originate {ringback=\'%(400,200,400,450); 
> %(400,2200,400,450)\',transfer_ringback=\'%(400,200,400,450); 
> %(400,2200,400,450)\',origination_caller_id_number=4156781020}sofia/ 
> gateway/voipms/4152309090 &park()
>
> To bridge, I then send the message:
>
> sendmsg e9dae14c-e473-466e-9d65-704e36a82e5f
> call-command: execute
> execute-app-name: bridge
> execute-app-arg: {{ringback=\'%(400,200,400,450); 
> %(400,2200,400,450\'},origination_caller_id_number=4152309090 }sofia/ 
> gateway/voipms/4156781020
>
> any idea?
>
> Max.
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org


From nagalenoj at gmail.com  Tue Feb  2 19:34:49 2010
From: nagalenoj at gmail.com (Nagalenoj H.)
Date: Wed, 3 Feb 2010 09:04:49 +0530
Subject: [Freeswitch-users] A leg disconnects after uuid_bridge?
Message-ID: <fe67a3341002021934m548cde89p4d48381c5abe02a0@mail.gmail.com>

Dear friends,
  In event socket, Why the session is closed for A leg when I do a
uuid_bridge with another uuid.
I've done the following operations(In nc),
  * Dial to he socket extension.
  * connect to the call.
  * Answered the call.
  * Bridged with extension X. When B leg terminates, A leg continues to be
alive.
  * Create an uuid.
  * Originated a call with origination_uuid and parked the leg.
  * I did a uuid_bridge for these 2 legs.
  * When B leg terminates the call, A leg is also getting exited.

I've tried setting the hangup_after_bridge to false explicitly. But, A leg
is getting exited.
I need it for further processing.

What is the way in which I can keep the A leg alive after uuid_bridge??

-- 
Regards,
Nagalenoj H.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100203/1ee810c1/attachment.html 

From anthony.minessale at gmail.com  Tue Feb  2 21:08:02 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 2 Feb 2010 23:08:02 -0600
Subject: [Freeswitch-users] A leg disconnects after uuid_bridge?
In-Reply-To: <fe67a3341002021934m548cde89p4d48381c5abe02a0@mail.gmail.com>
References: <fe67a3341002021934m548cde89p4d48381c5abe02a0@mail.gmail.com>
Message-ID: <191c3a031002022108ta22287ev40fc36420abba2e5@mail.gmail.com>

Tell one leg to execute intercept on the other instead.

On Feb 2, 2010 9:40 PM, "Nagalenoj H." <nagalenoj at gmail.com> wrote:

Dear friends,
  In event socket, Why the session is closed for A leg when I do a
uuid_bridge with another uuid.
I've done the following operations(In nc),
  * Dial to he socket extension.
  * connect to the call.
  * Answered the call.
  * Bridged with extension X. When B leg terminates, A leg continues to be
alive.
  * Create an uuid.
  * Originated a call with origination_uuid and parked the leg.
  * I did a uuid_bridge for these 2 legs.
  * When B leg terminates the call, A leg is also getting exited.

I've tried setting the hangup_after_bridge to false explicitly. But, A leg
is getting exited.
I need it for further processing.

What is the way in which I can keep the A leg alive after uuid_bridge??

-- 
Regards,
Nagalenoj H.

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100202/d95e34e7/attachment.html 

From thangappan143 at gmail.com  Tue Feb  2 21:22:25 2010
From: thangappan143 at gmail.com (Thangappan.M)
Date: Wed, 3 Feb 2010 10:52:25 +0530
Subject: [Freeswitch-users] Need Help to setup freeswitch with sangoma
	card
In-Reply-To: <7aa29e791001242150o3a582e2end69f1f5e9f1aef1c@mail.gmail.com>
References: <7aa29e791001192155l3c1f06e6w69769a69c9b761a1@mail.gmail.com>
	<7aa29e791001192313p73b20c8fk7e9b78e577c305a9@mail.gmail.com>
	<7aa29e791001200610m5a7b64cdx356e527ea7285b23@mail.gmail.com>
	<7aa29e791001202104n276ca48bo56d55ca82f3aac0f@mail.gmail.com>
	<7aa29e791001212045u20d85213wa0c8523761fee826@mail.gmail.com>
	<7aa29e791001222032t6f381e34x9f3c5eb099af9ba8@mail.gmail.com>
	<7aa29e791001242025y2adc91aes622bd22ea8d5ae1c@mail.gmail.com>
	<7aa29e791001242150o3a582e2end69f1f5e9f1aef1c@mail.gmail.com>
Message-ID: <7aa29e791002022122o289bc807p55f5ed20ccbd91b7@mail.gmail.com>

Any updates for this question. Still now I unable to make an outbound call
please help me...............

Or give the idea to change from boost to isdn?

On Mon, Jan 25, 2010 at 11:20 AM, Thangappan.M <thangappan143 at gmail.com>wrote:

> The following link have the openzap.conf,openzap.conf.xml ,smg_pri.conf, output of oz list
> and oz dump.
>
> http://www.pastebin.org/81929
>
>
>
> On Mon, Jan 25, 2010 at 9:55 AM, Thangappan.M <thangappan143 at gmail.com>wrote:
>
>> Here I mentioned the link which has the details of
>> /etc/wanpipe/smg_pri.conf
>> http://www.pastebin.org/81895
>>
>>
>> On Sat, Jan 23, 2010 at 10:02 AM, Thangappan.M <thangappan143 at gmail.com>wrote:
>>
>>> Updated the latest version of freeswitch ( 1.0.5-20100121-0400) and run
>>> the wanrouter then freeswitch. While executing the freeswtich it said the
>>> following error.
>>>
>>> [ERR] zap_io.c:2562 Error loading/usr/local/freeswitch/mod/ozmod_sangoma_boost.so
>>>
>>> [/usr/local/freeswitch/mod/ozmod_sangoma_boost.so: cannot open shared object
>>> file: No such file or directory]
>>> [ERR] zap_io.c:2722 can't find 'sangoma_boost
>>>
>>>
>>>
>>>
>>> Searched about this in freeswitch mailing list and found one post was
>>> there regarding the same problem. Finally found the problem. I missed to
>>> install the SCTP packages. Installed it and compiled the freeswitch again
>>> now the inbound call was landed on freeswitch.
>>>
>>> But I am unable to make a outbound call. When I was trying the following
>>> was get.
>>>
>>> freeswitch at internal> originate openzap/smg_prid/a/9940464753 at g1openzap/smg_prid/a/9940464753 at g1
>>> -ERR NORMAL_CIRCUIT_CONGESTION
>>>
>>> 2010-01-23 10:00:45.688854 [WARNING] ozmod_sangoma_boost.c:348 TX EVENT:
>>> CALL_START:(80) [w1g1] CSid=2 Seq=2 Cn=[FreeSWITCH] Cd=[9940464753]
>>> Ci=[0000000000] Rdnis=[]
>>> freeswitch at internal> 2010-01-23 10:00:46.709355 [WARNING]
>>> ozmod_sangoma_boost.c:1373 RX EVENT (N): CALL_START_NACK:(82) [w256g256]
>>> Rc=0 CSid=2 Seq=2
>>> 2010-01-23 10:00:46.709355 [WARNING] sangoma_boost_client.c:220 TX EVENT
>>> (N): CALL_START_NACK_ACK:(83) [w1g1] Rc=0 CSid=2 Seq=3
>>> 2010-01-23 10:00:46.709355 [ERR] mod_openzap.c:1162 No channels available
>>> 2010-01-23 10:00:46.709355 [ERR] switch_ivr_originate.c:2411 Cannot
>>> create outgoing channel of type [openzap] cause: [NORMAL_CIRCUIT_CONGESTION]
>>> 2010-01-23 10:00:46.709355 [DEBUG] switch_ivr_originate.c:3211 Originate
>>> Resulted in Error Cause: 34 [NORMAL_CIRCUIT_CONGESTION]
>>>
>>> Please help me...........
>>>
>>>
>>>
>>> On Fri, Jan 22, 2010 at 10:15 AM, Thangappan.M <thangappan143 at gmail.com>wrote:
>>>
>>>> The following link have the openzap.conf,openzap.conf.xml ,smg_prid.conf
>>>> , debug log of mod_openzap , oz list and oz dump 1 output.
>>>>
>>>> http://pastebin.org/80095
>>>>
>>>>
>>>>
>>>> On Thu, Jan 21, 2010 at 10:34 AM, Thangappan.M <thangappan143 at gmail.com
>>>> > wrote:
>>>>
>>>>> OpenZap is loading the ss7 signalling type. As per your concern openzap
>>>>> does not know the details of the signalling then how it is loading the
>>>>> ss7_boost libraries?
>>>>>
>>>>> FreeSWITCH log:
>>>>> -----------------------------
>>>>> 2010-01-21 10:10:46.707844 [INFO] zap_io.c:2374 Configured 30
>>>>> channel(s)
>>>>> 2010-01-21 10:10:46.708600 [INFO] zap_io.c:2468 Loading SIG from
>>>>> /usr/local/freeswitch/mod/ozmod_ss7_boost.so
>>>>> 2010-01-21 10:10:46.709031 [INFO] zap_io.c:2584 auto-loaded 'ss7_boost'
>>>>> 2010-01-21 10:10:46.709466 [DEBUG] ss7_boost_client.c:124 Creating L=
>>>>> 127.0.0.65:53000 R=127.0.0.66:53000
>>>>> 2010-01-21 10:10:46.709834 [DEBUG] ss7_boost_client.c:124 Creating L=
>>>>> 127.0.0.65:53001 R=127.0.0.66:53001
>>>>> 2010-01-21 10:10:46.710424 [WARNING] ss7_boost_client.c:244 TX EVENT
>>>>> (P): SYSTEM_RESTART:(87) [w1g1] Rc=0 CSid=0 Seq=0
>>>>>
>>>>> The signalling type might be anything but when I used the oz list
>>>>> command it showed the span details. But I am unable to make a inbound and
>>>>> outbound call.
>>>>>
>>>>> Outbound call result:
>>>>> ============
>>>>> >  originate openzap/smg_prid/a/9940464753
>>>>> openzap/smg_prid/a/9843171457
>>>>> -ERR NORMAL_CIRCUIT_CONGESTION
>>>>>
>>>>> 2010-01-21 10:26:14.304816 [CRIT] ozmod_ss7_boost.c:244 SPAN is not
>>>>> online.
>>>>> freeswitch at internal> 2010-01-21 10:26:14.304816 [ERR]
>>>>> mod_openzap.c:1043 No channels available
>>>>> 2010-01-21 10:26:14.304816 [ERR] switch_ivr_originate.c:1510 Cannot
>>>>> create outgoing channel of type [openzap] cause: [NORMAL_CIRCUIT_CONGESTION]
>>>>> 2010-01-21 10:26:14.304816 [DEBUG] switch_ivr_originate.c:2138
>>>>> Originate Resulted in Error Cause: 34 [NORMAL_CIRCUIT_CONGESTION]
>>>>>
>>>>> Inbound call result:
>>>>> -----------------------------
>>>>>
>>>>>      I made incoming call for the dial plan which is specified in the
>>>>> earlier post at that time it said the number is busy. We did the packet
>>>>> capture using the following command.
>>>>>
>>>>> wanpipemon -i w1g1 -pcap -pcap file isdn.pcap -port ISDN -full -systime
>>>>> -c trd
>>>>>
>>>>> Here I attached the pcap file for that.
>>>>>
>>>>>
>>>>> Where I did mistake or Did I miss any thing to do?
>>>>> Please help me.......
>>>>>
>>>>>
>>>>>
>>>>> On Wed, Jan 20, 2010 at 7:40 PM, Thangappan.M <thangappan143 at gmail.com
>>>>> > wrote:
>>>>>
>>>>>>
>>>>>>       I noticed the 'oz list' output  in that span type is 'ss7
>>>>>> (boost)'. How can I change this to isdn?
>>>>>>
>>>>>>
>>>>>>
>>>>>> On Wed, Jan 20, 2010 at 12:43 PM, Thangappan.M <
>>>>>> thangappan143 at gmail.com> wrote:
>>>>>>
>>>>>>> I found the error in it. The file name is used as openzap.conf.xml (
>>>>>>> smg_prid is specified here) and another file name as openzap.conf.wiki.xml (
>>>>>>> PRI_1 span is specified here ). FreeSWITCH referred the PRI_1 span from
>>>>>>> openzap.conf.wiki.xml  file.
>>>>>>>
>>>>>>> Now the another problem is raised here.
>>>>>>> When I was using oz list command , the details of the smg_prid shown.
>>>>>>> When I was using 'oz dump smg_prid' command it shows all the channels'
>>>>>>> details. But all the channels' states are DOWN. why? How can I make it the
>>>>>>> states to UP?
>>>>>>>
>>>>>>> When I was making the call , the number is busy message was get. The
>>>>>>> call was not  at all landed to the freeswitch.
>>>>>>>
>>>>>>> Dial plan Example:
>>>>>>> -------------------------------
>>>>>>>  <extension name="PRI_Testing">
>>>>>>>         <condition field="destination_number"
>>>>>>> expression="^39114600$">
>>>>>>>                 <action application="playback"
>>>>>>> data="ivr-welcome_to_freeswitch"/>
>>>>>>>         </condition>
>>>>>>>   </extension>
>>>>>>>
>>>>>>> Please help me...........
>>>>>>>
>>>>>>> *Output Reference:*
>>>>>>> http://pastebin.org/79074
>>>>>>>
>>>>>>>
>>>>>>> On Wed, Jan 20, 2010 at 11:25 AM, Thangappan.M <
>>>>>>> thangappan143 at gmail.com> wrote:
>>>>>>>
>>>>>>>> Dear all,
>>>>>>>>
>>>>>>>>          I have successfully configured wanpipe with freeswitch.
>>>>>>>> When I was the running wancfg_fs script the following files openzap.conf ,
>>>>>>>> autoload_confg/openzap.conf.xml , /etc/wanpipe/wanpipe1.xml, smg_pri.conf
>>>>>>>> are created.
>>>>>>>>
>>>>>>>>          I started the wanrouter command then executed the
>>>>>>>> freeswitch.
>>>>>>>>          When I was executing freeswitch mod_openzap.c said the
>>>>>>>> error as  "Error  for finding the span id. name:PRI_1".
>>>>>>>>          But in the openzap.conf and openzap.conf.xml files the span
>>>>>>>> name is smg_prid.
>>>>>>>>
>>>>>>>>          Why the freeswitch is referring the span name as PRI_1 ?
>>>>>>>>          Whether this has to configured in anywhere?
>>>>>>>>
>>>>>>>>          In the freeswitch CLI using oz command I tried to dump the
>>>>>>>> PRI_1 span id but it said te error  as "PRI_1 is not found".  When I was
>>>>>>>> trying the command 'oz dump smg_prid'  all the channel states and details
>>>>>>>> shown.
>>>>>>>>
>>>>>>>>         It seems that smg_prid span configured in openzap perfectly
>>>>>>>> (Its my assumption). Then Why freeswitch is referring the span name as
>>>>>>>> PRI_1.
>>>>>>>>
>>>>>>>> DID I MAKE ANY MISTAKE OR DID I MISS ANYTHING TO DO?
>>>>>>>>
>>>>>>>> Could anyone please help me?
>>>>>>>>
>>>>>>>> REFERENCE:
>>>>>>>>
>>>>>>>> openzap.conf
>>>>>>>> [span wanpipe smg_prid]
>>>>>>>> name => smg_prid
>>>>>>>> trunk_type =>e1
>>>>>>>> b-channel => 1:1-15
>>>>>>>> b-channel => 1:17-31
>>>>>>>>
>>>>>>>>
>>>>>>>> openzap.conf.xml
>>>>>>>> <configuration name="openzap.conf" description="OpenZAP
>>>>>>>> Configuration">
>>>>>>>>   <settings>
>>>>>>>>     <param name="debug" value="0"/>
>>>>>>>>     <!--<param name="hold-music" value="$${moh_uri}"/>-->
>>>>>>>>     <!--<param name="enable-analog-option" value="call-swap"/>-->
>>>>>>>>     <!--<param name="enable-analog-option" value="3-way"/>-->
>>>>>>>>   </settings>
>>>>>>>>   <boost_spans>
>>>>>>>>         <span id="smg_prid">
>>>>>>>>         <!--<param name="hold-music" value="$${moh_uri}"/>-->
>>>>>>>>         <param name="dialplan" value="XML"/>
>>>>>>>>         <param name="context" value="default"/>
>>>>>>>>          <!-- regex to stop dialing when it matches -->
>>>>>>>>         <!--<param name="dial-regex" value="5555"/>-->
>>>>>>>>         <!-- regex to stop dialing when it does not match -->
>>>>>>>>         <!--<param name="fail-dial-regex" value="^5"/>-->
>>>>>>>>         </span>
>>>>>>>>  </boost_spans>
>>>>>>>>  <analog_spans>
>>>>>>>>  </analog_spans>
>>>>>>>> </configuration>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> --
>>>>>>>> Regards,
>>>>>>>> Thangappan.M
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> --
>>>>>>> Regards,
>>>>>>> Thangappan.M
>>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>> Regards,
>>>>>> Thangappan.M
>>>>>>
>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> Regards,
>>>>> Thangappan.M
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> Regards,
>>>> Thangappan.M
>>>>
>>>
>>>
>>>
>>> --
>>> Regards,
>>> Thangappan.M
>>>
>>
>>
>>
>> --
>> Regards,
>> Thangappan.M
>>
>
>
>
> --
> Regards,
> Thangappan.M
>



-- 
Regards,
Thangappan.M
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100203/7b84ebe7/attachment-0001.html 

From nagalenoj at gmail.com  Wed Feb  3 01:12:03 2010
From: nagalenoj at gmail.com (Nagalenoj H.)
Date: Wed, 3 Feb 2010 14:42:03 +0530
Subject: [Freeswitch-users] A leg disconnects after uuid_bridge?
In-Reply-To: <191c3a031002022108ta22287ev40fc36420abba2e5@mail.gmail.com>
References: <fe67a3341002021934m548cde89p4d48381c5abe02a0@mail.gmail.com>
	<191c3a031002022108ta22287ev40fc36420abba2e5@mail.gmail.com>
Message-ID: <fe67a3341002030112yeea379u90c22c482614c57@mail.gmail.com>

I've used intercept application instead of uuid_bridge. I got the call
bridged with the given uuid. But, similar to uuid_bridge, A leg is getting
disconnected when B leg terminates.
I've did the same steps posted above, but used intercept instead of
uuid_bridge. Am I right?!

Tried using the hangup_after_bridge=false, but didn't see any difference.

On Wed, Feb 3, 2010 at 10:38 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> Tell one leg to execute intercept on the other instead.
>
> On Feb 2, 2010 9:40 PM, "Nagalenoj H." <nagalenoj at gmail.com> wrote:
>
> Dear friends,
>   In event socket, Why the session is closed for A leg when I do a
> uuid_bridge with another uuid.
> I've done the following operations(In nc),
>   * Dial to he socket extension.
>   * connect to the call.
>   * Answered the call.
>   * Bridged with extension X. When B leg terminates, A leg continues to be
> alive.
>   * Create an uuid.
>   * Originated a call with origination_uuid and parked the leg.
>   * I did a uuid_bridge for these 2 legs.
>   * When B leg terminates the call, A leg is also getting exited.
>
> I've tried setting the hangup_after_bridge to false explicitly. But, A leg
> is getting exited.
> I need it for further processing.
>
> What is the way in which I can keep the A leg alive after uuid_bridge??
>
> --
> Regards,
> Nagalenoj H.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100203/5cc696ce/attachment.html 

From Suneel.Papineni at mettoni.com  Wed Feb  3 02:56:21 2010
From: Suneel.Papineni at mettoni.com (Suneel Papineni)
Date: Wed, 3 Feb 2010 10:56:21 -0000
Subject: [Freeswitch-users] Attendant call transfer
In-Reply-To: <07247F72-DE53-4028-AFBF-BB5EB23FEA7B@freeswitch.org>
References: <3181A30B8C35AB4AA8577B78DDF461380668B5D3@nickel.mettonigroup.com>
	<07247F72-DE53-4028-AFBF-BB5EB23FEA7B@freeswitch.org>
Message-ID: <3181A30B8C35AB4AA8577B78DDF461380668B6E7@nickel.mettonigroup.com>

Hi Brian,

 

My apologies  for not sending required information.

 

I am using freeswitch 1.0.5_20100104-0400 and running on a Windows XP machine. I have written a .NET application to communicate with FS through Event Sockets.

 

In the scenario, once call is connected between Caller A and Caller B, I am expecting FS to come out of loop. 

 

Thanks in advance for helping me.

 

Thanks & Regards

Suneel

 

 

From: Brian West [mailto:brian at freeswitch.org] 
Sent: 02 February 2010 18:17
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Attendant call transfer

 

Suneel,

            After printing 100 copies of this email It dawned on me that you failed to include any details about what SVN revision you're using.  If you can reply with that info I can promptly print out 100 more copies and see if we can find your problem.

 

Thanks,

Brian

PS: just kidding about the printing part, but the svn rev would be helpful.

 

On Feb 2, 2010, at 12:09 PM, Suneel Papineni wrote:





Hi,

 

I am trying to establish attendant call transfer using event sockets.

1.       A call has come into Freeswitch from an external Gateway  and this call is parked (it is configured to park all calls coming to freeswitch) {Caller A ? FS}

2.       Once the call is parked, I am sending a command to originate a call to another number connected to external gateway. {FS ? Caller B}. Call is established between FS and caller B. ("api originate sofia/external/<another number>@<gateway IP address> 9999")

3.       On receiving event message as "Application: Answer", I am sending another command to bridge call between A & B. ("api uuid_bridge <callerA UUID> <caller UUID>")

4.       With this call is established between A & B, but there is a huge delay (appox 30 secs).

 

I believe that FS is still in the call and might be this is creating  delay (not sure).

 

Could you please tell me if I am doing something wrong or process to achieve this scenario working.

 

I tried in to transfer the call instead of bridging  using the command ("uuid_transfer <callerA UUID> intercept: <caller UUID> inline"), but the response is same as above with huge delay.

 

Thanks & Regards

Suneel

 

*************************************************************************
Please consider the environment before printing this e-mail
*************************************************************************
This email and any files transmitted with it are confidential and
intended solely for the use of the individual or entity to whom they
are addressed. If you have received this email in error please notify
the system manager.  http://www.mettoni.com
 
Mettoni Ltd
Registered in England and Wales: 4485956
9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN
*************************************************************************

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

 


*************************************************************************
Please consider the environment before printing this e-mail
*************************************************************************
This email and any files transmitted with it are confidential and
intended solely for the use of the individual or entity to whom they
are addressed. If you have received this email in error please notify
the system manager.  http://www.mettoni.com

Mettoni Ltd
Registered in England and Wales: 4485956
9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN
*************************************************************************

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100203/6871558f/attachment.html 

From nicolas at medularis.com  Wed Feb  3 03:11:49 2010
From: nicolas at medularis.com (Nicolas Brenner)
Date: Wed, 3 Feb 2010 08:11:49 -0300
Subject: [Freeswitch-users] Ringback after before` Bridge
In-Reply-To: <91DD5271-AE00-454A-A5E5-9AA933E0B459@freeswitch.org>
References: <c55d46951002021727v76e79adcg2281347ff8a0458f@mail.gmail.com>
	<91DD5271-AE00-454A-A5E5-9AA933E0B459@freeswitch.org>
Message-ID: <1b46b4e81002030311x48db5d8et4822bd8bd97c0094@mail.gmail.com>

Not sure about using ESL, but here's an example on how to do it with Lua:

http://wiki.freeswitch.org/wiki/Bridging_two_calls_with_retry

Simplifying the code on that example though, here are the basics:

session1 = freeswitch.Session(ostr1);
if (session1:ready()) then
-- Set ringback
session1:setVariable("ringback", "%(2000,4000,440,480)");

session2 = freeswitch.Session(ostr2, session1);
if (session2:ready()) then
freeswitch.bridge(session1, session2);

-- Hangup session2 if session1 is over
if (session2:ready()) then session2:hangup(); end
end
 -- hangup when done
    if (session1:ready()) then session1:hangup(); end
end


ostr1 and ostr2 should be your dialstrings, something like:
{ignore_early_media=true,originate_timeout=90,hangup_after_bridge=true}sofia/gateway/yourgateway/phonenumber


On Wed, Feb 3, 2010 at 12:19 AM, Michael S Collins <msc at freeswitch.org>wrote:

> Try the transfer_ringback var. Check the wiki for details.
> -MC
>
> Sent from my iPhone
>
> On Feb 2, 2010, at 5:27 PM, Max Bridgewater
> <max.bridgewater at gmail.com> wrote:
>
> > Hi,
> >
> > I'm trying to place a call to A and then bridge it to B. The problem
> > I'm having right now is that after A answers and while dialing B is
> > being dialed or rining, I want to send A a ringing tone. I don't
> > succeed in doing this. No tone/ringback is being sent to A.  Here is
> > what i did using ESL:
> >
> >  api originate {ringback=\'%(400,200,400,450);
> > %(400,2200,400,450)\',transfer_ringback=\'%(400,200,400,450);
> > %(400,2200,400,450)\',origination_caller_id_number=4156781020}sofia/
> > gateway/voipms/4152309090 &park()
> >
> > To bridge, I then send the message:
> >
> > sendmsg e9dae14c-e473-466e-9d65-704e36a82e5f
> > call-command: execute
> > execute-app-name: bridge
> > execute-app-arg: {{ringback=\'%(400,200,400,450);
> > %(400,2200,400,450\'},origination_caller_id_number=4152309090 }sofia/
> > gateway/voipms/4156781020
> >
> > any idea?
> >
> > Max.
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> > users
> > http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100203/9e7f221e/attachment-0001.html 

From moizchinoy at gmail.com  Wed Feb  3 04:12:02 2010
From: moizchinoy at gmail.com (Moiz Chinoy)
Date: Wed, 3 Feb 2010 16:12:02 +0400
Subject: [Freeswitch-users] Dialogic Board (JCT & DM3) Supported?
In-Reply-To: <D7BC4112-CA0B-4825-8ADA-A8F228C2F201@jerris.com>
References: <29b888f81002010308uf875b53h838453b6e77e9a9e@mail.gmail.com>
	<D7BC4112-CA0B-4825-8ADA-A8F228C2F201@jerris.com>
Message-ID: <29b888f81002030412x7856f1d4p3612ce49c9d23d8f@mail.gmail.com>

Sometime back it was posted that following cards are supported by
Freeswitch. Can anyone please guide me. We have a JCT and DMV card
available so I was thinking if we can do anything useful with it.

Analog cards: D/41JCT-LS, D/120JCT-LS  (jct serie)
Digital cards: D/600JCT-1E1 and DMV serie.

On Tue, Feb 2, 2010 at 12:14 PM, Michael Jerris <mike at jerris.com> wrote:
> Nope
>
> On Feb 1, 2010, at 6:08 AM, Moiz Chinoy wrote:
>> Can anyone please advise that whether Dialogic boards (JCT and DM3)
>> are supported by FS.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Regards,
Moiz Chinoy.


From steveu at coppice.org  Wed Feb  3 04:38:11 2010
From: steveu at coppice.org (Steve Underwood)
Date: Wed, 03 Feb 2010 20:38:11 +0800
Subject: [Freeswitch-users] Dialogic Board (JCT & DM3) Supported?
In-Reply-To: <29b888f81002030412x7856f1d4p3612ce49c9d23d8f@mail.gmail.com>
References: <29b888f81002010308uf875b53h838453b6e77e9a9e@mail.gmail.com>	<D7BC4112-CA0B-4825-8ADA-A8F228C2F201@jerris.com>
	<29b888f81002030412x7856f1d4p3612ce49c9d23d8f@mail.gmail.com>
Message-ID: <4B696E33.5040103@coppice.org>

Hi Moiz,

On 02/03/2010 08:12 PM, Moiz Chinoy wrote:
> Sometime back it was posted that following cards are supported by
> Freeswitch. Can anyone please guide me. We have a JCT and DMV card
> available so I was thinking if we can do anything useful with it.
>
> Analog cards: D/41JCT-LS, D/120JCT-LS  (jct serie)
> Digital cards: D/600JCT-1E1 and DMV serie.
>    
"For it is written" :-)

Someone randomly posting rubbish to the mailing list doesn't make it 
true. Those Dialogic cards have never been supported by Freeswitch.
> On Tue, Feb 2, 2010 at 12:14 PM, Michael Jerris<mike at jerris.com>  wrote:
>    
>> Nope
>>
>> On Feb 1, 2010, at 6:08 AM, Moiz Chinoy wrote:
>>      
>>> Can anyone please advise that whether Dialogic boards (JCT and DM3)
>>> are supported by FS.
>>>        
Steve



From dftoro at yahoo.com  Wed Feb  3 05:42:38 2010
From: dftoro at yahoo.com (Diego Toro)
Date: Wed, 3 Feb 2010 05:42:38 -0800 (PST)
Subject: [Freeswitch-users] Dialogic Board (JCT & DM3) Supported?
In-Reply-To: <29b888f81002030412x7856f1d4p3612ce49c9d23d8f@mail.gmail.com>
Message-ID: <963917.92014.qm@web33505.mail.mud.yahoo.com>

This would be a good opportunity to start the support of this hardware. The problem is that the "System Release" (API) has cost of licensing, this to avoid having to start from scratch. Now, this compared with good quality Sangoma hardware, Dialogic would not be competitive. Would be good to ask to Eicom people if they would be interested in the subject.

Diego Toro
http://lacarretade.blogspot.com/


--- On Wed, 2/3/10, Moiz Chinoy <moizchinoy at gmail.com> wrote:

> From: Moiz Chinoy <moizchinoy at gmail.com>
> Subject: Re: [Freeswitch-users] Dialogic Board (JCT & DM3) Supported?
> To: freeswitch-users at lists.freeswitch.org
> Date: Wednesday, February 3, 2010, 7:12 AM
> Sometime back it was posted that
> following cards are supported by
> Freeswitch. Can anyone please guide me. We have a JCT and
> DMV card
> available so I was thinking if we can do anything useful
> with it.
> 
> Analog cards: D/41JCT-LS, D/120JCT-LS? (jct serie)
> Digital cards: D/600JCT-1E1 and DMV serie.
> 
> On Tue, Feb 2, 2010 at 12:14 PM, Michael Jerris <mike at jerris.com>
> wrote:
> > Nope
> >
> > On Feb 1, 2010, at 6:08 AM, Moiz Chinoy wrote:
> >> Can anyone please advise that whether Dialogic
> boards (JCT and DM3)
> >> are supported by FS.
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> 
> 
> 
> -- 
> Regards,
> Moiz Chinoy.
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 


      


From msc at freeswitch.org  Wed Feb  3 06:05:16 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 3 Feb 2010 06:05:16 -0800
Subject: [Freeswitch-users] A leg disconnects after uuid_bridge?
In-Reply-To: <fe67a3341002030112yeea379u90c22c482614c57@mail.gmail.com>
References: <fe67a3341002021934m548cde89p4d48381c5abe02a0@mail.gmail.com>
	<191c3a031002022108ta22287ev40fc36420abba2e5@mail.gmail.com>
	<fe67a3341002030112yeea379u90c22c482614c57@mail.gmail.com>
Message-ID: <87f2f3b91002030605j7a1f9a02g665ba043c6ed7f7b@mail.gmail.com>

On Wed, Feb 3, 2010 at 1:12 AM, Nagalenoj H. <nagalenoj at gmail.com> wrote:

> I've used intercept application instead of uuid_bridge. I got the call
> bridged with the given uuid. But, similar to uuid_bridge, A leg is getting
> disconnected when B leg terminates.
> I've did the same steps posted above, but used intercept instead of
> uuid_bridge. Am I right?!
>
> Tried using the hangup_after_bridge=false, but didn't see any difference.
>
> Are you on the latest SVN of FreeSWITCH? Be sure to "make current" and try
again. If the issue remains then capture a complete debug log of the call
from start to finish and also pastebin your script so others can test and
analyze.
-MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100203/45071da9/attachment.html 

From marketing at cluecon.com  Wed Feb  3 06:28:04 2010
From: marketing at cluecon.com (Michael Collins)
Date: Wed, 3 Feb 2010 06:28:04 -0800
Subject: [Freeswitch-users] ClueCon MMX - Call For Speakers!
Message-ID: <87f2f3b91002030628n3b3bf512x68947758cf042ea@mail.gmail.com>

Hello everyone!

We are gearing up for ClueCon 2010 in August later this year. We are making
the necessary arrangements for the conference facilities and rooming. Things
are beginning to fall into place. Now we need to hear from you.

We would like to put out a call for speakers for this year's event. Please
contact us if you or your organization would like to give a presentation at
ClueCon this year. We want to get the speakers scheduled as early as
possible. Keep in mind that those organizations which sponsor ClueCon will
be given the highest priority when it comes to scheduling. Please contact
Brian West to discuss sponsorship opportunities for this year's event.

We also would like to hear from the conference attendees: what would you
like to see this year? Please give us your input. ClueCon is, of course, "By
Developers, For Developers." However, developers come in all shapes and
sizes and we would like have something for everyone. Please tell us what
would make ClueCon MMX the best conference of the year!

Stay tuned for more announcements. We look forward to hearing from you and
seeing everyone this August in Chicago.

-ClueCon team
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100203/24dc1efc/attachment.html 

From msc at freeswitch.org  Wed Feb  3 06:55:47 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 3 Feb 2010 06:55:47 -0800
Subject: [Freeswitch-users] Let's buy the FreeSWITCH developers dinner!
Message-ID: <87f2f3b91002030655x315159ads102f269dfdab200d@mail.gmail.com>

Hello all!

This is a reminder that next week the FreeSWITCH development team is
gathering together in one location for the final push to get version 1.0.5
released. You can help facilitate the timely release of the latest version
by helping to buy dinner for the FreeSWITCH developers. Remember, it's not
just the three core developers who are meeting together. There will be eight
FreeSWITCH team members gathering. Let's all pitch in a few dollars each and
give them a nice dinner! It's a great way to say thanks for all the hard
work they've done: building FreeSWITCH, answering questions on the mailing
list, and spending many hours in the IRC channel.

You can use the PayPal link on the main website. (http://www.freeswitch.org)
Alternatively, if PayPal is not available to you then please contact Brian
West (brian at freeswitch.org) to discuss alternate ways of donating.

Let's really pull together and support the guys for all of their hard work.
A nice meal paid for by the community would be greatly appreciated!

-Michael
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100203/277e82ea/attachment.html 

From moises.silva at gmail.com  Wed Feb  3 07:06:59 2010
From: moises.silva at gmail.com (Moises Silva)
Date: Wed, 3 Feb 2010 10:06:59 -0500
Subject: [Freeswitch-users] Need Help to setup freeswitch with sangoma
	card
In-Reply-To: <7aa29e791002022122o289bc807p55f5ed20ccbd91b7@mail.gmail.com>
References: <7aa29e791001192155l3c1f06e6w69769a69c9b761a1@mail.gmail.com>
	<7aa29e791001192313p73b20c8fk7e9b78e577c305a9@mail.gmail.com>
	<7aa29e791001200610m5a7b64cdx356e527ea7285b23@mail.gmail.com>
	<7aa29e791001202104n276ca48bo56d55ca82f3aac0f@mail.gmail.com>
	<7aa29e791001212045u20d85213wa0c8523761fee826@mail.gmail.com>
	<7aa29e791001222032t6f381e34x9f3c5eb099af9ba8@mail.gmail.com>
	<7aa29e791001242025y2adc91aes622bd22ea8d5ae1c@mail.gmail.com>
	<7aa29e791001242150o3a582e2end69f1f5e9f1aef1c@mail.gmail.com>
	<7aa29e791002022122o289bc807p55f5ed20ccbd91b7@mail.gmail.com>
Message-ID: <c4d05cbe1002030706x61342ee4hc9d5fe0389d123fc@mail.gmail.com>

Boost has been working fine, so there is no point in switching.

Try removing the d-channel from openzap.conf, boost spans do not need
d-channel declared, because that is done through the signaling binary
(sangoma_prid), declaring it is causing 2 different processes to open the
same channel, at best, data will be missing from one process or the other.

If you still cannot make calls after removing the d-channel from
openzap.conf (and restarting FreeSWITCH and sangoma_prid using smg_ctrl
script), then pastebin a debug log for both FreeSWITCH and sangoma_prid
after a single call attempt.

-- 
Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com

On Wed, Feb 3, 2010 at 12:22 AM, Thangappan.M <thangappan143 at gmail.com>wrote:

> Any updates for this question. Still now I unable to make an outbound call
> please help me...............
>
> Or give the idea to change from boost to isdn?
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100203/847db260/attachment.html 

From jerry.richards at teotech.com  Wed Feb  3 08:40:30 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Wed, 3 Feb 2010 08:40:30 -0800
Subject: [Freeswitch-users] FS Core Dump
In-Reply-To: <c4d05cbe1002021742m546286bcxc96930990e57d9e6@mail.gmail.com>
References: <40B8BCC90CB0486EA7730EB86BA1D1DB@greyhawk.tonecommander.com>
	<c4d05cbe1002021742m546286bcxc96930990e57d9e6@mail.gmail.com>
Message-ID: <C1EEC6F570BD4302ABDB7D4A0BEE7803@greyhawk.tonecommander.com>

Okay. I will get freeswitch-1.0.5-20100202-0400.tar.gz, which is one day
later than what I currently have.  I don't get openzap separately, because
it is included with this tarball.  True?
 
Thanks And Best Regards,
Jerry


  _____  

From: Moises Silva [mailto:moises.silva at gmail.com] 
Sent: Tuesday, February 02, 2010 5:42 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] FS Core Dump


On Tue, Feb 2, 2010 at 6:55 PM, Jerry Richards <jerry.richards at teotech.com>
wrote:


I spoke too soon.  Using Freeswitch version Feb 01, 2010 and Wanpipe Driver
ftp://ftp.sangoma.com/linux/custom/DavidYS/wanpipe-3.5.10.smg_pri.1.tgz, the
FS core will dump if I call my external cell phone, answer and then hangup.
See the pastebin: http://pastebin.freeswitch.org/12035.




Make sure you have latest revision of openzap, a few revisions ago I saw a
commit that set the zchan to null on hangup and you have a core dump on
hangup where the zchan is null, however on the latest trunk zchan is no
longer set to null on SIGEVENT_STOP ... suspicious ... if you can reproduce
with latest openzap trunk (at least revision 1021) then let me know and I
will take a look.

 -- 
Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com


-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100203/cd871bd6/attachment-0001.html 

From m.sobkow at marketelsystems.com  Wed Feb  3 10:49:11 2010
From: m.sobkow at marketelsystems.com (Mark Sobkow)
Date: Wed, 03 Feb 2010 12:49:11 -0600
Subject: [Freeswitch-users] Has anyone had Freeswitch crash on loading
 mod_voicemail?
In-Reply-To: <4B5E4A35.3060803@marketelsystems.com>
References: <4B5E4A35.3060803@marketelsystems.com>
Message-ID: <4B69C527.4090808@marketelsystems.com>

Mark Sobkow wrote:
> I tried a "load mod_voicemail" in fs_cli, hoping to see what 
> configuration section it requested from Erlang, but instead of loading 
> the module, Freeswitch crashed without any error messages.  SVN 15188 
> built on Ubuntu Hardy 32-bit.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>   
Updated Freeswitch to svn16561 this morning, captured the backtrace, and 
attached it to JIRA as requested.

-- 
Mark Sobkow
Senior Developer
MarkeTel Multi-Line Dialing Systems LTD.
428 Victoria Ave
Regina, SK S4N-0P6
Toll-Free: 800-289-8616-X533
Local: 306-359-6893-X533
Fax: 306-359-6879
Email: m.sobkow at marketelsystems.com
Web: http://www.marketelsystems.com 



From anthony.minessale at gmail.com  Wed Feb  3 11:19:30 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 3 Feb 2010 13:19:30 -0600
Subject: [Freeswitch-users] Has anyone had Freeswitch crash on loading
	mod_voicemail?
In-Reply-To: <4B69C527.4090808@marketelsystems.com>
References: <4B5E4A35.3060803@marketelsystems.com>
	<4B69C527.4090808@marketelsystems.com>
Message-ID: <191c3a031002031119q1efc36bwd5484aa0d640682d@mail.gmail.com>

Clear bug in mod_erlang.  Author has been notified.


On Wed, Feb 3, 2010 at 12:49 PM, Mark Sobkow
<m.sobkow at marketelsystems.com>wrote:

> Mark Sobkow wrote:
> > I tried a "load mod_voicemail" in fs_cli, hoping to see what
> > configuration section it requested from Erlang, but instead of loading
> > the module, Freeswitch crashed without any error messages.  SVN 15188
> > built on Ubuntu Hardy 32-bit.
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> Updated Freeswitch to svn16561 this morning, captured the backtrace, and
> attached it to JIRA as requested.
>
> --
> Mark Sobkow
> Senior Developer
> MarkeTel Multi-Line Dialing Systems LTD.
> 428 Victoria Ave
> Regina, SK S4N-0P6
> Toll-Free: 800-289-8616-X533
> Local: 306-359-6893-X533
> Fax: 306-359-6879
> Email: m.sobkow at marketelsystems.com
> Web: http://www.marketelsystems.com
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100203/96aa9f73/attachment.html 

From jerry.richards at teotech.com  Wed Feb  3 11:23:05 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Wed, 3 Feb 2010 11:23:05 -0800
Subject: [Freeswitch-users] FS Core Dump
In-Reply-To: <C1EEC6F570BD4302ABDB7D4A0BEE7803@greyhawk.tonecommander.com>
References: <40B8BCC90CB0486EA7730EB86BA1D1DB@greyhawk.tonecommander.com><c4d05cbe1002021742m546286bcxc96930990e57d9e6@mail.gmail.com>
	<C1EEC6F570BD4302ABDB7D4A0BEE7803@greyhawk.tonecommander.com>
Message-ID: <F940A2D994FA41EF812D9956EEB9A039@greyhawk.tonecommander.com>

Okay.  I got the latest trunk at 9:39AM PST and FS does not crash when I
call my cell phone, answer, and hangup.  I am using the
<ftp://ftp.sangoma.com/linux/custom/DavidYS/wanpipe-3.5.10.smg_pri.1.tgz>
ftp://ftp.sangoma.com/linux/custom/DavidYS/wanpipe-3.5.10.smg_pri.1.tgz
driver.
 
Thanks and Best Regards,
Jerry
 



  _____  

From: Jerry Richards [mailto:jerry.richards at teotech.com] 
Sent: Wednesday, February 03, 2010 8:41 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] FS Core Dump


Okay. I will get freeswitch-1.0.5-20100202-0400.tar.gz, which is one day
later than what I currently have.  I don't get openzap separately, because
it is included with this tarball.  True?
 
Thanks And Best Regards,
Jerry


  _____  

From: Moises Silva [mailto:moises.silva at gmail.com] 
Sent: Tuesday, February 02, 2010 5:42 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] FS Core Dump


On Tue, Feb 2, 2010 at 6:55 PM, Jerry Richards <jerry.richards at teotech.com>
wrote:


I spoke too soon.  Using Freeswitch version Feb 01, 2010 and Wanpipe Driver
ftp://ftp.sangoma.com/linux/custom/DavidYS/wanpipe-3.5.10.smg_pri.1.tgz, the
FS core will dump if I call my external cell phone, answer and then hangup.
See the pastebin: http://pastebin.freeswitch.org/12035.




Make sure you have latest revision of openzap, a few revisions ago I saw a
commit that set the zchan to null on hangup and you have a core dump on
hangup where the zchan is null, however on the latest trunk zchan is no
longer set to null on SIGEVENT_STOP ... suspicious ... if you can reproduce
with latest openzap trunk (at least revision 1021) then let me know and I
will take a look.

 -- 
Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com


-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100203/a56882f6/attachment.html 

From m.sobkow at marketelsystems.com  Wed Feb  3 11:40:45 2010
From: m.sobkow at marketelsystems.com (Mark Sobkow)
Date: Wed, 03 Feb 2010 13:40:45 -0600
Subject: [Freeswitch-users] Has anyone had Freeswitch crash on loading
 mod_voicemail?
In-Reply-To: <4B5E4A35.3060803@marketelsystems.com>
References: <4B5E4A35.3060803@marketelsystems.com>
Message-ID: <4B69D13D.7030205@marketelsystems.com>

Mark Sobkow wrote:
> I tried a "load mod_voicemail" in fs_cli, hoping to see what 
> configuration section it requested from Erlang, but instead of loading 
> the module, Freeswitch crashed without any error messages.  SVN 15188 
> built on Ubuntu Hardy 32-bit.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>   
Did a little digging through the traceback for this problem, and it 
looks to me like Freeswitch is passing an invalid UUID to the code that 
tries to get the configuration from Erlang.

http://jira.freeswitch.org/browse/FSCORE-542



-- 
Mark Sobkow
Senior Developer
MarkeTel Multi-Line Dialing Systems LTD.
428 Victoria Ave
Regina, SK S4N-0P6
Toll-Free: 800-289-8616-X533
Local: 306-359-6893-X533
Fax: 306-359-6879
Email: m.sobkow at marketelsystems.com
Web: http://www.marketelsystems.com 



From fvillarroel at yahoo.com  Wed Feb  3 12:29:54 2010
From: fvillarroel at yahoo.com (FERNANDO VILLARROEL)
Date: Wed, 3 Feb 2010 12:29:54 -0800 (PST)
Subject: [Freeswitch-users] max calls from a gateway
Message-ID: <867692.53487.qm@web34301.mail.mud.yahoo.com>

Dear.

How i can do for limits inbound calls from a gateway:


My config is like this:

~sip_profiles/external/

gateway1.xml

<include>
    <gateway name="foo">
      <param name="username" value="foo"/>
      <param name="password" value="foo"/>
      <param name="realm" value="xx.xxx.xxx.xx"/>
      <param name="proxy" value="xx.xxx.xxx.xx/>
      <param name="register" value="false"/>
      <param name="context" value="public"/>
      <param name="max_calls" value="30"/>-->
    </gateway>
</include>

It?s fine?

Fernando


      


From john at acsol.net  Wed Feb  3 14:30:37 2010
From: john at acsol.net (John)
Date: Wed, 03 Feb 2010 15:30:37 -0700
Subject: [Freeswitch-users] PAP2T issue
Message-ID: <4B69F90D.1070503@acsol.net>

I can register SNOM phones fine, one the same network, I am trying to 
get a Linksys PAP2T-NA registered. Anyone have a configuration example? 
Thanks



From christian.loeschenkohl at xpirio.com  Wed Feb  3 14:33:17 2010
From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=)
Date: Wed, 03 Feb 2010 23:33:17 +0100
Subject: [Freeswitch-users] adding sip header without X-
Message-ID: <4B69F9AD.8090904@xpirio.com>

hello

do anybody know a way to add "Alert-Info: <http://www.notused.com>;info=alert-group;x-line-id=0"
as a custom sip header to the invite message?

<action application="set" data="sip_h_X-Alert-Info=<http://www.notused.com>\;info=alert-internal\;x-line-id=0"/>
works but "X-Alert-Info" as a header isn't usefull at all

the purpose is to distinguish internal and external calls in a pbx. it is also described here
http://wiki.snom.com/Web_Interface/V7/Preferences/Alert-Info_Ringer

there was also a posting on the list by Kristian Kielhofner who suggested a very flexible solution

-----
mail from: 2009-10-09 21:10

2)  Make the behavior configurable with a channel variable and/or
     sofia config option: {sip_pass_headers=all|none|X-MyCustomHeaderByName}
-----

br


-- 
Ing. Christian L?schenkohl
Technische Leitung, Forschung & Entwicklung VoIP

xpirio
Telekommunikation & Service GmbH
Lakeside B04
9020 Klagenfurt
Austria

T  +43 (0) 5 77 11 - 1000
F  +43 (0) 5 77 11 - 1002
E  christian.loeschenkohl at xpirio.com


From john at acsol.net  Wed Feb  3 15:23:05 2010
From: john at acsol.net (John)
Date: Wed, 03 Feb 2010 16:23:05 -0700
Subject: [Freeswitch-users] PAP2T issue
In-Reply-To: <4B69F90D.1070503@acsol.net>
References: <4B69F90D.1070503@acsol.net>
Message-ID: <4B6A0559.2040200@acsol.net>

Found two issues and solved the problem. By default, the PAP2T has 
provisioning turned on, and won't try to register if the provisioning 
server isn't found. Turned that off. Secondly, turned off Stun and 
enabled rport and it registered fine. Hope this helps someone else.

thanks John
On 2/3/2010 3:30 PM, John wrote:
> I can register SNOM phones fine, one the same network, I am trying to
> get a Linksys PAP2T-NA registered. Anyone have a configuration example?
> Thanks
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>    



From kristian.kielhofner at gmail.com  Wed Feb  3 15:55:48 2010
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Wed, 3 Feb 2010 18:55:48 -0500
Subject: [Freeswitch-users] adding sip header without X-
In-Reply-To: <4B69F9AD.8090904@xpirio.com>
References: <4B69F9AD.8090904@xpirio.com>
Message-ID: <2d9149cd1002031555o150b63a2hfd76dac20725b06a@mail.gmail.com>

Hello Christian,

  Reading through the code (sofia_glue.c) it looks like all you have
to do is set the alert_info channel variable.  I also don't think you
have to include the X- when using set with sip_h.  It's just always
good (RFC compliant) to prefix "custom" headers with X-.

  Regarding my suggestion for that configuration option/channel
variable.  That was only to be used on bridged channels to make it
configurable which headers FreeSWITCH passes from the a leg to the b
leg when using bridge.  A "proper" B2BUA (according to me) should
completely rewrite the outgoing INVITE, including stripping any custom
headers.

2010/2/3 Christian L?schenkohl <christian.loeschenkohl at xpirio.com>:
> hello
>
> do anybody know a way to add "Alert-Info: <http://www.notused.com>;info=alert-group;x-line-id=0"
> as a custom sip header to the invite message?
>
> <action application="set" data="sip_h_X-Alert-Info=<http://www.notused.com>\;info=alert-internal\;x-line-id=0"/>
> works but "X-Alert-Info" as a header isn't usefull at all
>
> the purpose is to distinguish internal and external calls in a pbx. it is also described here
> http://wiki.snom.com/Web_Interface/V7/Preferences/Alert-Info_Ringer
>
> there was also a posting on the list by Kristian Kielhofner who suggested a very flexible solution
>
> -----
> mail from: 2009-10-09 21:10
>
> 2) ?Make the behavior configurable with a channel variable and/or
> ? ? sofia config option: {sip_pass_headers=all|none|X-MyCustomHeaderByName}
> -----
>
> br
>
>
> --
> Ing. Christian L?schenkohl
> Technische Leitung, Forschung & Entwicklung VoIP
>
> xpirio
> Telekommunikation & Service GmbH
> Lakeside B04
> 9020 Klagenfurt
> Austria
>
> T ?+43 (0) 5 77 11 - 1000
> F ?+43 (0) 5 77 11 - 1002
> E ?christian.loeschenkohl at xpirio.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com


From christian.loeschenkohl at xpirio.com  Wed Feb  3 16:19:35 2010
From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=)
Date: Thu, 04 Feb 2010 01:19:35 +0100
Subject: [Freeswitch-users] adding sip header without X-
In-Reply-To: <2d9149cd1002031555o150b63a2hfd76dac20725b06a@mail.gmail.com>
References: <4B69F9AD.8090904@xpirio.com>
	<2d9149cd1002031555o150b63a2hfd76dac20725b06a@mail.gmail.com>
Message-ID: <4B6A1297.9090802@xpirio.com>

hello

thank you for your answer
setting only sip_h_Alert-Info was my first try, didn't work at all

i use trun rev. 16456 in proxy mode - if it helps
every header i add with X-... shows up in the invite to the sip endpoint (snom phone)

when i read sofia_glue.c i do see the function sofia_glue_set_extra_headers
which only adds header starting with X- or P- (line 1407 for me)

br

Kristian Kielhofner wrote:

> Hello Christian,
> 
>   Reading through the code (sofia_glue.c) it looks like all you have
> to do is set the alert_info channel variable.  I also don't think you
> have to include the X- when using set with sip_h.  It's just always
> good (RFC compliant) to prefix "custom" headers with X-.
> 
>   Regarding my suggestion for that configuration option/channel
> variable.  That was only to be used on bridged channels to make it
> configurable which headers FreeSWITCH passes from the a leg to the b
> leg when using bridge.  A "proper" B2BUA (according to me) should
> completely rewrite the outgoing INVITE, including stripping any custom
> headers.
> 
> 2010/2/3 Christian L?schenkohl <christian.loeschenkohl at xpirio.com>:
>> hello
>>
>> do anybody know a way to add "Alert-Info: <http://www.notused.com>;info=alert-group;x-line-id=0"
>> as a custom sip header to the invite message?
>>
>> <action application="set" data="sip_h_X-Alert-Info=<http://www.notused.com>\;info=alert-internal\;x-line-id=0"/>
>> works but "X-Alert-Info" as a header isn't usefull at all
>>
>> the purpose is to distinguish internal and external calls in a pbx. it is also described here
>> http://wiki.snom.com/Web_Interface/V7/Preferences/Alert-Info_Ringer
>>
>> there was also a posting on the list by Kristian Kielhofner who suggested a very flexible solution
>>
>> -----
>> mail from: 2009-10-09 21:10
>>
>> 2)  Make the behavior configurable with a channel variable and/or
>>     sofia config option: {sip_pass_headers=all|none|X-MyCustomHeaderByName}
>> -----
>>
>> br
>>
>>
>> --
>> Ing. Christian L?schenkohl
>> Technische Leitung, Forschung & Entwicklung VoIP
>>
>> xpirio
>> Telekommunikation & Service GmbH
>> Lakeside B04
>> 9020 Klagenfurt
>> Austria
>>
>> T  +43 (0) 5 77 11 - 1000
>> F  +43 (0) 5 77 11 - 1002
>> E  christian.loeschenkohl at xpirio.com
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
> 
> 
> 


-- 
Ing. Christian L?schenkohl
Technische Leitung, Forschung & Entwicklung VoIP

xpirio
Telekommunikation & Service GmbH
Lakeside B04
9020 Klagenfurt
Austria

T  +43 (0) 5 77 11 - 1000
F  +43 (0) 5 77 11 - 1002
E  christian.loeschenkohl at xpirio.com


From mrene_lists at avgs.ca  Wed Feb  3 16:28:21 2010
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Wed, 3 Feb 2010 19:28:21 -0500
Subject: [Freeswitch-users] adding sip header without X-
In-Reply-To: <4B6A1297.9090802@xpirio.com>
References: <4B69F9AD.8090904@xpirio.com>
	<2d9149cd1002031555o150b63a2hfd76dac20725b06a@mail.gmail.com>
	<4B6A1297.9090802@xpirio.com>
Message-ID: <CA18157A-7FBB-47CB-83CC-9B60D1E5E28B@avgs.ca>

If you look carefully, you'll see that it imports X- headers as  
variables on an incoming invite, but it'll send out sip_h_<anything>  
anyways.

For alert info, FS uses a channel variable called "alert_info", its  
implemented directly, without using sip_h_*.

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 3-Feb-10, at 7:19 PM, Christian L?schenkohl wrote:

> hello
>
> thank you for your answer
> setting only sip_h_Alert-Info was my first try, didn't work at all
>
> i use trun rev. 16456 in proxy mode - if it helps
> every header i add with X-... shows up in the invite to the sip  
> endpoint (snom phone)
>
> when i read sofia_glue.c i do see the function  
> sofia_glue_set_extra_headers
> which only adds header starting with X- or P- (line 1407 for me)
>
> br
>
> Kristian Kielhofner wrote:
>
>> Hello Christian,
>>
>>  Reading through the code (sofia_glue.c) it looks like all you have
>> to do is set the alert_info channel variable.  I also don't think you
>> have to include the X- when using set with sip_h.  It's just always
>> good (RFC compliant) to prefix "custom" headers with X-.
>>
>>  Regarding my suggestion for that configuration option/channel
>> variable.  That was only to be used on bridged channels to make it
>> configurable which headers FreeSWITCH passes from the a leg to the b
>> leg when using bridge.  A "proper" B2BUA (according to me) should
>> completely rewrite the outgoing INVITE, including stripping any  
>> custom
>> headers.
>>
>> 2010/2/3 Christian L?schenkohl <christian.loeschenkohl at xpirio.com>:
>>> hello
>>>
>>> do anybody know a way to add "Alert-Info: <http:// 
>>> www.notused.com>;info=alert-group;x-line-id=0"
>>> as a custom sip header to the invite message?
>>>
>>> <action application="set" data="sip_h_X-Alert-Info=<http://www.notused.com 
>>> >\;info=alert-internal\;x-line-id=0"/>
>>> works but "X-Alert-Info" as a header isn't usefull at all
>>>
>>> the purpose is to distinguish internal and external calls in a  
>>> pbx. it is also described here
>>> http://wiki.snom.com/Web_Interface/V7/Preferences/Alert-Info_Ringer
>>>
>>> there was also a posting on the list by Kristian Kielhofner who  
>>> suggested a very flexible solution
>>>
>>> -----
>>> mail from: 2009-10-09 21:10
>>>
>>> 2)  Make the behavior configurable with a channel variable and/or
>>>    sofia config option: {sip_pass_headers=all|none|X- 
>>> MyCustomHeaderByName}
>>> -----
>>>
>>> br
>>>
>>>
>>> --
>>> Ing. Christian L?schenkohl
>>> Technische Leitung, Forschung & Entwicklung VoIP
>>>
>>> xpirio
>>> Telekommunikation & Service GmbH
>>> Lakeside B04
>>> 9020 Klagenfurt
>>> Austria
>>>
>>> T  +43 (0) 5 77 11 - 1000
>>> F  +43 (0) 5 77 11 - 1002
>>> E  christian.loeschenkohl at xpirio.com
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>
>
> -- 
> Ing. Christian L?schenkohl
> Technische Leitung, Forschung & Entwicklung VoIP
>
> xpirio
> Telekommunikation & Service GmbH
> Lakeside B04
> 9020 Klagenfurt
> Austria
>
> T  +43 (0) 5 77 11 - 1000
> F  +43 (0) 5 77 11 - 1002
> E  christian.loeschenkohl at xpirio.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From kristian.kielhofner at gmail.com  Wed Feb  3 16:29:55 2010
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Wed, 3 Feb 2010 19:29:55 -0500
Subject: [Freeswitch-users] adding sip header without X-
In-Reply-To: <4B6A1297.9090802@xpirio.com>
References: <4B69F9AD.8090904@xpirio.com>
	<2d9149cd1002031555o150b63a2hfd76dac20725b06a@mail.gmail.com>
	<4B6A1297.9090802@xpirio.com>
Message-ID: <2d9149cd1002031629k6353b8b9v1874da556fb1c245@mail.gmail.com>

Have you tried setting channel variable alert_info?

2010/2/3 Christian L?schenkohl <christian.loeschenkohl at xpirio.com>:
> hello
>
> thank you for your answer
> setting only sip_h_Alert-Info was my first try, didn't work at all
>
> i use trun rev. 16456 in proxy mode - if it helps
> every header i add with X-... shows up in the invite to the sip endpoint (snom phone)
>
> when i read sofia_glue.c i do see the function sofia_glue_set_extra_headers
> which only adds header starting with X- or P- (line 1407 for me)
>
> br
>

-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com


From msc at freeswitch.org  Wed Feb  3 16:54:54 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 3 Feb 2010 16:54:54 -0800
Subject: [Freeswitch-users] PAP2T issue
In-Reply-To: <4B6A0559.2040200@acsol.net>
References: <4B69F90D.1070503@acsol.net> <4B6A0559.2040200@acsol.net>
Message-ID: <87f2f3b91002031654o1838868bk49d75df6866bf351@mail.gmail.com>

Would you mind creating a small wiki page and linking to it here?
http://wiki.freeswitch.org/wiki/Interop_List#Linksys_PAP2T

It would be nice to have a known working config on the wiki. If you have any
issues let me know.

Thanks,
MC

On Wed, Feb 3, 2010 at 3:23 PM, John <john at acsol.net> wrote:

> Found two issues and solved the problem. By default, the PAP2T has
> provisioning turned on, and won't try to register if the provisioning
> server isn't found. Turned that off. Secondly, turned off Stun and
> enabled rport and it registered fine. Hope this helps someone else.
>
> thanks John
> On 2/3/2010 3:30 PM, John wrote:
> > I can register SNOM phones fine, one the same network, I am trying to
> > get a Linksys PAP2T-NA registered. Anyone have a configuration example?
> > Thanks
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100203/7e5e6f5c/attachment.html 

From christian.loeschenkohl at xpirio.com  Wed Feb  3 16:59:59 2010
From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=)
Date: Thu, 04 Feb 2010 01:59:59 +0100
Subject: [Freeswitch-users] adding sip header without X-
In-Reply-To: <2d9149cd1002031629k6353b8b9v1874da556fb1c245@mail.gmail.com>
References: <4B69F9AD.8090904@xpirio.com>	<2d9149cd1002031555o150b63a2hfd76dac20725b06a@mail.gmail.com>	<4B6A1297.9090802@xpirio.com>
	<2d9149cd1002031629k6353b8b9v1874da556fb1c245@mail.gmail.com>
Message-ID: <4B6A1C0F.9040806@xpirio.com>

now i did and i works

<action application="export" data="alert_info=http://www.notused.com;info=alert-internal;x-line-id=0"/>
did the job

thank you two
br

Kristian Kielhofner wrote:

> Have you tried setting channel variable alert_info?
> 
> 2010/2/3 Christian L?schenkohl <christian.loeschenkohl at xpirio.com>:
>> hello
>>
>> thank you for your answer
>> setting only sip_h_Alert-Info was my first try, didn't work at all
>>
>> i use trun rev. 16456 in proxy mode - if it helps
>> every header i add with X-... shows up in the invite to the sip endpoint (snom phone)
>>
>> when i read sofia_glue.c i do see the function sofia_glue_set_extra_headers
>> which only adds header starting with X- or P- (line 1407 for me)
>>
>> br
>>
> 


-- 
Ing. Christian L?schenkohl
Technische Leitung, Forschung & Entwicklung VoIP

xpirio
Telekommunikation & Service GmbH
Lakeside B04
9020 Klagenfurt
Austria

T  +43 (0) 5 77 11 - 1000
F  +43 (0) 5 77 11 - 1002
E  christian.loeschenkohl at xpirio.com


From scott.torr.fs at letterboxes.org  Wed Feb  3 23:16:10 2010
From: scott.torr.fs at letterboxes.org (Scott Torr)
Date: Thu, 04 Feb 2010 18:16:10 +1100
Subject: [Freeswitch-users] skypiax dtmf detection issue on Freeswitch
In-Reply-To: <A90E1E992A65E1428D12320969B90F2103042A99@USDALSMBS05.ad3.ad.alcatel.com>
References: <A90E1E992A65E1428D12320969B90F2103042A99@USDALSMBS05.ad3.ad.alcatel.com>
Message-ID: <1265267770.21239.1358206487@webmail.messagingengine.com>

Hi Majdi,

It is good to know that once the audio stream is in the right 'format',
Freeswitch is able to decode the DTMF's to digits. (as best it can given
the potential jitter and lossy quality of the signal)

I'm sure someone on the list has the answer and it will quite simple.

So here we go:


How do we convert the audio stream from mod_skypiax into to a suitable
format that <action application="start_dtmf" /> can attempt to decode
the 'Inband' DTMF tones into digits?

At present it seems that the decoder will not work on a 16Kbps stream.


Perhaps this has been explained and not understood?

Ideally Skype would decode the DTMF tones ingress to the Skype network
at the PSTN-Skype gateway and pass on as out of band signaling. Either
through bad design or intentionally they do not. It seems though at some
locations (New Zealand) they did, then did not?

Perhaps on their paid for (approx $9 AUD per month) SIP trunks they do?
Or is this just as hit and miss as well?
Does any body have any experience with those?


-Scott

On Wed, 03 Feb 2010 20:42 -0600, "BSOUL Majdi"
<Majdi.Bsoul at alcatel-lucent.com> wrote:
> I just saw your JIRA, so definitely by your comment date so close you
> might still blocked on that resolution.
> 
>  
> 
> I used SiptoSis with a single channel and because it passes the stream
> to Freeswitch as PCMU, it allows FS to detect correctly. And this proves
> that the inband DTMF quality coming from Skype is good enough.
> 
>  
> 
> I am trying to see if I can use stsTrunkBuilder to support multiple
> interfaces, but looks that it requires different skype account for each
> instance, rather than allowing me to use single instance for all
> instances as skypiax does.
> 
>  
> 
> Regards,
> 
>  
> 
> Majdi Bsoul
> 
> Mobile NGN R&D
> 
> Alcatel-Lucent
> 
> 3400 W. Plano Parkway
> 
> M/S 601-NGNRD
> 
> Plano, TX 75075  
> 
> Office: +1 972 477 0065
> 
> Fax: +1 972 519 3600
> Email: majdi.bsoul at alcatel-lucent.com
> 
> From: BSOUL Majdi 
> Sent: Wednesday, February 03, 2010 8:26 PM
> To: 'scott.torr.fs at letterboxes.org'
> Cc: 'mbsoul at hotmail.com'
> Subject: skypiax dtmf detection issue on Freeswitch
> 
>  
> 
> Hello Scott,
> 
>  
> 
> I thought I catch up with you and see if you had found a solution for
> the skypiax dtmf detection in FS issue that you reported in Dec?
> 
> I am running with the same issue.
> 
>  
> 
> Thanks,
> 
>  
> 
> Majdi Bsoul
> 
> Mobile NGN R&D
> 
> Alcatel-Lucent
> 
> 3400 W. Plano Parkway
> 
> M/S 601-NGNRD
> 
> Plano, TX 75075  
> 
> Office: +1 972 477 0065
> 
> Fax: +1 972 519 3600
> Email: majdi.bsoul at alcatel-lucent.com
> 


From nagalenoj at gmail.com  Thu Feb  4 01:32:58 2010
From: nagalenoj at gmail.com (Nagalenoj H.)
Date: Thu, 4 Feb 2010 15:02:58 +0530
Subject: [Freeswitch-users] A leg disconnects after uuid_bridge?
In-Reply-To: <87f2f3b91002030605j7a1f9a02g665ba043c6ed7f7b@mail.gmail.com>
References: <fe67a3341002021934m548cde89p4d48381c5abe02a0@mail.gmail.com>
	<191c3a031002022108ta22287ev40fc36420abba2e5@mail.gmail.com>
	<fe67a3341002030112yeea379u90c22c482614c57@mail.gmail.com>
	<87f2f3b91002030605j7a1f9a02g665ba043c6ed7f7b@mail.gmail.com>
Message-ID: <fe67a3341002040132n3aa2a897m60c5b9f1ca6b347f@mail.gmail.com>

On Wed, Feb 3, 2010 at 7:35 PM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Wed, Feb 3, 2010 at 1:12 AM, Nagalenoj H. <nagalenoj at gmail.com> wrote:
>
>> I've used intercept application instead of uuid_bridge. I got the call
>> bridged with the given uuid. But, similar to uuid_bridge, A leg is getting
>> disconnected when B leg terminates.
>> I've did the same steps posted above, but used intercept instead of
>> uuid_bridge. Am I right?!
>>
>> Tried using the hangup_after_bridge=false, but didn't see any difference.
>>
>> Are you on the latest SVN of FreeSWITCH? Be sure to "make current" and try
> again. If the issue remains then capture a complete debug log of the call
> from start to finish and also pastebin your script so others can test and
> analyze.
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
I've updated to latest SVN. Now, the version is 'FreeSWITCH Version
1.0.trunk (16565)'.
I tested intercept in this version and the problem persists here.

Freeswitch debug log:
     http://pastebin.freeswitch.org/12043

Script:
     Not done any scripts, tested through nc command. Used the following
commands,

sendmsg
call-command: execute
execute-app-name: answer

api originate
{origination_uuid=c6d2e0e2-0b2f-11df-9e84-fb15c3cd8565}user/1001 &park

sendmsg
call-command: execute
execute-app-name:intercept
execute-app-arg: c6d2e0e2-0b2f-11df-9e84-fb15c3cd8565

Steps:
* Make a call to the socket extension.
* Answer the call.
* Originate a call and park it.
* Intercept the originated call's uuid.
* Disconnect the 'B' leg.
* You will notice the 'A' leg is also getting disconnected.
-- 
Regards,
Nagalenoj H.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100204/b46049fb/attachment.html 

From nagalenoj at gmail.com  Thu Feb  4 01:47:44 2010
From: nagalenoj at gmail.com (Nagalenoj H.)
Date: Thu, 4 Feb 2010 15:17:44 +0530
Subject: [Freeswitch-users] uuid_bridge isn't working
Message-ID: <fe67a3341002040147i5892d5b8md65e327be6dea7c5@mail.gmail.com>

Dear friends,
    After upgrading to 'FreeSWITCH Version 1.0.trunk (16565)', uuid_bridge
isn't working. When I give uuid_bridge, both the legs are not bridged, and
they got disconnected.

Did the following,
* Made a call to socket extension.
* Answered the call.
* Originated a call and parked it.
* Did uuid_bridge with the uuids of the originated call and caller's uuid.

* Didn't get the legs bridged, instead both got disconnected,

Freeswitch debug log:
      http://pastebin.freeswitch.org/12044

Facing this problem only after upgrading to this trunk version.

-- 
Regards,
Nagalenoj H.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100204/cc2ad2f9/attachment.html 

From nagalenoj at gmail.com  Thu Feb  4 02:01:57 2010
From: nagalenoj at gmail.com (Nagalenoj H.)
Date: Thu, 4 Feb 2010 15:31:57 +0530
Subject: [Freeswitch-users] Event socket: filter delete isn't working
In-Reply-To: <191c3a031001271702l3e77c952lc8aba35bacbf3d58@mail.gmail.com>
References: <fe67a3341001270514g2715de2ag6aa2e3b099fa7816@mail.gmail.com>
	<191c3a031001271702l3e77c952lc8aba35bacbf3d58@mail.gmail.com>
Message-ID: <fe67a3341002040201i1b02a8bekdad0e649fdb323bf@mail.gmail.com>

I've now upgraded to 16565 trunk.  I've tested it and now it is working
fine.

But, when I give only 'filter delete' without the next parameters,
freeswitch is getting core dumped.

Freeswitch debug log:
   http://pastebin.freeswitch.org/12045

On Thu, Jan 28, 2010 at 6:32 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> in the future please report issues to jira http://jira.freeswitch.org
>
> please try svn trunk 16527 or higher
>
> This was not a bug but I made it work the way you describe since it made
> sense.
>
> you should have done
>
> filter delete unique-id
>
> which would have delete all the unique-id filters that was the only option
>
>
> you should be able to now say
>
> filter delete unique-id <value>
>
> To delete entry with specific value
>
> or
>
> filter delete unique-id
>
> to delete all entries with matching key
>
>
>
>
>
> On Wed, Jan 27, 2010 at 7:14 AM, Nagalenoj H. <nagalenoj at gmail.com> wrote:
>
>> Dear friends,
>>    I've tried to delete the filter which I applied for an unique id. But,
>> it doesn't work. After executing 'filter delete', I am receiving the events
>> from that uuid.
>>    I used the command as 'filter delete unique-id
>> c6d2e0e2-0b2f-11df-9e84-fb15c3cd8565'.
>>
>> I did the following operations.
>>     Made call to the event socket.
>>     Registered events for all. (events plain all).
>>     Applied filter for the uuid. (filter unique-id
>> aa3cb8ea-0b2f-11df-9e84-fb15c3cd8565).
>>     I've got a new uuid by using create_uuid.
>>     Applied filter for this new uuid. (filter unique-id
>> c6d2e0e2-0b2f-11df-9e84-fb15c3cd8565)
>>     Originated a call with that uuid.
>>     Now, I could receive events from both uuids. (Tested by giving DTMFs
>> in both end and checked unique-id in event header).
>>     Then, I wanted to delete a uuid from the filter. (filter delete
>> unique-id c6d2e0e2-0b2f-11df-9e84-fb15c3cd8565).
>>     I thought, i won't receive the events from this deleted unique-id.
>> But, I received the dtmfs from both unique-id.
>>
>> I'm using 'FreeSWITCH Version 1.0.trunk (15982)'.
>> --
>> Regards,
>> Nagalenoj H.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Regards,
Nagalenoj H.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100204/840d3f5d/attachment.html 

From xanlich at gmail.com  Thu Feb  4 03:33:54 2010
From: xanlich at gmail.com (Chia-Yen Wu)
Date: Thu, 4 Feb 2010 19:33:54 +0800
Subject: [Freeswitch-users] about bgapi
Message-ID: <314dc3f81002040333n28c15d92l5dd0aeef4dc5aa92@mail.gmail.com>

I tried to run a javascript by background API (bgapi jsrun test.js)

this javascript (test.js) wont automatically stop, I tried to kill it by
"uuid_kill" command with Job-UUID

but return "-ERR no such channel!", how can I kill this bgapi? btw my FS run
in windows.

thank you
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100204/3b5cb1b6/attachment.html 

From nepaligas at yahoo.com  Tue Feb  2 23:27:18 2010
From: nepaligas at yahoo.com (Prabin Shrestha)
Date: Tue, 2 Feb 2010 23:27:18 -0800 (PST)
Subject: [Freeswitch-users] need some hints on Softswitch deployment of
	FreeSwitch
Message-ID: <858472.38298.qm@web65402.mail.ac4.yahoo.com>

Dear all,

I had been browsing through all the wikis of freeswitch, googling more than 1 week and couldn't figure out where to start. 
I have been finding so many problems and IRC thing I don't understand. Basically, I am just a average linux user running Ubuntu, trying to build a softswitch. It there was some book on freeswitch it would have been much easier for newbie like me. 
Here are some problems I have been facing. 
After installation, I found freeswitch in /opt/freeswitch directory.
only creating freeswitch user, I can access it's fs_cli, and I have yet to learn the power of it. 
doing ps -A, I found freeswitch is running in background.

Now comes the hard part.
I wanted to test it using SPA3000 device with fxo and fxs ports, which after following guides in net,  is not working for me.

My reqirement is, 
1. to run freeswitch as a softswitch which can route calls from voip call providers to Quintum gateways.
2. to have complete CDR reports generated to sql database.

Some light on this matter will be highly appreciated.

Prabin.



      Your Mail works best with the New Yahoo Optimized IE8. Get it NOW! http://downloads.yahoo.com/in/internetexplorer/
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100202/fcbda4d8/attachment-0001.html 
-------------- next part --------------
A non-text attachment was scrubbed...
Name: not available
Type: image/gif
Size: 21362 bytes
Desc: not available
Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100202/fcbda4d8/attachment-0001.gif 

From nepaligas at yahoo.com  Wed Feb  3 22:00:33 2010
From: nepaligas at yahoo.com (Prabin Shrestha)
Date: Wed, 3 Feb 2010 22:00:33 -0800 (PST)
Subject: [Freeswitch-users] Fw: need some hints on Softswitch deployment of
	FreeSwitch
Message-ID: <609766.80114.qm@web65415.mail.ac4.yahoo.com>



--- On Tue, 2/2/10, Prabin Shrestha <nepaligas at yahoo.com> wrote:

From: Prabin Shrestha <nepaligas at yahoo.com>
Subject: need some hints on Softswitch deployment of FreeSwitch
To: freeswitch-users at lists.freeswitch.org
Date: Tuesday, 2 February, 2010, 11:27 PM

 
 
Dear all,

I had been browsing through all the wikis of freeswitch, googling more than 1 week and couldn't
 figure out where to start. 
I have been finding so many problems and IRC thing I don't understand. Basically, I am just a average linux user running Ubuntu, trying to build a softswitch. It there was some book on freeswitch it would have been much easier for newbie like me. 
Here are some problems I have been facing. 
After installation, I found freeswitch in /opt/freeswitch directory.
only creating freeswitch user, I can access it's fs_cli, and I have yet to learn the power of it. 
doing ps -A, I found freeswitch is running in background.

Now comes the hard part.
I wanted to test it using SPA3000 device with fxo and fxs ports, which after following guides in net,? is not working for me.

My reqirement is, 
1. to run freeswitch as a softswitch which can route calls from voip call providers to Quintum gateways.
2. to have complete CDR reports generated to sql database.

Some light on this matter will be
 highly appreciated.

Prabin.


 
 
 


       
Your Mail works best with the New Yahoo Optimized IE8. Get it NOW!.


      The INTERNET now has a personality. YOURS! See your Yahoo! Homepage. http://in.yahoo.com/
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100203/41a45589/attachment-0001.html 
-------------- next part --------------
A non-text attachment was scrubbed...
Name: not available
Type: image/gif
Size: 21362 bytes
Desc: not available
Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100203/41a45589/attachment-0001.gif 

From brian at freeswitch.org  Thu Feb  4 04:12:25 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 4 Feb 2010 06:12:25 -0600
Subject: [Freeswitch-users] A leg disconnects after uuid_bridge?
In-Reply-To: <fe67a3341002040132n3aa2a897m60c5b9f1ca6b347f@mail.gmail.com>
References: <fe67a3341002021934m548cde89p4d48381c5abe02a0@mail.gmail.com>
	<191c3a031002022108ta22287ev40fc36420abba2e5@mail.gmail.com>
	<fe67a3341002030112yeea379u90c22c482614c57@mail.gmail.com>
	<87f2f3b91002030605j7a1f9a02g665ba043c6ed7f7b@mail.gmail.com>
	<fe67a3341002040132n3aa2a897m60c5b9f1ca6b347f@mail.gmail.com>
Message-ID: <0A571E14-BB58-4A3D-996A-9D5205668E85@freeswitch.org>

Where are you getting this UUID?

/b

On Feb 4, 2010, at 3:32 AM, Nagalenoj H. wrote:

> api originate {origination_uuid=c6d2e0e2-0b2f-11df-9e84-fb15c3cd8565}user/1001 &park



From ustcorporation at yahoo.com  Wed Feb  3 18:12:09 2010
From: ustcorporation at yahoo.com (Darren C.)
Date: Wed, 3 Feb 2010 18:12:09 -0800 (PST)
Subject: [Freeswitch-users] Seeking Advice on SIP Phones like Aastra 6739i
	or Snom 870 that have good interoperability with FreeSWITCH
Message-ID: <894614.36103.qm@web33003.mail.mud.yahoo.com>

Hello,

I'm working on a project and are interested in using one of the newer SIP Phones with color displays, perhaps touchscreen, etc.?to implement PBX-like features like voice mail messages waiting, visual?address book,?conference call setup, and more.? 
?
We want to send some information about a call to the SIP Phone either via FS or our own Web Service.? These two phones have XML browsers that we may be able to utilize.? I'm concerned that these phones may work OK with Asterisk but not sure about FreeSWITCH.? There is a great guide for?XML Services for?Aastra 6739i but only mention Asterisk in the examples:
?
http://www.aastratelecom.com/cps/rde/xbcr/SID-3D8CCB6A-93A834D7/03/XML_Free_Services_-_PA-001005-00-05.pdf
?
I prefer to use phones that work well with FS but these are too new to show up on interoperability page.? Anyone have experience using these, any advise would be appreciated:


Aastra 6739i
http://www.aastratelecom.com/cps/rde/xbcr/SID-3D8CCB6A-EAA51F83/03/6739i_pds_en_1209.pdf
http://www.voipsupply.com/aastra-6739i?
?
SNOM 870
http://www.snom.com/uploads/docu/data_snom870_en.pdf
http://www.snom.com/products/ip-phones/snom-870-touchscreen-voip-phone
http://www.voipsupply.com/snom-870

Hoping someone has some positive or negative experiences working with these phones.? 
?
Thanks,

teldev



      
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100203/72c60834/attachment.html 

From nagalenoj at gmail.com  Thu Feb  4 04:52:35 2010
From: nagalenoj at gmail.com (Nagalenoj H.)
Date: Thu, 4 Feb 2010 18:22:35 +0530
Subject: [Freeswitch-users] A leg disconnects after uuid_bridge?
In-Reply-To: <0A571E14-BB58-4A3D-996A-9D5205668E85@freeswitch.org>
References: <fe67a3341002021934m548cde89p4d48381c5abe02a0@mail.gmail.com>
	<191c3a031002022108ta22287ev40fc36420abba2e5@mail.gmail.com>
	<fe67a3341002030112yeea379u90c22c482614c57@mail.gmail.com>
	<87f2f3b91002030605j7a1f9a02g665ba043c6ed7f7b@mail.gmail.com>
	<fe67a3341002040132n3aa2a897m60c5b9f1ca6b347f@mail.gmail.com>
	<0A571E14-BB58-4A3D-996A-9D5205668E85@freeswitch.org>
Message-ID: <fe67a3341002040452v5468fd1fh96d1c5062091474b@mail.gmail.com>

By using create_uuid. I've also tried without giving origination_uuid. But,
the result is same.

-- 
Regards,
Nagalenoj H.

On Thu, Feb 4, 2010 at 5:42 PM, Brian West <brian at freeswitch.org> wrote:

> Where are you getting this UUID?
>
> /b
>
> On Feb 4, 2010, at 3:32 AM, Nagalenoj H. wrote:
>
> > api originate
> {origination_uuid=c6d2e0e2-0b2f-11df-9e84-fb15c3cd8565}user/1001 &park
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100204/70acd9b7/attachment.html 

From christian.loeschenkohl at xpirio.com  Thu Feb  4 04:59:06 2010
From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=)
Date: Thu, 04 Feb 2010 13:59:06 +0100
Subject: [Freeswitch-users] presence with event socket
Message-ID: <4B6AC49A.8030409@xpirio.com>

hello

i have a little problem with setting the presence information manually (on and off)

with an outbound socket script i do send a event like this

sendevent PRESENCE_IN
proto: sip
from: xxxx at mydomain.com
login: xxxx at mydomain.com
event_type: presence
alt_event_type: dialog
Presence-Call-Direction: outbound
answer-state: confirmed

i monitor the estension on a snom phone with bfl (function key mode)

the problem is now that the led on the monitoring snom gets lit very short an the is off again
if i put a sleep in the outbound socket script the led lights up as long as i set the timeout.

the expected behaviour is that the led lights up and can be switched off with another
event (then answer-state: terminated)

we do use trunk rev. 16456 in proxy mode
br

-- 
Ing. Christian L?schenkohl
Technische Leitung, Forschung & Entwicklung VoIP

xpirio
Telekommunikation & Service GmbH
Lakeside B04
9020 Klagenfurt
Austria

T  +43 (0) 5 77 11 - 1000
F  +43 (0) 5 77 11 - 1002
E  christian.loeschenkohl at xpirio.com


From rupa at rupa.com  Thu Feb  4 05:19:21 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Thu, 4 Feb 2010 07:19:21 -0600
Subject: [Freeswitch-users] about bgapi
In-Reply-To: <314dc3f81002040333n28c15d92l5dd0aeef4dc5aa92@mail.gmail.com>
References: <314dc3f81002040333n28c15d92l5dd0aeef4dc5aa92@mail.gmail.com>
Message-ID: <b39273531002040519g71fa8a37uc9f0819c0a19772f@mail.gmail.com>

you can't.  If you want to terminate it, then you should set a global var
that the script periodically checks and if set the script should terminate
itself.

On Thu, Feb 4, 2010 at 5:33 AM, Chia-Yen Wu <xanlich at gmail.com> wrote:

> I tried to run a javascript by background API (bgapi jsrun test.js)
>
> this javascript (test.js) wont automatically stop, I tried to kill it by
> "uuid_kill" command with Job-UUID
>
> but return "-ERR no such channel!", how can I kill this bgapi? btw my FS
> run in windows.
>
> thank you
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100204/9c988b86/attachment.html 

From peder at networkoblivion.com  Thu Feb  4 05:57:20 2010
From: peder at networkoblivion.com (Peder)
Date: Thu, 4 Feb 2010 07:57:20 -0600
Subject: [Freeswitch-users] need some hints on Softswitch deployment
	of	FreeSwitch
In-Reply-To: <858472.38298.qm@web65402.mail.ac4.yahoo.com>
References: <858472.38298.qm@web65402.mail.ac4.yahoo.com>
Message-ID: <039401caa5a1$f8256b40$e87041c0$@com>

You will need to ask more specific questions, such as ?when I call from 1000 to 1001, I get xxx on the console and it doesn?t work?.  Just saying ?I need help? is too big of a topic for anyone to respond.  Have you tried the wiki?

 

http://wiki.freeswitch.org/wiki/Main_Page

 

 

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Prabin Shrestha
Sent: Wednesday, February 03, 2010 1:27 AM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] need some hints on Softswitch deployment of FreeSwitch

 


	
	


Dear all,

I had been browsing through all the wikis of freeswitch, googling more than 1 week and couldn't figure out where to start. 
I have been finding so many problems and IRC thing I don't understand. Basically, I am just a average linux user running Ubuntu, trying to build a softswitch. It there was some book on freeswitch it would have been much easier for newbie like me. 
Here are some problems I have been facing. 
After installation, I found freeswitch in /opt/freeswitch directory.
only creating freeswitch user, I can access it's fs_cli, and I have yet to learn the power of it. 
doing ps -A, I found freeswitch is running in background.

Now comes the hard part.
I wanted to test it using SPA3000 device with fxo and fxs ports, which after following guides in net,  is not working for me.

My reqirement is, 
1. to run freeswitch as a softswitch which can route calls from voip call providers to Quintum gateways.
2. to have complete CDR reports generated to sql database.

Some light on this matter will be highly appreciated.

Prabin.

	
	
	





  _____  

Your Mail works best with the New Yahoo Optimized IE8. Get it NOW! <http://in.rd.yahoo.com/tagline_ie8_new/*http:/downloads.yahoo.com/in/internetexplorer/> .

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100204/2d53841c/attachment-0001.html 

From maciej.aniserowicz at gmail.com  Thu Feb  4 06:53:10 2010
From: maciej.aniserowicz at gmail.com (Maciej Aniserowicz)
Date: Thu, 4 Feb 2010 06:53:10 -0800 (PST)
Subject: [Freeswitch-users] Trunk compilation error on Windows "Cannot open
 source file: '..\..\celt-0.7.0-1\libcelt\kfft_single.c'"
Message-ID: <1265295190863-4513865.post@n2.nabble.com>


Hi,
I downloaded the latest bits from svn but have troubles compiling FS. I get
21 similar errors:
Cannot open source file: '..\..\celt-0.7.0-1\libcelt\kfft_single.c'

Error 40 fatal error C1083: Cannot open source file:
'..\..\celt-0.7.0-1\libcelt\bands.c': No such file or directory
File: c1
Project: libcelt

I just opened Freeswitch.2008.sln in VS, changed configuration to Release
and hit Build. What am I missing?

Regards,
Maciej Aniserowicz
-- 
View this message in context: http://n2.nabble.com/Trunk-compilation-error-on-Windows-Cannot-open-source-file-celt-0-7-0-1-libcelt-kfft-single-c-tp4513865p4513865.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From brian at freeswitch.org  Thu Feb  4 07:02:31 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 4 Feb 2010 09:02:31 -0600
Subject: [Freeswitch-users] Trunk compilation error on Windows "Cannot
	open source file: '..\..\celt-0.7.0-1\libcelt\kfft_single.c'"
In-Reply-To: <1265295190863-4513865.post@n2.nabble.com>
References: <1265295190863-4513865.post@n2.nabble.com>
Message-ID: <FC7358A4-5486-4E61-BABE-8FDD5D8D52BF@freeswitch.org>

Please update.  That was already fixed... remember ALWAYS update.. check and then email... we move fast around these parts!  :P

/b

On Feb 4, 2010, at 8:53 AM, Maciej Aniserowicz wrote:

> 
> Hi,
> I downloaded the latest bits from svn but have troubles compiling FS. I get
> 21 similar errors:
> Cannot open source file: '..\..\celt-0.7.0-1\libcelt\kfft_single.c'
> 
> Error 40 fatal error C1083: Cannot open source file:
> '..\..\celt-0.7.0-1\libcelt\bands.c': No such file or directory
> File: c1
> Project: libcelt
> 
> I just opened Freeswitch.2008.sln in VS, changed configuration to Release
> and hit Build. What am I missing?
> 
> Regards,
> Maciej Aniserowicz
> -- 
> View this message in context: http://n2.nabble.com/Trunk-compilation-error-on-Windows-Cannot-open-source-file-celt-0-7-0-1-libcelt-kfft-single-c-tp4513865p4513865.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From tim at communicatefreely.net  Thu Feb  4 07:05:37 2010
From: tim at communicatefreely.net (Tim St. Pierre)
Date: Thu, 04 Feb 2010 10:05:37 -0500
Subject: [Freeswitch-users] Seeking Advice on SIP Phones like Aastra
 6739i or Snom 870 that have good interoperability with FreeSWITCH
In-Reply-To: <894614.36103.qm@web33003.mail.mud.yahoo.com>
References: <894614.36103.qm@web33003.mail.mud.yahoo.com>
Message-ID: <4B6AE241.4040106@communicatefreely.net>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hi Darren,

While I can't vouch for the 6739i yet, I have been doing all of my FS development with Aastra
phones.  I had a 57i connected up, and it worked just fine.  I could use the wideband codecs, BLFs
all worked correctly, I have intercom and distinctive ring working properly too.

As far as the XML features are concerned, these are all done outside of Freeswitch.  I have built
some simple applications that use the XML browser for or own network, and it was pretty straight
forward.  The phone interacts with our web server, that in turn manipulates the database used for
call handling and other parameters.  In some cases, the php script on the web server fires events to
the switch to initiate calls, or do some other call handling.

Right now, our production system is Asterisk, but I'm trying to migrate everything to freeswitch.
As long as everything is in a database, it's pretty easy to make some great features.  In many ways,
I have found freeswitch easier to integrate with.  You may want to enable core odbc, it opens up a
lot, since you can do a DB query to find out what voicemail messages a user has, or what calls are
active, etc.  I find it a lot easier to interact with than something like an event socket, and it
clusters better.

I ended up not using any of the Aastra supplied scripts, since they are very specific to a single
office Asterisk setup.  You will have to build your own for freeswitch, but they are easy to write.
 The Aastra php classes are still useful functions, just not their pre-built voicemail and other
Asterisk feature tools.

Good luck!

- -Tim

Darren C. wrote:
> Hello,
> 
> I'm working on a project and are interested in using one of the newer
> SIP Phones with color displays, perhaps touchscreen, etc. to implement
> PBX-like features like voice mail messages waiting, visual address
> book, conference call setup, and more. 
> 
>  
> 
> We want to send some information about a call to the SIP Phone either
> via FS or our own Web Service.  These two phones have XML browsers that
> we may be able to utilize.  I'm concerned that these phones may work OK
> with Asterisk but not sure about FreeSWITCH.  There is a great guide
> for XML Services for Aastra 6739i but only mention Asterisk in the examples:
> 
>  
> 
> http://www.aastratelecom.com/cps/rde/xbcr/SID-3D8CCB6A-93A834D7/03/XML_Free_Services_-_PA-001005-00-05.pdf
> 
>  
> 
> I prefer to use phones that work well with FS but these are too new to
> show up on interoperability page.  Anyone have experience using these,
> any advise would be appreciated:
> 
> *Aastra 6739i*
> 
> http://www.aastratelecom.com/cps/rde/xbcr/SID-3D8CCB6A-EAA51F83/03/6739i_pds_en_1209.pdf
> 
> http://www.voipsupply.com/aastra-6739i* *
> 
> * *
> 
> *SNOM 870*
> 
> http://www.snom.com/uploads/docu/data_snom870_en.pdf
> 
> http://www.snom.com/products/ip-phones/snom-870-touchscreen-voip-phone
> 
> http://www.voipsupply.com/snom-870
> 
> 
> Hoping someone has some positive or negative experiences working with
> these phones. 
> 
>  
> 
> Thanks,
> 
> teldev
> 
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

- --
Tim St. Pierre
IP Voice technician
Communicate Freely
1-877-291-8647 x5101
sip:5101 at communicatefreely.net
tim at communicatefreely.net
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v2.0.4 (FreeBSD)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org

iQCVAwUBS2riQYqVcvNCnHOrAQKQLgQAk9thDiDNp89QhI32cv4G3/RmPET2nFns
WHRovrKxo/gGLJV0eC7XAB9vysQqa/A7VGXQBjvrNZ/EylsRoKcGDHr95Tndz6jI
w82rqWUdiCl58ABae2jYW03sRUG47y9rwp6ujERO1XQ8iyUxLPdTju5pve0gmdyI
qO0PnOkWkhk=
=e6DD
-----END PGP SIGNATURE-----


From anthony.minessale at gmail.com  Thu Feb  4 07:46:25 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 4 Feb 2010 09:46:25 -0600
Subject: [Freeswitch-users] presence with event socket
In-Reply-To: <4B6AC49A.8030409@xpirio.com>
References: <4B6AC49A.8030409@xpirio.com>
Message-ID: <191c3a031002040746i494d5fe4s163f512afd34da79@mail.gmail.com>

the presence stuff is automatic so when the phone re-registers it probably
turns it back off.
snom does not do the nonce count thing right so it re-challenges every time
and causes a presence event.


2010/2/4 Christian L?schenkohl <christian.loeschenkohl at xpirio.com>

> hello
>
> i have a little problem with setting the presence information manually (on
> and off)
>
> with an outbound socket script i do send a event like this
>
> sendevent PRESENCE_IN
> proto: sip
> from: xxxx at mydomain.com
> login: xxxx at mydomain.com
> event_type: presence
> alt_event_type: dialog
> Presence-Call-Direction: outbound
> answer-state: confirmed
>
> i monitor the estension on a snom phone with bfl (function key mode)
>
> the problem is now that the led on the monitoring snom gets lit very short
> an the is off again
> if i put a sleep in the outbound socket script the led lights up as long as
> i set the timeout.
>
> the expected behaviour is that the led lights up and can be switched off
> with another
> event (then answer-state: terminated)
>
> we do use trunk rev. 16456 in proxy mode
> br
>
> --
> Ing. Christian L?schenkohl
> Technische Leitung, Forschung & Entwicklung VoIP
>
> xpirio
> Telekommunikation & Service GmbH
> Lakeside B04
> 9020 Klagenfurt
> Austria
>
> T  +43 (0) 5 77 11 - 1000
> F  +43 (0) 5 77 11 - 1002
> E  christian.loeschenkohl at xpirio.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100204/5956864d/attachment.html 

From anthony.minessale at gmail.com  Thu Feb  4 07:50:27 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 4 Feb 2010 09:50:27 -0600
Subject: [Freeswitch-users] Event socket: filter delete isn't working
In-Reply-To: <fe67a3341002040201i1b02a8bekdad0e649fdb323bf@mail.gmail.com>
References: <fe67a3341001270514g2715de2ag6aa2e3b099fa7816@mail.gmail.com>
	<191c3a031001271702l3e77c952lc8aba35bacbf3d58@mail.gmail.com>
	<fe67a3341002040201i1b02a8bekdad0e649fdb323bf@mail.gmail.com>
Message-ID: <191c3a031002040750x5c94f701p2f5dd7a6a8598cd4@mail.gmail.com>

not anymore, fixed r16568

On Thu, Feb 4, 2010 at 4:01 AM, Nagalenoj H. <nagalenoj at gmail.com> wrote:

> I've now upgraded to 16565 trunk.  I've tested it and now it is working
> fine.
>
> But, when I give only 'filter delete' without the next parameters,
> freeswitch is getting core dumped.
>
> Freeswitch debug log:
>    http://pastebin.freeswitch.org/12045
>
>
> On Thu, Jan 28, 2010 at 6:32 AM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> in the future please report issues to jira http://jira.freeswitch.org
>>
>> please try svn trunk 16527 or higher
>>
>> This was not a bug but I made it work the way you describe since it made
>> sense.
>>
>> you should have done
>>
>> filter delete unique-id
>>
>> which would have delete all the unique-id filters that was the only option
>>
>>
>> you should be able to now say
>>
>> filter delete unique-id <value>
>>
>> To delete entry with specific value
>>
>> or
>>
>> filter delete unique-id
>>
>> to delete all entries with matching key
>>
>>
>>
>>
>>
>> On Wed, Jan 27, 2010 at 7:14 AM, Nagalenoj H. <nagalenoj at gmail.com>wrote:
>>
>>> Dear friends,
>>>    I've tried to delete the filter which I applied for an unique id. But,
>>> it doesn't work. After executing 'filter delete', I am receiving the events
>>> from that uuid.
>>>    I used the command as 'filter delete unique-id
>>> c6d2e0e2-0b2f-11df-9e84-fb15c3cd8565'.
>>>
>>> I did the following operations.
>>>     Made call to the event socket.
>>>     Registered events for all. (events plain all).
>>>     Applied filter for the uuid. (filter unique-id
>>> aa3cb8ea-0b2f-11df-9e84-fb15c3cd8565).
>>>     I've got a new uuid by using create_uuid.
>>>     Applied filter for this new uuid. (filter unique-id
>>> c6d2e0e2-0b2f-11df-9e84-fb15c3cd8565)
>>>     Originated a call with that uuid.
>>>     Now, I could receive events from both uuids. (Tested by giving DTMFs
>>> in both end and checked unique-id in event header).
>>>     Then, I wanted to delete a uuid from the filter. (filter delete
>>> unique-id c6d2e0e2-0b2f-11df-9e84-fb15c3cd8565).
>>>     I thought, i won't receive the events from this deleted unique-id.
>>> But, I received the dtmfs from both unique-id.
>>>
>>> I'm using 'FreeSWITCH Version 1.0.trunk (15982)'.
>>> --
>>> Regards,
>>> Nagalenoj H.
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Regards,
> Nagalenoj H.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100204/5184adb5/attachment-0001.html 

From anthony.minessale at gmail.com  Thu Feb  4 07:54:22 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 4 Feb 2010 09:54:22 -0600
Subject: [Freeswitch-users] A leg disconnects after uuid_bridge?
In-Reply-To: <fe67a3341002040452v5468fd1fh96d1c5062091474b@mail.gmail.com>
References: <fe67a3341002021934m548cde89p4d48381c5abe02a0@mail.gmail.com>
	<191c3a031002022108ta22287ev40fc36420abba2e5@mail.gmail.com>
	<fe67a3341002030112yeea379u90c22c482614c57@mail.gmail.com>
	<87f2f3b91002030605j7a1f9a02g665ba043c6ed7f7b@mail.gmail.com>
	<fe67a3341002040132n3aa2a897m60c5b9f1ca6b347f@mail.gmail.com>
	<0A571E14-BB58-4A3D-996A-9D5205668E85@freeswitch.org>
	<fe67a3341002040452v5468fd1fh96d1c5062091474b@mail.gmail.com>
Message-ID: <191c3a031002040754r2987a03fo18ee2e1ceb239f6a@mail.gmail.com>

   1. 2010-02-04 14:30:09.574084 [DEBUG] switch_core_state_machine.c:354 (
   sofia/internal/1010 at 192.168.1.222) State SOFT_EXECUTE going to sleep
   2. 2010-02-04 14:30:09.574084 [DEBUG] switch_core_state_machine.c:314 (
   sofia/internal/1010 at 192.168.1.222) Running State Change CS_EXECUTE
   3. 2010-02-04 14:30:09.574084 [DEBUG] switch_core_state_machine.c:348 (
   sofia/internal/1010 at 192.168.1.222) State EXECUTE
   4. 2010-02-04 14:30:09.574084 [DEBUG] mod_sofia.c:181 sofia/internal/1010
   @192.168.1.222 SOFIA EXECUTE
   5. 2010-02-04 14:30:09.574084 [DEBUG]
switch_core_state_machine.c:159sofia/internal/
   1010 at 192.168.1.222 Standard EXECUTE
   6. 2010-02-04 14:30:09.574084 [NOTICE]
switch_core_state_machine.c:187Hangup sofia/internal/
   1010 at 192.168.1.222 [CS_EXECUTE] [NORMAL_CLEARING]



Your channel went back to EXECUTE as expected then it hungup because there
were no more instructions in your dial plan for it to execute.  So it is
working as expected.

Consider using transfer_after_bridge variable or park_after bridge to make
it stay around when the call is over.



On Thu, Feb 4, 2010 at 6:52 AM, Nagalenoj H. <nagalenoj at gmail.com> wrote:

> By using create_uuid. I've also tried without giving origination_uuid. But,
> the result is same.
>
> --
> Regards,
> Nagalenoj H.
>
>
> On Thu, Feb 4, 2010 at 5:42 PM, Brian West <brian at freeswitch.org> wrote:
>
>> Where are you getting this UUID?
>>
>> /b
>>
>> On Feb 4, 2010, at 3:32 AM, Nagalenoj H. wrote:
>>
>> > api originate
>> {origination_uuid=c6d2e0e2-0b2f-11df-9e84-fb15c3cd8565}user/1001 &park
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100204/7cc6eeff/attachment.html 

From rupa at rupa.com  Thu Feb  4 07:58:17 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Thu, 4 Feb 2010 09:58:17 -0600
Subject: [Freeswitch-users] max calls from a gateway
In-Reply-To: <867692.53487.qm@web34301.mail.mud.yahoo.com>
References: <867692.53487.qm@web34301.mail.mud.yahoo.com>
Message-ID: <b39273531002040758y6ef0491bo7d57accc83b1864b@mail.gmail.com>

If that works, let me know.  I just grepped the source for:

 ack-grep max[-_]calls

and while setting that will set profile->max_calls, I don't see anything
that actually acts on that.

My standard recommendation for setting limits is to use...  umm..  mod_limit
:)

On Wed, Feb 3, 2010 at 2:29 PM, FERNANDO VILLARROEL
<fvillarroel at yahoo.com>wrote:

> Dear.
>
> How i can do for limits inbound calls from a gateway:
>
>
> My config is like this:
>
> ~sip_profiles/external/
>
> gateway1.xml
>
> <include>
>    <gateway name="foo">
>      <param name="username" value="foo"/>
>      <param name="password" value="foo"/>
>      <param name="realm" value="xx.xxx.xxx.xx"/>
>      <param name="proxy" value="xx.xxx.xxx.xx/>
>      <param name="register" value="false"/>
>      <param name="context" value="public"/>
>      <param name="max_calls" value="30"/>-->
>    </gateway>
> </include>
>
> It?s fine?
>
> Fernando
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
-Rupa
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100204/59177e37/attachment.html 

From maciej.aniserowicz at gmail.com  Thu Feb  4 09:56:14 2010
From: maciej.aniserowicz at gmail.com (Maciej Aniserowicz)
Date: Thu, 4 Feb 2010 09:56:14 -0800 (PST)
Subject: [Freeswitch-users] Trunk compilation error on Windows "Cannot
 open source file: '..\..\celt-0.7.0-1\libcelt\kfft_single.c'"
In-Reply-To: <FC7358A4-5486-4E61-BABE-8FDD5D8D52BF@freeswitch.org>
References: <1265295190863-4513865.post@n2.nabble.com>
	<FC7358A4-5486-4E61-BABE-8FDD5D8D52BF@freeswitch.org>
Message-ID: <1265306174527-4515042.post@n2.nabble.com>


I updated, checked, took a break and then sent email -> the break should not
be there I guess :). Thanks, it works now.
-- 
View this message in context: http://n2.nabble.com/Trunk-compilation-error-on-Windows-Cannot-open-source-file-celt-0-7-0-1-libcelt-kfft-single-c-tp4513865p4515042.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From jerry.richards at teotech.com  Thu Feb  4 11:29:21 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Thu, 4 Feb 2010 11:29:21 -0800
Subject: [Freeswitch-users] Stripping Leading Digits of 10-digit Inbound
	Number
Message-ID: <59D52ECCA7234EB9ABC3EC66E241656A@greyhawk.tonecommander.com>

I am using a Sangoma PRI card.  When an inbound call is received, where do
the leading digits get stripped off?  For example, if the inbound called
number is 4257405381, I notice the call is routed to 5381 extension, but I
don't know what is stripping off the 425740 digits.

Best Regards,
Jerry



From msc at freeswitch.org  Thu Feb  4 11:45:05 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 4 Feb 2010 11:45:05 -0800
Subject: [Freeswitch-users] Fw: need some hints on Softswitch deployment
	of FreeSwitch
In-Reply-To: <609766.80114.qm@web65415.mail.ac4.yahoo.com>
References: <609766.80114.qm@web65415.mail.ac4.yahoo.com>
Message-ID: <87f2f3b91002041145y6b5e8a61m9ad86f330e9f952f@mail.gmail.com>

On Wed, Feb 3, 2010 at 10:00 PM, Prabin Shrestha <nepaligas at yahoo.com>wrote:

>
>
> --- On *Tue, 2/2/10, Prabin Shrestha <nepaligas at yahoo.com>* wrote:
>
>
> From: Prabin Shrestha <nepaligas at yahoo.com>
> Subject: need some hints on Softswitch deployment of FreeSwitch
> To: freeswitch-users at lists.freeswitch.org
> Date: Tuesday, 2 February, 2010, 11:27 PM
>
>
>
> Dear all,
>
> I had been browsing through all the wikis of freeswitch, googling more than
> 1 week and couldn't figure out where to start.
>
> Welcome to FreeSWITCH. Yes, it is overwhelming at first. Telephony and VoIP
are deep subjects. Give yourself a lot of time to learn - you've got a long
road ahead.

> I have been finding so many problems and IRC thing I don't understand.
> Basically, I am just a average linux user running Ubuntu, trying to build a
> softswitch. It there was some book on freeswitch it would have been much
> easier for newbie like me.
>
> Be careful what you wish for! ;) In the meantime you can check out this
article from Linux Pro magazine:
http://bit.ly/EpVrv

Also, are you installing from the source? We much prefer to see people on
the SVN trunk and installing from source. Using the Q&D install is very easy
and is very predictable. Check this out:
http://wiki.freeswitch.org/wiki/Quick_and_Dirty_Install

Here are some problems I have been facing.
> After installation, I found freeswitch in /opt/freeswitch directory.
> only creating freeswitch user, I can access it's fs_cli, and I have yet to
> learn the power of it.
> doing ps -A, I found freeswitch is running in background.
>
> Now comes the hard part.
> I wanted to test it using SPA3000 device with fxo and fxs ports, which
> after following guides in net,  is not working for me.
>
> Is the SPA3000 the old Sipura version of the newer Linksys/Cisco SPA3102? 1
FXO and 1 FXS port, right? The FXS ports are easy to configure - you just
need to have the Phone port register to FreeSWITCH as a user. The FXO port
is a bit more challenging because you have to handle inbound and outbound
calls differently. It would take too much time to write up how to do it here
in this email. I would look at the principles found here:
http://wiki.freeswitch.org/wiki/SPA400_FreeSwitch_HowTo


> My reqirement is,
> 1. to run freeswitch as a softswitch which can route calls from voip call
> providers to Quintum gateways.
>
> Should be doable. Anything crazy or unusual about the Quintum gateways?

> 2. to have complete CDR reports generated to sql database.
>
> Totally doable, just you need to do it after the call completes. FS always
writes CSV CDRs to disk. You can use the SQL CDR template to create easily
loadable CDR records that can drop right into a MySQL/PostgreSQL/etc. db
backed. (You can also use mod_xml_cdr but that's a much more difficult
proposition, so get to know FS first before going to the deep end of the
pool.)

>
> Some light on this matter will be highly appreciated.
>
> Relax and breath. This is a really long and drawn out process. It just
takes a lot of time and patience to learn it all. Sorry, there just aren't
any shortcuts.
-MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100204/acaa80b2/attachment-0001.html 

From msc at freeswitch.org  Thu Feb  4 11:48:30 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 4 Feb 2010 11:48:30 -0800
Subject: [Freeswitch-users] Stripping Leading Digits of 10-digit Inbound
	Number
In-Reply-To: <59D52ECCA7234EB9ABC3EC66E241656A@greyhawk.tonecommander.com>
References: <59D52ECCA7234EB9ABC3EC66E241656A@greyhawk.tonecommander.com>
Message-ID: <87f2f3b91002041148s33bc755p57d211282f3c8cd2@mail.gmail.com>

On Thu, Feb 4, 2010 at 11:29 AM, Jerry Richards
<jerry.richards at teotech.com>wrote:

> I am using a Sangoma PRI card.  When an inbound call is received, where do
> the leading digits get stripped off?  For example, if the inbound called
> number is 4257405381, I notice the call is routed to 5381 extension, but I
> don't know what is stripping off the 425740 digits.
>
> Are you receiving all 10 digits from the carrier? You might only be
receiving four digits. The log will show you. I think it's a green log line.
If you need help then capture the debug output and drop into pastebin.
-MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100204/06a86f9b/attachment.html 

From brian at freeswitch.org  Thu Feb  4 12:06:31 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 4 Feb 2010 14:06:31 -0600
Subject: [Freeswitch-users] Stripping Leading Digits of 10-digit Inbound
	Number
In-Reply-To: <87f2f3b91002041148s33bc755p57d211282f3c8cd2@mail.gmail.com>
References: <59D52ECCA7234EB9ABC3EC66E241656A@greyhawk.tonecommander.com>
	<87f2f3b91002041148s33bc755p57d211282f3c8cd2@mail.gmail.com>
Message-ID: <0A76C7F3-343A-4F2E-BF1A-65216E97CFD0@freeswitch.org>

Sounds like your provider just isn't delivering them to you.

/b

On Feb 4, 2010, at 1:48 PM, Michael Collins wrote:

> On Thu, Feb 4, 2010 at 11:29 AM, Jerry Richards <jerry.richards at teotech.com> wrote:
> I am using a Sangoma PRI card.  When an inbound call is received, where do
> the leading digits get stripped off?  For example, if the inbound called
> number is 4257405381, I notice the call is routed to 5381 extension, but I
> don't know what is stripping off the 425740 digits.
> 
> Are you receiving all 10 digits from the carrier? You might only be receiving four digits. The log will show you. I think it's a green log line. If you need help then capture the debug output and drop into pastebin.
> -MC

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100204/73d4c89d/attachment.html 

From jerry.richards at teotech.com  Thu Feb  4 15:56:05 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Thu, 4 Feb 2010 15:56:05 -0800
Subject: [Freeswitch-users] Difference Between "bridge" and "transfer"
Message-ID: <D123D09A29B942D988B3456528DD44AD@greyhawk.tonecommander.com>

What is the difference between "bridge" and "transfer"?  I'm looking at the
demo IVRs.

Thanks,
Jerry



From msc at freeswitch.org  Thu Feb  4 16:32:40 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 4 Feb 2010 16:32:40 -0800
Subject: [Freeswitch-users] Difference Between "bridge" and "transfer"
In-Reply-To: <D123D09A29B942D988B3456528DD44AD@greyhawk.tonecommander.com>
References: <D123D09A29B942D988B3456528DD44AD@greyhawk.tonecommander.com>
Message-ID: <87f2f3b91002041632t19f17aefoaf11e048150816c5@mail.gmail.com>

On Thu, Feb 4, 2010 at 3:56 PM, Jerry Richards
<jerry.richards at teotech.com>wrote:

> What is the difference between "bridge" and "transfer"?  I'm looking at the
> demo IVRs.
>
>
bridge will connect two endpoints together while transfer sends the endpoint
back through the dialplan again...

-MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100204/fa5fab9c/attachment.html 

From brian at freeswitch.org  Thu Feb  4 17:08:10 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 4 Feb 2010 19:08:10 -0600
Subject: [Freeswitch-users] Difference Between "bridge" and "transfer"
In-Reply-To: <87f2f3b91002041632t19f17aefoaf11e048150816c5@mail.gmail.com>
References: <D123D09A29B942D988B3456528DD44AD@greyhawk.tonecommander.com>
	<87f2f3b91002041632t19f17aefoaf11e048150816c5@mail.gmail.com>
Message-ID: <0F758845-5AE6-47C8-B1C8-F13C4CC1C756@freeswitch.org>

which then can result in a bridge being called again.

/b

On Feb 4, 2010, at 6:32 PM, Michael Collins wrote:

> bridge will connect two endpoints together while transfer sends the endpoint back through the dialplan again...



From andrew at hijacked.us  Thu Feb  4 17:28:24 2010
From: andrew at hijacked.us (Andrew Thompson)
Date: Thu, 4 Feb 2010 20:28:24 -0500
Subject: [Freeswitch-users] Difference Between "bridge" and "transfer"
In-Reply-To: <0F758845-5AE6-47C8-B1C8-F13C4CC1C756@freeswitch.org>
References: <D123D09A29B942D988B3456528DD44AD@greyhawk.tonecommander.com>
	<87f2f3b91002041632t19f17aefoaf11e048150816c5@mail.gmail.com>
	<0F758845-5AE6-47C8-B1C8-F13C4CC1C756@freeswitch.org>
Message-ID: <20100205012824.GD21394@hijacked.us>

On Thu, Feb 04, 2010 at 07:08:10PM -0600, Brian West wrote:
> which then can result in a bridge being called again.
> 
> /b
> 
> On Feb 4, 2010, at 6:32 PM, Michael Collins wrote:
> 
> > bridge will connect two endpoints together while transfer sends the endpoint back through the dialplan again...
> 
> 
Don't forget that the dialplan can be an endpoint too (via mod_loopback)
:P.

Andrew


From lists at redbonez.net  Thu Feb  4 18:09:59 2010
From: lists at redbonez.net (Adam Ford)
Date: Thu, 4 Feb 2010 19:09:59 -0700
Subject: [Freeswitch-users] Passing channel variables to mod_fifo and xml_cdr
Message-ID: <004301caa608$534747d0$f9d5d770$@net>

When sending a call through mod_fifo I seem to be losing my custom channel
variables that were assigned during prior processing of the call.  In my
example, I am trying to assign a unique identifier at the time the call
enters my FreeSWITCH system in order to more easily tie the xml_cdr logs
together.  This works great, until a call is processed through mod_fifo,
which drops my custom channel variable in the calls that it generates.  Is
it likely that I have something wrong with my config? Or does mod_fifo not
support the passing of custom channel variables?

 

The overall problem I am trying to solve is that mod_fifo generates a
separate a-leg for every time it rings an agent.  If the agent answers, the
a-leg log gets tied to the associated b-leg log with the uuids and I am able
to see the entire call in xml_cdr.  However, if the agent rejects the call
or doesn't answer, the a-leg is abandoned with seemingly no association back
to a b-leg log.  Anyone have a better suggestion for tying all these xml_cdr
logs together for a full picture of the call?  

 

-Adam

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100204/068d58f2/attachment.html 

From dujinfang at gmail.com  Thu Feb  4 18:46:01 2010
From: dujinfang at gmail.com (Seven Du)
Date: Fri, 5 Feb 2010 10:46:01 +0800
Subject: [Freeswitch-users] about bgapi
In-Reply-To: <b39273531002040519g71fa8a37uc9f0819c0a19772f@mail.gmail.com>
References: <314dc3f81002040333n28c15d92l5dd0aeef4dc5aa92@mail.gmail.com>
	<b39273531002040519g71fa8a37uc9f0819c0a19772f@mail.gmail.com>
Message-ID: <23f91031002041846n2ab49e30gb1e3e37a9b7d69ba@mail.gmail.com>

Or wait for a certain event.

http://fisheye.freeswitch.org/browse/~raw,r=14500/FreeSWITCH/contrib/seven/lua/gateway_report.lua

2010/2/4 Rupa Schomaker <rupa at rupa.com>:
> you can't. ?If you want to terminate it, then you should set a global var
> that the script periodically checks and if set the script should terminate
> itself.
>
> On Thu, Feb 4, 2010 at 5:33 AM, Chia-Yen Wu <xanlich at gmail.com> wrote:
>>
>> I tried to run a javascript by background API (bgapi jsrun test.js)
>> this javascript (test.js) wont automatically stop, I tried to kill it by
>> "uuid_kill" command with Job-UUID
>> but return "-ERR no such channel!", how can I kill this bgapi? btw my FS
>> run in windows.
>> thank you
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From wiltingtree at gmail.com  Thu Feb  4 18:48:07 2010
From: wiltingtree at gmail.com (Adam Wilt)
Date: Thu, 4 Feb 2010 21:48:07 -0500
Subject: [Freeswitch-users] Running a session asynchronously from a Lua
	script
Message-ID: <eb7797d71002041848h21e70a13t5ab3ba0004c6075a@mail.gmail.com>

Hi all,

I want to park an inbound call and play hold music while I simultaneously
place another outbound call.
But the hold music doesn't play while the lua script is placing the second
call. When the lua script ends, the hold music finally starts.
Here's my example code:

#!/usr/local/bin/lua

session:answer()
api = freeswitch.API()
api:executeString("bgapi uuid_park " .. tostring(session.uuid))
api:executeString("bgapi uuid_broadcast " .. tostring(session.uuid)  .. "
/freeswitch/sounds/music/8000/danza-espanola-op-37-h-142-xii-arabesca.wav")
local new_session =
freeswitch.Session("sofia/gateway/myprovider/15555555555")


So it seems like the script is blocking the original session, despite the
fact that I'm using bgapi.  I'd really appreciate if somebody could help me
with this.

By the way, I'm using FreeSWITCH 1.0.4 in Windows.

Thanks,
Adam
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100204/f4f4ea15/attachment-0001.html 

From brian at freeswitch.org  Thu Feb  4 19:23:46 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 4 Feb 2010 21:23:46 -0600
Subject: [Freeswitch-users] Running a session asynchronously from a Lua
	script
In-Reply-To: <eb7797d71002041848h21e70a13t5ab3ba0004c6075a@mail.gmail.com>
References: <eb7797d71002041848h21e70a13t5ab3ba0004c6075a@mail.gmail.com>
Message-ID: <0BD2ED36-2279-412A-AF43-554E15B596E2@freeswitch.org>

I wouldn't do it like that...

I would let FreeSWITCH do what its good at and stop trying to create and manage sessions manually.

-- You can optionally answer but you need to set transfer_ringback instead of ringback.
session:setVariable("ringback", "local_stream://moh");                                                                                                          
session:setVariable("ignore_early_media", "true");                                                                                                              
session:execute("bridge","user/1007");  

Which can also be expressed in pure XML as such:

<extension name="1234">
  <condition field="destination_number" expression="^1234$">                                                                                                
    <action application="set" data="ringback=local_stream://moh"/>
    <action application="set" data="ignore_early_media=true"/>
    <action application="bridge" data="user/1007"/>
  </condition>
</extension>

There is really no need to do this with Lua.  The XML dialplan can do some VERY complex things once you wrap your head around it.

In addition I would recommend you get the latest 1.0.5 build http://files-sync.freeswitch.org/windows_installer/  Its tagged as 1.0.4 and shouldn't be.

/b



On Feb 4, 2010, at 8:48 PM, Adam Wilt wrote:

> Hi all,
> 
> I want to park an inbound call and play hold music while I simultaneously place another outbound call.
> But the hold music doesn't play while the lua script is placing the second call. When the lua script ends, the hold music finally starts. 
> Here's my example code:
> 
> #!/usr/local/bin/lua
>    
> session:answer()
> api = freeswitch.API()
> api:executeString("bgapi uuid_park " .. tostring(session.uuid))
> api:executeString("bgapi uuid_broadcast " .. tostring(session.uuid)  .. " /freeswitch/sounds/music/8000/danza-espanola-op-37-h-142-xii-arabesca.wav")
> local new_session = freeswitch.Session("sofia/gateway/myprovider/15555555555")
> 
> So it seems like the script is blocking the original session, despite the fact that I'm using bgapi.  I'd really appreciate if somebody could help me with this.
> 
> By the way, I'm using FreeSWITCH 1.0.4 in Windows.
> 
> Thanks,
> Adam

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100204/c4c1fc03/attachment.html 

From ustcorporation at yahoo.com  Thu Feb  4 19:38:53 2010
From: ustcorporation at yahoo.com (Darren C.)
Date: Thu, 4 Feb 2010 19:38:53 -0800 (PST)
Subject: [Freeswitch-users]  Seeking Advice on SIP Phones like Aastra
Message-ID: <845952.61278.qm@web33007.mail.mud.yahoo.com>

Tim,
 
Many thanks for your response.  I posted this message on the Dev list and all I heard was crickets.  I would think a web GUI for a phone would be in demand by the FS community....
 
We are doing something similar to what you described.  We?re developing a rather complex IVR/Switching application and it currently does all its database writes via our Web Service to an MS SQL database.  We have a web site that is updated with call details via the web service?s backend database.  From this web interface a user can see counts of voicemails, see call activity, play voicemails, see calls in progress, record calls, etc.  It?s a specialized application so it doesn?t have every PBX feature but this is what we wanted to do with a high-end SIP Phone to replace our office PBX.
 
Currently we have an ESI (Estech) E-Class PBX that uses normal digital phones as well as proprietary VOIP (non-SIP) phones.  I think these really nice SIP phones with huge color touchscreens would be much better than even high-end proprietary digital phones + we?d get all the benefits of FS.  We?ll just need to add all the basic PBX capabilities to the phone?s GUI to see how many voicemails are waiting, how many lines are in use, button for call transfer, etc.
 
As you mentioned:  ?you can do a DB query to find out what voicemail messages a user has, or what calls are active, etc.?  We are keeping track of all this ourselves via our web service?s backend database.  I?m not sure I need to do this for everything but we are.  It?s a multi-tenant system with hundreds of tenants so I?m guessing I might lose some needed relationships by querying FS but I?m going to re-visit this?I will make sure I can?t just query FS like you?re doing for some of these things?we?ve never turned on ODBC or even looked for the documentation as to what FS stores.  We have to run FS in Linux for some Sangoma stuff but we?re Windows people so that is another reason we store via web services to an MS SQLbackend.
 
I was worried I?d get one of these fancy phones and find out it doesn?t support important SIP/FS features rendering the color touchscreen useless.  I?ve never owned one of these SIP phones, just used various softphones.  But thanks to you I?ll get an Aastra 6739i and give it a try.
 
I have a fulltime programmer working on this system on and off now for over a year but this has been mostly developing the IVR.  We haven't made any attempt at using FS as a PBX.  So based on your comments I think I?ll purchase an Aastra 6739i and develop a custom SIP Phone GUI interface with FS.
 
If you or others would like to collaborate on an Aastra 6739i phone GUI for FS, feel free to contact me.  We can try to make it extensible for other phones as well.  My email is ustcor<remove this junk>poration at yah<remove this too>oo.com.  
 
Thanks,

teldev


      
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100204/de6721c7/attachment-0001.html 

From wiltingtree at gmail.com  Thu Feb  4 19:49:36 2010
From: wiltingtree at gmail.com (Adam Wilt)
Date: Thu, 4 Feb 2010 22:49:36 -0500
Subject: [Freeswitch-users] Running a session asynchronously from a Lua
	script
In-Reply-To: <0BD2ED36-2279-412A-AF43-554E15B596E2@freeswitch.org>
References: <eb7797d71002041848h21e70a13t5ab3ba0004c6075a@mail.gmail.com>
	<0BD2ED36-2279-412A-AF43-554E15B596E2@freeswitch.org>
Message-ID: <eb7797d71002041949t7a3fd84dgcfa3f63c69465612@mail.gmail.com>

Brian,

Thanks for the reply.
The problem is I don't want the two parties to speak to each other. I want
one party to wait on hold while the system interacts with the other party.
Then the system will hang-up on the second party and start interacting with
the first party again.

Thanks,
Adam


On Thu, Feb 4, 2010 at 10:23 PM, Brian West <brian at freeswitch.org> wrote:

> I wouldn't do it like that...
>
> I would let FreeSWITCH do what its good at and stop trying to create and
> manage sessions manually.
>
> -- You can optionally answer but you need to set transfer_ringback instead
> of ringback.
> session:setVariable("ringback", "local_stream://moh");
>
>
> session:setVariable("ignore_early_media", "true");
>
>
> session:execute("bridge","user/1007");
>
> Which can also be expressed in pure XML as such:
>
> <extension name="1234">
>   <condition field="destination_number" expression="^1234$">
>
>
>     <action application="set" data="ringback=local_stream://moh"/>
>     <action application="set" data="ignore_early_media=true"/>
>     <action application="bridge" data="user/1007"/>
>   </condition>
> </extension>
>
> There is really no need to do this with Lua.  The XML dialplan can do some
> VERY complex things once you wrap your head around it.
>
> In addition I would recommend you get the latest 1.0.5 build
> http://files-sync.freeswitch.org/windows_installer/  Its tagged as 1.0.4
> and shouldn't be.
>
> /b
>
>
>
> On Feb 4, 2010, at 8:48 PM, Adam Wilt wrote:
>
> Hi all,
>
> I want to park an inbound call and play hold music while I simultaneously
> place another outbound call.
> But the hold music doesn't play while the lua script is placing the second
> call. When the lua script ends, the hold music finally starts.
> Here's my example code:
>
> #!/usr/local/bin/lua
>
> session:answer()
> api = freeswitch.API()
> api:executeString("bgapi uuid_park " .. tostring(session.uuid))
> api:executeString("bgapi uuid_broadcast " .. tostring(session.uuid)  .. "
> /freeswitch/sounds/music/8000/danza-espanola-op-37-h-142-xii-arabesca.wav")
> local new_session =
> freeswitch.Session("sofia/gateway/myprovider/15555555555")
>
>
> So it seems like the script is blocking the original session, despite the
> fact that I'm using bgapi.  I'd really appreciate if somebody could help me
> with this.
>
> By the way, I'm using FreeSWITCH 1.0.4 in Windows.
>
> Thanks,
> Adam
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100204/22e157d3/attachment.html 

From brian at freeswitch.org  Thu Feb  4 19:57:24 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 4 Feb 2010 21:57:24 -0600
Subject: [Freeswitch-users] Running a session asynchronously from a Lua
	script
In-Reply-To: <eb7797d71002041949t7a3fd84dgcfa3f63c69465612@mail.gmail.com>
References: <eb7797d71002041848h21e70a13t5ab3ba0004c6075a@mail.gmail.com>
	<0BD2ED36-2279-412A-AF43-554E15B596E2@freeswitch.org>
	<eb7797d71002041949t7a3fd84dgcfa3f63c69465612@mail.gmail.com>
Message-ID: <DD852357-BDC6-425D-81ED-1AE807EE2F48@freeswitch.org>

ok the use the api to "originate" the call to say an extension that plays music.  But if they never talk to each other what is the point of the two calls in the same script?

/b

On Feb 4, 2010, at 9:49 PM, Adam Wilt wrote:

> Brian, 
> 
> Thanks for the reply.  
> The problem is I don't want the two parties to speak to each other. I want one party to wait on hold while the system interacts with the other party. Then the system will hang-up on the second party and start interacting with the first party again.
> 
> Thanks,
> Adam



From nagalenoj at gmail.com  Thu Feb  4 20:41:39 2010
From: nagalenoj at gmail.com (Nagalenoj H.)
Date: Fri, 5 Feb 2010 10:11:39 +0530
Subject: [Freeswitch-users] A leg disconnects after uuid_bridge?
In-Reply-To: <191c3a031002040754r2987a03fo18ee2e1ceb239f6a@mail.gmail.com>
References: <fe67a3341002021934m548cde89p4d48381c5abe02a0@mail.gmail.com>
	<191c3a031002022108ta22287ev40fc36420abba2e5@mail.gmail.com>
	<fe67a3341002030112yeea379u90c22c482614c57@mail.gmail.com>
	<87f2f3b91002030605j7a1f9a02g665ba043c6ed7f7b@mail.gmail.com>
	<fe67a3341002040132n3aa2a897m60c5b9f1ca6b347f@mail.gmail.com>
	<0A571E14-BB58-4A3D-996A-9D5205668E85@freeswitch.org>
	<fe67a3341002040452v5468fd1fh96d1c5062091474b@mail.gmail.com>
	<191c3a031002040754r2987a03fo18ee2e1ceb239f6a@mail.gmail.com>
Message-ID: <fe67a3341002042041u1ee70b86u3bc91adbb6fb3d52@mail.gmail.com>

Sorry., I couldn't understand its behavior.

Let me ask the same question in this way.

* hangup_after_bridge is set to false.
* In outbound socket, first I answer the call.
* When I do a bridge to a extension (1001), after 1001 disconnects the call.
I am able to make another call.
      sendmsg
      call-command: execute
      execute-app-name: bridge
      execute-app-arg: user/1001

* When I originate a call to extension (1001), after 1001 disconnects the
call. I'm unable to make another call, because my session is also getting
closed.
      api originate user/1001 &park

      Content-Type: api/response
      Content-Length: 41

     +OK 1fac17ce-120b-11df-a878-d9c7fbcf71c4

      sendmsg
      call-command: execute
      execute-app-name: intercept
      execute-app-arg: 1fac17ce-120b-11df-a878-d9c7fbcf71c4

* In both the case, the call is getting bridged to an extension and
hangup_after_bridge is false.
* When bridge doesn't need any other variables to set to continue, why
intercept needs a explicit park after bridge.?

Hope, this has some clarity.,

On Thu, Feb 4, 2010 at 9:24 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

>
>
>    1. 2010-02-04 14:30:09.574084 [DEBUG] switch_core_state_machine.c:354 (
>    sofia/internal/1010 at 192.168.1.222) State SOFT_EXECUTE going to sleep
>    2. 2010-02-04 14:30:09.574084 [DEBUG] switch_core_state_machine.c:314 (
>    sofia/internal/1010 at 192.168.1.222) Running State Change CS_EXECUTE
>    3. 2010-02-04 14:30:09.574084 [DEBUG] switch_core_state_machine.c:348 (
>    sofia/internal/1010 at 192.168.1.222) State EXECUTE
>    4. 2010-02-04 14:30:09.574084 [DEBUG] mod_sofia.c:181 sofia/internal/
>    1010 at 192.168.1.222 SOFIA EXECUTE
>    5. 2010-02-04 14:30:09.574084 [DEBUG] switch_core_state_machine.c:159sofia/internal/
>    1010 at 192.168.1.222 Standard EXECUTE
>    6. 2010-02-04 14:30:09.574084 [NOTICE] switch_core_state_machine.c:187Hangup sofia/internal/
>    1010 at 192.168.1.222 [CS_EXECUTE] [NORMAL_CLEARING]
>
>
>
> Your channel went back to EXECUTE as expected then it hungup because there
> were no more instructions in your dial plan for it to execute.  So it is
> working as expected.
>
> Consider using transfer_after_bridge variable or park_after bridge to make
> it stay around when the call is over.
>
>
>
>
> On Thu, Feb 4, 2010 at 6:52 AM, Nagalenoj H. <nagalenoj at gmail.com> wrote:
>
>> By using create_uuid. I've also tried without giving origination_uuid.
>> But, the result is same.
>>
>> --
>> Regards,
>> Nagalenoj H.
>>
>>
>> On Thu, Feb 4, 2010 at 5:42 PM, Brian West <brian at freeswitch.org> wrote:
>>
>>> Where are you getting this UUID?
>>>
>>> /b
>>>
>>> On Feb 4, 2010, at 3:32 AM, Nagalenoj H. wrote:
>>>
>>> > api originate
>>> {origination_uuid=c6d2e0e2-0b2f-11df-9e84-fb15c3cd8565}user/1001 &park
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Regards,
Nagalenoj H.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100205/203be6f6/attachment.html 

From matt at webcontracts.co.uk  Thu Feb  4 04:22:13 2010
From: matt at webcontracts.co.uk (Matthew Law)
Date: Thu, 4 Feb 2010 12:22:13 -0000
Subject: [Freeswitch-users] Simple IAX2 setup - help with converting from
 asterisk to freeswitch
Message-ID: <91093d24f2166d220acf2ab19115f95d.squirrel@www.webcontracts.co.uk>

Hi,

I have compiled freeswitch trunk from svn on debian.  I am trying to
convert the following simple working asterisk config to freeswitch and I
would be really grateful if someone could point me in the right direction:

pbx:/etc/asterisk# cat iax.conf
[general]
bindport=4569
bindaddr=my.ip.address
jitterbuffer=yes
disallow=all
allow=ulaw
allow=alaw
context=deadend

[voiptalk]
type=peer
username=XXXXXX
secret=XXXXXX
host=iax.voiptalk.org

[0843XXXXXX]
type=friend
username=08433XXXXXX
context=incoming
requirecalltoken=auto

[1000]
type=friend
host=dynamic
mailbox=1000
secret=XXXXXX
context=phones
requirecalltoken=auto

pbx:/etc/asterisk# cat extensions.conf
[general]
autofallthrough=yes

[outgoing]
exten => _0[1-9].,1,Dial(IAX2/XXXXXX at voiptalk/44${EXTEN:1})
exten => _00.,1,Dial(IAX2/XXXXXX at voiptalk/${EXTEN:2})

[internal]
exten => 901,1,VoiceMailMain()
exten => 901,2,Hangup()
exten => 902,1,MeetMe(1234,cdM)
exten => 902,2,Hangup()

[incoming]
exten => XXXXXXXX,1,Dial(IAX2/1000,30)
exten => XXXXXXXX,2,VoiceMail(1000 at internal)

[phones]
include => internal
include => outgoing

pbx:/etc/asterisk# cat voicemail.conf
[internal]
1000 => XXXX,Matt,me at myomain.com
format=wav49
maxsilence=0


Sorry to be a bonehead, but I'm struggling with the wiki docs (especially
with regard to IAX2) and also concerned about security.  I installed the
samples when I compiled freeswitch but wondering if that is a security
risk as the PBX box is on the public internet?

Many thanks,

Matt





From tim at novion.ru  Thu Feb  4 11:47:57 2010
From: tim at novion.ru (=?KOI8-R?B?9MnN1dIg98HMydvF1w==?=)
Date: Thu, 4 Feb 2010 22:47:57 +0300
Subject: [Freeswitch-users] No reINVITE when bridging two sessions from
	JavaScript with bypass_media_after_bridge=true
Message-ID: <8e9d67561002041147q6c135546u1f7e1887529a916d@mail.gmail.com>

Dear colleagues,

The task is to start two sessions from JS script and then bridge them
in no-media mode. Unfirtunately, FreeSwitch does not reINVITE the
peers after bridging.

Here is my script:

SCRIPT #1 - 'callback-session.js' <<<<<<<<<<<<<<<<<<<<<<<<<<
session = new Session("{bypass_media_after_bridge=true,ignore_early_media=true,hangup_after_bridge=true}user/1001");
session2 = new Session("{ignore_early_media=true}user/1001");
bridge(session, session2);
>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>

Then I run this script from the freeswitch console:

freeswitch at internal> jsrun callback-session.js

And there is no re-invites between peers, peers get connected and the
traffic goes through FS

BUT!!! Here is another scipt:

SCRIPT #2 - 'callback-bridge.js' <<<<<<<<<<<<<<<<<<<<<<<<<<
session = new Session("{bypass_media_after_bridge=true,continue_on_fail=true,ignore_early_media=true}user/1001");
session.execute("bridge","{ignore_early_media=false,originate_timeout=90}user/1001");
>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>

When I run this script, FreeSwitch successfuly sends reINVITES to both
users after bridge, so they exchange media directly, not through FS.
In my task, I need to have control when B-leg establishes (to start
billing correctly), so I need to get the first scenario working.

What I've already tried and did not succeed:
1) set bypass_media=true, on A leg only, on B leg only, on both legs
2) set bypass_media_after_bridge=true, on A leg only, on B leg only,
on both legs

Is there something I do wrong in the first script? What should I do to
make FS reINVITE peers? Many thanks in advance!

Best regards,
Timur Valishev
sip:tim at novion.ru


From christian at officepools.com  Thu Feb  4 14:08:56 2010
From: christian at officepools.com (Christain Jensen)
Date: Thu, 4 Feb 2010 14:08:56 -0800
Subject: [Freeswitch-users] Looking for some good/cheap desktop phones
Message-ID: <3f2e071b5aea5e11bb37a37e435a33b7@mail.gmail.com>

Hi,



I am looking for a vendor for some (3-5) desktop voip phones. Any
suggestions?



Christian
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100204/e206ac28/attachment.html 

From tim at novion.ru  Thu Feb  4 21:46:00 2010
From: tim at novion.ru (=?KOI8-R?B?9MnN1dIg98HMydvF1w==?=)
Date: Fri, 5 Feb 2010 08:46:00 +0300
Subject: [Freeswitch-users] No reINVITE when bridging two sessions from
	JavaScript with bypass_media_after_bridge=true
In-Reply-To: <8e9d67561002041147q6c135546u1f7e1887529a916d@mail.gmail.com>
References: <8e9d67561002041147q6c135546u1f7e1887529a916d@mail.gmail.com>
Message-ID: <8e9d67561002042146l3b4265f9sb87270d5b0adac68@mail.gmail.com>

Dear colleagues,

The task is to start two sessions from JS script and then bridge them
in no-media mode. Unfirtunately, FreeSwitch does not reINVITE the
peers after bridging.

Here is my script:

SCRIPT #1 - 'callback-session.js' <<<<<<<<<<<<<<<<<<<<<<<<<<
session = new Session("{bypass_media_after_bridge=true,ignore_early_media=true,hangup_after_bridge=true}user/1001");
session2 = new Session("{ignore_early_media=true}user/1001");
bridge(session, session2);
>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>

Then I run this script from the freeswitch console:

freeswitch at internal> jsrun callback-session.js

And there is no re-invites between peers, peers get connected and the
traffic goes through FS

BUT!!! Here is another scipt:

SCRIPT #2 - 'callback-bridge.js' <<<<<<<<<<<<<<<<<<<<<<<<<<
session = new Session("{bypass_media_after_bridge=true,continue_on_fail=true,ignore_early_media=true}user/1001");
session.execute("bridge","{ignore_early_media=false,originate_timeout=90}user/1001");
>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>

When I run this script, FreeSwitch successfuly sends reINVITES to both
users after bridge, so they exchange media directly, not through FS.
In my task, I need to have control when B-leg establishes (to start
billing correctly), so I need to get the first scenario working.

What I've already tried and did not succeed:
1) set bypass_media=true, on A leg only, on B leg only, on both legs
2) set bypass_media_after_bridge=true, on A leg only, on B leg only,
on both legs

Is there something I do wrong in the first script? What should I do to
make FS reINVITE peers? Many thanks in advance!

Best regards,
Timur Valishev
sip:tim at novion.ru


From tim at novion.ru  Thu Feb  4 22:20:41 2010
From: tim at novion.ru (=?KOI8-R?B?9MnN1dIg98HMydvF1w==?=)
Date: Fri, 5 Feb 2010 09:20:41 +0300
Subject: [Freeswitch-users] No reINVITE when bridging two sessions from
	JavaScript with bypass_media_after_bridge=true
In-Reply-To: <8e9d67561002041147q6c135546u1f7e1887529a916d@mail.gmail.com>
References: <8e9d67561002041147q6c135546u1f7e1887529a916d@mail.gmail.com>
Message-ID: <8e9d67561002042220t59dfbf97x50c2a0bf804c5e5f@mail.gmail.com>

Dear colleagues,

The task is to start two sessions from JS script and then bridge them
in no-media mode. Unfirtunately, FreeSwitch does not reINVITE the
peers after bridging.

Here is my script:

SCRIPT #1 - 'callback-session.js' <<<<<<<<<<<<<<<<<<<<<<<<<<
session = new Session("{bypass_media_after_bridge=true,ignore_early_media=true,hangup_after_bridge=true}user/1001");
session2 = new Session("{ignore_early_media=true}user/1001");
bridge(session, session2);
>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>

Then I run this script from the freeswitch console:

freeswitch at internal> jsrun callback-session.js

And there is no re-invites between peers, peers get connected and the
traffic goes through FS. Is there something I do wrong in the script?
What should I do to
make FS reINVITE peers? Many thanks in advance!

Best regards,
Timur Valishev


From tim at novion.ru  Thu Feb  4 22:36:19 2010
From: tim at novion.ru (=?KOI8-R?B?9MnN1dIg98HMydvF1w==?=)
Date: Fri, 5 Feb 2010 09:36:19 +0300
Subject: [Freeswitch-users] Looking for some good/cheap desktop phones
In-Reply-To: <3f2e071b5aea5e11bb37a37e435a33b7@mail.gmail.com>
References: <3f2e071b5aea5e11bb37a37e435a33b7@mail.gmail.com>
Message-ID: <8e9d67561002042236s434e0643od322551de8720be1@mail.gmail.com>

Have a look at Yealink (Skypemate) and Fanvill

2010/2/5 Christain Jensen <christian at officepools.com>:
> Hi,
>
>
>
> I am looking for a vendor for some (3-5) desktop voip phones. Any
> suggestions?
>
>
>
> Christian
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From brian at freeswitch.org  Thu Feb  4 22:41:27 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 5 Feb 2010 00:41:27 -0600
Subject: [Freeswitch-users] No reINVITE when bridging two sessions from
	JavaScript with bypass_media_after_bridge=true
In-Reply-To: <8e9d67561002041147q6c135546u1f7e1887529a916d@mail.gmail.com>
References: <8e9d67561002041147q6c135546u1f7e1887529a916d@mail.gmail.com>
Message-ID: <3D739D7F-F5C6-4266-B04E-9413DCFDA6AF@freeswitch.org>

http://wiki.freeswitch.org/wiki/Bypass_Media

Also make sure you're on SVN trunk.

/b

On Feb 4, 2010, at 1:47 PM, ????? ??????? wrote:

> Is there something I do wrong in the first script? What should I do to
> make FS reINVITE peers? Many thanks in advance!



From mike at jerris.com  Thu Feb  4 22:43:00 2010
From: mike at jerris.com (Michael Jerris)
Date: Fri, 5 Feb 2010 01:43:00 -0500
Subject: [Freeswitch-users] Simple IAX2 setup - help with converting
	from asterisk to freeswitch
In-Reply-To: <91093d24f2166d220acf2ab19115f95d.squirrel@www.webcontracts.co.uk>
References: <91093d24f2166d220acf2ab19115f95d.squirrel@www.webcontracts.co.uk>
Message-ID: <E1E3F6EF-1A1E-4891-B371-7A3BF0988E95@jerris.com>

iax2 support has been removed from FreeSWITCH in current trunk and will not be in the 1.0.5 release.

On Feb 4, 2010, at 7:22 AM, Matthew Law wrote:

> Hi,
> 
> I have compiled freeswitch trunk from svn on debian.  I am trying to
> convert the following simple working asterisk config to freeswitch and I
> would be really grateful if someone could point me in the right direction:
> 
<snip>

> Sorry to be a bonehead, but I'm struggling with the wiki docs (especially
> with regard to IAX2) and also concerned about security.  I installed the
> samples when I compiled freeswitch but wondering if that is a security
> risk as the PBX box is on the public internet?

Mike



From brian at freeswitch.org  Thu Feb  4 22:43:37 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 5 Feb 2010 00:43:37 -0600
Subject: [Freeswitch-users] Looking for some good/cheap desktop phones
In-Reply-To: <8e9d67561002042236s434e0643od322551de8720be1@mail.gmail.com>
References: <3f2e071b5aea5e11bb37a37e435a33b7@mail.gmail.com>
	<8e9d67561002042236s434e0643od322551de8720be1@mail.gmail.com>
Message-ID: <409D0ACF-ADA4-4D02-8B7E-71EC1B1BAFBC@freeswitch.org>

And all of those are awful phones.  They don't even make good paper weights.

You can't have good and cheap in the same sentence when talking about VoIP phones.  You have to take your pick between quality (good) and price (cheap) you can't have both at once.

/b


On Feb 5, 2010, at 12:36 AM, ????? ??????? wrote:

> Have a look at Yealink (Skypemate) and Fanvill



From tim at novion.ru  Thu Feb  4 23:01:42 2010
From: tim at novion.ru (=?KOI8-R?B?9MnN1dIg98HMydvF1w==?=)
Date: Fri, 5 Feb 2010 10:01:42 +0300
Subject: [Freeswitch-users] Looking for some good/cheap desktop phones
In-Reply-To: <409D0ACF-ADA4-4D02-8B7E-71EC1B1BAFBC@freeswitch.org>
References: <3f2e071b5aea5e11bb37a37e435a33b7@mail.gmail.com>
	<8e9d67561002042236s434e0643od322551de8720be1@mail.gmail.com>
	<409D0ACF-ADA4-4D02-8B7E-71EC1B1BAFBC@freeswitch.org>
Message-ID: <8e9d67561002042301v76129abfj3dcbc70ed073752c@mail.gmail.com>

Sure, those phones do not deliver superior usability, but they at
least give the best sound among budget models.

2010/2/5 Brian West <brian at freeswitch.org>:
> And all of those are awful phones. ?They don't even make good paper weights.
>
> You can't have good and cheap in the same sentence when talking about VoIP phones. ?You have to take your pick between quality (good) and price (cheap) you can't have both at once.
>
> /b
>
>
> On Feb 5, 2010, at 12:36 AM, ????? ??????? wrote:
>
>> Have a look at Yealink (Skypemate) and Fanvill
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From ustcorporation at yahoo.com  Thu Feb  4 23:11:54 2010
From: ustcorporation at yahoo.com (Darren C.)
Date: Thu, 4 Feb 2010 23:11:54 -0800 (PST)
Subject: [Freeswitch-users] Seeking Advice on SIP Phones like Aastra 6739i
	or Snom 870 that have good interoperability with FreeSWITCH
Message-ID: <174266.76158.qm@web33008.mail.mud.yahoo.com>

Tim,

Many thanks for your response.  My first response became an orphan/new post on the list...I think if subject line is too long reply looses thread.

We are doing something similar to what you described.  We?re developing a rather complex IVR/Switching application and it currently does all its database writes via our Web Service to an MS SQL database.  We have a web site that is updated with call details via the web service?s backend database.  From this web interface a user can see counts of voicemails, see call activity, play voicemails, see calls in progress, record calls, etc.  It?s a specialized application so it doesn?t have every PBX feature but this is what we wanted to do with a high-end SIP Phone to replace our office PBX.

Currently we have an ESI (Estech) E-Class PBX that uses normal digital phones as well as proprietary VOIP (non-SIP) phones.  I think these really nice SIP phones with huge color touchscreens would be much better than even high-end proprietary digital phones + we?d get all the benefits of FS.  We?ll just need to add all the basic PBX capabilities to the phone?s GUI to see how many voicemails are waiting, how many lines are in use, button for call transfer, etc.

As you mentioned:  ?you can do a DB query to find out what voicemail messages a user has, or what calls are active, etc.?  We are keeping track of all this ourselves via our web service?s backend database.  I?m not sure I need to do this for everything but we are.  It?s a multi-tenant system with hundreds of tenants so I?m guessing I might lose some needed relationships by querying FS but I?m going to re-visit this?I will make sure I can?t just query FS like you?re doing for some of these things?we?ve never turned on ODBC or even looked for the documentation as to what FS stores.  We have to run FS in Linux for some Sangoma stuff but we?re Windows people so that is another reason we store via web services to an MS SQLbackend.

I was worried I?d get one of these fancy phones and find out it doesn?t support important SIP/FS features rendering the color touchscreen useless.  I?ve never owned one of these SIP phones, just used various softphones.  But thanks to you I?ll get an Aastra 6739i and give it a try.

I have a fulltime programmer working on this system on and off now for over a year but this has been mostly developing the IVR.  We haven't made any attempt at using FS as a PBX.  So based on your comments I think I?ll purchase an Aastra 6739i and develop a custom SIP Phone GUI interface with FS.

If you or others would like to collaborate on an Aastra 6739i phone GUI for FS, feel free to contact me.  We can try to make it extensible for other phones as well.  My email is ustcor<remove this junk>poration at yah<remove this too>oo.com.  

Thanks,

teldev


      
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100204/eb1b26d0/attachment-0001.html 

From tim at novion.ru  Fri Feb  5 00:18:15 2010
From: tim at novion.ru (Timur Valishev)
Date: Fri, 5 Feb 2010 11:18:15 +0300
Subject: [Freeswitch-users] No reINVITE when bridging two sessions from
	JavaScript with bypass_media_after_bridge=true
In-Reply-To: <3D739D7F-F5C6-4266-B04E-9413DCFDA6AF@freeswitch.org>
References: <8e9d67561002041147q6c135546u1f7e1887529a916d@mail.gmail.com>
	<3D739D7F-F5C6-4266-B04E-9413DCFDA6AF@freeswitch.org>
Message-ID: <8e9d67561002050018q24e3a806kc8807b55268e759b@mail.gmail.com>

Thank you for reply, Brian!

http://wiki.freeswitch.org/wiki/Bypass_Media says:
*>Can I use bypass media when executing the bridge application from a
javascript? *
*>Of course you can, all it takes is setting the bypass_media session
variable to true before the bridge: *
*>session.setVariable('bypass_media', 'true'*);

I've modified my script to make sure: <<<<<<<<<<<<<<<<<<<<<<<<<<<<,
*session = new Session(*
*"{ignore_early_media=true,hangup_after_bridge=true}sofia/external/
timwork at novion.ru"*
*);*
*session2 = new Session(*
*"{ignore_early_media=true}sofia/external/timwork at novion.ru"*
*);*
*session.setVariable('bypass_media', 'true');*
*session2.setVariable('bypass_media', 'true');*
*bridge(session, session2);*
>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>

But there is still no reINVITE =(

SVN trunk I'm building from seems to be fresh: <<<<<<<<<<<<<<<
*[root at sip freeswitch.trunk]# svn info*
*Path: .*
*URL: http://svn.freeswitch.org:/svn/freeswitch/trunk*
*Repository Root: http://svn.freeswitch.org:/svn*
*Repository UUID: d0543943-73ff-0310-b7d9-9358b9ac24b2*
*Revision: 16561*
*Node Kind: directory*
*Schedule: normal*
*Last Changed Author: mcollins*
*Last Changed Rev: 16561*
*Last Changed Date: 2010-02-03 04:53:31 +0300 (Wed, 03 Feb 2010)*
>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>

Do you have any ideas how to make FS reINVITE in this scenario?

2010/2/5 Brian West <brian at freeswitch.org>:
> http://wiki.freeswitch.org/wiki/Bypass_Media
>
> Also make sure you're on SVN trunk.
>
> /b
>
> On Feb 4, 2010, at 1:47 PM, ????? ??????? wrote:
>
>> Is there something I do wrong in the first script? What should I do to
>> make FS reINVITE peers? Many thanks in advance!
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100205/f5f42fee/attachment.html 

From tayeb.meftah at gmail.com  Fri Feb  5 04:09:08 2010
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Fri, 05 Feb 2010 13:09:08 +0100
Subject: [Freeswitch-users] Looking for some good/cheap desktop phones
In-Reply-To: <3f2e071b5aea5e11bb37a37e435a33b7@mail.gmail.com>
References: <3f2e071b5aea5e11bb37a37e435a33b7@mail.gmail.com>
Message-ID: <4B6C0A64.3060500@gmail.com>

hi
try linksys SPA901
Le 04/02/2010 23:08, Christain Jensen a ?crit :
>
> Hi,
>
> I am looking for a vendor for some (3-5) desktop voip phones. Any 
> suggestions?
>
> Christian
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>    

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100205/0b83da11/attachment.html 

From jcasale at activenetwerx.com  Fri Feb  5 04:41:41 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Fri, 5 Feb 2010 12:41:41 +0000
Subject: [Freeswitch-users] Looking for some good/cheap desktop phones
In-Reply-To: <4B6C0A64.3060500@gmail.com>
References: <3f2e071b5aea5e11bb37a37e435a33b7@mail.gmail.com>
	<4B6C0A64.3060500@gmail.com>
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E90CD9E1@prato.activenetwerx.local>

>try linksys SPA901

Use Linksys support just once, then tell me if you still want
any of their product...


From dave at 3c.co.uk  Fri Feb  5 05:04:05 2010
From: dave at 3c.co.uk (David Knell)
Date: Fri, 05 Feb 2010 06:04:05 -0700
Subject: [Freeswitch-users] Looking for some good/cheap desktop phones
In-Reply-To: <8e9d67561002042301v76129abfj3dcbc70ed073752c@mail.gmail.com>
References: <3f2e071b5aea5e11bb37a37e435a33b7@mail.gmail.com>
	<8e9d67561002042236s434e0643od322551de8720be1@mail.gmail.com>
	<409D0ACF-ADA4-4D02-8B7E-71EC1B1BAFBC@freeswitch.org>
	<8e9d67561002042301v76129abfj3dcbc70ed073752c@mail.gmail.com>
Message-ID: <1265375045.12871.27.camel@local.freepabx.com>

Some notes from a grumpy old luddite:

I have one of the Yealink USB desk speakerphones, and I don't think you
can get a better usability and audio quality to price ratio anywhere on
the market.  And it just worked.

The Aastra 6757i (also on my somewhat cluttered desk) cost more than ten
times as much, was a PITA to set up (its UPnP sporadically crashed my
WiFi router which was entertaining until I worked out what was going on
and turned it off), has a configuration interface that would send Steve
Jobs out hunting those responsible with an elephant gun were it an Apple
product and, once it was finally configured and working, gave me really
no more usable functionality on the deskphone than the el cheapo one
above.  The cordless handset's nice to have, though.

--Dave

> Sure, those phones do not deliver superior usability, but they at
> least give the best sound among budget models.
> 
> 2010/2/5 Brian West <brian at freeswitch.org>:
> > And all of those are awful phones.  They don't even make good paper weights.
> >
> > You can't have good and cheap in the same sentence when talking about VoIP phones.  You have to take your pick between quality (good) and price (cheap) you can't have both at once.
> >
> > /b
> >
> >
> > On Feb 5, 2010, at 12:36 AM, ????? ??????? wrote:
> >
> >> Have a look at Yealink (Skypemate) and Fanvill
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From yehavi.bourvine at gmail.com  Fri Feb  5 05:07:09 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Fri, 5 Feb 2010 15:07:09 +0200
Subject: [Freeswitch-users] Looking for some good/cheap desktop phones
In-Reply-To: <8e9d67561002042301v76129abfj3dcbc70ed073752c@mail.gmail.com>
References: <3f2e071b5aea5e11bb37a37e435a33b7@mail.gmail.com>
	<8e9d67561002042236s434e0643od322551de8720be1@mail.gmail.com>
	<409D0ACF-ADA4-4D02-8B7E-71EC1B1BAFBC@freeswitch.org>
	<8e9d67561002042301v76129abfj3dcbc70ed073752c@mail.gmail.com>
Message-ID: <d361ca7b1002050507n5e564621oa6cf7ace005dd5b8@mail.gmail.com>

>From my experience Polycom and SNOM are expensive but give you what you
need. Polycom is more intutive to the users but more cumbersome for the
manager to deploy; SNOM is somewhat less intuitive to the user but
everything can be set via the WEB interface.

If you talk about 4-5 phones, then probably SNOM is the choice. It also
depends about the specific functions you want to use. I our specific
environment (high use of BLF and shared lines) Polycom wins because it
handles these functions just as the user expects.

I did not try Aastra so cannot testify. We did test Yealink, Thomson,
Asterphone, SipTip and maybe others I forgot. Cisco also seems good but
Cisco does not supply the required socumentation to make them fully working.

                    Regards, __Yehavi:

2010/2/5 ????? ??????? <tim at novion.ru>

> Sure, those phones do not deliver superior usability, but they at
> least give the best sound among budget models.
>
> 2010/2/5 Brian West <brian at freeswitch.org>:
>  > And all of those are awful phones.  They don't even make good paper
> weights.
> >
> > You can't have good and cheap in the same sentence when talking about
> VoIP phones.  You have to take your pick between quality (good) and price
> (cheap) you can't have both at once.
> >
> > /b
> >
> >
> > On Feb 5, 2010, at 12:36 AM, ????? ??????? wrote:
> >
> >> Have a look at Yealink (Skypemate) and Fanvill
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100205/183cf9e4/attachment.html 

From tculjaga at gmail.com  Fri Feb  5 05:21:25 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Fri, 5 Feb 2010 14:21:25 +0100
Subject: [Freeswitch-users] Looking for some good/cheap desktop phones
In-Reply-To: <d361ca7b1002050507n5e564621oa6cf7ace005dd5b8@mail.gmail.com>
References: <3f2e071b5aea5e11bb37a37e435a33b7@mail.gmail.com>
	<8e9d67561002042236s434e0643od322551de8720be1@mail.gmail.com>
	<409D0ACF-ADA4-4D02-8B7E-71EC1B1BAFBC@freeswitch.org>
	<8e9d67561002042301v76129abfj3dcbc70ed073752c@mail.gmail.com>
	<d361ca7b1002050507n5e564621oa6cf7ace005dd5b8@mail.gmail.com>
Message-ID: <65d96fc81002050521w746707b1q4398e6d8a0692d1e@mail.gmail.com>

Atcom AT-620 (
http://www.atcom.cn/AT620.html#~tab-small_midsized_large_enterprises<http://www.atcom.cn/AT620.html#%7Etab-small_midsized_large_enterprises>)
is quite ok and cheap (~30$)... also we have been talking to Atcom to add a
sort of auto-provissioning (dhcp/http) and this is going to happen next
week.


T.


2010/2/5 Yehavi Bourvine <yehavi.bourvine at gmail.com>

> From my experience Polycom and SNOM are expensive but give you what you
> need. Polycom is more intutive to the users but more cumbersome for the
> manager to deploy; SNOM is somewhat less intuitive to the user but
> everything can be set via the WEB interface.
>
> If you talk about 4-5 phones, then probably SNOM is the choice. It also
> depends about the specific functions you want to use. I our specific
> environment (high use of BLF and shared lines) Polycom wins because it
> handles these functions just as the user expects.
>
> I did not try Aastra so cannot testify. We did test Yealink, Thomson,
> Asterphone, SipTip and maybe others I forgot. Cisco also seems good but
> Cisco does not supply the required socumentation to make them fully working.
>
>                     Regards, __Yehavi:
>
> 2010/2/5 ????? ??????? <tim at novion.ru>
>
> Sure, those phones do not deliver superior usability, but they at
>> least give the best sound among budget models.
>>
>> 2010/2/5 Brian West <brian at freeswitch.org>:
>>  > And all of those are awful phones.  They don't even make good paper
>> weights.
>> >
>> > You can't have good and cheap in the same sentence when talking about
>> VoIP phones.  You have to take your pick between quality (good) and price
>> (cheap) you can't have both at once.
>> >
>> > /b
>> >
>> >
>> > On Feb 5, 2010, at 12:36 AM, ????? ??????? wrote:
>> >
>> >> Have a look at Yealink (Skypemate) and Fanvill
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100205/df4afa81/attachment-0001.html 

From Prometheus001 at gmx.net  Fri Feb  5 05:27:39 2010
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Fri, 05 Feb 2010 14:27:39 +0100
Subject: [Freeswitch-users] mod_skypiax - help needed with
 modified	snd-dummy (one way audio)
In-Reply-To: <7b197bef1001290520x37179cb1o63273410e9ef5bc@mail.gmail.com>
References: <4B60555B.2020004@gmx.net> <4B607944.4040700@gmx.net>
	<4B608014.4030902@gmx.net>	<7b197bef1001271215o64918f56s3d16e51528fc5f66@mail.gmail.com>
	<4B61ECE0.10409@gmx.net>	<7b197bef1001281210u43a907edi13afe794cf0e1a2e@mail.gmail.com>
	<4B61FCAB.5040707@gmx.net>	<7b197bef1001281341h7ee2c58fj79b3886630901f29@mail.gmail.com>
	<4B62C962.7000601@gmx.net>	<8213d6071001290403g2c08fe39gd92eb0446bb6485a@mail.gmail.com>
	<7b197bef1001290520x37179cb1o63273410e9ef5bc@mail.gmail.com>
Message-ID: <4B6C1CCB.4080606@gmx.net>

Hello Giovanni,

as I couldn't even get skype again working again with the standard alsa
driver, I would like to setup the machine from scratch based on a
working machine.
The latest errors I received from Skype was:
  snd_pcm_avail_update() returned a value that is exceptionally large:
715706624 bytes (3727638 ms).
  Most likely this is a bug in the ALSA driver. Please report this issue
to the ALSA developers.
I think that may be the reason for one-way-audio.

For setting up my machine from scratch, please advise:
- which OS you are you using und recommending exactly?
- I would like to use 64bit OS in order to use 8GB of memory, does this
work?
- any other hints?

Best regards
Peter

Giovanni Maruzzelli schrieb:
> Peter,
>
> Can you connect on IRC (irc.freenode.net #freeswitch)? I'm gmaruzz there.
>
> Can you restate your problems? I've lost connection :)
>
> with snd-dummy custom you can create *one only* snd-dummy instance, so
> *one only* fake soundcard. If you create more, will not work. But with
> that one fake soundcard you can use 64 skype client instances, all
> with the same soundcard hardware device (hw:n).
>
> with original snd-dummy you can create a max of 8 instances, so 8 fake
> soundcards, and with each fake soundcard you can  use a max of 8 skype
> client instances.
>
> use the hardware devices, not the default devices (use the "hw:n")
>
> -giovanni
>
> On Fri, Jan 29, 2010 at 1:03 PM, Ghulam Mustafa <mustafa.pk at gmail.com> wrote:
>   
>> did you enable debug mode while compiling custom snd-dummy? if  yes
>> try re-compiling with debug mode disabled.
>>
>> -m
>>
>> On Fri, Jan 29, 2010 at 4:41 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>     
>>> I now reinstalled the original sound drivers
>>> Unfortunaltely the sound problems remain, not that worse but they are there:
>>> Audio is still (almost) one way. Almost means:
>>>
>>>    * SIP -> Skype ok
>>>    * Skype=> SIP I hear only some scratching on very loud audio
>>>
>>> Could it be a volume problem? But snd-dummy should have no volume
>>> properties, right?
>>>
>>> Best regards
>>> Peter
>>>
>>>
>>> Giovanni Maruzzelli schrieb:
>>>       
>>>> with three instances you will assign the hw:0 device to skype client
>>>> 0...7, hw:1 to skype client 8...15 and hw:2 to skype client 16...23.
>>>> Must work. Pay attention to assign the same device name to all devices
>>>> needed by a skype instance (sound devices window): playback, capture
>>>> AND ring.
>>>>
>>>> Or maybe is a bug of ALSA on Debian...
>>>>
>>>> -giovanni
>>>>
>>>> On Thu, Jan 28, 2010 at 10:07 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>
>>>>         
>>>>> I crated 3 instances of snd-dummy, this worked. I assigned then Instance
>>>>> #2 to the Skype accounts. Still no sound.
>>>>> On the frist call there is one way audio, on the following calls there
>>>>> is no audio at all.
>>>>> This is weird.
>>>>>
>>>>> Best regards
>>>>> Peter
>>>>>
>>>>> Giovanni Maruzzelli schrieb:
>>>>>
>>>>>           
>>>>>> Ciao Peter,
>>>>>>
>>>>>> Never tested on Debian 5.
>>>>>>
>>>>>> When you write "same problem" you are referring to the audio going one
>>>>>> way only (btw, which way?) with the custom audio driver?
>>>>>>
>>>>>> Have you tried with multiple instances of the regular Debian
>>>>>> snd-dummy, as I wrote in a mail before?
>>>>>>
>>>>>> -gm
>>>>>>
>>>>>>
>>>>>>
>>>>>> On Thu, Jan 28, 2010 at 9:00 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>>
>>>>>>
>>>>>>             
>>>>>>> Hello Giovanni,
>>>>>>>
>>>>>>> I did so but the same problem again.
>>>>>>>
>>>>>>> Did you ever test in on Debian 5.0?
>>>>>>>
>>>>>>> Best reards
>>>>>>> Peter
>>>>>>>
>>>>>>> Giovanni Maruzzelli schrieb:
>>>>>>>
>>>>>>>
>>>>>>>               
>>>>>>>> good, so you have only one sound device, the right one.
>>>>>>>>
>>>>>>>> Use the one with hw:0 in the window that skype gives you to set sound devices
>>>>>>>>
>>>>>>>> -gm
>>>>>>>>
>>>>>>>> On Wed, Jan 27, 2010 at 7:04 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>                 
>>>>>>>>> I installed alsa-utile,
>>>>>>>>>
>>>>>>>>> now I get:
>>>>>>>>>
>>>>>>>>> skype:/var/cache/apt/archives# aplay -l
>>>>>>>>> **** List of PLAYBACK Hardware Devices ****
>>>>>>>>> card 0: Dummy [Dummy], device 0: Dummy PCM [Dummy PCM]
>>>>>>>>>  Subdevices: 127/128
>>>>>>>>>  Subdevice #0: subdevice #0
>>>>>>>>>  Subdevice #1: subdevice #1
>>>>>>>>>  Subdevice #2: subdevice #2
>>>>>>>>>  Subdevice #3: subdevice #3
>>>>>>>>>  Subdevice #4: subdevice #4
>>>>>>>>>  Subdevice #5: subdevice #5
>>>>>>>>>  Subdevice #6: subdevice #6
>>>>>>>>>  Subdevice #7: subdevice #7
>>>>>>>>>  Subdevice #8: subdevice #8
>>>>>>>>>  Subdevice #9: subdevice #9
>>>>>>>>>  Subdevice #10: subdevice #10
>>>>>>>>>  Subdevice #11: subdevice #11
>>>>>>>>>  Subdevice #12: subdevice #12
>>>>>>>>>  Subdevice #13: subdevice #13
>>>>>>>>>  Subdevice #14: subdevice #14
>>>>>>>>>  Subdevice #15: subdevice #15
>>>>>>>>>  Subdevice #16: subdevice #16
>>>>>>>>>  Subdevice #17: subdevice #17
>>>>>>>>>  Subdevice #18: subdevice #18
>>>>>>>>>  Subdevice #19: subdevice #19
>>>>>>>>>  Subdevice #20: subdevice #20
>>>>>>>>>  Subdevice #21: subdevice #21
>>>>>>>>>  Subdevice #22: subdevice #22
>>>>>>>>>  Subdevice #23: subdevice #23
>>>>>>>>>  Subdevice #24: subdevice #24
>>>>>>>>>  Subdevice #25: subdevice #25
>>>>>>>>>  Subdevice #26: subdevice #26
>>>>>>>>>  Subdevice #27: subdevice #27
>>>>>>>>>  Subdevice #28: subdevice #28
>>>>>>>>>  Subdevice #29: subdevice #29
>>>>>>>>>  Subdevice #30: subdevice #30
>>>>>>>>>  Subdevice #31: subdevice #31
>>>>>>>>>  Subdevice #32: subdevice #32
>>>>>>>>>  Subdevice #33: subdevice #33
>>>>>>>>>  Subdevice #34: subdevice #34
>>>>>>>>>  Subdevice #35: subdevice #35
>>>>>>>>>  Subdevice #36: subdevice #36
>>>>>>>>>  Subdevice #37: subdevice #37
>>>>>>>>>  Subdevice #38: subdevice #38
>>>>>>>>>  Subdevice #39: subdevice #39
>>>>>>>>>  Subdevice #40: subdevice #40
>>>>>>>>>  Subdevice #41: subdevice #41
>>>>>>>>>  Subdevice #42: subdevice #42
>>>>>>>>>  Subdevice #43: subdevice #43
>>>>>>>>>  Subdevice #44: subdevice #44
>>>>>>>>>  Subdevice #45: subdevice #45
>>>>>>>>>  Subdevice #46: subdevice #46
>>>>>>>>>  Subdevice #47: subdevice #47
>>>>>>>>>  Subdevice #48: subdevice #48
>>>>>>>>>  Subdevice #49: subdevice #49
>>>>>>>>>  Subdevice #50: subdevice #50
>>>>>>>>>  Subdevice #51: subdevice #51
>>>>>>>>>  Subdevice #52: subdevice #52
>>>>>>>>>  Subdevice #53: subdevice #53
>>>>>>>>>  Subdevice #54: subdevice #54
>>>>>>>>>  Subdevice #55: subdevice #55
>>>>>>>>>  Subdevice #56: subdevice #56
>>>>>>>>>  Subdevice #57: subdevice #57
>>>>>>>>>  Subdevice #58: subdevice #58
>>>>>>>>>  Subdevice #59: subdevice #59
>>>>>>>>>  Subdevice #60: subdevice #60
>>>>>>>>>  Subdevice #61: subdevice #61
>>>>>>>>>  Subdevice #62: subdevice #62
>>>>>>>>>  Subdevice #63: subdevice #63
>>>>>>>>>  Subdevice #64: subdevice #64
>>>>>>>>>  Subdevice #65: subdevice #65
>>>>>>>>>  Subdevice #66: subdevice #66
>>>>>>>>>  Subdevice #67: subdevice #67
>>>>>>>>>  Subdevice #68: subdevice #68
>>>>>>>>>  Subdevice #69: subdevice #69
>>>>>>>>>  Subdevice #70: subdevice #70
>>>>>>>>>  Subdevice #71: subdevice #71
>>>>>>>>>  Subdevice #72: subdevice #72
>>>>>>>>>  Subdevice #73: subdevice #73
>>>>>>>>>  Subdevice #74: subdevice #74
>>>>>>>>>  Subdevice #75: subdevice #75
>>>>>>>>>  Subdevice #76: subdevice #76
>>>>>>>>>  Subdevice #77: subdevice #77
>>>>>>>>>  Subdevice #78: subdevice #78
>>>>>>>>>  Subdevice #79: subdevice #79
>>>>>>>>>  Subdevice #80: subdevice #80
>>>>>>>>>  Subdevice #81: subdevice #81
>>>>>>>>>  Subdevice #82: subdevice #82
>>>>>>>>>  Subdevice #83: subdevice #83
>>>>>>>>>  Subdevice #84: subdevice #84
>>>>>>>>>  Subdevice #85: subdevice #85
>>>>>>>>>  Subdevice #86: subdevice #86
>>>>>>>>>  Subdevice #87: subdevice #87
>>>>>>>>>  Subdevice #88: subdevice #88
>>>>>>>>>  Subdevice #89: subdevice #89
>>>>>>>>>  Subdevice #90: subdevice #90
>>>>>>>>>  Subdevice #91: subdevice #91
>>>>>>>>>  Subdevice #92: subdevice #92
>>>>>>>>>  Subdevice #93: subdevice #93
>>>>>>>>>  Subdevice #94: subdevice #94
>>>>>>>>>  Subdevice #95: subdevice #95
>>>>>>>>>  Subdevice #96: subdevice #96
>>>>>>>>>  Subdevice #97: subdevice #97
>>>>>>>>>  Subdevice #98: subdevice #98
>>>>>>>>>  Subdevice #99: subdevice #99
>>>>>>>>>  Subdevice #100: subdevice #100
>>>>>>>>>  Subdevice #101: subdevice #101
>>>>>>>>>  Subdevice #102: subdevice #102
>>>>>>>>>  Subdevice #103: subdevice #103
>>>>>>>>>  Subdevice #104: subdevice #104
>>>>>>>>>  Subdevice #105: subdevice #105
>>>>>>>>>  Subdevice #106: subdevice #106
>>>>>>>>>  Subdevice #107: subdevice #107
>>>>>>>>>  Subdevice #108: subdevice #108
>>>>>>>>>  Subdevice #109: subdevice #109
>>>>>>>>>  Subdevice #110: subdevice #110
>>>>>>>>>  Subdevice #111: subdevice #111
>>>>>>>>>  Subdevice #112: subdevice #112
>>>>>>>>>  Subdevice #113: subdevice #113
>>>>>>>>>  Subdevice #114: subdevice #114
>>>>>>>>>  Subdevice #115: subdevice #115
>>>>>>>>>  Subdevice #116: subdevice #116
>>>>>>>>>  Subdevice #117: subdevice #117
>>>>>>>>>  Subdevice #118: subdevice #118
>>>>>>>>>  Subdevice #119: subdevice #119
>>>>>>>>>  Subdevice #120: subdevice #120
>>>>>>>>>  Subdevice #121: subdevice #121
>>>>>>>>>  Subdevice #122: subdevice #122
>>>>>>>>>  Subdevice #123: subdevice #123
>>>>>>>>>  Subdevice #124: subdevice #124
>>>>>>>>>  Subdevice #125: subdevice #125
>>>>>>>>>  Subdevice #126: subdevice #126
>>>>>>>>>  Subdevice #127: subdevice #127
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> Peter P GMX schrieb:
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>                   
>>>>>>>>>> Her's the output:
>>>>>>>>>>
>>>>>>>>>> skype:~# aplay -l
>>>>>>>>>> bash: aplay: command not found
>>>>>>>>>>
>>>>>>>>>> Giovanni Maruzzelli schrieb:
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>                     
>>>>>>>>>>> I don't think you got two snd-dummy loaded (but maybe yes)
>>>>>>>>>>> what's the output of:
>>>>>>>>>>>
>>>>>>>>>>> aplay -l
>>>>>>>>>>>
>>>>>>>>>>> ?
>>>>>>>>>>>
>>>>>>>>>>> If instead you are referring to the choices that skype clients offers
>>>>>>>>>>> you in the "set audio devices" window, choose Dummy PCM (hw0:0)
>>>>>>>>>>>
>>>>>>>>>>> Eg: not the "default", but the "hardware" one
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> On Wed, Jan 27, 2010 at 5:58 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>                       
>>>>>>>>>>>> Thanks Giovanni,
>>>>>>>>>>>>
>>>>>>>>>>>> I think there may be the problem, that I have 2 sound devices now:
>>>>>>>>>>>> - Dummy PCM (hw0:0) (this is from debian install)
>>>>>>>>>>>> - Dummy PCM Default Audio device (defauzlt: CARD=Dummy) (this is new
>>>>>>>>>>>> since I compiled alsa newly)
>>>>>>>>>>>>
>>>>>>>>>>>> I tried both, but both do not work. How do I get rid of the old alsa device?
>>>>>>>>>>>> By the way: I uninstalled Alsa before I installed the new driver
>>>>>>>>>>>> (apt-get remove alsa-utils alsa-base).
>>>>>>>>>>>>
>>>>>>>>>>>> Best regards
>>>>>>>>>>>> Peter
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>> Giovanni Maruzzelli schrieb:
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>                         
>>>>>>>>>>>>> This warning is harmless:
>>>>>>>>>>>>>
>>>>>>>>>>>>> Jan 27 15:28:50 skype kernel: [ 3994.795786] process `skype' is using
>>>>>>>>>>>>> obsolete setsockopt SO_BSDCOMPAT
>>>>>>>>>>>>>
>>>>>>>>>>>>> On Wed, Jan 27, 2010 at 4:26 PM, Giovanni Maruzzelli
>>>>>>>>>>>>> <gmaruzz at celliax.org> wrote:
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>                           
>>>>>>>>>>>>>> Ciao Peter
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> one instance of snd-dummy "customized" is enough for 64 instances of
>>>>>>>>>>>>>> skype clients, no need (and do not works) with more instances of
>>>>>>>>>>>>>> snd-dummy-customized.
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> Maybe you got the one-way problem because of kernel at 250HZ (don't
>>>>>>>>>>>>>> know). It uses to works well on a tickless kernel at 100HZ (eg: ubuntu
>>>>>>>>>>>>>> 8.04).
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> Or maybe you have to check and modify which sound devices the skype
>>>>>>>>>>>>>> clients are using (try to check that with snd-summy-custom loaded,
>>>>>>>>>>>>>> maybe with the ssh -X trick (as in the wiki page).
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> To load more than one snd-dummy-original (the non modified one), you
>>>>>>>>>>>>>> do this with the modprobe command, as in:
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> rmmod snd-dummy
>>>>>>>>>>>>>> modprobe snd-dummy enable=1,1,1
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> this command will enable three instances of snd-dummy original, so
>>>>>>>>>>>>>> you'll have three fake soundcards, and you'll have to setup each group
>>>>>>>>>>>>>> of 8 skype instances to use sound devices from one fake soundcard, RG:
>>>>>>>>>>>>>> no more than 8 skype client instances can use one instance of fake
>>>>>>>>>>>>>> soundcard.
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> Also, please update the mod_skypiax code (svn up in its directory) I
>>>>>>>>>>>>>> just committed some improvements.
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> If you have any other doubts, or need more info, don't hesitate to
>>>>>>>>>>>>>> write the mailing list again,
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> ciao for now,
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> -giovanni
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> On Wed, Jan 27, 2010 at 4:01 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>                             
>>>>>>>>>>>>>>> I have mod_skypiax working nicely so far with 2 Skype channels. Thanks
>>>>>>>>>>>>>>> to all contributors, excellent work!
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> In order to have more than 8 channels working, I have followed the
>>>>>>>>>>>>>>> instructions in
>>>>>>>>>>>>>>> http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk
>>>>>>>>>>>>>>> and compiled alsa-driver-1.0.20 with the modified dummy.c file. (System
>>>>>>>>>>>>>>> ist Debian 5.0R3)
>>>>>>>>>>>>>>> It compiled well however when I start snd-dummy I only have
>>>>>>>>>>>>>>> one-way-audio and my logs show
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> Jan 27 15:28:41 skype kernel: [ 3984.318403] snd-dummy skypiax driver,
>>>>>>>>>>>>>>> /usr/src/alsa-driver-1.0.20/drivers/../alsa-kernel/drivers/dummy.c:920
>>>>>>>>>>>>>>> working on a machine with 250HZ kernel
>>>>>>>>>>>>>>> Jan 27 15:28:50 skype kernel: [ 3994.795786] process `skype' is using
>>>>>>>>>>>>>>> obsolete setsockopt SO_BSDCOMPAT
>>>>>>>>>>>>>>> Jan 27 15:28:56 skype kernel: [ 4005.289907] __ratelimit: 490 messages
>>>>>>>>>>>>>>> suppressed
>>>>>>>>>>>>>>> Jan 27 15:29:01 skype kernel: [ 4012.458310] process `skype' is using
>>>>>>>>>>>>>>> obsolete setsockopt SO_BSDCOMPAT
>>>>>>>>>>>>>>> Jan 27 15:29:01 skype kernel: [ 4013.326290] __ratelimit: 499 messages
>>>>>>>>>>>>>>> suppressed
>>>>>>>>>>>>>>> If I reinstall alsa from deb everything sworks fine again (of course
>>>>>>>>>>>>>>> with the current limitations).
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> First question: Has anybody had this issue before? How can I solve this?
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> Second question:
>>>>>>>>>>>>>>> As I do not need 64 channels or more: how do I manage, that Skype
>>>>>>>>>>>>>>> instances 9..15 use a second instance of snd-dummy as addressed in the wiki?
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> Best regards
>>>>>>>>>>>>>>> Peter
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>                               
>>>>>>>>>>>>>> --
>>>>>>>>>>>>>> Sincerely,
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> Giovanni Maruzzelli
>>>>>>>>>>>>>> Cell : +39-347-2665618
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>                             
>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>                         
>>>>>>>>> _______________________________________________
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>                   
>>>>>>>>                 
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>               
>>>>>>
>>>>>>             
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>>           
>>>>
>>>>
>>>>         
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>       
>>
>> --
>> Ghulam Mustafa
>> cell: +92 333.611.7681
>> sip: cyrenity at ekiga.net
>> mail: mustafa.pk at gmail.com
>> web: cyrenity.wordpress.com
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>     
>
>
>
>   


From gmaruzz at celliax.org  Fri Feb  5 05:36:28 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Fri, 5 Feb 2010 14:36:28 +0100
Subject: [Freeswitch-users] mod_skypiax - help needed with modified
	snd-dummy (one way audio)
In-Reply-To: <4B6C1CCB.4080606@gmx.net>
References: <4B60555B.2020004@gmx.net>
	<7b197bef1001271215o64918f56s3d16e51528fc5f66@mail.gmail.com> 
	<4B61ECE0.10409@gmx.net>
	<7b197bef1001281210u43a907edi13afe794cf0e1a2e@mail.gmail.com> 
	<4B61FCAB.5040707@gmx.net>
	<7b197bef1001281341h7ee2c58fj79b3886630901f29@mail.gmail.com> 
	<4B62C962.7000601@gmx.net>
	<8213d6071001290403g2c08fe39gd92eb0446bb6485a@mail.gmail.com> 
	<7b197bef1001290520x37179cb1o63273410e9ef5bc@mail.gmail.com> 
	<4B6C1CCB.4080606@gmx.net>
Message-ID: <7b197bef1002050536j7ea09684v913434296ae52b97@mail.gmail.com>

Ciao Peter,

I would use ubuntu 8.04 (hardy) LTS server 64bit or CentOS 5.4 64bit.

-giovanni

On Fri, Feb 5, 2010 at 2:27 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
> Hello Giovanni,
>
> as I couldn't even get skype again working again with the standard alsa
> driver, I would like to setup the machine from scratch based on a
> working machine.
> The latest errors I received from Skype was:
> ?snd_pcm_avail_update() returned a value that is exceptionally large:
> 715706624 bytes (3727638 ms).
> ?Most likely this is a bug in the ALSA driver. Please report this issue
> to the ALSA developers.
> I think that may be the reason for one-way-audio.
>
> For setting up my machine from scratch, please advise:
> - which OS you are you using und recommending exactly?
> - I would like to use 64bit OS in order to use 8GB of memory, does this
> work?
> - any other hints?
>
> Best regards
> Peter
>
> Giovanni Maruzzelli schrieb:
>> Peter,
>>
>> Can you connect on IRC (irc.freenode.net #freeswitch)? I'm gmaruzz there.
>>
>> Can you restate your problems? I've lost connection :)
>>
>> with snd-dummy custom you can create *one only* snd-dummy instance, so
>> *one only* fake soundcard. If you create more, will not work. But with
>> that one fake soundcard you can use 64 skype client instances, all
>> with the same soundcard hardware device (hw:n).
>>
>> with original snd-dummy you can create a max of 8 instances, so 8 fake
>> soundcards, and with each fake soundcard you can ?use a max of 8 skype
>> client instances.
>>
>> use the hardware devices, not the default devices (use the "hw:n")
>>
>> -giovanni
>>
>> On Fri, Jan 29, 2010 at 1:03 PM, Ghulam Mustafa <mustafa.pk at gmail.com> wrote:
>>
>>> did you enable debug mode while compiling custom snd-dummy? if ?yes
>>> try re-compiling with debug mode disabled.
>>>
>>> -m
>>>
>>> On Fri, Jan 29, 2010 at 4:41 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>
>>>> I now reinstalled the original sound drivers
>>>> Unfortunaltely the sound problems remain, not that worse but they are there:
>>>> Audio is still (almost) one way. Almost means:
>>>>
>>>> ? ?* SIP -> Skype ok
>>>> ? ?* Skype=> SIP I hear only some scratching on very loud audio
>>>>
>>>> Could it be a volume problem? But snd-dummy should have no volume
>>>> properties, right?
>>>>
>>>> Best regards
>>>> Peter
>>>>
>>>>
>>>> Giovanni Maruzzelli schrieb:
>>>>
>>>>> with three instances you will assign the hw:0 device to skype client
>>>>> 0...7, hw:1 to skype client 8...15 and hw:2 to skype client 16...23.
>>>>> Must work. Pay attention to assign the same device name to all devices
>>>>> needed by a skype instance (sound devices window): playback, capture
>>>>> AND ring.
>>>>>
>>>>> Or maybe is a bug of ALSA on Debian...
>>>>>
>>>>> -giovanni
>>>>>
>>>>> On Thu, Jan 28, 2010 at 10:07 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>
>>>>>
>>>>>> I crated 3 instances of snd-dummy, this worked. I assigned then Instance
>>>>>> #2 to the Skype accounts. Still no sound.
>>>>>> On the frist call there is one way audio, on the following calls there
>>>>>> is no audio at all.
>>>>>> This is weird.
>>>>>>
>>>>>> Best regards
>>>>>> Peter
>>>>>>
>>>>>> Giovanni Maruzzelli schrieb:
>>>>>>
>>>>>>
>>>>>>> Ciao Peter,
>>>>>>>
>>>>>>> Never tested on Debian 5.
>>>>>>>
>>>>>>> When you write "same problem" you are referring to the audio going one
>>>>>>> way only (btw, which way?) with the custom audio driver?
>>>>>>>
>>>>>>> Have you tried with multiple instances of the regular Debian
>>>>>>> snd-dummy, as I wrote in a mail before?
>>>>>>>
>>>>>>> -gm
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> On Thu, Jan 28, 2010 at 9:00 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>> Hello Giovanni,
>>>>>>>>
>>>>>>>> I did so but the same problem again.
>>>>>>>>
>>>>>>>> Did you ever test in on Debian 5.0?
>>>>>>>>
>>>>>>>> Best reards
>>>>>>>> Peter
>>>>>>>>
>>>>>>>> Giovanni Maruzzelli schrieb:
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>> good, so you have only one sound device, the right one.
>>>>>>>>>
>>>>>>>>> Use the one with hw:0 in the window that skype gives you to set sound devices
>>>>>>>>>
>>>>>>>>> -gm
>>>>>>>>>
>>>>>>>>> On Wed, Jan 27, 2010 at 7:04 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>> I installed alsa-utile,
>>>>>>>>>>
>>>>>>>>>> now I get:
>>>>>>>>>>
>>>>>>>>>> skype:/var/cache/apt/archives# aplay -l
>>>>>>>>>> **** List of PLAYBACK Hardware Devices ****
>>>>>>>>>> card 0: Dummy [Dummy], device 0: Dummy PCM [Dummy PCM]
>>>>>>>>>> ?Subdevices: 127/128
>>>>>>>>>> ?Subdevice #0: subdevice #0
>>>>>>>>>> ?Subdevice #1: subdevice #1
>>>>>>>>>> ?Subdevice #2: subdevice #2
>>>>>>>>>> ?Subdevice #3: subdevice #3
>>>>>>>>>> ?Subdevice #4: subdevice #4
>>>>>>>>>> ?Subdevice #5: subdevice #5
>>>>>>>>>> ?Subdevice #6: subdevice #6
>>>>>>>>>> ?Subdevice #7: subdevice #7
>>>>>>>>>> ?Subdevice #8: subdevice #8
>>>>>>>>>> ?Subdevice #9: subdevice #9
>>>>>>>>>> ?Subdevice #10: subdevice #10
>>>>>>>>>> ?Subdevice #11: subdevice #11
>>>>>>>>>> ?Subdevice #12: subdevice #12
>>>>>>>>>> ?Subdevice #13: subdevice #13
>>>>>>>>>> ?Subdevice #14: subdevice #14
>>>>>>>>>> ?Subdevice #15: subdevice #15
>>>>>>>>>> ?Subdevice #16: subdevice #16
>>>>>>>>>> ?Subdevice #17: subdevice #17
>>>>>>>>>> ?Subdevice #18: subdevice #18
>>>>>>>>>> ?Subdevice #19: subdevice #19
>>>>>>>>>> ?Subdevice #20: subdevice #20
>>>>>>>>>> ?Subdevice #21: subdevice #21
>>>>>>>>>> ?Subdevice #22: subdevice #22
>>>>>>>>>> ?Subdevice #23: subdevice #23
>>>>>>>>>> ?Subdevice #24: subdevice #24
>>>>>>>>>> ?Subdevice #25: subdevice #25
>>>>>>>>>> ?Subdevice #26: subdevice #26
>>>>>>>>>> ?Subdevice #27: subdevice #27
>>>>>>>>>> ?Subdevice #28: subdevice #28
>>>>>>>>>> ?Subdevice #29: subdevice #29
>>>>>>>>>> ?Subdevice #30: subdevice #30
>>>>>>>>>> ?Subdevice #31: subdevice #31
>>>>>>>>>> ?Subdevice #32: subdevice #32
>>>>>>>>>> ?Subdevice #33: subdevice #33
>>>>>>>>>> ?Subdevice #34: subdevice #34
>>>>>>>>>> ?Subdevice #35: subdevice #35
>>>>>>>>>> ?Subdevice #36: subdevice #36
>>>>>>>>>> ?Subdevice #37: subdevice #37
>>>>>>>>>> ?Subdevice #38: subdevice #38
>>>>>>>>>> ?Subdevice #39: subdevice #39
>>>>>>>>>> ?Subdevice #40: subdevice #40
>>>>>>>>>> ?Subdevice #41: subdevice #41
>>>>>>>>>> ?Subdevice #42: subdevice #42
>>>>>>>>>> ?Subdevice #43: subdevice #43
>>>>>>>>>> ?Subdevice #44: subdevice #44
>>>>>>>>>> ?Subdevice #45: subdevice #45
>>>>>>>>>> ?Subdevice #46: subdevice #46
>>>>>>>>>> ?Subdevice #47: subdevice #47
>>>>>>>>>> ?Subdevice #48: subdevice #48
>>>>>>>>>> ?Subdevice #49: subdevice #49
>>>>>>>>>> ?Subdevice #50: subdevice #50
>>>>>>>>>> ?Subdevice #51: subdevice #51
>>>>>>>>>> ?Subdevice #52: subdevice #52
>>>>>>>>>> ?Subdevice #53: subdevice #53
>>>>>>>>>> ?Subdevice #54: subdevice #54
>>>>>>>>>> ?Subdevice #55: subdevice #55
>>>>>>>>>> ?Subdevice #56: subdevice #56
>>>>>>>>>> ?Subdevice #57: subdevice #57
>>>>>>>>>> ?Subdevice #58: subdevice #58
>>>>>>>>>> ?Subdevice #59: subdevice #59
>>>>>>>>>> ?Subdevice #60: subdevice #60
>>>>>>>>>> ?Subdevice #61: subdevice #61
>>>>>>>>>> ?Subdevice #62: subdevice #62
>>>>>>>>>> ?Subdevice #63: subdevice #63
>>>>>>>>>> ?Subdevice #64: subdevice #64
>>>>>>>>>> ?Subdevice #65: subdevice #65
>>>>>>>>>> ?Subdevice #66: subdevice #66
>>>>>>>>>> ?Subdevice #67: subdevice #67
>>>>>>>>>> ?Subdevice #68: subdevice #68
>>>>>>>>>> ?Subdevice #69: subdevice #69
>>>>>>>>>> ?Subdevice #70: subdevice #70
>>>>>>>>>> ?Subdevice #71: subdevice #71
>>>>>>>>>> ?Subdevice #72: subdevice #72
>>>>>>>>>> ?Subdevice #73: subdevice #73
>>>>>>>>>> ?Subdevice #74: subdevice #74
>>>>>>>>>> ?Subdevice #75: subdevice #75
>>>>>>>>>> ?Subdevice #76: subdevice #76
>>>>>>>>>> ?Subdevice #77: subdevice #77
>>>>>>>>>> ?Subdevice #78: subdevice #78
>>>>>>>>>> ?Subdevice #79: subdevice #79
>>>>>>>>>> ?Subdevice #80: subdevice #80
>>>>>>>>>> ?Subdevice #81: subdevice #81
>>>>>>>>>> ?Subdevice #82: subdevice #82
>>>>>>>>>> ?Subdevice #83: subdevice #83
>>>>>>>>>> ?Subdevice #84: subdevice #84
>>>>>>>>>> ?Subdevice #85: subdevice #85
>>>>>>>>>> ?Subdevice #86: subdevice #86
>>>>>>>>>> ?Subdevice #87: subdevice #87
>>>>>>>>>> ?Subdevice #88: subdevice #88
>>>>>>>>>> ?Subdevice #89: subdevice #89
>>>>>>>>>> ?Subdevice #90: subdevice #90
>>>>>>>>>> ?Subdevice #91: subdevice #91
>>>>>>>>>> ?Subdevice #92: subdevice #92
>>>>>>>>>> ?Subdevice #93: subdevice #93
>>>>>>>>>> ?Subdevice #94: subdevice #94
>>>>>>>>>> ?Subdevice #95: subdevice #95
>>>>>>>>>> ?Subdevice #96: subdevice #96
>>>>>>>>>> ?Subdevice #97: subdevice #97
>>>>>>>>>> ?Subdevice #98: subdevice #98
>>>>>>>>>> ?Subdevice #99: subdevice #99
>>>>>>>>>> ?Subdevice #100: subdevice #100
>>>>>>>>>> ?Subdevice #101: subdevice #101
>>>>>>>>>> ?Subdevice #102: subdevice #102
>>>>>>>>>> ?Subdevice #103: subdevice #103
>>>>>>>>>> ?Subdevice #104: subdevice #104
>>>>>>>>>> ?Subdevice #105: subdevice #105
>>>>>>>>>> ?Subdevice #106: subdevice #106
>>>>>>>>>> ?Subdevice #107: subdevice #107
>>>>>>>>>> ?Subdevice #108: subdevice #108
>>>>>>>>>> ?Subdevice #109: subdevice #109
>>>>>>>>>> ?Subdevice #110: subdevice #110
>>>>>>>>>> ?Subdevice #111: subdevice #111
>>>>>>>>>> ?Subdevice #112: subdevice #112
>>>>>>>>>> ?Subdevice #113: subdevice #113
>>>>>>>>>> ?Subdevice #114: subdevice #114
>>>>>>>>>> ?Subdevice #115: subdevice #115
>>>>>>>>>> ?Subdevice #116: subdevice #116
>>>>>>>>>> ?Subdevice #117: subdevice #117
>>>>>>>>>> ?Subdevice #118: subdevice #118
>>>>>>>>>> ?Subdevice #119: subdevice #119
>>>>>>>>>> ?Subdevice #120: subdevice #120
>>>>>>>>>> ?Subdevice #121: subdevice #121
>>>>>>>>>> ?Subdevice #122: subdevice #122
>>>>>>>>>> ?Subdevice #123: subdevice #123
>>>>>>>>>> ?Subdevice #124: subdevice #124
>>>>>>>>>> ?Subdevice #125: subdevice #125
>>>>>>>>>> ?Subdevice #126: subdevice #126
>>>>>>>>>> ?Subdevice #127: subdevice #127
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> Peter P GMX schrieb:
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>> Her's the output:
>>>>>>>>>>>
>>>>>>>>>>> skype:~# aplay -l
>>>>>>>>>>> bash: aplay: command not found
>>>>>>>>>>>
>>>>>>>>>>> Giovanni Maruzzelli schrieb:
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>> I don't think you got two snd-dummy loaded (but maybe yes)
>>>>>>>>>>>> what's the output of:
>>>>>>>>>>>>
>>>>>>>>>>>> aplay -l
>>>>>>>>>>>>
>>>>>>>>>>>> ?
>>>>>>>>>>>>
>>>>>>>>>>>> If instead you are referring to the choices that skype clients offers
>>>>>>>>>>>> you in the "set audio devices" window, choose Dummy PCM (hw0:0)
>>>>>>>>>>>>
>>>>>>>>>>>> Eg: not the "default", but the "hardware" one
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>> On Wed, Jan 27, 2010 at 5:58 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>> Thanks Giovanni,
>>>>>>>>>>>>>
>>>>>>>>>>>>> I think there may be the problem, that I have 2 sound devices now:
>>>>>>>>>>>>> - Dummy PCM (hw0:0) (this is from debian install)
>>>>>>>>>>>>> - Dummy PCM Default Audio device (defauzlt: CARD=Dummy) (this is new
>>>>>>>>>>>>> since I compiled alsa newly)
>>>>>>>>>>>>>
>>>>>>>>>>>>> I tried both, but both do not work. How do I get rid of the old alsa device?
>>>>>>>>>>>>> By the way: I uninstalled Alsa before I installed the new driver
>>>>>>>>>>>>> (apt-get remove alsa-utils alsa-base).
>>>>>>>>>>>>>
>>>>>>>>>>>>> Best regards
>>>>>>>>>>>>> Peter
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>> Giovanni Maruzzelli schrieb:
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>> This warning is harmless:
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> Jan 27 15:28:50 skype kernel: [ 3994.795786] process `skype' is using
>>>>>>>>>>>>>> obsolete setsockopt SO_BSDCOMPAT
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> On Wed, Jan 27, 2010 at 4:26 PM, Giovanni Maruzzelli
>>>>>>>>>>>>>> <gmaruzz at celliax.org> wrote:
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> Ciao Peter
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> one instance of snd-dummy "customized" is enough for 64 instances of
>>>>>>>>>>>>>>> skype clients, no need (and do not works) with more instances of
>>>>>>>>>>>>>>> snd-dummy-customized.
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> Maybe you got the one-way problem because of kernel at 250HZ (don't
>>>>>>>>>>>>>>> know). It uses to works well on a tickless kernel at 100HZ (eg: ubuntu
>>>>>>>>>>>>>>> 8.04).
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> Or maybe you have to check and modify which sound devices the skype
>>>>>>>>>>>>>>> clients are using (try to check that with snd-summy-custom loaded,
>>>>>>>>>>>>>>> maybe with the ssh -X trick (as in the wiki page).
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> To load more than one snd-dummy-original (the non modified one), you
>>>>>>>>>>>>>>> do this with the modprobe command, as in:
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> rmmod snd-dummy
>>>>>>>>>>>>>>> modprobe snd-dummy enable=1,1,1
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> this command will enable three instances of snd-dummy original, so
>>>>>>>>>>>>>>> you'll have three fake soundcards, and you'll have to setup each group
>>>>>>>>>>>>>>> of 8 skype instances to use sound devices from one fake soundcard, RG:
>>>>>>>>>>>>>>> no more than 8 skype client instances can use one instance of fake
>>>>>>>>>>>>>>> soundcard.
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> Also, please update the mod_skypiax code (svn up in its directory) I
>>>>>>>>>>>>>>> just committed some improvements.
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> If you have any other doubts, or need more info, don't hesitate to
>>>>>>>>>>>>>>> write the mailing list again,
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> ciao for now,
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> -giovanni
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> On Wed, Jan 27, 2010 at 4:01 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> I have mod_skypiax working nicely so far with 2 Skype channels. Thanks
>>>>>>>>>>>>>>>> to all contributors, excellent work!
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> In order to have more than 8 channels working, I have followed the
>>>>>>>>>>>>>>>> instructions in
>>>>>>>>>>>>>>>> http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk
>>>>>>>>>>>>>>>> and compiled alsa-driver-1.0.20 with the modified dummy.c file. (System
>>>>>>>>>>>>>>>> ist Debian 5.0R3)
>>>>>>>>>>>>>>>> It compiled well however when I start snd-dummy I only have
>>>>>>>>>>>>>>>> one-way-audio and my logs show
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> Jan 27 15:28:41 skype kernel: [ 3984.318403] snd-dummy skypiax driver,
>>>>>>>>>>>>>>>> /usr/src/alsa-driver-1.0.20/drivers/../alsa-kernel/drivers/dummy.c:920
>>>>>>>>>>>>>>>> working on a machine with 250HZ kernel
>>>>>>>>>>>>>>>> Jan 27 15:28:50 skype kernel: [ 3994.795786] process `skype' is using
>>>>>>>>>>>>>>>> obsolete setsockopt SO_BSDCOMPAT
>>>>>>>>>>>>>>>> Jan 27 15:28:56 skype kernel: [ 4005.289907] __ratelimit: 490 messages
>>>>>>>>>>>>>>>> suppressed
>>>>>>>>>>>>>>>> Jan 27 15:29:01 skype kernel: [ 4012.458310] process `skype' is using
>>>>>>>>>>>>>>>> obsolete setsockopt SO_BSDCOMPAT
>>>>>>>>>>>>>>>> Jan 27 15:29:01 skype kernel: [ 4013.326290] __ratelimit: 499 messages
>>>>>>>>>>>>>>>> suppressed
>>>>>>>>>>>>>>>> If I reinstall alsa from deb everything sworks fine again (of course
>>>>>>>>>>>>>>>> with the current limitations).
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> First question: Has anybody had this issue before? How can I solve this?
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> Second question:
>>>>>>>>>>>>>>>> As I do not need 64 channels or more: how do I manage, that Skype
>>>>>>>>>>>>>>>> instances 9..15 use a second instance of snd-dummy as addressed in the wiki?
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> Best regards
>>>>>>>>>>>>>>>> Peter
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> --
>>>>>>>>>>>>>>> Sincerely,
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> Giovanni Maruzzelli
>>>>>>>>>>>>>>> Cell : +39-347-2665618
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>> _______________________________________________
>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> --
>>> Ghulam Mustafa
>>> cell: +92 333.611.7681
>>> sip: cyrenity at ekiga.net
>>> mail: mustafa.pk at gmail.com
>>> web: cyrenity.wordpress.com
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From matt at webcontracts.co.uk  Fri Feb  5 05:53:49 2010
From: matt at webcontracts.co.uk (Matthew Law)
Date: Fri, 5 Feb 2010 13:53:49 -0000
Subject: [Freeswitch-users] Simple IAX2 setup - help with converting
 from asterisk to freeswitch
In-Reply-To: <E1E3F6EF-1A1E-4891-B371-7A3BF0988E95@jerris.com>
References: <91093d24f2166d220acf2ab19115f95d.squirrel@www.webcontracts.co.uk>
	<E1E3F6EF-1A1E-4891-B371-7A3BF0988E95@jerris.com>
Message-ID: <d2976552e9ba0ce1ed095b7cef22e1ad.squirrel@www.webcontracts.co.uk>

Why is that? - a lot of web pages I have read claim that IAX is more
secure and efficient.  I have no problem with using SIP whatsoever and it
certainly appears to be ubiquitous.  I am a complete newcomer to VoIP and
I am trying to do this as securely as possible since I want to run
freeswitch on a Xen VPS which will be visible on the internet.

Anyway, since my first question, I have worked my way through the wiki,
read a lot of example configs and made some sense of the docs.  I now have
a very basic config working (with SIP) on a local vmware image using the
'quick and dirty' Makefile method.  I removed all of the example configs
from the xml files (those default extensions and passwords scared me) and
added just one for extension 1000, plus my dialplan and inbound/outbound
settings.

One question: is there any reason not to use a smaller extension number
range, like 200-210, for example?

Now to figure out how to get time based roaming working...


Thanks,

Matt.


On Fri, February 5, 2010 6:43 am, Michael Jerris wrote:
> iax2 support has been removed from FreeSWITCH in current trunk and will
> not be in the 1.0.5 release.
>
> Mike



From Prometheus001 at gmx.net  Fri Feb  5 06:23:26 2010
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Fri, 05 Feb 2010 15:23:26 +0100
Subject: [Freeswitch-users] mod_skypiax - help needed with
 modified	snd-dummy (one way audio)
In-Reply-To: <7b197bef1002050536j7ea09684v913434296ae52b97@mail.gmail.com>
References: <4B60555B.2020004@gmx.net>	<7b197bef1001271215o64918f56s3d16e51528fc5f66@mail.gmail.com>
	<4B61ECE0.10409@gmx.net>	<7b197bef1001281210u43a907edi13afe794cf0e1a2e@mail.gmail.com>
	<4B61FCAB.5040707@gmx.net>	<7b197bef1001281341h7ee2c58fj79b3886630901f29@mail.gmail.com>
	<4B62C962.7000601@gmx.net>	<8213d6071001290403g2c08fe39gd92eb0446bb6485a@mail.gmail.com>
	<7b197bef1001290520x37179cb1o63273410e9ef5bc@mail.gmail.com>
	<4B6C1CCB.4080606@gmx.net>
	<7b197bef1002050536j7ea09684v913434296ae52b97@mail.gmail.com>
Message-ID: <4B6C29DE.3010107@gmx.net>

Hello Giovanni,

I will then try ubuntu 8.04 (hardy) LTS server 64bit now 
and report the results.

Best regards
Peter



Giovanni Maruzzelli schrieb:
> Ciao Peter,
>
> I would use ubuntu 8.04 (hardy) LTS server 64bit or CentOS 5.4 64bit.
>
> -giovanni
>
> On Fri, Feb 5, 2010 at 2:27 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>   
>> Hello Giovanni,
>>
>> as I couldn't even get skype again working again with the standard alsa
>> driver, I would like to setup the machine from scratch based on a
>> working machine.
>> The latest errors I received from Skype was:
>>  snd_pcm_avail_update() returned a value that is exceptionally large:
>> 715706624 bytes (3727638 ms).
>>  Most likely this is a bug in the ALSA driver. Please report this issue
>> to the ALSA developers.
>> I think that may be the reason for one-way-audio.
>>
>> For setting up my machine from scratch, please advise:
>> - which OS you are you using und recommending exactly?
>> - I would like to use 64bit OS in order to use 8GB of memory, does this
>> work?
>> - any other hints?
>>
>> Best regards
>> Peter
>>
>> Giovanni Maruzzelli schrieb:
>>     
>>> Peter,
>>>
>>> Can you connect on IRC (irc.freenode.net #freeswitch)? I'm gmaruzz there.
>>>
>>> Can you restate your problems? I've lost connection :)
>>>
>>> with snd-dummy custom you can create *one only* snd-dummy instance, so
>>> *one only* fake soundcard. If you create more, will not work. But with
>>> that one fake soundcard you can use 64 skype client instances, all
>>> with the same soundcard hardware device (hw:n).
>>>
>>> with original snd-dummy you can create a max of 8 instances, so 8 fake
>>> soundcards, and with each fake soundcard you can  use a max of 8 skype
>>> client instances.
>>>
>>> use the hardware devices, not the default devices (use the "hw:n")
>>>
>>> -giovanni
>>>
>>> On Fri, Jan 29, 2010 at 1:03 PM, Ghulam Mustafa <mustafa.pk at gmail.com> wrote:
>>>
>>>       
>>>> did you enable debug mode while compiling custom snd-dummy? if  yes
>>>> try re-compiling with debug mode disabled.
>>>>
>>>> -m
>>>>
>>>> On Fri, Jan 29, 2010 at 4:41 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>
>>>>         
>>>>> I now reinstalled the original sound drivers
>>>>> Unfortunaltely the sound problems remain, not that worse but they are there:
>>>>> Audio is still (almost) one way. Almost means:
>>>>>
>>>>>    * SIP -> Skype ok
>>>>>    * Skype=> SIP I hear only some scratching on very loud audio
>>>>>
>>>>> Could it be a volume problem? But snd-dummy should have no volume
>>>>> properties, right?
>>>>>
>>>>> Best regards
>>>>> Peter
>>>>>
>>>>>
>>>>> Giovanni Maruzzelli schrieb:
>>>>>
>>>>>           
>>>>>> with three instances you will assign the hw:0 device to skype client
>>>>>> 0...7, hw:1 to skype client 8...15 and hw:2 to skype client 16...23.
>>>>>> Must work. Pay attention to assign the same device name to all devices
>>>>>> needed by a skype instance (sound devices window): playback, capture
>>>>>> AND ring.
>>>>>>
>>>>>> Or maybe is a bug of ALSA on Debian...
>>>>>>
>>>>>> -giovanni
>>>>>>
>>>>>> On Thu, Jan 28, 2010 at 10:07 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>>
>>>>>>
>>>>>>             
>>>>>>> I crated 3 instances of snd-dummy, this worked. I assigned then Instance
>>>>>>> #2 to the Skype accounts. Still no sound.
>>>>>>> On the frist call there is one way audio, on the following calls there
>>>>>>> is no audio at all.
>>>>>>> This is weird.
>>>>>>>
>>>>>>> Best regards
>>>>>>> Peter
>>>>>>>
>>>>>>> Giovanni Maruzzelli schrieb:
>>>>>>>
>>>>>>>
>>>>>>>               
>>>>>>>> Ciao Peter,
>>>>>>>>
>>>>>>>> Never tested on Debian 5.
>>>>>>>>
>>>>>>>> When you write "same problem" you are referring to the audio going one
>>>>>>>> way only (btw, which way?) with the custom audio driver?
>>>>>>>>
>>>>>>>> Have you tried with multiple instances of the regular Debian
>>>>>>>> snd-dummy, as I wrote in a mail before?
>>>>>>>>
>>>>>>>> -gm
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> On Thu, Jan 28, 2010 at 9:00 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>                 
>>>>>>>>> Hello Giovanni,
>>>>>>>>>
>>>>>>>>> I did so but the same problem again.
>>>>>>>>>
>>>>>>>>> Did you ever test in on Debian 5.0?
>>>>>>>>>
>>>>>>>>> Best reards
>>>>>>>>> Peter
>>>>>>>>>
>>>>>>>>> Giovanni Maruzzelli schrieb:
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>                   
>>>>>>>>>> good, so you have only one sound device, the right one.
>>>>>>>>>>
>>>>>>>>>> Use the one with hw:0 in the window that skype gives you to set sound devices
>>>>>>>>>>
>>>>>>>>>> -gm
>>>>>>>>>>
>>>>>>>>>> On Wed, Jan 27, 2010 at 7:04 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>                     
>>>>>>>>>>> I installed alsa-utile,
>>>>>>>>>>>
>>>>>>>>>>> now I get:
>>>>>>>>>>>
>>>>>>>>>>> skype:/var/cache/apt/archives# aplay -l
>>>>>>>>>>> **** List of PLAYBACK Hardware Devices ****
>>>>>>>>>>> card 0: Dummy [Dummy], device 0: Dummy PCM [Dummy PCM]
>>>>>>>>>>>  Subdevices: 127/128
>>>>>>>>>>>  Subdevice #0: subdevice #0
>>>>>>>>>>>  Subdevice #1: subdevice #1
>>>>>>>>>>>  Subdevice #2: subdevice #2
>>>>>>>>>>>  Subdevice #3: subdevice #3
>>>>>>>>>>>  Subdevice #4: subdevice #4
>>>>>>>>>>>  Subdevice #5: subdevice #5
>>>>>>>>>>>  Subdevice #6: subdevice #6
>>>>>>>>>>>  Subdevice #7: subdevice #7
>>>>>>>>>>>  Subdevice #8: subdevice #8
>>>>>>>>>>>  Subdevice #9: subdevice #9
>>>>>>>>>>>  Subdevice #10: subdevice #10
>>>>>>>>>>>  Subdevice #11: subdevice #11
>>>>>>>>>>>  Subdevice #12: subdevice #12
>>>>>>>>>>>  Subdevice #13: subdevice #13
>>>>>>>>>>>  Subdevice #14: subdevice #14
>>>>>>>>>>>  Subdevice #15: subdevice #15
>>>>>>>>>>>  Subdevice #16: subdevice #16
>>>>>>>>>>>  Subdevice #17: subdevice #17
>>>>>>>>>>>  Subdevice #18: subdevice #18
>>>>>>>>>>>  Subdevice #19: subdevice #19
>>>>>>>>>>>  Subdevice #20: subdevice #20
>>>>>>>>>>>  Subdevice #21: subdevice #21
>>>>>>>>>>>  Subdevice #22: subdevice #22
>>>>>>>>>>>  Subdevice #23: subdevice #23
>>>>>>>>>>>  Subdevice #24: subdevice #24
>>>>>>>>>>>  Subdevice #25: subdevice #25
>>>>>>>>>>>  Subdevice #26: subdevice #26
>>>>>>>>>>>  Subdevice #27: subdevice #27
>>>>>>>>>>>  Subdevice #28: subdevice #28
>>>>>>>>>>>  Subdevice #29: subdevice #29
>>>>>>>>>>>  Subdevice #30: subdevice #30
>>>>>>>>>>>  Subdevice #31: subdevice #31
>>>>>>>>>>>  Subdevice #32: subdevice #32
>>>>>>>>>>>  Subdevice #33: subdevice #33
>>>>>>>>>>>  Subdevice #34: subdevice #34
>>>>>>>>>>>  Subdevice #35: subdevice #35
>>>>>>>>>>>  Subdevice #36: subdevice #36
>>>>>>>>>>>  Subdevice #37: subdevice #37
>>>>>>>>>>>  Subdevice #38: subdevice #38
>>>>>>>>>>>  Subdevice #39: subdevice #39
>>>>>>>>>>>  Subdevice #40: subdevice #40
>>>>>>>>>>>  Subdevice #41: subdevice #41
>>>>>>>>>>>  Subdevice #42: subdevice #42
>>>>>>>>>>>  Subdevice #43: subdevice #43
>>>>>>>>>>>  Subdevice #44: subdevice #44
>>>>>>>>>>>  Subdevice #45: subdevice #45
>>>>>>>>>>>  Subdevice #46: subdevice #46
>>>>>>>>>>>  Subdevice #47: subdevice #47
>>>>>>>>>>>  Subdevice #48: subdevice #48
>>>>>>>>>>>  Subdevice #49: subdevice #49
>>>>>>>>>>>  Subdevice #50: subdevice #50
>>>>>>>>>>>  Subdevice #51: subdevice #51
>>>>>>>>>>>  Subdevice #52: subdevice #52
>>>>>>>>>>>  Subdevice #53: subdevice #53
>>>>>>>>>>>  Subdevice #54: subdevice #54
>>>>>>>>>>>  Subdevice #55: subdevice #55
>>>>>>>>>>>  Subdevice #56: subdevice #56
>>>>>>>>>>>  Subdevice #57: subdevice #57
>>>>>>>>>>>  Subdevice #58: subdevice #58
>>>>>>>>>>>  Subdevice #59: subdevice #59
>>>>>>>>>>>  Subdevice #60: subdevice #60
>>>>>>>>>>>  Subdevice #61: subdevice #61
>>>>>>>>>>>  Subdevice #62: subdevice #62
>>>>>>>>>>>  Subdevice #63: subdevice #63
>>>>>>>>>>>  Subdevice #64: subdevice #64
>>>>>>>>>>>  Subdevice #65: subdevice #65
>>>>>>>>>>>  Subdevice #66: subdevice #66
>>>>>>>>>>>  Subdevice #67: subdevice #67
>>>>>>>>>>>  Subdevice #68: subdevice #68
>>>>>>>>>>>  Subdevice #69: subdevice #69
>>>>>>>>>>>  Subdevice #70: subdevice #70
>>>>>>>>>>>  Subdevice #71: subdevice #71
>>>>>>>>>>>  Subdevice #72: subdevice #72
>>>>>>>>>>>  Subdevice #73: subdevice #73
>>>>>>>>>>>  Subdevice #74: subdevice #74
>>>>>>>>>>>  Subdevice #75: subdevice #75
>>>>>>>>>>>  Subdevice #76: subdevice #76
>>>>>>>>>>>  Subdevice #77: subdevice #77
>>>>>>>>>>>  Subdevice #78: subdevice #78
>>>>>>>>>>>  Subdevice #79: subdevice #79
>>>>>>>>>>>  Subdevice #80: subdevice #80
>>>>>>>>>>>  Subdevice #81: subdevice #81
>>>>>>>>>>>  Subdevice #82: subdevice #82
>>>>>>>>>>>  Subdevice #83: subdevice #83
>>>>>>>>>>>  Subdevice #84: subdevice #84
>>>>>>>>>>>  Subdevice #85: subdevice #85
>>>>>>>>>>>  Subdevice #86: subdevice #86
>>>>>>>>>>>  Subdevice #87: subdevice #87
>>>>>>>>>>>  Subdevice #88: subdevice #88
>>>>>>>>>>>  Subdevice #89: subdevice #89
>>>>>>>>>>>  Subdevice #90: subdevice #90
>>>>>>>>>>>  Subdevice #91: subdevice #91
>>>>>>>>>>>  Subdevice #92: subdevice #92
>>>>>>>>>>>  Subdevice #93: subdevice #93
>>>>>>>>>>>  Subdevice #94: subdevice #94
>>>>>>>>>>>  Subdevice #95: subdevice #95
>>>>>>>>>>>  Subdevice #96: subdevice #96
>>>>>>>>>>>  Subdevice #97: subdevice #97
>>>>>>>>>>>  Subdevice #98: subdevice #98
>>>>>>>>>>>  Subdevice #99: subdevice #99
>>>>>>>>>>>  Subdevice #100: subdevice #100
>>>>>>>>>>>  Subdevice #101: subdevice #101
>>>>>>>>>>>  Subdevice #102: subdevice #102
>>>>>>>>>>>  Subdevice #103: subdevice #103
>>>>>>>>>>>  Subdevice #104: subdevice #104
>>>>>>>>>>>  Subdevice #105: subdevice #105
>>>>>>>>>>>  Subdevice #106: subdevice #106
>>>>>>>>>>>  Subdevice #107: subdevice #107
>>>>>>>>>>>  Subdevice #108: subdevice #108
>>>>>>>>>>>  Subdevice #109: subdevice #109
>>>>>>>>>>>  Subdevice #110: subdevice #110
>>>>>>>>>>>  Subdevice #111: subdevice #111
>>>>>>>>>>>  Subdevice #112: subdevice #112
>>>>>>>>>>>  Subdevice #113: subdevice #113
>>>>>>>>>>>  Subdevice #114: subdevice #114
>>>>>>>>>>>  Subdevice #115: subdevice #115
>>>>>>>>>>>  Subdevice #116: subdevice #116
>>>>>>>>>>>  Subdevice #117: subdevice #117
>>>>>>>>>>>  Subdevice #118: subdevice #118
>>>>>>>>>>>  Subdevice #119: subdevice #119
>>>>>>>>>>>  Subdevice #120: subdevice #120
>>>>>>>>>>>  Subdevice #121: subdevice #121
>>>>>>>>>>>  Subdevice #122: subdevice #122
>>>>>>>>>>>  Subdevice #123: subdevice #123
>>>>>>>>>>>  Subdevice #124: subdevice #124
>>>>>>>>>>>  Subdevice #125: subdevice #125
>>>>>>>>>>>  Subdevice #126: subdevice #126
>>>>>>>>>>>  Subdevice #127: subdevice #127
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> Peter P GMX schrieb:
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>                       
>>>>>>>>>>>> Her's the output:
>>>>>>>>>>>>
>>>>>>>>>>>> skype:~# aplay -l
>>>>>>>>>>>> bash: aplay: command not found
>>>>>>>>>>>>
>>>>>>>>>>>> Giovanni Maruzzelli schrieb:
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>                         
>>>>>>>>>>>>> I don't think you got two snd-dummy loaded (but maybe yes)
>>>>>>>>>>>>> what's the output of:
>>>>>>>>>>>>>
>>>>>>>>>>>>> aplay -l
>>>>>>>>>>>>>
>>>>>>>>>>>>> ?
>>>>>>>>>>>>>
>>>>>>>>>>>>> If instead you are referring to the choices that skype clients offers
>>>>>>>>>>>>> you in the "set audio devices" window, choose Dummy PCM (hw0:0)
>>>>>>>>>>>>>
>>>>>>>>>>>>> Eg: not the "default", but the "hardware" one
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>> On Wed, Jan 27, 2010 at 5:58 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>                           
>>>>>>>>>>>>>> Thanks Giovanni,
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> I think there may be the problem, that I have 2 sound devices now:
>>>>>>>>>>>>>> - Dummy PCM (hw0:0) (this is from debian install)
>>>>>>>>>>>>>> - Dummy PCM Default Audio device (defauzlt: CARD=Dummy) (this is new
>>>>>>>>>>>>>> since I compiled alsa newly)
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> I tried both, but both do not work. How do I get rid of the old alsa device?
>>>>>>>>>>>>>> By the way: I uninstalled Alsa before I installed the new driver
>>>>>>>>>>>>>> (apt-get remove alsa-utils alsa-base).
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> Best regards
>>>>>>>>>>>>>> Peter
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> Giovanni Maruzzelli schrieb:
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>                             
>>>>>>>>>>>>>>> This warning is harmless:
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> Jan 27 15:28:50 skype kernel: [ 3994.795786] process `skype' is using
>>>>>>>>>>>>>>> obsolete setsockopt SO_BSDCOMPAT
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> On Wed, Jan 27, 2010 at 4:26 PM, Giovanni Maruzzelli
>>>>>>>>>>>>>>> <gmaruzz at celliax.org> wrote:
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>                               
>>>>>>>>>>>>>>>> Ciao Peter
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> one instance of snd-dummy "customized" is enough for 64 instances of
>>>>>>>>>>>>>>>> skype clients, no need (and do not works) with more instances of
>>>>>>>>>>>>>>>> snd-dummy-customized.
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> Maybe you got the one-way problem because of kernel at 250HZ (don't
>>>>>>>>>>>>>>>> know). It uses to works well on a tickless kernel at 100HZ (eg: ubuntu
>>>>>>>>>>>>>>>> 8.04).
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> Or maybe you have to check and modify which sound devices the skype
>>>>>>>>>>>>>>>> clients are using (try to check that with snd-summy-custom loaded,
>>>>>>>>>>>>>>>> maybe with the ssh -X trick (as in the wiki page).
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> To load more than one snd-dummy-original (the non modified one), you
>>>>>>>>>>>>>>>> do this with the modprobe command, as in:
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> rmmod snd-dummy
>>>>>>>>>>>>>>>> modprobe snd-dummy enable=1,1,1
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> this command will enable three instances of snd-dummy original, so
>>>>>>>>>>>>>>>> you'll have three fake soundcards, and you'll have to setup each group
>>>>>>>>>>>>>>>> of 8 skype instances to use sound devices from one fake soundcard, RG:
>>>>>>>>>>>>>>>> no more than 8 skype client instances can use one instance of fake
>>>>>>>>>>>>>>>> soundcard.
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> Also, please update the mod_skypiax code (svn up in its directory) I
>>>>>>>>>>>>>>>> just committed some improvements.
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> If you have any other doubts, or need more info, don't hesitate to
>>>>>>>>>>>>>>>> write the mailing list again,
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> ciao for now,
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> -giovanni
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> On Wed, Jan 27, 2010 at 4:01 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>                                 
>>>>>>>>>>>>>>>>> I have mod_skypiax working nicely so far with 2 Skype channels. Thanks
>>>>>>>>>>>>>>>>> to all contributors, excellent work!
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> In order to have more than 8 channels working, I have followed the
>>>>>>>>>>>>>>>>> instructions in
>>>>>>>>>>>>>>>>> http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk
>>>>>>>>>>>>>>>>> and compiled alsa-driver-1.0.20 with the modified dummy.c file. (System
>>>>>>>>>>>>>>>>> ist Debian 5.0R3)
>>>>>>>>>>>>>>>>> It compiled well however when I start snd-dummy I only have
>>>>>>>>>>>>>>>>> one-way-audio and my logs show
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> Jan 27 15:28:41 skype kernel: [ 3984.318403] snd-dummy skypiax driver,
>>>>>>>>>>>>>>>>> /usr/src/alsa-driver-1.0.20/drivers/../alsa-kernel/drivers/dummy.c:920
>>>>>>>>>>>>>>>>> working on a machine with 250HZ kernel
>>>>>>>>>>>>>>>>> Jan 27 15:28:50 skype kernel: [ 3994.795786] process `skype' is using
>>>>>>>>>>>>>>>>> obsolete setsockopt SO_BSDCOMPAT
>>>>>>>>>>>>>>>>> Jan 27 15:28:56 skype kernel: [ 4005.289907] __ratelimit: 490 messages
>>>>>>>>>>>>>>>>> suppressed
>>>>>>>>>>>>>>>>> Jan 27 15:29:01 skype kernel: [ 4012.458310] process `skype' is using
>>>>>>>>>>>>>>>>> obsolete setsockopt SO_BSDCOMPAT
>>>>>>>>>>>>>>>>> Jan 27 15:29:01 skype kernel: [ 4013.326290] __ratelimit: 499 messages
>>>>>>>>>>>>>>>>> suppressed
>>>>>>>>>>>>>>>>> If I reinstall alsa from deb everything sworks fine again (of course
>>>>>>>>>>>>>>>>> with the current limitations).
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> First question: Has anybody had this issue before? How can I solve this?
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> Second question:
>>>>>>>>>>>>>>>>> As I do not need 64 channels or more: how do I manage, that Skype
>>>>>>>>>>>>>>>>> instances 9..15 use a second instance of snd-dummy as addressed in the wiki?
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> Best regards
>>>>>>>>>>>>>>>>> Peter
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>                                   
>>>>>>>>>>>>>>>> --
>>>>>>>>>>>>>>>> Sincerely,
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> Giovanni Maruzzelli
>>>>>>>>>>>>>>>> Cell : +39-347-2665618
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>                                 
>>>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>                             
>>>>>>>>>>> _______________________________________________
>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>                       
>>>>>>>>> _______________________________________________
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>                   
>>>>>>>>                 
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>               
>>>>>>
>>>>>>             
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>>           
>>>> --
>>>> Ghulam Mustafa
>>>> cell: +92 333.611.7681
>>>> sip: cyrenity at ekiga.net
>>>> mail: mustafa.pk at gmail.com
>>>> web: cyrenity.wordpress.com
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>         
>>>
>>>
>>>       
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>     
>
>
>
>   


From rupa at rupa.com  Fri Feb  5 06:57:18 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Fri, 5 Feb 2010 08:57:18 -0600
Subject: [Freeswitch-users] Simple IAX2 setup - help with converting
	from asterisk to freeswitch
In-Reply-To: <d2976552e9ba0ce1ed095b7cef22e1ad.squirrel@www.webcontracts.co.uk>
References: <91093d24f2166d220acf2ab19115f95d.squirrel@www.webcontracts.co.uk>
	<E1E3F6EF-1A1E-4891-B371-7A3BF0988E95@jerris.com>
	<d2976552e9ba0ce1ed095b7cef22e1ad.squirrel@www.webcontracts.co.uk>
Message-ID: <b39273531002050657x4f0f921euce8a683b8018ed63@mail.gmail.com>

the lib that we used to provide iax support is pretty much abandonware (no
longer updated) and newer iax implementations (like latest asterisk) can
cause it to crash.  There are no license compatible iax implementations that
work, so.. mod_iax has been moved to the unsupported column.

Default passwords -- that is a single var in vars.xml that controls the
passwords.

number ranges - up to you.  The sample configs supplied are just that,
samples.  I use a smaller range personally.

On Fri, Feb 5, 2010 at 7:53 AM, Matthew Law <matt at webcontracts.co.uk> wrote:

> Why is that? - a lot of web pages I have read claim that IAX is more
> secure and efficient.  I have no problem with using SIP whatsoever and it
> certainly appears to be ubiquitous.  I am a complete newcomer to VoIP and
> I am trying to do this as securely as possible since I want to run
> freeswitch on a Xen VPS which will be visible on the internet.
>
> Anyway, since my first question, I have worked my way through the wiki,
> read a lot of example configs and made some sense of the docs.  I now have
> a very basic config working (with SIP) on a local vmware image using the
> 'quick and dirty' Makefile method.  I removed all of the example configs
> from the xml files (those default extensions and passwords scared me) and
> added just one for extension 1000, plus my dialplan and inbound/outbound
> settings.
>
> One question: is there any reason not to use a smaller extension number
> range, like 200-210, for example?
>
> Now to figure out how to get time based roaming working...
>
>
> Thanks,
>
> Matt.
>
>
> On Fri, February 5, 2010 6:43 am, Michael Jerris wrote:
> > iax2 support has been removed from FreeSWITCH in current trunk and will
> > not be in the 1.0.5 release.
> >
> > Mike
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
-Rupa
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100205/61672bcc/attachment-0001.html 

From brian at freeswitch.org  Fri Feb  5 07:09:39 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 5 Feb 2010 09:09:39 -0600
Subject: [Freeswitch-users] No reINVITE when bridging two sessions from
	JavaScript with bypass_media_after_bridge=true
In-Reply-To: <8e9d67561002050018q24e3a806kc8807b55268e759b@mail.gmail.com>
References: <8e9d67561002041147q6c135546u1f7e1887529a916d@mail.gmail.com>
	<3D739D7F-F5C6-4266-B04E-9413DCFDA6AF@freeswitch.org>
	<8e9d67561002050018q24e3a806kc8807b55268e759b@mail.gmail.com>
Message-ID: <B66289FF-C2A9-4AC4-A56C-66E51B4726F4@freeswitch.org>

set it inside each of the {} for each session you create its not set after the fact the call is up already...  you're setting it too late.

you an also issue uuid_media off <uuid>

/b

On Feb 5, 2010, at 2:18 AM, Timur Valishev wrote:

> I've modified my script to make sure: <<<<<<<<<<<<<<<<<<<<<<<<<<<<,
> session = new Session(
> "{ignore_early_media=true,hangup_after_bridge=true}sofia/external/timwork at novion.ru"
> );
> session2 = new Session(
> "{ignore_early_media=true}sofia/external/timwork at novion.ru"
> );
> session.setVariable('bypass_media', 'true');
> session2.setVariable('bypass_media', 'true');
> bridge(session, session2);

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100205/48a0ce6f/attachment.html 

From john at acsol.net  Fri Feb  5 07:11:26 2010
From: john at acsol.net (John)
Date: Fri, 05 Feb 2010 08:11:26 -0700
Subject: [Freeswitch-users] Switch Security
Message-ID: <4B6C351E.6080608@acsol.net>

Freeswitch is to be used by phones external to my lan. Many of the 
phones will be coming from DSL connections without static IP. I have 
disabled the default password. What is a good procedure to secure the 
switch from non-customers registering? I know that I could use an ACL; 
however it's difficult with all the non-static users.

Thanks John


From brian at freeswitch.org  Fri Feb  5 07:13:29 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 5 Feb 2010 09:13:29 -0600
Subject: [Freeswitch-users] Looking for some good/cheap desktop phones
In-Reply-To: <65d96fc81002050521w746707b1q4398e6d8a0692d1e@mail.gmail.com>
References: <3f2e071b5aea5e11bb37a37e435a33b7@mail.gmail.com>
	<8e9d67561002042236s434e0643od322551de8720be1@mail.gmail.com>
	<409D0ACF-ADA4-4D02-8B7E-71EC1B1BAFBC@freeswitch.org>
	<8e9d67561002042301v76129abfj3dcbc70ed073752c@mail.gmail.com>
	<d361ca7b1002050507n5e564621oa6cf7ace005dd5b8@mail.gmail.com>
	<65d96fc81002050521w746707b1q4398e6d8a0692d1e@mail.gmail.com>
Message-ID: <6B520D1B-EDD6-44F7-9DE5-A384D3286138@freeswitch.org>

Sigh... When is someone actually going to build an open platform voip hardware phone... Its just a linux box that happens to be shaped like a phone, with a touch screen, 48kHz sound card... and possibly video too.

/b
PS: Most of those eastern made phones are crap. 

On Feb 5, 2010, at 7:21 AM, Tihomir Culjaga wrote:

> Atcom AT-620 (http://www.atcom.cn/AT620.html#~tab-small_midsized_large_enterprises) is quite ok and cheap (~30$)... also we have been talking to Atcom to add a sort of auto-provissioning (dhcp/http) and this is going to happen next week.
> 
> 
> T.

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100205/b1a26745/attachment.html 

From brian at freeswitch.org  Fri Feb  5 07:28:42 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 5 Feb 2010 09:28:42 -0600
Subject: [Freeswitch-users] Switch Security
In-Reply-To: <4B6C351E.6080608@acsol.net>
References: <4B6C351E.6080608@acsol.net>
Message-ID: <2ED7083F-F294-40F8-B46C-1D2E0E46509F@freeswitch.org>

Give them passwords... install fail2ban... http://wiki.freeswitch.org/wiki/Fail2ban

/b

On Feb 5, 2010, at 9:11 AM, John wrote:

> Freeswitch is to be used by phones external to my lan. Many of the 
> phones will be coming from DSL connections without static IP. I have 
> disabled the default password. What is a good procedure to secure the 
> switch from non-customers registering? I know that I could use an ACL; 
> however it's difficult with all the non-static users.
> 
> Thanks John



From dave at 3c.co.uk  Fri Feb  5 07:31:11 2010
From: dave at 3c.co.uk (David Knell)
Date: Fri, 05 Feb 2010 08:31:11 -0700
Subject: [Freeswitch-users] Looking for some good/cheap desktop phones
In-Reply-To: <6B520D1B-EDD6-44F7-9DE5-A384D3286138@freeswitch.org>
References: <3f2e071b5aea5e11bb37a37e435a33b7@mail.gmail.com>
	<8e9d67561002042236s434e0643od322551de8720be1@mail.gmail.com>
	<409D0ACF-ADA4-4D02-8B7E-71EC1B1BAFBC@freeswitch.org>
	<8e9d67561002042301v76129abfj3dcbc70ed073752c@mail.gmail.com>
	<d361ca7b1002050507n5e564621oa6cf7ace005dd5b8@mail.gmail.com>
	<65d96fc81002050521w746707b1q4398e6d8a0692d1e@mail.gmail.com>
	<6B520D1B-EDD6-44F7-9DE5-A384D3286138@freeswitch.org>
Message-ID: <1265383871.12871.71.camel@local.freepabx.com>

Hi Brian,

This is a start:
http://www.digitmat.com/ - you need to follow some links, but it's open
source.

--Dave

> Sigh... When is someone actually going to build an open platform voip
> hardware phone... Its just a linux box that happens to be shaped like
> a phone, with a touch screen, 48kHz sound card... and possibly video
> too.
> 
> 
> /b
> PS: Most of those eastern made phones are crap. 
> 
> On Feb 5, 2010, at 7:21 AM, Tihomir Culjaga wrote:
> 
> > Atcom AT-620
> > (http://www.atcom.cn/AT620.html#~tab-small_midsized_large_enterprises) is quite ok and cheap (~30$)... also we have been talking to Atcom to add a sort of auto-provissioning (dhcp/http) and this is going to happen next week.
> > 
> > 
> > T.
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From mike at jerris.com  Fri Feb  5 07:54:09 2010
From: mike at jerris.com (Michael Jerris)
Date: Fri, 5 Feb 2010 10:54:09 -0500
Subject: [Freeswitch-users] Simple IAX2 setup - help with converting
	from asterisk to freeswitch
In-Reply-To: <d2976552e9ba0ce1ed095b7cef22e1ad.squirrel@www.webcontracts.co.uk>
References: <91093d24f2166d220acf2ab19115f95d.squirrel@www.webcontracts.co.uk>
	<E1E3F6EF-1A1E-4891-B371-7A3BF0988E95@jerris.com>
	<d2976552e9ba0ce1ed095b7cef22e1ad.squirrel@www.webcontracts.co.uk>
Message-ID: <4D6421C5-9336-40D2-B54C-F773B2E6BA0E@jerris.com>

I find the secure and efficiency claims on IAX to be pretty much a farce.  IAX offers less overhead on rtp, but the same thing can be accomplished with rtp using packetization, as for security, I don't see any credible claim on that.  IAX also forces the program to sort out a ton of audio for different users going to 1 socket, something that a network stack is quite good at when using different ports, but is a lot more work where we are getting the packets.  As for default passwords and users, of course I wouldn't use those in production, those are for you to see how the pieces work together out of the box.  I wouldn't however quickly scrap the entire default config, just read through them and think about what you need and do not.  The extension ranges you use is totally at your discretion.

Mike


On Feb 5, 2010, at 8:53 AM, Matthew Law wrote:

> Why is that? - a lot of web pages I have read claim that IAX is more
> secure and efficient.  I have no problem with using SIP whatsoever and it
> certainly appears to be ubiquitous.  I am a complete newcomer to VoIP and
> I am trying to do this as securely as possible since I want to run
> freeswitch on a Xen VPS which will be visible on the internet.
> 
> Anyway, since my first question, I have worked my way through the wiki,
> read a lot of example configs and made some sense of the docs.  I now have
> a very basic config working (with SIP) on a local vmware image using the
> 'quick and dirty' Makefile method.  I removed all of the example configs
> from the xml files (those default extensions and passwords scared me) and
> added just one for extension 1000, plus my dialplan and inbound/outbound
> settings.
> 
> One question: is there any reason not to use a smaller extension number
> range, like 200-210, for example?
> 
> Now to figure out how to get time based roaming working?



From dave at 3c.co.uk  Fri Feb  5 08:03:54 2010
From: dave at 3c.co.uk (David Knell)
Date: Fri, 05 Feb 2010 09:03:54 -0700
Subject: [Freeswitch-users] Simple IAX2 setup - help with
	converting	from asterisk to freeswitch
In-Reply-To: <b39273531002050657x4f0f921euce8a683b8018ed63@mail.gmail.com>
References: <91093d24f2166d220acf2ab19115f95d.squirrel@www.webcontracts.co.uk>
	<E1E3F6EF-1A1E-4891-B371-7A3BF0988E95@jerris.com>
	<d2976552e9ba0ce1ed095b7cef22e1ad.squirrel@www.webcontracts.co.uk>
	<b39273531002050657x4f0f921euce8a683b8018ed63@mail.gmail.com>
Message-ID: <1265385834.12871.83.camel@local.freepabx.com>

There's a fairly simple solution to IAX needs, which is to run Asterisk,
probably on the same box, as a protocol converter - you just need to
tell it to use a non-standard port in sip.conf so that it doesn't clash
with FreeSWITCH.

--Dave

> the lib that we used to provide iax support is pretty much abandonware
> (no longer updated) and newer iax implementations (like latest
> asterisk) can cause it to crash.  There are no license compatible iax
> implementations that work, so.. mod_iax has been moved to the
> unsupported column.
> 
> 
> Default passwords -- that is a single var in vars.xml that controls
> the passwords.
> 
> 
> number ranges - up to you.  The sample configs supplied are just that,
> samples.  I use a smaller range personally.
> 
> On Fri, Feb 5, 2010 at 7:53 AM, Matthew Law <matt at webcontracts.co.uk>
> wrote:
>         Why is that? - a lot of web pages I have read claim that IAX
>         is more
>         secure and efficient.  I have no problem with using SIP
>         whatsoever and it
>         certainly appears to be ubiquitous.  I am a complete newcomer
>         to VoIP and
>         I am trying to do this as securely as possible since I want to
>         run
>         freeswitch on a Xen VPS which will be visible on the internet.
>         
>         Anyway, since my first question, I have worked my way through
>         the wiki,
>         read a lot of example configs and made some sense of the
>         docs.  I now have
>         a very basic config working (with SIP) on a local vmware image
>         using the
>         'quick and dirty' Makefile method.  I removed all of the
>         example configs
>         from the xml files (those default extensions and passwords
>         scared me) and
>         added just one for extension 1000, plus my dialplan and
>         inbound/outbound
>         settings.
>         
>         One question: is there any reason not to use a smaller
>         extension number
>         range, like 200-210, for example?
>         
>         Now to figure out how to get time based roaming working...
>         
>         
>         Thanks,
>         
>         Matt.
>         
>         
>         On Fri, February 5, 2010 6:43 am, Michael Jerris wrote:
>         > iax2 support has been removed from FreeSWITCH in current
>         trunk and will
>         > not be in the 1.0.5 release.
>         >
>         
>         
>         > Mike
>         
>         
>         _______________________________________________
>         FreeSWITCH-users mailing list
>         FreeSWITCH-users at lists.freeswitch.org
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         http://www.freeswitch.org
>         
> 
> 
> 
> -- 
> -Rupa
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From rupa at rupa.com  Fri Feb  5 08:18:24 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Fri, 5 Feb 2010 10:18:24 -0600
Subject: [Freeswitch-users] Switch Security
In-Reply-To: <2ED7083F-F294-40F8-B46C-1D2E0E46509F@freeswitch.org>
References: <4B6C351E.6080608@acsol.net>
	<2ED7083F-F294-40F8-B46C-1D2E0E46509F@freeswitch.org>
Message-ID: <b39273531002050818m39b2b45er762ffc491b0e80cd@mail.gmail.com>

That wiki needs to actually have the fail2ban rules for freeswitch
documented....

hmm... ok, I finished up the documentation with what is needed to actually
configure/verify fail2ban.

On Fri, Feb 5, 2010 at 9:28 AM, Brian West <brian at freeswitch.org> wrote:

> Give them passwords... install fail2ban...
> http://wiki.freeswitch.org/wiki/Fail2ban
>
> /b
>
> On Feb 5, 2010, at 9:11 AM, John wrote:
>
> > Freeswitch is to be used by phones external to my lan. Many of the
> > phones will be coming from DSL connections without static IP. I have
> > disabled the default password. What is a good procedure to secure the
> > switch from non-customers registering? I know that I could use an ACL;
> > however it's difficult with all the non-static users.
> >
> > Thanks John
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
-Rupa
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100205/f6f118e9/attachment-0001.html 

From jmesquita at freeswitch.org  Fri Feb  5 08:23:02 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Fri, 5 Feb 2010 14:23:02 -0200
Subject: [Freeswitch-users] Switch Security
In-Reply-To: <b39273531002050818m39b2b45er762ffc491b0e80cd@mail.gmail.com>
References: <4B6C351E.6080608@acsol.net>
	<2ED7083F-F294-40F8-B46C-1D2E0E46509F@freeswitch.org>
	<b39273531002050818m39b2b45er762ffc491b0e80cd@mail.gmail.com>
Message-ID: <ebbc2d551002050823s39f17451vd21cbe46a9509a19@mail.gmail.com>

Rupa, like usual, thank you.

Regards,
Jo?o Mesquita


On Fri, Feb 5, 2010 at 2:18 PM, Rupa Schomaker <rupa at rupa.com> wrote:

> That wiki needs to actually have the fail2ban rules for freeswitch
> documented....
>
> hmm... ok, I finished up the documentation with what is needed to actually
> configure/verify fail2ban.
>
>
> On Fri, Feb 5, 2010 at 9:28 AM, Brian West <brian at freeswitch.org> wrote:
>
>> Give them passwords... install fail2ban...
>> http://wiki.freeswitch.org/wiki/Fail2ban
>>
>> /b
>>
>> On Feb 5, 2010, at 9:11 AM, John wrote:
>>
>> > Freeswitch is to be used by phones external to my lan. Many of the
>> > phones will be coming from DSL connections without static IP. I have
>> > disabled the default password. What is a good procedure to secure the
>> > switch from non-customers registering? I know that I could use an ACL;
>> > however it's difficult with all the non-static users.
>> >
>> > Thanks John
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100205/eee5c222/attachment.html 

From tculjaga at gmail.com  Fri Feb  5 08:42:06 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Fri, 5 Feb 2010 17:42:06 +0100
Subject: [Freeswitch-users] Looking for some good/cheap desktop phones
In-Reply-To: <1265383871.12871.71.camel@local.freepabx.com>
References: <3f2e071b5aea5e11bb37a37e435a33b7@mail.gmail.com>
	<8e9d67561002042236s434e0643od322551de8720be1@mail.gmail.com>
	<409D0ACF-ADA4-4D02-8B7E-71EC1B1BAFBC@freeswitch.org>
	<8e9d67561002042301v76129abfj3dcbc70ed073752c@mail.gmail.com>
	<d361ca7b1002050507n5e564621oa6cf7ace005dd5b8@mail.gmail.com>
	<65d96fc81002050521w746707b1q4398e6d8a0692d1e@mail.gmail.com>
	<6B520D1B-EDD6-44F7-9DE5-A384D3286138@freeswitch.org>
	<1265383871.12871.71.camel@local.freepabx.com>
Message-ID: <65d96fc81002050842l544d012eg2b3d43aba1e0d8dc@mail.gmail.com>

impressive!

This really looks nice: http://www.digitmat.com/res.html

i'm tempted to five it a shot :)


T.

On Fri, Feb 5, 2010 at 4:31 PM, David Knell <dave at 3c.co.uk> wrote:

> Hi Brian,
>
> This is a start:
> http://www.digitmat.com/ - you need to follow some links, but it's open
> source.
>
> --Dave
>
> > Sigh... When is someone actually going to build an open platform voip
> > hardware phone... Its just a linux box that happens to be shaped like
> > a phone, with a touch screen, 48kHz sound card... and possibly video
> > too.
> >
> >
> > /b
> > PS: Most of those eastern made phones are crap.
> >
> > On Feb 5, 2010, at 7:21 AM, Tihomir Culjaga wrote:
> >
> > > Atcom AT-620
> > > (http://www.atcom.cn/AT620.html#~tab-small_midsized_large_enterprises<http://www.atcom.cn/AT620.html#%7Etab-small_midsized_large_enterprises>)
> is quite ok and cheap (~30$)... also we have been talking to Atcom to add a
> sort of auto-provissioning (dhcp/http) and this is going to happen next
> week.
> > >
> > >
> > > T.
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100205/83717744/attachment.html 

From msc at freeswitch.org  Fri Feb  5 08:45:04 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 5 Feb 2010 08:45:04 -0800
Subject: [Freeswitch-users] FreeSWITCH Conference Call Starting Shortly!
Message-ID: <87f2f3b91002050845w70bb52s434dff55c11fec92@mail.gmail.com>

Come join us today!

http://wiki.freeswitch.org/wiki/FS_weekly_2010_02_5

-Michael
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100205/74f048cf/attachment.html 

From jerry.richards at teotech.com  Fri Feb  5 09:16:21 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Fri, 5 Feb 2010 09:16:21 -0800
Subject: [Freeswitch-users] Can Freeswitch and OpenSER Co-reside on Same
	Server?
Message-ID: <C4DC856C75094358AEDBE3FA0E242741@greyhawk.tonecommander.com>

If I use OpenSER for a session border controller, does anyone see an issue
if it resides on the same server as Freeswitch?  So I would have a LAN and
WAN socket?  Are there any drawbacks (other than loading) to worry about?

Thanks And Best Regards,
Jerry



From steveu at coppice.org  Fri Feb  5 09:44:33 2010
From: steveu at coppice.org (Steve Underwood)
Date: Sat, 06 Feb 2010 01:44:33 +0800
Subject: [Freeswitch-users] Looking for some good/cheap desktop phones
In-Reply-To: <1265383871.12871.71.camel@local.freepabx.com>
References: <3f2e071b5aea5e11bb37a37e435a33b7@mail.gmail.com>	<8e9d67561002042236s434e0643od322551de8720be1@mail.gmail.com>	<409D0ACF-ADA4-4D02-8B7E-71EC1B1BAFBC@freeswitch.org>	<8e9d67561002042301v76129abfj3dcbc70ed073752c@mail.gmail.com>	<d361ca7b1002050507n5e564621oa6cf7ace005dd5b8@mail.gmail.com>	<65d96fc81002050521w746707b1q4398e6d8a0692d1e@mail.gmail.com>	<6B520D1B-EDD6-44F7-9DE5-A384D3286138@freeswitch.org>
	<1265383871.12871.71.camel@local.freepabx.com>
Message-ID: <4B6C5901.1010509@coppice.org>

On 02/05/2010 11:31 PM, David Knell wrote:
> Hi Brian,
>
> This is a start:
> http://www.digitmat.com/ - you need to follow some links, but it's open
> source.
>
> --Dave
>    
Those people have a troubled history. They were the people behind many 
of the early cheap, but less than stellar, VoIP phones, with a chip that 
isn't made any more - the PA1688. They seem to have regrouped, and have 
a newer chip. I wonder if they have got their act together this time.
>> Sigh... When is someone actually going to build an open platform voip
>> hardware phone... Its just a linux box that happens to be shaped like
>> a phone, with a touch screen, 48kHz sound card... and possibly video
>> too.
>>
>>
>> /b
>> PS: Most of those eastern made phones are crap.
>>
>> On Feb 5, 2010, at 7:21 AM, Tihomir Culjaga wrote:
>>
>>      
>>> Atcom AT-620
>>> (http://www.atcom.cn/AT620.html#~tab-small_midsized_large_enterprises) is quite ok and cheap (~30$)... also we have been talking to Atcom to add a sort of auto-provissioning (dhcp/http) and this is going to happen next week.
>>>
>>>
>>> T.
>>>        
>>
ATCOM are amenable to the idea of supplying hardware for other people to 
put their own software on. Some of their phones use Infineon chip sets, 
and I think those are running software based on the Infineon Linux 
reference platform. All it takes is a few good people with the 
commitment to actually do something. I expect there are other Chinese 
makers who would be delighted to supply bare hardware to people, and a 
lot of the cheaper current Chinese phones have very similar Infineon 
based designs.

Steve



From joel.sisko at iconverged.com  Fri Feb  5 10:08:12 2010
From: joel.sisko at iconverged.com (joel.sisko at iconverged.com)
Date: Fri, 5 Feb 2010 12:08:12 -0600 (CST)
Subject: [Freeswitch-users] Use of mod_Conference
In-Reply-To: <472380364.66881265393034269.JavaMail.root@mail-2.01.com>
Message-ID: <648878714.68411265393292092.JavaMail.root@mail-2.01.com>

Group, 

I have a simple question I think about mod_Conference, does each conference room have to be created dynamically? My question is born from how Asterisk MeetMe room can work where you can have all the meetme rooms programmed statically in the meetme.conf file. 

What I am looking to do is rather than using the Asterisk meetme room application, I want to use FreeSwitch to provide this functionality. So I am trying to figure out what the integration would look like from a 10,000 foot view. 

Thanks for the help in advance. 

Joel 



-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100205/e52cbc2b/attachment.html 

From kristian.kielhofner at gmail.com  Fri Feb  5 10:11:35 2010
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Fri, 5 Feb 2010 13:11:35 -0500
Subject: [Freeswitch-users] Can Freeswitch and OpenSER Co-reside on Same
	Server?
In-Reply-To: <C4DC856C75094358AEDBE3FA0E242741@greyhawk.tonecommander.com>
References: <C4DC856C75094358AEDBE3FA0E242741@greyhawk.tonecommander.com>
Message-ID: <2d9149cd1002051011x3b4f548dqabe615ed5f991687@mail.gmail.com>

On Fri, Feb 5, 2010 at 12:16 PM, Jerry Richards
<jerry.richards at teotech.com> wrote:
> If I use OpenSER for a session border controller, does anyone see an issue
> if it resides on the same server as Freeswitch? ?So I would have a LAN and
> WAN socket? ?Are there any drawbacks (other than loading) to worry about?
>
> Thanks And Best Regards,
> Jerry
>

You can use different IP addresses or ports.  I do this all of the time.

I question why you are using OpenSER (OpenSIPS?) as a SBC.  FreeSWITCH
is actually more well suited to most of the functions served by
something called* a "session border controller".

For example, FreeSWITCH in bypass media mode is a signaling only SBC
where you can (cleanly) do the header rewriting, number formatting,
and SIP topology hiding typically done by a SBC without touching the
media.  Proxy media mode can do the same while proxying media
(traversing NAT and hiding real RTP addresses).  FreeSWITCH in normal
bridging mode can transcode, convert between different types of DTMF
and do everything else mentioned above.

OpenSER as a proxy can't even (per RFC3261) rewrite To or From, nor
will it hide topology (it simple adds Record-Route/Via).

* Session borders controllers are very ill-defined and mean different
things to different people.

-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com


From kristian.kielhofner at gmail.com  Fri Feb  5 10:20:01 2010
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Fri, 5 Feb 2010 13:20:01 -0500
Subject: [Freeswitch-users] Use of mod_Conference
In-Reply-To: <648878714.68411265393292092.JavaMail.root@mail-2.01.com>
References: <472380364.66881265393034269.JavaMail.root@mail-2.01.com>
	<648878714.68411265393292092.JavaMail.root@mail-2.01.com>
Message-ID: <2d9149cd1002051020n2035e423y701f9608b098d1c8@mail.gmail.com>

On Fri, Feb 5, 2010 at 1:08 PM,  <joel.sisko at iconverged.com> wrote:
> Group,
>
> I have a simple question I think about mod_Conference, does each conference
> room have to be created dynamically? My question is born from how Asterisk
> MeetMe room can work where you can have all the meetme rooms programmed
> statically in the meetme.conf file.
>
> What I am looking to do is rather than using the Asterisk meetme room
> application, I want to use FreeSwitch to provide this functionality. So I am
> trying to figure out what the integration would look like from a 10,000 foot
> view.
>
> Thanks for the help in advance.
>
> Joel

Joel,

  I think what you are looking for is "profile" support in
mod_conference, where you can specify most of the configuration
parameters for a specific conference statically:

http://wiki.freeswitch.org/wiki/Mod_conference#.3Cprofiles.3E

-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com


From anthony.minessale at gmail.com  Fri Feb  5 10:24:10 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 5 Feb 2010 12:24:10 -0600
Subject: [Freeswitch-users] A leg disconnects after uuid_bridge?
In-Reply-To: <fe67a3341002042041u1ee70b86u3bc91adbb6fb3d52@mail.gmail.com>
References: <fe67a3341002021934m548cde89p4d48381c5abe02a0@mail.gmail.com>
	<191c3a031002022108ta22287ev40fc36420abba2e5@mail.gmail.com>
	<fe67a3341002030112yeea379u90c22c482614c57@mail.gmail.com>
	<87f2f3b91002030605j7a1f9a02g665ba043c6ed7f7b@mail.gmail.com>
	<fe67a3341002040132n3aa2a897m60c5b9f1ca6b347f@mail.gmail.com>
	<0A571E14-BB58-4A3D-996A-9D5205668E85@freeswitch.org>
	<fe67a3341002040452v5468fd1fh96d1c5062091474b@mail.gmail.com>
	<191c3a031002040754r2987a03fo18ee2e1ceb239f6a@mail.gmail.com>
	<fe67a3341002042041u1ee70b86u3bc91adbb6fb3d52@mail.gmail.com>
Message-ID: <191c3a031002051024w1f484febj34a34d9e06421c52@mail.gmail.com>

try latest trunk i think your issue is fixed.


On Thu, Feb 4, 2010 at 10:41 PM, Nagalenoj H. <nagalenoj at gmail.com> wrote:

> Sorry., I couldn't understand its behavior.
>
> Let me ask the same question in this way.
>
> * hangup_after_bridge is set to false.
> * In outbound socket, first I answer the call.
> * When I do a bridge to a extension (1001), after 1001 disconnects the
> call. I am able to make another call.
>       sendmsg
>       call-command: execute
>       execute-app-name: bridge
>       execute-app-arg: user/1001
>
> * When I originate a call to extension (1001), after 1001 disconnects the
> call. I'm unable to make another call, because my session is also getting
> closed.
>       api originate user/1001 &park
>
>       Content-Type: api/response
>       Content-Length: 41
>
>      +OK 1fac17ce-120b-11df-a878-d9c7fbcf71c4
>
>
>       sendmsg
>       call-command: execute
>       execute-app-name: intercept
>       execute-app-arg: 1fac17ce-120b-11df-a878-d9c7fbcf71c4
>
> * In both the case, the call is getting bridged to an extension and
> hangup_after_bridge is false.
> * When bridge doesn't need any other variables to set to continue, why
> intercept needs a explicit park after bridge.?
>
> Hope, this has some clarity.,
>
>
> On Thu, Feb 4, 2010 at 9:24 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>>
>>
>>    1. 2010-02-04 14:30:09.574084 [DEBUG] switch_core_state_machine.c:354
>>    (sofia/internal/1010 at 192.168.1.222) State SOFT_EXECUTE going to sleep
>>    2. 2010-02-04 14:30:09.574084 [DEBUG] switch_core_state_machine.c:314
>>    (sofia/internal/1010 at 192.168.1.222) Running State Change CS_EXECUTE
>>    3. 2010-02-04 14:30:09.574084 [DEBUG] switch_core_state_machine.c:348
>>    (sofia/internal/1010 at 192.168.1.222) State EXECUTE
>>    4. 2010-02-04 14:30:09.574084 [DEBUG] mod_sofia.c:181 sofia/internal/
>>    1010 at 192.168.1.222 SOFIA EXECUTE
>>    5. 2010-02-04 14:30:09.574084 [DEBUG] switch_core_state_machine.c:159sofia/internal/
>>    1010 at 192.168.1.222 Standard EXECUTE
>>    6. 2010-02-04 14:30:09.574084 [NOTICE] switch_core_state_machine.c:187Hangup sofia/internal/
>>    1010 at 192.168.1.222 [CS_EXECUTE] [NORMAL_CLEARING]
>>
>>
>>
>> Your channel went back to EXECUTE as expected then it hungup because there
>> were no more instructions in your dial plan for it to execute.  So it is
>> working as expected.
>>
>> Consider using transfer_after_bridge variable or park_after bridge to make
>> it stay around when the call is over.
>>
>>
>>
>>
>> On Thu, Feb 4, 2010 at 6:52 AM, Nagalenoj H. <nagalenoj at gmail.com> wrote:
>>
>>> By using create_uuid. I've also tried without giving origination_uuid.
>>> But, the result is same.
>>>
>>> --
>>> Regards,
>>> Nagalenoj H.
>>>
>>>
>>> On Thu, Feb 4, 2010 at 5:42 PM, Brian West <brian at freeswitch.org> wrote:
>>>
>>>> Where are you getting this UUID?
>>>>
>>>> /b
>>>>
>>>> On Feb 4, 2010, at 3:32 AM, Nagalenoj H. wrote:
>>>>
>>>> > api originate
>>>> {origination_uuid=c6d2e0e2-0b2f-11df-9e84-fb15c3cd8565}user/1001 &park
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Regards,
> Nagalenoj H.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100205/3696c4c0/attachment.html 

From red.rain.seven at gmail.com  Fri Feb  5 10:36:13 2010
From: red.rain.seven at gmail.com (Henry Huang)
Date: Sat, 6 Feb 2010 02:36:13 +0800
Subject: [Freeswitch-users] Can Freeswitch and OpenSER Co-reside on Same
	Server?
In-Reply-To: <2d9149cd1002051011x3b4f548dqabe615ed5f991687@mail.gmail.com>
References: <C4DC856C75094358AEDBE3FA0E242741@greyhawk.tonecommander.com>
	<2d9149cd1002051011x3b4f548dqabe615ed5f991687@mail.gmail.com>
Message-ID: <59ad9ca11002051036w5817127ai66cad23046c100c1@mail.gmail.com>

Kristian:

Can you point me to the wiki link where it describes how to do the header
rewriting and number formatting and topology hiding?
I am also looking into OpenSIPS to be a session boarder controller, but if
freeswitch is already able to do all these, then I think it's easier for me
to stick with it since I already learn and do a lot with it already.

Thanks,

On Sat, Feb 6, 2010 at 2:11 AM, Kristian Kielhofner <
kristian.kielhofner at gmail.com> wrote:

> On Fri, Feb 5, 2010 at 12:16 PM, Jerry Richards
> <jerry.richards at teotech.com> wrote:
> > If I use OpenSER for a session border controller, does anyone see an
> issue
> > if it resides on the same server as Freeswitch?  So I would have a LAN
> and
> > WAN socket?  Are there any drawbacks (other than loading) to worry about?
> >
> > Thanks And Best Regards,
> > Jerry
> >
>
> You can use different IP addresses or ports.  I do this all of the time.
>
> I question why you are using OpenSER (OpenSIPS?) as a SBC.  FreeSWITCH
> is actually more well suited to most of the functions served by
> something called* a "session border controller".
>
> For example, FreeSWITCH in bypass media mode is a signaling only SBC
> where you can (cleanly) do the header rewriting, number formatting,
> and SIP topology hiding typically done by a SBC without touching the
> media.  Proxy media mode can do the same while proxying media
> (traversing NAT and hiding real RTP addresses).  FreeSWITCH in normal
> bridging mode can transcode, convert between different types of DTMF
> and do everything else mentioned above.
>
> OpenSER as a proxy can't even (per RFC3261) rewrite To or From, nor
> will it hide topology (it simple adds Record-Route/Via).
>
> * Session borders controllers are very ill-defined and mean different
> things to different people.
>
> --
> Kristian Kielhofner
> http://www.astlinux.org
> http://blog.krisk.org
> http://www.star2star.com
> http://www.submityoursip.com
> http://www.voalte.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Henry Huang
UniC Solution - Communication Unified
VoIP & Open Source software Consultant
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100206/675cf29a/attachment.html 

From costa.zikalala at gmail.com  Fri Feb  5 10:44:42 2010
From: costa.zikalala at gmail.com (Costa Zikalala)
Date: Fri, 5 Feb 2010 20:44:42 +0200
Subject: [Freeswitch-users] Difference Between "bridge" and "transfer"
In-Reply-To: <87f2f3b91002041632t19f17aefoaf11e048150816c5@mail.gmail.com>
References: <D123D09A29B942D988B3456528DD44AD@greyhawk.tonecommander.com> 
	<87f2f3b91002041632t19f17aefoaf11e048150816c5@mail.gmail.com>
Message-ID: <59daa2cd1002051044p1bee852ele8d0d972c303ea0e@mail.gmail.com>

Whilst on this subject, if I receive a call from PSTN and I 'bridge' it to
another PSTN extension, will the PSTN Provider charge me for the 'b-leg' or
will the original caller be charged the entire call?



On 5 February 2010 02:32, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Thu, Feb 4, 2010 at 3:56 PM, Jerry Richards <jerry.richards at teotech.com
> > wrote:
>
>> What is the difference between "bridge" and "transfer"?  I'm looking at
>> the
>> demo IVRs.
>>
>>
> bridge will connect two endpoints together while transfer sends the
> endpoint back through the dialplan again...
>
> -MC
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100205/9ac3de36/attachment-0001.html 

From joel.sisko at iconverged.com  Fri Feb  5 10:59:31 2010
From: joel.sisko at iconverged.com (Joel Sisko)
Date: Fri, 5 Feb 2010 12:59:31 -0600 (CST)
Subject: [Freeswitch-users] Use of mod_Conference
In-Reply-To: <2d9149cd1002051020n2035e423y701f9608b098d1c8@mail.gmail.com>
Message-ID: <875390421.80391265396371614.JavaMail.root@mail-2.01.com>

Kristian,

Thank you for the quick reply.

The profile looks like it meant to be specific to any or all conferences that are created (a template).

I think I have a different way to clarify my question:

Lets assume on my Asterisk system that someone wants to enter conference room 201. I want to pass them into conference room 201 on my FreeSwitch server, so to over simplify I will pass the caller to conference201 at myFreeSwitch.server.com, but if  the conference room has not been created then there is nothing to transfer to.(?) So is there a way that to use  a static configuration so the conference room is already created? Or must I create that conference room prior to my first request for that room, then transfer the caller after the creation of the conference room?

Joel
----- Original Message -----
From: "Kristian Kielhofner" <kristian.kielhofner at gmail.com>
To: freeswitch-users at lists.freeswitch.org
Sent: Friday, February 5, 2010 10:20:01 AM GMT -08:00 US/Canada Pacific
Subject: Re: [Freeswitch-users] Use of mod_Conference

On Fri, Feb 5, 2010 at 1:08 PM,  <joel.sisko at iconverged.com> wrote:
> Group,
>
> I have a simple question I think about mod_Conference, does each conference
> room have to be created dynamically? My question is born from how Asterisk
> MeetMe room can work where you can have all the meetme rooms programmed
> statically in the meetme.conf file.
>
> What I am looking to do is rather than using the Asterisk meetme room
> application, I want to use FreeSwitch to provide this functionality. So I am
> trying to figure out what the integration would look like from a 10,000 foot
> view.
>
> Thanks for the help in advance.
>
> Joel

Joel,

  I think what you are looking for is "profile" support in
mod_conference, where you can specify most of the configuration
parameters for a specific conference statically:

http://wiki.freeswitch.org/wiki/Mod_conference#.3Cprofiles.3E

-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


From kristian.kielhofner at gmail.com  Fri Feb  5 11:12:53 2010
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Fri, 5 Feb 2010 14:12:53 -0500
Subject: [Freeswitch-users] Can Freeswitch and OpenSER Co-reside on Same
	Server?
In-Reply-To: <59ad9ca11002051036w5817127ai66cad23046c100c1@mail.gmail.com>
References: <C4DC856C75094358AEDBE3FA0E242741@greyhawk.tonecommander.com>
	<2d9149cd1002051011x3b4f548dqabe615ed5f991687@mail.gmail.com>
	<59ad9ca11002051036w5817127ai66cad23046c100c1@mail.gmail.com>
Message-ID: <2d9149cd1002051112y4c82bb77m25aa43d70c1005d3@mail.gmail.com>

On Fri, Feb 5, 2010 at 1:36 PM, Henry Huang <red.rain.seven at gmail.com> wrote:
> Kristian:
>
> Can you point me to the wiki link where it describes how to do the header
> rewriting and number formatting and topology hiding?
> I am also looking into OpenSIPS to be a session boarder controller, but if
> freeswitch is already able to do all these, then I think it's easier for me
> to stick with it since I already learn and do a lot with it already.
>
> Thanks,
>

Henry,

  By default (when using bridge) FreeSWITCH will generate a new leg
with a fresh set of headers.  Various channel variables can be used to
influence the values of some of these:

Request URI: based on bridge string (411 at sip.provider.com) - can also
include transport, port, uri params, etc
To: same as Request URI (possible minus some of the params)
From: Determined by gateway config (if used: use-callerid-in-from) and
effective_caller_id_number/effective_caller_id_name
RPID/PAI: sip_cid_type

  The other params will be taken from the Sofia config (session
timers, etc).  Use multiple profiles for internal and external
networks (just like in the samples).

  FreeSWITCH does topology hiding by default - as a B2BUA (regardless
of "mode") it will do topology hiding by creating a new channel/leg.
None of the IP addresses etc, from the original channel are visible.
Note that this doesn't count the SDP if you are using bypass media or
proxy media.  If you want *full* SBC style topology hiding with media
you can't use these modes but you'll pay for it in performance.

  Unless you use the uac/uas modules and/or some textops based
manipulation in OpenSER, all OpenSER can do is copy most of the
headers/body, add Record-Route and Via headers, and forward the
message to the next hop while leaving all IP addresses, etc intact.
OpenSER/OpenSIPS can be a fairly general purpose SIP server but it's
main function is a fairly strict RFC 3261 compliant proxy.

  I should also point out that OpenSIPS does have a B2BUA module.  I
myself would much rather just use FreeSWITCH.  You know - the best
tool for the job.  They're both EXCELLENT pieces of software and
between the two of them you can build incredible VoIP solutions and
networks.

  I use FreeSWITCH to interface with each of my carriers.  FreeSWITCH
runs on the same machine as our main SIP proxy running in
bypass_media.  The SIP proxy (OpenSER) handles all of the requests for
our servers/customers (servers we provision and control) while
FreeSWITCH (usually in bypass media) interfaces with each of our
carriers to make everyone happy at the signaling level.  I have a
profile for our network and a profile for each carrier.  No need to
worry about different caller id formats, number formats (e.164),
transports, caller id, etc.  On one machine FreeSWITCH regularly does
over 1200 channels using about %20 CPU.  It's a four year old Dell
1850 :).

-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com


From lists at redbonez.net  Fri Feb  5 11:21:17 2010
From: lists at redbonez.net (Adam Ford)
Date: Fri, 5 Feb 2010 12:21:17 -0700
Subject: [Freeswitch-users] Looking for some good/cheap desktop phones
In-Reply-To: <d361ca7b1002050507n5e564621oa6cf7ace005dd5b8@mail.gmail.com>
References: <3f2e071b5aea5e11bb37a37e435a33b7@mail.gmail.com>	<8e9d67561002042236s434e0643od322551de8720be1@mail.gmail.com>	<409D0ACF-ADA4-4D02-8B7E-71EC1B1BAFBC@freeswitch.org>	<8e9d67561002042301v76129abfj3dcbc70ed073752c@mail.gmail.com>
	<d361ca7b1002050507n5e564621oa6cf7ace005dd5b8@mail.gmail.com>
Message-ID: <00ae01caa698$65b570f0$312052d0$@net>

I just picked up old model Polycoms.  You can get the IP301's for ~$60-70
new and the IP501s for ~$100 new.  They don't have some of the fancier
features of the new Polycoms, but they carry the same quality and
configurability(with the exception of NAT).

 

-Adam

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Yehavi
Bourvine
Sent: Friday, February 05, 2010 6:07 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Looking for some good/cheap desktop phones

 

>From my experience Polycom and SNOM are expensive but give you what you
need. Polycom is more intutive to the users but more cumbersome for the
manager to deploy; SNOM is somewhat less intuitive to the user but
everything can be set via the WEB interface.

 

If you talk about 4-5 phones, then probably SNOM is the choice. It also
depends about the specific functions you want to use. I our specific
environment (high use of BLF and shared lines) Polycom wins because it
handles these functions just as the user expects.

 

I did not try Aastra so cannot testify. We did test Yealink, Thomson,
Asterphone, SipTip and maybe others I forgot. Cisco also seems good but
Cisco does not supply the required socumentation to make them fully working.

 

                    Regards, __Yehavi:

2010/2/5 ????? ??????? <tim at novion.ru>

Sure, those phones do not deliver superior usability, but they at
least give the best sound among budget models.


2010/2/5 Brian West <brian at freeswitch.org>:

> And all of those are awful phones.  They don't even make good paper
weights.
>
> You can't have good and cheap in the same sentence when talking about VoIP
phones.  You have to take your pick between quality (good) and price (cheap)
you can't have both at once.
>
> /b
>
>
> On Feb 5, 2010, at 12:36 AM, ????? ??????? wrote:
>
>> Have a look at Yealink (Skypemate) and Fanvill
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org <http://www.freeswitch.org/> 
>

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org <http://www.freeswitch.org/> 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100205/4417ceba/attachment.html 

From msc at freeswitch.org  Fri Feb  5 11:25:10 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 5 Feb 2010 11:25:10 -0800
Subject: [Freeswitch-users] Difference Between "bridge" and "transfer"
In-Reply-To: <59daa2cd1002051044p1bee852ele8d0d972c303ea0e@mail.gmail.com>
References: <D123D09A29B942D988B3456528DD44AD@greyhawk.tonecommander.com>
	<87f2f3b91002041632t19f17aefoaf11e048150816c5@mail.gmail.com>
	<59daa2cd1002051044p1bee852ele8d0d972c303ea0e@mail.gmail.com>
Message-ID: <87f2f3b91002051125u31dbf290h531d34d078c0f42e@mail.gmail.com>

On Fri, Feb 5, 2010 at 10:44 AM, Costa Zikalala <costa.zikalala at gmail.com>wrote:

> Whilst on this subject, if I receive a call from PSTN and I 'bridge' it to
> another PSTN extension, will the PSTN Provider charge me for the 'b-leg' or
> will the original caller be charged the entire call?
>

That depends... is the "other" leg an outbound call? Is the other leg an
inbound call to a toll-free number?
-MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100205/b1f33908/attachment.html 

From kristian.kielhofner at gmail.com  Fri Feb  5 11:25:11 2010
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Fri, 5 Feb 2010 14:25:11 -0500
Subject: [Freeswitch-users] Use of mod_Conference
In-Reply-To: <875390421.80391265396371614.JavaMail.root@mail-2.01.com>
References: <2d9149cd1002051020n2035e423y701f9608b098d1c8@mail.gmail.com>
	<875390421.80391265396371614.JavaMail.root@mail-2.01.com>
Message-ID: <2d9149cd1002051125u20eef2c8jf9ce39a0223238b@mail.gmail.com>

On Fri, Feb 5, 2010 at 1:59 PM, Joel Sisko <joel.sisko at iconverged.com> wrote:
> Kristian,
>
> Thank you for the quick reply.
>
> The profile looks like it meant to be specific to any or all conferences that are created (a template).
>
> I think I have a different way to clarify my question:
>
> Lets assume on my Asterisk system that someone wants to enter conference room 201. I want to pass them into conference room 201 on my FreeSwitch server, so to over simplify I will pass the caller to conference201 at myFreeSwitch.server.com, but if ?the conference room has not been created then there is nothing to transfer to.(?) So is there a way that to use ?a static configuration so the conference room is already created? Or must I create that conference room prior to my first request for that room, then transfer the caller after the creation of the conference room?
>
> Joel

Joel,

  Let me try to explain using a sample taken from the default dialplan:

    <extension name="conference201">
      <condition field="destination_number" expression="^conference201$">
        <action application="answer"/>
        <action application="conference" data="201 at default"/>
      </condition>
    </extension>

  (Yes I could use variables here but I wanted to keep it simple).

  This would create the conference with the first caller that called
in.  The conference number would be 201 and it would use the "default"
profile from the conference configuration.  Any number of conferences
can be defined in the dialplan with different settings, pins, etc
depending on the profile used and the arguments passed to conference
via data=.

-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com


From jerry.richards at teotech.com  Fri Feb  5 11:38:09 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Fri, 5 Feb 2010 11:38:09 -0800
Subject: [Freeswitch-users] Presence PUBLISH Not Updating After Softphone
	OffLine Then Available
Message-ID: <31FEE9072F104AE18B26812978FCA341@greyhawk.tonecommander.com>

I found a scenario where presence status is not distributed to subscribers.
This is using the latest changes (as of Feb 03, 2010).  The scenario
follows:

1) Register two Bria softphones (A and B), which each have the other as a
contact.
2) Set softphone B's presence status to 'Appear Offline'.
3) Softphone A correctly reflects contact B is offline.
4) Set softphone B's presence status to 'Available'.
5) ******* There is no change to contact B's status at softphone A *******

I posted a log at http://pastebin.freeswitch.org/12054.  At line 773, there
is an error when FS is processing the PUBLISH from softphone B:

773.2010-02-05 10:29:21.254221 [ERR] sofia_presence.c:674 DUMP PRESENCE SQL:

I did notice that after about 30 minutes, softphone B's status gets
reflected at softphone A.

Thanks and Best Regards,
Jerry



From joel.sisko at iconverged.com  Fri Feb  5 11:39:59 2010
From: joel.sisko at iconverged.com (Joel Sisko)
Date: Fri, 5 Feb 2010 13:39:59 -0600 (CST)
Subject: [Freeswitch-users] Use of mod_Conference
In-Reply-To: <2d9149cd1002051125u20eef2c8jf9ce39a0223238b@mail.gmail.com>
Message-ID: <1852904235.89881265398799084.JavaMail.root@mail-2.01.com>

Thanks for the insight.

Joel
----- Original Message -----
From: "Kristian Kielhofner" <kristian.kielhofner at gmail.com>
To: freeswitch-users at lists.freeswitch.org
Sent: Friday, February 5, 2010 11:25:11 AM GMT -08:00 US/Canada Pacific
Subject: Re: [Freeswitch-users] Use of mod_Conference

On Fri, Feb 5, 2010 at 1:59 PM, Joel Sisko <joel.sisko at iconverged.com> wrote:
> Kristian,
>
> Thank you for the quick reply.
>
> The profile looks like it meant to be specific to any or all conferences that are created (a template).
>
> I think I have a different way to clarify my question:
>
> Lets assume on my Asterisk system that someone wants to enter conference room 201. I want to pass them into conference room 201 on my FreeSwitch server, so to over simplify I will pass the caller to conference201 at myFreeSwitch.server.com, but if ?the conference room has not been created then there is nothing to transfer to.(?) So is there a way that to use ?a static configuration so the conference room is already created? Or must I create that conference room prior to my first request for that room, then transfer the caller after the creation of the conference room?
>
> Joel

Joel,

  Let me try to explain using a sample taken from the default dialplan:

    <extension name="conference201">
      <condition field="destination_number" expression="^conference201$">
        <action application="answer"/>
        <action application="conference" data="201 at default"/>
      </condition>
    </extension>

  (Yes I could use variables here but I wanted to keep it simple).

  This would create the conference with the first caller that called
in.  The conference number would be 201 and it would use the "default"
profile from the conference configuration.  Any number of conferences
can be defined in the dialplan with different settings, pins, etc
depending on the profile used and the arguments passed to conference
via data=.

-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


From jerry.richards at teotech.com  Fri Feb  5 11:40:06 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Fri, 5 Feb 2010 11:40:06 -0800
Subject: [Freeswitch-users] Can Freeswitch and OpenSER Co-reside on
	SameServer?
In-Reply-To: <2d9149cd1002051011x3b4f548dqabe615ed5f991687@mail.gmail.com>
References: <C4DC856C75094358AEDBE3FA0E242741@greyhawk.tonecommander.com>
	<2d9149cd1002051011x3b4f548dqabe615ed5f991687@mail.gmail.com>
Message-ID: <0203851F538F44B18BFD40EC383EBEBF@greyhawk.tonecommander.com>

So do you build your server with two FS instances running?  One as the SBC
and one as Proxy/PBX?

Thanks,
Jerry
 

-----Original Message-----
From: Kristian Kielhofner [mailto:kristian.kielhofner at gmail.com] 
Sent: Friday, February 05, 2010 10:12 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Can Freeswitch and OpenSER Co-reside on
SameServer?

On Fri, Feb 5, 2010 at 12:16 PM, Jerry Richards <jerry.richards at teotech.com>
wrote:
> If I use OpenSER for a session border controller, does anyone see an 
> issue if it resides on the same server as Freeswitch? ?So I would have 
> a LAN and WAN socket? ?Are there any drawbacks (other than loading) to
worry about?
>
> Thanks And Best Regards,
> Jerry
>

You can use different IP addresses or ports.  I do this all of the time.

I question why you are using OpenSER (OpenSIPS?) as a SBC.  FreeSWITCH is
actually more well suited to most of the functions served by something
called* a "session border controller".

For example, FreeSWITCH in bypass media mode is a signaling only SBC where
you can (cleanly) do the header rewriting, number formatting, and SIP
topology hiding typically done by a SBC without touching the media.  Proxy
media mode can do the same while proxying media (traversing NAT and hiding
real RTP addresses).  FreeSWITCH in normal bridging mode can transcode,
convert between different types of DTMF and do everything else mentioned
above.

OpenSER as a proxy can't even (per RFC3261) rewrite To or From, nor will it
hide topology (it simple adds Record-Route/Via).

* Session borders controllers are very ill-defined and mean different things
to different people.

--
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com





From lon at kickasspixels.com  Fri Feb  5 11:42:50 2010
From: lon at kickasspixels.com (Lon Baker)
Date: Fri, 5 Feb 2010 11:42:50 -0800
Subject: [Freeswitch-users] Testing Config Changes
Message-ID: <5d3e0dc61002051142h65c3f533t9524ef0510af02fd@mail.gmail.com>

I'm looking for the best practice for testing configuration changes on
a live Fresswitch server.

Is it best to use reloadxml in the cli? Will that alert us to issues
with syntax and other errors without bringing down the server?

Lon


From kristian.kielhofner at gmail.com  Fri Feb  5 11:50:00 2010
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Fri, 5 Feb 2010 14:50:00 -0500
Subject: [Freeswitch-users] Can Freeswitch and OpenSER Co-reside on
	SameServer?
In-Reply-To: <0203851F538F44B18BFD40EC383EBEBF@greyhawk.tonecommander.com>
References: <C4DC856C75094358AEDBE3FA0E242741@greyhawk.tonecommander.com>
	<2d9149cd1002051011x3b4f548dqabe615ed5f991687@mail.gmail.com>
	<0203851F538F44B18BFD40EC383EBEBF@greyhawk.tonecommander.com>
Message-ID: <2d9149cd1002051150i3fba0945s4332af51d261274c@mail.gmail.com>

On Fri, Feb 5, 2010 at 2:40 PM, Jerry Richards
<jerry.richards at teotech.com> wrote:
> So do you build your server with two FS instances running? ?One as the SBC
> and one as Proxy/PBX?
>
> Thanks,
> Jerry

Jerry,

  No.  One instance of FreeSWITCH and one instance of OpenSER.  As I
said, just make sure they use separate IPs and/or ports.  I prefer
standard ports and separate IPs because then (in the future) if I need
to split them (scaling, redundancy, etc) all I have to do is bring up
the second IP on a different host and move the software/config.

-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com


From brian at freeswitch.org  Fri Feb  5 11:53:30 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 5 Feb 2010 13:53:30 -0600
Subject: [Freeswitch-users] Testing Config Changes
In-Reply-To: <5d3e0dc61002051142h65c3f533t9524ef0510af02fd@mail.gmail.com>
References: <5d3e0dc61002051142h65c3f533t9524ef0510af02fd@mail.gmail.com>
Message-ID: <14964F2E-0D87-4E40-8CB9-E7ACA815FB34@freeswitch.org>

This all depends on what all you're changing.. some things you can't change no matter how hard you try.

/b

On Feb 5, 2010, at 1:42 PM, Lon Baker wrote:

> I'm looking for the best practice for testing configuration changes on
> a live Fresswitch server.
> 
> Is it best to use reloadxml in the cli? Will that alert us to issues
> with syntax and other errors without bringing down the server?
> 
> Lon



From kristian.kielhofner at gmail.com  Fri Feb  5 11:54:30 2010
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Fri, 5 Feb 2010 14:54:30 -0500
Subject: [Freeswitch-users] Use of mod_Conference
In-Reply-To: <1852904235.89881265398799084.JavaMail.root@mail-2.01.com>
References: <2d9149cd1002051125u20eef2c8jf9ce39a0223238b@mail.gmail.com>
	<1852904235.89881265398799084.JavaMail.root@mail-2.01.com>
Message-ID: <2d9149cd1002051154w507c31e9y8c7dfc9c64956542@mail.gmail.com>

On Fri, Feb 5, 2010 at 2:39 PM, Joel Sisko <joel.sisko at iconverged.com> wrote:
> Thanks for the insight.
>
> Joel

No problem.  Maybe we'll be able to put this stuff in writing someday
if O'Reilly ever publishes a FreeSWITCH book...  If you recall, you
and I worked on "VoIP Hacks" and "The Future of Asterisk".

P.S. - Ted Wallingford friend requested me on Facebook today.  What
are the chances? ;)

-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com


From rupa at rupa.com  Fri Feb  5 11:54:40 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Fri, 5 Feb 2010 13:54:40 -0600
Subject: [Freeswitch-users] Looking for some good/cheap desktop phones
In-Reply-To: <00ae01caa698$65b570f0$312052d0$@net>
References: <3f2e071b5aea5e11bb37a37e435a33b7@mail.gmail.com>
	<8e9d67561002042236s434e0643od322551de8720be1@mail.gmail.com>
	<409D0ACF-ADA4-4D02-8B7E-71EC1B1BAFBC@freeswitch.org>
	<8e9d67561002042301v76129abfj3dcbc70ed073752c@mail.gmail.com>
	<d361ca7b1002050507n5e564621oa6cf7ace005dd5b8@mail.gmail.com>
	<00ae01caa698$65b570f0$312052d0$@net>
Message-ID: <b39273531002051154h5132faa9h9fb024ebdfb15b77@mail.gmail.com>

Also be aware they are EOL so no new firmware for them.  The 301 also is not
backlit which can be a pain depending on environment.

2010/2/5 Adam Ford <lists at redbonez.net>

>  I just picked up old model Polycoms.  You can get the IP301's for ~$60-70
> new and the IP501s for ~$100 new.  They don't have some of the fancier
> features of the new Polycoms, but they carry the same quality and
> configurability(with the exception of NAT).
>
>
>
> -Adam
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Yehavi
> Bourvine
> *Sent:* Friday, February 05, 2010 6:07 AM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Looking for some good/cheap desktop
> phones
>
>
>
> From my experience Polycom and SNOM are expensive but give you what you
> need. Polycom is more intutive to the users but more cumbersome for the
> manager to deploy; SNOM is somewhat less intuitive to the user but
> everything can be set via the WEB interface.
>
>
>
> If you talk about 4-5 phones, then probably SNOM is the choice. It also
> depends about the specific functions you want to use. I our specific
> environment (high use of BLF and shared lines) Polycom wins because it
> handles these functions just as the user expects.
>
>
>
> I did not try Aastra so cannot testify. We did test Yealink, Thomson,
> Asterphone, SipTip and maybe others I forgot. Cisco also seems good but
> Cisco does not supply the required socumentation to make them fully working.
>
>
>
>                     Regards, __Yehavi:
>
> 2010/2/5 ????? ??????? <tim at novion.ru>
>
> Sure, those phones do not deliver superior usability, but they at
> least give the best sound among budget models.
>
>
> 2010/2/5 Brian West <brian at freeswitch.org>:
>
> > And all of those are awful phones.  They don't even make good paper
> weights.
> >
> > You can't have good and cheap in the same sentence when talking about
> VoIP phones.  You have to take your pick between quality (good) and price
> (cheap) you can't have both at once.
> >
> > /b
> >
> >
> > On Feb 5, 2010, at 12:36 AM, ????? ??????? wrote:
> >
> >> Have a look at Yealink (Skypemate) and Fanvill
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100205/260ede3d/attachment-0001.html 

From rupa at rupa.com  Fri Feb  5 11:58:13 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Fri, 5 Feb 2010 13:58:13 -0600
Subject: [Freeswitch-users] Testing Config Changes
In-Reply-To: <5d3e0dc61002051142h65c3f533t9524ef0510af02fd@mail.gmail.com>
References: <5d3e0dc61002051142h65c3f533t9524ef0510af02fd@mail.gmail.com>
Message-ID: <b39273531002051158y18ea21a5ka5b174927778a24f@mail.gmail.com>

it'll alert you to syntax errors, but pretty much any other mistake you'll
find out when calls start failing or misbehaving.  Better to have a staging
server where you test stuff out (with a test plan!).

On Fri, Feb 5, 2010 at 1:42 PM, Lon Baker <lon at kickasspixels.com> wrote:

> I'm looking for the best practice for testing configuration changes on
> a live Fresswitch server.
>
> Is it best to use reloadxml in the cli? Will that alert us to issues
> with syntax and other errors without bringing down the server?
>
> Lon
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
-Rupa
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100205/ff170705/attachment.html 

From tim at novion.ru  Fri Feb  5 12:02:38 2010
From: tim at novion.ru (Timur Valishev)
Date: Fri, 5 Feb 2010 23:02:38 +0300
Subject: [Freeswitch-users] No reINVITE when bridging two sessions from
	JavaScript with bypass_media_after_bridge=true
In-Reply-To: <B66289FF-C2A9-4AC4-A56C-66E51B4726F4@freeswitch.org>
References: <8e9d67561002041147q6c135546u1f7e1887529a916d@mail.gmail.com>
	<3D739D7F-F5C6-4266-B04E-9413DCFDA6AF@freeswitch.org>
	<8e9d67561002050018q24e3a806kc8807b55268e759b@mail.gmail.com>
	<B66289FF-C2A9-4AC4-A56C-66E51B4726F4@freeswitch.org>
Message-ID: <8e9d67561002051202qeaf1551v8fdd29565f8c02fd@mail.gmail.com>

I think we are on the right way) still does not work, but there is hope)

First of all, this script does not produce any reinvite either (even if
replace bypass_media to bypass_media_after_bridge, or set bypass_media only
on one channel):

<<<<<<<<<<<<<<<<<<<<<<<<<<<<<
session = new
Session("{bypass_media=true,ignore_early_media=true,hangup_after_bridge=true}user/1001");
session2 = new
Session("{bypass_media=true,ignore_early_media=true} user/1001");
bridge(session, session2);
>>>>>>>>>>>>>>>>>>>>>>>>>>>>>

BUT! if I run the following script:
<<<<<<<<<<<<<<<<<<<<<<<<<<<<<
session = new
Session("{bypass_media=true,ignore_early_media=true,hangup_after_bridge=true}
user/1001");
session2 = new
Session("{bypass_media=true,ignore_early_media=true}user/1001");
>>>>>>>>>>>>>>>>>>>>>>>>>>>>>

And then manually type in the console
uuid_media off <uuid of my session>

- then I get the reINVITE!

BUT! When I try to write it to the script:
<<<<<<<<<<<<<<<<<<<<<<<<<<<<<
session = new
Session("{bypass_media=true,ignore_early_media=true,hangup_after_bridge=true}sofia/external/
timwork at novion.ru");
session2 = new
Session("{bypass_media=true,ignore_early_media=true}sofia/external/
timwork at novion.ru");
bridge(session, session2);
apiExecute('uuid_media off '+session.uuid); // <-- this line is not
executed, because bridge hangs up untill BYE
>>>>>>>>>>>>>>>>>>>>>>>>>>>>>

the last line is not executed, because bridge hangs up untill BYE

Then I've tried to do like this:
<<<<<<<<<<<<<<<<<<<<<<<<<<<<<
session = new
Session("{bypass_media=true,ignore_early_media=true,hangup_after_bridge=true}user/1001");
session2 = new
Session("{bypass_media=true,ignore_early_media=true}user/1001");

session.setAutoHangup(false)
session2.setAutoHangup(false)

apiExecute("uuid_bridge "+session.uuid+" "+session2.uuid);
apiExecute('uuid_media off '+session.uuid);
>>>>>>>>>>>>>>>>>>>>>>>>>>>>>

But sessions do not get bridged -( Even if I insert session.ready() after
each call.

Any ideas on how to call the functions correctly to get the reINVITE?

Best regards,
Timur Valishev

2010/2/5 Brian West <brian at freeswitch.org>

> set it inside each of the {} for each session you create its not set after
> the fact the call is up already...  you're setting it too late.
>
> you an also issue uuid_media off <uuid>
>
> /b
>
> On Feb 5, 2010, at 2:18 AM, Timur Valishev wrote:
>
> I've modified my script to make sure: <<<<<<<<<<<<<<<<<<<<<<<<<<<<,
> *session = new Session(*
> *"{ignore_early_media=true,hangup_after_bridge=true}sofia/external/
> timwork at novion.ru"*
> *);*
> *session2 = new Session(*
> *"{ignore_early_media=true}sofia/external/timwork at novion.ru"*
> *);*
> *session.setVariable('bypass_media', 'true');*
> *session2.setVariable('bypass_media', 'true');*
> *bridge(session, session2);*
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100205/3bc58a3a/attachment.html 

From lon at kickasspixels.com  Fri Feb  5 12:15:37 2010
From: lon at kickasspixels.com (Lon Baker)
Date: Fri, 5 Feb 2010 12:15:37 -0800
Subject: [Freeswitch-users] Testing Config Changes
In-Reply-To: <b39273531002051158y18ea21a5ka5b174927778a24f@mail.gmail.com>
References: <5d3e0dc61002051142h65c3f533t9524ef0510af02fd@mail.gmail.com> 
	<b39273531002051158y18ea21a5ka5b174927778a24f@mail.gmail.com>
Message-ID: <5d3e0dc61002051215i619d7cb2t6293edbe6397e1bc@mail.gmail.com>

I agree with the staging and test plan, this is something we do.

I want to implement a procedure on the production servers to help
insure that we are as bullet proof as possible.

On Fri, Feb 5, 2010 at 11:58 AM, Rupa Schomaker <rupa at rupa.com> wrote:
> it'll alert you to syntax errors, but pretty much any other mistake you'll
> find out when calls start failing or misbehaving. ?Better to have a staging
> server where you test stuff out (with a test plan!).
>
> On Fri, Feb 5, 2010 at 1:42 PM, Lon Baker <lon at kickasspixels.com> wrote:
>>
>> I'm looking for the best practice for testing configuration changes on
>> a live Fresswitch server.
>>
>> Is it best to use reloadxml in the cli? Will that alert us to issues
>> with syntax and other errors without bringing down the server?
>>
>> Lon
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From shyjuk at live.com  Fri Feb  5 12:33:19 2010
From: shyjuk at live.com (Shyju Kanaprath)
Date: Sat, 6 Feb 2010 02:03:19 +0530
Subject: [Freeswitch-users] Looking for some good/cheap desktop phones
In-Reply-To: <b39273531002051154h5132faa9h9fb024ebdfb15b77@mail.gmail.com>
References: <3f2e071b5aea5e11bb37a37e435a33b7@mail.gmail.com>,
	<8e9d67561002042236s434e0643od322551de8720be1@mail.gmail.com>,
	<409D0ACF-ADA4-4D02-8B7E-71EC1B1BAFBC@freeswitch.org>,
	<8e9d67561002042301v76129abfj3dcbc70ed073752c@mail.gmail.com>,
	<d361ca7b1002050507n5e564621oa6cf7ace005dd5b8@mail.gmail.com>,
	<00ae01caa698$65b570f0$312052d0$@net>,
	<b39273531002051154h5132faa9h9fb024ebdfb15b77@mail.gmail.com>
Message-ID: <SNT119-W1223C808A5E092C7E2A590B5540@phx.gbl>


Grandstream is also a good choice.. GXP2000 has got 5-6 blf/speed dial buttons and is very user friendly.
Configuring through web interface is also very easy.

Regards,
Shyju
 		 	   		  
_________________________________________________________________
Post free property ads on Yello Classifieds now! www.yello.in
http://ss1.richmedia.in/recurl.asp?pid=219
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100206/0be9922e/attachment.html 

From Prometheus001 at gmx.net  Fri Feb  5 12:58:51 2010
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Fri, 05 Feb 2010 21:58:51 +0100
Subject: [Freeswitch-users] mod_skypiax - help needed with
 modified	snd-dummy (one way audio)
In-Reply-To: <7b197bef1002050536j7ea09684v913434296ae52b97@mail.gmail.com>
References: <4B60555B.2020004@gmx.net>	<7b197bef1001271215o64918f56s3d16e51528fc5f66@mail.gmail.com>
	<4B61ECE0.10409@gmx.net>	<7b197bef1001281210u43a907edi13afe794cf0e1a2e@mail.gmail.com>
	<4B61FCAB.5040707@gmx.net>	<7b197bef1001281341h7ee2c58fj79b3886630901f29@mail.gmail.com>
	<4B62C962.7000601@gmx.net>	<8213d6071001290403g2c08fe39gd92eb0446bb6485a@mail.gmail.com>
	<7b197bef1001290520x37179cb1o63273410e9ef5bc@mail.gmail.com>
	<4B6C1CCB.4080606@gmx.net>
	<7b197bef1002050536j7ea09684v913434296ae52b97@mail.gmail.com>
Message-ID: <4B6C868B.3040406@gmx.net>

Hello Giovanni,

I am now at the point to install Skype. But there is only an Intrepid
version available (no 8.04 version).
The current verison crashed on 8.04x because of dbus error.
    process 8408: D-Bus library appears to be incorrectly set up; failed
to read machine uuid: Failed to open "/var/lib/dbus/machine-id": No such
file or directory
    See the manual page for dbus-uuidgen to correct this issue.
    /usr/bin/skype: symbol lookup error: /usr/bin/skype: undefined
symbol: _ZN10QBoxLayout13addSpacerItemEP11QSpacerIte

Any idea where I can download the older version for 8.04?

Best regards
Peter


Giovanni Maruzzelli schrieb:
> Ciao Peter,
>
> I would use ubuntu 8.04 (hardy) LTS server 64bit or CentOS 5.4 64bit.
>
> -giovanni
>
> On Fri, Feb 5, 2010 at 2:27 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>   
>> Hello Giovanni,
>>
>> as I couldn't even get skype again working again with the standard alsa
>> driver, I would like to setup the machine from scratch based on a
>> working machine.
>> The latest errors I received from Skype was:
>>  snd_pcm_avail_update() returned a value that is exceptionally large:
>> 715706624 bytes (3727638 ms).
>>  Most likely this is a bug in the ALSA driver. Please report this issue
>> to the ALSA developers.
>> I think that may be the reason for one-way-audio.
>>
>> For setting up my machine from scratch, please advise:
>> - which OS you are you using und recommending exactly?
>> - I would like to use 64bit OS in order to use 8GB of memory, does this
>> work?
>> - any other hints?
>>
>> Best regards
>> Peter
>>
>> Giovanni Maruzzelli schrieb:
>>     
>>> Peter,
>>>
>>> Can you connect on IRC (irc.freenode.net #freeswitch)? I'm gmaruzz there.
>>>
>>> Can you restate your problems? I've lost connection :)
>>>
>>> with snd-dummy custom you can create *one only* snd-dummy instance, so
>>> *one only* fake soundcard. If you create more, will not work. But with
>>> that one fake soundcard you can use 64 skype client instances, all
>>> with the same soundcard hardware device (hw:n).
>>>
>>> with original snd-dummy you can create a max of 8 instances, so 8 fake
>>> soundcards, and with each fake soundcard you can  use a max of 8 skype
>>> client instances.
>>>
>>> use the hardware devices, not the default devices (use the "hw:n")
>>>
>>> -giovanni
>>>
>>> On Fri, Jan 29, 2010 at 1:03 PM, Ghulam Mustafa <mustafa.pk at gmail.com> wrote:
>>>
>>>       
>>>> did you enable debug mode while compiling custom snd-dummy? if  yes
>>>> try re-compiling with debug mode disabled.
>>>>
>>>> -m
>>>>
>>>> On Fri, Jan 29, 2010 at 4:41 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>
>>>>         
>>>>> I now reinstalled the original sound drivers
>>>>> Unfortunaltely the sound problems remain, not that worse but they are there:
>>>>> Audio is still (almost) one way. Almost means:
>>>>>
>>>>>    * SIP -> Skype ok
>>>>>    * Skype=> SIP I hear only some scratching on very loud audio
>>>>>
>>>>> Could it be a volume problem? But snd-dummy should have no volume
>>>>> properties, right?
>>>>>
>>>>> Best regards
>>>>> Peter
>>>>>
>>>>>
>>>>> Giovanni Maruzzelli schrieb:
>>>>>
>>>>>           
>>>>>> with three instances you will assign the hw:0 device to skype client
>>>>>> 0...7, hw:1 to skype client 8...15 and hw:2 to skype client 16...23.
>>>>>> Must work. Pay attention to assign the same device name to all devices
>>>>>> needed by a skype instance (sound devices window): playback, capture
>>>>>> AND ring.
>>>>>>
>>>>>> Or maybe is a bug of ALSA on Debian...
>>>>>>
>>>>>> -giovanni
>>>>>>
>>>>>> On Thu, Jan 28, 2010 at 10:07 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>>
>>>>>>
>>>>>>             
>>>>>>> I crated 3 instances of snd-dummy, this worked. I assigned then Instance
>>>>>>> #2 to the Skype accounts. Still no sound.
>>>>>>> On the frist call there is one way audio, on the following calls there
>>>>>>> is no audio at all.
>>>>>>> This is weird.
>>>>>>>
>>>>>>> Best regards
>>>>>>> Peter
>>>>>>>
>>>>>>> Giovanni Maruzzelli schrieb:
>>>>>>>
>>>>>>>
>>>>>>>               
>>>>>>>> Ciao Peter,
>>>>>>>>
>>>>>>>> Never tested on Debian 5.
>>>>>>>>
>>>>>>>> When you write "same problem" you are referring to the audio going one
>>>>>>>> way only (btw, which way?) with the custom audio driver?
>>>>>>>>
>>>>>>>> Have you tried with multiple instances of the regular Debian
>>>>>>>> snd-dummy, as I wrote in a mail before?
>>>>>>>>
>>>>>>>> -gm
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> On Thu, Jan 28, 2010 at 9:00 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>                 
>>>>>>>>> Hello Giovanni,
>>>>>>>>>
>>>>>>>>> I did so but the same problem again.
>>>>>>>>>
>>>>>>>>> Did you ever test in on Debian 5.0?
>>>>>>>>>
>>>>>>>>> Best reards
>>>>>>>>> Peter
>>>>>>>>>
>>>>>>>>> Giovanni Maruzzelli schrieb:
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>                   
>>>>>>>>>> good, so you have only one sound device, the right one.
>>>>>>>>>>
>>>>>>>>>> Use the one with hw:0 in the window that skype gives you to set sound devices
>>>>>>>>>>
>>>>>>>>>> -gm
>>>>>>>>>>
>>>>>>>>>> On Wed, Jan 27, 2010 at 7:04 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>                     
>>>>>>>>>>> I installed alsa-utile,
>>>>>>>>>>>
>>>>>>>>>>> now I get:
>>>>>>>>>>>
>>>>>>>>>>> skype:/var/cache/apt/archives# aplay -l
>>>>>>>>>>> **** List of PLAYBACK Hardware Devices ****
>>>>>>>>>>> card 0: Dummy [Dummy], device 0: Dummy PCM [Dummy PCM]
>>>>>>>>>>>  Subdevices: 127/128
>>>>>>>>>>>  Subdevice #0: subdevice #0
>>>>>>>>>>>  Subdevice #1: subdevice #1
>>>>>>>>>>>  Subdevice #2: subdevice #2
>>>>>>>>>>>  Subdevice #3: subdevice #3
>>>>>>>>>>>  Subdevice #4: subdevice #4
>>>>>>>>>>>  Subdevice #5: subdevice #5
>>>>>>>>>>>  Subdevice #6: subdevice #6
>>>>>>>>>>>  Subdevice #7: subdevice #7
>>>>>>>>>>>  Subdevice #8: subdevice #8
>>>>>>>>>>>  Subdevice #9: subdevice #9
>>>>>>>>>>>  Subdevice #10: subdevice #10
>>>>>>>>>>>  Subdevice #11: subdevice #11
>>>>>>>>>>>  Subdevice #12: subdevice #12
>>>>>>>>>>>  Subdevice #13: subdevice #13
>>>>>>>>>>>  Subdevice #14: subdevice #14
>>>>>>>>>>>  Subdevice #15: subdevice #15
>>>>>>>>>>>  Subdevice #16: subdevice #16
>>>>>>>>>>>  Subdevice #17: subdevice #17
>>>>>>>>>>>  Subdevice #18: subdevice #18
>>>>>>>>>>>  Subdevice #19: subdevice #19
>>>>>>>>>>>  Subdevice #20: subdevice #20
>>>>>>>>>>>  Subdevice #21: subdevice #21
>>>>>>>>>>>  Subdevice #22: subdevice #22
>>>>>>>>>>>  Subdevice #23: subdevice #23
>>>>>>>>>>>  Subdevice #24: subdevice #24
>>>>>>>>>>>  Subdevice #25: subdevice #25
>>>>>>>>>>>  Subdevice #26: subdevice #26
>>>>>>>>>>>  Subdevice #27: subdevice #27
>>>>>>>>>>>  Subdevice #28: subdevice #28
>>>>>>>>>>>  Subdevice #29: subdevice #29
>>>>>>>>>>>  Subdevice #30: subdevice #30
>>>>>>>>>>>  Subdevice #31: subdevice #31
>>>>>>>>>>>  Subdevice #32: subdevice #32
>>>>>>>>>>>  Subdevice #33: subdevice #33
>>>>>>>>>>>  Subdevice #34: subdevice #34
>>>>>>>>>>>  Subdevice #35: subdevice #35
>>>>>>>>>>>  Subdevice #36: subdevice #36
>>>>>>>>>>>  Subdevice #37: subdevice #37
>>>>>>>>>>>  Subdevice #38: subdevice #38
>>>>>>>>>>>  Subdevice #39: subdevice #39
>>>>>>>>>>>  Subdevice #40: subdevice #40
>>>>>>>>>>>  Subdevice #41: subdevice #41
>>>>>>>>>>>  Subdevice #42: subdevice #42
>>>>>>>>>>>  Subdevice #43: subdevice #43
>>>>>>>>>>>  Subdevice #44: subdevice #44
>>>>>>>>>>>  Subdevice #45: subdevice #45
>>>>>>>>>>>  Subdevice #46: subdevice #46
>>>>>>>>>>>  Subdevice #47: subdevice #47
>>>>>>>>>>>  Subdevice #48: subdevice #48
>>>>>>>>>>>  Subdevice #49: subdevice #49
>>>>>>>>>>>  Subdevice #50: subdevice #50
>>>>>>>>>>>  Subdevice #51: subdevice #51
>>>>>>>>>>>  Subdevice #52: subdevice #52
>>>>>>>>>>>  Subdevice #53: subdevice #53
>>>>>>>>>>>  Subdevice #54: subdevice #54
>>>>>>>>>>>  Subdevice #55: subdevice #55
>>>>>>>>>>>  Subdevice #56: subdevice #56
>>>>>>>>>>>  Subdevice #57: subdevice #57
>>>>>>>>>>>  Subdevice #58: subdevice #58
>>>>>>>>>>>  Subdevice #59: subdevice #59
>>>>>>>>>>>  Subdevice #60: subdevice #60
>>>>>>>>>>>  Subdevice #61: subdevice #61
>>>>>>>>>>>  Subdevice #62: subdevice #62
>>>>>>>>>>>  Subdevice #63: subdevice #63
>>>>>>>>>>>  Subdevice #64: subdevice #64
>>>>>>>>>>>  Subdevice #65: subdevice #65
>>>>>>>>>>>  Subdevice #66: subdevice #66
>>>>>>>>>>>  Subdevice #67: subdevice #67
>>>>>>>>>>>  Subdevice #68: subdevice #68
>>>>>>>>>>>  Subdevice #69: subdevice #69
>>>>>>>>>>>  Subdevice #70: subdevice #70
>>>>>>>>>>>  Subdevice #71: subdevice #71
>>>>>>>>>>>  Subdevice #72: subdevice #72
>>>>>>>>>>>  Subdevice #73: subdevice #73
>>>>>>>>>>>  Subdevice #74: subdevice #74
>>>>>>>>>>>  Subdevice #75: subdevice #75
>>>>>>>>>>>  Subdevice #76: subdevice #76
>>>>>>>>>>>  Subdevice #77: subdevice #77
>>>>>>>>>>>  Subdevice #78: subdevice #78
>>>>>>>>>>>  Subdevice #79: subdevice #79
>>>>>>>>>>>  Subdevice #80: subdevice #80
>>>>>>>>>>>  Subdevice #81: subdevice #81
>>>>>>>>>>>  Subdevice #82: subdevice #82
>>>>>>>>>>>  Subdevice #83: subdevice #83
>>>>>>>>>>>  Subdevice #84: subdevice #84
>>>>>>>>>>>  Subdevice #85: subdevice #85
>>>>>>>>>>>  Subdevice #86: subdevice #86
>>>>>>>>>>>  Subdevice #87: subdevice #87
>>>>>>>>>>>  Subdevice #88: subdevice #88
>>>>>>>>>>>  Subdevice #89: subdevice #89
>>>>>>>>>>>  Subdevice #90: subdevice #90
>>>>>>>>>>>  Subdevice #91: subdevice #91
>>>>>>>>>>>  Subdevice #92: subdevice #92
>>>>>>>>>>>  Subdevice #93: subdevice #93
>>>>>>>>>>>  Subdevice #94: subdevice #94
>>>>>>>>>>>  Subdevice #95: subdevice #95
>>>>>>>>>>>  Subdevice #96: subdevice #96
>>>>>>>>>>>  Subdevice #97: subdevice #97
>>>>>>>>>>>  Subdevice #98: subdevice #98
>>>>>>>>>>>  Subdevice #99: subdevice #99
>>>>>>>>>>>  Subdevice #100: subdevice #100
>>>>>>>>>>>  Subdevice #101: subdevice #101
>>>>>>>>>>>  Subdevice #102: subdevice #102
>>>>>>>>>>>  Subdevice #103: subdevice #103
>>>>>>>>>>>  Subdevice #104: subdevice #104
>>>>>>>>>>>  Subdevice #105: subdevice #105
>>>>>>>>>>>  Subdevice #106: subdevice #106
>>>>>>>>>>>  Subdevice #107: subdevice #107
>>>>>>>>>>>  Subdevice #108: subdevice #108
>>>>>>>>>>>  Subdevice #109: subdevice #109
>>>>>>>>>>>  Subdevice #110: subdevice #110
>>>>>>>>>>>  Subdevice #111: subdevice #111
>>>>>>>>>>>  Subdevice #112: subdevice #112
>>>>>>>>>>>  Subdevice #113: subdevice #113
>>>>>>>>>>>  Subdevice #114: subdevice #114
>>>>>>>>>>>  Subdevice #115: subdevice #115
>>>>>>>>>>>  Subdevice #116: subdevice #116
>>>>>>>>>>>  Subdevice #117: subdevice #117
>>>>>>>>>>>  Subdevice #118: subdevice #118
>>>>>>>>>>>  Subdevice #119: subdevice #119
>>>>>>>>>>>  Subdevice #120: subdevice #120
>>>>>>>>>>>  Subdevice #121: subdevice #121
>>>>>>>>>>>  Subdevice #122: subdevice #122
>>>>>>>>>>>  Subdevice #123: subdevice #123
>>>>>>>>>>>  Subdevice #124: subdevice #124
>>>>>>>>>>>  Subdevice #125: subdevice #125
>>>>>>>>>>>  Subdevice #126: subdevice #126
>>>>>>>>>>>  Subdevice #127: subdevice #127
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> Peter P GMX schrieb:
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>                       
>>>>>>>>>>>> Her's the output:
>>>>>>>>>>>>
>>>>>>>>>>>> skype:~# aplay -l
>>>>>>>>>>>> bash: aplay: command not found
>>>>>>>>>>>>
>>>>>>>>>>>> Giovanni Maruzzelli schrieb:
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>                         
>>>>>>>>>>>>> I don't think you got two snd-dummy loaded (but maybe yes)
>>>>>>>>>>>>> what's the output of:
>>>>>>>>>>>>>
>>>>>>>>>>>>> aplay -l
>>>>>>>>>>>>>
>>>>>>>>>>>>> ?
>>>>>>>>>>>>>
>>>>>>>>>>>>> If instead you are referring to the choices that skype clients offers
>>>>>>>>>>>>> you in the "set audio devices" window, choose Dummy PCM (hw0:0)
>>>>>>>>>>>>>
>>>>>>>>>>>>> Eg: not the "default", but the "hardware" one
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>> On Wed, Jan 27, 2010 at 5:58 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>                           
>>>>>>>>>>>>>> Thanks Giovanni,
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> I think there may be the problem, that I have 2 sound devices now:
>>>>>>>>>>>>>> - Dummy PCM (hw0:0) (this is from debian install)
>>>>>>>>>>>>>> - Dummy PCM Default Audio device (defauzlt: CARD=Dummy) (this is new
>>>>>>>>>>>>>> since I compiled alsa newly)
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> I tried both, but both do not work. How do I get rid of the old alsa device?
>>>>>>>>>>>>>> By the way: I uninstalled Alsa before I installed the new driver
>>>>>>>>>>>>>> (apt-get remove alsa-utils alsa-base).
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> Best regards
>>>>>>>>>>>>>> Peter
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> Giovanni Maruzzelli schrieb:
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>                             
>>>>>>>>>>>>>>> This warning is harmless:
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> Jan 27 15:28:50 skype kernel: [ 3994.795786] process `skype' is using
>>>>>>>>>>>>>>> obsolete setsockopt SO_BSDCOMPAT
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> On Wed, Jan 27, 2010 at 4:26 PM, Giovanni Maruzzelli
>>>>>>>>>>>>>>> <gmaruzz at celliax.org> wrote:
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>                               
>>>>>>>>>>>>>>>> Ciao Peter
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> one instance of snd-dummy "customized" is enough for 64 instances of
>>>>>>>>>>>>>>>> skype clients, no need (and do not works) with more instances of
>>>>>>>>>>>>>>>> snd-dummy-customized.
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> Maybe you got the one-way problem because of kernel at 250HZ (don't
>>>>>>>>>>>>>>>> know). It uses to works well on a tickless kernel at 100HZ (eg: ubuntu
>>>>>>>>>>>>>>>> 8.04).
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> Or maybe you have to check and modify which sound devices the skype
>>>>>>>>>>>>>>>> clients are using (try to check that with snd-summy-custom loaded,
>>>>>>>>>>>>>>>> maybe with the ssh -X trick (as in the wiki page).
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> To load more than one snd-dummy-original (the non modified one), you
>>>>>>>>>>>>>>>> do this with the modprobe command, as in:
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> rmmod snd-dummy
>>>>>>>>>>>>>>>> modprobe snd-dummy enable=1,1,1
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> this command will enable three instances of snd-dummy original, so
>>>>>>>>>>>>>>>> you'll have three fake soundcards, and you'll have to setup each group
>>>>>>>>>>>>>>>> of 8 skype instances to use sound devices from one fake soundcard, RG:
>>>>>>>>>>>>>>>> no more than 8 skype client instances can use one instance of fake
>>>>>>>>>>>>>>>> soundcard.
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> Also, please update the mod_skypiax code (svn up in its directory) I
>>>>>>>>>>>>>>>> just committed some improvements.
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> If you have any other doubts, or need more info, don't hesitate to
>>>>>>>>>>>>>>>> write the mailing list again,
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> ciao for now,
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> -giovanni
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> On Wed, Jan 27, 2010 at 4:01 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>                                 
>>>>>>>>>>>>>>>>> I have mod_skypiax working nicely so far with 2 Skype channels. Thanks
>>>>>>>>>>>>>>>>> to all contributors, excellent work!
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> In order to have more than 8 channels working, I have followed the
>>>>>>>>>>>>>>>>> instructions in
>>>>>>>>>>>>>>>>> http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk
>>>>>>>>>>>>>>>>> and compiled alsa-driver-1.0.20 with the modified dummy.c file. (System
>>>>>>>>>>>>>>>>> ist Debian 5.0R3)
>>>>>>>>>>>>>>>>> It compiled well however when I start snd-dummy I only have
>>>>>>>>>>>>>>>>> one-way-audio and my logs show
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> Jan 27 15:28:41 skype kernel: [ 3984.318403] snd-dummy skypiax driver,
>>>>>>>>>>>>>>>>> /usr/src/alsa-driver-1.0.20/drivers/../alsa-kernel/drivers/dummy.c:920
>>>>>>>>>>>>>>>>> working on a machine with 250HZ kernel
>>>>>>>>>>>>>>>>> Jan 27 15:28:50 skype kernel: [ 3994.795786] process `skype' is using
>>>>>>>>>>>>>>>>> obsolete setsockopt SO_BSDCOMPAT
>>>>>>>>>>>>>>>>> Jan 27 15:28:56 skype kernel: [ 4005.289907] __ratelimit: 490 messages
>>>>>>>>>>>>>>>>> suppressed
>>>>>>>>>>>>>>>>> Jan 27 15:29:01 skype kernel: [ 4012.458310] process `skype' is using
>>>>>>>>>>>>>>>>> obsolete setsockopt SO_BSDCOMPAT
>>>>>>>>>>>>>>>>> Jan 27 15:29:01 skype kernel: [ 4013.326290] __ratelimit: 499 messages
>>>>>>>>>>>>>>>>> suppressed
>>>>>>>>>>>>>>>>> If I reinstall alsa from deb everything sworks fine again (of course
>>>>>>>>>>>>>>>>> with the current limitations).
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> First question: Has anybody had this issue before? How can I solve this?
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> Second question:
>>>>>>>>>>>>>>>>> As I do not need 64 channels or more: how do I manage, that Skype
>>>>>>>>>>>>>>>>> instances 9..15 use a second instance of snd-dummy as addressed in the wiki?
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> Best regards
>>>>>>>>>>>>>>>>> Peter
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>                                   
>>>>>>>>>>>>>>>> --
>>>>>>>>>>>>>>>> Sincerely,
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> Giovanni Maruzzelli
>>>>>>>>>>>>>>>> Cell : +39-347-2665618
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>                                 
>>>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>                             
>>>>>>>>>>> _______________________________________________
>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>                       
>>>>>>>>> _______________________________________________
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>                   
>>>>>>>>                 
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>               
>>>>>>
>>>>>>             
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>>           
>>>> --
>>>> Ghulam Mustafa
>>>> cell: +92 333.611.7681
>>>> sip: cyrenity at ekiga.net
>>>> mail: mustafa.pk at gmail.com
>>>> web: cyrenity.wordpress.com
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>         
>>>
>>>
>>>       
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>     
>
>
>
>   


From tayeb.meftah at gmail.com  Fri Feb  5 13:05:46 2010
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Fri, 05 Feb 2010 22:05:46 +0100
Subject: [Freeswitch-users] Switch Security
In-Reply-To: <4B6C351E.6080608@acsol.net>
References: <4B6C351E.6080608@acsol.net>
Message-ID: <4B6C882A.4050101@gmail.com>

hi
you can use acl with mod_xml_curl
thanks
Le 05/02/2010 16:11, John a ?crit :
> Freeswitch is to be used by phones external to my lan. Many of the
> phones will be coming from DSL connections without static IP. I have
> disabled the default password. What is a good procedure to secure the
> switch from non-customers registering? I know that I could use an ACL;
> however it's difficult with all the non-static users.
>
> Thanks John
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>    



From tayeb.meftah at gmail.com  Fri Feb  5 13:15:44 2010
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Fri, 05 Feb 2010 22:15:44 +0100
Subject: [Freeswitch-users] Simple IAX2 setup - help with
 converting	from asterisk to freeswitch
In-Reply-To: <1265385834.12871.83.camel@local.freepabx.com>
References: <91093d24f2166d220acf2ab19115f95d.squirrel@www.webcontracts.co.uk>	<E1E3F6EF-1A1E-4891-B371-7A3BF0988E95@jerris.com>	<d2976552e9ba0ce1ed095b7cef22e1ad.squirrel@www.webcontracts.co.uk>	<b39273531002050657x4f0f921euce8a683b8018ed63@mail.gmail.com>
	<1265385834.12871.83.camel@local.freepabx.com>
Message-ID: <4B6C8A80.9050700@gmail.com>

hi,
iax2 is secure
but, is not a good idea to avoid rtp and pass all packet including audio 
and signalisation troug the same port
and digium added some change to the IAX2 protocol so freeswitch is not 
up to date
no one want to update the iax2 stack in fs
so fs mod_iax have bean removedfrom the trunk
Le 05/02/2010 17:03, David Knell a ?crit :
> There's a fairly simple solution to IAX needs, which is to run Asterisk,
> probably on the same box, as a protocol converter - you just need to
> tell it to use a non-standard port in sip.conf so that it doesn't clash
> with FreeSWITCH.
>
> --Dave
>
>    
>> the lib that we used to provide iax support is pretty much abandonware
>> (no longer updated) and newer iax implementations (like latest
>> asterisk) can cause it to crash.  There are no license compatible iax
>> implementations that work, so.. mod_iax has been moved to the
>> unsupported column.
>>
>>
>> Default passwords -- that is a single var in vars.xml that controls
>> the passwords.
>>
>>
>> number ranges - up to you.  The sample configs supplied are just that,
>> samples.  I use a smaller range personally.
>>
>> On Fri, Feb 5, 2010 at 7:53 AM, Matthew Law<matt at webcontracts.co.uk>
>> wrote:
>>          Why is that? - a lot of web pages I have read claim that IAX
>>          is more
>>          secure and efficient.  I have no problem with using SIP
>>          whatsoever and it
>>          certainly appears to be ubiquitous.  I am a complete newcomer
>>          to VoIP and
>>          I am trying to do this as securely as possible since I want to
>>          run
>>          freeswitch on a Xen VPS which will be visible on the internet.
>>
>>          Anyway, since my first question, I have worked my way through
>>          the wiki,
>>          read a lot of example configs and made some sense of the
>>          docs.  I now have
>>          a very basic config working (with SIP) on a local vmware image
>>          using the
>>          'quick and dirty' Makefile method.  I removed all of the
>>          example configs
>>          from the xml files (those default extensions and passwords
>>          scared me) and
>>          added just one for extension 1000, plus my dialplan and
>>          inbound/outbound
>>          settings.
>>
>>          One question: is there any reason not to use a smaller
>>          extension number
>>          range, like 200-210, for example?
>>
>>          Now to figure out how to get time based roaming working...
>>
>>
>>          Thanks,
>>
>>          Matt.
>>
>>
>>          On Fri, February 5, 2010 6:43 am, Michael Jerris wrote:
>>          >  iax2 support has been removed from FreeSWITCH in current
>>          trunk and will
>>          >  not be in the 1.0.5 release.
>>          >
>>
>>
>>          >  Mike
>>
>>
>>          _______________________________________________
>>          FreeSWITCH-users mailing list
>>          FreeSWITCH-users at lists.freeswitch.org
>>          http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>          UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>          http://www.freeswitch.org
>>
>>
>>
>>
>> -- 
>> -Rupa
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>      
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>    



From gmaruzz at celliax.org  Fri Feb  5 13:19:24 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Fri, 5 Feb 2010 22:19:24 +0100
Subject: [Freeswitch-users] mod_skypiax - help needed with modified
	snd-dummy (one way audio)
In-Reply-To: <4B6C868B.3040406@gmx.net>
References: <4B60555B.2020004@gmx.net>
	<7b197bef1001281210u43a907edi13afe794cf0e1a2e@mail.gmail.com> 
	<4B61FCAB.5040707@gmx.net>
	<7b197bef1001281341h7ee2c58fj79b3886630901f29@mail.gmail.com> 
	<4B62C962.7000601@gmx.net>
	<8213d6071001290403g2c08fe39gd92eb0446bb6485a@mail.gmail.com> 
	<7b197bef1001290520x37179cb1o63273410e9ef5bc@mail.gmail.com> 
	<4B6C1CCB.4080606@gmx.net>
	<7b197bef1002050536j7ea09684v913434296ae52b97@mail.gmail.com> 
	<4B6C868B.3040406@gmx.net>
Message-ID: <7b197bef1002051319l2fcb5617p1691692f58c65372@mail.gmail.com>

that's not at all a fatal error.
I believe it works the same.
Are you sure it does not work?

-gm


On Fri, Feb 5, 2010 at 9:58 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
> Hello Giovanni,
>
> I am now at the point to install Skype. But there is only an Intrepid
> version available (no 8.04 version).
> The current verison crashed on 8.04x because of dbus error.
> ? ?process 8408: D-Bus library appears to be incorrectly set up; failed
> to read machine uuid: Failed to open "/var/lib/dbus/machine-id": No such
> file or directory
> ? ?See the manual page for dbus-uuidgen to correct this issue.
> ? ?/usr/bin/skype: symbol lookup error: /usr/bin/skype: undefined
> symbol: _ZN10QBoxLayout13addSpacerItemEP11QSpacerIte
>
> Any idea where I can download the older version for 8.04?
>
> Best regards
> Peter
>
>
> Giovanni Maruzzelli schrieb:
>> Ciao Peter,
>>
>> I would use ubuntu 8.04 (hardy) LTS server 64bit or CentOS 5.4 64bit.
>>
>> -giovanni
>>
>> On Fri, Feb 5, 2010 at 2:27 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>
>>> Hello Giovanni,
>>>
>>> as I couldn't even get skype again working again with the standard alsa
>>> driver, I would like to setup the machine from scratch based on a
>>> working machine.
>>> The latest errors I received from Skype was:
>>> ?snd_pcm_avail_update() returned a value that is exceptionally large:
>>> 715706624 bytes (3727638 ms).
>>> ?Most likely this is a bug in the ALSA driver. Please report this issue
>>> to the ALSA developers.
>>> I think that may be the reason for one-way-audio.
>>>
>>> For setting up my machine from scratch, please advise:
>>> - which OS you are you using und recommending exactly?
>>> - I would like to use 64bit OS in order to use 8GB of memory, does this
>>> work?
>>> - any other hints?
>>>
>>> Best regards
>>> Peter
>>>
>>> Giovanni Maruzzelli schrieb:
>>>
>>>> Peter,
>>>>
>>>> Can you connect on IRC (irc.freenode.net #freeswitch)? I'm gmaruzz there.
>>>>
>>>> Can you restate your problems? I've lost connection :)
>>>>
>>>> with snd-dummy custom you can create *one only* snd-dummy instance, so
>>>> *one only* fake soundcard. If you create more, will not work. But with
>>>> that one fake soundcard you can use 64 skype client instances, all
>>>> with the same soundcard hardware device (hw:n).
>>>>
>>>> with original snd-dummy you can create a max of 8 instances, so 8 fake
>>>> soundcards, and with each fake soundcard you can ?use a max of 8 skype
>>>> client instances.
>>>>
>>>> use the hardware devices, not the default devices (use the "hw:n")
>>>>
>>>> -giovanni
>>>>
>>>> On Fri, Jan 29, 2010 at 1:03 PM, Ghulam Mustafa <mustafa.pk at gmail.com> wrote:
>>>>
>>>>
>>>>> did you enable debug mode while compiling custom snd-dummy? if ?yes
>>>>> try re-compiling with debug mode disabled.
>>>>>
>>>>> -m
>>>>>
>>>>> On Fri, Jan 29, 2010 at 4:41 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>
>>>>>
>>>>>> I now reinstalled the original sound drivers
>>>>>> Unfortunaltely the sound problems remain, not that worse but they are there:
>>>>>> Audio is still (almost) one way. Almost means:
>>>>>>
>>>>>> ? ?* SIP -> Skype ok
>>>>>> ? ?* Skype=> SIP I hear only some scratching on very loud audio
>>>>>>
>>>>>> Could it be a volume problem? But snd-dummy should have no volume
>>>>>> properties, right?
>>>>>>
>>>>>> Best regards
>>>>>> Peter
>>>>>>
>>>>>>
>>>>>> Giovanni Maruzzelli schrieb:
>>>>>>
>>>>>>
>>>>>>> with three instances you will assign the hw:0 device to skype client
>>>>>>> 0...7, hw:1 to skype client 8...15 and hw:2 to skype client 16...23.
>>>>>>> Must work. Pay attention to assign the same device name to all devices
>>>>>>> needed by a skype instance (sound devices window): playback, capture
>>>>>>> AND ring.
>>>>>>>
>>>>>>> Or maybe is a bug of ALSA on Debian...
>>>>>>>
>>>>>>> -giovanni
>>>>>>>
>>>>>>> On Thu, Jan 28, 2010 at 10:07 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>> I crated 3 instances of snd-dummy, this worked. I assigned then Instance
>>>>>>>> #2 to the Skype accounts. Still no sound.
>>>>>>>> On the frist call there is one way audio, on the following calls there
>>>>>>>> is no audio at all.
>>>>>>>> This is weird.
>>>>>>>>
>>>>>>>> Best regards
>>>>>>>> Peter
>>>>>>>>
>>>>>>>> Giovanni Maruzzelli schrieb:
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>> Ciao Peter,
>>>>>>>>>
>>>>>>>>> Never tested on Debian 5.
>>>>>>>>>
>>>>>>>>> When you write "same problem" you are referring to the audio going one
>>>>>>>>> way only (btw, which way?) with the custom audio driver?
>>>>>>>>>
>>>>>>>>> Have you tried with multiple instances of the regular Debian
>>>>>>>>> snd-dummy, as I wrote in a mail before?
>>>>>>>>>
>>>>>>>>> -gm
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> On Thu, Jan 28, 2010 at 9:00 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>> Hello Giovanni,
>>>>>>>>>>
>>>>>>>>>> I did so but the same problem again.
>>>>>>>>>>
>>>>>>>>>> Did you ever test in on Debian 5.0?
>>>>>>>>>>
>>>>>>>>>> Best reards
>>>>>>>>>> Peter
>>>>>>>>>>
>>>>>>>>>> Giovanni Maruzzelli schrieb:
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>> good, so you have only one sound device, the right one.
>>>>>>>>>>>
>>>>>>>>>>> Use the one with hw:0 in the window that skype gives you to set sound devices
>>>>>>>>>>>
>>>>>>>>>>> -gm
>>>>>>>>>>>
>>>>>>>>>>> On Wed, Jan 27, 2010 at 7:04 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>> I installed alsa-utile,
>>>>>>>>>>>>
>>>>>>>>>>>> now I get:
>>>>>>>>>>>>
>>>>>>>>>>>> skype:/var/cache/apt/archives# aplay -l
>>>>>>>>>>>> **** List of PLAYBACK Hardware Devices ****
>>>>>>>>>>>> card 0: Dummy [Dummy], device 0: Dummy PCM [Dummy PCM]
>>>>>>>>>>>> ?Subdevices: 127/128
>>>>>>>>>>>> ?Subdevice #0: subdevice #0
>>>>>>>>>>>> ?Subdevice #1: subdevice #1
>>>>>>>>>>>> ?Subdevice #2: subdevice #2
>>>>>>>>>>>> ?Subdevice #3: subdevice #3
>>>>>>>>>>>> ?Subdevice #4: subdevice #4
>>>>>>>>>>>> ?Subdevice #5: subdevice #5
>>>>>>>>>>>> ?Subdevice #6: subdevice #6
>>>>>>>>>>>> ?Subdevice #7: subdevice #7
>>>>>>>>>>>> ?Subdevice #8: subdevice #8
>>>>>>>>>>>> ?Subdevice #9: subdevice #9
>>>>>>>>>>>> ?Subdevice #10: subdevice #10
>>>>>>>>>>>> ?Subdevice #11: subdevice #11
>>>>>>>>>>>> ?Subdevice #12: subdevice #12
>>>>>>>>>>>> ?Subdevice #13: subdevice #13
>>>>>>>>>>>> ?Subdevice #14: subdevice #14
>>>>>>>>>>>> ?Subdevice #15: subdevice #15
>>>>>>>>>>>> ?Subdevice #16: subdevice #16
>>>>>>>>>>>> ?Subdevice #17: subdevice #17
>>>>>>>>>>>> ?Subdevice #18: subdevice #18
>>>>>>>>>>>> ?Subdevice #19: subdevice #19
>>>>>>>>>>>> ?Subdevice #20: subdevice #20
>>>>>>>>>>>> ?Subdevice #21: subdevice #21
>>>>>>>>>>>> ?Subdevice #22: subdevice #22
>>>>>>>>>>>> ?Subdevice #23: subdevice #23
>>>>>>>>>>>> ?Subdevice #24: subdevice #24
>>>>>>>>>>>> ?Subdevice #25: subdevice #25
>>>>>>>>>>>> ?Subdevice #26: subdevice #26
>>>>>>>>>>>> ?Subdevice #27: subdevice #27
>>>>>>>>>>>> ?Subdevice #28: subdevice #28
>>>>>>>>>>>> ?Subdevice #29: subdevice #29
>>>>>>>>>>>> ?Subdevice #30: subdevice #30
>>>>>>>>>>>> ?Subdevice #31: subdevice #31
>>>>>>>>>>>> ?Subdevice #32: subdevice #32
>>>>>>>>>>>> ?Subdevice #33: subdevice #33
>>>>>>>>>>>> ?Subdevice #34: subdevice #34
>>>>>>>>>>>> ?Subdevice #35: subdevice #35
>>>>>>>>>>>> ?Subdevice #36: subdevice #36
>>>>>>>>>>>> ?Subdevice #37: subdevice #37
>>>>>>>>>>>> ?Subdevice #38: subdevice #38
>>>>>>>>>>>> ?Subdevice #39: subdevice #39
>>>>>>>>>>>> ?Subdevice #40: subdevice #40
>>>>>>>>>>>> ?Subdevice #41: subdevice #41
>>>>>>>>>>>> ?Subdevice #42: subdevice #42
>>>>>>>>>>>> ?Subdevice #43: subdevice #43
>>>>>>>>>>>> ?Subdevice #44: subdevice #44
>>>>>>>>>>>> ?Subdevice #45: subdevice #45
>>>>>>>>>>>> ?Subdevice #46: subdevice #46
>>>>>>>>>>>> ?Subdevice #47: subdevice #47
>>>>>>>>>>>> ?Subdevice #48: subdevice #48
>>>>>>>>>>>> ?Subdevice #49: subdevice #49
>>>>>>>>>>>> ?Subdevice #50: subdevice #50
>>>>>>>>>>>> ?Subdevice #51: subdevice #51
>>>>>>>>>>>> ?Subdevice #52: subdevice #52
>>>>>>>>>>>> ?Subdevice #53: subdevice #53
>>>>>>>>>>>> ?Subdevice #54: subdevice #54
>>>>>>>>>>>> ?Subdevice #55: subdevice #55
>>>>>>>>>>>> ?Subdevice #56: subdevice #56
>>>>>>>>>>>> ?Subdevice #57: subdevice #57
>>>>>>>>>>>> ?Subdevice #58: subdevice #58
>>>>>>>>>>>> ?Subdevice #59: subdevice #59
>>>>>>>>>>>> ?Subdevice #60: subdevice #60
>>>>>>>>>>>> ?Subdevice #61: subdevice #61
>>>>>>>>>>>> ?Subdevice #62: subdevice #62
>>>>>>>>>>>> ?Subdevice #63: subdevice #63
>>>>>>>>>>>> ?Subdevice #64: subdevice #64
>>>>>>>>>>>> ?Subdevice #65: subdevice #65
>>>>>>>>>>>> ?Subdevice #66: subdevice #66
>>>>>>>>>>>> ?Subdevice #67: subdevice #67
>>>>>>>>>>>> ?Subdevice #68: subdevice #68
>>>>>>>>>>>> ?Subdevice #69: subdevice #69
>>>>>>>>>>>> ?Subdevice #70: subdevice #70
>>>>>>>>>>>> ?Subdevice #71: subdevice #71
>>>>>>>>>>>> ?Subdevice #72: subdevice #72
>>>>>>>>>>>> ?Subdevice #73: subdevice #73
>>>>>>>>>>>> ?Subdevice #74: subdevice #74
>>>>>>>>>>>> ?Subdevice #75: subdevice #75
>>>>>>>>>>>> ?Subdevice #76: subdevice #76
>>>>>>>>>>>> ?Subdevice #77: subdevice #77
>>>>>>>>>>>> ?Subdevice #78: subdevice #78
>>>>>>>>>>>> ?Subdevice #79: subdevice #79
>>>>>>>>>>>> ?Subdevice #80: subdevice #80
>>>>>>>>>>>> ?Subdevice #81: subdevice #81
>>>>>>>>>>>> ?Subdevice #82: subdevice #82
>>>>>>>>>>>> ?Subdevice #83: subdevice #83
>>>>>>>>>>>> ?Subdevice #84: subdevice #84
>>>>>>>>>>>> ?Subdevice #85: subdevice #85
>>>>>>>>>>>> ?Subdevice #86: subdevice #86
>>>>>>>>>>>> ?Subdevice #87: subdevice #87
>>>>>>>>>>>> ?Subdevice #88: subdevice #88
>>>>>>>>>>>> ?Subdevice #89: subdevice #89
>>>>>>>>>>>> ?Subdevice #90: subdevice #90
>>>>>>>>>>>> ?Subdevice #91: subdevice #91
>>>>>>>>>>>> ?Subdevice #92: subdevice #92
>>>>>>>>>>>> ?Subdevice #93: subdevice #93
>>>>>>>>>>>> ?Subdevice #94: subdevice #94
>>>>>>>>>>>> ?Subdevice #95: subdevice #95
>>>>>>>>>>>> ?Subdevice #96: subdevice #96
>>>>>>>>>>>> ?Subdevice #97: subdevice #97
>>>>>>>>>>>> ?Subdevice #98: subdevice #98
>>>>>>>>>>>> ?Subdevice #99: subdevice #99
>>>>>>>>>>>> ?Subdevice #100: subdevice #100
>>>>>>>>>>>> ?Subdevice #101: subdevice #101
>>>>>>>>>>>> ?Subdevice #102: subdevice #102
>>>>>>>>>>>> ?Subdevice #103: subdevice #103
>>>>>>>>>>>> ?Subdevice #104: subdevice #104
>>>>>>>>>>>> ?Subdevice #105: subdevice #105
>>>>>>>>>>>> ?Subdevice #106: subdevice #106
>>>>>>>>>>>> ?Subdevice #107: subdevice #107
>>>>>>>>>>>> ?Subdevice #108: subdevice #108
>>>>>>>>>>>> ?Subdevice #109: subdevice #109
>>>>>>>>>>>> ?Subdevice #110: subdevice #110
>>>>>>>>>>>> ?Subdevice #111: subdevice #111
>>>>>>>>>>>> ?Subdevice #112: subdevice #112
>>>>>>>>>>>> ?Subdevice #113: subdevice #113
>>>>>>>>>>>> ?Subdevice #114: subdevice #114
>>>>>>>>>>>> ?Subdevice #115: subdevice #115
>>>>>>>>>>>> ?Subdevice #116: subdevice #116
>>>>>>>>>>>> ?Subdevice #117: subdevice #117
>>>>>>>>>>>> ?Subdevice #118: subdevice #118
>>>>>>>>>>>> ?Subdevice #119: subdevice #119
>>>>>>>>>>>> ?Subdevice #120: subdevice #120
>>>>>>>>>>>> ?Subdevice #121: subdevice #121
>>>>>>>>>>>> ?Subdevice #122: subdevice #122
>>>>>>>>>>>> ?Subdevice #123: subdevice #123
>>>>>>>>>>>> ?Subdevice #124: subdevice #124
>>>>>>>>>>>> ?Subdevice #125: subdevice #125
>>>>>>>>>>>> ?Subdevice #126: subdevice #126
>>>>>>>>>>>> ?Subdevice #127: subdevice #127
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>> Peter P GMX schrieb:
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>> Her's the output:
>>>>>>>>>>>>>
>>>>>>>>>>>>> skype:~# aplay -l
>>>>>>>>>>>>> bash: aplay: command not found
>>>>>>>>>>>>>
>>>>>>>>>>>>> Giovanni Maruzzelli schrieb:
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>> I don't think you got two snd-dummy loaded (but maybe yes)
>>>>>>>>>>>>>> what's the output of:
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> aplay -l
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> ?
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> If instead you are referring to the choices that skype clients offers
>>>>>>>>>>>>>> you in the "set audio devices" window, choose Dummy PCM (hw0:0)
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> Eg: not the "default", but the "hardware" one
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> On Wed, Jan 27, 2010 at 5:58 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> Thanks Giovanni,
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> I think there may be the problem, that I have 2 sound devices now:
>>>>>>>>>>>>>>> - Dummy PCM (hw0:0) (this is from debian install)
>>>>>>>>>>>>>>> - Dummy PCM Default Audio device (defauzlt: CARD=Dummy) (this is new
>>>>>>>>>>>>>>> since I compiled alsa newly)
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> I tried both, but both do not work. How do I get rid of the old alsa device?
>>>>>>>>>>>>>>> By the way: I uninstalled Alsa before I installed the new driver
>>>>>>>>>>>>>>> (apt-get remove alsa-utils alsa-base).
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> Best regards
>>>>>>>>>>>>>>> Peter
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> Giovanni Maruzzelli schrieb:
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> This warning is harmless:
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> Jan 27 15:28:50 skype kernel: [ 3994.795786] process `skype' is using
>>>>>>>>>>>>>>>> obsolete setsockopt SO_BSDCOMPAT
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> On Wed, Jan 27, 2010 at 4:26 PM, Giovanni Maruzzelli
>>>>>>>>>>>>>>>> <gmaruzz at celliax.org> wrote:
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> Ciao Peter
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> one instance of snd-dummy "customized" is enough for 64 instances of
>>>>>>>>>>>>>>>>> skype clients, no need (and do not works) with more instances of
>>>>>>>>>>>>>>>>> snd-dummy-customized.
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> Maybe you got the one-way problem because of kernel at 250HZ (don't
>>>>>>>>>>>>>>>>> know). It uses to works well on a tickless kernel at 100HZ (eg: ubuntu
>>>>>>>>>>>>>>>>> 8.04).
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> Or maybe you have to check and modify which sound devices the skype
>>>>>>>>>>>>>>>>> clients are using (try to check that with snd-summy-custom loaded,
>>>>>>>>>>>>>>>>> maybe with the ssh -X trick (as in the wiki page).
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> To load more than one snd-dummy-original (the non modified one), you
>>>>>>>>>>>>>>>>> do this with the modprobe command, as in:
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> rmmod snd-dummy
>>>>>>>>>>>>>>>>> modprobe snd-dummy enable=1,1,1
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> this command will enable three instances of snd-dummy original, so
>>>>>>>>>>>>>>>>> you'll have three fake soundcards, and you'll have to setup each group
>>>>>>>>>>>>>>>>> of 8 skype instances to use sound devices from one fake soundcard, RG:
>>>>>>>>>>>>>>>>> no more than 8 skype client instances can use one instance of fake
>>>>>>>>>>>>>>>>> soundcard.
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> Also, please update the mod_skypiax code (svn up in its directory) I
>>>>>>>>>>>>>>>>> just committed some improvements.
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> If you have any other doubts, or need more info, don't hesitate to
>>>>>>>>>>>>>>>>> write the mailing list again,
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> ciao for now,
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> -giovanni
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> On Wed, Jan 27, 2010 at 4:01 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>> I have mod_skypiax working nicely so far with 2 Skype channels. Thanks
>>>>>>>>>>>>>>>>>> to all contributors, excellent work!
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>> In order to have more than 8 channels working, I have followed the
>>>>>>>>>>>>>>>>>> instructions in
>>>>>>>>>>>>>>>>>> http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk
>>>>>>>>>>>>>>>>>> and compiled alsa-driver-1.0.20 with the modified dummy.c file. (System
>>>>>>>>>>>>>>>>>> ist Debian 5.0R3)
>>>>>>>>>>>>>>>>>> It compiled well however when I start snd-dummy I only have
>>>>>>>>>>>>>>>>>> one-way-audio and my logs show
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>> Jan 27 15:28:41 skype kernel: [ 3984.318403] snd-dummy skypiax driver,
>>>>>>>>>>>>>>>>>> /usr/src/alsa-driver-1.0.20/drivers/../alsa-kernel/drivers/dummy.c:920
>>>>>>>>>>>>>>>>>> working on a machine with 250HZ kernel
>>>>>>>>>>>>>>>>>> Jan 27 15:28:50 skype kernel: [ 3994.795786] process `skype' is using
>>>>>>>>>>>>>>>>>> obsolete setsockopt SO_BSDCOMPAT
>>>>>>>>>>>>>>>>>> Jan 27 15:28:56 skype kernel: [ 4005.289907] __ratelimit: 490 messages
>>>>>>>>>>>>>>>>>> suppressed
>>>>>>>>>>>>>>>>>> Jan 27 15:29:01 skype kernel: [ 4012.458310] process `skype' is using
>>>>>>>>>>>>>>>>>> obsolete setsockopt SO_BSDCOMPAT
>>>>>>>>>>>>>>>>>> Jan 27 15:29:01 skype kernel: [ 4013.326290] __ratelimit: 499 messages
>>>>>>>>>>>>>>>>>> suppressed
>>>>>>>>>>>>>>>>>> If I reinstall alsa from deb everything sworks fine again (of course
>>>>>>>>>>>>>>>>>> with the current limitations).
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>> First question: Has anybody had this issue before? How can I solve this?
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>> Second question:
>>>>>>>>>>>>>>>>>> As I do not need 64 channels or more: how do I manage, that Skype
>>>>>>>>>>>>>>>>>> instances 9..15 use a second instance of snd-dummy as addressed in the wiki?
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>> Best regards
>>>>>>>>>>>>>>>>>> Peter
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> --
>>>>>>>>>>>>>>>>> Sincerely,
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> Giovanni Maruzzelli
>>>>>>>>>>>>>>>>> Cell : +39-347-2665618
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>> _______________________________________________
>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>>
>>>>> --
>>>>> Ghulam Mustafa
>>>>> cell: +92 333.611.7681
>>>>> sip: cyrenity at ekiga.net
>>>>> mail: mustafa.pk at gmail.com
>>>>> web: cyrenity.wordpress.com
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>>
>>>>
>>>>
>>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From brian at freeswitch.org  Fri Feb  5 13:23:41 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 5 Feb 2010 15:23:41 -0600
Subject: [Freeswitch-users] Simple IAX2 setup - help with
	converting	from asterisk to freeswitch
In-Reply-To: <4B6C8A80.9050700@gmail.com>
References: <91093d24f2166d220acf2ab19115f95d.squirrel@www.webcontracts.co.uk>	<E1E3F6EF-1A1E-4891-B371-7A3BF0988E95@jerris.com>	<d2976552e9ba0ce1ed095b7cef22e1ad.squirrel@www.webcontracts.co.uk>	<b39273531002050657x4f0f921euce8a683b8018ed63@mail.gmail.com>
	<1265385834.12871.83.camel@local.freepabx.com>
	<4B6C8A80.9050700@gmail.com>
Message-ID: <0C996EBC-3E28-404B-9160-3692080D6A19@freeswitch.org>

Its not that we didn't want to... nobody stepped up to help out so we had no choice but to tag it as unsupported. 

/b

On Feb 5, 2010, at 3:15 PM, Meftah Tayeb wrote:

> no one want to update the iax2 stack in fs
> so fs mod_iax have bean removedfrom the trunk



From jerry.richards at teotech.com  Fri Feb  5 13:24:13 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Fri, 5 Feb 2010 13:24:13 -0800
Subject: [Freeswitch-users] Can Freeswitch and OpenSER Co-reside
	onSameServer?
In-Reply-To: <2d9149cd1002051150i3fba0945s4332af51d261274c@mail.gmail.com>
References: <C4DC856C75094358AEDBE3FA0E242741@greyhawk.tonecommander.com><2d9149cd1002051011x3b4f548dqabe615ed5f991687@mail.gmail.com><0203851F538F44B18BFD40EC383EBEBF@greyhawk.tonecommander.com>
	<2d9149cd1002051150i3fba0945s4332af51d261274c@mail.gmail.com>
Message-ID: <7C537AE7D6064AF080EA20CD9C3D43D6@greyhawk.tonecommander.com>

Okay, so you use both FreeSWITCH and OpenSER in one box.  But just to be
clear, if I want to I should be able to use two FreeSWITCH instances in the
same box, one as a SBC and one as a PBX.  True?

Jerry


-----Original Message-----
From: Kristian Kielhofner [mailto:kristian.kielhofner at gmail.com] 
Sent: Friday, February 05, 2010 11:50 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Can Freeswitch and OpenSER Co-reside
onSameServer?

On Fri, Feb 5, 2010 at 2:40 PM, Jerry Richards <jerry.richards at teotech.com>
wrote:
> So do you build your server with two FS instances running? ?One as the 
> SBC and one as Proxy/PBX?
>
> Thanks,
> Jerry

Jerry,

  No.  One instance of FreeSWITCH and one instance of OpenSER.  As I said,
just make sure they use separate IPs and/or ports.  I prefer standard ports
and separate IPs because then (in the future) if I need to split them
(scaling, redundancy, etc) all I have to do is bring up the second IP on a
different host and move the software/config.

--
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com





From sos at sokhapkin.dyndns.org  Fri Feb  5 13:32:57 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Fri, 5 Feb 2010 16:32:57 -0500
Subject: [Freeswitch-users]
	=?iso-8859-1?q?Simple_IAX2_setup_-_help_with_c?=
	=?iso-8859-1?q?onverting=09from_asterisk_to_freeswitch?=
In-Reply-To: <4B6C8A80.9050700@gmail.com>
References: <91093d24f2166d220acf2ab19115f95d.squirrel@www.webcontracts.co.uk>
	<1265385834.12871.83.camel@local.freepabx.com>
	<4B6C8A80.9050700@gmail.com>
Message-ID: <201002051632.57936.sos@sokhapkin.dyndns.org>

I had random crashes on IAX outgoing calls in mod_iax (all calls went to the 
same provider). I gave up and now use asterisk as protocol converter.

On Friday 05 February 2010, Meftah Tayeb wrote:
> hi,
> iax2 is secure
> but, is not a good idea to avoid rtp and pass all packet including audio
> and signalisation troug the same port
> and digium added some change to the IAX2 protocol so freeswitch is not
> up to date
> no one want to update the iax2 stack in fs
> so fs mod_iax have bean removedfrom the trunk
>
> Le 05/02/2010 17:03, David Knell a ?crit :
> > There's a fairly simple solution to IAX needs, which is to run Asterisk,
> > probably on the same box, as a protocol converter - you just need to
> > tell it to use a non-standard port in sip.conf so that it doesn't clash
> > with FreeSWITCH.
> >
> > --Dave
> >
> >> the lib that we used to provide iax support is pretty much abandonware
> >> (no longer updated) and newer iax implementations (like latest
> >> asterisk) can cause it to crash.  There are no license compatible iax
> >> implementations that work, so.. mod_iax has been moved to the
> >> unsupported column.
> >>
> >>
> >> Default passwords -- that is a single var in vars.xml that controls
> >> the passwords.
> >>
> >>
> >> number ranges - up to you.  The sample configs supplied are just that,
> >> samples.  I use a smaller range personally.
> >>
> >> On Fri, Feb 5, 2010 at 7:53 AM, Matthew Law<matt at webcontracts.co.uk>
> >> wrote:
> >>          Why is that? - a lot of web pages I have read claim that IAX
> >>          is more
> >>          secure and efficient.  I have no problem with using SIP
> >>          whatsoever and it
> >>          certainly appears to be ubiquitous.  I am a complete newcomer
> >>          to VoIP and
> >>          I am trying to do this as securely as possible since I want to
> >>          run
> >>          freeswitch on a Xen VPS which will be visible on the internet.
> >>
> >>          Anyway, since my first question, I have worked my way through
> >>          the wiki,
> >>          read a lot of example configs and made some sense of the
> >>          docs.  I now have
> >>          a very basic config working (with SIP) on a local vmware image
> >>          using the
> >>          'quick and dirty' Makefile method.  I removed all of the
> >>          example configs
> >>          from the xml files (those default extensions and passwords
> >>          scared me) and
> >>          added just one for extension 1000, plus my dialplan and
> >>          inbound/outbound
> >>          settings.
> >>
> >>          One question: is there any reason not to use a smaller
> >>          extension number
> >>          range, like 200-210, for example?
> >>
> >>          Now to figure out how to get time based roaming working...
> >>
> >>
> >>          Thanks,
> >>
> >>          Matt.
> >>
> >>          On Fri, February 5, 2010 6:43 am, Michael Jerris wrote:
> >>          >  iax2 support has been removed from FreeSWITCH in current
> >>
> >>          trunk and will
> >>
> >>          >  not be in the 1.0.5 release.
> >>          >
> >>          >
> >>          >
> >>          >  Mike
> >>
> >>          _______________________________________________
> >>          FreeSWITCH-users mailing list
> >>          FreeSWITCH-users at lists.freeswitch.org
> >>          http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>         
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >>
> >>
> >> --
> >> -Rupa
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From jerry.richards at teotech.com  Fri Feb  5 13:34:27 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Fri, 5 Feb 2010 13:34:27 -0800
Subject: [Freeswitch-users] Blind Transfer Not Working
Message-ID: <6E6877B337EC4FF683536961971841D8@greyhawk.tonecommander.com>

Does anyone know why my blind transfer is not working?  I posted a trace in
th pastebin at http://pastebin.freeswitch.org/12065.  Attended transfer is
not working either.

Thanks And Best Regards,
Jerry



From sos at sokhapkin.dyndns.org  Fri Feb  5 13:37:43 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Fri, 5 Feb 2010 16:37:43 -0500
Subject: [Freeswitch-users] Can Freeswitch and OpenSER Co-reside
	onSameServer?
In-Reply-To: <7C537AE7D6064AF080EA20CD9C3D43D6@greyhawk.tonecommander.com>
References: <C4DC856C75094358AEDBE3FA0E242741@greyhawk.tonecommander.com>
	<2d9149cd1002051150i3fba0945s4332af51d261274c@mail.gmail.com>
	<7C537AE7D6064AF080EA20CD9C3D43D6@greyhawk.tonecommander.com>
Message-ID: <201002051637.43994.sos@sokhapkin.dyndns.org>

You can use either multiple FS instances on the same box, or use different SIP 
profiles of single FS instance to perform different functions.

I use openser as registrar/load balancer and multiple FS boxes for call 
handling and billing.

On Friday 05 February 2010, Jerry Richards wrote:
> Okay, so you use both FreeSWITCH and OpenSER in one box.  But just to be
> clear, if I want to I should be able to use two FreeSWITCH instances in the
> same box, one as a SBC and one as a PBX.  True?
>
> Jerry
>
>
> -----Original Message-----
> From: Kristian Kielhofner [mailto:kristian.kielhofner at gmail.com]
> Sent: Friday, February 05, 2010 11:50 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Can Freeswitch and OpenSER Co-reside
> onSameServer?
>
> On Fri, Feb 5, 2010 at 2:40 PM, Jerry Richards <jerry.richards at teotech.com>
>
> wrote:
> > So do you build your server with two FS instances running? ?One as the
> > SBC and one as Proxy/PBX?
> >
> > Thanks,
> > Jerry
>
> Jerry,
>
>   No.  One instance of FreeSWITCH and one instance of OpenSER.  As I said,
> just make sure they use separate IPs and/or ports.  I prefer standard ports
> and separate IPs because then (in the future) if I need to split them
> (scaling, redundancy, etc) all I have to do is bring up the second IP on a
> different host and move the software/config.
>
> --
> Kristian Kielhofner
> http://www.astlinux.org
> http://blog.krisk.org
> http://www.star2star.com
> http://www.submityoursip.com
> http://www.voalte.com
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From anthony.minessale at gmail.com  Fri Feb  5 13:39:51 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 5 Feb 2010 15:39:51 -0600
Subject: [Freeswitch-users] Simple IAX2 setup - help with converting
	from asterisk to freeswitch
In-Reply-To: <201002051632.57936.sos@sokhapkin.dyndns.org>
References: <91093d24f2166d220acf2ab19115f95d.squirrel@www.webcontracts.co.uk>
	<1265385834.12871.83.camel@local.freepabx.com>
	<4B6C8A80.9050700@gmail.com>
	<201002051632.57936.sos@sokhapkin.dyndns.org>
Message-ID: <191c3a031002051339g786fc85dwdbd139d9d097d70a@mail.gmail.com>

exactly,

The random crashes started happening by themselves, the protocol on the
other end has exploited it's ancestor code and we really don't feel like
bothering to fix it.


On Fri, Feb 5, 2010 at 3:32 PM, Sergey Okhapkin <sos at sokhapkin.dyndns.org>wrote:

> I had random crashes on IAX outgoing calls in mod_iax (all calls went to
> the
> same provider). I gave up and now use asterisk as protocol converter.
>
> On Friday 05 February 2010, Meftah Tayeb wrote:
> > hi,
> > iax2 is secure
> > but, is not a good idea to avoid rtp and pass all packet including audio
> > and signalisation troug the same port
> > and digium added some change to the IAX2 protocol so freeswitch is not
> > up to date
> > no one want to update the iax2 stack in fs
> > so fs mod_iax have bean removedfrom the trunk
> >
> > Le 05/02/2010 17:03, David Knell a ?crit :
> > > There's a fairly simple solution to IAX needs, which is to run
> Asterisk,
> > > probably on the same box, as a protocol converter - you just need to
> > > tell it to use a non-standard port in sip.conf so that it doesn't clash
> > > with FreeSWITCH.
> > >
> > > --Dave
> > >
> > >> the lib that we used to provide iax support is pretty much abandonware
> > >> (no longer updated) and newer iax implementations (like latest
> > >> asterisk) can cause it to crash.  There are no license compatible iax
> > >> implementations that work, so.. mod_iax has been moved to the
> > >> unsupported column.
> > >>
> > >>
> > >> Default passwords -- that is a single var in vars.xml that controls
> > >> the passwords.
> > >>
> > >>
> > >> number ranges - up to you.  The sample configs supplied are just that,
> > >> samples.  I use a smaller range personally.
> > >>
> > >> On Fri, Feb 5, 2010 at 7:53 AM, Matthew Law<matt at webcontracts.co.uk>
> > >> wrote:
> > >>          Why is that? - a lot of web pages I have read claim that IAX
> > >>          is more
> > >>          secure and efficient.  I have no problem with using SIP
> > >>          whatsoever and it
> > >>          certainly appears to be ubiquitous.  I am a complete newcomer
> > >>          to VoIP and
> > >>          I am trying to do this as securely as possible since I want
> to
> > >>          run
> > >>          freeswitch on a Xen VPS which will be visible on the
> internet.
> > >>
> > >>          Anyway, since my first question, I have worked my way through
> > >>          the wiki,
> > >>          read a lot of example configs and made some sense of the
> > >>          docs.  I now have
> > >>          a very basic config working (with SIP) on a local vmware
> image
> > >>          using the
> > >>          'quick and dirty' Makefile method.  I removed all of the
> > >>          example configs
> > >>          from the xml files (those default extensions and passwords
> > >>          scared me) and
> > >>          added just one for extension 1000, plus my dialplan and
> > >>          inbound/outbound
> > >>          settings.
> > >>
> > >>          One question: is there any reason not to use a smaller
> > >>          extension number
> > >>          range, like 200-210, for example?
> > >>
> > >>          Now to figure out how to get time based roaming working...
> > >>
> > >>
> > >>          Thanks,
> > >>
> > >>          Matt.
> > >>
> > >>          On Fri, February 5, 2010 6:43 am, Michael Jerris wrote:
> > >>          >  iax2 support has been removed from FreeSWITCH in current
> > >>
> > >>          trunk and will
> > >>
> > >>          >  not be in the 1.0.5 release.
> > >>          >
> > >>          >
> > >>          >
> > >>          >  Mike
> > >>
> > >>          _______________________________________________
> > >>          FreeSWITCH-users mailing list
> > >>          FreeSWITCH-users at lists.freeswitch.org
> > >>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >>
> > >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > >> http://www.freeswitch.org
> > >>
> > >>
> > >>
> > >>
> > >> --
> > >> -Rupa
> > >>
> > >> _______________________________________________
> > >> FreeSWITCH-users mailing list
> > >> FreeSWITCH-users at lists.freeswitch.org
> > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > >> http://www.freeswitch.org
> > >
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100205/6b7da7e7/attachment-0001.html 

From kristian.kielhofner at gmail.com  Fri Feb  5 13:44:12 2010
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Fri, 5 Feb 2010 16:44:12 -0500
Subject: [Freeswitch-users] Can Freeswitch and OpenSER Co-reside
	onSameServer?
In-Reply-To: <201002051637.43994.sos@sokhapkin.dyndns.org>
References: <C4DC856C75094358AEDBE3FA0E242741@greyhawk.tonecommander.com>
	<2d9149cd1002051150i3fba0945s4332af51d261274c@mail.gmail.com>
	<7C537AE7D6064AF080EA20CD9C3D43D6@greyhawk.tonecommander.com>
	<201002051637.43994.sos@sokhapkin.dyndns.org>
Message-ID: <2d9149cd1002051344r4cbe8d7ha95149ec6e392e15@mail.gmail.com>

On Fri, Feb 5, 2010 at 4:37 PM, Sergey Okhapkin
<sos at sokhapkin.dyndns.org> wrote:
> You can use either multiple FS instances on the same box, or use different SIP
> profiles of single FS instance to perform different functions.
>

Exactly.

-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com


From carlos.talbot at gmail.com  Fri Feb  5 13:46:41 2010
From: carlos.talbot at gmail.com (Carlos Talbot)
Date: Fri, 5 Feb 2010 15:46:41 -0600
Subject: [Freeswitch-users] SIP over TCP with Sipdroid, an Android SIP client
Message-ID: <5800526b1002051346g3890f152o1d939faa054811e6@mail.gmail.com>

Anyone use sipdroid on their Andorid phone? For the most part it works with
the exception of when using SIP over TCP. For some reason, after 30 seconds
into a call FreeSWITCH sends a bye and drops the call. Why use TCP? The
author claims significantly increased standby times using SIP TCP over 3g:
http://code.google.com/p/sipdroid/wiki/NewStandbyTechnique

According to Brian it might be because the phone is not setting a transport
in the contact field and FS is falling back to UDP.

This is on r16557. Here's a sip trace along with call graph:

http://pastebin.freeswitch.org/12064

regards,

Carlos
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100205/a529ca80/attachment.html 

From jimthomasembedded at yahoo.com  Fri Feb  5 13:51:50 2010
From: jimthomasembedded at yahoo.com (Jim Thomas)
Date: Fri, 5 Feb 2010 13:51:50 -0800 (PST)
Subject: [Freeswitch-users] Can Freeswitch and OpenSER Co-reside on Same
	Server?
In-Reply-To: <2d9149cd1002051011x3b4f548dqabe615ed5f991687@mail.gmail.com>
References: <C4DC856C75094358AEDBE3FA0E242741@greyhawk.tonecommander.com>
	<2d9149cd1002051011x3b4f548dqabe615ed5f991687@mail.gmail.com>
Message-ID: <970116.27641.qm@web44811.mail.sp1.yahoo.com>

Kristian,

I enjoyed your recent blog praising FreeSWITCH.

The world needs an O'Reilly book about FreeSWITCH.? Perhaps you could write one in your spare time?

Whoever gets there first and does it well is likely to own that shelf space for a long time to come.

Just an idea.

Jim



----- Original Message ----
From: Kristian Kielhofner <kristian.kielhofner at gmail.com>
To: freeswitch-users at lists.freeswitch.org
Sent: Fri, February 5, 2010 12:11:35 PM
Subject: Re: [Freeswitch-users] Can Freeswitch and OpenSER Co-reside on Same Server?

On Fri, Feb 5, 2010 at 12:16 PM, Jerry Richards
<jerry.richards at teotech.com> wrote:
> If I use OpenSER for a session border controller, does anyone see an issue
> if it resides on the same server as Freeswitch? ?So I would have a LAN and
> WAN socket? ?Are there any drawbacks (other than loading) to worry about?
>
> Thanks And Best Regards,
> Jerry
>

You can use different IP addresses or ports.? I do this all of the time.

I question why you are using OpenSER (OpenSIPS?) as a SBC.? FreeSWITCH
is actually more well suited to most of the functions served by
something called* a "session border controller".

For example, FreeSWITCH in bypass media mode is a signaling only SBC
where you can (cleanly) do the header rewriting, number formatting,
and SIP topology hiding typically done by a SBC without touching the
media.? Proxy media mode can do the same while proxying media
(traversing NAT and hiding real RTP addresses).? FreeSWITCH in normal
bridging mode can transcode, convert between different types of DTMF
and do everything else mentioned above.

OpenSER as a proxy can't even (per RFC3261) rewrite To or From, nor
will it hide topology (it simple adds Record-Route/Via).

* Session borders controllers are very ill-defined and mean different
things to different people.

-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



      



From anthony.minessale at gmail.com  Fri Feb  5 13:53:27 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 5 Feb 2010 15:53:27 -0600
Subject: [Freeswitch-users] Blind Transfer Not Working
In-Reply-To: <6E6877B337EC4FF683536961971841D8@greyhawk.tonecommander.com>
References: <6E6877B337EC4FF683536961971841D8@greyhawk.tonecommander.com>
Message-ID: <191c3a031002051353u78163bafv1fcd2cd4fae244a7@mail.gmail.com>

look at line 481


   1.  v=0
   2.    o=TC 251112858 251112858 IN IP4 192.168.72.58
   3.    s=session rtp/2
   4.    c=IN IP4 192.168.72.58
   5.    t=0 0
   6.    m=audio 1760 RTP/AVP 101
   7.    a=rtpmap:101 telephone-event/8000/1


the sdp on the remote end does not have any codec info.



On Fri, Feb 5, 2010 at 3:34 PM, Jerry Richards
<jerry.richards at teotech.com>wrote:

> Does anyone know why my blind transfer is not working?  I posted a trace in
> th pastebin at http://pastebin.freeswitch.org/12065.  Attended transfer is
> not working either.
>
> Thanks And Best Regards,
> Jerry
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100205/33a535b5/attachment.html 

From sos at sokhapkin.dyndns.org  Fri Feb  5 13:58:49 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Fri, 5 Feb 2010 16:58:49 -0500
Subject: [Freeswitch-users] Simple IAX2 setup - help with converting
	from asterisk to freeswitch
In-Reply-To: <191c3a031002051339g786fc85dwdbd139d9d097d70a@mail.gmail.com>
References: <91093d24f2166d220acf2ab19115f95d.squirrel@www.webcontracts.co.uk>
	<201002051632.57936.sos@sokhapkin.dyndns.org>
	<191c3a031002051339g786fc85dwdbd139d9d097d70a@mail.gmail.com>
Message-ID: <201002051658.49267.sos@sokhapkin.dyndns.org>

From the beginning Digium made a big mistake when released libiax as 
independent  from chan_iax code. Protocol changes/bug fixes in chan_iax never 
(or maybe rarely) went back to libiax. This ended up in 2 different IAX 
protocol implementations.

I wish Digium make chan_iax as a wrapper on top of supported and actively 
developed libiax...


On Friday 05 February 2010, Anthony Minessale wrote:
> exactly,
>
> The random crashes started happening by themselves, the protocol on the
> other end has exploited it's ancestor code and we really don't feel like
> bothering to fix it.
>
> On Fri, Feb 5, 2010 at 3:32 PM, Sergey Okhapkin 
<sos at sokhapkin.dyndns.org>wrote:
> > I had random crashes on IAX outgoing calls in mod_iax (all calls went to
> > the
> > same provider). I gave up and now use asterisk as protocol converter.
> >
> > On Friday 05 February 2010, Meftah Tayeb wrote:
> > > hi,
> > > iax2 is secure
> > > but, is not a good idea to avoid rtp and pass all packet including
> > > audio and signalisation troug the same port
> > > and digium added some change to the IAX2 protocol so freeswitch is not
> > > up to date
> > > no one want to update the iax2 stack in fs
> > > so fs mod_iax have bean removedfrom the trunk
> > >
> > > Le 05/02/2010 17:03, David Knell a ?crit :
> > > > There's a fairly simple solution to IAX needs, which is to run
> >
> > Asterisk,
> >
> > > > probably on the same box, as a protocol converter - you just need to
> > > > tell it to use a non-standard port in sip.conf so that it doesn't
> > > > clash with FreeSWITCH.
> > > >
> > > > --Dave
> > > >
> > > >> the lib that we used to provide iax support is pretty much
> > > >> abandonware (no longer updated) and newer iax implementations (like
> > > >> latest asterisk) can cause it to crash.  There are no license
> > > >> compatible iax implementations that work, so.. mod_iax has been
> > > >> moved to the unsupported column.
> > > >>
> > > >>
> > > >> Default passwords -- that is a single var in vars.xml that controls
> > > >> the passwords.
> > > >>
> > > >>
> > > >> number ranges - up to you.  The sample configs supplied are just
> > > >> that, samples.  I use a smaller range personally.
> > > >>
> > > >> On Fri, Feb 5, 2010 at 7:53 AM, Matthew Law<matt at webcontracts.co.uk>
> > > >> wrote:
> > > >>          Why is that? - a lot of web pages I have read claim that
> > > >> IAX is more
> > > >>          secure and efficient.  I have no problem with using SIP
> > > >>          whatsoever and it
> > > >>          certainly appears to be ubiquitous.  I am a complete
> > > >> newcomer to VoIP and
> > > >>          I am trying to do this as securely as possible since I want
> >
> > to
> >
> > > >>          run
> > > >>          freeswitch on a Xen VPS which will be visible on the
> >
> > internet.
> >
> > > >>          Anyway, since my first question, I have worked my way
> > > >> through the wiki,
> > > >>          read a lot of example configs and made some sense of the
> > > >>          docs.  I now have
> > > >>          a very basic config working (with SIP) on a local vmware
> >
> > image
> >
> > > >>          using the
> > > >>          'quick and dirty' Makefile method.  I removed all of the
> > > >>          example configs
> > > >>          from the xml files (those default extensions and passwords
> > > >>          scared me) and
> > > >>          added just one for extension 1000, plus my dialplan and
> > > >>          inbound/outbound
> > > >>          settings.
> > > >>
> > > >>          One question: is there any reason not to use a smaller
> > > >>          extension number
> > > >>          range, like 200-210, for example?
> > > >>
> > > >>          Now to figure out how to get time based roaming working...
> > > >>
> > > >>
> > > >>          Thanks,
> > > >>
> > > >>          Matt.
> > > >>
> > > >>          On Fri, February 5, 2010 6:43 am, Michael Jerris wrote:
> > > >>          >  iax2 support has been removed from FreeSWITCH in current
> > > >>
> > > >>          trunk and will
> > > >>
> > > >>          >  not be in the 1.0.5 release.
> > > >>          >
> > > >>          >
> > > >>          >
> > > >>          >  Mike
> > > >>
> > > >>          _______________________________________________
> > > >>          FreeSWITCH-users mailing list
> > > >>          FreeSWITCH-users at lists.freeswitch.org
> >
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >
> > > >> UNSUBSCRIBE:
> >
> > http://lists.freeswitch.org/mailman/options/freeswitch-users
> >
> > > >> http://www.freeswitch.org
> > > >>
> > > >>
> > > >>
> > > >>
> > > >> --
> > > >> -Rupa
> > > >>
> > > >> _______________________________________________
> > > >> FreeSWITCH-users mailing list
> > > >> FreeSWITCH-users at lists.freeswitch.org
> > > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > >> UNSUBSCRIBE:
> >
> > http://lists.freeswitch.org/mailman/options/freeswitch-users
> >
> > > >> http://www.freeswitch.org
> > > >
> > > > _______________________________________________
> > > > FreeSWITCH-users mailing list
> > > > FreeSWITCH-users at lists.freeswitch.org
> > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > > UNSUBSCRIBE:
> >
> > http://lists.freeswitch.org/mailman/options/freeswitch-users
> >
> > > > http://www.freeswitch.org
> > >
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-user
> > >s http://www.freeswitch.org
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org




From msc at freeswitch.org  Fri Feb  5 14:07:10 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 5 Feb 2010 14:07:10 -0800
Subject: [Freeswitch-users] Can Freeswitch and OpenSER Co-reside on Same
	Server?
In-Reply-To: <970116.27641.qm@web44811.mail.sp1.yahoo.com>
References: <C4DC856C75094358AEDBE3FA0E242741@greyhawk.tonecommander.com>
	<2d9149cd1002051011x3b4f548dqabe615ed5f991687@mail.gmail.com>
	<970116.27641.qm@web44811.mail.sp1.yahoo.com>
Message-ID: <87f2f3b91002051407r2ed4eb26hf6d8ff14f3a61c89@mail.gmail.com>

On Fri, Feb 5, 2010 at 1:51 PM, Jim Thomas <jimthomasembedded at yahoo.com>wrote:

> Kristian,
>
> I enjoyed your recent blog praising FreeSWITCH.
>
> The world needs an O'Reilly book about FreeSWITCH.  Perhaps you could write
> one in your spare time?
>
>
Would you guys settle for a FreeSWITCH book from Packt Publishing? Maybe we
could get them to put an animal on the cover... :)
-MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100205/33956e5d/attachment.html 

From kristian.kielhofner at gmail.com  Fri Feb  5 14:10:05 2010
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Fri, 5 Feb 2010 17:10:05 -0500
Subject: [Freeswitch-users] Can Freeswitch and OpenSER Co-reside on Same
	Server?
In-Reply-To: <970116.27641.qm@web44811.mail.sp1.yahoo.com>
References: <C4DC856C75094358AEDBE3FA0E242741@greyhawk.tonecommander.com>
	<2d9149cd1002051011x3b4f548dqabe615ed5f991687@mail.gmail.com>
	<970116.27641.qm@web44811.mail.sp1.yahoo.com>
Message-ID: <2d9149cd1002051410o73076e0fs94ca76b4d2458801@mail.gmail.com>

On Fri, Feb 5, 2010 at 4:51 PM, Jim Thomas <jimthomasembedded at yahoo.com> wrote:
> Kristian,
>
> I enjoyed your recent blog praising FreeSWITCH.

  Thanks.

> The world needs an O'Reilly book about FreeSWITCH.? Perhaps you could write one in your spare time?

  O'Reilly authors have a joke: "The only people that get rich writing
books are Stephen King and J.K. Rowling".  I'm not picking on
O'Reilly; it doesn't matter who the publisher is.  Writing a tech book
takes an immense amount of time and compared to consulting and other
ways to spend your limited time it (often) doesn't pay well enough.

  Certainly some people are in different situation.  In the next
couple of days I'll be reviewing Packt Publishing's new OpenSIPS book
written by Flavio Goncalves.  Flavio has an OpenSIPS training business
so writing a book is perfect for him.  You get to train with the guy
who literally wrote the book!

> Whoever gets there first and does it well is likely to own that shelf space for a long time to come.

  There are some people working on a FreeSWITCH book and it makes
sense for them too.  I'm sure they'll do a great job.

-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com


From costa.zikalala at gmail.com  Fri Feb  5 15:52:48 2010
From: costa.zikalala at gmail.com (Costa Zikalala)
Date: Sat, 6 Feb 2010 01:52:48 +0200
Subject: [Freeswitch-users] Difference Between "bridge" and "transfer"
In-Reply-To: <87f2f3b91002051125u31dbf290h531d34d078c0f42e@mail.gmail.com>
References: <D123D09A29B942D988B3456528DD44AD@greyhawk.tonecommander.com> 
	<87f2f3b91002041632t19f17aefoaf11e048150816c5@mail.gmail.com> 
	<59daa2cd1002051044p1bee852ele8d0d972c303ea0e@mail.gmail.com> 
	<87f2f3b91002051125u31dbf290h531d34d078c0f42e@mail.gmail.com>
Message-ID: <59daa2cd1002051552l5d349c92w9a26baa55e5dbd1e@mail.gmail.com>

The incoming call is to a normal DID number, but I don't bridge to that
internal extension instead to a normal external PSTN number.
Will I then be charged for the b-leg?

Thanks
Costa


On 5 February 2010 21:25, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Fri, Feb 5, 2010 at 10:44 AM, Costa Zikalala <costa.zikalala at gmail.com>wrote:
>
>> Whilst on this subject, if I receive a call from PSTN and I 'bridge' it to
>> another PSTN extension, will the PSTN Provider charge me for the 'b-leg' or
>> will the original caller be charged the entire call?
>>
>
> That depends... is the "other" leg an outbound call? Is the other leg an
> inbound call to a toll-free number?
> -MC
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100206/76b43aa6/attachment.html 

From jmesquita at freeswitch.org  Fri Feb  5 16:00:16 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Fri, 5 Feb 2010 22:00:16 -0200
Subject: [Freeswitch-users] Can Freeswitch and OpenSER Co-reside on Same
	Server?
In-Reply-To: <2d9149cd1002051410o73076e0fs94ca76b4d2458801@mail.gmail.com>
References: <C4DC856C75094358AEDBE3FA0E242741@greyhawk.tonecommander.com>
	<2d9149cd1002051011x3b4f548dqabe615ed5f991687@mail.gmail.com>
	<970116.27641.qm@web44811.mail.sp1.yahoo.com>
	<2d9149cd1002051410o73076e0fs94ca76b4d2458801@mail.gmail.com>
Message-ID: <ebbc2d551002051600yedce872m1c949db909c3be3b@mail.gmail.com>

Kudos for Flavio! Vamos brazukas! :-)

Abra?os,
Jo?o Mesquita


On Fri, Feb 5, 2010 at 8:10 PM, Kristian Kielhofner <
kristian.kielhofner at gmail.com> wrote:

> On Fri, Feb 5, 2010 at 4:51 PM, Jim Thomas <jimthomasembedded at yahoo.com>
> wrote:
> > Kristian,
> >
> > I enjoyed your recent blog praising FreeSWITCH.
>
>   Thanks.
>
> > The world needs an O'Reilly book about FreeSWITCH.  Perhaps you could
> write one in your spare time?
>
>   O'Reilly authors have a joke: "The only people that get rich writing
> books are Stephen King and J.K. Rowling".  I'm not picking on
> O'Reilly; it doesn't matter who the publisher is.  Writing a tech book
> takes an immense amount of time and compared to consulting and other
> ways to spend your limited time it (often) doesn't pay well enough.
>
>  Certainly some people are in different situation.  In the next
> couple of days I'll be reviewing Packt Publishing's new OpenSIPS book
> written by Flavio Goncalves.  Flavio has an OpenSIPS training business
> so writing a book is perfect for him.  You get to train with the guy
> who literally wrote the book!
>
> > Whoever gets there first and does it well is likely to own that shelf
> space for a long time to come.
>
>   There are some people working on a FreeSWITCH book and it makes
> sense for them too.  I'm sure they'll do a great job.
>
> --
> Kristian Kielhofner
> http://www.astlinux.org
> http://blog.krisk.org
> http://www.star2star.com
> http://www.submityoursip.com
> http://www.voalte.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100205/2b03afc0/attachment.html 

From brian at freeswitch.org  Fri Feb  5 16:27:51 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 5 Feb 2010 18:27:51 -0600
Subject: [Freeswitch-users] Difference Between "bridge" and "transfer"
In-Reply-To: <59daa2cd1002051552l5d349c92w9a26baa55e5dbd1e@mail.gmail.com>
References: <D123D09A29B942D988B3456528DD44AD@greyhawk.tonecommander.com>
	<87f2f3b91002041632t19f17aefoaf11e048150816c5@mail.gmail.com>
	<59daa2cd1002051044p1bee852ele8d0d972c303ea0e@mail.gmail.com>
	<87f2f3b91002051125u31dbf290h531d34d078c0f42e@mail.gmail.com>
	<59daa2cd1002051552l5d349c92w9a26baa55e5dbd1e@mail.gmail.com>
Message-ID: <B074C6EB-C1FC-45D4-9577-AE01556CA343@freeswitch.org>

Depends dos your provider charge you for outbound calls?  Most do... so I suspect YES.

/b

On Feb 5, 2010, at 5:52 PM, Costa Zikalala wrote:

> The incoming call is to a normal DID number, but I don't bridge to that internal extension instead to a normal external PSTN number. 
> Will I then be charged for the b-leg? 
> 
> Thanks 
> Costa



From Prometheus001 at gmx.net  Fri Feb  5 17:36:18 2010
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Sat, 06 Feb 2010 02:36:18 +0100
Subject: [Freeswitch-users] mod_skypiax - help needed with
 modified	snd-dummy (one way audio)
In-Reply-To: <7b197bef1002051319l2fcb5617p1691692f58c65372@mail.gmail.com>
References: <4B60555B.2020004@gmx.net>	<7b197bef1001281210u43a907edi13afe794cf0e1a2e@mail.gmail.com>
	<4B61FCAB.5040707@gmx.net>	<7b197bef1001281341h7ee2c58fj79b3886630901f29@mail.gmail.com>
	<4B62C962.7000601@gmx.net>	<8213d6071001290403g2c08fe39gd92eb0446bb6485a@mail.gmail.com>
	<7b197bef1001290520x37179cb1o63273410e9ef5bc@mail.gmail.com>
	<4B6C1CCB.4080606@gmx.net>	<7b197bef1002050536j7ea09684v913434296ae52b97@mail.gmail.com>
	<4B6C868B.3040406@gmx.net>
	<7b197bef1002051319l2fcb5617p1691692f58c65372@mail.gmail.com>
Message-ID: <4B6CC792.5060608@gmx.net>

Skype starts, but as soon as it receives a call it crashes with:
  /usr/bin/skype: symbol lookup error: /usr/bin/skype: undefined symbol:
_ZN10QBoxLayout13addSpacerItemEP11QSpacerItem

I think the 8.10 version dos not work with8.04.

Any hints, where I may get an older Skype client? I may also try the static skype client.

Best regards
Peter



.
Giovanni Maruzzelli schrieb:
> that's not at all a fatal error.
> I believe it works the same.
> Are you sure it does not work?
>
> -gm
>
>
> On Fri, Feb 5, 2010 at 9:58 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>   
>> Hello Giovanni,
>>
>> I am now at the point to install Skype. But there is only an Intrepid
>> version available (no 8.04 version).
>> The current verison crashed on 8.04x because of dbus error.
>>    process 8408: D-Bus library appears to be incorrectly set up; failed
>> to read machine uuid: Failed to open "/var/lib/dbus/machine-id": No such
>> file or directory
>>    See the manual page for dbus-uuidgen to correct this issue.
>>    /usr/bin/skype: symbol lookup error: /usr/bin/skype: undefined
>> symbol: _ZN10QBoxLayout13addSpacerItemEP11QSpacerIte
>>
>> Any idea where I can download the older version for 8.04?
>>
>> Best regards
>> Peter
>>
>>
>> Giovanni Maruzzelli schrieb:
>>     
>>> Ciao Peter,
>>>
>>> I would use ubuntu 8.04 (hardy) LTS server 64bit or CentOS 5.4 64bit.
>>>
>>> -giovanni
>>>
>>> On Fri, Feb 5, 2010 at 2:27 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>
>>>       
>>>> Hello Giovanni,
>>>>
>>>> as I couldn't even get skype again working again with the standard alsa
>>>> driver, I would like to setup the machine from scratch based on a
>>>> working machine.
>>>> The latest errors I received from Skype was:
>>>>  snd_pcm_avail_update() returned a value that is exceptionally large:
>>>> 715706624 bytes (3727638 ms).
>>>>  Most likely this is a bug in the ALSA driver. Please report this issue
>>>> to the ALSA developers.
>>>> I think that may be the reason for one-way-audio.
>>>>
>>>> For setting up my machine from scratch, please advise:
>>>> - which OS you are you using und recommending exactly?
>>>> - I would like to use 64bit OS in order to use 8GB of memory, does this
>>>> work?
>>>> - any other hints?
>>>>
>>>> Best regards
>>>> Peter
>>>>
>>>> Giovanni Maruzzelli schrieb:
>>>>
>>>>         
>>>>> Peter,
>>>>>
>>>>> Can you connect on IRC (irc.freenode.net #freeswitch)? I'm gmaruzz there.
>>>>>
>>>>> Can you restate your problems? I've lost connection :)
>>>>>
>>>>> with snd-dummy custom you can create *one only* snd-dummy instance, so
>>>>> *one only* fake soundcard. If you create more, will not work. But with
>>>>> that one fake soundcard you can use 64 skype client instances, all
>>>>> with the same soundcard hardware device (hw:n).
>>>>>
>>>>> with original snd-dummy you can create a max of 8 instances, so 8 fake
>>>>> soundcards, and with each fake soundcard you can  use a max of 8 skype
>>>>> client instances.
>>>>>
>>>>> use the hardware devices, not the default devices (use the "hw:n")
>>>>>
>>>>> -giovanni
>>>>>
>>>>> On Fri, Jan 29, 2010 at 1:03 PM, Ghulam Mustafa <mustafa.pk at gmail.com> wrote:
>>>>>
>>>>>
>>>>>           
>>>>>> did you enable debug mode while compiling custom snd-dummy? if  yes
>>>>>> try re-compiling with debug mode disabled.
>>>>>>
>>>>>> -m
>>>>>>
>>>>>> On Fri, Jan 29, 2010 at 4:41 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>>
>>>>>>
>>>>>>             
>>>>>>> I now reinstalled the original sound drivers
>>>>>>> Unfortunaltely the sound problems remain, not that worse but they are there:
>>>>>>> Audio is still (almost) one way. Almost means:
>>>>>>>
>>>>>>>    * SIP -> Skype ok
>>>>>>>    * Skype=> SIP I hear only some scratching on very loud audio
>>>>>>>
>>>>>>> Could it be a volume problem? But snd-dummy should have no volume
>>>>>>> properties, right?
>>>>>>>
>>>>>>> Best regards
>>>>>>> Peter
>>>>>>>
>>>>>>>
>>>>>>> Giovanni Maruzzelli schrieb:
>>>>>>>
>>>>>>>
>>>>>>>               
>>>>>>>> with three instances you will assign the hw:0 device to skype client
>>>>>>>> 0...7, hw:1 to skype client 8...15 and hw:2 to skype client 16...23.
>>>>>>>> Must work. Pay attention to assign the same device name to all devices
>>>>>>>> needed by a skype instance (sound devices window): playback, capture
>>>>>>>> AND ring.
>>>>>>>>
>>>>>>>> Or maybe is a bug of ALSA on Debian...
>>>>>>>>
>>>>>>>> -giovanni
>>>>>>>>
>>>>>>>> On Thu, Jan 28, 2010 at 10:07 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>                 
>>>>>>>>> I crated 3 instances of snd-dummy, this worked. I assigned then Instance
>>>>>>>>> #2 to the Skype accounts. Still no sound.
>>>>>>>>> On the frist call there is one way audio, on the following calls there
>>>>>>>>> is no audio at all.
>>>>>>>>> This is weird.
>>>>>>>>>
>>>>>>>>> Best regards
>>>>>>>>> Peter
>>>>>>>>>
>>>>>>>>> Giovanni Maruzzelli schrieb:
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>                   
>>>>>>>>>> Ciao Peter,
>>>>>>>>>>
>>>>>>>>>> Never tested on Debian 5.
>>>>>>>>>>
>>>>>>>>>> When you write "same problem" you are referring to the audio going one
>>>>>>>>>> way only (btw, which way?) with the custom audio driver?
>>>>>>>>>>
>>>>>>>>>> Have you tried with multiple instances of the regular Debian
>>>>>>>>>> snd-dummy, as I wrote in a mail before?
>>>>>>>>>>
>>>>>>>>>> -gm
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> On Thu, Jan 28, 2010 at 9:00 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>                     
>>>>>>>>>>> Hello Giovanni,
>>>>>>>>>>>
>>>>>>>>>>> I did so but the same problem again.
>>>>>>>>>>>
>>>>>>>>>>> Did you ever test in on Debian 5.0?
>>>>>>>>>>>
>>>>>>>>>>> Best reards
>>>>>>>>>>> Peter
>>>>>>>>>>>
>>>>>>>>>>> Giovanni Maruzzelli schrieb:
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>                       
>>>>>>>>>>>> good, so you have only one sound device, the right one.
>>>>>>>>>>>>
>>>>>>>>>>>> Use the one with hw:0 in the window that skype gives you to set sound devices
>>>>>>>>>>>>
>>>>>>>>>>>> -gm
>>>>>>>>>>>>
>>>>>>>>>>>> On Wed, Jan 27, 2010 at 7:04 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>                         
>>>>>>>>>>>>> I installed alsa-utile,
>>>>>>>>>>>>>
>>>>>>>>>>>>> now I get:
>>>>>>>>>>>>>
>>>>>>>>>>>>> skype:/var/cache/apt/archives# aplay -l
>>>>>>>>>>>>> **** List of PLAYBACK Hardware Devices ****
>>>>>>>>>>>>> card 0: Dummy [Dummy], device 0: Dummy PCM [Dummy PCM]
>>>>>>>>>>>>>  Subdevices: 127/128
>>>>>>>>>>>>>  Subdevice #0: subdevice #0
>>>>>>>>>>>>>  Subdevice #1: subdevice #1
>>>>>>>>>>>>>  Subdevice #2: subdevice #2
>>>>>>>>>>>>>  Subdevice #3: subdevice #3
>>>>>>>>>>>>>  Subdevice #4: subdevice #4
>>>>>>>>>>>>>  Subdevice #5: subdevice #5
>>>>>>>>>>>>>  Subdevice #6: subdevice #6
>>>>>>>>>>>>>  Subdevice #7: subdevice #7
>>>>>>>>>>>>>  Subdevice #8: subdevice #8
>>>>>>>>>>>>>  Subdevice #9: subdevice #9
>>>>>>>>>>>>>  Subdevice #10: subdevice #10
>>>>>>>>>>>>>  Subdevice #11: subdevice #11
>>>>>>>>>>>>>  Subdevice #12: subdevice #12
>>>>>>>>>>>>>  Subdevice #13: subdevice #13
>>>>>>>>>>>>>  Subdevice #14: subdevice #14
>>>>>>>>>>>>>  Subdevice #15: subdevice #15
>>>>>>>>>>>>>  Subdevice #16: subdevice #16
>>>>>>>>>>>>>  Subdevice #17: subdevice #17
>>>>>>>>>>>>>  Subdevice #18: subdevice #18
>>>>>>>>>>>>>  Subdevice #19: subdevice #19
>>>>>>>>>>>>>  Subdevice #20: subdevice #20
>>>>>>>>>>>>>  Subdevice #21: subdevice #21
>>>>>>>>>>>>>  Subdevice #22: subdevice #22
>>>>>>>>>>>>>  Subdevice #23: subdevice #23
>>>>>>>>>>>>>  Subdevice #24: subdevice #24
>>>>>>>>>>>>>  Subdevice #25: subdevice #25
>>>>>>>>>>>>>  Subdevice #26: subdevice #26
>>>>>>>>>>>>>  Subdevice #27: subdevice #27
>>>>>>>>>>>>>  Subdevice #28: subdevice #28
>>>>>>>>>>>>>  Subdevice #29: subdevice #29
>>>>>>>>>>>>>  Subdevice #30: subdevice #30
>>>>>>>>>>>>>  Subdevice #31: subdevice #31
>>>>>>>>>>>>>  Subdevice #32: subdevice #32
>>>>>>>>>>>>>  Subdevice #33: subdevice #33
>>>>>>>>>>>>>  Subdevice #34: subdevice #34
>>>>>>>>>>>>>  Subdevice #35: subdevice #35
>>>>>>>>>>>>>  Subdevice #36: subdevice #36
>>>>>>>>>>>>>  Subdevice #37: subdevice #37
>>>>>>>>>>>>>  Subdevice #38: subdevice #38
>>>>>>>>>>>>>  Subdevice #39: subdevice #39
>>>>>>>>>>>>>  Subdevice #40: subdevice #40
>>>>>>>>>>>>>  Subdevice #41: subdevice #41
>>>>>>>>>>>>>  Subdevice #42: subdevice #42
>>>>>>>>>>>>>  Subdevice #43: subdevice #43
>>>>>>>>>>>>>  Subdevice #44: subdevice #44
>>>>>>>>>>>>>  Subdevice #45: subdevice #45
>>>>>>>>>>>>>  Subdevice #46: subdevice #46
>>>>>>>>>>>>>  Subdevice #47: subdevice #47
>>>>>>>>>>>>>  Subdevice #48: subdevice #48
>>>>>>>>>>>>>  Subdevice #49: subdevice #49
>>>>>>>>>>>>>  Subdevice #50: subdevice #50
>>>>>>>>>>>>>  Subdevice #51: subdevice #51
>>>>>>>>>>>>>  Subdevice #52: subdevice #52
>>>>>>>>>>>>>  Subdevice #53: subdevice #53
>>>>>>>>>>>>>  Subdevice #54: subdevice #54
>>>>>>>>>>>>>  Subdevice #55: subdevice #55
>>>>>>>>>>>>>  Subdevice #56: subdevice #56
>>>>>>>>>>>>>  Subdevice #57: subdevice #57
>>>>>>>>>>>>>  Subdevice #58: subdevice #58
>>>>>>>>>>>>>  Subdevice #59: subdevice #59
>>>>>>>>>>>>>  Subdevice #60: subdevice #60
>>>>>>>>>>>>>  Subdevice #61: subdevice #61
>>>>>>>>>>>>>  Subdevice #62: subdevice #62
>>>>>>>>>>>>>  Subdevice #63: subdevice #63
>>>>>>>>>>>>>  Subdevice #64: subdevice #64
>>>>>>>>>>>>>  Subdevice #65: subdevice #65
>>>>>>>>>>>>>  Subdevice #66: subdevice #66
>>>>>>>>>>>>>  Subdevice #67: subdevice #67
>>>>>>>>>>>>>  Subdevice #68: subdevice #68
>>>>>>>>>>>>>  Subdevice #69: subdevice #69
>>>>>>>>>>>>>  Subdevice #70: subdevice #70
>>>>>>>>>>>>>  Subdevice #71: subdevice #71
>>>>>>>>>>>>>  Subdevice #72: subdevice #72
>>>>>>>>>>>>>  Subdevice #73: subdevice #73
>>>>>>>>>>>>>  Subdevice #74: subdevice #74
>>>>>>>>>>>>>  Subdevice #75: subdevice #75
>>>>>>>>>>>>>  Subdevice #76: subdevice #76
>>>>>>>>>>>>>  Subdevice #77: subdevice #77
>>>>>>>>>>>>>  Subdevice #78: subdevice #78
>>>>>>>>>>>>>  Subdevice #79: subdevice #79
>>>>>>>>>>>>>  Subdevice #80: subdevice #80
>>>>>>>>>>>>>  Subdevice #81: subdevice #81
>>>>>>>>>>>>>  Subdevice #82: subdevice #82
>>>>>>>>>>>>>  Subdevice #83: subdevice #83
>>>>>>>>>>>>>  Subdevice #84: subdevice #84
>>>>>>>>>>>>>  Subdevice #85: subdevice #85
>>>>>>>>>>>>>  Subdevice #86: subdevice #86
>>>>>>>>>>>>>  Subdevice #87: subdevice #87
>>>>>>>>>>>>>  Subdevice #88: subdevice #88
>>>>>>>>>>>>>  Subdevice #89: subdevice #89
>>>>>>>>>>>>>  Subdevice #90: subdevice #90
>>>>>>>>>>>>>  Subdevice #91: subdevice #91
>>>>>>>>>>>>>  Subdevice #92: subdevice #92
>>>>>>>>>>>>>  Subdevice #93: subdevice #93
>>>>>>>>>>>>>  Subdevice #94: subdevice #94
>>>>>>>>>>>>>  Subdevice #95: subdevice #95
>>>>>>>>>>>>>  Subdevice #96: subdevice #96
>>>>>>>>>>>>>  Subdevice #97: subdevice #97
>>>>>>>>>>>>>  Subdevice #98: subdevice #98
>>>>>>>>>>>>>  Subdevice #99: subdevice #99
>>>>>>>>>>>>>  Subdevice #100: subdevice #100
>>>>>>>>>>>>>  Subdevice #101: subdevice #101
>>>>>>>>>>>>>  Subdevice #102: subdevice #102
>>>>>>>>>>>>>  Subdevice #103: subdevice #103
>>>>>>>>>>>>>  Subdevice #104: subdevice #104
>>>>>>>>>>>>>  Subdevice #105: subdevice #105
>>>>>>>>>>>>>  Subdevice #106: subdevice #106
>>>>>>>>>>>>>  Subdevice #107: subdevice #107
>>>>>>>>>>>>>  Subdevice #108: subdevice #108
>>>>>>>>>>>>>  Subdevice #109: subdevice #109
>>>>>>>>>>>>>  Subdevice #110: subdevice #110
>>>>>>>>>>>>>  Subdevice #111: subdevice #111
>>>>>>>>>>>>>  Subdevice #112: subdevice #112
>>>>>>>>>>>>>  Subdevice #113: subdevice #113
>>>>>>>>>>>>>  Subdevice #114: subdevice #114
>>>>>>>>>>>>>  Subdevice #115: subdevice #115
>>>>>>>>>>>>>  Subdevice #116: subdevice #116
>>>>>>>>>>>>>  Subdevice #117: subdevice #117
>>>>>>>>>>>>>  Subdevice #118: subdevice #118
>>>>>>>>>>>>>  Subdevice #119: subdevice #119
>>>>>>>>>>>>>  Subdevice #120: subdevice #120
>>>>>>>>>>>>>  Subdevice #121: subdevice #121
>>>>>>>>>>>>>  Subdevice #122: subdevice #122
>>>>>>>>>>>>>  Subdevice #123: subdevice #123
>>>>>>>>>>>>>  Subdevice #124: subdevice #124
>>>>>>>>>>>>>  Subdevice #125: subdevice #125
>>>>>>>>>>>>>  Subdevice #126: subdevice #126
>>>>>>>>>>>>>  Subdevice #127: subdevice #127
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>> Peter P GMX schrieb:
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>                           
>>>>>>>>>>>>>> Her's the output:
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> skype:~# aplay -l
>>>>>>>>>>>>>> bash: aplay: command not found
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> Giovanni Maruzzelli schrieb:
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>                             
>>>>>>>>>>>>>>> I don't think you got two snd-dummy loaded (but maybe yes)
>>>>>>>>>>>>>>> what's the output of:
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> aplay -l
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> ?
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> If instead you are referring to the choices that skype clients offers
>>>>>>>>>>>>>>> you in the "set audio devices" window, choose Dummy PCM (hw0:0)
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> Eg: not the "default", but the "hardware" one
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> On Wed, Jan 27, 2010 at 5:58 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>                               
>>>>>>>>>>>>>>>> Thanks Giovanni,
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> I think there may be the problem, that I have 2 sound devices now:
>>>>>>>>>>>>>>>> - Dummy PCM (hw0:0) (this is from debian install)
>>>>>>>>>>>>>>>> - Dummy PCM Default Audio device (defauzlt: CARD=Dummy) (this is new
>>>>>>>>>>>>>>>> since I compiled alsa newly)
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> I tried both, but both do not work. How do I get rid of the old alsa device?
>>>>>>>>>>>>>>>> By the way: I uninstalled Alsa before I installed the new driver
>>>>>>>>>>>>>>>> (apt-get remove alsa-utils alsa-base).
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> Best regards
>>>>>>>>>>>>>>>> Peter
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> Giovanni Maruzzelli schrieb:
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>                                 
>>>>>>>>>>>>>>>>> This warning is harmless:
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> Jan 27 15:28:50 skype kernel: [ 3994.795786] process `skype' is using
>>>>>>>>>>>>>>>>> obsolete setsockopt SO_BSDCOMPAT
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> On Wed, Jan 27, 2010 at 4:26 PM, Giovanni Maruzzelli
>>>>>>>>>>>>>>>>> <gmaruzz at celliax.org> wrote:
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>                                   
>>>>>>>>>>>>>>>>>> Ciao Peter
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>> one instance of snd-dummy "customized" is enough for 64 instances of
>>>>>>>>>>>>>>>>>> skype clients, no need (and do not works) with more instances of
>>>>>>>>>>>>>>>>>> snd-dummy-customized.
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>> Maybe you got the one-way problem because of kernel at 250HZ (don't
>>>>>>>>>>>>>>>>>> know). It uses to works well on a tickless kernel at 100HZ (eg: ubuntu
>>>>>>>>>>>>>>>>>> 8.04).
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>> Or maybe you have to check and modify which sound devices the skype
>>>>>>>>>>>>>>>>>> clients are using (try to check that with snd-summy-custom loaded,
>>>>>>>>>>>>>>>>>> maybe with the ssh -X trick (as in the wiki page).
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>> To load more than one snd-dummy-original (the non modified one), you
>>>>>>>>>>>>>>>>>> do this with the modprobe command, as in:
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>> rmmod snd-dummy
>>>>>>>>>>>>>>>>>> modprobe snd-dummy enable=1,1,1
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>> this command will enable three instances of snd-dummy original, so
>>>>>>>>>>>>>>>>>> you'll have three fake soundcards, and you'll have to setup each group
>>>>>>>>>>>>>>>>>> of 8 skype instances to use sound devices from one fake soundcard, RG:
>>>>>>>>>>>>>>>>>> no more than 8 skype client instances can use one instance of fake
>>>>>>>>>>>>>>>>>> soundcard.
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>> Also, please update the mod_skypiax code (svn up in its directory) I
>>>>>>>>>>>>>>>>>> just committed some improvements.
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>> If you have any other doubts, or need more info, don't hesitate to
>>>>>>>>>>>>>>>>>> write the mailing list again,
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>> ciao for now,
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>> -giovanni
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>> On Wed, Jan 27, 2010 at 4:01 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>                                     
>>>>>>>>>>>>>>>>>>> I have mod_skypiax working nicely so far with 2 Skype channels. Thanks
>>>>>>>>>>>>>>>>>>> to all contributors, excellent work!
>>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>> In order to have more than 8 channels working, I have followed the
>>>>>>>>>>>>>>>>>>> instructions in
>>>>>>>>>>>>>>>>>>> http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk
>>>>>>>>>>>>>>>>>>> and compiled alsa-driver-1.0.20 with the modified dummy.c file. (System
>>>>>>>>>>>>>>>>>>> ist Debian 5.0R3)
>>>>>>>>>>>>>>>>>>> It compiled well however when I start snd-dummy I only have
>>>>>>>>>>>>>>>>>>> one-way-audio and my logs show
>>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>> Jan 27 15:28:41 skype kernel: [ 3984.318403] snd-dummy skypiax driver,
>>>>>>>>>>>>>>>>>>> /usr/src/alsa-driver-1.0.20/drivers/../alsa-kernel/drivers/dummy.c:920
>>>>>>>>>>>>>>>>>>> working on a machine with 250HZ kernel
>>>>>>>>>>>>>>>>>>> Jan 27 15:28:50 skype kernel: [ 3994.795786] process `skype' is using
>>>>>>>>>>>>>>>>>>> obsolete setsockopt SO_BSDCOMPAT
>>>>>>>>>>>>>>>>>>> Jan 27 15:28:56 skype kernel: [ 4005.289907] __ratelimit: 490 messages
>>>>>>>>>>>>>>>>>>> suppressed
>>>>>>>>>>>>>>>>>>> Jan 27 15:29:01 skype kernel: [ 4012.458310] process `skype' is using
>>>>>>>>>>>>>>>>>>> obsolete setsockopt SO_BSDCOMPAT
>>>>>>>>>>>>>>>>>>> Jan 27 15:29:01 skype kernel: [ 4013.326290] __ratelimit: 499 messages
>>>>>>>>>>>>>>>>>>> suppressed
>>>>>>>>>>>>>>>>>>> If I reinstall alsa from deb everything sworks fine again (of course
>>>>>>>>>>>>>>>>>>> with the current limitations).
>>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>> First question: Has anybody had this issue before? How can I solve this?
>>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>> Second question:
>>>>>>>>>>>>>>>>>>> As I do not need 64 channels or more: how do I manage, that Skype
>>>>>>>>>>>>>>>>>>> instances 9..15 use a second instance of snd-dummy as addressed in the wiki?
>>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>> Best regards
>>>>>>>>>>>>>>>>>>> Peter
>>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>>                                       
>>>>>>>>>>>>>>>>>> --
>>>>>>>>>>>>>>>>>> Sincerely,
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>> Giovanni Maruzzelli
>>>>>>>>>>>>>>>>>> Cell : +39-347-2665618
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>                                     
>>>>>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>                                 
>>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>                           
>>>>>>>>>>> _______________________________________________
>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>                       
>>>>>>>>> _______________________________________________
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>                   
>>>>>>>>                 
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>               
>>>>>> --
>>>>>> Ghulam Mustafa
>>>>>> cell: +92 333.611.7681
>>>>>> sip: cyrenity at ekiga.net
>>>>>> mail: mustafa.pk at gmail.com
>>>>>> web: cyrenity.wordpress.com
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>>
>>>>>>             
>>>>>
>>>>>           
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>         
>>>
>>>
>>>       
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>     
>
>
>
>   


From Russell.Mosemann at cune.org  Fri Feb  5 18:51:57 2010
From: Russell.Mosemann at cune.org (Russell Mosemann)
Date: Fri, 5 Feb 2010 20:51:57 -0600
Subject: [Freeswitch-users] Difference Between "bridge" and "transfer"
In-Reply-To: <59daa2cd1002051552l5d349c92w9a26baa55e5dbd1e@mail.gmail.com>
References: <D123D09A29B942D988B3456528DD44AD@greyhawk.tonecommander.com>
	<87f2f3b91002041632t19f17aefoaf11e048150816c5@mail.gmail.com>
	<59daa2cd1002051044p1bee852ele8d0d972c303ea0e@mail.gmail.com>
	<87f2f3b91002051125u31dbf290h531d34d078c0f42e@mail.gmail.com>
	<59daa2cd1002051552l5d349c92w9a26baa55e5dbd1e@mail.gmail.com>
Message-ID: <1467DE0871D34F5DB81DC9C2ADE0463C@cune.pri>

Costa Zikalala asked

> The incoming call is to a normal DID number, but I don't bridge to that
> internal extension instead to a normal external PSTN number.
> Will I then be charged for the b-leg?

You are making the call on the b-leg. If you are normally charged for making a call, then you will be charged. A call is a call. It doesn't matter if your cousin is making the call, your grandmother is making the call or FS is making the call. It is all happening on your PSTN line. The telephone company doesn't care who is making the call. They only care that your line is being used to make a call.

--
Russell Mosemann



From nagalenoj at gmail.com  Fri Feb  5 22:48:36 2010
From: nagalenoj at gmail.com (Nagalenoj H.)
Date: Sat, 6 Feb 2010 12:18:36 +0530
Subject: [Freeswitch-users] A leg disconnects after uuid_bridge?
In-Reply-To: <191c3a031002051024w1f484febj34a34d9e06421c52@mail.gmail.com>
References: <fe67a3341002021934m548cde89p4d48381c5abe02a0@mail.gmail.com>
	<191c3a031002022108ta22287ev40fc36420abba2e5@mail.gmail.com>
	<fe67a3341002030112yeea379u90c22c482614c57@mail.gmail.com>
	<87f2f3b91002030605j7a1f9a02g665ba043c6ed7f7b@mail.gmail.com>
	<fe67a3341002040132n3aa2a897m60c5b9f1ca6b347f@mail.gmail.com>
	<0A571E14-BB58-4A3D-996A-9D5205668E85@freeswitch.org>
	<fe67a3341002040452v5468fd1fh96d1c5062091474b@mail.gmail.com>
	<191c3a031002040754r2987a03fo18ee2e1ceb239f6a@mail.gmail.com>
	<fe67a3341002042041u1ee70b86u3bc91adbb6fb3d52@mail.gmail.com>
	<191c3a031002051024w1f484febj34a34d9e06421c52@mail.gmail.com>
Message-ID: <fe67a3341002052248y3bff9d07g78d0a70e6a0ddb17@mail.gmail.com>

I've upgraded to trunk number - 16580. But, Intercept isn't working as per
the my understanding(in my last mail).

While doing 'make current', faced the following errors.. But, after
installing, I executed freeswitch and it showed the version as 16580.

installing mod_voicemail.so
make[6]: Leaving directory
`/root/files/freeswitch/src/mod/applications/mod_voicemail'
make[5]: Leaving directory
`/root/files/freeswitch/src/mod/applications/mod_voicemail'

making install mod_voipcodecs
make[5]: Entering directory
`/root/files/freeswitch/src/mod/codecs/mod_voipcodecs'
make[6]: Entering directory
`/root/files/freeswitch/src/mod/codecs/mod_voipcodecs'
make[7]: Entering directory `/root/files/freeswitch/libs/tiff-3.8.2'
 cd . && /bin/sh /root/files/freeswitch/libs/tiff-3.8.2/config/missing --run
automake-1.9 --foreign  Makefile
configure.ac: no proper invocation of AM_INIT_AUTOMAKE was found.
configure.ac: You should verify that configure.ac invokes AM_INIT_AUTOMAKE,
configure.ac: that aclocal.m4 is present in the top-level directory,
configure.ac: and that aclocal.m4 was recently regenerated (using aclocal).
configure.ac: required file `config/install-sh' not found
configure.ac:37: required file `config/config.guess' not found
configure.ac:37: required file `config/config.sub' not found
make[7]: *** [Makefile.in] Error 1
make[7]: Leaving directory `/root/files/freeswitch/libs/tiff-3.8.2'
make[6]: *** [/root/files/freeswitch/libs/tiff-3.8.2/libtiff/libtiff.la]
Error 2
make[6]: Leaving directory
`/root/files/freeswitch/src/mod/codecs/mod_voipcodecs'
make[5]: *** [install] Error 1
make[5]: Leaving directory
`/root/files/freeswitch/src/mod/codecs/mod_voipcodecs'
make[4]: *** [mod_voipcodecs-install] Error 1
make[4]: Leaving directory `/root/files/freeswitch/src/mod'
make[4]: Entering directory `/root/files/freeswitch/src'
make[4]: Leaving directory `/root/files/freeswitch/src'
make[3]: *** [install-recursive] Error 1
make[3]: Leaving directory `/root/files/freeswitch/src'
make[2]: Leaving directory `/root/files/freeswitch'
make[1]: Leaving directory `/root/files/freeswitch'


On Fri, Feb 5, 2010 at 11:54 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> try latest trunk i think your issue is fixed.
>
>
>
> On Thu, Feb 4, 2010 at 10:41 PM, Nagalenoj H. <nagalenoj at gmail.com> wrote:
>
>> Sorry., I couldn't understand its behavior.
>>
>> Let me ask the same question in this way.
>>
>> * hangup_after_bridge is set to false.
>> * In outbound socket, first I answer the call.
>> * When I do a bridge to a extension (1001), after 1001 disconnects the
>> call. I am able to make another call.
>>       sendmsg
>>       call-command: execute
>>       execute-app-name: bridge
>>       execute-app-arg: user/1001
>>
>> * When I originate a call to extension (1001), after 1001 disconnects the
>> call. I'm unable to make another call, because my session is also getting
>> closed.
>>       api originate user/1001 &park
>>
>>       Content-Type: api/response
>>       Content-Length: 41
>>
>>      +OK 1fac17ce-120b-11df-a878-d9c7fbcf71c4
>>
>>
>>       sendmsg
>>       call-command: execute
>>       execute-app-name: intercept
>>       execute-app-arg: 1fac17ce-120b-11df-a878-d9c7fbcf71c4
>>
>> * In both the case, the call is getting bridged to an extension and
>> hangup_after_bridge is false.
>> * When bridge doesn't need any other variables to set to continue, why
>> intercept needs a explicit park after bridge.?
>>
>> Hope, this has some clarity.,
>>
>>
>> On Thu, Feb 4, 2010 at 9:24 PM, Anthony Minessale <
>> anthony.minessale at gmail.com> wrote:
>>
>>>
>>>
>>>    1. 2010-02-04 14:30:09.574084 [DEBUG] switch_core_state_machine.c:354
>>>    (sofia/internal/1010 at 192.168.1.222) State SOFT_EXECUTE going to sleep
>>>    2. 2010-02-04 14:30:09.574084 [DEBUG] switch_core_state_machine.c:314
>>>    (sofia/internal/1010 at 192.168.1.222) Running State Change CS_EXECUTE
>>>    3. 2010-02-04 14:30:09.574084 [DEBUG] switch_core_state_machine.c:348
>>>    (sofia/internal/1010 at 192.168.1.222) State EXECUTE
>>>    4. 2010-02-04 14:30:09.574084 [DEBUG] mod_sofia.c:181 sofia/internal/
>>>    1010 at 192.168.1.222 SOFIA EXECUTE
>>>    5. 2010-02-04 14:30:09.574084 [DEBUG] switch_core_state_machine.c:159sofia/internal/
>>>    1010 at 192.168.1.222 Standard EXECUTE
>>>    6. 2010-02-04 14:30:09.574084 [NOTICE] switch_core_state_machine.c:
>>>    187 Hangup sofia/internal/1010 at 192.168.1.222 [CS_EXECUTE] [
>>>    NORMAL_CLEARING]
>>>
>>>
>>>
>>> Your channel went back to EXECUTE as expected then it hungup because
>>> there were no more instructions in your dial plan for it to execute.  So it
>>> is working as expected.
>>>
>>> Consider using transfer_after_bridge variable or park_after bridge to
>>> make it stay around when the call is over.
>>>
>>>
>>>
>>>
>>> On Thu, Feb 4, 2010 at 6:52 AM, Nagalenoj H. <nagalenoj at gmail.com>wrote:
>>>
>>>> By using create_uuid. I've also tried without giving origination_uuid.
>>>> But, the result is same.
>>>>
>>>> --
>>>> Regards,
>>>> Nagalenoj H.
>>>>
>>>>
>>>> On Thu, Feb 4, 2010 at 5:42 PM, Brian West <brian at freeswitch.org>wrote:
>>>>
>>>>> Where are you getting this UUID?
>>>>>
>>>>> /b
>>>>>
>>>>> On Feb 4, 2010, at 3:32 AM, Nagalenoj H. wrote:
>>>>>
>>>>> > api originate
>>>>> {origination_uuid=c6d2e0e2-0b2f-11df-9e84-fb15c3cd8565}user/1001 &park
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>> iax:guest at conference.freeswitch.org/888
>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> pstn:+19193869900
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Regards,
>> Nagalenoj H.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Regards,
Nagalenoj H.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100206/207a5e32/attachment.html 

From oseslija at gmail.com  Sat Feb  6 02:55:01 2010
From: oseslija at gmail.com (Ognjen Seslija)
Date: Sat, 6 Feb 2010 11:55:01 +0100
Subject: [Freeswitch-users] Difference Between "bridge" and "transfer"
In-Reply-To: <59daa2cd1002051044p1bee852ele8d0d972c303ea0e@mail.gmail.com>
References: <D123D09A29B942D988B3456528DD44AD@greyhawk.tonecommander.com>
	<87f2f3b91002041632t19f17aefoaf11e048150816c5@mail.gmail.com>
	<59daa2cd1002051044p1bee852ele8d0d972c303ea0e@mail.gmail.com>
Message-ID: <4468a6771002060255l582856e7g577d63d0b596b173@mail.gmail.com>

This happens when most of my customers forward call from their sip phones to
pstn extensions (mobile phones i.e.).
FS will then bridge two pstn calls and hence you'll be charged for the b-leg
call. Also, most likely is that you will not be able to present the original
caller id of a-leg to the forwarded extension, because of pstn settings
(inbound acls for clid on your line).

There is a way on most ISDN lines to "deflect" call back to the telco switch
with the forward extension information, so the switch can call the b-leg
itself and properly present the original clid. FS then doesn't need to
anything but to respond with the information.
I'm pretty sure, though, that telco will charge you for that second call,
also.

Regards,
Ognjen

On Fri, Feb 5, 2010 at 7:44 PM, Costa Zikalala <costa.zikalala at gmail.com>wrote:

> Whilst on this subject, if I receive a call from PSTN and I 'bridge' it to
> another PSTN extension, will the PSTN Provider charge me for the 'b-leg' or
> will the original caller be charged the entire call?
>
>
>
> On 5 February 2010 02:32, Michael Collins <msc at freeswitch.org> wrote:
>
>>
>>
>> On Thu, Feb 4, 2010 at 3:56 PM, Jerry Richards <
>> jerry.richards at teotech.com> wrote:
>>
>>> What is the difference between "bridge" and "transfer"?  I'm looking at
>>> the
>>> demo IVRs.
>>>
>>>
>> bridge will connect two endpoints together while transfer sends the
>> endpoint back through the dialplan again...
>>
>> -MC
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100206/601917d3/attachment-0001.html 

From woodydickson at gmail.com  Sat Feb  6 05:06:38 2010
From: woodydickson at gmail.com (Woody Dickson)
Date: Sat, 6 Feb 2010 21:06:38 +0800
Subject: [Freeswitch-users] AUDIO RTP REPORTS ERROR: [Bind Error!]
Message-ID: <a44688a1002060506q8679d19u5e403207df073d30@mail.gmail.com>

Hi,

While running FreeSwitch with 300 CPS, I start to get the following error:


2010-02-05 01:08:39.756510 [ERR] sofia_glue.c:2503 AUDIO RTP REPORTS
ERROR: [Bind Error!]
2010-02-05 01:08:39.759524 [WARNING] switch_ivr_bridge.c:992 Bridge
Failed


Each time such an error pops up, the memory used by FS increases.
This is a potential memory leaking there.


Does anyone know what is the problem and how to solve it?


woody


From tayeb.meftah at gmail.com  Sat Feb  6 05:41:18 2010
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Sat, 06 Feb 2010 14:41:18 +0100
Subject: [Freeswitch-users] Billing IVR
In-Reply-To: <4468a6771002060255l582856e7g577d63d0b596b173@mail.gmail.com>
References: <D123D09A29B942D988B3456528DD44AD@greyhawk.tonecommander.com>	<87f2f3b91002041632t19f17aefoaf11e048150816c5@mail.gmail.com>	<59daa2cd1002051044p1bee852ele8d0d972c303ea0e@mail.gmail.com>
	<4468a6771002060255l582856e7g577d63d0b596b173@mail.gmail.com>
Message-ID: <4B6D717E.4000306@gmail.com>

hi,
  i am creating a simple prepay IVR to bill calls troug mod_nibblebill
     using javascript and some recorded sounds
the user call 222
call get answered and request a pin
the pin is the nibblebill account number
if the pin is corect, the user will heare the actual account cache
and if want to recharge, the user will enter the recharge card pin 
number (8 digites)
if is corecte, we move cache from the card table to account table and 
the account is debited
but if user inter the card pin number, the IVR is returning the user to 
the pin number prompt
any error in this
Pastebin <http://pastebin:freeswitch at pastebin.freeswitch.org/12068>
(http://pastebin.freeswitch.org/12068)
please help me folk
thank you
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100206/3a3403d2/attachment.html 

From anthony.minessale at gmail.com  Sat Feb  6 05:44:27 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 6 Feb 2010 07:44:27 -0600
Subject: [Freeswitch-users] A leg disconnects after uuid_bridge?
In-Reply-To: <191c3a031002060543x27fb6ce8t6b96f05630dec77d@mail.gmail.com>
References: <fe67a3341002021934m548cde89p4d48381c5abe02a0@mail.gmail.com>
	<87f2f3b91002030605j7a1f9a02g665ba043c6ed7f7b@mail.gmail.com>
	<fe67a3341002040132n3aa2a897m60c5b9f1ca6b347f@mail.gmail.com>
	<0A571E14-BB58-4A3D-996A-9D5205668E85@freeswitch.org>
	<fe67a3341002040452v5468fd1fh96d1c5062091474b@mail.gmail.com>
	<191c3a031002040754r2987a03fo18ee2e1ceb239f6a@mail.gmail.com>
	<fe67a3341002042041u1ee70b86u3bc91adbb6fb3d52@mail.gmail.com>
	<191c3a031002051024w1f484febj34a34d9e06421c52@mail.gmail.com>
	<fe67a3341002052248y3bff9d07g78d0a70e6a0ddb17@mail.gmail.com>
	<191c3a031002060543x27fb6ce8t6b96f05630dec77d@mail.gmail.com>
Message-ID: <191c3a031002060544v4707e0abmbbc239e6469f0ba0@mail.gmail.com>

You probably need to do a fresh build those errors mean its not fully
building correctly.

On Feb 6, 2010 12:56 AM, "Nagalenoj H." <nagalenoj at gmail.com> wrote:

I've upgraded to trunk number - 16580. But, Intercept isn't working as per
the my understanding(in my last mail).

While doing 'make current', faced the following errors.. But, after
installing, I executed freeswitch and it showed the version as 16580.

installing mod_voicemail.so
make[6]: Leaving directory
`/root/files/freeswitch/src/mod/applications/mod_voicemail'
make[5]: Leaving directory
`/root/files/freeswitch/src/mod/applications/mod_voicemail'

making install mod_voipcodecs
make[5]: Entering directory
`/root/files/freeswitch/src/mod/codecs/mod_voipcodecs'
make[6]: Entering directory
`/root/files/freeswitch/src/mod/codecs/mod_voipcodecs'
make[7]: Entering directory `/root/files/freeswitch/libs/tiff-3.8.2'
 cd . && /bin/sh /root/files/freeswitch/libs/tiff-3.8.2/config/missing --run
automake-1.9 --foreign  Makefile
configure.ac: no proper invocation of AM_INIT_AUTOMAKE was found.
configure.ac: You should verify that configure.ac invokes AM_INIT_AUTOMAKE,
configure.ac: that aclocal.m4 is present in the top-level directory,
configure.ac: and that aclocal.m4 was recently regenerated (using aclocal).
configure.ac: required file `config/install-sh' not found
configure.ac:37: required file `config/config.guess' not found
configure.ac:37: required file `config/config.sub' not found
make[7]: *** [Makefile.in] Error 1
make[7]: Leaving directory `/root/files/freeswitch/libs/tiff-3.8.2'
make[6]: *** [/root/files/freeswitch/libs/tiff-3.8.2/libtiff/libtiff.la]
Error 2
make[6]: Leaving directory
`/root/files/freeswitch/src/mod/codecs/mod_voipcodecs'
make[5]: *** [install] Error 1
make[5]: Leaving directory
`/root/files/freeswitch/src/mod/codecs/mod_voipcodecs'
make[4]: *** [mod_voipcodecs-install] Error 1
make[4]: Leaving directory `/root/files/freeswitch/src/mod'
make[4]: Entering directory `/root/files/freeswitch/src'
make[4]: Leaving directory `/root/files/freeswitch/src'
make[3]: *** [install-recursive] Error 1
make[3]: Leaving directory `/root/files/freeswitch/src'
make[2]: Leaving directory `/root/files/freeswitch'
make[1]: Leaving directory `/root/files/freeswitch'




On Fri, Feb 5, 2010 at 11:54 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:
>
> try l...

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100206/9aa099a6/attachment.html 

From anthony.minessale at gmail.com  Sat Feb  6 05:55:01 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 6 Feb 2010 07:55:01 -0600
Subject: [Freeswitch-users] AUDIO RTP REPORTS ERROR: [Bind Error!]
In-Reply-To: <191c3a031002060552g4f241f83wb4d881ed07de7dd5@mail.gmail.com>
References: <a44688a1002060506q8679d19u5e403207df073d30@mail.gmail.com>
	<191c3a031002060549h57c256fapb1b7e56aaab8a12f@mail.gmail.com>
	<191c3a031002060550h1fb60638q16b201536685b532@mail.gmail.com>
	<191c3a031002060552g4f241f83wb4d881ed07de7dd5@mail.gmail.com>
Message-ID: <191c3a031002060555x153637d9iec16fb1d8dc44d7c@mail.gmail.com>

That's pretty funny.
Let me stop laughing about the nonchalant expectation of 300cps
first.........................

Ok so,

You ran out of rtp ports.
The configured range is from 16384 to 32768 evens so do the math, 8192 simo
ports.

At that rate you are going to run out or possibly collide with other rtp
enabled devices on the same box.

If you are just using sipp to torture fs expect it to back up if you go
beyond the limitations of your hardware.

That said we do not entertain load testing threads......

On Feb 6, 2010 7:13 AM, "Woody Dickson" <woodydickson at gmail.com> wrote:

Hi,

While running FreeSwitch with 300 CPS, I start to get the following error:


2010-02-05 01:08:39.756510 [ERR] sofia_glue.c:2503 AUDIO RTP REPORTS
ERROR: [Bind Error!]
2010-02-05 01:08:39.759524 [WARNING] switch_ivr_bridge.c:992 Bridge
Failed


Each time such an error pops up, the memory used by FS increases.
This is a potential memory leaking there.


Does anyone know what is the problem and how to solve it?


woody

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100206/09c70177/attachment.html 

From anthony.minessale at gmail.com  Sat Feb  6 05:58:48 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 6 Feb 2010 07:58:48 -0600
Subject: [Freeswitch-users] mod_skypiax - help needed with modified
	snd-dummy (one way audio)
In-Reply-To: <4B6CC792.5060608@gmx.net>
References: <4B60555B.2020004@gmx.net>
	<7b197bef1001281341h7ee2c58fj79b3886630901f29@mail.gmail.com>
	<4B62C962.7000601@gmx.net>
	<8213d6071001290403g2c08fe39gd92eb0446bb6485a@mail.gmail.com>
	<7b197bef1001290520x37179cb1o63273410e9ef5bc@mail.gmail.com>
	<4B6C1CCB.4080606@gmx.net>
	<7b197bef1002050536j7ea09684v913434296ae52b97@mail.gmail.com>
	<4B6C868B.3040406@gmx.net>
	<7b197bef1002051319l2fcb5617p1691692f58c65372@mail.gmail.com>
	<4B6CC792.5060608@gmx.net>
Message-ID: <191c3a031002060558r36947d93vcbb46df9e0a7dfde@mail.gmail.com>

Quick fyi centos 5.4 has some memory bugs in libc don't go beyond 5.3 until
its fixed.

On Feb 5, 2010 7:42 PM, "Peter P GMX" <Prometheus001 at gmx.net> wrote:

Skype starts, but as soon as it receives a call it crashes with:

/usr/bin/skype: symbol lookup error: /usr/bin/skype: undefined symbol:
_ZN10QBoxLayout13addSpacerItemEP11QSpacerItem

I think the 8.10 version dos not work with8.04.

Any hints, where I may get an older Skype client? I may also try the static
skype client.


Best regards
Peter



.
Giovanni Maruzzelli schrieb:
> that's not at all a fatal error.
> I believe...
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100206/7472dca5/attachment.html 

From steveu at coppice.org  Sat Feb  6 10:21:13 2010
From: steveu at coppice.org (Steve Underwood)
Date: Sun, 07 Feb 2010 02:21:13 +0800
Subject: [Freeswitch-users] The brilliance of hyper-links
Message-ID: <4B6DB319.2090308@coppice.org>

I would like to strongly object to a story posted on the front page at 
www.freeswitch.org, about patent reexamination. Its the kind of thing 
that gets proponents of patent reform a bad name. It says "Did you know 
that someone actually got a patent on the oh-so-clever concept of the 
hyperlink 
<http://patft.uspto.gov/netacgi/nph-Parser?Sect1=PTO1&Sect2=HITOFF&d=PALL&p=1&u=%2Fnetahtml%2FPTO%2Fsrchnum.htm&r=1&f=G&l=50&s1=4,873,662.PN.&OS=PN/4,873,662&RS=PN/4,873,662>? 
Enough said.". If you can't see the brilliance of the hyperlink you must 
be a fool. When Doug Engelbart demostrated that concept in 1968 it was 
so brilliant it went over the heads of most of the audience, some of 
them making dumb remarks that entirely missed the point. That's pretty 
much conclusive proof that it was not something obvious to practitioners 
in the art.

The patent on hyperlinks was not bad because the idea was obvious. It 
was bad because it was applied for in 1980, 12 years after the concept 
was demonstrated. Luckily, video exists of the 1968 demo, and the patent 
was shot down.

If you can't get your act together about where real innovation lies, 
just shut up. It just makes arguing a meaningful case for a better 
patent system hard for the rest of us.

Steve



From jimthomasembedded at yahoo.com  Sat Feb  6 11:14:19 2010
From: jimthomasembedded at yahoo.com (Jim Thomas)
Date: Sat, 6 Feb 2010 11:14:19 -0800 (PST)
Subject: [Freeswitch-users] Can Freeswitch and OpenSER Co-reside on Same
	Server?
In-Reply-To: <87f2f3b91002051407r2ed4eb26hf6d8ff14f3a61c89@mail.gmail.com>
References: <C4DC856C75094358AEDBE3FA0E242741@greyhawk.tonecommander.com>
	<2d9149cd1002051011x3b4f548dqabe615ed5f991687@mail.gmail.com>
	<970116.27641.qm@web44811.mail.sp1.yahoo.com>
	<87f2f3b91002051407r2ed4eb26hf6d8ff14f3a61c89@mail.gmail.com>
Message-ID: <344660.64876.qm@web44803.mail.sp1.yahoo.com>

Packt would be fine, and kudos to Packt for being so present in the open source telephony publishing space.

Actually, I would like to see two FreeSWITCH books, one for users (how to employ it as a turnkey element) and another for developers (FreeSWITCH internal architecture, structure, Sofia-SIP, etc.).



________________________________
From: Michael Collins <msc at freeswitch.org>
To: freeswitch-users at lists.freeswitch.org
Sent: Fri, February 5, 2010 4:07:10 PM
Subject: Re: [Freeswitch-users] Can Freeswitch and OpenSER Co-reside on Same Server?




On Fri, Feb 5, 2010 at 1:51 PM, Jim Thomas <jimthomasembedded at yahoo.com> wrote:

>Kristian,
>
>>I enjoyed your recent blog praising FreeSWITCH.
>
>>The world needs an O'Reilly book about FreeSWITCH.  Perhaps you could write one in your spare time?
>
>

Would you guys settle for a FreeSWITCH book from Packt Publishing? Maybe we could get them to put an animal on the cover... :)
-MC


      
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100206/89974506/attachment-0001.html 

From christian at officepools.com  Sat Feb  6 11:24:05 2010
From: christian at officepools.com (Christian Jensen)
Date: Sat, 6 Feb 2010 11:24:05 -0800
Subject: [Freeswitch-users] Looking for some good/cheap desktop phones
In-Reply-To: <SNT119-W1223C808A5E092C7E2A590B5540@phx.gbl>
References: <3f2e071b5aea5e11bb37a37e435a33b7@mail.gmail.com>
	<8e9d67561002042236s434e0643od322551de8720be1@mail.gmail.com>
	<409D0ACF-ADA4-4D02-8B7E-71EC1B1BAFBC@freeswitch.org>
	<8e9d67561002042301v76129abfj3dcbc70ed073752c@mail.gmail.com>
	<d361ca7b1002050507n5e564621oa6cf7ace005dd5b8@mail.gmail.com>
	<00ae01caa698$65b570f0$312052d0$@net>
	<b39273531002051154h5132faa9h9fb024ebdfb15b77@mail.gmail.com>
	<SNT119-W1223C808A5E092C7E2A590B5540@phx.gbl>
Message-ID: <e90351e71002061124v129b081cg501fb0c6ae6a17dd@mail.gmail.com>

The response on this thread has been great!

I wonder if there should be a link off the FreeSwitch site for vendors for
these phones.... hint hint.

>From what I have been hearing here and elsewhere, the preference is Polycom,
Cisco, Grandstream, Aastra in that order.

Does anyone have a URL for where I could buy the units without crazy stupid
shipping costs? VoipSupply quoted me ~$60 for a shipping a single phone to
Vancouver, Canada, Yick!

I looked on EBay but there isn't anything I could consider a steal.

Did I hear that someone was putting FS into a phone or something like that?
Cluecon and the secure sip guy was talking about this I think.

Christian
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100206/78214d55/attachment.html 

From frank at carmickle.com  Sat Feb  6 12:05:47 2010
From: frank at carmickle.com (Frank Carmickle)
Date: Sat, 6 Feb 2010 15:05:47 -0500
Subject: [Freeswitch-users] Looking for some good/cheap desktop phones
In-Reply-To: <e90351e71002061124v129b081cg501fb0c6ae6a17dd@mail.gmail.com>
References: <3f2e071b5aea5e11bb37a37e435a33b7@mail.gmail.com>
	<8e9d67561002042236s434e0643od322551de8720be1@mail.gmail.com>
	<409D0ACF-ADA4-4D02-8B7E-71EC1B1BAFBC@freeswitch.org>
	<8e9d67561002042301v76129abfj3dcbc70ed073752c@mail.gmail.com>
	<d361ca7b1002050507n5e564621oa6cf7ace005dd5b8@mail.gmail.com>
	<00ae01caa698$65b570f0$312052d0$@net>
	<b39273531002051154h5132faa9h9fb024ebdfb15b77@mail.gmail.com>
	<SNT119-W1223C808A5E092C7E2A590B5540@phx.gbl>
	<e90351e71002061124v129b081cg501fb0c6ae6a17dd@mail.gmail.com>
Message-ID: <20100206200547.GD31942@base.carmickle.com>

On Sat, Feb 06, Christian Jensen wrote:
> The response on this thread has been great!
> 
> I wonder if there should be a link off the FreeSwitch site for vendors for
> these phones.... hint hint.
> 
> >From what I have been hearing here and elsewhere, the preference is Polycom,
> Cisco, Grandstream, Aastra in that order.

Hmmm...  That's not the impression that I got at all.  From what I picked up on this thread it would look more like

polycom snom aastra cisco grandstream

I'm sure others can speak to how Cisco doesn't do presence right.  They are harder to provision then Snoms and Polycoms and in my opinion don't sound as good either.  To me they sound dull and fuzzy on transmit.  I've heard about four different versions of 7940/7960.  

The Snom 3x0 series have suffered in the speaker phone department until version 7 firmware.  Version 8 betas have sounded even better yet.  Make sure that if your going to be using headsets with Snoms that you ground the phone to something.  This can be done by either having a shielded network cable attached to a grounded switch or computer, or order them with the grounded power adaptor.  They give you the not so grounded adaptor by default.  Finding shielded ethernet can be kind of tricky but it is nice to have.

Polycom speech quality is my favorite if you can handle the ringy under water sounds of the background noise especially on speaker.  It sounds like there's a fish tank running near by if there's any white noise.  I believe this to be the case because the way they split the audio up in to many bands to process it.  Maybe less steep filters would help.  

It's hard to believe how many people just don't care about call quality.  It starts with a good sounding phone that doesn't echo or hum.  I'd like to hope that we are doing our part to help the world sound a little better all the time.

--FC


From msc at freeswitch.org  Sat Feb  6 13:48:20 2010
From: msc at freeswitch.org (Michael S Collins)
Date: Sat, 6 Feb 2010 13:48:20 -0800
Subject: [Freeswitch-users] The brilliance of hyper-links
In-Reply-To: <4B6DB319.2090308@coppice.org>
References: <4B6DB319.2090308@coppice.org>
Message-ID: <7B471A70-40F2-4F8D-A3EA-009DDA20FCD6@freeswitch.org>

Sir,

You've gotten unnecessarily upset. A poor choice of words written in a  
bit of a hurry. You are exactly right about the reason *why* the  
patent is bogus. "Clever" was the wrong word here. It should have been  
"the oh so new, original, never-been-done-before, not even a hint of  
prior art, concept of the hyperlink."

No offense intended to the real originator(s) of the very useful  
hyperlink.

-MC

Sent from my iPhone

On Feb 6, 2010, at 10:21 AM, Steve Underwood <steveu at coppice.org> wrote:

> I would like to strongly object to a story posted on the front page at
> www.freeswitch.org, about patent reexamination. Its the kind of thing
> that gets proponents of patent reform a bad name. It says "Did you  
> know
> that someone actually got a patent on the oh-so-clever concept of the
> hyperlink
> <http://patft.uspto.gov/netacgi/nph-Parser?Sect1=PTO1&Sect2=HITOFF&d=PALL&p=1&u=%2Fnetahtml%2FPTO%2Fsrchnum.htm&r=1&f=G&l=50&s1=4,873,662.PN.&OS=PN/4,873,662&RS=PN/4,873,662 
> >?
> Enough said.". If you can't see the brilliance of the hyperlink you  
> must
> be a fool. When Doug Engelbart demostrated that concept in 1968 it was
> so brilliant it went over the heads of most of the audience, some of
> them making dumb remarks that entirely missed the point. That's pretty
> much conclusive proof that it was not something obvious to  
> practitioners
> in the art.
>
> The patent on hyperlinks was not bad because the idea was obvious. It
> was bad because it was applied for in 1980, 12 years after the concept
> was demonstrated. Luckily, video exists of the 1968 demo, and the  
> patent
> was shot down.
>
> If you can't get your act together about where real innovation lies,
> just shut up. It just makes arguing a meaningful case for a better
> patent system hard for the rest of us.
>
> Steve
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org


From derek at indranet.co.nz  Sat Feb  6 15:11:05 2010
From: derek at indranet.co.nz (Derek Smithies)
Date: Sun, 7 Feb 2010 12:11:05 +1300 (NZDT)
Subject: [Freeswitch-users] Simple IAX2 setup - help with converting
 from asterisk to freeswitch
In-Reply-To: <d2976552e9ba0ce1ed095b7cef22e1ad.squirrel@www.webcontracts.co.uk>
References: <91093d24f2166d220acf2ab19115f95d.squirrel@www.webcontracts.co.uk>
	<E1E3F6EF-1A1E-4891-B371-7A3BF0988E95@jerris.com>
	<d2976552e9ba0ce1ed095b7cef22e1ad.squirrel@www.webcontracts.co.uk>
Message-ID: <alpine.DEB.2.00.1002071152520.4493@monster>

On Fri, 5 Feb 2010, Matthew Law wrote:

> Why is that? - a lot of web pages I have read claim that IAX is more
> secure and efficient.  I have no problem with using SIP whatsoever and it
> certainly appears to be ubiquitous.  I am a complete newcomer to VoIP and
> I am trying to do this as securely as possible since I want to run
> freeswitch on a Xen VPS which will be visible on the internet.
>

Sigh.
  the web pages are wrong.

I have implemented the earlier version of IAX2 inside the opal library. At 
one stage, I could make a voip call using IAX2 to digiums test server.

The standard IAX2 codebase did not (at one stage) support silence 
suppression. With silence suppression, the bandwidth usage is halved. 
Using efficient packing of audio makes a few percent of difference. Doing 
silence suppression makes 50% of difference to bandwidth.

IAX2 is more efficient? Most(>90%) of the cpu work when doing voip is in 
the codec. By using a non standard approach (iax2 audio packets) for 
carrying audio, you will have a minimal gain in effiency (or cpu usage). 
Remember that iax2 uses the same codecs as in H.323/SIP. Oh - you are 
right, g711 is a codec, and has trivial cpu cost. However, with g711 any 
bandwidth you save on the header is negligible compared to the size of the 
encoded audio block.

Secure? Security protocols (HIP, ZRTP, etc) take years of careful 
development to get something that works well. I do not recall seeing 
evidence of years of development going into IAX2 security. Either the 
author of the security in IAX2 is a pure genius, able to do in 
days/weeks what other do in years,,,, or the iax2 security is 
average (at best).

Yes, all the iax2 packets go to the same port. This has huge advantages 
for getting through firewalls, and setting up firewalls to accept incoming 
calls. It does (slightly) increase the complexity inside the iax2 code.

===================================

if you want iax2 inside freeswitch, my suggestion is that the opal library 
needs to have the iax2 code there brought up to spec, and then used (in 
the same way as the H.323 component of the opal library).

Derek.
-- 
Derek Smithies Ph.D.
IndraNet Technologies Ltd.
Email: derek at indranet.co.nz
ph +64 3 365 6485
Web: http://www.indranet-technologies.com/



From gavin.henry at gmail.com  Sat Feb  6 15:47:01 2010
From: gavin.henry at gmail.com (Gavin Henry)
Date: Sat, 6 Feb 2010 23:47:01 +0000
Subject: [Freeswitch-users] Looking for some good/cheap desktop phones
In-Reply-To: <20100206200547.GD31942@base.carmickle.com>
References: <3f2e071b5aea5e11bb37a37e435a33b7@mail.gmail.com>
	<8e9d67561002042236s434e0643od322551de8720be1@mail.gmail.com>
	<409D0ACF-ADA4-4D02-8B7E-71EC1B1BAFBC@freeswitch.org>
	<8e9d67561002042301v76129abfj3dcbc70ed073752c@mail.gmail.com>
	<d361ca7b1002050507n5e564621oa6cf7ace005dd5b8@mail.gmail.com>
	<00ae01caa698$65b570f0$312052d0$@net>
	<b39273531002051154h5132faa9h9fb024ebdfb15b77@mail.gmail.com>
	<SNT119-W1223C808A5E092C7E2A590B5540@phx.gbl>
	<e90351e71002061124v129b081cg501fb0c6ae6a17dd@mail.gmail.com>
	<20100206200547.GD31942@base.carmickle.com>
Message-ID: <13ca621c1002061547n22353b27tefe78f28c66f073d@mail.gmail.com>

I would put Aastra before a snom, but then again Aastra always wants a reboot!

On 06/02/2010, Frank Carmickle <frank at carmickle.com> wrote:
> On Sat, Feb 06, Christian Jensen wrote:
>> The response on this thread has been great!
>>
>> I wonder if there should be a link off the FreeSwitch site for vendors for
>> these phones.... hint hint.
>>
>> >From what I have been hearing here and elsewhere, the preference is
>> Polycom,
>> Cisco, Grandstream, Aastra in that order.
>
> Hmmm...  That's not the impression that I got at all.  From what I picked up
> on this thread it would look more like
>
> polycom snom aastra cisco grandstream
>
> I'm sure others can speak to how Cisco doesn't do presence right.  They are
> harder to provision then Snoms and Polycoms and in my opinion don't sound as
> good either.  To me they sound dull and fuzzy on transmit.  I've heard about
> four different versions of 7940/7960.
>
> The Snom 3x0 series have suffered in the speaker phone department until
> version 7 firmware.  Version 8 betas have sounded even better yet.  Make
> sure that if your going to be using headsets with Snoms that you ground the
> phone to something.  This can be done by either having a shielded network
> cable attached to a grounded switch or computer, or order them with the
> grounded power adaptor.  They give you the not so grounded adaptor by
> default.  Finding shielded ethernet can be kind of tricky but it is nice to
> have.
>
> Polycom speech quality is my favorite if you can handle the ringy under
> water sounds of the background noise especially on speaker.  It sounds like
> there's a fish tank running near by if there's any white noise.  I believe
> this to be the case because the way they split the audio up in to many bands
> to process it.  Maybe less steep filters would help.
>
> It's hard to believe how many people just don't care about call quality.  It
> starts with a good sounding phone that doesn't echo or hum.  I'd like to
> hope that we are doing our part to help the world sound a little better all
> the time.
>
> --FC
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>

-- 
Sent from my mobile device

http://www.suretecsystems.com/services/openldap/
http://www.suretectelecom.com


From gavin.henry at gmail.com  Sat Feb  6 16:00:07 2010
From: gavin.henry at gmail.com (Gavin Henry)
Date: Sun, 7 Feb 2010 00:00:07 +0000
Subject: [Freeswitch-users] Can Freeswitch and OpenSER Co-reside on Same
	Server?
In-Reply-To: <2d9149cd1002051410o73076e0fs94ca76b4d2458801@mail.gmail.com>
References: <C4DC856C75094358AEDBE3FA0E242741@greyhawk.tonecommander.com>
	<2d9149cd1002051011x3b4f548dqabe615ed5f991687@mail.gmail.com>
	<970116.27641.qm@web44811.mail.sp1.yahoo.com>
	<2d9149cd1002051410o73076e0fs94ca76b4d2458801@mail.gmail.com>
Message-ID: <13ca621c1002061600x2bfe7ee4l166e6d5cec18f1ad@mail.gmail.com>

Hi,

I got asked to review this too as I submitted a few bugs in the book.
Their review guidelines are long and I don't have enough time to write
such an in depth review. I'll watch your blog though!

I hope the index is better this time as Packt always let me down that way.

Gavin.


On 05/02/2010, Kristian Kielhofner <kristian.kielhofner at gmail.com> wrote:
> On Fri, Feb 5, 2010 at 4:51 PM, Jim Thomas <jimthomasembedded at yahoo.com>
> wrote:
>> Kristian,
>>
>> I enjoyed your recent blog praising FreeSWITCH.
>
>   Thanks.
>
>> The world needs an O'Reilly book about FreeSWITCH.? Perhaps you could
>> write one in your spare time?
>
>   O'Reilly authors have a joke: "The only people that get rich writing
> books are Stephen King and J.K. Rowling".  I'm not picking on
> O'Reilly; it doesn't matter who the publisher is.  Writing a tech book
> takes an immense amount of time and compared to consulting and other
> ways to spend your limited time it (often) doesn't pay well enough.
>
>   Certainly some people are in different situation.  In the next
> couple of days I'll be reviewing Packt Publishing's new OpenSIPS book
> written by Flavio Goncalves.  Flavio has an OpenSIPS training business
> so writing a book is perfect for him.  You get to train with the guy
> who literally wrote the book!
>
>> Whoever gets there first and does it well is likely to own that shelf
>> space for a long time to come.
>
>   There are some people working on a FreeSWITCH book and it makes
> sense for them too.  I'm sure they'll do a great job.
>
> --
> Kristian Kielhofner
> http://www.astlinux.org
> http://blog.krisk.org
> http://www.star2star.com
> http://www.submityoursip.com
> http://www.voalte.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>

-- 
Sent from my mobile device

http://www.suretecsystems.com/services/openldap/
http://www.suretectelecom.com


From matt at webcontracts.co.uk  Sat Feb  6 16:05:47 2010
From: matt at webcontracts.co.uk (Matthew Law)
Date: Sun, 7 Feb 2010 00:05:47 -0000
Subject: [Freeswitch-users] Simple IAX2 setup - help with converting
 from asterisk to freeswitch
In-Reply-To: <alpine.DEB.2.00.1002071152520.4493@monster>
References: <91093d24f2166d220acf2ab19115f95d.squirrel@www.webcontracts.co.uk>
	<E1E3F6EF-1A1E-4891-B371-7A3BF0988E95@jerris.com>
	<d2976552e9ba0ce1ed095b7cef22e1ad.squirrel@www.webcontracts.co.uk>
	<alpine.DEB.2.00.1002071152520.4493@monster>
Message-ID: <65e4725010ae312e26887c33624c1bb9.squirrel@www.webcontracts.co.uk>

Thanks for the comprehensive reply.  I have switched to using plain SIP
and the experiments continue...

Matt

On Sat, February 6, 2010 11:11 pm, Derek Smithies wrote:
> Sigh.
>   the web pages are wrong.
>
> I have implemented the earlier version of IAX2 inside the opal library. At
> one stage, I could make a voip call using IAX2 to digiums test server.
>
> The standard IAX2 codebase did not (at one stage) support silence
> suppression. With silence suppression, the bandwidth usage is halved.
> Using efficient packing of audio makes a few percent of difference. Doing
> silence suppression makes 50% of difference to bandwidth.
>
> IAX2 is more efficient? Most(>90%) of the cpu work when doing voip is in
> the codec. By using a non standard approach (iax2 audio packets) for
> carrying audio, you will have a minimal gain in effiency (or cpu usage).
> Remember that iax2 uses the same codecs as in H.323/SIP. Oh - you are
> right, g711 is a codec, and has trivial cpu cost. However, with g711 any
> bandwidth you save on the header is negligible compared to the size of the
> encoded audio block.
>
> Secure? Security protocols (HIP, ZRTP, etc) take years of careful
> development to get something that works well. I do not recall seeing
> evidence of years of development going into IAX2 security. Either the
> author of the security in IAX2 is a pure genius, able to do in
> days/weeks what other do in years,,,, or the iax2 security is
> average (at best).
>
> Yes, all the iax2 packets go to the same port. This has huge advantages
> for getting through firewalls, and setting up firewalls to accept incoming
> calls. It does (slightly) increase the complexity inside the iax2 code.
>
> ===================================
>
> if you want iax2 inside freeswitch, my suggestion is that the opal library
> needs to have the iax2 code there brought up to spec, and then used (in
> the same way as the H.323 component of the opal library).



From matt at webcontracts.co.uk  Sat Feb  6 16:24:39 2010
From: matt at webcontracts.co.uk (Matthew Law)
Date: Sun, 7 Feb 2010 00:24:39 -0000
Subject: [Freeswitch-users] ACL question and js error
Message-ID: <ea05d45dca9be2d979fb386f960bc077.squirrel@www.webcontracts.co.uk>

After some more experiments I have a working replacement for the asterisk
box we were using before, which is great.

I had problems getting incoming calls to work. Changing the entry in
acl.conf.xml from:

<list name="domains" default="deny">
   <node type="allow" domain="$${domain}"/>
</list>

to:

<list name="domains" default="allow">
   <node type="allow" domain="$${domain}"/>
</list>

and reloading xml works but this gets reverted every time FS starts up. 
I've scanned the wiki docs and can't see anything pertaining to that. 
Why/where is this happening and how do I make it the default?  Actually,
the question should probably be is it sensible to do that? - the box is
out on the internet and I really only want to take incoming calls from
voiptalk.org, but I can't find a list of IPs on their site which I could
create an acl from...

Second question: I have tried this example for an answer machine (mainly
because it looked the shortest and simplest of the examples listed):

http://wiki.freeswitch.org/wiki/Examples_answermachine

I get this error:

2010-02-06 19:01:26.799118 [ERR] answermachine.js:135  ReferenceError:
email is not defined

Which is relating to this line:

email(eMailFrom, eMailTo, "Subject: " + tmp_eMailSubject, eMailBody,
tmp_Filename);

The script says it requires mod_spidermonkey_etpan, but I can't find any
reference to that anywhere (I'm using svn trunk).  Where is it?

Thanks,

Matt - roughly 1% up the FreeSWITCH learning curve and climbing...

PS: a nice FreeSWITCH 'cookbook' which starts out at the most simple
example and goes on to add more juicy, *working* features would get my
money any day. Hint, hint! :-)



From costa.zikalala at gmail.com  Sun Feb  7 06:36:43 2010
From: costa.zikalala at gmail.com (Costa Zikalala)
Date: Sun, 7 Feb 2010 16:36:43 +0200
Subject: [Freeswitch-users] Difference Between "bridge" and "transfer"
In-Reply-To: <4468a6771002060255l582856e7g577d63d0b596b173@mail.gmail.com>
References: <D123D09A29B942D988B3456528DD44AD@greyhawk.tonecommander.com> 
	<87f2f3b91002041632t19f17aefoaf11e048150816c5@mail.gmail.com> 
	<59daa2cd1002051044p1bee852ele8d0d972c303ea0e@mail.gmail.com> 
	<4468a6771002060255l582856e7g577d63d0b596b173@mail.gmail.com>
Message-ID: <59daa2cd1002070636p7d28b1b3s4bae8ee7c9c69fbf@mail.gmail.com>

Thanks
Thanks Ognjen for that extra bit of info.


On 6 February 2010 12:55, Ognjen Seslija <oseslija at gmail.com> wrote:
>
> There is a way on most ISDN lines to "deflect" call back to the telco
> switch with the forward extension information, so the switch can call the
> b-leg itself and properly present the original clid. FS then doesn't need to
> anything but to respond with the information.
>
>
> On Fri, Feb 5, 2010 at 7:44 PM, Costa Zikalala <costa.zikalala at gmail.com>wrote:
>
>> Whilst on this subject, if I receive a call from PSTN and I 'bridge' it to
>> another PSTN extension, will the PSTN Provider charge me for the 'b-leg' or
>> will the original caller be charged the entire call?
>>
>>
>>
>> On 5 February 2010 02:32, Michael Collins <msc at freeswitch.org> wrote:
>>
>>>
>>>
>>> On Thu, Feb 4, 2010 at 3:56 PM, Jerry Richards <
>>> jerry.richards at teotech.com> wrote:
>>>
>>>> What is the difference between "bridge" and "transfer"?  I'm looking at
>>>> the
>>>> demo IVRs.
>>>>
>>>>
>>> bridge will connect two endpoints together while transfer sends the
>>> endpoint back through the dialplan again...
>>>
>>> -MC
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100207/0393f070/attachment.html 

From frank at carmickle.com  Sun Feb  7 06:59:07 2010
From: frank at carmickle.com (Frank Carmickle)
Date: Sun, 7 Feb 2010 09:59:07 -0500
Subject: [Freeswitch-users] ACL question and js error
In-Reply-To: <ea05d45dca9be2d979fb386f960bc077.squirrel@www.webcontracts.co.uk>
References: <ea05d45dca9be2d979fb386f960bc077.squirrel@www.webcontracts.co.uk>
Message-ID: <20100207145907.GF31942@base.carmickle.com>

On Sun, Feb 07, Matthew Law wrote:
> After some more experiments I have a working replacement for the asterisk
> box we were using before, which is great.
> 
> I had problems getting incoming calls to work. Changing the entry in
> acl.conf.xml from:
> 
> <list name="domains" default="deny">
>    <node type="allow" domain="$${domain}"/>
> </list>
> 
> to:
> 
> <list name="domains" default="allow">
>    <node type="allow" domain="$${domain}"/>
> </list>
> 
> and reloading xml works but this gets reverted every time FS starts up. 
> I've scanned the wiki docs and can't see anything pertaining to that. 
> Why/where is this happening and how do I make it the default?  Actually,
> the question should probably be is it sensible to do that? - the box is
> out on the internet and I really only want to take incoming calls from
> voiptalk.org, but I can't find a list of IPs on their site which I could
> create an acl from...

This is what gateway definitions are for in sofia.

> 
> Second question: I have tried this example for an answer machine (mainly
> because it looked the shortest and simplest of the examples listed):
> 
> http://wiki.freeswitch.org/wiki/Examples_answermachine

Is voicemail not what your looking for?

I understand the frustration of trying to get things working first run.  I found reading rereading and rereading the wiki to be most helpful.  You start to get a sense for how things work.  There are usually 100 ways to accomplish the same task in fs.  Over time you'll start to figure which ones work best for your setup.  You should jump in the weekly conf call.  Lots of people there can give you a hand.

--FC


From christian at officepools.com  Sun Feb  7 09:16:08 2010
From: christian at officepools.com (Christian Jensen)
Date: Sun, 7 Feb 2010 09:16:08 -0800
Subject: [Freeswitch-users] Looking for some good/cheap desktop phones
In-Reply-To: <00ae01caa698$65b570f0$312052d0$@net>
References: <3f2e071b5aea5e11bb37a37e435a33b7@mail.gmail.com>	
	<8e9d67561002042236s434e0643od322551de8720be1@mail.gmail.com>	
	<409D0ACF-ADA4-4D02-8B7E-71EC1B1BAFBC@freeswitch.org>	<8e9d67561002042301v76129abfj3dcbc70ed073752c@mail.gmail.com>	
	<d361ca7b1002050507n5e564621oa6cf7ace005dd5b8@mail.gmail.com> 
	<00ae01caa698$65b570f0$312052d0$@net>
Message-ID: <c7987ee2c6cbc7c0c148302dee8ee343@mail.gmail.com>

Where did you get these cheap phones?



*From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Adam Ford
*Sent:* Friday, February 05, 2010 11:21 AM
*To:* freeswitch-users at lists.freeswitch.org
*Subject:* Re: [Freeswitch-users] Looking for some good/cheap desktop phones



I just picked up old model Polycoms.  You can get the IP301?s for ~$60-70
new and the IP501s for ~$100 new.  They don?t have some of the fancier
features of the new Polycoms, but they carry the same quality and
configurability(with the exception of NAT).



-Adam



*From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Yehavi
Bourvine
*Sent:* Friday, February 05, 2010 6:07 AM
*To:* freeswitch-users at lists.freeswitch.org
*Subject:* Re: [Freeswitch-users] Looking for some good/cheap desktop phones



>From my experience Polycom and SNOM are expensive but give you what you
need. Polycom is more intutive to the users but more cumbersome for the
manager to deploy; SNOM is somewhat less intuitive to the user but
everything can be set via the WEB interface.



If you talk about 4-5 phones, then probably SNOM is the choice. It also
depends about the specific functions you want to use. I our specific
environment (high use of BLF and shared lines) Polycom wins because it
handles these functions just as the user expects.



I did not try Aastra so cannot testify. We did test Yealink, Thomson,
Asterphone, SipTip and maybe others I forgot. Cisco also seems good but
Cisco does not supply the required socumentation to make them fully working.



                    Regards, __Yehavi:

2010/2/5 ????? ??????? <tim at novion.ru>

Sure, those phones do not deliver superior usability, but they at
least give the best sound among budget models.


2010/2/5 Brian West <brian at freeswitch.org>:

> And all of those are awful phones.  They don't even make good paper
weights.
>
> You can't have good and cheap in the same sentence when talking about VoIP
phones.  You have to take your pick between quality (good) and price (cheap)
you can't have both at once.
>
> /b
>
>
> On Feb 5, 2010, at 12:36 AM, ????? ??????? wrote:
>
>> Have a look at Yealink (Skypemate) and Fanvill
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100207/4444caf4/attachment.html 

From vmaruani at interwise.com  Sun Feb  7 07:00:55 2010
From: vmaruani at interwise.com (Victor Maruani)
Date: Sun, 7 Feb 2010 17:00:55 +0200
Subject: [Freeswitch-users] Bypass-media and REFER method
Message-ID: <B816EF85C732094BA2016B2112F3DC2B0402C777@isr-brass.interwise.com>

Hi,

 

I'm trying to do a POC using FS, the goal is to have FS handle REFERs
containing proprietary data. 

I want to have some logic on top of FS and also use the fail over
mechanism.

in short, I have something like this:

(third party) A side  --- FS ---- B side (IVR server)

 

the IVR the sends a REFER to FS. I don't want A to deal with it. 

now say B refers to C, it would be considered as a "group" C1, C2 ... to
which I want FS to failover.

only when one has answered should A be updated (REINVITE) and B notified
and disconnected.

if all fails I would expect B to be notified of the failure and proceed
as I wish without "losing" A.

 

from what I've read FS should be OK for the job but I have a couple
issues:

 

1 ) I have some issues getting FS handle a REFER while in bypass-media
mode.

(I tried with the release and some revisions including latest)

first when I bridge A and B everything is fine and media is bypassed.

When B sends REFER to C:

-           FS immediately NOTIFY B of success and send a reinvite to A
with SDP containing its own media IP/port.

-          then it does INVITE C with A's SDP.

-          B gets disconnected. A is not updated with C's sdp.

so at this point A sends RTP to FS and C sends RTP to A. ...

 

I basically have one extension for B: (set bypass-media and bridge to B)

and another extension to C which does the same actions.

what do you think I do wrong? 

 

 

2 ) how can I catch the REFER and set variables from it? (like ref-by or
ref-to)

in the dial plan I do catch the INVITE sent to C, but how to do it with
the REFER itself?

 

 

thanks for your help!

 

 

Best Regards,

Victor.

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100207/2992971b/attachment-0001.html 

From tculjaga at gmail.com  Sun Feb  7 13:17:40 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Sun, 7 Feb 2010 22:17:40 +0100
Subject: [Freeswitch-users] Looking for some good/cheap desktop phones
In-Reply-To: <20100206200547.GD31942@base.carmickle.com>
References: <3f2e071b5aea5e11bb37a37e435a33b7@mail.gmail.com>
	<8e9d67561002042236s434e0643od322551de8720be1@mail.gmail.com>
	<409D0ACF-ADA4-4D02-8B7E-71EC1B1BAFBC@freeswitch.org>
	<8e9d67561002042301v76129abfj3dcbc70ed073752c@mail.gmail.com>
	<d361ca7b1002050507n5e564621oa6cf7ace005dd5b8@mail.gmail.com>
	<00ae01caa698$65b570f0$312052d0$@net>
	<b39273531002051154h5132faa9h9fb024ebdfb15b77@mail.gmail.com>
	<SNT119-W1223C808A5E092C7E2A590B5540@phx.gbl>
	<e90351e71002061124v129b081cg501fb0c6ae6a17dd@mail.gmail.com>
	<20100206200547.GD31942@base.carmickle.com>
Message-ID: <65d96fc81002071317l69f8d8ccw8f6abee2f4265bd4@mail.gmail.com>

On Sat, Feb 6, 2010 at 9:05 PM, Frank Carmickle <frank at carmickle.com> wrote:

> On Sat, Feb 06, Christian Jensen wrote:
> > The response on this thread has been great!
> >
> > I wonder if there should be a link off the FreeSwitch site for vendors
> for
> > these phones.... hint hint.
> >
> > >From what I have been hearing here and elsewhere, the preference is
> Polycom,
> > Cisco, Grandstream, Aastra in that order.
>
> Hmmm...  That's not the impression that I got at all.  From what I picked
> up on this thread it would look more like
>
> polycom snom aastra cisco grandstream
>

Try Avaya 9600 SIP it is really great but it costs :)
It's always the same story ... there is not cheap and good ... it just
depends of what you can be satisfied for..
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100207/b1f6ec7e/attachment.html 

From matt at webcontracts.co.uk  Sun Feb  7 14:23:15 2010
From: matt at webcontracts.co.uk (Matthew Law)
Date: Sun, 7 Feb 2010 22:23:15 -0000
Subject: [Freeswitch-users] ACL question and js error
In-Reply-To: <20100207145907.GF31942@base.carmickle.com>
References: <ea05d45dca9be2d979fb386f960bc077.squirrel@www.webcontracts.co.uk>
	<20100207145907.GF31942@base.carmickle.com>
Message-ID: <f14a5f708124b6c1ca692e2ef390749e.squirrel@www.webcontracts.co.uk>


On Sun, February 7, 2010 2:59 pm, Frank Carmickle wrote:
> This is what gateway definitions are for in sofia.
>
>>
>> Second question: I have tried this example for an answer machine (mainly
>> because it looked the shortest and simplest of the examples listed):
>>
>> http://wiki.freeswitch.org/wiki/Examples_answermachine
>
> Is voicemail not what your looking for?
>
> I understand the frustration of trying to get things working first run.  I
> found reading rereading and rereading the wiki to be most helpful.  You
> start to get a sense for how things work.  There are usually 100 ways to
> accomplish the same task in fs.  Over time you'll start to figure which
> ones work best for your setup.  You should jump in the weekly conf call.
> Lots of people there can give you a hand.

I'll read the wiki again :-)

What I would like to do for the moment is route all calls to extension 200
if it is logged in, ring for 20 seconds then go to the answermachine.  If
200 is not logged in, then go straight to answermachine.  Answermachine in
the current context could, I suppose, be a custom voicemail message for
200, but I do need it to be emailed to our group address.

Eventually I want several extensions in a group called 'support'.  Each
one with their own voicemail so they can receive messages from other
extensions but I want external callers to go to the 'answermachine'
voicemail as before.

What would be really cool is if I could wire that to RT so that it raises
a support ticket with the message attached if the caller ID was recognised
as one of our customers.  If not, it should just get emailed to our group
email address to be picked up by someone as per usual.

Yes, I am finding FS difficult mainly because all of the information is
there somewhere, it is just difficult to piece it all together.  One thing
is becoming clear: it is very powerful indeed and like many things in Unix
land, it is the way you can chain each individual component together that
makes it so capable.

Thanks,

Matt.





From Prometheus001 at gmx.net  Sun Feb  7 14:28:42 2010
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Sun, 07 Feb 2010 23:28:42 +0100
Subject: [Freeswitch-users] mod_skypiax - help needed with
 modified	snd-dummy (one way audio)
In-Reply-To: <191c3a031002060558r36947d93vcbb46df9e0a7dfde@mail.gmail.com>
References: <4B60555B.2020004@gmx.net>	<7b197bef1001281341h7ee2c58fj79b3886630901f29@mail.gmail.com>	<4B62C962.7000601@gmx.net>	<8213d6071001290403g2c08fe39gd92eb0446bb6485a@mail.gmail.com>	<7b197bef1001290520x37179cb1o63273410e9ef5bc@mail.gmail.com>	<4B6C1CCB.4080606@gmx.net>	<7b197bef1002050536j7ea09684v913434296ae52b97@mail.gmail.com>	<4B6C868B.3040406@gmx.net>	<7b197bef1002051319l2fcb5617p1691692f58c65372@mail.gmail.com>	<4B6CC792.5060608@gmx.net>
	<191c3a031002060558r36947d93vcbb46df9e0a7dfde@mail.gmail.com>
Message-ID: <4B6F3E9A.2020103@gmx.net>

I now used the static Skype binary in order to avoid missing constraints
to other libraries: It still crashes
1st it starts with:
  process 15431: D-Bus library appears to be incorrectly set up; failed
to read machine uuid: Failed to open "/var/lib/dbus/machine-id": No such
file or directory
  See the manual page for dbus-uuidgen to correct this issue.
After calling this client it crashes with:
   /usr/bin/skype: symbol lookup error: /usr/bin/skype: undefined
symbol: _ZN10QBoxLayout13addSpacerItemEP11QSpacerItem

Any hints, where I may get an older Skype client?

Best regards
Peter

Anthony Minessale schrieb:
>
> Quick fyi centos 5.4 has some memory bugs in libc don't go beyond 5.3
> until its fixed.
>
>> On Feb 5, 2010 7:42 PM, "Peter P GMX" <Prometheus001 at gmx.net
>> <mailto:Prometheus001 at gmx.net>> wrote:
>>
>> Skype starts, but as soon as it receives a call it crashes with:
>>
>> /usr/bin/skype: symbol lookup error: /usr/bin/skype: undefined symbol:
>>
>> _ZN10QBoxLayout13addSpacerItemEP11QSpacerItem
>>
>> I think the 8.10 version dos not work with8.04.
>>
>> Any hints, where I may get an older Skype client? I may also try the
>> static skype client.
>>
>>
>> Best regards
>> Peter
>>
>>
>>
>> .
>> Giovanni Maruzzelli schrieb:
>> > that's not at all a fatal error.
>> > I believe...
>>
> ------------------------------------------------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   


From wiltingtree at gmail.com  Sun Feb  7 18:31:40 2010
From: wiltingtree at gmail.com (Adam Wilt)
Date: Sun, 7 Feb 2010 21:31:40 -0500
Subject: [Freeswitch-users] Bridging sessions from two separate lua scripts;
	uuid_bridge? 	intercept?
Message-ID: <eb7797d71002071831w27301c5dva767b82b38c2ce6f@mail.gmail.com>

Hi. I have two sessions running in two separate Lua scripts, and I want to
bridge them so that the bridged call is being controlled by the first
(a-leg) script.
If I simply use uuid_bridge, I get no error but the calls don't bridge.
I've tried intercept, but I don't understand how it should be used; nothing
I try seems to work.
Here's what I have:

function bridge_calls(session,api,b_leg_uuid, call_len)
  session:setAutoHangup(false)
  session:execute("sched_hangup","+" .. tostring(call_len) .. " " ..
tostring(session.uuid))
  session:execute("set","continue_on_fail=true")
  api:executeString("intercept -bleg " .. tostring(b_leg_uuid))
  api:executeString("uuid_bridge " .. tostring(session.uuid) .. " " ..
tostring(b_leg_uuid))
end

I'd really appreciate any help.

Thanks,
Adam
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100207/115218cc/attachment.html 

From msc at freeswitch.org  Sun Feb  7 18:50:33 2010
From: msc at freeswitch.org (Michael S Collins)
Date: Sun, 7 Feb 2010 20:50:33 -0600
Subject: [Freeswitch-users] Bridging sessions from two separate lua
	scripts; uuid_bridge? 	intercept?
In-Reply-To: <eb7797d71002071831w27301c5dva767b82b38c2ce6f@mail.gmail.com>
References: <eb7797d71002071831w27301c5dva767b82b38c2ce6f@mail.gmail.com>
Message-ID: <C23AA196-E851-4D5B-B1B5-2C2F32F6BB68@freeswitch.org>

Pastebin a debug log so we can see what is happening when the script  
runs.

-MC

Sent from my iPhone

On Feb 7, 2010, at 8:31 PM, Adam Wilt <wiltingtree at gmail.com> wrote:

> Hi. I have two sessions running in two separate Lua scripts, and I  
> want to bridge them so that the bridged call is being controlled by  
> the first (a-leg) script.
> If I simply use uuid_bridge, I get no error but the calls don't  
> bridge.
> I've tried intercept, but I don't understand how it should be used;  
> nothing I try seems to work.
> Here's what I have:
>
> function bridge_calls(session,api,b_leg_uuid, call_len)
>   session:setAutoHangup(false)
>   session:execute("sched_hangup","+" .. tostring(call_len) .. " " ..  
> tostring(session.uuid))
>   session:execute("set","continue_on_fail=true")
>   api:executeString("intercept -bleg " .. tostring(b_leg_uuid))
>   api:executeString("uuid_bridge " .. tostring(session.uuid) .. "  
> " .. tostring(b_leg_uuid))
> end
>
> I'd really appreciate any help.
>
> Thanks,
> Adam
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100207/655fb680/attachment.html 

From wiltingtree at gmail.com  Sun Feb  7 19:29:30 2010
From: wiltingtree at gmail.com (Adam Wilt)
Date: Sun, 7 Feb 2010 22:29:30 -0500
Subject: [Freeswitch-users] Bridging sessions from two separate lua
	scripts; uuid_bridge? intercept?
In-Reply-To: <C23AA196-E851-4D5B-B1B5-2C2F32F6BB68@freeswitch.org>
References: <eb7797d71002071831w27301c5dva767b82b38c2ce6f@mail.gmail.com>
	<C23AA196-E851-4D5B-B1B5-2C2F32F6BB68@freeswitch.org>
Message-ID: <eb7797d71002071929pb53e1adue2e91e6bdb598251@mail.gmail.com>

Thanks Michael for the reply.
Here's the pastebin link: http://pastebin.freeswitch.org/12084


On Sun, Feb 7, 2010 at 9:50 PM, Michael S Collins <msc at freeswitch.org>wrote:

> Pastebin a debug log so we can see what is happening when the script runs.
>
> -MC
>
> Sent from my iPhone
>
> On Feb 7, 2010, at 8:31 PM, Adam Wilt <wiltingtree at gmail.com> wrote:
>
> Hi. I have two sessions running in two separate Lua scripts, and I want to
> bridge them so that the bridged call is being controlled by the first
> (a-leg) script.
> If I simply use uuid_bridge, I get no error but the calls don't bridge.
> I've tried intercept, but I don't understand how it should be used; nothing
> I try seems to work.
> Here's what I have:
>
> function bridge_calls(session,api,b_leg_uuid, call_len)
>   session:setAutoHangup(false)
>   session:execute("sched_hangup","+" .. tostring(call_len) .. " " ..
> tostring(session.uuid))
>   session:execute("set","continue_on_fail=true")
>   api:executeString("intercept -bleg " .. tostring(b_leg_uuid))
>   api:executeString("uuid_bridge " .. tostring(session.uuid) .. " " ..
> tostring(b_leg_uuid))
> end
>
> I'd really appreciate any help.
>
> Thanks,
> Adam
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100207/079392c4/attachment.html 

From kristian.kielhofner at gmail.com  Sun Feb  7 21:06:03 2010
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Mon, 8 Feb 2010 00:06:03 -0500
Subject: [Freeswitch-users] mod_limit requires media?
Message-ID: <2d9149cd1002072106n7272cb80se3329700f0074d52@mail.gmail.com>

Hello everyone,

  I was playing with mod_limit earlier tonight and I noticed that it
essentially stopped hashing/tracking calls once bypass media was set.
Is this by design?  Is there some reason mod_limit requires media?
Other than that mod_limit looks to be very well implemented (no
surprise there) and I'm excited to put it to more use.

Thanks!

-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com


From codecomplete at free.fr  Mon Feb  8 03:07:05 2010
From: codecomplete at free.fr (Fred-145)
Date: Mon, 08 Feb 2010 12:07:05 +0100
Subject: [Freeswitch-users] Driving peripherals through Freeswitch?
Message-ID: <2srvm5945qgcno44oetn9ngii0u3aed73p@4ax.com>

Hello

I don't know anything about this, but I was wondering if someone had
successfully used a Freeswitch server to drive peripherals like
switching on a heater by sending an SMS or calling an extension, etc.?

I'm thinking of tools like X10 to drive peripherals from a PC.

Has someone played with this kind of tool and could tell me what is
technically possible?

Thank you for any feedback.



From codecomplete at free.fr  Mon Feb  8 03:14:31 2010
From: codecomplete at free.fr (Fred-145)
Date: Mon, 08 Feb 2010 12:14:31 +0100
Subject: [Freeswitch-users] Looking for some good/cheap desktop phones
References: <3f2e071b5aea5e11bb37a37e435a33b7@mail.gmail.com>
Message-ID: <2asvm5105hn2d6pdtdqlnrd8hmc3btg80a@4ax.com>

On Thu, 4 Feb 2010 14:08:56 -0800, Christain Jensen
<christian at officepools.com> wrote:
>I am looking for a vendor for some (3-5) desktop voip phones. Any
>suggestions?

Based on positive feedback from VoIP forums, I bought the DECT Siemens
A580IP a couple of weeks ago for about 70? (sales tax excluded).

Don't know how well it compares to higher-end solutions, but it's been
working fine so far once I was told how to solve some compatibility
issue to get it working with Eyebeam/XLite.



From codecomplete at free.fr  Mon Feb  8 03:10:55 2010
From: codecomplete at free.fr (Fred-145)
Date: Mon, 08 Feb 2010 12:10:55 +0100
Subject: [Freeswitch-users] Looking for some good/cheap desktop phones
References: <8e9d67561002042236s434e0643od322551de8720be1@mail.gmail.com>
	<409D0ACF-ADA4-4D02-8B7E-71EC1B1BAFBC@freeswitch.org>
	<8e9d67561002042301v76129abfj3dcbc70ed073752c@mail.gmail.com>
	<d361ca7b1002050507n5e564621oa6cf7ace005dd5b8@mail.gmail.com>
	<65d96fc81002050521w746707b1q4398e6d8a0692d1e@mail.gmail.com>
	<65d96fc81002050521w746707b1q4398e6d8a0692d1e-JsoAwUIsXosN+BqQ9rBEUg@public.gmane.org>
	<6B520D1B-EDD6-44F7-9DE5-A384D3286138@freeswitch.org>
Message-ID: <u6svm5t18meap4cd6t673o03f8gvosvgos@4ax.com>

On Fri, 5 Feb 2010 09:13:29 -0600, Brian West
<brian at freeswitch.org> wrote:
>Sigh... When is someone actually going to build an open platform voip hardware phone

I guess David Rowe is the person who could pull this off

http://www.rowetel.com/blog/?page_id=2



From mike at yes.net.ua  Mon Feb  8 03:19:09 2010
From: mike at yes.net.ua (Mike Tkachuk)
Date: Mon, 8 Feb 2010 13:19:09 +0200
Subject: [Freeswitch-users] Dialplan search order
Message-ID: <798899361.20100208131909@yes.net.ua>

Hello FS gurus,

  I'm using xml_curl for external dialplan fetch, but I like to
  split static and dynamic parts of configuration, so for example,
  leave call unloop logic, and voicemail extension in static xml file
  while having all other parts dynamic. It will allow to avoid
  unnecessary call to costly external source and also avoid xml parse
  of content that is static.

  Currently FS first look in xml_curl and only after that falls back
  to static files. Is that behavior possible to change, so FS will work
  like that:

  1 - Look in static xml file and execute all extensions that have
  'continue="true"'
  2 - If previous step didn't stop on matching extension than look in
  xml_curl or other source specified in dialplan param of sofia
  config.

  Looks like <param name="dialplan" value="XML:/path_to_dialplan_default.xml,XML"/>
  don't do the trick.

  I'm sorry if my question is really noobish or goes against FS
  logic. Thanks in advance.


--
Mike Tkachuk



From gmaruzz at celliax.org  Mon Feb  8 05:29:36 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Mon, 8 Feb 2010 14:29:36 +0100
Subject: [Freeswitch-users] mod_skypiax - help needed with modified
	snd-dummy (one way audio)
In-Reply-To: <4B6F3E9A.2020103@gmx.net>
References: <4B60555B.2020004@gmx.net>
	<8213d6071001290403g2c08fe39gd92eb0446bb6485a@mail.gmail.com> 
	<7b197bef1001290520x37179cb1o63273410e9ef5bc@mail.gmail.com> 
	<4B6C1CCB.4080606@gmx.net>
	<7b197bef1002050536j7ea09684v913434296ae52b97@mail.gmail.com> 
	<4B6C868B.3040406@gmx.net>
	<7b197bef1002051319l2fcb5617p1691692f58c65372@mail.gmail.com> 
	<4B6CC792.5060608@gmx.net>
	<191c3a031002060558r36947d93vcbb46df9e0a7dfde@mail.gmail.com> 
	<4B6F3E9A.2020103@gmx.net>
Message-ID: <7b197bef1002080529u1afc7f9av492b0c221ee2d6af@mail.gmail.com>

Peter,

I just tested with the static build you find on skype.com

I never tested for performances or other issues, there may be (it's a beta).

But it do not crash on me.

I have no problem at all.

If you can give me ssh access I can try to understand why you have so
many problems.

Or, alternatively, try to follow the wiki. You know, I've not heard
about those problems.

root at hardy64:~/skype.beta/skype_static-2.1.0.81# ldd skype
        linux-gate.so.1 =>  (0xffffe000)
        libasound.so.2 => /usr/lib32/libasound.so.2 (0xf7eb7000)
        libXv.so.1 => /usr/lib32/libXv.so.1 (0xf7eb2000)
        libXss.so.1 => /usr/lib32/libXss.so.1 (0xf7eae000)
        libSM.so.6 => /usr/lib32/libSM.so.6 (0xf7ea6000)
        libICE.so.6 => /usr/lib32/libICE.so.6 (0xf7e8e000)
        libXi.so.6 => /usr/lib32/libXi.so.6 (0xf7e86000)
        libXrender.so.1 => /usr/lib32/libXrender.so.1 (0xf7e7e000)
        libXrandr.so.2 => /usr/lib32/libXrandr.so.2 (0xf7e78000)
        libfreetype.so.6 => /usr/lib32/libfreetype.so.6 (0xf7e0a000)
        libfontconfig.so.1 => /usr/lib32/libfontconfig.so.1 (0xf7de0000)
        libXext.so.6 => /usr/lib32/libXext.so.6 (0xf7dd2000)
        libX11.so.6 => /usr/lib32/libX11.so.6 (0xf7ceb000)
        libz.so.1 => /usr/lib32/libz.so.1 (0xf7cd6000)
        libgthread-2.0.so.0 => /usr/lib32/libgthread-2.0.so.0 (0xf7cd1000)
        libglib-2.0.so.0 => /usr/lib32/libglib-2.0.so.0 (0xf7c1f000)
        librt.so.1 => /lib32/librt.so.1 (0xf7c16000)
        libdl.so.2 => /lib32/libdl.so.2 (0xf7c12000)
        libpthread.so.0 => /lib32/libpthread.so.0 (0xf7bfa000)
        libstdc++.so.6 => /usr/lib32/libstdc++.so.6 (0xf7b07000)
        libm.so.6 => /lib32/libm.so.6 (0xf7ae2000)
        libgcc_s.so.1 => /usr/lib32/libgcc_s.so.1 (0xf7ad6000)
        libc.so.6 => /lib32/libc.so.6 (0xf7987000)
        libexpat.so.1 => /usr/lib32/libexpat.so.1 (0xf7966000)
        libXau.so.6 => /usr/lib32/libXau.so.6 (0xf7963000)
        libxcb-xlib.so.0 => /usr/lib32/libxcb-xlib.so.0 (0xf7961000)
        libxcb.so.1 => /usr/lib32/libxcb.so.1 (0xf7948000)
        libpcre.so.3 => /usr/lib32/libpcre.so.3 (0xf7921000)
        /lib/ld-linux.so.2 (0xf7f86000)
        libXdmcp.so.6 => /usr/lib32/libXdmcp.so.6 (0xf791c000)


On Sun, Feb 7, 2010 at 11:28 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
> I now used the static Skype binary in order to avoid missing constraints
> to other libraries: It still crashes
> 1st it starts with:
> ?process 15431: D-Bus library appears to be incorrectly set up; failed
> to read machine uuid: Failed to open "/var/lib/dbus/machine-id": No such
> file or directory
> ?See the manual page for dbus-uuidgen to correct this issue.
> After calling this client it crashes with:
> ? /usr/bin/skype: symbol lookup error: /usr/bin/skype: undefined
> symbol: _ZN10QBoxLayout13addSpacerItemEP11QSpacerItem
>
> Any hints, where I may get an older Skype client?
>
> Best regards
> Peter
>
> Anthony Minessale schrieb:
>>
>> Quick fyi centos 5.4 has some memory bugs in libc don't go beyond 5.3
>> until its fixed.
>>
>>> On Feb 5, 2010 7:42 PM, "Peter P GMX" <Prometheus001 at gmx.net
>>> <mailto:Prometheus001 at gmx.net>> wrote:
>>>
>>> Skype starts, but as soon as it receives a call it crashes with:
>>>
>>> /usr/bin/skype: symbol lookup error: /usr/bin/skype: undefined symbol:
>>>
>>> _ZN10QBoxLayout13addSpacerItemEP11QSpacerItem
>>>
>>> I think the 8.10 version dos not work with8.04.
>>>
>>> Any hints, where I may get an older Skype client? I may also try the
>>> static skype client.
>>>
>>>
>>> Best regards
>>> Peter
>>>
>>>
>>>
>>> .
>>> Giovanni Maruzzelli schrieb:
>>> > that's not at all a fatal error.
>>> > I believe...
>>>
>> ------------------------------------------------------------------------
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From max.bridgewater at gmail.com  Mon Feb  8 05:46:42 2010
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Mon, 8 Feb 2010 08:46:42 -0500
Subject: [Freeswitch-users] mod_skypiax - help needed with modified
	snd-dummy (one way audio)
In-Reply-To: <7b197bef1002080529u1afc7f9av492b0c221ee2d6af@mail.gmail.com>
References: <4B60555B.2020004@gmx.net>
	<7b197bef1001290520x37179cb1o63273410e9ef5bc@mail.gmail.com>
	<4B6C1CCB.4080606@gmx.net>
	<7b197bef1002050536j7ea09684v913434296ae52b97@mail.gmail.com>
	<4B6C868B.3040406@gmx.net>
	<7b197bef1002051319l2fcb5617p1691692f58c65372@mail.gmail.com>
	<4B6CC792.5060608@gmx.net>
	<191c3a031002060558r36947d93vcbb46df9e0a7dfde@mail.gmail.com>
	<4B6F3E9A.2020103@gmx.net>
	<7b197bef1002080529u1afc7f9av492b0c221ee2d6af@mail.gmail.com>
Message-ID: <c55d46951002080546p829a881w256bfdb19334c548@mail.gmail.com>

Interesting; a while back I tried to install Skypiax with the latest
static build on Skype.com. I had QT library compatibility problem on a
CentOS 5.4 (Final). Giovannni, may I ask what Linux distro you are
using?

Thanks,
max.

On Mon, Feb 8, 2010 at 8:29 AM, Giovanni Maruzzelli <gmaruzz at celliax.org> wrote:
> Peter,
>
> I just tested with the static build you find on skype.com
>
> I never tested for performances or other issues, there may be (it's a beta).
>
> But it do not crash on me.
>
> I have no problem at all.
>
> If you can give me ssh access I can try to understand why you have so
> many problems.
>
> Or, alternatively, try to follow the wiki. You know, I've not heard
> about those problems.
>
> root at hardy64:~/skype.beta/skype_static-2.1.0.81# ldd skype
> ? ? ? ?linux-gate.so.1 => ?(0xffffe000)
> ? ? ? ?libasound.so.2 => /usr/lib32/libasound.so.2 (0xf7eb7000)
> ? ? ? ?libXv.so.1 => /usr/lib32/libXv.so.1 (0xf7eb2000)
> ? ? ? ?libXss.so.1 => /usr/lib32/libXss.so.1 (0xf7eae000)
> ? ? ? ?libSM.so.6 => /usr/lib32/libSM.so.6 (0xf7ea6000)
> ? ? ? ?libICE.so.6 => /usr/lib32/libICE.so.6 (0xf7e8e000)
> ? ? ? ?libXi.so.6 => /usr/lib32/libXi.so.6 (0xf7e86000)
> ? ? ? ?libXrender.so.1 => /usr/lib32/libXrender.so.1 (0xf7e7e000)
> ? ? ? ?libXrandr.so.2 => /usr/lib32/libXrandr.so.2 (0xf7e78000)
> ? ? ? ?libfreetype.so.6 => /usr/lib32/libfreetype.so.6 (0xf7e0a000)
> ? ? ? ?libfontconfig.so.1 => /usr/lib32/libfontconfig.so.1 (0xf7de0000)
> ? ? ? ?libXext.so.6 => /usr/lib32/libXext.so.6 (0xf7dd2000)
> ? ? ? ?libX11.so.6 => /usr/lib32/libX11.so.6 (0xf7ceb000)
> ? ? ? ?libz.so.1 => /usr/lib32/libz.so.1 (0xf7cd6000)
> ? ? ? ?libgthread-2.0.so.0 => /usr/lib32/libgthread-2.0.so.0 (0xf7cd1000)
> ? ? ? ?libglib-2.0.so.0 => /usr/lib32/libglib-2.0.so.0 (0xf7c1f000)
> ? ? ? ?librt.so.1 => /lib32/librt.so.1 (0xf7c16000)
> ? ? ? ?libdl.so.2 => /lib32/libdl.so.2 (0xf7c12000)
> ? ? ? ?libpthread.so.0 => /lib32/libpthread.so.0 (0xf7bfa000)
> ? ? ? ?libstdc++.so.6 => /usr/lib32/libstdc++.so.6 (0xf7b07000)
> ? ? ? ?libm.so.6 => /lib32/libm.so.6 (0xf7ae2000)
> ? ? ? ?libgcc_s.so.1 => /usr/lib32/libgcc_s.so.1 (0xf7ad6000)
> ? ? ? ?libc.so.6 => /lib32/libc.so.6 (0xf7987000)
> ? ? ? ?libexpat.so.1 => /usr/lib32/libexpat.so.1 (0xf7966000)
> ? ? ? ?libXau.so.6 => /usr/lib32/libXau.so.6 (0xf7963000)
> ? ? ? ?libxcb-xlib.so.0 => /usr/lib32/libxcb-xlib.so.0 (0xf7961000)
> ? ? ? ?libxcb.so.1 => /usr/lib32/libxcb.so.1 (0xf7948000)
> ? ? ? ?libpcre.so.3 => /usr/lib32/libpcre.so.3 (0xf7921000)
> ? ? ? ?/lib/ld-linux.so.2 (0xf7f86000)
> ? ? ? ?libXdmcp.so.6 => /usr/lib32/libXdmcp.so.6 (0xf791c000)
>
>
> On Sun, Feb 7, 2010 at 11:28 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>> I now used the static Skype binary in order to avoid missing constraints
>> to other libraries: It still crashes
>> 1st it starts with:
>> ?process 15431: D-Bus library appears to be incorrectly set up; failed
>> to read machine uuid: Failed to open "/var/lib/dbus/machine-id": No such
>> file or directory
>> ?See the manual page for dbus-uuidgen to correct this issue.
>> After calling this client it crashes with:
>> ? /usr/bin/skype: symbol lookup error: /usr/bin/skype: undefined
>> symbol: _ZN10QBoxLayout13addSpacerItemEP11QSpacerItem
>>
>> Any hints, where I may get an older Skype client?
>>
>> Best regards
>> Peter
>>
>> Anthony Minessale schrieb:
>>>
>>> Quick fyi centos 5.4 has some memory bugs in libc don't go beyond 5.3
>>> until its fixed.
>>>
>>>> On Feb 5, 2010 7:42 PM, "Peter P GMX" <Prometheus001 at gmx.net
>>>> <mailto:Prometheus001 at gmx.net>> wrote:
>>>>
>>>> Skype starts, but as soon as it receives a call it crashes with:
>>>>
>>>> /usr/bin/skype: symbol lookup error: /usr/bin/skype: undefined symbol:
>>>>
>>>> _ZN10QBoxLayout13addSpacerItemEP11QSpacerItem
>>>>
>>>> I think the 8.10 version dos not work with8.04.
>>>>
>>>> Any hints, where I may get an older Skype client? I may also try the
>>>> static skype client.
>>>>
>>>>
>>>> Best regards
>>>> Peter
>>>>
>>>>
>>>>
>>>> .
>>>> Giovanni Maruzzelli schrieb:
>>>> > that's not at all a fatal error.
>>>> > I believe...
>>>>
>>> ------------------------------------------------------------------------
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From gmaruzz at celliax.org  Mon Feb  8 05:53:04 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Mon, 8 Feb 2010 14:53:04 +0100
Subject: [Freeswitch-users] mod_skypiax - help needed with modified
	snd-dummy (one way audio)
In-Reply-To: <c55d46951002080546p829a881w256bfdb19334c548@mail.gmail.com>
References: <4B60555B.2020004@gmx.net> <4B6C1CCB.4080606@gmx.net> 
	<7b197bef1002050536j7ea09684v913434296ae52b97@mail.gmail.com> 
	<4B6C868B.3040406@gmx.net>
	<7b197bef1002051319l2fcb5617p1691692f58c65372@mail.gmail.com> 
	<4B6CC792.5060608@gmx.net>
	<191c3a031002060558r36947d93vcbb46df9e0a7dfde@mail.gmail.com> 
	<4B6F3E9A.2020103@gmx.net>
	<7b197bef1002080529u1afc7f9av492b0c221ee2d6af@mail.gmail.com> 
	<c55d46951002080546p829a881w256bfdb19334c548@mail.gmail.com>
Message-ID: <7b197bef1002080553qd1cb42dla26425942de83c43@mail.gmail.com>

Peter is using hardy 64 bit. I checked on that.

But, let me understand: if you're using a static build, why you have a
problem with QT?
Is actually Qt to be statically linked...

what is the results of:

ldd skype

Anyway, in a short while I'll check that on centos 5.4 too, and I'll report here

-giovanni

On Mon, Feb 8, 2010 at 2:46 PM, Max Bridgewater
<max.bridgewater at gmail.com> wrote:
> Interesting; a while back I tried to install Skypiax with the latest
> static build on Skype.com. I had QT library compatibility problem on a
> CentOS 5.4 (Final). Giovannni, may I ask what Linux distro you are
> using?
>
> Thanks,
> max.
>
> On Mon, Feb 8, 2010 at 8:29 AM, Giovanni Maruzzelli <gmaruzz at celliax.org> wrote:
>> Peter,
>>
>> I just tested with the static build you find on skype.com
>>
>> I never tested for performances or other issues, there may be (it's a beta).
>>
>> But it do not crash on me.
>>
>> I have no problem at all.
>>
>> If you can give me ssh access I can try to understand why you have so
>> many problems.
>>
>> Or, alternatively, try to follow the wiki. You know, I've not heard
>> about those problems.
>>
>> root at hardy64:~/skype.beta/skype_static-2.1.0.81# ldd skype
>> ? ? ? ?linux-gate.so.1 => ?(0xffffe000)
>> ? ? ? ?libasound.so.2 => /usr/lib32/libasound.so.2 (0xf7eb7000)
>> ? ? ? ?libXv.so.1 => /usr/lib32/libXv.so.1 (0xf7eb2000)
>> ? ? ? ?libXss.so.1 => /usr/lib32/libXss.so.1 (0xf7eae000)
>> ? ? ? ?libSM.so.6 => /usr/lib32/libSM.so.6 (0xf7ea6000)
>> ? ? ? ?libICE.so.6 => /usr/lib32/libICE.so.6 (0xf7e8e000)
>> ? ? ? ?libXi.so.6 => /usr/lib32/libXi.so.6 (0xf7e86000)
>> ? ? ? ?libXrender.so.1 => /usr/lib32/libXrender.so.1 (0xf7e7e000)
>> ? ? ? ?libXrandr.so.2 => /usr/lib32/libXrandr.so.2 (0xf7e78000)
>> ? ? ? ?libfreetype.so.6 => /usr/lib32/libfreetype.so.6 (0xf7e0a000)
>> ? ? ? ?libfontconfig.so.1 => /usr/lib32/libfontconfig.so.1 (0xf7de0000)
>> ? ? ? ?libXext.so.6 => /usr/lib32/libXext.so.6 (0xf7dd2000)
>> ? ? ? ?libX11.so.6 => /usr/lib32/libX11.so.6 (0xf7ceb000)
>> ? ? ? ?libz.so.1 => /usr/lib32/libz.so.1 (0xf7cd6000)
>> ? ? ? ?libgthread-2.0.so.0 => /usr/lib32/libgthread-2.0.so.0 (0xf7cd1000)
>> ? ? ? ?libglib-2.0.so.0 => /usr/lib32/libglib-2.0.so.0 (0xf7c1f000)
>> ? ? ? ?librt.so.1 => /lib32/librt.so.1 (0xf7c16000)
>> ? ? ? ?libdl.so.2 => /lib32/libdl.so.2 (0xf7c12000)
>> ? ? ? ?libpthread.so.0 => /lib32/libpthread.so.0 (0xf7bfa000)
>> ? ? ? ?libstdc++.so.6 => /usr/lib32/libstdc++.so.6 (0xf7b07000)
>> ? ? ? ?libm.so.6 => /lib32/libm.so.6 (0xf7ae2000)
>> ? ? ? ?libgcc_s.so.1 => /usr/lib32/libgcc_s.so.1 (0xf7ad6000)
>> ? ? ? ?libc.so.6 => /lib32/libc.so.6 (0xf7987000)
>> ? ? ? ?libexpat.so.1 => /usr/lib32/libexpat.so.1 (0xf7966000)
>> ? ? ? ?libXau.so.6 => /usr/lib32/libXau.so.6 (0xf7963000)
>> ? ? ? ?libxcb-xlib.so.0 => /usr/lib32/libxcb-xlib.so.0 (0xf7961000)
>> ? ? ? ?libxcb.so.1 => /usr/lib32/libxcb.so.1 (0xf7948000)
>> ? ? ? ?libpcre.so.3 => /usr/lib32/libpcre.so.3 (0xf7921000)
>> ? ? ? ?/lib/ld-linux.so.2 (0xf7f86000)
>> ? ? ? ?libXdmcp.so.6 => /usr/lib32/libXdmcp.so.6 (0xf791c000)
>>
>>
>> On Sun, Feb 7, 2010 at 11:28 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>> I now used the static Skype binary in order to avoid missing constraints
>>> to other libraries: It still crashes
>>> 1st it starts with:
>>> ?process 15431: D-Bus library appears to be incorrectly set up; failed
>>> to read machine uuid: Failed to open "/var/lib/dbus/machine-id": No such
>>> file or directory
>>> ?See the manual page for dbus-uuidgen to correct this issue.
>>> After calling this client it crashes with:
>>> ? /usr/bin/skype: symbol lookup error: /usr/bin/skype: undefined
>>> symbol: _ZN10QBoxLayout13addSpacerItemEP11QSpacerItem
>>>
>>> Any hints, where I may get an older Skype client?
>>>
>>> Best regards
>>> Peter
>>>
>>> Anthony Minessale schrieb:
>>>>
>>>> Quick fyi centos 5.4 has some memory bugs in libc don't go beyond 5.3
>>>> until its fixed.
>>>>
>>>>> On Feb 5, 2010 7:42 PM, "Peter P GMX" <Prometheus001 at gmx.net
>>>>> <mailto:Prometheus001 at gmx.net>> wrote:
>>>>>
>>>>> Skype starts, but as soon as it receives a call it crashes with:
>>>>>
>>>>> /usr/bin/skype: symbol lookup error: /usr/bin/skype: undefined symbol:
>>>>>
>>>>> _ZN10QBoxLayout13addSpacerItemEP11QSpacerItem
>>>>>
>>>>> I think the 8.10 version dos not work with8.04.
>>>>>
>>>>> Any hints, where I may get an older Skype client? I may also try the
>>>>> static skype client.
>>>>>
>>>>>
>>>>> Best regards
>>>>> Peter
>>>>>
>>>>>
>>>>>
>>>>> .
>>>>> Giovanni Maruzzelli schrieb:
>>>>> > that's not at all a fatal error.
>>>>> > I believe...
>>>>>
>>>> ------------------------------------------------------------------------
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> Cell : +39-347-2665618
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From max.bridgewater at gmail.com  Mon Feb  8 06:04:15 2010
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Mon, 8 Feb 2010 09:04:15 -0500
Subject: [Freeswitch-users] mod_skypiax - help needed with modified
	snd-dummy (one way audio)
In-Reply-To: <7b197bef1002080553qd1cb42dla26425942de83c43@mail.gmail.com>
References: <4B60555B.2020004@gmx.net>
	<7b197bef1002050536j7ea09684v913434296ae52b97@mail.gmail.com>
	<4B6C868B.3040406@gmx.net>
	<7b197bef1002051319l2fcb5617p1691692f58c65372@mail.gmail.com>
	<4B6CC792.5060608@gmx.net>
	<191c3a031002060558r36947d93vcbb46df9e0a7dfde@mail.gmail.com>
	<4B6F3E9A.2020103@gmx.net>
	<7b197bef1002080529u1afc7f9av492b0c221ee2d6af@mail.gmail.com>
	<c55d46951002080546p829a881w256bfdb19334c548@mail.gmail.com>
	<7b197bef1002080553qd1cb42dla26425942de83c43@mail.gmail.com>
Message-ID: <c55d46951002080604k400344b0i8add00b5c6014958@mail.gmail.com>

Well, it was a couple of weeks ago. I then switched back to 2.0.4. I'm
going to try it again and let you know.

Max.

On Mon, Feb 8, 2010 at 8:53 AM, Giovanni Maruzzelli <gmaruzz at celliax.org> wrote:
> Peter is using hardy 64 bit. I checked on that.
>
> But, let me understand: if you're using a static build, why you have a
> problem with QT?
> Is actually Qt to be statically linked...
>
> what is the results of:
>
> ldd skype
>
> Anyway, in a short while I'll check that on centos 5.4 too, and I'll report here
>
> -giovanni
>
> On Mon, Feb 8, 2010 at 2:46 PM, Max Bridgewater
> <max.bridgewater at gmail.com> wrote:
>> Interesting; a while back I tried to install Skypiax with the latest
>> static build on Skype.com. I had QT library compatibility problem on a
>> CentOS 5.4 (Final). Giovannni, may I ask what Linux distro you are
>> using?
>>
>> Thanks,
>> max.
>>
>> On Mon, Feb 8, 2010 at 8:29 AM, Giovanni Maruzzelli <gmaruzz at celliax.org> wrote:
>>> Peter,
>>>
>>> I just tested with the static build you find on skype.com
>>>
>>> I never tested for performances or other issues, there may be (it's a beta).
>>>
>>> But it do not crash on me.
>>>
>>> I have no problem at all.
>>>
>>> If you can give me ssh access I can try to understand why you have so
>>> many problems.
>>>
>>> Or, alternatively, try to follow the wiki. You know, I've not heard
>>> about those problems.
>>>
>>> root at hardy64:~/skype.beta/skype_static-2.1.0.81# ldd skype
>>> ? ? ? ?linux-gate.so.1 => ?(0xffffe000)
>>> ? ? ? ?libasound.so.2 => /usr/lib32/libasound.so.2 (0xf7eb7000)
>>> ? ? ? ?libXv.so.1 => /usr/lib32/libXv.so.1 (0xf7eb2000)
>>> ? ? ? ?libXss.so.1 => /usr/lib32/libXss.so.1 (0xf7eae000)
>>> ? ? ? ?libSM.so.6 => /usr/lib32/libSM.so.6 (0xf7ea6000)
>>> ? ? ? ?libICE.so.6 => /usr/lib32/libICE.so.6 (0xf7e8e000)
>>> ? ? ? ?libXi.so.6 => /usr/lib32/libXi.so.6 (0xf7e86000)
>>> ? ? ? ?libXrender.so.1 => /usr/lib32/libXrender.so.1 (0xf7e7e000)
>>> ? ? ? ?libXrandr.so.2 => /usr/lib32/libXrandr.so.2 (0xf7e78000)
>>> ? ? ? ?libfreetype.so.6 => /usr/lib32/libfreetype.so.6 (0xf7e0a000)
>>> ? ? ? ?libfontconfig.so.1 => /usr/lib32/libfontconfig.so.1 (0xf7de0000)
>>> ? ? ? ?libXext.so.6 => /usr/lib32/libXext.so.6 (0xf7dd2000)
>>> ? ? ? ?libX11.so.6 => /usr/lib32/libX11.so.6 (0xf7ceb000)
>>> ? ? ? ?libz.so.1 => /usr/lib32/libz.so.1 (0xf7cd6000)
>>> ? ? ? ?libgthread-2.0.so.0 => /usr/lib32/libgthread-2.0.so.0 (0xf7cd1000)
>>> ? ? ? ?libglib-2.0.so.0 => /usr/lib32/libglib-2.0.so.0 (0xf7c1f000)
>>> ? ? ? ?librt.so.1 => /lib32/librt.so.1 (0xf7c16000)
>>> ? ? ? ?libdl.so.2 => /lib32/libdl.so.2 (0xf7c12000)
>>> ? ? ? ?libpthread.so.0 => /lib32/libpthread.so.0 (0xf7bfa000)
>>> ? ? ? ?libstdc++.so.6 => /usr/lib32/libstdc++.so.6 (0xf7b07000)
>>> ? ? ? ?libm.so.6 => /lib32/libm.so.6 (0xf7ae2000)
>>> ? ? ? ?libgcc_s.so.1 => /usr/lib32/libgcc_s.so.1 (0xf7ad6000)
>>> ? ? ? ?libc.so.6 => /lib32/libc.so.6 (0xf7987000)
>>> ? ? ? ?libexpat.so.1 => /usr/lib32/libexpat.so.1 (0xf7966000)
>>> ? ? ? ?libXau.so.6 => /usr/lib32/libXau.so.6 (0xf7963000)
>>> ? ? ? ?libxcb-xlib.so.0 => /usr/lib32/libxcb-xlib.so.0 (0xf7961000)
>>> ? ? ? ?libxcb.so.1 => /usr/lib32/libxcb.so.1 (0xf7948000)
>>> ? ? ? ?libpcre.so.3 => /usr/lib32/libpcre.so.3 (0xf7921000)
>>> ? ? ? ?/lib/ld-linux.so.2 (0xf7f86000)
>>> ? ? ? ?libXdmcp.so.6 => /usr/lib32/libXdmcp.so.6 (0xf791c000)
>>>
>>>
>>> On Sun, Feb 7, 2010 at 11:28 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>> I now used the static Skype binary in order to avoid missing constraints
>>>> to other libraries: It still crashes
>>>> 1st it starts with:
>>>> ?process 15431: D-Bus library appears to be incorrectly set up; failed
>>>> to read machine uuid: Failed to open "/var/lib/dbus/machine-id": No such
>>>> file or directory
>>>> ?See the manual page for dbus-uuidgen to correct this issue.
>>>> After calling this client it crashes with:
>>>> ? /usr/bin/skype: symbol lookup error: /usr/bin/skype: undefined
>>>> symbol: _ZN10QBoxLayout13addSpacerItemEP11QSpacerItem
>>>>
>>>> Any hints, where I may get an older Skype client?
>>>>
>>>> Best regards
>>>> Peter
>>>>
>>>> Anthony Minessale schrieb:
>>>>>
>>>>> Quick fyi centos 5.4 has some memory bugs in libc don't go beyond 5.3
>>>>> until its fixed.
>>>>>
>>>>>> On Feb 5, 2010 7:42 PM, "Peter P GMX" <Prometheus001 at gmx.net
>>>>>> <mailto:Prometheus001 at gmx.net>> wrote:
>>>>>>
>>>>>> Skype starts, but as soon as it receives a call it crashes with:
>>>>>>
>>>>>> /usr/bin/skype: symbol lookup error: /usr/bin/skype: undefined symbol:
>>>>>>
>>>>>> _ZN10QBoxLayout13addSpacerItemEP11QSpacerItem
>>>>>>
>>>>>> I think the 8.10 version dos not work with8.04.
>>>>>>
>>>>>> Any hints, where I may get an older Skype client? I may also try the
>>>>>> static skype client.
>>>>>>
>>>>>>
>>>>>> Best regards
>>>>>> Peter
>>>>>>
>>>>>>
>>>>>>
>>>>>> .
>>>>>> Giovanni Maruzzelli schrieb:
>>>>>> > that's not at all a fatal error.
>>>>>> > I believe...
>>>>>>
>>>>> ------------------------------------------------------------------------
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>> Sincerely,
>>>
>>> Giovanni Maruzzelli
>>> Cell : +39-347-2665618
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From max.bridgewater at gmail.com  Mon Feb  8 07:06:22 2010
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Mon, 8 Feb 2010 10:06:22 -0500
Subject: [Freeswitch-users] CentOS 5.3 (Final) with Skype Static Build
Message-ID: <c55d46951002080706t6bb8531i4f12cb35af12e6de@mail.gmail.com>

Hi Giovanni,

Let me start a new thread so I don't hijack Peter's one. Here is what
I get for ldd /usr/bin/skype. What is the deal with these libXv.so.1,
libXss.so.1?

        linux-gate.so.1 =>  (0x00697000)
        libasound.so.2 => /lib/libasound.so.2 (0x008b7000)
        libXv.so.1 => not found
        libXss.so.1 => not found
        libSM.so.6 => /usr/lib/libSM.so.6 (0x00d02000)
        libICE.so.6 => /usr/lib/libICE.so.6 (0x0036b000)
        libXi.so.6 => /usr/lib/libXi.so.6 (0x00111000)
        libXrender.so.1 => /usr/lib/libXrender.so.1 (0x00119000)
        libXrandr.so.2 => /usr/lib/libXrandr.so.2 (0x00b77000)
        libfreetype.so.6 => /usr/lib/libfreetype.so.6 (0x00122000)
        libfontconfig.so.1 => /usr/lib/libfontconfig.so.1 (0x009fb000)
        libXext.so.6 => /usr/lib/libXext.so.6 (0x00eb0000)
        libX11.so.6 => /usr/lib/libX11.so.6 (0x001a2000)
        libz.so.1 => /usr/lib/libz.so.1 (0x002a5000)
        libgthread-2.0.so.0 => /lib/libgthread-2.0.so.0 (0x006bd000)
        libglib-2.0.so.0 => /lib/libglib-2.0.so.0 (0x002be000)
        librt.so.1 => /lib/librt.so.1 (0x009a1000)
        libdl.so.2 => /lib/libdl.so.2 (0x00604000)
        libpthread.so.0 => /lib/libpthread.so.0 (0x004e6000)
        libstdc++.so.6 => /usr/lib/libstdc++.so.6 (0x004fd000)
        libm.so.6 => /lib/libm.so.6 (0x00385000)
        libgcc_s.so.1 => /lib/libgcc_s.so.1 (0x00bcc000)
        libc.so.6 => /lib/libc.so.6 (0x006c2000)
        libexpat.so.0 => /lib/libexpat.so.0 (0x00c2a000)
        libXau.so.6 => /usr/lib/libXau.so.6 (0x002ba000)
        libXdmcp.so.6 => /usr/lib/libXdmcp.so.6 (0x0087e000)
        /lib/ld-linux.so.2 (0x00476000)

Max


From m.sobkow at marketelsystems.com  Mon Feb  8 07:06:55 2010
From: m.sobkow at marketelsystems.com (Mark Sobkow)
Date: Mon, 08 Feb 2010 09:06:55 -0600
Subject: [Freeswitch-users] IVR and Erlang
Message-ID: <4B70288F.6040003@marketelsystems.com>

I'm having a little trouble with Erlang and IVRs.  Specifically, the IVR 
configuration file is auto-loaded before the initialization code has a 
chance to tell Freeswitch to get it's configs from Erlang.  As a result, 
it's not querying Erlang for the configuration file.

Also, because there is no mod_ivr, I can't just do a "reload mod_ivr" 
after the Erlang communications has been initialized.

Any idea how to get Freeswitch to load it's IVR configs from Erlang?

-- 
Mark Sobkow
Senior Developer
MarkeTel Multi-Line Dialing Systems LTD.
428 Victoria Ave
Regina, SK S4N-0P6
Toll-Free: 800-289-8616-X533
Local: 306-359-6893-X533
Fax: 306-359-6879
Email: m.sobkow at marketelsystems.com
Web: http://www.marketelsystems.com 



From wiltingtree at gmail.com  Mon Feb  8 07:37:59 2010
From: wiltingtree at gmail.com (Adam Wilt)
Date: Mon, 8 Feb 2010 10:37:59 -0500
Subject: [Freeswitch-users] Bridging sessions from two separate lua
	scripts; uuid_bridge? intercept?
In-Reply-To: <eb7797d71002071929pb53e1adue2e91e6bdb598251@mail.gmail.com>
References: <eb7797d71002071831w27301c5dva767b82b38c2ce6f@mail.gmail.com>
	<C23AA196-E851-4D5B-B1B5-2C2F32F6BB68@freeswitch.org>
	<eb7797d71002071929pb53e1adue2e91e6bdb598251@mail.gmail.com>
Message-ID: <eb7797d71002080737k3f8fa5c9m8bf3ef2fcf42881e@mail.gmail.com>

One other thing I should mention. I'm running FreeSWITCH version 1.4 (build
14460) in Windows.
Brian suggested I upgrade to the build in the
http://files-sync.freeswitch.org/windows_installer/  folder, but it turned
out to be the exact same build I already had. I'd love to try upgrade to 1.5
in case this problem has been fixed already.


On Sun, Feb 7, 2010 at 10:29 PM, Adam Wilt <wiltingtree at gmail.com> wrote:

> Thanks Michael for the reply.
> Here's the pastebin link: http://pastebin.freeswitch.org/12084
>
>
> On Sun, Feb 7, 2010 at 9:50 PM, Michael S Collins <msc at freeswitch.org>wrote:
>
>> Pastebin a debug log so we can see what is happening when the script
>> runs.
>>
>> -MC
>>
>> Sent from my iPhone
>>
>> On Feb 7, 2010, at 8:31 PM, Adam Wilt <wiltingtree at gmail.com> wrote:
>>
>> Hi. I have two sessions running in two separate Lua scripts, and I want to
>> bridge them so that the bridged call is being controlled by the first
>> (a-leg) script.
>> If I simply use uuid_bridge, I get no error but the calls don't bridge.
>> I've tried intercept, but I don't understand how it should be used;
>> nothing I try seems to work.
>> Here's what I have:
>>
>> function bridge_calls(session,api,b_leg_uuid, call_len)
>>   session:setAutoHangup(false)
>>   session:execute("sched_hangup","+" .. tostring(call_len) .. " " ..
>> tostring(session.uuid))
>>   session:execute("set","continue_on_fail=true")
>>   api:executeString("intercept -bleg " .. tostring(b_leg_uuid))
>>   api:executeString("uuid_bridge " .. tostring(session.uuid) .. " " ..
>> tostring(b_leg_uuid))
>> end
>>
>> I'd really appreciate any help.
>>
>> Thanks,
>> Adam
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100208/e88d0652/attachment.html 

From m.sobkow at marketelsystems.com  Mon Feb  8 08:17:37 2010
From: m.sobkow at marketelsystems.com (Mark Sobkow)
Date: Mon, 08 Feb 2010 10:17:37 -0600
Subject: [Freeswitch-users] IVR and Erlang
In-Reply-To: <4B70288F.6040003@marketelsystems.com>
References: <4B70288F.6040003@marketelsystems.com>
Message-ID: <4B703921.4080909@marketelsystems.com>

Mark Sobkow wrote:
> I'm having a little trouble with Erlang and IVRs.  Specifically, the IVR 
> configuration file is auto-loaded before the initialization code has a 
> chance to tell Freeswitch to get it's configs from Erlang.  As a result, 
> it's not querying Erlang for the configuration file.
>
> Also, because there is no mod_ivr, I can't just do a "reload mod_ivr" 
> after the Erlang communications has been initialized.
>
> Any idea how to get Freeswitch to load it's IVR configs from Erlang?
>
>   
Mucking about this morning, I tried doing a "reloadxml", but that only 
seems to reload the timezones.conf from Erlang.

-- 
Mark Sobkow
Senior Developer
MarkeTel Multi-Line Dialing Systems LTD.
428 Victoria Ave
Regina, SK S4N-0P6
Toll-Free: 800-289-8616-X533
Local: 306-359-6893-X533
Fax: 306-359-6879
Email: m.sobkow at marketelsystems.com
Web: http://www.marketelsystems.com 



From Prometheus001 at gmx.net  Mon Feb  8 08:24:48 2010
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Mon, 08 Feb 2010 17:24:48 +0100
Subject: [Freeswitch-users] mod_skypiax - help needed with
 modified	snd-dummy (one way audio)
In-Reply-To: <c55d46951002080604k400344b0i8add00b5c6014958@mail.gmail.com>
References: <4B60555B.2020004@gmx.net>	<7b197bef1002050536j7ea09684v913434296ae52b97@mail.gmail.com>	<4B6C868B.3040406@gmx.net>	<7b197bef1002051319l2fcb5617p1691692f58c65372@mail.gmail.com>	<4B6CC792.5060608@gmx.net>	<191c3a031002060558r36947d93vcbb46df9e0a7dfde@mail.gmail.com>	<4B6F3E9A.2020103@gmx.net>	<7b197bef1002080529u1afc7f9av492b0c221ee2d6af@mail.gmail.com>	<c55d46951002080546p829a881w256bfdb19334c548@mail.gmail.com>	<7b197bef1002080553qd1cb42dla26425942de83c43@mail.gmail.com>
	<c55d46951002080604k400344b0i8add00b5c6014958@mail.gmail.com>
Message-ID: <4B703AD0.2080909@gmx.net>

I got it working now with static build and an older version of skype
(skype_static-2.1.0.47).

But I still have a problem ongoing with sound quality, resp. one way audio:
With original Alsa driver: When Skype calls mod_skypiax => SIP Phone,
then the sound from the SIP phone is interupted regularly 2 times a second.
Example:
When a person on the sip phone speaks
"aaaaaaaaaaaaaaaaaaaaaa"
the other site hears
"aaatataaaatataaaatataaaa"
With "t" meaning the interruption of the sound.

So I compliled and installed the modified alsa driver as described in
the wiki (configure, make and make install, remove old ubuntu sound dir
in /lib/modules/2.6.24-24-server/ubuntu/sound + depmod -a + reboot the
server.

Now the SIP phone is heard loud and clearly without interruption.
However the other direction is not heard, so we're at the beginning of
the post (one way audio). Only when I really scratch the microphone then
I hear some parts of this scratching on the SIP side.

So, some more hints are needed.

Here's the log, when I start the skype client:
  su root -c "/bin/echo 'username password'| DISPLAY=:101 
/usr/bin/skype1 --pipelogin &" &
  /usr/bin/Xvfb :102 -ac &
  error opening security policy file /etc/X11/xserver/SecurityPolicy
  expected keysym, got XF86KbdLightOnOff: line 70 of pc
  expected keysym, got XF86KbdBrightnessDown: line 71 of pc 
  expected keysym, got XF86KbdBrightnessUp: line 72 of pc
  Could not init font path element /usr/share/fonts/X11/cyrillic,
removing from list!
  Could not init font path element
/var/lib/defoma/x-ttcidfont-conf.d/dirs/TrueType, removing from list!
But these messages are not critical, right?

Best regards
Peter


Max Bridgewater schrieb:
> Well, it was a couple of weeks ago. I then switched back to 2.0.4. I'm
> going to try it again and let you know.
>
> Max.
>
> On Mon, Feb 8, 2010 at 8:53 AM, Giovanni Maruzzelli <gmaruzz at celliax.org> wrote:
>   
>> Peter is using hardy 64 bit. I checked on that.
>>
>> But, let me understand: if you're using a static build, why you have a
>> problem with QT?
>> Is actually Qt to be statically linked...
>>
>> what is the results of:
>>
>> ldd skype
>>
>> Anyway, in a short while I'll check that on centos 5.4 too, and I'll report here
>>
>> -giovanni
>>
>> On Mon, Feb 8, 2010 at 2:46 PM, Max Bridgewater
>> <max.bridgewater at gmail.com> wrote:
>>     
>>> Interesting; a while back I tried to install Skypiax with the latest
>>> static build on Skype.com. I had QT library compatibility problem on a
>>> CentOS 5.4 (Final). Giovannni, may I ask what Linux distro you are
>>> using?
>>>
>>> Thanks,
>>> max.
>>>
>>> On Mon, Feb 8, 2010 at 8:29 AM, Giovanni Maruzzelli <gmaruzz at celliax.org> wrote:
>>>       
>>>> Peter,
>>>>
>>>> I just tested with the static build you find on skype.com
>>>>
>>>> I never tested for performances or other issues, there may be (it's a beta).
>>>>
>>>> But it do not crash on me.
>>>>
>>>> I have no problem at all.
>>>>
>>>> If you can give me ssh access I can try to understand why you have so
>>>> many problems.
>>>>
>>>> Or, alternatively, try to follow the wiki. You know, I've not heard
>>>> about those problems.
>>>>
>>>> root at hardy64:~/skype.beta/skype_static-2.1.0.81# ldd skype
>>>>        linux-gate.so.1 =>  (0xffffe000)
>>>>        libasound.so.2 => /usr/lib32/libasound.so.2 (0xf7eb7000)
>>>>        libXv.so.1 => /usr/lib32/libXv.so.1 (0xf7eb2000)
>>>>        libXss.so.1 => /usr/lib32/libXss.so.1 (0xf7eae000)
>>>>        libSM.so.6 => /usr/lib32/libSM.so.6 (0xf7ea6000)
>>>>        libICE.so.6 => /usr/lib32/libICE.so.6 (0xf7e8e000)
>>>>        libXi.so.6 => /usr/lib32/libXi.so.6 (0xf7e86000)
>>>>        libXrender.so.1 => /usr/lib32/libXrender.so.1 (0xf7e7e000)
>>>>        libXrandr.so.2 => /usr/lib32/libXrandr.so.2 (0xf7e78000)
>>>>        libfreetype.so.6 => /usr/lib32/libfreetype.so.6 (0xf7e0a000)
>>>>        libfontconfig.so.1 => /usr/lib32/libfontconfig.so.1 (0xf7de0000)
>>>>        libXext.so.6 => /usr/lib32/libXext.so.6 (0xf7dd2000)
>>>>        libX11.so.6 => /usr/lib32/libX11.so.6 (0xf7ceb000)
>>>>        libz.so.1 => /usr/lib32/libz.so.1 (0xf7cd6000)
>>>>        libgthread-2.0.so.0 => /usr/lib32/libgthread-2.0.so.0 (0xf7cd1000)
>>>>        libglib-2.0.so.0 => /usr/lib32/libglib-2.0.so.0 (0xf7c1f000)
>>>>        librt.so.1 => /lib32/librt.so.1 (0xf7c16000)
>>>>        libdl.so.2 => /lib32/libdl.so.2 (0xf7c12000)
>>>>        libpthread.so.0 => /lib32/libpthread.so.0 (0xf7bfa000)
>>>>        libstdc++.so.6 => /usr/lib32/libstdc++.so.6 (0xf7b07000)
>>>>        libm.so.6 => /lib32/libm.so.6 (0xf7ae2000)
>>>>        libgcc_s.so.1 => /usr/lib32/libgcc_s.so.1 (0xf7ad6000)
>>>>        libc.so.6 => /lib32/libc.so.6 (0xf7987000)
>>>>        libexpat.so.1 => /usr/lib32/libexpat.so.1 (0xf7966000)
>>>>        libXau.so.6 => /usr/lib32/libXau.so.6 (0xf7963000)
>>>>        libxcb-xlib.so.0 => /usr/lib32/libxcb-xlib.so.0 (0xf7961000)
>>>>        libxcb.so.1 => /usr/lib32/libxcb.so.1 (0xf7948000)
>>>>        libpcre.so.3 => /usr/lib32/libpcre.so.3 (0xf7921000)
>>>>        /lib/ld-linux.so.2 (0xf7f86000)
>>>>        libXdmcp.so.6 => /usr/lib32/libXdmcp.so.6 (0xf791c000)
>>>>
>>>>
>>>> On Sun, Feb 7, 2010 at 11:28 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>         
>>>>> I now used the static Skype binary in order to avoid missing constraints
>>>>> to other libraries: It still crashes
>>>>> 1st it starts with:
>>>>>  process 15431: D-Bus library appears to be incorrectly set up; failed
>>>>> to read machine uuid: Failed to open "/var/lib/dbus/machine-id": No such
>>>>> file or directory
>>>>>  See the manual page for dbus-uuidgen to correct this issue.
>>>>> After calling this client it crashes with:
>>>>>   /usr/bin/skype: symbol lookup error: /usr/bin/skype: undefined
>>>>> symbol: _ZN10QBoxLayout13addSpacerItemEP11QSpacerItem
>>>>>
>>>>> Any hints, where I may get an older Skype client?
>>>>>
>>>>> Best regards
>>>>> Peter
>>>>>
>>>>> Anthony Minessale schrieb:
>>>>>           
>>>>>> Quick fyi centos 5.4 has some memory bugs in libc don't go beyond 5.3
>>>>>> until its fixed.
>>>>>>
>>>>>>             
>>>>>>> On Feb 5, 2010 7:42 PM, "Peter P GMX" <Prometheus001 at gmx.net
>>>>>>> <mailto:Prometheus001 at gmx.net>> wrote:
>>>>>>>
>>>>>>> Skype starts, but as soon as it receives a call it crashes with:
>>>>>>>
>>>>>>> /usr/bin/skype: symbol lookup error: /usr/bin/skype: undefined symbol:
>>>>>>>
>>>>>>> _ZN10QBoxLayout13addSpacerItemEP11QSpacerItem
>>>>>>>
>>>>>>> I think the 8.10 version dos not work with8.04.
>>>>>>>
>>>>>>> Any hints, where I may get an older Skype client? I may also try the
>>>>>>> static skype client.
>>>>>>>
>>>>>>>
>>>>>>> Best regards
>>>>>>> Peter
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> .
>>>>>>> Giovanni Maruzzelli schrieb:
>>>>>>>               
>>>>>>>> that's not at all a fatal error.
>>>>>>>> I believe...
>>>>>>>>                 
>>>>>> ------------------------------------------------------------------------
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>             
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>           
>>>>
>>>> --
>>>> Sincerely,
>>>>
>>>> Giovanni Maruzzelli
>>>> Cell : +39-347-2665618
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>         
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>       
>>
>> --
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> Cell : +39-347-2665618
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>     
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>   


From gmaruzz at celliax.org  Mon Feb  8 08:39:54 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Mon, 8 Feb 2010 17:39:54 +0100
Subject: [Freeswitch-users] mod_skypiax - help needed with modified
	snd-dummy (one way audio)
In-Reply-To: <4B703AD0.2080909@gmx.net>
References: <4B60555B.2020004@gmx.net>
	<7b197bef1002051319l2fcb5617p1691692f58c65372@mail.gmail.com> 
	<4B6CC792.5060608@gmx.net>
	<191c3a031002060558r36947d93vcbb46df9e0a7dfde@mail.gmail.com> 
	<4B6F3E9A.2020103@gmx.net>
	<7b197bef1002080529u1afc7f9av492b0c221ee2d6af@mail.gmail.com> 
	<c55d46951002080546p829a881w256bfdb19334c548@mail.gmail.com> 
	<7b197bef1002080553qd1cb42dla26425942de83c43@mail.gmail.com> 
	<c55d46951002080604k400344b0i8add00b5c6014958@mail.gmail.com> 
	<4B703AD0.2080909@gmx.net>
Message-ID: <7b197bef1002080839o61b4b1b5k6d5cc39bb05d9a32@mail.gmail.com>

Peter,

excuse me but I really do not follow you.

Why you have the normal static build not working?

Also, this is really taking too much of my time. You continue to
change things, and report issues, without waiting for solutions you
ask for, then you report something else, and so on... we'll never get
at the end of this.

If you want, please connect via IRC, and contact me (gmaruzz).
Or let me connect ssh to your machine.

-gm


On Mon, Feb 8, 2010 at 5:24 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
> I got it working now with static build and an older version of skype
> (skype_static-2.1.0.47).
>
> But I still have a problem ongoing with sound quality, resp. one way audio:
> With original Alsa driver: When Skype calls mod_skypiax => SIP Phone,
> then the sound from the SIP phone is interupted regularly 2 times a second.
> Example:
> When a person on the sip phone speaks
> "aaaaaaaaaaaaaaaaaaaaaa"
> the other site hears
> "aaatataaaatataaaatataaaa"
> With "t" meaning the interruption of the sound.
>
> So I compliled and installed the modified alsa driver as described in
> the wiki (configure, make and make install, remove old ubuntu sound dir
> in /lib/modules/2.6.24-24-server/ubuntu/sound + depmod -a + reboot the
> server.
>
> Now the SIP phone is heard loud and clearly without interruption.
> However the other direction is not heard, so we're at the beginning of
> the post (one way audio). Only when I really scratch the microphone then
> I hear some parts of this scratching on the SIP side.
>
> So, some more hints are needed.
>
> Here's the log, when I start the skype client:
> ?su root -c "/bin/echo 'username password'| DISPLAY=:101
> /usr/bin/skype1 --pipelogin &" &
> ?/usr/bin/Xvfb :102 -ac &
> ?error opening security policy file /etc/X11/xserver/SecurityPolicy
> ?expected keysym, got XF86KbdLightOnOff: line 70 of pc
> ?expected keysym, got XF86KbdBrightnessDown: line 71 of pc
> ?expected keysym, got XF86KbdBrightnessUp: line 72 of pc
> ?Could not init font path element /usr/share/fonts/X11/cyrillic,
> removing from list!
> ?Could not init font path element
> /var/lib/defoma/x-ttcidfont-conf.d/dirs/TrueType, removing from list!
> But these messages are not critical, right?
>
> Best regards
> Peter
>
>
> Max Bridgewater schrieb:
>> Well, it was a couple of weeks ago. I then switched back to 2.0.4. I'm
>> going to try it again and let you know.
>>
>> Max.
>>
>> On Mon, Feb 8, 2010 at 8:53 AM, Giovanni Maruzzelli <gmaruzz at celliax.org> wrote:
>>
>>> Peter is using hardy 64 bit. I checked on that.
>>>
>>> But, let me understand: if you're using a static build, why you have a
>>> problem with QT?
>>> Is actually Qt to be statically linked...
>>>
>>> what is the results of:
>>>
>>> ldd skype
>>>
>>> Anyway, in a short while I'll check that on centos 5.4 too, and I'll report here
>>>
>>> -giovanni
>>>
>>> On Mon, Feb 8, 2010 at 2:46 PM, Max Bridgewater
>>> <max.bridgewater at gmail.com> wrote:
>>>
>>>> Interesting; a while back I tried to install Skypiax with the latest
>>>> static build on Skype.com. I had QT library compatibility problem on a
>>>> CentOS 5.4 (Final). Giovannni, may I ask what Linux distro you are
>>>> using?
>>>>
>>>> Thanks,
>>>> max.
>>>>
>>>> On Mon, Feb 8, 2010 at 8:29 AM, Giovanni Maruzzelli <gmaruzz at celliax.org> wrote:
>>>>
>>>>> Peter,
>>>>>
>>>>> I just tested with the static build you find on skype.com
>>>>>
>>>>> I never tested for performances or other issues, there may be (it's a beta).
>>>>>
>>>>> But it do not crash on me.
>>>>>
>>>>> I have no problem at all.
>>>>>
>>>>> If you can give me ssh access I can try to understand why you have so
>>>>> many problems.
>>>>>
>>>>> Or, alternatively, try to follow the wiki. You know, I've not heard
>>>>> about those problems.
>>>>>
>>>>> root at hardy64:~/skype.beta/skype_static-2.1.0.81# ldd skype
>>>>> ? ? ? ?linux-gate.so.1 => ?(0xffffe000)
>>>>> ? ? ? ?libasound.so.2 => /usr/lib32/libasound.so.2 (0xf7eb7000)
>>>>> ? ? ? ?libXv.so.1 => /usr/lib32/libXv.so.1 (0xf7eb2000)
>>>>> ? ? ? ?libXss.so.1 => /usr/lib32/libXss.so.1 (0xf7eae000)
>>>>> ? ? ? ?libSM.so.6 => /usr/lib32/libSM.so.6 (0xf7ea6000)
>>>>> ? ? ? ?libICE.so.6 => /usr/lib32/libICE.so.6 (0xf7e8e000)
>>>>> ? ? ? ?libXi.so.6 => /usr/lib32/libXi.so.6 (0xf7e86000)
>>>>> ? ? ? ?libXrender.so.1 => /usr/lib32/libXrender.so.1 (0xf7e7e000)
>>>>> ? ? ? ?libXrandr.so.2 => /usr/lib32/libXrandr.so.2 (0xf7e78000)
>>>>> ? ? ? ?libfreetype.so.6 => /usr/lib32/libfreetype.so.6 (0xf7e0a000)
>>>>> ? ? ? ?libfontconfig.so.1 => /usr/lib32/libfontconfig.so.1 (0xf7de0000)
>>>>> ? ? ? ?libXext.so.6 => /usr/lib32/libXext.so.6 (0xf7dd2000)
>>>>> ? ? ? ?libX11.so.6 => /usr/lib32/libX11.so.6 (0xf7ceb000)
>>>>> ? ? ? ?libz.so.1 => /usr/lib32/libz.so.1 (0xf7cd6000)
>>>>> ? ? ? ?libgthread-2.0.so.0 => /usr/lib32/libgthread-2.0.so.0 (0xf7cd1000)
>>>>> ? ? ? ?libglib-2.0.so.0 => /usr/lib32/libglib-2.0.so.0 (0xf7c1f000)
>>>>> ? ? ? ?librt.so.1 => /lib32/librt.so.1 (0xf7c16000)
>>>>> ? ? ? ?libdl.so.2 => /lib32/libdl.so.2 (0xf7c12000)
>>>>> ? ? ? ?libpthread.so.0 => /lib32/libpthread.so.0 (0xf7bfa000)
>>>>> ? ? ? ?libstdc++.so.6 => /usr/lib32/libstdc++.so.6 (0xf7b07000)
>>>>> ? ? ? ?libm.so.6 => /lib32/libm.so.6 (0xf7ae2000)
>>>>> ? ? ? ?libgcc_s.so.1 => /usr/lib32/libgcc_s.so.1 (0xf7ad6000)
>>>>> ? ? ? ?libc.so.6 => /lib32/libc.so.6 (0xf7987000)
>>>>> ? ? ? ?libexpat.so.1 => /usr/lib32/libexpat.so.1 (0xf7966000)
>>>>> ? ? ? ?libXau.so.6 => /usr/lib32/libXau.so.6 (0xf7963000)
>>>>> ? ? ? ?libxcb-xlib.so.0 => /usr/lib32/libxcb-xlib.so.0 (0xf7961000)
>>>>> ? ? ? ?libxcb.so.1 => /usr/lib32/libxcb.so.1 (0xf7948000)
>>>>> ? ? ? ?libpcre.so.3 => /usr/lib32/libpcre.so.3 (0xf7921000)
>>>>> ? ? ? ?/lib/ld-linux.so.2 (0xf7f86000)
>>>>> ? ? ? ?libXdmcp.so.6 => /usr/lib32/libXdmcp.so.6 (0xf791c000)
>>>>>
>>>>>
>>>>> On Sun, Feb 7, 2010 at 11:28 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>
>>>>>> I now used the static Skype binary in order to avoid missing constraints
>>>>>> to other libraries: It still crashes
>>>>>> 1st it starts with:
>>>>>> ?process 15431: D-Bus library appears to be incorrectly set up; failed
>>>>>> to read machine uuid: Failed to open "/var/lib/dbus/machine-id": No such
>>>>>> file or directory
>>>>>> ?See the manual page for dbus-uuidgen to correct this issue.
>>>>>> After calling this client it crashes with:
>>>>>> ? /usr/bin/skype: symbol lookup error: /usr/bin/skype: undefined
>>>>>> symbol: _ZN10QBoxLayout13addSpacerItemEP11QSpacerItem
>>>>>>
>>>>>> Any hints, where I may get an older Skype client?
>>>>>>
>>>>>> Best regards
>>>>>> Peter
>>>>>>
>>>>>> Anthony Minessale schrieb:
>>>>>>
>>>>>>> Quick fyi centos 5.4 has some memory bugs in libc don't go beyond 5.3
>>>>>>> until its fixed.
>>>>>>>
>>>>>>>
>>>>>>>> On Feb 5, 2010 7:42 PM, "Peter P GMX" <Prometheus001 at gmx.net
>>>>>>>> <mailto:Prometheus001 at gmx.net>> wrote:
>>>>>>>>
>>>>>>>> Skype starts, but as soon as it receives a call it crashes with:
>>>>>>>>
>>>>>>>> /usr/bin/skype: symbol lookup error: /usr/bin/skype: undefined symbol:
>>>>>>>>
>>>>>>>> _ZN10QBoxLayout13addSpacerItemEP11QSpacerItem
>>>>>>>>
>>>>>>>> I think the 8.10 version dos not work with8.04.
>>>>>>>>
>>>>>>>> Any hints, where I may get an older Skype client? I may also try the
>>>>>>>> static skype client.
>>>>>>>>
>>>>>>>>
>>>>>>>> Best regards
>>>>>>>> Peter
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> .
>>>>>>>> Giovanni Maruzzelli schrieb:
>>>>>>>>
>>>>>>>>> that's not at all a fatal error.
>>>>>>>>> I believe...
>>>>>>>>>
>>>>>>> ------------------------------------------------------------------------
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>> --
>>>>> Sincerely,
>>>>>
>>>>> Giovanni Maruzzelli
>>>>> Cell : +39-347-2665618
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> --
>>> Sincerely,
>>>
>>> Giovanni Maruzzelli
>>> Cell : +39-347-2665618
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From gmaruzz at celliax.org  Mon Feb  8 08:42:49 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Mon, 8 Feb 2010 17:42:49 +0100
Subject: [Freeswitch-users] mod_skypiax - help needed with modified
	snd-dummy (one way audio)
In-Reply-To: <c55d46951002080604k400344b0i8add00b5c6014958@mail.gmail.com>
References: <4B60555B.2020004@gmx.net> <4B6C868B.3040406@gmx.net> 
	<7b197bef1002051319l2fcb5617p1691692f58c65372@mail.gmail.com> 
	<4B6CC792.5060608@gmx.net>
	<191c3a031002060558r36947d93vcbb46df9e0a7dfde@mail.gmail.com> 
	<4B6F3E9A.2020103@gmx.net>
	<7b197bef1002080529u1afc7f9av492b0c221ee2d6af@mail.gmail.com> 
	<c55d46951002080546p829a881w256bfdb19334c548@mail.gmail.com> 
	<7b197bef1002080553qd1cb42dla26425942de83c43@mail.gmail.com> 
	<c55d46951002080604k400344b0i8add00b5c6014958@mail.gmail.com>
Message-ID: <7b197bef1002080842v7a970d00l19df72d4f48a6ef6@mail.gmail.com>

Max,

Just checked, the static build of skype-beta works on latest centos (5.4)

you have to install:

yum install libXv
yum install libXScrnSaver

then use the snd-dummy supplied with centos (the modified one do not
works at the moment - because the skype-beta does some alsa calls was
not doing in previous versions, that are not implemented, I'll fix it
soon).

-gm

On Mon, Feb 8, 2010 at 3:04 PM, Max Bridgewater
<max.bridgewater at gmail.com> wrote:
> Well, it was a couple of weeks ago. I then switched back to 2.0.4. I'm
> going to try it again and let you know.
>
> Max.
>
> On Mon, Feb 8, 2010 at 8:53 AM, Giovanni Maruzzelli <gmaruzz at celliax.org> wrote:
>> Peter is using hardy 64 bit. I checked on that.
>>
>> But, let me understand: if you're using a static build, why you have a
>> problem with QT?
>> Is actually Qt to be statically linked...
>>
>> what is the results of:
>>
>> ldd skype
>>
>> Anyway, in a short while I'll check that on centos 5.4 too, and I'll report here
>>
>> -giovanni
>>
>> On Mon, Feb 8, 2010 at 2:46 PM, Max Bridgewater
>> <max.bridgewater at gmail.com> wrote:
>>> Interesting; a while back I tried to install Skypiax with the latest
>>> static build on Skype.com. I had QT library compatibility problem on a
>>> CentOS 5.4 (Final). Giovannni, may I ask what Linux distro you are
>>> using?
>>>
>>> Thanks,
>>> max.
>>>
>>> On Mon, Feb 8, 2010 at 8:29 AM, Giovanni Maruzzelli <gmaruzz at celliax.org> wrote:
>>>> Peter,
>>>>
>>>> I just tested with the static build you find on skype.com
>>>>
>>>> I never tested for performances or other issues, there may be (it's a beta).
>>>>
>>>> But it do not crash on me.
>>>>
>>>> I have no problem at all.
>>>>
>>>> If you can give me ssh access I can try to understand why you have so
>>>> many problems.
>>>>
>>>> Or, alternatively, try to follow the wiki. You know, I've not heard
>>>> about those problems.
>>>>
>>>> root at hardy64:~/skype.beta/skype_static-2.1.0.81# ldd skype
>>>> ? ? ? ?linux-gate.so.1 => ?(0xffffe000)
>>>> ? ? ? ?libasound.so.2 => /usr/lib32/libasound.so.2 (0xf7eb7000)
>>>> ? ? ? ?libXv.so.1 => /usr/lib32/libXv.so.1 (0xf7eb2000)
>>>> ? ? ? ?libXss.so.1 => /usr/lib32/libXss.so.1 (0xf7eae000)
>>>> ? ? ? ?libSM.so.6 => /usr/lib32/libSM.so.6 (0xf7ea6000)
>>>> ? ? ? ?libICE.so.6 => /usr/lib32/libICE.so.6 (0xf7e8e000)
>>>> ? ? ? ?libXi.so.6 => /usr/lib32/libXi.so.6 (0xf7e86000)
>>>> ? ? ? ?libXrender.so.1 => /usr/lib32/libXrender.so.1 (0xf7e7e000)
>>>> ? ? ? ?libXrandr.so.2 => /usr/lib32/libXrandr.so.2 (0xf7e78000)
>>>> ? ? ? ?libfreetype.so.6 => /usr/lib32/libfreetype.so.6 (0xf7e0a000)
>>>> ? ? ? ?libfontconfig.so.1 => /usr/lib32/libfontconfig.so.1 (0xf7de0000)
>>>> ? ? ? ?libXext.so.6 => /usr/lib32/libXext.so.6 (0xf7dd2000)
>>>> ? ? ? ?libX11.so.6 => /usr/lib32/libX11.so.6 (0xf7ceb000)
>>>> ? ? ? ?libz.so.1 => /usr/lib32/libz.so.1 (0xf7cd6000)
>>>> ? ? ? ?libgthread-2.0.so.0 => /usr/lib32/libgthread-2.0.so.0 (0xf7cd1000)
>>>> ? ? ? ?libglib-2.0.so.0 => /usr/lib32/libglib-2.0.so.0 (0xf7c1f000)
>>>> ? ? ? ?librt.so.1 => /lib32/librt.so.1 (0xf7c16000)
>>>> ? ? ? ?libdl.so.2 => /lib32/libdl.so.2 (0xf7c12000)
>>>> ? ? ? ?libpthread.so.0 => /lib32/libpthread.so.0 (0xf7bfa000)
>>>> ? ? ? ?libstdc++.so.6 => /usr/lib32/libstdc++.so.6 (0xf7b07000)
>>>> ? ? ? ?libm.so.6 => /lib32/libm.so.6 (0xf7ae2000)
>>>> ? ? ? ?libgcc_s.so.1 => /usr/lib32/libgcc_s.so.1 (0xf7ad6000)
>>>> ? ? ? ?libc.so.6 => /lib32/libc.so.6 (0xf7987000)
>>>> ? ? ? ?libexpat.so.1 => /usr/lib32/libexpat.so.1 (0xf7966000)
>>>> ? ? ? ?libXau.so.6 => /usr/lib32/libXau.so.6 (0xf7963000)
>>>> ? ? ? ?libxcb-xlib.so.0 => /usr/lib32/libxcb-xlib.so.0 (0xf7961000)
>>>> ? ? ? ?libxcb.so.1 => /usr/lib32/libxcb.so.1 (0xf7948000)
>>>> ? ? ? ?libpcre.so.3 => /usr/lib32/libpcre.so.3 (0xf7921000)
>>>> ? ? ? ?/lib/ld-linux.so.2 (0xf7f86000)
>>>> ? ? ? ?libXdmcp.so.6 => /usr/lib32/libXdmcp.so.6 (0xf791c000)
>>>>
>>>>
>>>> On Sun, Feb 7, 2010 at 11:28 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>> I now used the static Skype binary in order to avoid missing constraints
>>>>> to other libraries: It still crashes
>>>>> 1st it starts with:
>>>>> ?process 15431: D-Bus library appears to be incorrectly set up; failed
>>>>> to read machine uuid: Failed to open "/var/lib/dbus/machine-id": No such
>>>>> file or directory
>>>>> ?See the manual page for dbus-uuidgen to correct this issue.
>>>>> After calling this client it crashes with:
>>>>> ? /usr/bin/skype: symbol lookup error: /usr/bin/skype: undefined
>>>>> symbol: _ZN10QBoxLayout13addSpacerItemEP11QSpacerItem
>>>>>
>>>>> Any hints, where I may get an older Skype client?
>>>>>
>>>>> Best regards
>>>>> Peter
>>>>>
>>>>> Anthony Minessale schrieb:
>>>>>>
>>>>>> Quick fyi centos 5.4 has some memory bugs in libc don't go beyond 5.3
>>>>>> until its fixed.
>>>>>>
>>>>>>> On Feb 5, 2010 7:42 PM, "Peter P GMX" <Prometheus001 at gmx.net
>>>>>>> <mailto:Prometheus001 at gmx.net>> wrote:
>>>>>>>
>>>>>>> Skype starts, but as soon as it receives a call it crashes with:
>>>>>>>
>>>>>>> /usr/bin/skype: symbol lookup error: /usr/bin/skype: undefined symbol:
>>>>>>>
>>>>>>> _ZN10QBoxLayout13addSpacerItemEP11QSpacerItem
>>>>>>>
>>>>>>> I think the 8.10 version dos not work with8.04.
>>>>>>>
>>>>>>> Any hints, where I may get an older Skype client? I may also try the
>>>>>>> static skype client.
>>>>>>>
>>>>>>>
>>>>>>> Best regards
>>>>>>> Peter
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> .
>>>>>>> Giovanni Maruzzelli schrieb:
>>>>>>> > that's not at all a fatal error.
>>>>>>> > I believe...
>>>>>>>
>>>>>> ------------------------------------------------------------------------
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> Sincerely,
>>>>
>>>> Giovanni Maruzzelli
>>>> Cell : +39-347-2665618
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> Cell : +39-347-2665618
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From gmaruzz at celliax.org  Mon Feb  8 08:47:03 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Mon, 8 Feb 2010 17:47:03 +0100
Subject: [Freeswitch-users] CentOS 5.3 (Final) with Skype Static Build
In-Reply-To: <c55d46951002080706t6bb8531i4f12cb35af12e6de@mail.gmail.com>
References: <c55d46951002080706t6bb8531i4f12cb35af12e6de@mail.gmail.com>
Message-ID: <7b197bef1002080847v2524f99dt638c1668ac71425a@mail.gmail.com>

Max,

Just checked, the static build of skype-beta works on latest centos (5.4)

But you can have the same results if just tried :).

ldd tells you which libraries are missing, the you install those libraries.
If you dont know which package a library is into, just do, eg:

yum search libXss

and you'll be told.

========================
So, all in all:

you have to install:

yum install libXv
yum install libXScrnSaver

then use the original snd-dummy supplied with centos (the modified one do not
works at the moment - because the skype-beta does some alsa calls it was
not doing in previous versions, that are not implemented in the
modified alsa driver, I'll fix it soon).

===============================================

-gm



On Mon, Feb 8, 2010 at 4:06 PM, Max Bridgewater
<max.bridgewater at gmail.com> wrote:
> Hi Giovanni,
>
> Let me start a new thread so I don't hijack Peter's one. Here is what
> I get for ldd /usr/bin/skype. What is the deal with these libXv.so.1,
> libXss.so.1?
>
> ? ? ? ?linux-gate.so.1 => ?(0x00697000)
> ? ? ? ?libasound.so.2 => /lib/libasound.so.2 (0x008b7000)
> ? ? ? ?libXv.so.1 => not found
> ? ? ? ?libXss.so.1 => not found
> ? ? ? ?libSM.so.6 => /usr/lib/libSM.so.6 (0x00d02000)
> ? ? ? ?libICE.so.6 => /usr/lib/libICE.so.6 (0x0036b000)
> ? ? ? ?libXi.so.6 => /usr/lib/libXi.so.6 (0x00111000)
> ? ? ? ?libXrender.so.1 => /usr/lib/libXrender.so.1 (0x00119000)
> ? ? ? ?libXrandr.so.2 => /usr/lib/libXrandr.so.2 (0x00b77000)
> ? ? ? ?libfreetype.so.6 => /usr/lib/libfreetype.so.6 (0x00122000)
> ? ? ? ?libfontconfig.so.1 => /usr/lib/libfontconfig.so.1 (0x009fb000)
> ? ? ? ?libXext.so.6 => /usr/lib/libXext.so.6 (0x00eb0000)
> ? ? ? ?libX11.so.6 => /usr/lib/libX11.so.6 (0x001a2000)
> ? ? ? ?libz.so.1 => /usr/lib/libz.so.1 (0x002a5000)
> ? ? ? ?libgthread-2.0.so.0 => /lib/libgthread-2.0.so.0 (0x006bd000)
> ? ? ? ?libglib-2.0.so.0 => /lib/libglib-2.0.so.0 (0x002be000)
> ? ? ? ?librt.so.1 => /lib/librt.so.1 (0x009a1000)
> ? ? ? ?libdl.so.2 => /lib/libdl.so.2 (0x00604000)
> ? ? ? ?libpthread.so.0 => /lib/libpthread.so.0 (0x004e6000)
> ? ? ? ?libstdc++.so.6 => /usr/lib/libstdc++.so.6 (0x004fd000)
> ? ? ? ?libm.so.6 => /lib/libm.so.6 (0x00385000)
> ? ? ? ?libgcc_s.so.1 => /lib/libgcc_s.so.1 (0x00bcc000)
> ? ? ? ?libc.so.6 => /lib/libc.so.6 (0x006c2000)
> ? ? ? ?libexpat.so.0 => /lib/libexpat.so.0 (0x00c2a000)
> ? ? ? ?libXau.so.6 => /usr/lib/libXau.so.6 (0x002ba000)
> ? ? ? ?libXdmcp.so.6 => /usr/lib/libXdmcp.so.6 (0x0087e000)
> ? ? ? ?/lib/ld-linux.so.2 (0x00476000)
>
> Max
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From max.bridgewater at gmail.com  Mon Feb  8 09:12:31 2010
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Mon, 8 Feb 2010 12:12:31 -0500
Subject: [Freeswitch-users] CentOS 5.3 (Final) with Skype Static Build
In-Reply-To: <7b197bef1002080847v2524f99dt638c1668ac71425a@mail.gmail.com>
References: <c55d46951002080706t6bb8531i4f12cb35af12e6de@mail.gmail.com>
	<7b197bef1002080847v2524f99dt638c1668ac71425a@mail.gmail.com>
Message-ID: <c55d46951002080912l4123a589qd5546462b7a6b30@mail.gmail.com>

Great thanks, Giovanni. Just completed the installation. No sound test
yet though.

Max.

On Mon, Feb 8, 2010 at 11:47 AM, Giovanni Maruzzelli
<gmaruzz at celliax.org> wrote:
> Max,
>
> Just checked, the static build of skype-beta works on latest centos (5.4)
>
> But you can have the same results if just tried :).
>
> ldd tells you which libraries are missing, the you install those libraries.
> If you dont know which package a library is into, just do, eg:
>
> yum search libXss
>
> and you'll be told.
>
> ========================
> So, all in all:
>
> you have to install:
>
> yum install libXv
> yum install libXScrnSaver
>
> then use the original snd-dummy supplied with centos (the modified one do not
> works at the moment - because the skype-beta does some alsa calls it was
> not doing in previous versions, that are not implemented in the
> modified alsa driver, I'll fix it soon).
>
> ===============================================
>
> -gm
>
>
>
> On Mon, Feb 8, 2010 at 4:06 PM, Max Bridgewater
> <max.bridgewater at gmail.com> wrote:
>> Hi Giovanni,
>>
>> Let me start a new thread so I don't hijack Peter's one. Here is what
>> I get for ldd /usr/bin/skype. What is the deal with these libXv.so.1,
>> libXss.so.1?
>>
>> ? ? ? ?linux-gate.so.1 => ?(0x00697000)
>> ? ? ? ?libasound.so.2 => /lib/libasound.so.2 (0x008b7000)
>> ? ? ? ?libXv.so.1 => not found
>> ? ? ? ?libXss.so.1 => not found
>> ? ? ? ?libSM.so.6 => /usr/lib/libSM.so.6 (0x00d02000)
>> ? ? ? ?libICE.so.6 => /usr/lib/libICE.so.6 (0x0036b000)
>> ? ? ? ?libXi.so.6 => /usr/lib/libXi.so.6 (0x00111000)
>> ? ? ? ?libXrender.so.1 => /usr/lib/libXrender.so.1 (0x00119000)
>> ? ? ? ?libXrandr.so.2 => /usr/lib/libXrandr.so.2 (0x00b77000)
>> ? ? ? ?libfreetype.so.6 => /usr/lib/libfreetype.so.6 (0x00122000)
>> ? ? ? ?libfontconfig.so.1 => /usr/lib/libfontconfig.so.1 (0x009fb000)
>> ? ? ? ?libXext.so.6 => /usr/lib/libXext.so.6 (0x00eb0000)
>> ? ? ? ?libX11.so.6 => /usr/lib/libX11.so.6 (0x001a2000)
>> ? ? ? ?libz.so.1 => /usr/lib/libz.so.1 (0x002a5000)
>> ? ? ? ?libgthread-2.0.so.0 => /lib/libgthread-2.0.so.0 (0x006bd000)
>> ? ? ? ?libglib-2.0.so.0 => /lib/libglib-2.0.so.0 (0x002be000)
>> ? ? ? ?librt.so.1 => /lib/librt.so.1 (0x009a1000)
>> ? ? ? ?libdl.so.2 => /lib/libdl.so.2 (0x00604000)
>> ? ? ? ?libpthread.so.0 => /lib/libpthread.so.0 (0x004e6000)
>> ? ? ? ?libstdc++.so.6 => /usr/lib/libstdc++.so.6 (0x004fd000)
>> ? ? ? ?libm.so.6 => /lib/libm.so.6 (0x00385000)
>> ? ? ? ?libgcc_s.so.1 => /lib/libgcc_s.so.1 (0x00bcc000)
>> ? ? ? ?libc.so.6 => /lib/libc.so.6 (0x006c2000)
>> ? ? ? ?libexpat.so.0 => /lib/libexpat.so.0 (0x00c2a000)
>> ? ? ? ?libXau.so.6 => /usr/lib/libXau.so.6 (0x002ba000)
>> ? ? ? ?libXdmcp.so.6 => /usr/lib/libXdmcp.so.6 (0x0087e000)
>> ? ? ? ?/lib/ld-linux.so.2 (0x00476000)
>>
>> Max
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From christian at officepools.com  Mon Feb  8 10:39:46 2010
From: christian at officepools.com (Christian Jensen)
Date: Mon, 8 Feb 2010 10:39:46 -0800
Subject: [Freeswitch-users] FS based Softphone?
Message-ID: <e90351e71002081039y7c7723a6x7181d9fdd208bdde@mail.gmail.com>

Hi,

I am setting up our office with phones and softphones. I heard that there is
a softphone that was based on FS - is this ready for primetime? If not, what
is the best softphone for use with a mostly windows but some ubuntu and mac
environment?

Thanks!
Christian
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100208/c938948c/attachment.html 

From jmesquita at freeswitch.org  Mon Feb  8 11:09:13 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Mon, 8 Feb 2010 17:09:13 -0200
Subject: [Freeswitch-users] FS based Softphone?
In-Reply-To: <e90351e71002081039y7c7723a6x7181d9fdd208bdde@mail.gmail.com>
References: <e90351e71002081039y7c7723a6x7181d9fdd208bdde@mail.gmail.com>
Message-ID: <ebbc2d551002081109p2e451894y45aa9bfae3c548bb@mail.gmail.com>

Christian,

The project you are mentioning is called FSComm and can be found on the SVN
source. The project is still under intense development with very limited
resources and time. We are constantly looking for sponsors to make this
dream possible, if you are a potential one, please send me and email
offlist.

As for the project maturity, it is up to you to decide since only you know
the features needed. We have users that use it on a daily basis and we have
a couple of bugs filed that I still haven't had time to fix.

We don't have a mac installer but the wiki makes it pretty clear on how to
compile it: http://wiki.freeswitch.org/wiki/FSComm

We welcome any type of criticism or request.

Regards,

Jo?o Mesquita
FSComm Developer


On Mon, Feb 8, 2010 at 4:39 PM, Christian Jensen
<christian at officepools.com>wrote:

> Hi,
>
> I am setting up our office with phones and softphones. I heard that there
> is a softphone that was based on FS - is this ready for primetime? If not,
> what is the best softphone for use with a mostly windows but some ubuntu and
> mac environment?
>
> Thanks!
> Christian
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100208/edb3d29d/attachment.html 

From mrene_lists at avgs.ca  Mon Feb  8 11:32:25 2010
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Mon, 8 Feb 2010 14:32:25 -0500
Subject: [Freeswitch-users] mod_limit requires media?
In-Reply-To: <2d9149cd1002072106n7272cb80se3329700f0074d52@mail.gmail.com>
References: <2d9149cd1002072106n7272cb80se3329700f0074d52@mail.gmail.com>
Message-ID: <AE10DF9B-B6AC-41D1-AC28-EFEF5946C44D@avgs.ca>

Hi,

mod_limit doesn't require any media. ca you post some logs of your  
problem?

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 8-Feb-10, at 12:06 AM, Kristian Kielhofner wrote:

> Hello everyone,
>
>  I was playing with mod_limit earlier tonight and I noticed that it
> essentially stopped hashing/tracking calls once bypass media was set.
> Is this by design?  Is there some reason mod_limit requires media?
> Other than that mod_limit looks to be very well implemented (no
> surprise there) and I'm excited to put it to more use.
>
> Thanks!
>
> -- 
> Kristian Kielhofner
> http://www.astlinux.org
> http://blog.krisk.org
> http://www.star2star.com
> http://www.submityoursip.com
> http://www.voalte.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From kristian.kielhofner at gmail.com  Mon Feb  8 12:10:56 2010
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Mon, 8 Feb 2010 15:10:56 -0500
Subject: [Freeswitch-users] mod_limit requires media?
In-Reply-To: <AE10DF9B-B6AC-41D1-AC28-EFEF5946C44D@avgs.ca>
References: <2d9149cd1002072106n7272cb80se3329700f0074d52@mail.gmail.com>
	<AE10DF9B-B6AC-41D1-AC28-EFEF5946C44D@avgs.ca>
Message-ID: <2d9149cd1002081210u3662ffa6h3eb3d1c479512e80@mail.gmail.com>

Hi,

  I just sent this to Tony off-list:

It doesn't seem like it should be but testing it shows there is some issue...

no bypass_media (working):
http://pastebin.freeswitch.org/12085

with bypass_media (not working):
http://pastebin.freeswitch.org/12086

This is SVN rev 16584 on my Mac running Snow Leopard.  I'm trying to
call my AstLinux conf from a local softphone, bridging through FS.
Completely default configs except I added the limit config shown in
the PBs in conf/dialplan/default/00_astlinux-conf.xml

You can see that with bypass media set when the call is up
limit_hash_usage outbound carrier1 shows 0.  Without bypass media set
it shows 1.  However in both cases the FS console reports that usage
is 1/1 at some point...

On Mon, Feb 8, 2010 at 2:32 PM, Mathieu Rene <mrene_lists at avgs.ca> wrote:
> Hi,
>
> mod_limit doesn't require any media. ca you post some logs of your
> problem?
>
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>
>
>
>
> On 8-Feb-10, at 12:06 AM, Kristian Kielhofner wrote:
>
>> Hello everyone,
>>
>> ?I was playing with mod_limit earlier tonight and I noticed that it
>> essentially stopped hashing/tracking calls once bypass media was set.
>> Is this by design? ?Is there some reason mod_limit requires media?
>> Other than that mod_limit looks to be very well implemented (no
>> surprise there) and I'm excited to put it to more use.
>>
>> Thanks!
>>
>> --
>> Kristian Kielhofner
>> http://www.astlinux.org
>> http://blog.krisk.org
>> http://www.star2star.com
>> http://www.submityoursip.com
>> http://www.voalte.com
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com


From gavin.henry at gmail.com  Mon Feb  8 12:40:57 2010
From: gavin.henry at gmail.com (Gavin Henry)
Date: Mon, 8 Feb 2010 20:40:57 +0000
Subject: [Freeswitch-users] mod_nibblebill, loading A-Z rates and CDRs
Message-ID: <13ca621c1002081240h223942e7vb4f9e1034822ee28@mail.gmail.com>

Hi all,

We're testing FS and mod_nibblebill for a wholesale platform. So far
it's working ok via ODBC. I'd like to know how you'd recommend loading
A-Z rates that have peak, offpeak and weekend rates.

I was thinking about using the conditions for destination to match the
time of day:

http://wiki.freeswitch.org/wiki/Dialplan_XML#Condition

so each rate would be listed a few times in order to match the time of day.

For the destinations I was going to do
(http://wiki.freeswitch.org/wiki/Mod_nibblebill#Different_Rates_per_Area_Code):

 <extension name="uk_landline">
   <condition field="destination_number" expression="^0([1-3]\d{9})$">
    <action application="set" data="nibble_account=${accountcode}"/>
    <action application="set" data="nibble_rate=rate_tbc"/>
    <action application="bridge"
data="sofia/gateway/sip.myprovider.co.uk/44$1"/>
   </condition>
 </extension>


I presume I can have (newbie here) multiple conditions for the time of day?

My last question is if there is a better way to load the A-Z rates
rather than a massive XML file, like from a DB. I'm still new to FS so
not sure how to pull in this data. I presume I could just pull it in
via mod_xml_curl and do the same with saving the CDRs to a RDBMS using
mod_xml_cdr?

Is this a best practice going to the RDBMS (I'm part of the OpenLDAP
project so always call it RDBMS so as not confuse a user thinking
about the OpenLDAP database backend) via mod_xml_curl? I would thought
going directly to the RDBMS is better? I suppose it depends on the
system architecture.

Thanks,

Gavin.


-- 
http://www.suretecsystems.com/services/openldap/
http://www.suretectelecom.com


From rupa at rupa.com  Mon Feb  8 13:10:49 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Mon, 8 Feb 2010 15:10:49 -0600
Subject: [Freeswitch-users] mod_nibblebill, loading A-Z rates and CDRs
In-Reply-To: <13ca621c1002081240h223942e7vb4f9e1034822ee28@mail.gmail.com>
References: <13ca621c1002081240h223942e7vb4f9e1034822ee28@mail.gmail.com>
Message-ID: <b39273531002081310m7d37d9f0gf700b1ecd9ee56dd@mail.gmail.com>

Look at using mod_lcr for doing your A-Z rates.  You can categorize your
peak/offpeak/wkend rates as different profiles and then lookup based on
those profiles. mod_lcr is very flexible in that you can also specify your
own custom sql so if you are familiar with sql you could even put your
peak/offpeak/wkend etc decisions in the SQL or in a stored procedure in the
db.

If you are going down the path of custom_sql, then use the patch in jira
since custom_sql behavior will change post 1.0.5 and the new behavior is
much more flexible.

On Mon, Feb 8, 2010 at 2:40 PM, Gavin Henry <gavin.henry at gmail.com> wrote:

> Hi all,
>
> We're testing FS and mod_nibblebill for a wholesale platform. So far
> it's working ok via ODBC. I'd like to know how you'd recommend loading
> A-Z rates that have peak, offpeak and weekend rates.
>
> I was thinking about using the conditions for destination to match the
> time of day:
>
> http://wiki.freeswitch.org/wiki/Dialplan_XML#Condition
>
> so each rate would be listed a few times in order to match the time of day.
>
> For the destinations I was going to do
> (
> http://wiki.freeswitch.org/wiki/Mod_nibblebill#Different_Rates_per_Area_Code
> ):
>
>  <extension name="uk_landline">
>   <condition field="destination_number" expression="^0([1-3]\d{9})$">
>    <action application="set" data="nibble_account=${accountcode}"/>
>    <action application="set" data="nibble_rate=rate_tbc"/>
>    <action application="bridge"
> data="sofia/gateway/sip.myprovider.co.uk/44$1"/>
>   </condition>
>  </extension>
>
>
> I presume I can have (newbie here) multiple conditions for the time of day?
>
> My last question is if there is a better way to load the A-Z rates
> rather than a massive XML file, like from a DB. I'm still new to FS so
> not sure how to pull in this data. I presume I could just pull it in
> via mod_xml_curl and do the same with saving the CDRs to a RDBMS using
> mod_xml_cdr?
>
> Is this a best practice going to the RDBMS (I'm part of the OpenLDAP
> project so always call it RDBMS so as not confuse a user thinking
> about the OpenLDAP database backend) via mod_xml_curl? I would thought
> going directly to the RDBMS is better? I suppose it depends on the
> system architecture.
>
> Thanks,
>
> Gavin.
>
>
> --
> http://www.suretecsystems.com/services/openldap/
> http://www.suretectelecom.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
-Rupa
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100208/3d2778ec/attachment.html 

From gavin.henry at gmail.com  Mon Feb  8 13:20:44 2010
From: gavin.henry at gmail.com (Gavin Henry)
Date: Mon, 8 Feb 2010 21:20:44 +0000
Subject: [Freeswitch-users] mod_nibblebill, loading A-Z rates and CDRs
In-Reply-To: <b39273531002081310m7d37d9f0gf700b1ecd9ee56dd@mail.gmail.com>
References: <13ca621c1002081240h223942e7vb4f9e1034822ee28@mail.gmail.com>
	<b39273531002081310m7d37d9f0gf700b1ecd9ee56dd@mail.gmail.com>
Message-ID: <13ca621c1002081320m3b9a7779l6ee6daa6382ffe56@mail.gmail.com>

On 8 February 2010 21:10, Rupa Schomaker <rupa at rupa.com> wrote:
> Look at using mod_lcr for doing your A-Z rates. ?You can categorize your
> peak/offpeak/wkend rates as different profiles and then lookup based on
> those profiles. mod_lcr is very flexible in that you can also specify your
> own custom sql so if you are familiar with sql you could even put your
> peak/offpeak/wkend etc decisions in the SQL or in a stored procedure in the
> db.
> If you are going down the path of custom_sql, then use the patch in jira
> since custom_sql behavior will change post 1.0.5 and the new behavior is
> much more flexible.

OK, thanks I will do. What about the real time billing features of
nibblebill? Can you use them together?

Will do read the docs now for mod_lcr.

> On Mon, Feb 8, 2010 at 2:40 PM, Gavin Henry <gavin.henry at gmail.com> wrote:
>>
>> Hi all,
>>
>> We're testing FS and mod_nibblebill for a wholesale platform. So far
>> it's working ok via ODBC. I'd like to know how you'd recommend loading
>> A-Z rates that have peak, offpeak and weekend rates.
>>
>> I was thinking about using the conditions for destination to match the
>> time of day:
>>
>> http://wiki.freeswitch.org/wiki/Dialplan_XML#Condition
>>
>> so each rate would be listed a few times in order to match the time of
>> day.
>>
>> For the destinations I was going to do
>>
>> (http://wiki.freeswitch.org/wiki/Mod_nibblebill#Different_Rates_per_Area_Code):
>>
>> ?<extension name="uk_landline">
>> ? <condition field="destination_number" expression="^0([1-3]\d{9})$">
>> ? ?<action application="set" data="nibble_account=${accountcode}"/>
>> ? ?<action application="set" data="nibble_rate=rate_tbc"/>
>> ? ?<action application="bridge"
>> data="sofia/gateway/sip.myprovider.co.uk/44$1"/>
>> ? </condition>
>> ?</extension>
>>
>>
>> I presume I can have (newbie here) multiple conditions for the time of
>> day?
>>
>> My last question is if there is a better way to load the A-Z rates
>> rather than a massive XML file, like from a DB. I'm still new to FS so
>> not sure how to pull in this data. I presume I could just pull it in
>> via mod_xml_curl and do the same with saving the CDRs to a RDBMS using
>> mod_xml_cdr?
>>
>> Is this a best practice going to the RDBMS (I'm part of the OpenLDAP
>> project so always call it RDBMS so as not confuse a user thinking
>> about the OpenLDAP database backend) via mod_xml_curl? I would thought
>> going directly to the RDBMS is better? I suppose it depends on the
>> system architecture.
>>
>> Thanks,
>>
>> Gavin.
>>
>>
>> --
>> http://www.suretecsystems.com/services/openldap/
>> http://www.suretectelecom.com
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
http://www.suretecsystems.com/services/openldap/
http://www.suretectelecom.com


From gavin.henry at gmail.com  Mon Feb  8 13:39:38 2010
From: gavin.henry at gmail.com (Gavin Henry)
Date: Mon, 8 Feb 2010 21:39:38 +0000
Subject: [Freeswitch-users] mod_nibblebill, loading A-Z rates and CDRs
In-Reply-To: <13ca621c1002081320m3b9a7779l6ee6daa6382ffe56@mail.gmail.com>
References: <13ca621c1002081240h223942e7vb4f9e1034822ee28@mail.gmail.com>
	<b39273531002081310m7d37d9f0gf700b1ecd9ee56dd@mail.gmail.com>
	<13ca621c1002081320m3b9a7779l6ee6daa6382ffe56@mail.gmail.com>
Message-ID: <13ca621c1002081339y574aabdg1408052b2bb28bf@mail.gmail.com>

On 8 February 2010 21:20, Gavin Henry <gavin.henry at gmail.com> wrote:
> On 8 February 2010 21:10, Rupa Schomaker <rupa at rupa.com> wrote:
>> Look at using mod_lcr for doing your A-Z rates. ?You can categorize your
>> peak/offpeak/wkend rates as different profiles and then lookup based on
>> those profiles. mod_lcr is very flexible in that you can also specify your
>> own custom sql so if you are familiar with sql you could even put your
>> peak/offpeak/wkend etc decisions in the SQL or in a stored procedure in the
>> db.
>> If you are going down the path of custom_sql, then use the patch in jira
>> since custom_sql behavior will change post 1.0.5 and the new behavior is
>> much more flexible.
>
> OK, thanks I will do. What about the real time billing features of
> nibblebill? Can you use them together?

For others, there is some info at
http://wiki.freeswitch.org/wiki/Mod_lcr#User_Rates that mentions
nibblebill.

Thanks.

-- 
http://www.suretecsystems.com/services/openldap/
http://www.suretectelecom.com


From mrene_lists at avgs.ca  Mon Feb  8 14:27:08 2010
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Mon, 8 Feb 2010 17:27:08 -0500
Subject: [Freeswitch-users] mod_limit requires media?
In-Reply-To: <2d9149cd1002081210u3662ffa6h3eb3d1c479512e80@mail.gmail.com>
References: <2d9149cd1002072106n7272cb80se3329700f0074d52@mail.gmail.com>
	<AE10DF9B-B6AC-41D1-AC28-EFEF5946C44D@avgs.ca>
	<2d9149cd1002081210u3662ffa6h3eb3d1c479512e80@mail.gmail.com>
Message-ID: <A2EB2618-601A-487C-A754-7B258250D6FC@avgs.ca>

Try r16586

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 8-Feb-10, at 3:10 PM, Kristian Kielhofner wrote:

> Hi,
>
>  I just sent this to Tony off-list:
>
> It doesn't seem like it should be but testing it shows there is some  
> issue...
>
> no bypass_media (working):
> http://pastebin.freeswitch.org/12085
>
> with bypass_media (not working):
> http://pastebin.freeswitch.org/12086
>
> This is SVN rev 16584 on my Mac running Snow Leopard.  I'm trying to
> call my AstLinux conf from a local softphone, bridging through FS.
> Completely default configs except I added the limit config shown in
> the PBs in conf/dialplan/default/00_astlinux-conf.xml
>
> You can see that with bypass media set when the call is up
> limit_hash_usage outbound carrier1 shows 0.  Without bypass media set
> it shows 1.  However in both cases the FS console reports that usage
> is 1/1 at some point...
>
> On Mon, Feb 8, 2010 at 2:32 PM, Mathieu Rene <mrene_lists at avgs.ca>  
> wrote:
>> Hi,
>>
>> mod_limit doesn't require any media. ca you post some logs of your
>> problem?
>>
>> Mathieu Rene
>> Avant-Garde Solutions Inc
>> Office: + 1 (514) 664-1044 x100
>> Cell: +1 (514) 664-1044 x200
>> mrene at avgs.ca
>>
>>
>>
>>
>> On 8-Feb-10, at 12:06 AM, Kristian Kielhofner wrote:
>>
>>> Hello everyone,
>>>
>>>  I was playing with mod_limit earlier tonight and I noticed that it
>>> essentially stopped hashing/tracking calls once bypass media was  
>>> set.
>>> Is this by design?  Is there some reason mod_limit requires media?
>>> Other than that mod_limit looks to be very well implemented (no
>>> surprise there) and I'm excited to put it to more use.
>>>
>>> Thanks!
>>>
>>> --
>>> Kristian Kielhofner
>>> http://www.astlinux.org
>>> http://blog.krisk.org
>>> http://www.star2star.com
>>> http://www.submityoursip.com
>>> http://www.voalte.com
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> -- 
> Kristian Kielhofner
> http://www.astlinux.org
> http://blog.krisk.org
> http://www.star2star.com
> http://www.submityoursip.com
> http://www.voalte.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From kristian.kielhofner at gmail.com  Mon Feb  8 14:55:56 2010
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Mon, 8 Feb 2010 17:55:56 -0500
Subject: [Freeswitch-users] mod_limit requires media?
In-Reply-To: <A2EB2618-601A-487C-A754-7B258250D6FC@avgs.ca>
References: <2d9149cd1002072106n7272cb80se3329700f0074d52@mail.gmail.com>
	<AE10DF9B-B6AC-41D1-AC28-EFEF5946C44D@avgs.ca>
	<2d9149cd1002081210u3662ffa6h3eb3d1c479512e80@mail.gmail.com>
	<A2EB2618-601A-487C-A754-7B258250D6FC@avgs.ca>
Message-ID: <2d9149cd1002081455q6a84e61fpac46be8082efc3d3@mail.gmail.com>

I'm going to perform further testing but it looks good so far.

Thanks a lot!

On Mon, Feb 8, 2010 at 5:27 PM, Mathieu Rene <mrene_lists at avgs.ca> wrote:
> Try r16586
>
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>

-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com


From lon at kickasspixels.com  Mon Feb  8 17:05:04 2010
From: lon at kickasspixels.com (Lon Baker)
Date: Mon, 8 Feb 2010 17:05:04 -0800
Subject: [Freeswitch-users] Redirect during Bridge
Message-ID: <5d3e0dc61002081705n24992492yd28194b9a5791bac@mail.gmail.com>

Hi,

I need to bridge calls to a proxy that issues a redirect command. Its
an internal proxy that translates internal SIP URIs into outbound
URIs.

I have one interface on a private network where the proxy is running
and another interface on my public network.

If I am reading the debug information correctly, the bridge to the
proxy is getting the redirect correctly, but since it is on the
internal network (profile), the call never transitions to the public
network to complete the call.

Is it possible to do this?

Lon


From r.wilczynski at gmail.com  Mon Feb  8 11:37:40 2010
From: r.wilczynski at gmail.com (=?ISO-8859-2?Q?Robert_Wilczy=F1ski?=)
Date: Mon, 8 Feb 2010 20:37:40 +0100
Subject: [Freeswitch-users] Trunk Version Number
In-Reply-To: <8976E2C4-CF8C-433A-AC79-B58F65105D08@jerris.com>
References: <2360B060EDF44D368707DD12064BD416@bp1.ad.bp.com>
	<8976E2C4-CF8C-433A-AC79-B58F65105D08@jerris.com>
Message-ID: <56bfa1be1002081137q3f2aad44wc2742aa3505bf961@mail.gmail.com>

Hi Michael,

You mean a set of windows binaries matching those in w32\Library (this
seems to work for me)? Where should I email it?

Robert.

On Mon, Feb 1, 2010 at 7:11 AM, Michael Jerris <mike at jerris.com> wrote:
> it should. ?This can happen if you build from an svn checkout and the svn
> client your using is newer than our static linked svnversion.exe. ?If anyone
> can make me a newer stripped down version like that I would appreciate it I
> have not had the time.
> On Jan 31, 2010, at 9:30 AM, Dave Stevenson wrote:
>
> Hi,
>
> Running the latest SVN (16453)?under Windows, the console "Version" command
> displays :-
>
> "FreeSWITCH Version 1.0.trunk (UNKNOWN)"
>
> Should the version number not include a meaningful build version?in the
> brackets ?
>
> regards
> Dave
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From troy at tlainvestments.com  Mon Feb  8 20:21:01 2010
From: troy at tlainvestments.com (Troy Anderson)
Date: Mon, 8 Feb 2010 21:21:01 -0700
Subject: [Freeswitch-users] UPnP Timeout
Message-ID: <9706F7A8-7644-4602-B1DE-289A35F788AA@tlainvestments.com>

I have been using pfSense (1.2.3) and FS.  FS nicely uses UPnP to poke holes in the firewall, but it seems that the holes close after a while.  I cannot find any documentation in FS nor in pfSense as to what the timeout is.  Is there a setting in FS to do some kind of keep-alive thing with UPnP to keep, e.g. 5060, open?  Or is it already doing that and pfSense is the issue?

Thanks!

From yehavi.bourvine at gmail.com  Mon Feb  8 21:10:04 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Tue, 9 Feb 2010 07:10:04 +0200
Subject: [Freeswitch-users] Announcement: FreeSWITCH Add Broadsoft SCA
	Support
In-Reply-To: <87f2f3b91001111742k691966cfs82515db77c13d4d4@mail.gmail.com>
References: <87f2f3b91001111742k691966cfs82515db77c13d4d4@mail.gmail.com>
Message-ID: <d361ca7b1002082110h4e3c60d0p88412f753444ec3d@mail.gmail.com>

Hello,

  We currently use the "old" type of presence which is activated by
"manage-presence" coupled with "dbname" and "presence-hosts".
With the new method, does "manage-shared-presence" replace all of the above
or comes in addition?

                       Thanks! __Yehavi:

 2010/1/12 Michael Collins <msc at freeswitch.org>

> We want to let everyone know that FreeSWITCH now supports the Broadsoft SCA
> method of doing shared lines. The story is here:
>
> http://www.freeswitch.org/node/227
>
> Tony and Brian spent many hours laboring over this, so please be sure to
> show your appreciation to them for this new feature and all of the great
> things they do for the FreeSWITCH community and VoIP in general!
>
> -Michael
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100209/fc020b4c/attachment.html 

From mrene_lists at avgs.ca  Mon Feb  8 21:36:55 2010
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Tue, 9 Feb 2010 00:36:55 -0500
Subject: [Freeswitch-users] Redirect during Bridge
In-Reply-To: <5d3e0dc61002081705n24992492yd28194b9a5791bac@mail.gmail.com>
References: <5d3e0dc61002081705n24992492yd28194b9a5791bac@mail.gmail.com>
Message-ID: <ABEBD044-4E95-4916-A125-2556F2E47ED6@avgs.ca>

Hi,

mod_sofia's default behavior is to process 302s within the sip stack,  
automatically, in some cases, like yours, this behavior isnt desired.
You can enable manual redirects which will make the call go back in  
the dialplan whenever an invite hits a 302.
Add the following to your sip profile:
<param name="manual-redirect" value="true" />

You can then, in your dialplan, set the sip_redirect_profile variable  
to your internal network's sip profile. When a call is redirected,   
the sip_redirect_dialstring variable will contain the dialstring you  
need to pass to bridge.

your outbound dialplan should look like this
<extension name="something">
  <condition field="destination_number" expression="....">
    <action application="set" data="sip_redirect_profile=internal" />
  </condition>
</extension>

Once a 302 is received, your call with redirect to the "redirected"  
dialplan context (or to the one specified in sip_redirect_context and  
sip_redirect_dialplan). It should like the following:

<context name="redirected">
  <extension name="redirected">
   <condition>
     <action application="bridge" data="${sip_redirect_dialstring}" />
   </condition>
  </extension>
</context>

Note: you can also check if the user is allowed to do redirects while  
using this method, so, lets say, one of your carrier can't 302 you to  
another of your carriers.


Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 8-Feb-10, at 8:05 PM, Lon Baker wrote:

> Hi,
>
> I need to bridge calls to a proxy that issues a redirect command. Its
> an internal proxy that translates internal SIP URIs into outbound
> URIs.
>
> I have one interface on a private network where the proxy is running
> and another interface on my public network.
>
> If I am reading the debug information correctly, the bridge to the
> proxy is getting the redirect correctly, but since it is on the
> internal network (profile), the call never transitions to the public
> network to complete the call.
>
> Is it possible to do this?
>
> Lon
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From dujinfang at gmail.com  Mon Feb  8 22:20:05 2010
From: dujinfang at gmail.com (Seven Du)
Date: Tue, 9 Feb 2010 14:20:05 +0800
Subject: [Freeswitch-users] CentOS 5.3 (Final) with Skype Static Build
In-Reply-To: <7b197bef1002080847v2524f99dt638c1668ac71425a@mail.gmail.com>
References: <c55d46951002080706t6bb8531i4f12cb35af12e6de@mail.gmail.com>
	<7b197bef1002080847v2524f99dt638c1668ac71425a@mail.gmail.com>
Message-ID: <23f91031002082220p770fcfdep9a3ae60296a44654@mail.gmail.com>

Ciao Giovanni,

I'm about to install a new server and will try to migrate to CentOS
from Ubuntu, and based on this list Anthony mentioned CentOS5.4 has
some problems on some tool chain, but I noticed that you said you are
on the latest 5.4 with skypiax, do you have any problems?

The new server will mainly be used to do SIP and skype gateway. And I
have no experience on CentOS 5.

Thanks.

2010/2/9 Giovanni Maruzzelli <gmaruzz at celliax.org>:
> Max,
>
> Just checked, the static build of skype-beta works on latest centos (5.4)
>
> But you can have the same results if just tried :).
>
> ldd tells you which libraries are missing, the you install those libraries.
> If you dont know which package a library is into, just do, eg:
>
> yum search libXss
>
> and you'll be told.
>
> ========================
> So, all in all:
>
> you have to install:
>
> yum install libXv
> yum install libXScrnSaver
>
> then use the original snd-dummy supplied with centos (the modified one do not
> works at the moment - because the skype-beta does some alsa calls it was
> not doing in previous versions, that are not implemented in the
> modified alsa driver, I'll fix it soon).
>
> ===============================================
>
> -gm
>
>
>
> On Mon, Feb 8, 2010 at 4:06 PM, Max Bridgewater
> <max.bridgewater at gmail.com> wrote:
>> Hi Giovanni,
>>
>> Let me start a new thread so I don't hijack Peter's one. Here is what
>> I get for ldd /usr/bin/skype. What is the deal with these libXv.so.1,
>> libXss.so.1?
>>
>> ? ? ? ?linux-gate.so.1 => ?(0x00697000)
>> ? ? ? ?libasound.so.2 => /lib/libasound.so.2 (0x008b7000)
>> ? ? ? ?libXv.so.1 => not found
>> ? ? ? ?libXss.so.1 => not found
>> ? ? ? ?libSM.so.6 => /usr/lib/libSM.so.6 (0x00d02000)
>> ? ? ? ?libICE.so.6 => /usr/lib/libICE.so.6 (0x0036b000)
>> ? ? ? ?libXi.so.6 => /usr/lib/libXi.so.6 (0x00111000)
>> ? ? ? ?libXrender.so.1 => /usr/lib/libXrender.so.1 (0x00119000)
>> ? ? ? ?libXrandr.so.2 => /usr/lib/libXrandr.so.2 (0x00b77000)
>> ? ? ? ?libfreetype.so.6 => /usr/lib/libfreetype.so.6 (0x00122000)
>> ? ? ? ?libfontconfig.so.1 => /usr/lib/libfontconfig.so.1 (0x009fb000)
>> ? ? ? ?libXext.so.6 => /usr/lib/libXext.so.6 (0x00eb0000)
>> ? ? ? ?libX11.so.6 => /usr/lib/libX11.so.6 (0x001a2000)
>> ? ? ? ?libz.so.1 => /usr/lib/libz.so.1 (0x002a5000)
>> ? ? ? ?libgthread-2.0.so.0 => /lib/libgthread-2.0.so.0 (0x006bd000)
>> ? ? ? ?libglib-2.0.so.0 => /lib/libglib-2.0.so.0 (0x002be000)
>> ? ? ? ?librt.so.1 => /lib/librt.so.1 (0x009a1000)
>> ? ? ? ?libdl.so.2 => /lib/libdl.so.2 (0x00604000)
>> ? ? ? ?libpthread.so.0 => /lib/libpthread.so.0 (0x004e6000)
>> ? ? ? ?libstdc++.so.6 => /usr/lib/libstdc++.so.6 (0x004fd000)
>> ? ? ? ?libm.so.6 => /lib/libm.so.6 (0x00385000)
>> ? ? ? ?libgcc_s.so.1 => /lib/libgcc_s.so.1 (0x00bcc000)
>> ? ? ? ?libc.so.6 => /lib/libc.so.6 (0x006c2000)
>> ? ? ? ?libexpat.so.0 => /lib/libexpat.so.0 (0x00c2a000)
>> ? ? ? ?libXau.so.6 => /usr/lib/libXau.so.6 (0x002ba000)
>> ? ? ? ?libXdmcp.so.6 => /usr/lib/libXdmcp.so.6 (0x0087e000)
>> ? ? ? ?/lib/ld-linux.so.2 (0x00476000)
>>
>> Max
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From dujinfang at gmail.com  Mon Feb  8 22:22:27 2010
From: dujinfang at gmail.com (Seven Du)
Date: Tue, 9 Feb 2010 14:22:27 +0800
Subject: [Freeswitch-users] CentOS 5.3 (Final) with Skype Static Build
In-Reply-To: <23f91031002082220p770fcfdep9a3ae60296a44654@mail.gmail.com>
References: <c55d46951002080706t6bb8531i4f12cb35af12e6de@mail.gmail.com>
	<7b197bef1002080847v2524f99dt638c1668ac71425a@mail.gmail.com>
	<23f91031002082220p770fcfdep9a3ae60296a44654@mail.gmail.com>
Message-ID: <23f91031002082222q75228efai24d0d89933713cdc@mail.gmail.com>

or I will try CentOS 5.3 and Skype2.1 beta.   But it sames skype has a
64bit native on Ubuntu 8.10 which not available on CentOS.

2010/2/9 Seven Du <dujinfang at gmail.com>:
> Ciao Giovanni,
>
> I'm about to install a new server and will try to migrate to CentOS
> from Ubuntu, and based on this list Anthony mentioned CentOS5.4 has
> some problems on some tool chain, but I noticed that you said you are
> on the latest 5.4 with skypiax, do you have any problems?
>
> The new server will mainly be used to do SIP and skype gateway. And I
> have no experience on CentOS 5.
>
> Thanks.
>
> 2010/2/9 Giovanni Maruzzelli <gmaruzz at celliax.org>:
>> Max,
>>
>> Just checked, the static build of skype-beta works on latest centos (5.4)
>>
>> But you can have the same results if just tried :).
>>
>> ldd tells you which libraries are missing, the you install those libraries.
>> If you dont know which package a library is into, just do, eg:
>>
>> yum search libXss
>>
>> and you'll be told.
>>
>> ========================
>> So, all in all:
>>
>> you have to install:
>>
>> yum install libXv
>> yum install libXScrnSaver
>>
>> then use the original snd-dummy supplied with centos (the modified one do not
>> works at the moment - because the skype-beta does some alsa calls it was
>> not doing in previous versions, that are not implemented in the
>> modified alsa driver, I'll fix it soon).
>>
>> ===============================================
>>
>> -gm
>>
>>
>>
>> On Mon, Feb 8, 2010 at 4:06 PM, Max Bridgewater
>> <max.bridgewater at gmail.com> wrote:
>>> Hi Giovanni,
>>>
>>> Let me start a new thread so I don't hijack Peter's one. Here is what
>>> I get for ldd /usr/bin/skype. What is the deal with these libXv.so.1,
>>> libXss.so.1?
>>>
>>> ? ? ? ?linux-gate.so.1 => ?(0x00697000)
>>> ? ? ? ?libasound.so.2 => /lib/libasound.so.2 (0x008b7000)
>>> ? ? ? ?libXv.so.1 => not found
>>> ? ? ? ?libXss.so.1 => not found
>>> ? ? ? ?libSM.so.6 => /usr/lib/libSM.so.6 (0x00d02000)
>>> ? ? ? ?libICE.so.6 => /usr/lib/libICE.so.6 (0x0036b000)
>>> ? ? ? ?libXi.so.6 => /usr/lib/libXi.so.6 (0x00111000)
>>> ? ? ? ?libXrender.so.1 => /usr/lib/libXrender.so.1 (0x00119000)
>>> ? ? ? ?libXrandr.so.2 => /usr/lib/libXrandr.so.2 (0x00b77000)
>>> ? ? ? ?libfreetype.so.6 => /usr/lib/libfreetype.so.6 (0x00122000)
>>> ? ? ? ?libfontconfig.so.1 => /usr/lib/libfontconfig.so.1 (0x009fb000)
>>> ? ? ? ?libXext.so.6 => /usr/lib/libXext.so.6 (0x00eb0000)
>>> ? ? ? ?libX11.so.6 => /usr/lib/libX11.so.6 (0x001a2000)
>>> ? ? ? ?libz.so.1 => /usr/lib/libz.so.1 (0x002a5000)
>>> ? ? ? ?libgthread-2.0.so.0 => /lib/libgthread-2.0.so.0 (0x006bd000)
>>> ? ? ? ?libglib-2.0.so.0 => /lib/libglib-2.0.so.0 (0x002be000)
>>> ? ? ? ?librt.so.1 => /lib/librt.so.1 (0x009a1000)
>>> ? ? ? ?libdl.so.2 => /lib/libdl.so.2 (0x00604000)
>>> ? ? ? ?libpthread.so.0 => /lib/libpthread.so.0 (0x004e6000)
>>> ? ? ? ?libstdc++.so.6 => /usr/lib/libstdc++.so.6 (0x004fd000)
>>> ? ? ? ?libm.so.6 => /lib/libm.so.6 (0x00385000)
>>> ? ? ? ?libgcc_s.so.1 => /lib/libgcc_s.so.1 (0x00bcc000)
>>> ? ? ? ?libc.so.6 => /lib/libc.so.6 (0x006c2000)
>>> ? ? ? ?libexpat.so.0 => /lib/libexpat.so.0 (0x00c2a000)
>>> ? ? ? ?libXau.so.6 => /usr/lib/libXau.so.6 (0x002ba000)
>>> ? ? ? ?libXdmcp.so.6 => /usr/lib/libXdmcp.so.6 (0x0087e000)
>>> ? ? ? ?/lib/ld-linux.so.2 (0x00476000)
>>>
>>> Max
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> Cell : +39-347-2665618
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>


From tayeb.meftah at gmail.com  Mon Feb  8 23:03:06 2010
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Tue, 09 Feb 2010 08:03:06 +0100
Subject: [Freeswitch-users] UPnP Timeout
In-Reply-To: <9706F7A8-7644-4602-B1DE-289A35F788AA@tlainvestments.com>
References: <9706F7A8-7644-4602-B1DE-289A35F788AA@tlainvestments.com>
Message-ID: <4B7108AA.9000101@gmail.com>

hi,
no problem in windows or debian GNU/Linux
mayb is a pfsense problem
if the problem percist start fs with: ./freeswitch -nonat and redirect 
your required ports;)
thanks
Le 09/02/2010 05:21, Troy Anderson a ?crit :
> I have been using pfSense (1.2.3) and FS.  FS nicely uses UPnP to poke holes in the firewall, but it seems that the holes close after a while.  I cannot find any documentation in FS nor in pfSense as to what the timeout is.  Is there a setting in FS to do some kind of keep-alive thing with UPnP to keep, e.g. 5060, open?  Or is it already doing that and pfSense is the issue?
>
> Thanks!
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>    



From matt at webcontracts.co.uk  Tue Feb  9 01:53:47 2010
From: matt at webcontracts.co.uk (Matthew Law)
Date: Tue, 9 Feb 2010 09:53:47 -0000
Subject: [Freeswitch-users] ACL question and js error
In-Reply-To: <20100207145907.GF31942@base.carmickle.com>
References: <ea05d45dca9be2d979fb386f960bc077.squirrel@www.webcontracts.co.uk>
	<20100207145907.GF31942@base.carmickle.com>
Message-ID: <cfe871e3305675257b8c74ed4fb5a11b.squirrel@www.webcontracts.co.uk>

On Sun, February 7, 2010 2:59 pm, Frank Carmickle wrote:
> On Sun, Feb 07, Matthew Law wrote:
>> After some more experiments I have a working replacement for the
>> asterisk
>> box we were using before, which is great.
>>
>> I had problems getting incoming calls to work. Changing the entry in
>> acl.conf.xml from:
>>
>> <list name="domains" default="deny">
>>    <node type="allow" domain="$${domain}"/>
>> </list>
>>
>> to:
>>
>> <list name="domains" default="allow">
>>    <node type="allow" domain="$${domain}"/>
>> </list>
>>
>> and reloading xml works but this gets reverted every time FS starts up.
>> I've scanned the wiki docs and can't see anything pertaining to that.
>> Why/where is this happening and how do I make it the default?  Actually,
>> the question should probably be is it sensible to do that? - the box is
>> out on the internet and I really only want to take incoming calls from
>> voiptalk.org, but I can't find a list of IPs on their site which I could
>> create an acl from...
>
> This is what gateway definitions are for in sofia.

I'm still struggling with this.  How where do I tell sofia to allow
incoming connections from this gateway?

Here's my sip_profiles/external/voiptalk.org.xml with the sensitive stuff
removed:

<include>
        <gateway name="voiptalk">
                <param name="username" value="XXXXXXX" />
                <param name="realm" value="voiptalk.org" />
                <param name="from-user" value="XXXXXXX"/>
                <param name="from-domain" value="voiptalk.org"/>
                <param name="password" value="XXXXXXX" />
                <param name="extension" value="voiptalk-incoming"/>
                <param name="proxy" value="voiptalk.org"/>
                <param name="register-proxy" value="voiptalk.org"/>
                <param name="expire-seconds" value="600"/>
                <param name="register" value="true"/>
                <param name="ping" value="25"/>
        </gateway>
</include>

Do I need to add something to this file or maybe sofia.conf.xml to allow
connections from this domain?  Most everything else is working now, just
banging my head on this.


Thanks,

Matt.



From rupa at rupa.com  Tue Feb  9 02:07:24 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Tue, 9 Feb 2010 04:07:24 -0600
Subject: [Freeswitch-users] UPnP Timeout
In-Reply-To: <9706F7A8-7644-4602-B1DE-289A35F788AA@tlainvestments.com>
References: <9706F7A8-7644-4602-B1DE-289A35F788AA@tlainvestments.com>
Message-ID: <b39273531002090207n11d54e69h8d4ff6f7e71c731c@mail.gmail.com>

I believe FS opens the ports with an indefinite timeout (never close).  I'd
have to double check.  In addition, FS refreshes the NAT mappings on every
keep-alive packet sent by the upnp gateway.  Have you done a nat_map status
once the ports are missing in pfsense to see if fs still thinks the ports
should be open?  What if you do a nat_map republish?  Do the maps get pushed
to pfsense and then stay open for a whlie?

Perhaps pfsense is sending a keep-alive packet that we don't process right
or is invalid? If so, I'd need a packet trace to do analysis.

On Mon, Feb 8, 2010 at 10:21 PM, Troy Anderson <troy at tlainvestments.com>wrote:

> I have been using pfSense (1.2.3) and FS.  FS nicely uses UPnP to poke
> holes in the firewall, but it seems that the holes close after a while.  I
> cannot find any documentation in FS nor in pfSense as to what the timeout
> is.  Is there a setting in FS to do some kind of keep-alive thing with UPnP
> to keep, e.g. 5060, open?  Or is it already doing that and pfSense is the
> issue?
>
> Thanks!
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
-Rupa
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100209/ae52bf49/attachment.html 

From nagalenoj at gmail.com  Tue Feb  9 05:19:34 2010
From: nagalenoj at gmail.com (Nagalenoj H.)
Date: Tue, 9 Feb 2010 18:49:34 +0530
Subject: [Freeswitch-users] Play music to A leg.
Message-ID: <fe67a3341002090519r2cc6e8edm7702b8d012fc2be3@mail.gmail.com>

Dear friends,
  In event socket, I'm originating a call to a number from A leg and till
the person answers the call, I would want to play some music to the A leg,
till I bridge these A leg and originated call.

  I don't want to use bridge, in which I could use ringback.

  So, what is the way to do this??

-- 
Regards,
Nagalenoj H.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100209/67e63d5f/attachment.html 

From peder at networkoblivion.com  Tue Feb  9 05:51:22 2010
From: peder at networkoblivion.com (Peder)
Date: Tue, 9 Feb 2010 07:51:22 -0600
Subject: [Freeswitch-users] Announcement: FreeSWITCH Add Broadsoft
	SCA	Support
In-Reply-To: <d361ca7b1002082110h4e3c60d0p88412f753444ec3d@mail.gmail.com>
References: <87f2f3b91001111742k691966cfs82515db77c13d4d4@mail.gmail.com>
	<d361ca7b1002082110h4e3c60d0p88412f753444ec3d@mail.gmail.com>
Message-ID: <016d01caa98e$f6df25f0$e49d71d0$@com>

It is in addition to the existing settings.  It is for SCA presence on
shared lines.  The "manage presence" setting is for regular registrations.

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Yehavi
Bourvine
Sent: Monday, February 08, 2010 11:10 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Announcement: FreeSWITCH Add Broadsoft SCA
Support

 

Hello,

 

  We currently use the "old" type of presence which is activated by
"manage-presence" coupled with "dbname" and "presence-hosts".

With the new method, does "manage-shared-presence" replace all of the above
or comes in addition?

 

                       Thanks! __Yehavi:

2010/1/12 Michael Collins <msc at freeswitch.org>

We want to let everyone know that FreeSWITCH now supports the Broadsoft SCA
method of doing shared lines. The story is here:

http://www.freeswitch.org/node/227

Tony and Brian spent many hours laboring over this, so please be sure to
show your appreciation to them for this new feature and all of the great
things they do for the FreeSWITCH community and VoIP in general!

-Michael


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org <http://www.freeswitch.org/> 

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100209/16814a2a/attachment-0001.html 

From Prometheus001 at gmx.net  Tue Feb  9 06:09:20 2010
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Tue, 09 Feb 2010 15:09:20 +0100
Subject: [Freeswitch-users] mod_skypiax - help needed with
 modified	snd-dummy (one way audio)
In-Reply-To: <7b197bef1002080839o61b4b1b5k6d5cc39bb05d9a32@mail.gmail.com>
References: <4B60555B.2020004@gmx.net>	<7b197bef1002051319l2fcb5617p1691692f58c65372@mail.gmail.com>
	<4B6CC792.5060608@gmx.net>	<191c3a031002060558r36947d93vcbb46df9e0a7dfde@mail.gmail.com>
	<4B6F3E9A.2020103@gmx.net>	<7b197bef1002080529u1afc7f9av492b0c221ee2d6af@mail.gmail.com>
	<c55d46951002080546p829a881w256bfdb19334c548@mail.gmail.com>
	<7b197bef1002080553qd1cb42dla26425942de83c43@mail.gmail.com>
	<c55d46951002080604k400344b0i8add00b5c6014958@mail.gmail.com>
	<4B703AD0.2080909@gmx.net>
	<7b197bef1002080839o61b4b1b5k6d5cc39bb05d9a32@mail.gmail.com>
Message-ID: <4B716C90.8070109@gmx.net>

Hello Giovanni,

I will try to contact you via IRC (stony)

Best regards
Peter

Giovanni Maruzzelli schrieb:
> Peter,
>
> excuse me but I really do not follow you.
>
> Why you have the normal static build not working?
>
> Also, this is really taking too much of my time. You continue to
> change things, and report issues, without waiting for solutions you
> ask for, then you report something else, and so on... we'll never get
> at the end of this.
>
> If you want, please connect via IRC, and contact me (gmaruzz).
> Or let me connect ssh to your machine.
>
> -gm
>
>
> On Mon, Feb 8, 2010 at 5:24 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>   
>> I got it working now with static build and an older version of skype
>> (skype_static-2.1.0.47).
>>
>> But I still have a problem ongoing with sound quality, resp. one way audio:
>> With original Alsa driver: When Skype calls mod_skypiax => SIP Phone,
>> then the sound from the SIP phone is interupted regularly 2 times a second.
>> Example:
>> When a person on the sip phone speaks
>> "aaaaaaaaaaaaaaaaaaaaaa"
>> the other site hears
>> "aaatataaaatataaaatataaaa"
>> With "t" meaning the interruption of the sound.
>>
>> So I compliled and installed the modified alsa driver as described in
>> the wiki (configure, make and make install, remove old ubuntu sound dir
>> in /lib/modules/2.6.24-24-server/ubuntu/sound + depmod -a + reboot the
>> server.
>>
>> Now the SIP phone is heard loud and clearly without interruption.
>> However the other direction is not heard, so we're at the beginning of
>> the post (one way audio). Only when I really scratch the microphone then
>> I hear some parts of this scratching on the SIP side.
>>
>> So, some more hints are needed.
>>
>> Here's the log, when I start the skype client:
>>  su root -c "/bin/echo 'username password'| DISPLAY=:101
>> /usr/bin/skype1 --pipelogin &" &
>>  /usr/bin/Xvfb :102 -ac &
>>  error opening security policy file /etc/X11/xserver/SecurityPolicy
>>  expected keysym, got XF86KbdLightOnOff: line 70 of pc
>>  expected keysym, got XF86KbdBrightnessDown: line 71 of pc
>>  expected keysym, got XF86KbdBrightnessUp: line 72 of pc
>>  Could not init font path element /usr/share/fonts/X11/cyrillic,
>> removing from list!
>>  Could not init font path element
>> /var/lib/defoma/x-ttcidfont-conf.d/dirs/TrueType, removing from list!
>> But these messages are not critical, right?
>>
>> Best regards
>> Peter
>>
>>
>> Max Bridgewater schrieb:
>>     
>>> Well, it was a couple of weeks ago. I then switched back to 2.0.4. I'm
>>> going to try it again and let you know.
>>>
>>> Max.
>>>
>>> On Mon, Feb 8, 2010 at 8:53 AM, Giovanni Maruzzelli <gmaruzz at celliax.org> wrote:
>>>
>>>       
>>>> Peter is using hardy 64 bit. I checked on that.
>>>>
>>>> But, let me understand: if you're using a static build, why you have a
>>>> problem with QT?
>>>> Is actually Qt to be statically linked...
>>>>
>>>> what is the results of:
>>>>
>>>> ldd skype
>>>>
>>>> Anyway, in a short while I'll check that on centos 5.4 too, and I'll report here
>>>>
>>>> -giovanni
>>>>
>>>> On Mon, Feb 8, 2010 at 2:46 PM, Max Bridgewater
>>>> <max.bridgewater at gmail.com> wrote:
>>>>
>>>>         
>>>>> Interesting; a while back I tried to install Skypiax with the latest
>>>>> static build on Skype.com. I had QT library compatibility problem on a
>>>>> CentOS 5.4 (Final). Giovannni, may I ask what Linux distro you are
>>>>> using?
>>>>>
>>>>> Thanks,
>>>>> max.
>>>>>
>>>>> On Mon, Feb 8, 2010 at 8:29 AM, Giovanni Maruzzelli <gmaruzz at celliax.org> wrote:
>>>>>
>>>>>           
>>>>>> Peter,
>>>>>>
>>>>>> I just tested with the static build you find on skype.com
>>>>>>
>>>>>> I never tested for performances or other issues, there may be (it's a beta).
>>>>>>
>>>>>> But it do not crash on me.
>>>>>>
>>>>>> I have no problem at all.
>>>>>>
>>>>>> If you can give me ssh access I can try to understand why you have so
>>>>>> many problems.
>>>>>>
>>>>>> Or, alternatively, try to follow the wiki. You know, I've not heard
>>>>>> about those problems.
>>>>>>
>>>>>> root at hardy64:~/skype.beta/skype_static-2.1.0.81# ldd skype
>>>>>>        linux-gate.so.1 =>  (0xffffe000)
>>>>>>        libasound.so.2 => /usr/lib32/libasound.so.2 (0xf7eb7000)
>>>>>>        libXv.so.1 => /usr/lib32/libXv.so.1 (0xf7eb2000)
>>>>>>        libXss.so.1 => /usr/lib32/libXss.so.1 (0xf7eae000)
>>>>>>        libSM.so.6 => /usr/lib32/libSM.so.6 (0xf7ea6000)
>>>>>>        libICE.so.6 => /usr/lib32/libICE.so.6 (0xf7e8e000)
>>>>>>        libXi.so.6 => /usr/lib32/libXi.so.6 (0xf7e86000)
>>>>>>        libXrender.so.1 => /usr/lib32/libXrender.so.1 (0xf7e7e000)
>>>>>>        libXrandr.so.2 => /usr/lib32/libXrandr.so.2 (0xf7e78000)
>>>>>>        libfreetype.so.6 => /usr/lib32/libfreetype.so.6 (0xf7e0a000)
>>>>>>        libfontconfig.so.1 => /usr/lib32/libfontconfig.so.1 (0xf7de0000)
>>>>>>        libXext.so.6 => /usr/lib32/libXext.so.6 (0xf7dd2000)
>>>>>>        libX11.so.6 => /usr/lib32/libX11.so.6 (0xf7ceb000)
>>>>>>        libz.so.1 => /usr/lib32/libz.so.1 (0xf7cd6000)
>>>>>>        libgthread-2.0.so.0 => /usr/lib32/libgthread-2.0.so.0 (0xf7cd1000)
>>>>>>        libglib-2.0.so.0 => /usr/lib32/libglib-2.0.so.0 (0xf7c1f000)
>>>>>>        librt.so.1 => /lib32/librt.so.1 (0xf7c16000)
>>>>>>        libdl.so.2 => /lib32/libdl.so.2 (0xf7c12000)
>>>>>>        libpthread.so.0 => /lib32/libpthread.so.0 (0xf7bfa000)
>>>>>>        libstdc++.so.6 => /usr/lib32/libstdc++.so.6 (0xf7b07000)
>>>>>>        libm.so.6 => /lib32/libm.so.6 (0xf7ae2000)
>>>>>>        libgcc_s.so.1 => /usr/lib32/libgcc_s.so.1 (0xf7ad6000)
>>>>>>        libc.so.6 => /lib32/libc.so.6 (0xf7987000)
>>>>>>        libexpat.so.1 => /usr/lib32/libexpat.so.1 (0xf7966000)
>>>>>>        libXau.so.6 => /usr/lib32/libXau.so.6 (0xf7963000)
>>>>>>        libxcb-xlib.so.0 => /usr/lib32/libxcb-xlib.so.0 (0xf7961000)
>>>>>>        libxcb.so.1 => /usr/lib32/libxcb.so.1 (0xf7948000)
>>>>>>        libpcre.so.3 => /usr/lib32/libpcre.so.3 (0xf7921000)
>>>>>>        /lib/ld-linux.so.2 (0xf7f86000)
>>>>>>        libXdmcp.so.6 => /usr/lib32/libXdmcp.so.6 (0xf791c000)
>>>>>>
>>>>>>
>>>>>> On Sun, Feb 7, 2010 at 11:28 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>>
>>>>>>             
>>>>>>> I now used the static Skype binary in order to avoid missing constraints
>>>>>>> to other libraries: It still crashes
>>>>>>> 1st it starts with:
>>>>>>>  process 15431: D-Bus library appears to be incorrectly set up; failed
>>>>>>> to read machine uuid: Failed to open "/var/lib/dbus/machine-id": No such
>>>>>>> file or directory
>>>>>>>  See the manual page for dbus-uuidgen to correct this issue.
>>>>>>> After calling this client it crashes with:
>>>>>>>   /usr/bin/skype: symbol lookup error: /usr/bin/skype: undefined
>>>>>>> symbol: _ZN10QBoxLayout13addSpacerItemEP11QSpacerItem
>>>>>>>
>>>>>>> Any hints, where I may get an older Skype client?
>>>>>>>
>>>>>>> Best regards
>>>>>>> Peter
>>>>>>>
>>>>>>> Anthony Minessale schrieb:
>>>>>>>
>>>>>>>               
>>>>>>>> Quick fyi centos 5.4 has some memory bugs in libc don't go beyond 5.3
>>>>>>>> until its fixed.
>>>>>>>>
>>>>>>>>
>>>>>>>>                 
>>>>>>>>> On Feb 5, 2010 7:42 PM, "Peter P GMX" <Prometheus001 at gmx.net
>>>>>>>>> <mailto:Prometheus001 at gmx.net>> wrote:
>>>>>>>>>
>>>>>>>>> Skype starts, but as soon as it receives a call it crashes with:
>>>>>>>>>
>>>>>>>>> /usr/bin/skype: symbol lookup error: /usr/bin/skype: undefined symbol:
>>>>>>>>>
>>>>>>>>> _ZN10QBoxLayout13addSpacerItemEP11QSpacerItem
>>>>>>>>>
>>>>>>>>> I think the 8.10 version dos not work with8.04.
>>>>>>>>>
>>>>>>>>> Any hints, where I may get an older Skype client? I may also try the
>>>>>>>>> static skype client.
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> Best regards
>>>>>>>>> Peter
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> .
>>>>>>>>> Giovanni Maruzzelli schrieb:
>>>>>>>>>
>>>>>>>>>                   
>>>>>>>>>> that's not at all a fatal error.
>>>>>>>>>> I believe...
>>>>>>>>>>
>>>>>>>>>>                     
>>>>>>>> ------------------------------------------------------------------------
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>>                 
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>>               
>>>>>> --
>>>>>> Sincerely,
>>>>>>
>>>>>> Giovanni Maruzzelli
>>>>>> Cell : +39-347-2665618
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>>             
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>>           
>>>> --
>>>> Sincerely,
>>>>
>>>> Giovanni Maruzzelli
>>>> Cell : +39-347-2665618
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>         
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>       
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>     
>
>
>
>   


From msc at freeswitch.org  Tue Feb  9 07:35:52 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 9 Feb 2010 07:35:52 -0800
Subject: [Freeswitch-users] Last call: buy the devs dinner!
Message-ID: <87f2f3b91002090735g1c6c69eby5274f6dfa9127fc5@mail.gmail.com>

Hey all,

Thanks so much for the donations that have come in already! We appreciate
your generosity. The dev team really wants to release 1.0.5 but they're
kinda hungry! :) Please hit the PayPal button on the main
freeswitch.orgpage to drop a few dollars in the hat. Also, keep in
mind that we have the
"extended family" of developers all here so it's not just Tony, Mike, and
Brian. Let's all pitch in and have a great dinner for them.

Thanks!
-Michael
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100209/31eaa19d/attachment.html 

From msc at freeswitch.org  Tue Feb  9 07:37:13 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 9 Feb 2010 07:37:13 -0800
Subject: [Freeswitch-users] Play music to A leg.
In-Reply-To: <fe67a3341002090519r2cc6e8edm7702b8d012fc2be3@mail.gmail.com>
References: <fe67a3341002090519r2cc6e8edm7702b8d012fc2be3@mail.gmail.com>
Message-ID: <87f2f3b91002090737l4efa8en29bde9457fef7ea5@mail.gmail.com>

On Tue, Feb 9, 2010 at 5:19 AM, Nagalenoj H. <nagalenoj at gmail.com> wrote:

> Dear friends,
>   In event socket, I'm originating a call to a number from A leg and till
> the person answers the call, I would want to play some music to the A leg,
> till I bridge these A leg and originated call.
>
>   I don't want to use bridge, in which I could use ringback.
>
You don't want to use bridge because... why?
-MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100209/dc809e44/attachment.html 

From troy at tlainvestments.com  Tue Feb  9 07:44:02 2010
From: troy at tlainvestments.com (Troy Anderson)
Date: Tue, 9 Feb 2010 08:44:02 -0700
Subject: [Freeswitch-users] UPnP Timeout
In-Reply-To: <b39273531002090207n11d54e69h8d4ff6f7e71c731c@mail.gmail.com>
References: <9706F7A8-7644-4602-B1DE-289A35F788AA@tlainvestments.com>
	<b39273531002090207n11d54e69h8d4ff6f7e71c731c@mail.gmail.com>
Message-ID: <47D355D1-CDF6-4D79-8A64-1134EBEB36BA@tlainvestments.com>

I did do a nap_map status when the ports were missing from pfSense and FS thought they were still open.  I didn't know about nat_map republish, but will try next time.  I think the timeframe is days, so this is kind of hard to diagnose.  I may add a periodic nat_map republish from fs_cli to our production systems.

In any case, I'll keep an eye on it and try nat_map republish next time pfSense drops the ports to be sure that is working in this environment.

In the meantime, which .c file(s) can I peruse to learn more?

Thanks!
Troy

On Feb 9, 2010, at 3:07 AM, Rupa Schomaker wrote:

> I believe FS opens the ports with an indefinite timeout (never close).  I'd have to double check.  In addition, FS refreshes the NAT mappings on every keep-alive packet sent by the upnp gateway.  Have you done a nat_map status once the ports are missing in pfsense to see if fs still thinks the ports should be open?  What if you do a nat_map republish?  Do the maps get pushed to pfsense and then stay open for a whlie?
> 
> Perhaps pfsense is sending a keep-alive packet that we don't process right or is invalid? If so, I'd need a packet trace to do analysis.  
> 
> On Mon, Feb 8, 2010 at 10:21 PM, Troy Anderson <troy at tlainvestments.com> wrote:
> I have been using pfSense (1.2.3) and FS.  FS nicely uses UPnP to poke holes in the firewall, but it seems that the holes close after a while.  I cannot find any documentation in FS nor in pfSense as to what the timeout is.  Is there a setting in FS to do some kind of keep-alive thing with UPnP to keep, e.g. 5060, open?  Or is it already doing that and pfSense is the issue?
> 
> Thanks!
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> -- 
> -Rupa
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100209/5ac7b97c/attachment.html 

From msc at freeswitch.org  Tue Feb  9 07:45:26 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 9 Feb 2010 07:45:26 -0800
Subject: [Freeswitch-users] ACL question and js error
In-Reply-To: <cfe871e3305675257b8c74ed4fb5a11b.squirrel@www.webcontracts.co.uk>
References: <ea05d45dca9be2d979fb386f960bc077.squirrel@www.webcontracts.co.uk>
	<20100207145907.GF31942@base.carmickle.com>
	<cfe871e3305675257b8c74ed4fb5a11b.squirrel@www.webcontracts.co.uk>
Message-ID: <87f2f3b91002090745v2128714byf1f7574d75f4449c@mail.gmail.com>

On Tue, Feb 9, 2010 at 1:53 AM, Matthew Law <matt at webcontracts.co.uk> wrote:

> On Sun, February 7, 2010 2:59 pm, Frank Carmickle wrote:
> > On Sun, Feb 07, Matthew Law wrote:
> >> After some more experiments I have a working replacement for the
> >> asterisk
> >> box we were using before, which is great.
> >>
> >> I had problems getting incoming calls to work. Changing the entry in
> >> acl.conf.xml from:
> >>
> >> <list name="domains" default="deny">
> >>    <node type="allow" domain="$${domain}"/>
> >> </list>
> >>
> >> to:
> >>
> >> <list name="domains" default="allow">
> >>    <node type="allow" domain="$${domain}"/>
> >> </list>
> >>
> >> and reloading xml works but this gets reverted every time FS starts up.
> >> I've scanned the wiki docs and can't see anything pertaining to that.
> >> Why/where is this happening and how do I make it the default?  Actually,
> >> the question should probably be is it sensible to do that? - the box is
> >> out on the internet and I really only want to take incoming calls from
> >> voiptalk.org, but I can't find a list of IPs on their site which I
> could
> >> create an acl from...
> >
> > This is what gateway definitions are for in sofia.
>
> I'm still struggling with this.  How where do I tell sofia to allow
> incoming connections from this gateway?
>
> Here's my sip_profiles/external/voiptalk.org.xml with the sensitive stuff
> removed:
>
> <include>
>        <gateway name="voiptalk">
>                <param name="username" value="XXXXXXX" />
>                <param name="realm" value="voiptalk.org" />
>                <param name="from-user" value="XXXXXXX"/>
>                <param name="from-domain" value="voiptalk.org"/>
>                <param name="password" value="XXXXXXX" />
>                <param name="extension" value="voiptalk-incoming"/>
>                <param name="proxy" value="voiptalk.org"/>
>                <param name="register-proxy" value="voiptalk.org"/>
>                <param name="expire-seconds" value="600"/>
>                <param name="register" value="true"/>
>                <param name="ping" value="25"/>
>        </gateway>
> </include>
>
> Do I need to add something to this file or maybe sofia.conf.xml to allow
> connections from this domain?  Most everything else is working now, just
> banging my head on this.
>
> Matt,

Are you trying to let calls in from voiptalk.org? Do you want to auth all
inbound calls or do you just want to blanket allow them and handle them in
the dialplan? If you just want to allow calls in from the voiptalk.org IP
address then you need to use the cidr tag in acl.conf.xml:
<list name="domains" default="deny">
    <node type="allow" domain="$${domain}"/>
    <node type="allow" cidr="1.2.3.4/32"/>
 </list>

-MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100209/7685f314/attachment-0001.html 

From rupa at rupa.com  Tue Feb  9 08:06:11 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Tue, 9 Feb 2010 10:06:11 -0600
Subject: [Freeswitch-users] UPnP Timeout
In-Reply-To: <47D355D1-CDF6-4D79-8A64-1134EBEB36BA@tlainvestments.com>
References: <9706F7A8-7644-4602-B1DE-289A35F788AA@tlainvestments.com>
	<b39273531002090207n11d54e69h8d4ff6f7e71c731c@mail.gmail.com>
	<47D355D1-CDF6-4D79-8A64-1134EBEB36BA@tlainvestments.com>
Message-ID: <b39273531002090806k7fa7f02dyeabd79586736041b@mail.gmail.com>

perhaps pfSense isn't sending the keep-alive packets like we expect?

You can look in switch_nat.c for details.

On Tue, Feb 9, 2010 at 9:44 AM, Troy Anderson <troy at tlainvestments.com>wrote:

> I did do a nap_map status when the ports were missing from pfSense and FS
> thought they were still open.  I didn't know about nat_map republish, but
> will try next time.  I think the timeframe is days, so this is kind of hard
> to diagnose.  I may add a periodic nat_map republish from fs_cli to our
> production systems.
>
> In any case, I'll keep an eye on it and try nat_map republish next time
> pfSense drops the ports to be sure that is working in this environment.
>
> In the meantime, which .c file(s) can I peruse to learn more?
>
> Thanks!
> Troy
>
>
> On Feb 9, 2010, at 3:07 AM, Rupa Schomaker wrote:
>
> I believe FS opens the ports with an indefinite timeout (never close).  I'd
> have to double check.  In addition, FS refreshes the NAT mappings on every
> keep-alive packet sent by the upnp gateway.  Have you done a nat_map status
> once the ports are missing in pfsense to see if fs still thinks the ports
> should be open?  What if you do a nat_map republish?  Do the maps get pushed
> to pfsense and then stay open for a whlie?
>
> Perhaps pfsense is sending a keep-alive packet that we don't process right
> or is invalid? If so, I'd need a packet trace to do analysis.
>
> On Mon, Feb 8, 2010 at 10:21 PM, Troy Anderson <troy at tlainvestments.com>wrote:
>
>> I have been using pfSense (1.2.3) and FS.  FS nicely uses UPnP to poke
>> holes in the firewall, but it seems that the holes close after a while.  I
>> cannot find any documentation in FS nor in pfSense as to what the timeout
>> is.  Is there a setting in FS to do some kind of keep-alive thing with UPnP
>> to keep, e.g. 5060, open?  Or is it already doing that and pfSense is the
>> issue?
>>
>> Thanks!
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> -Rupa
>  _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100209/5b58daae/attachment.html 

From Prometheus001 at gmx.net  Tue Feb  9 08:07:59 2010
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Tue, 09 Feb 2010 17:07:59 +0100
Subject: [Freeswitch-users] Last call: buy the devs dinner!
In-Reply-To: <87f2f3b91002090735g1c6c69eby5274f6dfa9127fc5@mail.gmail.com>
References: <87f2f3b91002090735g1c6c69eby5274f6dfa9127fc5@mail.gmail.com>
Message-ID: <4B71885F.5090908@gmx.net>

Hello Michael,

just hit the paypal button. Enjoy your dinner! I think it's not just
dinner, it will be also 50% work I think (discussing about issues and
new features etc.) which brings additional benefits to the copmmunity.
Thanks for the great work you all have done so far.

Best regards
Peter

Michael Collins schrieb:
> Hey all,
>
> Thanks so much for the donations that have come in already! We
> appreciate your generosity. The dev team really wants to release 1.0.5
> but they're kinda hungry! :) Please hit the PayPal button on the main
> freeswitch.org <http://freeswitch.org> page to drop a few dollars in
> the hat. Also, keep in mind that we have the "extended family" of
> developers all here so it's not just Tony, Mike, and Brian. Let's all
> pitch in and have a great dinner for them.
>
> Thanks!
> -Michael
> ------------------------------------------------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   


From yehavi.bourvine at gmail.com  Tue Feb  9 08:17:57 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Tue, 9 Feb 2010 18:17:57 +0200
Subject: [Freeswitch-users] Announcement: FreeSWITCH Add Broadsoft SCA
	Support
In-Reply-To: <016d01caa98e$f6df25f0$e49d71d0$@com>
References: <87f2f3b91001111742k691966cfs82515db77c13d4d4@mail.gmail.com>
	<d361ca7b1002082110h4e3c60d0p88412f753444ec3d@mail.gmail.com>
	<016d01caa98e$f6df25f0$e49d71d0$@com>
Message-ID: <d361ca7b1002090817h4aa3c13dka4d4db8b4adc7606@mail.gmail.com>

Thanks!
I've managed to make it work today with Polcyoms. It works inside a profile
but does not across profiles. In our case this is not a limitation.

                           Thanks! __Yehavi:

 2010/2/9 Peder <peder at networkoblivion.com>

>   It is in addition to the existing settings.  It is for SCA presence on
> shared lines.  The ?manage presence? setting is for regular registrations.
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Yehavi
> Bourvine
> *Sent:* Monday, February 08, 2010 11:10 PM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Announcement: FreeSWITCH Add Broadsoft
> SCA Support
>
>
>
> Hello,
>
>
>
>   We currently use the "old" type of presence which is activated by
> "manage-presence" coupled with "dbname" and "presence-hosts".
>
> With the new method, does "manage-shared-presence" replace all of the above
> or comes in addition?
>
>
>
>                        Thanks! __Yehavi:
>
> 2010/1/12 Michael Collins <msc at freeswitch.org>
>
>  We want to let everyone know that FreeSWITCH now supports the Broadsoft
> SCA method of doing shared lines. The story is here:
>
> http://www.freeswitch.org/node/227
>
> Tony and Brian spent many hours laboring over this, so please be sure to
> show your appreciation to them for this new feature and all of the great
> things they do for the FreeSWITCH community and VoIP in general!
>
> -Michael
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100209/dfb636c2/attachment.html 

From brian at freeswitch.org  Tue Feb  9 08:21:41 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 9 Feb 2010 10:21:41 -0600
Subject: [Freeswitch-users] Announcement: FreeSWITCH Add Broadsoft SCA
	Support
In-Reply-To: <d361ca7b1002090817h4aa3c13dka4d4db8b4adc7606@mail.gmail.com>
References: <87f2f3b91001111742k691966cfs82515db77c13d4d4@mail.gmail.com>
	<d361ca7b1002082110h4e3c60d0p88412f753444ec3d@mail.gmail.com>
	<016d01caa98e$f6df25f0$e49d71d0$@com>
	<d361ca7b1002090817h4aa3c13dka4d4db8b4adc7606@mail.gmail.com>
Message-ID: <1C0A1BEE-3480-4A33-916F-52067AF5B9C9@freeswitch.org>

It will work across profiles if you bond them.  :P

/b

On Feb 9, 2010, at 10:17 AM, Yehavi Bourvine wrote:

> Thanks!
> I've managed to make it work today with Polcyoms. It works inside a profile but does not across profiles. In our case this is not a limitation.
>  
>                            Thanks! __Yehavi:



From yehavi.bourvine at gmail.com  Tue Feb  9 08:38:53 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Tue, 9 Feb 2010 18:38:53 +0200
Subject: [Freeswitch-users] Announcement: FreeSWITCH Add Broadsoft SCA
	Support
In-Reply-To: <1C0A1BEE-3480-4A33-916F-52067AF5B9C9@freeswitch.org>
References: <87f2f3b91001111742k691966cfs82515db77c13d4d4@mail.gmail.com>
	<d361ca7b1002082110h4e3c60d0p88412f753444ec3d@mail.gmail.com>
	<016d01caa98e$f6df25f0$e49d71d0$@com>
	<d361ca7b1002090817h4aa3c13dka4d4db8b4adc7606@mail.gmail.com>
	<1C0A1BEE-3480-4A33-916F-52067AF5B9C9@freeswitch.org>
Message-ID: <d361ca7b1002090838j6c42dcd2of8affb5f281900fd@mail.gmail.com>

What do you mean by "bonding them"?

             Thanks! __Yehavi:

2010/2/9 Brian West <brian at freeswitch.org>

> It will work across profiles if you bond them.  :P
>
> /b
>
> On Feb 9, 2010, at 10:17 AM, Yehavi Bourvine wrote:
>
> > Thanks!
> > I've managed to make it work today with Polcyoms. It works inside a
> profile but does not across profiles. In our case this is not a limitation.
> >
> >                            Thanks! __Yehavi:
>
>
>  _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100209/48f4b4b2/attachment.html 

From carlos.talbot at gmail.com  Tue Feb  9 08:51:02 2010
From: carlos.talbot at gmail.com (Carlos Talbot)
Date: Tue, 9 Feb 2010 10:51:02 -0600
Subject: [Freeswitch-users] SIP over TCP with Sipdroid,
	an Android SIP client
In-Reply-To: <5800526b1002051346g3890f152o1d939faa054811e6@mail.gmail.com>
References: <5800526b1002051346g3890f152o1d939faa054811e6@mail.gmail.com>
Message-ID: <5800526b1002090851x578310ddy205fa5a373b3d8be@mail.gmail.com>

FYI,

for those interested this issue has been identified and fixed with a simple
patch:

http://code.google.com/p/sipdroid/issues/detail?id=311#c2
<http://code.google.com/p/sipdroid/issues/detail?id=311#c2>

On Fri, Feb 5, 2010 at 3:46 PM, Carlos Talbot <carlos.talbot at gmail.com>wrote:

>
> Anyone use sipdroid on their Andorid phone? For the most part it works with
> the exception of when using SIP over TCP. For some reason, after 30 seconds
> into a call FreeSWITCH sends a bye and drops the call. Why use TCP? The
> author claims significantly increased standby times using SIP TCP over 3g:
> http://code.google.com/p/sipdroid/wiki/NewStandbyTechnique
>
> According to Brian it might be because the phone is not setting a transport
> in the contact field and FS is falling back to UDP.
>
> This is on r16557. Here's a sip trace along with call graph:
>
> http://pastebin.freeswitch.org/12064
>
> regards,
>
> Carlos
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100209/73f5464d/attachment-0001.html 

From matt at webcontracts.co.uk  Tue Feb  9 08:59:39 2010
From: matt at webcontracts.co.uk (Matthew Law)
Date: Tue, 9 Feb 2010 16:59:39 -0000
Subject: [Freeswitch-users] ACL question and js error
In-Reply-To: <87f2f3b91002090745v2128714byf1f7574d75f4449c@mail.gmail.com>
References: <ea05d45dca9be2d979fb386f960bc077.squirrel@www.webcontracts.co.uk>
	<20100207145907.GF31942@base.carmickle.com>
	<cfe871e3305675257b8c74ed4fb5a11b.squirrel@www.webcontracts.co.uk>
	<87f2f3b91002090745v2128714byf1f7574d75f4449c@mail.gmail.com>
Message-ID: <1a2b9e340124c8ef35c7fa5991bf3a5f.squirrel@www.webcontracts.co.uk>

On Tue, February 9, 2010 3:45 pm, Michael Collins wrote:
> Are you trying to let calls in from voiptalk.org? Do you want to auth all
> inbound calls or do you just want to blanket allow them and handle them in
> the dialplan? If you just want to allow calls in from the voiptalk.org IP
> address then you need to use the cidr tag in acl.conf.xml:
> <list name="domains" default="deny">
>     <node type="allow" domain="$${domain}"/>
>     <node type="allow" cidr="1.2.3.4/32"/>
>  </list>

AFAIK, they don't publish their IP addresses (I see incoming calls from
lots of different IPs in various subnets).  So at the moment I just want
to allow all and filter in the dialplan (which I think I am doing now).  I
just need a config that will survive a reboot and changing acl.conf.xml
doesn't at the moment.


Thanks,

Matt.



From brian at freeswitch.org  Tue Feb  9 09:02:30 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 9 Feb 2010 11:02:30 -0600
Subject: [Freeswitch-users] Announcement: FreeSWITCH Add Broadsoft SCA
	Support
In-Reply-To: <d361ca7b1002090838j6c42dcd2of8affb5f281900fd@mail.gmail.com>
References: <87f2f3b91001111742k691966cfs82515db77c13d4d4@mail.gmail.com>
	<d361ca7b1002082110h4e3c60d0p88412f753444ec3d@mail.gmail.com>
	<016d01caa98e$f6df25f0$e49d71d0$@com>
	<d361ca7b1002090817h4aa3c13dka4d4db8b4adc7606@mail.gmail.com>
	<1C0A1BEE-3480-4A33-916F-52067AF5B9C9@freeswitch.org>
	<d361ca7b1002090838j6c42dcd2of8affb5f281900fd@mail.gmail.com>
Message-ID: <767B677E-0D19-43A9-89DE-D96EC4A715EC@freeswitch.org>

    <!--<param name="dbname" value="share_presence"/>-->
    <!--<param name="presence-hosts" value="$${domain}"/>-->


/b

On Feb 9, 2010, at 10:38 AM, Yehavi Bourvine wrote:

> What do you mean by "bonding them"?
>  
>              Thanks! __Yehavi:
> 
> 2010/2/9 Brian West <brian at freeswitch.org>
> It will work across profiles if you bond them.  :P
> 
> /b
> 
> On Feb 9, 2010, at 10:17 AM, Yehavi Bourvine wrote:
> 
> > Thanks!
> > I've managed to make it work today with Polcyoms. It works inside a profile but does not across profiles. In our case this is not a limitation.
> >
> >                            Thanks! __Yehavi:
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100209/4432324d/attachment.html 

From m.sobkow at marketelsystems.com  Tue Feb  9 10:46:54 2010
From: m.sobkow at marketelsystems.com (Mark Sobkow)
Date: Tue, 09 Feb 2010 12:46:54 -0600
Subject: [Freeswitch-users] Having trouble establishing a call
Message-ID: <4B71AD9E.7090606@marketelsystems.com>

We're using Erlang to serve up the configurations to Freeswitch.  I've 
got things configured such that I can place a call from a SIP phone 
registered to extension 5000 to our "external" SIP provider (our 
Asterisk installation), but I can't place a call to extension 5001 from 
5000.  Below is the trace log Freeswitch produces when I attempt to do so.

Any suggestions as to what I should be looking at?  The directory seems 
to be getting served up correctly, as it provides the passwords both SIP 
softphones are using to register with Freeswitch.  I'd have thought that 
once they've registered with FS, the extension would automatically be 
recognized when an incoming call is placed or bridged, but such does not 
seem to be the case.



2010-02-09 12:43:10.394647 [NOTICE] switch_channel.c:660 New Channel 
sofia/external/5000 at testsrv.marketel [9854f2ea-cf04-4a48-a560-467cbd86bb1d]
2010-02-09 12:43:10.394647 [DEBUG] switch_core_state_machine.c:316 
(sofia/external/5000 at testsrv.marketel) Running State Change CS_NEW
2010-02-09 12:43:10.394647 [DEBUG] switch_core_state_machine.c:322 
(sofia/external/5000 at testsrv.marketel) State NEW
2010-02-09 12:43:10.394647 [DEBUG] sofia.c:4019 Channel 
sofia/external/5000 at testsrv.marketel entering state [received][100]
2010-02-09 12:43:10.394647 [DEBUG] sofia.c:4030 Remote SDP:
v=0
o=- 4197664938 0 IN IP4 10.77.0.126
s=SIPPER for phoner
c=IN IP4 10.77.0.126
t=0 0
m=audio 5062 RTP/AVP 8 0 2 3 97 9 111 112 113 114 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:9 G722/8000
a=rtpmap:111 speex/16000
a=rtpmap:112 G726-16/8000
a=rtpmap:113 G726-24/8000
a=rtpmap:114 G726-40/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

2010-02-09 12:43:10.394647 [DEBUG] sofia_glue.c:3388 Audio Codec Compare 
[PCMA:8:8000:20]/[PCMU:0:8000:20]
2010-02-09 12:43:10.394647 [DEBUG] sofia_glue.c:3388 Audio Codec Compare 
[PCMA:8:8000:20]/[PCMA:8:8000:20]
2010-02-09 12:43:10.394647 [DEBUG] sofia_glue.c:2211 Set Codec 
sofia/external/5000 at testsrv.marketel PCMA/8000 20 ms 160 samples
2010-02-09 12:43:10.394647 [DEBUG] sofia_glue.c:3344 Set 2833 dtmf 
payload to 101
2010-02-09 12:43:10.394647 [DEBUG] sofia.c:4178 
(sofia/external/5000 at testsrv.marketel) State Change CS_NEW -> CS_INIT
2010-02-09 12:43:10.394647 [DEBUG] switch_core_session.c:1019 Send 
signal sofia/external/5000 at testsrv.marketel [BREAK]
2010-02-09 12:43:10.394647 [DEBUG] switch_core_state_machine.c:316 
(sofia/external/5000 at testsrv.marketel) Running State Change CS_INIT
2010-02-09 12:43:10.394647 [DEBUG] switch_core_state_machine.c:340 
(sofia/external/5000 at testsrv.marketel) State INIT
2010-02-09 12:43:10.394647 [DEBUG] mod_sofia.c:83 
sofia/external/5000 at testsrv.marketel SOFIA INIT
2010-02-09 12:43:10.394647 [DEBUG] mod_sofia.c:111 
(sofia/external/5000 at testsrv.marketel) State Change CS_INIT -> CS_ROUTING
2010-02-09 12:43:10.394647 [DEBUG] switch_core_session.c:1019 Send 
signal sofia/external/5000 at testsrv.marketel [BREAK]
2010-02-09 12:43:10.394647 [DEBUG] switch_core_state_machine.c:340 
(sofia/external/5000 at testsrv.marketel) State INIT going to sleep
2010-02-09 12:43:10.394647 [DEBUG] switch_core_state_machine.c:316 
(sofia/external/5000 at testsrv.marketel) Running State Change CS_ROUTING
2010-02-09 12:43:10.394647 [DEBUG] switch_core_state_machine.c:343 
(sofia/external/5000 at testsrv.marketel) State ROUTING
2010-02-09 12:43:10.394647 [DEBUG] mod_sofia.c:132 
sofia/external/5000 at testsrv.marketel SOFIA ROUTING
2010-02-09 12:43:10.394647 [DEBUG] switch_core_state_machine.c:78 
sofia/external/5000 at testsrv.marketel Standard ROUTING
2010-02-09 12:43:10.394647 [INFO] mod_dialplan_xml.c:408 Processing MSS 
Testing->5001 in context public
2010-02-09 12:43:10.394647 [DEBUG] mod_erlang_event.c:387 looking for 
bindings
2010-02-09 12:43:10.394647 [DEBUG] mod_erlang_event.c:403 binding for 
(null) in section dialplan with key (null) and value (null) requested 
from node pursuit at testsrv
2010-02-09 12:43:10.406527 [DEBUG] handle_msg.c:191 Found waiting slot 
for 7077ba5c-0aae-44fd-87fd-e7aa48f62659
2010-02-09 12:43:10.406527 [DEBUG] mod_erlang_event.c:456 got data 
<document type="freeswitch/xml">
  <section name="dialplan" description="Dial Plan For FreeSwitch">
    <context name="public">
      <extension name="Operator">
        <condition field="destination_number" expression="^(0)$">
          <action application="bridge" 
data="sofia/external/160 at rats.marketel"/>
        </condition>
      </extension>
      <extension name="InternalFS">
        <condition field="destination_number" expression="^(\d\d\d\d)$">
          <action application="bridge" 
data="sofia/external/$0 at testsrv.marketel"/>
        </condition>
      </extension>
      <extension name="RatsExtension">
        <condition field="destination_number" expression="^(\d\d\d)$">
          <action application="bridge" 
data="sofia/external/$0 at rats.marketel"/>
        </condition>
      </extension>
      <extension name="LocalCall">
        <condition field="destination_number" 
expression="^(9\d\d\d\d\d\d\d)$">
          <action application="bridge" 
data="sofia/external/$0 at rats.marketel"/>
        </condition>
      </extension>
      <extension name="LongDistanceCall">
        <condition field="destination_number" 
expression="^91(\d\d\d\d\d\d\d\d\d\d)$">
          <action application="bridge" 
data="sofia/external/$0 at rats.marketel"/>
        </condition>
      </extension>
      <extension name="DialerConference">
        <condition field="destination_number" expression="^8(\d\d\d)$">
          <action application="conference" data="conf$0 at dialer"/>
        </condition>
      </extension>
    </context>
    <context name="default">
      <extension name="InternalFS">
        <condition field="destination_number" expression="^(\d\d\d\d)$">
          <action application="bridge" 
data="sofia/external/$0 at testsrv.marketel"/>
        </condition>
      </extension>
      <extension name="RatsExtension">
        <condition field="destination_number" expression="^(\d\d\d)$">
          <action application="bridge" 
data="sofia/external/$0 at rats.marketel"/>
        </condition>
      </extension>
      <extension name="LocalCall">
        <condition field="destination_number" 
expression="^(9\d\d\d\d\d\d\d)$">
          <action application="bridge" 
data="sofia/external/$0 at rats.marketel"/>
        </condition>
      </extension>
      <extension name="LongDistanceCall">
        <condition field="destination_number" 
expression="^91(\d\d\d\d\d\d\d\d\d\d)$">
          <action application="bridge" 
data="sofia/external/$0 at rats.marketel"/>
        </condition>
      </extension>
      <extension name="DialerConference">
        <condition field="destination_number" expression="^8(\d\d\d)$">
          <action application="conference" data="conf$0 at dialer"/>
        </condition>
      </extension>
    </context>
  </section>
</document> after 10 milliseconds!
2010-02-09 12:43:10.406527 [DEBUG] mod_erlang_event.c:463 XML parsed OK!
Dialplan: sofia/external/5000 at testsrv.marketel parsing 
[public->Operator] continue=false
Dialplan: sofia/external/5000 at testsrv.marketel Regex (FAIL) [Operator] 
destination_number(5001) =~ /^(0)$/ break=on-false
Dialplan: sofia/external/5000 at testsrv.marketel parsing 
[public->InternalFS] continue=false
Dialplan: sofia/external/5000 at testsrv.marketel Regex (PASS) [InternalFS] 
destination_number(5001) =~ /^(\d\d\d\d)$/ break=on-false
Dialplan: sofia/external/5000 at testsrv.marketel Action 
bridge(sofia/external/5001 at testsrv.marketel)
2010-02-09 12:43:10.406527 [DEBUG] switch_core_state_machine.c:122 
(sofia/external/5000 at testsrv.marketel) State Change CS_ROUTING -> CS_EXECUTE
2010-02-09 12:43:10.406527 [DEBUG] switch_core_session.c:1019 Send 
signal sofia/external/5000 at testsrv.marketel [BREAK]
2010-02-09 12:43:10.406527 [DEBUG] switch_core_state_machine.c:343 
(sofia/external/5000 at testsrv.marketel) State ROUTING going to sleep
2010-02-09 12:43:10.406527 [DEBUG] switch_core_state_machine.c:316 
(sofia/external/5000 at testsrv.marketel) Running State Change CS_EXECUTE
2010-02-09 12:43:10.406527 [DEBUG] switch_core_state_machine.c:350 
(sofia/external/5000 at testsrv.marketel) State EXECUTE
2010-02-09 12:43:10.406527 [DEBUG] mod_sofia.c:181 
sofia/external/5000 at testsrv.marketel SOFIA EXECUTE
2010-02-09 12:43:10.406527 [DEBUG] switch_core_state_machine.c:159 
sofia/external/5000 at testsrv.marketel Standard EXECUTE
EXECUTE sofia/external/5000 at testsrv.marketel 
bridge(sofia/external/5001 at testsrv.marketel)
2010-02-09 12:43:10.406527 [NOTICE] switch_channel.c:660 New Channel 
sofia/external/5001 at testsrv.marketel [813c2e55-8acf-486d-b3b6-3244edf5529a]
2010-02-09 12:43:10.406527 [DEBUG] mod_sofia.c:3317 
(sofia/external/5001 at testsrv.marketel) State Change CS_NEW -> CS_INIT
2010-02-09 12:43:10.406527 [DEBUG] switch_core_session.c:1019 Send 
signal sofia/external/5001 at testsrv.marketel [BREAK]
2010-02-09 12:43:10.414137 [DEBUG] switch_core_state_machine.c:316 
(sofia/external/5001 at testsrv.marketel) Running State Change CS_INIT
2010-02-09 12:43:10.414137 [DEBUG] switch_core_state_machine.c:340 
(sofia/external/5001 at testsrv.marketel) State INIT
2010-02-09 12:43:10.414137 [DEBUG] mod_sofia.c:83 
sofia/external/5001 at testsrv.marketel SOFIA INIT
2010-02-09 12:43:10.414137 [DEBUG] mod_sofia.c:111 
(sofia/external/5001 at testsrv.marketel) State Change CS_INIT -> CS_ROUTING
2010-02-09 12:43:10.414137 [DEBUG] switch_core_session.c:1019 Send 
signal sofia/external/5001 at testsrv.marketel [BREAK]
2010-02-09 12:43:10.414137 [DEBUG] switch_core_state_machine.c:340 
(sofia/external/5001 at testsrv.marketel) State INIT going to sleep
2010-02-09 12:43:10.414137 [DEBUG] switch_core_state_machine.c:316 
(sofia/external/5001 at testsrv.marketel) Running State Change CS_ROUTING
2010-02-09 12:43:10.414137 [DEBUG] switch_core_state_machine.c:343 
(sofia/external/5001 at testsrv.marketel) State ROUTING
2010-02-09 12:43:10.414137 [DEBUG] mod_sofia.c:132 
sofia/external/5001 at testsrv.marketel SOFIA ROUTING
2010-02-09 12:43:10.414137 [DEBUG] switch_ivr_originate.c:66 
(sofia/external/5001 at testsrv.marketel) State Change CS_ROUTING -> 
CS_CONSUME_MEDIA
2010-02-09 12:43:10.414137 [DEBUG] switch_core_session.c:1019 Send 
signal sofia/external/5001 at testsrv.marketel [BREAK]
2010-02-09 12:43:10.414137 [DEBUG] switch_core_state_machine.c:343 
(sofia/external/5001 at testsrv.marketel) State ROUTING going to sleep
2010-02-09 12:43:10.414137 [DEBUG] switch_core_state_machine.c:316 
(sofia/external/5001 at testsrv.marketel) Running State Change CS_CONSUME_MEDIA
2010-02-09 12:43:10.414137 [DEBUG] switch_core_state_machine.c:362 
(sofia/external/5001 at testsrv.marketel) State CONSUME_MEDIA
2010-02-09 12:43:10.414137 [DEBUG] switch_core_state_machine.c:362 
(sofia/external/5001 at testsrv.marketel) State CONSUME_MEDIA going to sleep
2010-02-09 12:43:10.414137 [DEBUG] sofia.c:4019 Channel 
sofia/external/5001 at testsrv.marketel entering state [calling][0]
2010-02-09 12:43:10.414137 [DEBUG] sofia.c:4019 Channel 
sofia/external/5001 at testsrv.marketel entering state [terminated][503]
2010-02-09 12:43:10.414137 [NOTICE] sofia.c:4663 Hangup 
sofia/external/5001 at testsrv.marketel [CS_CONSUME_MEDIA] 
[NORMAL_TEMPORARY_FAILURE]
2010-02-09 12:43:10.414137 [DEBUG] switch_ivr_originate.c:3209 Originate 
Resulted in Error Cause: 41 [NORMAL_TEMPORARY_FAILURE]
2010-02-09 12:43:10.414137 [DEBUG] switch_channel.c:1994 Send signal 
sofia/external/5001 at testsrv.marketel [KILL]
2010-02-09 12:43:10.414137 [DEBUG] switch_core_state_machine.c:316 
(sofia/external/5001 at testsrv.marketel) Running State Change CS_HANGUP
2010-02-09 12:43:10.414137 [INFO] mod_dptools.c:2346 Originate Failed.  
Cause: NORMAL_TEMPORARY_FAILURE
2010-02-09 12:43:10.414137 [NOTICE] mod_dptools.c:2409 Hangup 
sofia/external/5000 at testsrv.marketel [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE]
2010-02-09 12:43:10.414137 [DEBUG] switch_channel.c:1994 Send signal 
sofia/external/5000 at testsrv.marketel [KILL]
2010-02-09 12:43:10.414137 [DEBUG] switch_core_session.c:1019 Send 
signal sofia/external/5000 at testsrv.marketel [BREAK]
2010-02-09 12:43:10.414137 [DEBUG] switch_core_state_machine.c:350 
(sofia/external/5000 at testsrv.marketel) State EXECUTE going to sleep
2010-02-09 12:43:10.414137 [DEBUG] switch_core_state_machine.c:496 
(sofia/external/5001 at testsrv.marketel) State HANGUP
2010-02-09 12:43:10.414137 [DEBUG] switch_core_state_machine.c:316 
(sofia/external/5000 at testsrv.marketel) Running State Change CS_HANGUP
2010-02-09 12:43:10.414137 [DEBUG] switch_core_session.c:1019 Send 
signal sofia/external/5001 at testsrv.marketel [BREAK]
2010-02-09 12:43:10.414137 [DEBUG] mod_sofia.c:352 
sofia/external/5001 at testsrv.marketel Overriding SIP cause 503 with 503 
from the other leg
2010-02-09 12:43:10.414137 [DEBUG] mod_sofia.c:358 Channel 
sofia/external/5001 at testsrv.marketel hanging up, cause: 
NORMAL_TEMPORARY_FAILURE
2010-02-09 12:43:10.414137 [DEBUG] switch_core_state_machine.c:46 
sofia/external/5001 at testsrv.marketel Standard HANGUP, cause: 
NORMAL_TEMPORARY_FAILURE
2010-02-09 12:43:10.414137 [DEBUG] switch_core_state_machine.c:496 
(sofia/external/5001 at testsrv.marketel) State HANGUP going to sleep
2010-02-09 12:43:10.414137 [DEBUG] switch_core_state_machine.c:335 
(sofia/external/5001 at testsrv.marketel) State Change CS_HANGUP -> 
CS_REPORTING
2010-02-09 12:43:10.414137 [DEBUG] switch_core_session.c:1019 Send 
signal sofia/external/5001 at testsrv.marketel [BREAK]
2010-02-09 12:43:10.414137 [DEBUG] switch_core_state_machine.c:316 
(sofia/external/5001 at testsrv.marketel) Running State Change CS_REPORTING
2010-02-09 12:43:10.414137 [DEBUG] switch_core_state_machine.c:587 
(sofia/external/5001 at testsrv.marketel) State REPORTING
2010-02-09 12:43:10.414137 [DEBUG] switch_core_state_machine.c:53 
sofia/external/5001 at testsrv.marketel Standard REPORTING, cause: 
NORMAL_TEMPORARY_FAILURE
2010-02-09 12:43:10.414137 [DEBUG] switch_core_state_machine.c:587 
(sofia/external/5001 at testsrv.marketel) State REPORTING going to sleep
2010-02-09 12:43:10.414137 [DEBUG] switch_core_state_machine.c:329 
(sofia/external/5001 at testsrv.marketel) State Change CS_REPORTING -> 
CS_DESTROY
2010-02-09 12:43:10.414137 [DEBUG] switch_core_session.c:1019 Send 
signal sofia/external/5001 at testsrv.marketel [BREAK]
2010-02-09 12:43:10.414137 [DEBUG] switch_core_session.c:1161 Session 18 
(sofia/external/5001 at testsrv.marketel) Locked, Waiting on external entities
2010-02-09 12:43:10.414137 [NOTICE] switch_core_session.c:1179 Session 
18 (sofia/external/5001 at testsrv.marketel) Ended
2010-02-09 12:43:10.414137 [NOTICE] switch_core_session.c:1181 Close 
Channel sofia/external/5001 at testsrv.marketel [CS_DESTROY]
2010-02-09 12:43:10.414137 [DEBUG] switch_core_state_machine.c:425 
(sofia/external/5001 at testsrv.marketel) Running State Change CS_DESTROY
2010-02-09 12:43:10.414137 [DEBUG] switch_core_state_machine.c:436 
(sofia/external/5001 at testsrv.marketel) State DESTROY
2010-02-09 12:43:10.414137 [DEBUG] mod_sofia.c:293 
sofia/external/5001 at testsrv.marketel SOFIA DESTROY
2010-02-09 12:43:10.414137 [DEBUG] switch_core_state_machine.c:60 
sofia/external/5001 at testsrv.marketel Standard DESTROY
2010-02-09 12:43:10.414137 [DEBUG] switch_core_state_machine.c:436 
(sofia/external/5001 at testsrv.marketel) State DESTROY going to sleep
2010-02-09 12:43:10.414137 [DEBUG] switch_core_state_machine.c:496 
(sofia/external/5000 at testsrv.marketel) State HANGUP
2010-02-09 12:43:10.414137 [DEBUG] mod_sofia.c:352 
sofia/external/5000 at testsrv.marketel Overriding SIP cause 503 with 503 
from the other leg
2010-02-09 12:43:10.414137 [DEBUG] mod_sofia.c:358 Channel 
sofia/external/5000 at testsrv.marketel hanging up, cause: 
NORMAL_TEMPORARY_FAILURE
2010-02-09 12:43:10.414137 [DEBUG] mod_sofia.c:424 Responding to INVITE 
with: 503
2010-02-09 12:43:10.414137 [DEBUG] switch_core_state_machine.c:46 
sofia/external/5000 at testsrv.marketel Standard HANGUP, cause: 
NORMAL_TEMPORARY_FAILURE
2010-02-09 12:43:10.414137 [DEBUG] switch_core_state_machine.c:496 
(sofia/external/5000 at testsrv.marketel) State HANGUP going to sleep
2010-02-09 12:43:10.414137 [DEBUG] switch_core_state_machine.c:335 
(sofia/external/5000 at testsrv.marketel) State Change CS_HANGUP -> 
CS_REPORTING
2010-02-09 12:43:10.414137 [DEBUG] switch_core_session.c:1019 Send 
signal sofia/external/5000 at testsrv.marketel [BREAK]
2010-02-09 12:43:10.414137 [DEBUG] switch_core_state_machine.c:316 
(sofia/external/5000 at testsrv.marketel) Running State Change CS_REPORTING
2010-02-09 12:43:10.414137 [DEBUG] switch_core_state_machine.c:587 
(sofia/external/5000 at testsrv.marketel) State REPORTING
2010-02-09 12:43:10.414137 [DEBUG] switch_core_state_machine.c:53 
sofia/external/5000 at testsrv.marketel Standard REPORTING, cause: 
NORMAL_TEMPORARY_FAILURE
2010-02-09 12:43:10.414137 [DEBUG] switch_core_state_machine.c:587 
(sofia/external/5000 at testsrv.marketel) State REPORTING going to sleep
2010-02-09 12:43:10.414137 [DEBUG] switch_core_state_machine.c:329 
(sofia/external/5000 at testsrv.marketel) State Change CS_REPORTING -> 
CS_DESTROY
2010-02-09 12:43:10.414137 [DEBUG] switch_core_session.c:1019 Send 
signal sofia/external/5000 at testsrv.marketel [BREAK]
2010-02-09 12:43:10.414137 [DEBUG] switch_core_session.c:1161 Session 17 
(sofia/external/5000 at testsrv.marketel) Locked, Waiting on external entities
2010-02-09 12:43:10.414137 [NOTICE] switch_core_session.c:1179 Session 
17 (sofia/external/5000 at testsrv.marketel) Ended
2010-02-09 12:43:10.414137 [NOTICE] switch_core_session.c:1181 Close 
Channel sofia/external/5000 at testsrv.marketel [CS_DESTROY]
2010-02-09 12:43:10.414137 [DEBUG] switch_core_state_machine.c:425 
(sofia/external/5000 at testsrv.marketel) Running State Change CS_DESTROY
2010-02-09 12:43:10.414137 [DEBUG] switch_core_state_machine.c:436 
(sofia/external/5000 at testsrv.marketel) State DESTROY
2010-02-09 12:43:10.414137 [DEBUG] mod_sofia.c:293 
sofia/external/5000 at testsrv.marketel SOFIA DESTROY
2010-02-09 12:43:10.414137 [DEBUG] switch_core_state_machine.c:60 
sofia/external/5000 at testsrv.marketel Standard DESTROY
2010-02-09 12:43:10.414137 [DEBUG] switch_core_state_machine.c:436 
(sofia/external/5000 at testsrv.marketel) State DESTROY going to sleep


-- 
Mark Sobkow
Senior Developer
MarkeTel Multi-Line Dialing Systems LTD.
428 Victoria Ave
Regina, SK S4N-0P6
Toll-Free: 800-289-8616-X533
Local: 306-359-6893-X533
Fax: 306-359-6879
Email: m.sobkow at marketelsystems.com
Web: http://www.marketelsystems.com 



From jerry.richards at teotech.com  Tue Feb  9 11:15:26 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Tue, 9 Feb 2010 11:15:26 -0800
Subject: [Freeswitch-users] Deleting Single Registered Device in the FS
	Console
Message-ID: <326DBA7172E24B42A1B466E7BAC0B562@greyhawk.tonecommander.com>

How do I delete a single registered user at the FS Console?  I know how to
flush_inbound_reg (which is all user) using "sofia profile internal
flush_inbound_reg".

Best Regards,
Jerry



From jerry.richards at teotech.com  Tue Feb  9 11:19:50 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Tue, 9 Feb 2010 11:19:50 -0800
Subject: [Freeswitch-users] Deleting Single Registered Device in the FS
	Console
Message-ID: <7377B7F5DEF54DB8843C47A7EBC8888F@greyhawk.tonecommander.com>

Actually I found out how to do it using the Call-ID (which is very long),
but can't I do it using the user's extension number?

Best Regards,
Jerry


-----Original Message-----
From: Jerry Richards [mailto:jerry.richards at teotech.com] 
Sent: Tuesday, February 09, 2010 11:15 AM
To: 'freeswitch-users at lists.freeswitch.org'
Subject: Deleting Single Registered Device in the FS Console

How do I delete a single registered user at the FS Console?  I know how to
flush_inbound_reg (which is all user) using "sofia profile internal
flush_inbound_reg".

Best Regards,
Jerry



From brian at freeswitch.org  Tue Feb  9 11:21:12 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 9 Feb 2010 13:21:12 -0600
Subject: [Freeswitch-users] Deleting Single Registered Device in the FS
	Console
In-Reply-To: <326DBA7172E24B42A1B466E7BAC0B562@greyhawk.tonecommander.com>
References: <326DBA7172E24B42A1B466E7BAC0B562@greyhawk.tonecommander.com>
Message-ID: <B1554204-BDAA-44D4-AD40-3E4FD3CBA04F@freeswitch.org>

Read the sofia command... you can add the call-id of the registration to flush the single one.

/b

On Feb 9, 2010, at 1:15 PM, Jerry Richards wrote:

> How do I delete a single registered user at the FS Console?  I know how to
> flush_inbound_reg (which is all user) using "sofia profile internal
> flush_inbound_reg".
> 
> Best Regards,
> Jerry



From brian at freeswitch.org  Tue Feb  9 11:23:35 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 9 Feb 2010 13:23:35 -0600
Subject: [Freeswitch-users] Deleting Single Registered Device in the FS
	Console
In-Reply-To: <7377B7F5DEF54DB8843C47A7EBC8888F@greyhawk.tonecommander.com>
References: <7377B7F5DEF54DB8843C47A7EBC8888F@greyhawk.tonecommander.com>
Message-ID: <D2E44884-0DB3-4CE8-BAD0-68D88DC297D4@freeswitch.org>

what if they have 10 phones registered?  That kinda goes beyond a single registration record.

/b

On Feb 9, 2010, at 1:19 PM, Jerry Richards wrote:

> Actually I found out how to do it using the Call-ID (which is very long),
> but can't I do it using the user's extension number?
> 
> Best Regards,
> Jerry



From andrew at hijacked.us  Tue Feb  9 11:25:37 2010
From: andrew at hijacked.us (Andrew Thompson)
Date: Tue, 9 Feb 2010 14:25:37 -0500
Subject: [Freeswitch-users] Having trouble establishing a call
In-Reply-To: <4B71AD9E.7090606@marketelsystems.com>
References: <4B71AD9E.7090606@marketelsystems.com>
Message-ID: <20100209192537.GA24616@hijacked.us>

On Tue, Feb 09, 2010 at 12:46:54PM -0600, Mark Sobkow wrote:
> We're using Erlang to serve up the configurations to Freeswitch.  I've 
> got things configured such that I can place a call from a SIP phone 
> registered to extension 5000 to our "external" SIP provider (our 
> Asterisk installation), but I can't place a call to extension 5001 from 
> 5000.  Below is the trace log Freeswitch produces when I attempt to do so.
> 
> Any suggestions as to what I should be looking at?  The directory seems 
> to be getting served up correctly, as it provides the passwords both SIP 
> softphones are using to register with Freeswitch.  I'd have thought that 
> once they've registered with FS, the extension would automatically be 
> recognized when an incoming call is placed or bridged, but such does not 
> seem to be the case.
>

Looks like the other sip box (sofia/external/5001 at testsrv.marketel) is
returning the failure code. Check the config on the other side? Don't
you have a gateway setup for this other machine so you couls do
sofia/gateway/testsrv.marketel/5001 instead?

Andrew


From robert.hadley at teotech.com  Tue Feb  9 11:28:13 2010
From: robert.hadley at teotech.com (Robert Hadley)
Date: Tue, 9 Feb 2010 11:28:13 -0800
Subject: [Freeswitch-users] Where are new sounds_dir and recordings_dir
	variables set?
Message-ID: <28788FA27BE74B9FA6BFFE52BED11B60@greyhawk.tonecommander.com>

 

The XML conf files have been recently modified to replace
"$${base_dir}/sounds" with $${sounds_dir}.  The same replacement was done
for $${base_dir}/recordings and $${recordings_dir}.  In vars.xml, the
X-PRE-PROCESS cmd to setting the old variable sound_prefix was removed but
set commands were not added for the new variables sounds_dir and
recordings_dir.  How is FS finding determining the value of $${sounds_dir}
and $${recordings_dir} in the XML files?

 

I ask because I have 2 cloned FS servers where one can find sound files in
the IVR when an invalid extension is entered, but the other does not and is
using the wrong path to search for sound files.

 

Server 2 Fails:

2010-02-09 11:11:16.735745 [DEBUG] switch_ivr_menu.c:381 digits '9999'

2010-02-09 11:11:16.735745 [DEBUG] switch_ivr_menu.c:475 action regex [9999]
[/^((5[34][8901][0-9])||(10[0-5][0-9])||(30\d{2}))$/] [0]

2010-02-09 11:11:16.735745 [DEBUG] switch_ivr_menu.c:565 IVR menu 'teo_ivr'
caught invalid input '9999'

2010-02-09 11:11:16.735745 [ERR] mod_sndfile.c:194 Error Opening File
[/opt/teoswitch/sounds/ivr/ivr-that_was_an_invalid_entry.wav] [System error
: No such file or directory.]

 

Server 1 Works:

2010-02-09 11:17:25.085660 [DEBUG] switch_ivr_menu.c:378 digits '9999'

2010-02-09 11:17:25.085660 [DEBUG] switch_ivr_menu.c:472 action regex [9999]
[/^((5[34][8901][0-9])||(10[0-5][0-9])||(30\d{2}))$/] [0]

2010-02-09 11:17:25.085660 [DEBUG] switch_ivr_menu.c:562 IVR menu 'teo_ivr'
caught invalid input '9999'

2010-02-09 11:17:25.085660 [DEBUG] switch_ivr_play_say.c:1162 Codec
Activated L16 at 8000hz 1 channels 20ms

2010-02-09 11:17:25.085660 [DEBUG] switch_core_io.c:652
sofia/internal/1045 at 192.168.72.141:5060 receive message
[TRANSCODING_NECESSARY]

2010-02-09 11:17:26.765470 [DEBUG] switch_ivr_play_say.c:1454 done playing
file

 

I have compared the conf folders and they are nearly identical.  The only
difference I know of is Server 2 was originally compiled with mod_flite
enabled but it is not loaded at runtime.  Any suggestions why one server can
find sound files but the other looks in the wrong path?

 

Thanks,

Robert

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100209/a1c12974/attachment.html 

From brian at freeswitch.org  Tue Feb  9 11:33:53 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 9 Feb 2010 13:33:53 -0600
Subject: [Freeswitch-users] Where are new sounds_dir and recordings_dir
	variables set?
In-Reply-To: <28788FA27BE74B9FA6BFFE52BED11B60@greyhawk.tonecommander.com>
References: <28788FA27BE74B9FA6BFFE52BED11B60@greyhawk.tonecommander.com>
Message-ID: <6B8839F5-DA53-4565-A7E0-876FC53BD6F7@freeswitch.org>

did you configure it with --prefix=/opt/teoswitch or did you move it?

/b

On Feb 9, 2010, at 1:28 PM, Robert Hadley wrote:

>  
> The XML conf files have been recently modified to replace ?$${base_dir}/sounds? with $${sounds_dir}.  The same replacement was done for $${base_dir}/recordings and $${recordings_dir}.  In vars.xml, the X-PRE-PROCESS cmd to setting the old variable sound_prefix was removed but set commands were not added for the new variables sounds_dir and recordings_dir.  How is FS finding determining the value of $${sounds_dir} and $${recordings_dir} in the XML files?
>  
> I ask because I have 2 cloned FS servers where one can find sound files in the IVR when an invalid extension is entered, but the other does not and is using the wrong path to search for sound files.
>  
> Server 2 Fails:
> 2010-02-09 11:11:16.735745 [DEBUG] switch_ivr_menu.c:381 digits '9999'
> 2010-02-09 11:11:16.735745 [DEBUG] switch_ivr_menu.c:475 action regex [9999] [/^((5[34][8901][0-9])||(10[0-5][0-9])||(30\d{2}))$/] [0]
> 2010-02-09 11:11:16.735745 [DEBUG] switch_ivr_menu.c:565 IVR menu 'teo_ivr' caught invalid input '9999'
> 2010-02-09 11:11:16.735745 [ERR] mod_sndfile.c:194 Error Opening File [/opt/teoswitch/sounds/ivr/ivr-that_was_an_invalid_entry.wav] [System error : No such file or directory.]
>  
> Server 1 Works:
> 2010-02-09 11:17:25.085660 [DEBUG] switch_ivr_menu.c:378 digits '9999'
> 2010-02-09 11:17:25.085660 [DEBUG] switch_ivr_menu.c:472 action regex [9999] [/^((5[34][8901][0-9])||(10[0-5][0-9])||(30\d{2}))$/] [0]
> 2010-02-09 11:17:25.085660 [DEBUG] switch_ivr_menu.c:562 IVR menu 'teo_ivr' caught invalid input '9999'
> 2010-02-09 11:17:25.085660 [DEBUG] switch_ivr_play_say.c:1162 Codec Activated L16 at 8000hz 1 channels 20ms
> 2010-02-09 11:17:25.085660 [DEBUG] switch_core_io.c:652 sofia/internal/1045 at 192.168.72.141:5060 receive message [TRANSCODING_NECESSARY]
> 2010-02-09 11:17:26.765470 [DEBUG] switch_ivr_play_say.c:1454 done playing file
>  
> I have compared the conf folders and they are nearly identical.  The only difference I know of is Server 2 was originally compiled with mod_flite enabled but it is not loaded at runtime.  Any suggestions why one server can find sound files but the other looks in the wrong path?
>  
> Thanks,
> Robert



From m.sobkow at marketelsystems.com  Tue Feb  9 11:54:04 2010
From: m.sobkow at marketelsystems.com (Mark Sobkow)
Date: Tue, 09 Feb 2010 13:54:04 -0600
Subject: [Freeswitch-users] Having trouble establishing a call
In-Reply-To: <20100209192537.GA24616@hijacked.us>
References: <4B71AD9E.7090606@marketelsystems.com>
	<20100209192537.GA24616@hijacked.us>
Message-ID: <4B71BD5C.1080201@marketelsystems.com>

Andrew Thompson wrote:
> On Tue, Feb 09, 2010 at 12:46:54PM -0600, Mark Sobkow wrote:
>   
>> We're using Erlang to serve up the configurations to Freeswitch.  I've 
>> got things configured such that I can place a call from a SIP phone 
>> registered to extension 5000 to our "external" SIP provider (our 
>> Asterisk installation), but I can't place a call to extension 5001 from 
>> 5000.  Below is the trace log Freeswitch produces when I attempt to do so.
>>
>> Any suggestions as to what I should be looking at?  The directory seems 
>> to be getting served up correctly, as it provides the passwords both SIP 
>> softphones are using to register with Freeswitch.  I'd have thought that 
>> once they've registered with FS, the extension would automatically be 
>> recognized when an incoming call is placed or bridged, but such does not 
>> seem to be the case.
>>
>>     
>
> Looks like the other sip box (sofia/external/5001 at testsrv.marketel) is
> returning the failure code. Check the config on the other side? Don't
> you have a gateway setup for this other machine so you couls do
> sofia/gateway/testsrv.marketel/5001 instead?
>
> Andrew
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>   

Actually rats.marketel is our Asterisk box, which acts as our SIP 
trunk.  testsrv.marketel is the one running Freeswitch.

Am I maybe using an incorrect syntax in the dialplan for specifying that 
the call should be routed to a local extension on the Freeswitch box?  I 
thought the sofia/<profile>/<number>@<server> syntax is supposed to be 
used for placing any SIP calls, not just "remote" ones.


From m.sobkow at marketelsystems.com  Tue Feb  9 12:26:09 2010
From: m.sobkow at marketelsystems.com (Mark Sobkow)
Date: Tue, 09 Feb 2010 14:26:09 -0600
Subject: [Freeswitch-users] Having trouble establishing a call
In-Reply-To: <20100209192537.GA24616@hijacked.us>
References: <4B71AD9E.7090606@marketelsystems.com>
	<20100209192537.GA24616@hijacked.us>
Message-ID: <4B71C4E1.705@marketelsystems.com>

Andrew Thompson wrote:
> On Tue, Feb 09, 2010 at 12:46:54PM -0600, Mark Sobkow wrote:
>   
>> We're using Erlang to serve up the configurations to Freeswitch.  I've 
>> got things configured such that I can place a call from a SIP phone 
>> registered to extension 5000 to our "external" SIP provider (our 
>> Asterisk installation), but I can't place a call to extension 5001 from 
>> 5000.  Below is the trace log Freeswitch produces when I attempt to do so.
>>
>> Any suggestions as to what I should be looking at?  The directory seems 
>> to be getting served up correctly, as it provides the passwords both SIP 
>> softphones are using to register with Freeswitch.  I'd have thought that 
>> once they've registered with FS, the extension would automatically be 
>> recognized when an incoming call is placed or bridged, but such does not 
>> seem to be the case.
>>
>>     
>
> Looks like the other sip box (sofia/external/5001 at testsrv.marketel) is
> returning the failure code. Check the config on the other side? Don't
> you have a gateway setup for this other machine so you couls do
> sofia/gateway/testsrv.marketel/5001 instead?
>
> Andrew
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>   
After about 6 hours of debugging and digging, I finally got the 
Freeswitch installation to dial extensions attached/registered to it.

Buried away in the Freeswitch wikis are 2-3 lines of example for 
mod_sofia that show using a % to separate the extension/number and the 
server name instead of the @ sign that's using in 99% of the 
documentation.  The % syntax means "local".

*sigh*

Now I can get on with working on the conference automated dialing code 
that I was originally trying to prototype through the command line.  
Those commands just weren't working 'cause I was using the @ syntax so 
it was attempting to find a remote server named testsrv.marketel instead 
of routing the call through the local registration list.

-- 
Mark Sobkow
Senior Developer
MarkeTel Multi-Line Dialing Systems LTD.
428 Victoria Ave
Regina, SK S4N-0P6
Toll-Free: 800-289-8616-X533
Local: 306-359-6893-X533
Fax: 306-359-6879
Email: m.sobkow at marketelsystems.com
Web: http://www.marketelsystems.com 



From jerry.richards at teotech.com  Tue Feb  9 12:26:57 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Tue, 9 Feb 2010 12:26:57 -0800
Subject: [Freeswitch-users] Any Known Dual-Registration Issue?
Message-ID: <5AD33F8B1F0848CF9C3A314A60039366@greyhawk.tonecommander.com>

I've noticed that sometimes my phones end up with two registrations with two
Call-IDs at Freeswitch.  Is there any known bug that would cause this?  I've
seen it on different phone models, so I'm thinking there is some timing
issue with Freeswitch.

Best Regards,
Jerry



From Prometheus001 at gmx.net  Tue Feb  9 12:27:40 2010
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Tue, 09 Feb 2010 21:27:40 +0100
Subject: [Freeswitch-users] mod_skypiax - help needed with
 modified	snd-dummy (one way audio)
In-Reply-To: <4B716C90.8070109@gmx.net>
References: <4B60555B.2020004@gmx.net>	<7b197bef1002051319l2fcb5617p1691692f58c65372@mail.gmail.com>	<4B6CC792.5060608@gmx.net>	<191c3a031002060558r36947d93vcbb46df9e0a7dfde@mail.gmail.com>	<4B6F3E9A.2020103@gmx.net>	<7b197bef1002080529u1afc7f9av492b0c221ee2d6af@mail.gmail.com>	<c55d46951002080546p829a881w256bfdb19334c548@mail.gmail.com>	<7b197bef1002080553qd1cb42dla26425942de83c43@mail.gmail.com>	<c55d46951002080604k400344b0i8add00b5c6014958@mail.gmail.com>	<4B703AD0.2080909@gmx.net>	<7b197bef1002080839o61b4b1b5k6d5cc39bb05d9a32@mail.gmail.com>
	<4B716C90.8070109@gmx.net>
Message-ID: <4B71C53C.2020202@gmx.net>

Hello,

thanks to Giovanni's help we solved the problem.
It was an incompatiblity with the current Skype 2.1 Beta with Ubuntu Hardy.
The version skype_static-2.0.0.72 works with good sound quality in both
directions.

Best regards
Peter



Peter P GMX schrieb:
> Hello Giovanni,
>
> I will try to contact you via IRC (stony)
>
> Best regards
> Peter
>
> Giovanni Maruzzelli schrieb:
>   
>> Peter,
>>
>> excuse me but I really do not follow you.
>>
>> Why you have the normal static build not working?
>>
>> Also, this is really taking too much of my time. You continue to
>> change things, and report issues, without waiting for solutions you
>> ask for, then you report something else, and so on... we'll never get
>> at the end of this.
>>
>> If you want, please connect via IRC, and contact me (gmaruzz).
>> Or let me connect ssh to your machine.
>>
>> -gm
>>
>>
>> On Mon, Feb 8, 2010 at 5:24 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>   
>>     
>>> I got it working now with static build and an older version of skype
>>> (skype_static-2.1.0.47).
>>>
>>> But I still have a problem ongoing with sound quality, resp. one way audio:
>>> With original Alsa driver: When Skype calls mod_skypiax => SIP Phone,
>>> then the sound from the SIP phone is interupted regularly 2 times a second.
>>> Example:
>>> When a person on the sip phone speaks
>>> "aaaaaaaaaaaaaaaaaaaaaa"
>>> the other site hears
>>> "aaatataaaatataaaatataaaa"
>>> With "t" meaning the interruption of the sound.
>>>
>>> So I compliled and installed the modified alsa driver as described in
>>> the wiki (configure, make and make install, remove old ubuntu sound dir
>>> in /lib/modules/2.6.24-24-server/ubuntu/sound + depmod -a + reboot the
>>> server.
>>>
>>> Now the SIP phone is heard loud and clearly without interruption.
>>> However the other direction is not heard, so we're at the beginning of
>>> the post (one way audio). Only when I really scratch the microphone then
>>> I hear some parts of this scratching on the SIP side.
>>>
>>> So, some more hints are needed.
>>>
>>> Here's the log, when I start the skype client:
>>>  su root -c "/bin/echo 'username password'| DISPLAY=:101
>>> /usr/bin/skype1 --pipelogin &" &
>>>  /usr/bin/Xvfb :102 -ac &
>>>  error opening security policy file /etc/X11/xserver/SecurityPolicy
>>>  expected keysym, got XF86KbdLightOnOff: line 70 of pc
>>>  expected keysym, got XF86KbdBrightnessDown: line 71 of pc
>>>  expected keysym, got XF86KbdBrightnessUp: line 72 of pc
>>>  Could not init font path element /usr/share/fonts/X11/cyrillic,
>>> removing from list!
>>>  Could not init font path element
>>> /var/lib/defoma/x-ttcidfont-conf.d/dirs/TrueType, removing from list!
>>> But these messages are not critical, right?
>>>
>>> Best regards
>>> Peter
>>>
>>>
>>> Max Bridgewater schrieb:
>>>     
>>>       
>>>> Well, it was a couple of weeks ago. I then switched back to 2.0.4. I'm
>>>> going to try it again and let you know.
>>>>
>>>> Max.
>>>>
>>>> On Mon, Feb 8, 2010 at 8:53 AM, Giovanni Maruzzelli <gmaruzz at celliax.org> wrote:
>>>>
>>>>       
>>>>         
>>>>> Peter is using hardy 64 bit. I checked on that.
>>>>>
>>>>> But, let me understand: if you're using a static build, why you have a
>>>>> problem with QT?
>>>>> Is actually Qt to be statically linked...
>>>>>
>>>>> what is the results of:
>>>>>
>>>>> ldd skype
>>>>>
>>>>> Anyway, in a short while I'll check that on centos 5.4 too, and I'll report here
>>>>>
>>>>> -giovanni
>>>>>
>>>>> On Mon, Feb 8, 2010 at 2:46 PM, Max Bridgewater
>>>>> <max.bridgewater at gmail.com> wrote:
>>>>>
>>>>>         
>>>>>           
>>>>>> Interesting; a while back I tried to install Skypiax with the latest
>>>>>> static build on Skype.com. I had QT library compatibility problem on a
>>>>>> CentOS 5.4 (Final). Giovannni, may I ask what Linux distro you are
>>>>>> using?
>>>>>>
>>>>>> Thanks,
>>>>>> max.
>>>>>>
>>>>>> On Mon, Feb 8, 2010 at 8:29 AM, Giovanni Maruzzelli <gmaruzz at celliax.org> wrote:
>>>>>>
>>>>>>           
>>>>>>             
>>>>>>> Peter,
>>>>>>>
>>>>>>> I just tested with the static build you find on skype.com
>>>>>>>
>>>>>>> I never tested for performances or other issues, there may be (it's a beta).
>>>>>>>
>>>>>>> But it do not crash on me.
>>>>>>>
>>>>>>> I have no problem at all.
>>>>>>>
>>>>>>> If you can give me ssh access I can try to understand why you have so
>>>>>>> many problems.
>>>>>>>
>>>>>>> Or, alternatively, try to follow the wiki. You know, I've not heard
>>>>>>> about those problems.
>>>>>>>
>>>>>>> root at hardy64:~/skype.beta/skype_static-2.1.0.81# ldd skype
>>>>>>>        linux-gate.so.1 =>  (0xffffe000)
>>>>>>>        libasound.so.2 => /usr/lib32/libasound.so.2 (0xf7eb7000)
>>>>>>>        libXv.so.1 => /usr/lib32/libXv.so.1 (0xf7eb2000)
>>>>>>>        libXss.so.1 => /usr/lib32/libXss.so.1 (0xf7eae000)
>>>>>>>        libSM.so.6 => /usr/lib32/libSM.so.6 (0xf7ea6000)
>>>>>>>        libICE.so.6 => /usr/lib32/libICE.so.6 (0xf7e8e000)
>>>>>>>        libXi.so.6 => /usr/lib32/libXi.so.6 (0xf7e86000)
>>>>>>>        libXrender.so.1 => /usr/lib32/libXrender.so.1 (0xf7e7e000)
>>>>>>>        libXrandr.so.2 => /usr/lib32/libXrandr.so.2 (0xf7e78000)
>>>>>>>        libfreetype.so.6 => /usr/lib32/libfreetype.so.6 (0xf7e0a000)
>>>>>>>        libfontconfig.so.1 => /usr/lib32/libfontconfig.so.1 (0xf7de0000)
>>>>>>>        libXext.so.6 => /usr/lib32/libXext.so.6 (0xf7dd2000)
>>>>>>>        libX11.so.6 => /usr/lib32/libX11.so.6 (0xf7ceb000)
>>>>>>>        libz.so.1 => /usr/lib32/libz.so.1 (0xf7cd6000)
>>>>>>>        libgthread-2.0.so.0 => /usr/lib32/libgthread-2.0.so.0 (0xf7cd1000)
>>>>>>>        libglib-2.0.so.0 => /usr/lib32/libglib-2.0.so.0 (0xf7c1f000)
>>>>>>>        librt.so.1 => /lib32/librt.so.1 (0xf7c16000)
>>>>>>>        libdl.so.2 => /lib32/libdl.so.2 (0xf7c12000)
>>>>>>>        libpthread.so.0 => /lib32/libpthread.so.0 (0xf7bfa000)
>>>>>>>        libstdc++.so.6 => /usr/lib32/libstdc++.so.6 (0xf7b07000)
>>>>>>>        libm.so.6 => /lib32/libm.so.6 (0xf7ae2000)
>>>>>>>        libgcc_s.so.1 => /usr/lib32/libgcc_s.so.1 (0xf7ad6000)
>>>>>>>        libc.so.6 => /lib32/libc.so.6 (0xf7987000)
>>>>>>>        libexpat.so.1 => /usr/lib32/libexpat.so.1 (0xf7966000)
>>>>>>>        libXau.so.6 => /usr/lib32/libXau.so.6 (0xf7963000)
>>>>>>>        libxcb-xlib.so.0 => /usr/lib32/libxcb-xlib.so.0 (0xf7961000)
>>>>>>>        libxcb.so.1 => /usr/lib32/libxcb.so.1 (0xf7948000)
>>>>>>>        libpcre.so.3 => /usr/lib32/libpcre.so.3 (0xf7921000)
>>>>>>>        /lib/ld-linux.so.2 (0xf7f86000)
>>>>>>>        libXdmcp.so.6 => /usr/lib32/libXdmcp.so.6 (0xf791c000)
>>>>>>>
>>>>>>>
>>>>>>> On Sun, Feb 7, 2010 at 11:28 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>>>
>>>>>>>             
>>>>>>>               
>>>>>>>> I now used the static Skype binary in order to avoid missing constraints
>>>>>>>> to other libraries: It still crashes
>>>>>>>> 1st it starts with:
>>>>>>>>  process 15431: D-Bus library appears to be incorrectly set up; failed
>>>>>>>> to read machine uuid: Failed to open "/var/lib/dbus/machine-id": No such
>>>>>>>> file or directory
>>>>>>>>  See the manual page for dbus-uuidgen to correct this issue.
>>>>>>>> After calling this client it crashes with:
>>>>>>>>   /usr/bin/skype: symbol lookup error: /usr/bin/skype: undefined
>>>>>>>> symbol: _ZN10QBoxLayout13addSpacerItemEP11QSpacerItem
>>>>>>>>
>>>>>>>> Any hints, where I may get an older Skype client?
>>>>>>>>
>>>>>>>> Best regards
>>>>>>>> Peter
>>>>>>>>
>>>>>>>> Anthony Minessale schrieb:
>>>>>>>>
>>>>>>>>               
>>>>>>>>                 
>>>>>>>>> Quick fyi centos 5.4 has some memory bugs in libc don't go beyond 5.3
>>>>>>>>> until its fixed.
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>                 
>>>>>>>>>                   
>>>>>>>>>> On Feb 5, 2010 7:42 PM, "Peter P GMX" <Prometheus001 at gmx.net
>>>>>>>>>> <mailto:Prometheus001 at gmx.net>> wrote:
>>>>>>>>>>
>>>>>>>>>> Skype starts, but as soon as it receives a call it crashes with:
>>>>>>>>>>
>>>>>>>>>> /usr/bin/skype: symbol lookup error: /usr/bin/skype: undefined symbol:
>>>>>>>>>>
>>>>>>>>>> _ZN10QBoxLayout13addSpacerItemEP11QSpacerItem
>>>>>>>>>>
>>>>>>>>>> I think the 8.10 version dos not work with8.04.
>>>>>>>>>>
>>>>>>>>>> Any hints, where I may get an older Skype client? I may also try the
>>>>>>>>>> static skype client.
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> Best regards
>>>>>>>>>> Peter
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> .
>>>>>>>>>> Giovanni Maruzzelli schrieb:
>>>>>>>>>>
>>>>>>>>>>                   
>>>>>>>>>>                     
>>>>>>>>>>> that's not at all a fatal error.
>>>>>>>>>>> I believe...
>>>>>>>>>>>
>>>>>>>>>>>                     
>>>>>>>>>>>                       
>>>>>>>>> ------------------------------------------------------------------------
>>>>>>>>>
>>>>>>>>> _______________________________________________
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>                 
>>>>>>>>>                   
>>>>>>>> _______________________________________________
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>>               
>>>>>>>>                 
>>>>>>> --
>>>>>>> Sincerely,
>>>>>>>
>>>>>>> Giovanni Maruzzelli
>>>>>>> Cell : +39-347-2665618
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>>             
>>>>>>>               
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>>           
>>>>>>             
>>>>> --
>>>>> Sincerely,
>>>>>
>>>>> Giovanni Maruzzelli
>>>>> Cell : +39-347-2665618
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>>         
>>>>>           
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>       
>>>>         
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>     
>>>       
>>
>>   
>>     
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>   


From brian at freeswitch.org  Tue Feb  9 12:36:19 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 9 Feb 2010 14:36:19 -0600
Subject: [Freeswitch-users] Any Known Dual-Registration Issue?
In-Reply-To: <5AD33F8B1F0848CF9C3A314A60039366@greyhawk.tonecommander.com>
References: <5AD33F8B1F0848CF9C3A314A60039366@greyhawk.tonecommander.com>
Message-ID: <0FF21E6C-39DE-4973-8790-9E02C5ED8BDC@freeswitch.org>

The phone is at fault.  It prob. uses a different call-id for each new reg... and or you restart the phone without it unregistering and you'll end up with TWO.

/b

On Feb 9, 2010, at 2:26 PM, Jerry Richards wrote:

> I've noticed that sometimes my phones end up with two registrations with two
> Call-IDs at Freeswitch.  Is there any known bug that would cause this?  I've
> seen it on different phone models, so I'm thinking there is some timing
> issue with Freeswitch.
> 
> Best Regards,
> Jerry



From brian at freeswitch.org  Tue Feb  9 12:36:52 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 9 Feb 2010 14:36:52 -0600
Subject: [Freeswitch-users] Having trouble establishing a call
In-Reply-To: <4B71C4E1.705@marketelsystems.com>
References: <4B71AD9E.7090606@marketelsystems.com>
	<20100209192537.GA24616@hijacked.us>
	<4B71C4E1.705@marketelsystems.com>
Message-ID: <FB6BCA40-7B12-4BF1-9A2D-892A4FDDA909@freeswitch.org>

Question 2 on the FAQ

http://wiki.freeswitch.org/wiki/FreeSwitch_FAQ#Q:_What_is_the_difference_between_using_a_.25_and_.40_in_a_sofia_dial_string.3F

/b

On Feb 9, 2010, at 2:26 PM, Mark Sobkow wrote:

> Andrew Thompson wrote:
>> On Tue, Feb 09, 2010 at 12:46:54PM -0600, Mark Sobkow wrote:
>> 
>>> We're using Erlang to serve up the configurations to Freeswitch.  I've 
>>> got things configured such that I can place a call from a SIP phone 
>>> registered to extension 5000 to our "external" SIP provider (our 
>>> Asterisk installation), but I can't place a call to extension 5001 from 
>>> 5000.  Below is the trace log Freeswitch produces when I attempt to do so.
>>> 
>>> Any suggestions as to what I should be looking at?  The directory seems 
>>> to be getting served up correctly, as it provides the passwords both SIP 
>>> softphones are using to register with Freeswitch.  I'd have thought that 
>>> once they've registered with FS, the extension would automatically be 
>>> recognized when an incoming call is placed or bridged, but such does not 
>>> seem to be the case.
>>> 
>>> 
>> 
>> Looks like the other sip box (sofia/external/5001 at testsrv.marketel) is
>> returning the failure code. Check the config on the other side? Don't
>> you have a gateway setup for this other machine so you couls do
>> sofia/gateway/testsrv.marketel/5001 instead?
>> 
>> Andrew
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
> After about 6 hours of debugging and digging, I finally got the 
> Freeswitch installation to dial extensions attached/registered to it.
> 
> Buried away in the Freeswitch wikis are 2-3 lines of example for 
> mod_sofia that show using a % to separate the extension/number and the 
> server name instead of the @ sign that's using in 99% of the 
> documentation.  The % syntax means "local".
> 
> *sigh*
> 
> Now I can get on with working on the conference automated dialing code 
> that I was originally trying to prototype through the command line.  
> Those commands just weren't working 'cause I was using the @ syntax so 
> it was attempting to find a remote server named testsrv.marketel instead 
> of routing the call through the local registration list.
> 
> -- 
> Mark Sobkow
> Senior Developer
> MarkeTel Multi-Line Dialing Systems LTD.
> 428 Victoria Ave
> Regina, SK S4N-0P6
> Toll-Free: 800-289-8616-X533
> Local: 306-359-6893-X533
> Fax: 306-359-6879
> Email: m.sobkow at marketelsystems.com
> Web: http://www.marketelsystems.com 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From peder at networkoblivion.com  Tue Feb  9 12:40:55 2010
From: peder at networkoblivion.com (Peder)
Date: Tue, 9 Feb 2010 14:40:55 -0600
Subject: [Freeswitch-users] Any Known Dual-Registration Issue?
In-Reply-To: <5AD33F8B1F0848CF9C3A314A60039366@greyhawk.tonecommander.com>
References: <5AD33F8B1F0848CF9C3A314A60039366@greyhawk.tonecommander.com>
Message-ID: <035001caa9c8$2dca81c0$895f8540$@com>

What kind of phones?  If you have multiple registartion, this can happen
sometimes if you reboot a phone.  Crappy phones, like Grandstream, don't
un-register when you reboot and then when they come back up, they register
again and thus two registrations until the lifetime of the registration ends
and it gets flushed.  Changing the multiple-registration to contact can help
as I believe that uses port and source IP as part of the registration info:

<param name="multiple-registrations" value="contact"/>



-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jerry
Richards
Sent: Tuesday, February 09, 2010 2:27 PM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Any Known Dual-Registration Issue?

I've noticed that sometimes my phones end up with two registrations with two
Call-IDs at Freeswitch.  Is there any known bug that would cause this?  I've
seen it on different phone models, so I'm thinking there is some timing
issue with Freeswitch.

Best Regards,
Jerry


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



From w8hdkim at gmail.com  Tue Feb  9 06:52:51 2010
From: w8hdkim at gmail.com (Kim Culhan)
Date: Tue, 9 Feb 2010 09:52:51 -0500
Subject: [Freeswitch-users] UPnP Timeout
Message-ID: <89dbfdc31002090652h884bc2av82d53c6958fdc409@mail.gmail.com>

On Mon, February 8, 2010 11:21 pm, Troy Anderson wrote:
> I have been using pfSense (1.2.3) and FS.  FS nicely uses UPnP to poke
> holes in the firewall, but it seems that the holes close after a while.  I
> cannot find any documentation in FS nor in pfSense as to what the timeout
> is.  Is there a setting in FS to do some kind of keep-alive thing with
> UPnP to keep, e.g. 5060, open?  Or is it already doing that and pfSense is
> the issue?

FS has provisions for keep-alive, see the bottom of the page for ping
time value:

http://wiki.freeswitch.org/wiki/SIP_Provider_Examples

To watch the pf firewall hole timing you can install pftop from
FreeBSD ports/sysutils
which displays the filter states 'and more'.

-kim


From w8hdkim at gmail.com  Tue Feb  9 08:22:05 2010
From: w8hdkim at gmail.com (Kim Culhan)
Date: Tue, 9 Feb 2010 11:22:05 -0500
Subject: [Freeswitch-users] UPnP Timeout
Message-ID: <89dbfdc31002090822r3f4fd352ncbeb90955b5fbc14@mail.gmail.com>

On Mon, February 8, 2010 11:21 pm, Troy Anderson wrote:
> I have been using pfSense (1.2.3) and FS.  FS nicely uses UPnP to poke
> holes in the firewall, but it seems that the holes close after a while.  I
> cannot find any documentation in FS nor in pfSense as to what the timeout
> is.  Is there a setting in FS to do some kind of keep-alive thing with
> UPnP to keep, e.g. 5060, open?  Or is it already doing that and pfSense is
> the issue?

FS has provisions for keep-alive, see the bottom of the page for ping
time value:

http://wiki.freeswitch.org/wiki/SIP_Provider_Examples

To watch the pf firewall hole timing you can install pftop from
FreeBSD ports/sysutils
which displays the filter states 'and more'.

-kim


From sergey.kobzar at mail.ru  Tue Feb  9 08:57:02 2010
From: sergey.kobzar at mail.ru (Sergey Kobzar)
Date: Tue, 9 Feb 2010 18:57:02 +0200
Subject: [Freeswitch-users] Video conferencing
Message-ID: <57499143.20100209185702@mail.ru>

Hello.

Does anybody have a success story of implementing video conferencing?

I've spend some time with Goole and found that Asterisk has many
limitations, hardware solutions are quite expensive. I played with FS
without luck.

Any ideas?

Thanks.

-- 
Sergey



From jbrucehopkins at gmail.com  Tue Feb  9 12:03:08 2010
From: jbrucehopkins at gmail.com (Bruce Hopkins)
Date: Tue, 9 Feb 2010 20:03:08 +0000
Subject: [Freeswitch-users] g.722 --> SPEEX/16000 or SPEEX/32000 transcoding
	issue
Message-ID: <d03a10e01002091203qde4caabi1a7565929e60918f@mail.gmail.com>

Hi,

Using the FusionPBX ISO (FreeSWITCH 1.0.4) I find I cannot make a call which
requires transcoding from g.722 (or other codec which declares 8kHz sampling
rate in the SDP) to SPEEX/16000 or SPEEX/32000.

If the calling extension uses only g.722, alaw, ulaw, etc, then only the
SPEEX/8000 narrowband variety of SPEEX is offered to the recipient extension
in the SDP of the SIP invite.

If the call is initiated the other way round - e.g. Client using SPEEX/32000
--> FreeSWITCH --> Client using g.722,  then the call is transcoded with no
problem.

I am wondering if this is the intended behaviour, to avoid transcoding
narrowband --> wideband.  However what I am finding is that transcodinig
g722/8000 (wideband) to SPEEX (wideband or ultrawideband) does not seem to
work.

I would be most grateful if anyone were able to let me know if there is a
configuration option I can set to alter this behavious and allow the full
range of SPEEX sampling rates to be offered in the SDP to the receiving
party, regardless of the codec used by the calling party.

Also, is this perhaps different in a more recent version?

Many thanks
Bruce
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100209/959a3d07/attachment.html 

From mike at jerris.com  Tue Feb  9 12:57:26 2010
From: mike at jerris.com (Michael Jerris)
Date: Tue, 9 Feb 2010 15:57:26 -0500
Subject: [Freeswitch-users] uuid_bridge isn't working
In-Reply-To: <fe67a3341002040147i5892d5b8md65e327be6dea7c5@mail.gmail.com>
References: <fe67a3341002040147i5892d5b8md65e327be6dea7c5@mail.gmail.com>
Message-ID: <BDBBA43F-8BC9-4DE9-8A6A-AA808C4FDEA3@jerris.com>

Fixed in rev 16574.

Mike

On Feb 4, 2010, at 4:47 AM, Nagalenoj H. wrote:

> Dear friends,
>     After upgrading to 'FreeSWITCH Version 1.0.trunk (16565)', uuid_bridge isn't working. When I give uuid_bridge, both the legs are not bridged, and they got disconnected.
> 
> Did the following,
> * Made a call to socket extension.
> * Answered the call.
> * Originated a call and parked it.
> * Did uuid_bridge with the uuids of the originated call and caller's uuid.
> 
> * Didn't get the legs bridged, instead both got disconnected,
> 
> Freeswitch debug log:
>       http://pastebin.freeswitch.org/12044
> 
> Facing this problem only after upgrading to this trunk version.

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100209/636a886e/attachment.html 

From robert.hadley at teotech.com  Tue Feb  9 12:58:25 2010
From: robert.hadley at teotech.com (Robert Hadley)
Date: Tue, 9 Feb 2010 12:58:25 -0800
Subject: [Freeswitch-users] Where are new sounds_dir and
	recordings_dirvariables set?
In-Reply-To: <6B8839F5-DA53-4565-A7E0-876FC53BD6F7@freeswitch.org>
References: <28788FA27BE74B9FA6BFFE52BED11B60@greyhawk.tonecommander.com>
	<6B8839F5-DA53-4565-A7E0-876FC53BD6F7@freeswitch.org>
Message-ID: <21D89D46DD7A47DFB60DDF25C3E86814@greyhawk.tonecommander.com>


Configured it with --prefix=/opt/teoswitch.
/r

-----Original Message-----
From: Brian West [mailto:brian at freeswitch.org] 
Sent: Tuesday, February 09, 2010 11:34 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Where are new sounds_dir and
recordings_dirvariables set?

did you configure it with --prefix=/opt/teoswitch or did you move it?

/b

On Feb 9, 2010, at 1:28 PM, Robert Hadley wrote:

>  
> The XML conf files have been recently modified to replace
"$${base_dir}/sounds" with $${sounds_dir}.  The same replacement was done
for $${base_dir}/recordings and $${recordings_dir}.  In vars.xml, the
X-PRE-PROCESS cmd to setting the old variable sound_prefix was removed but
set commands were not added for the new variables sounds_dir and
recordings_dir.  How is FS finding determining the value of $${sounds_dir}
and $${recordings_dir} in the XML files?
>  
> I ask because I have 2 cloned FS servers where one can find sound files in
the IVR when an invalid extension is entered, but the other does not and is
using the wrong path to search for sound files.
>  
> Server 2 Fails:
> 2010-02-09 11:11:16.735745 [DEBUG] switch_ivr_menu.c:381 digits '9999'
> 2010-02-09 11:11:16.735745 [DEBUG] switch_ivr_menu.c:475 action regex
[9999] [/^((5[34][8901][0-9])||(10[0-5][0-9])||(30\d{2}))$/] [0]
> 2010-02-09 11:11:16.735745 [DEBUG] switch_ivr_menu.c:565 IVR menu
'teo_ivr' caught invalid input '9999'
> 2010-02-09 11:11:16.735745 [ERR] mod_sndfile.c:194 Error Opening File
[/opt/teoswitch/sounds/ivr/ivr-that_was_an_invalid_entry.wav] [System error
: No such file or directory.]
>  
> Server 1 Works:
> 2010-02-09 11:17:25.085660 [DEBUG] switch_ivr_menu.c:378 digits '9999'
> 2010-02-09 11:17:25.085660 [DEBUG] switch_ivr_menu.c:472 action regex
[9999] [/^((5[34][8901][0-9])||(10[0-5][0-9])||(30\d{2}))$/] [0]
> 2010-02-09 11:17:25.085660 [DEBUG] switch_ivr_menu.c:562 IVR menu
'teo_ivr' caught invalid input '9999'
> 2010-02-09 11:17:25.085660 [DEBUG] switch_ivr_play_say.c:1162 Codec
Activated L16 at 8000hz 1 channels 20ms
> 2010-02-09 11:17:25.085660 [DEBUG] switch_core_io.c:652
sofia/internal/1045 at 192.168.72.141:5060 receive message
[TRANSCODING_NECESSARY]
> 2010-02-09 11:17:26.765470 [DEBUG] switch_ivr_play_say.c:1454 done playing
file
>  
> I have compared the conf folders and they are nearly identical.  The only
difference I know of is Server 2 was originally compiled with mod_flite
enabled but it is not loaded at runtime.  Any suggestions why one server can
find sound files but the other looks in the wrong path?
>  
> Thanks,
> Robert






From brian at freeswitch.org  Tue Feb  9 12:59:48 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 9 Feb 2010 14:59:48 -0600
Subject: [Freeswitch-users] g.722 --> SPEEX/16000 or SPEEX/32000
	transcoding issue
In-Reply-To: <d03a10e01002091203qde4caabi1a7565929e60918f@mail.gmail.com>
References: <d03a10e01002091203qde4caabi1a7565929e60918f@mail.gmail.com>
Message-ID: <03A2F6D0-754A-4067-A106-9D91B0D68A92@freeswitch.org>

I would recommend you contact the FusionPBX project as 1.0.4 is no longer supported by our team as we are about to release 1.0.5 this week.

/b

On Feb 9, 2010, at 2:03 PM, Bruce Hopkins wrote:

> 
> Using the FusionPBX ISO (FreeSWITCH 1.0.4) I find I cannot make a call which requires transcoding from g.722 (or other codec which declares 8kHz sampling rate in the SDP) to SPEEX/16000 or SPEEX/32000.



From mike at jerris.com  Tue Feb  9 13:02:30 2010
From: mike at jerris.com (Michael Jerris)
Date: Tue, 9 Feb 2010 16:02:30 -0500
Subject: [Freeswitch-users] Passing channel variables to mod_fifo and
	xml_cdr
In-Reply-To: <004301caa608$534747d0$f9d5d770$@net>
References: <004301caa608$534747d0$f9d5d770$@net>
Message-ID: <3B778B9C-8884-4E07-B856-2F7C0B317F36@jerris.com>

There is no association if the agent does not accept the call.  When fifo has calls waiting for agents, it will call as many agents as it needs for calls waiting to be handled, when those agents accept the call, only then does it go and grab a specific caller to attach to the agent.  What most people assume incorrectly is that it is calling the agent using the waiting caller.  There is no connection at all at the time the call is made.  This is the same reason that you do not get caller id of the caller until you answer.  This is how mod_fifo is architected and a behavior that can not be changed without writing a completely different call queue system.

Mike

On Feb 4, 2010, at 9:09 PM, Adam Ford wrote:

> When sending a call through mod_fifo I seem to be losing my custom channel variables that were assigned during prior processing of the call.  In my example, I am trying to assign a unique identifier at the time the call enters my FreeSWITCH system in order to more easily tie the xml_cdr logs together.  This works great, until a call is processed through mod_fifo, which drops my custom channel variable in the calls that it generates.  Is it likely that I have something wrong with my config? Or does mod_fifo not support the passing of custom channel variables?
>  
> The overall problem I am trying to solve is that mod_fifo generates a separate a-leg for every time it rings an agent.  If the agent answers, the a-leg log gets tied to the associated b-leg log with the uuids and I am able to see the entire call in xml_cdr.  However, if the agent rejects the call or doesn?t answer, the a-leg is abandoned with seemingly no association back to a b-leg log.  Anyone have a better suggestion for tying all these xml_cdr logs together for a full picture of the call?  
>  
> -Adam
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100209/12063867/attachment.html 

From mike at jerris.com  Tue Feb  9 13:08:45 2010
From: mike at jerris.com (Michael Jerris)
Date: Tue, 9 Feb 2010 16:08:45 -0500
Subject: [Freeswitch-users] Seeking Advice on SIP Phones like Aastra
In-Reply-To: <845952.61278.qm@web33007.mail.mud.yahoo.com>
References: <845952.61278.qm@web33007.mail.mud.yahoo.com>
Message-ID: <9DFA265F-8632-466B-8910-1E2F9D956908@jerris.com>


On Feb 4, 2010, at 10:38 PM, Darren C. wrote:

> Tim,
>  
> Many thanks for your response.  I posted this message on the Dev list and all I heard was crickets.  I would think a web GUI for a phone would be in demand by the FS community....
>  
> We are doing something similar to what you described.  We?re developing a rather complex IVR/Switching application and it currently does all its database writes via our Web Service to an MS SQL database.  We have a web site that is updated with call details via the web service?s backend database.  From this web interface a user can see counts of voicemails, see call activity, play voicemails, see calls in progress, record calls, etc.  It?s a specialized application so it doesn?t have every PBX feature but this is what we wanted to do with a high-end SIP Phone to replace our office PBX.
>  
> Currently we have an ESI (Estech) E-Class PBX that uses normal digital phones as well as proprietary VOIP (non-SIP) phones.  I think these really nice SIP phones with huge color touchscreens would be much better than even high-end proprietary digital phones + we?d get all the benefits of FS.  We?ll just need to add all the basic PBX capabilities to the phone?s GUI to see how many voicemails are waiting, how many lines are in use, button for call transfer, etc.
>  
> As you mentioned:  ?you can do a DB query to find out what voicemail messages a user has, or what calls are active, etc.?  We are keeping track of all this ourselves via our web service?s backend database.  I?m not sure I need to do this for everything but we are.  It?s a multi-tenant system with hundreds of tenants so I?m guessing I might lose some needed relationships by querying FS but I?m going to re-visit this?I will make sure I can?t just query FS like you?re doing for some of these things?we?ve never turned on ODBC or even looked for the documentation as to what FS stores.  We have to run FS in Linux for some Sangoma stuff but we?re Windows people so that is another reason we store via web services to an MS SQL backend.

FreeSWITCH can use ODBC to talk directly to your db, and for this sort of setup, its probably a requirement to use a db other than the embedded database.  Sangoma wanpipe on windows I am told works or basically works.  I would talk to sangoma for more information.

> I was worried I?d get one of these fancy phones and find out it doesn?t support important SIP/FS features rendering the color touchscreen useless.  I?ve never owned one of these SIP phones, just used various softphones.  But thanks to you I?ll get an Aastra 6739i and give it a try.

Typically the applications on the phone are not related to sip at all.

> I have a fulltime programmer working on this system on and off now for over a year but this has been mostly developing the IVR.  We haven't made any attempt at using FS as a PBX.  So based on your comments I think I?ll purchase an Aastra 6739i and develop a custom SIP Phone GUI interface with FS.

I am not sure the phone gui has much to do with FreeSWITCH at all, other than pulling a little data from the databases (again, ODBC highly recommended)
 
> If you or others would like to collaborate on an Aastra 6739i phone GUI for FS, feel free to contact me.  We can try to make it extensible for other phones as well.  My email is ustcor<remove this junk>poration at yah<remove this too>oo.com. 

Can you post for all to see some idea of how these applications work, lanagages used, some samples so we can see if we can get some interest in this?

Mike

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100209/558fc211/attachment-0001.html 

From mike at jerris.com  Tue Feb  9 13:10:06 2010
From: mike at jerris.com (Michael Jerris)
Date: Tue, 9 Feb 2010 16:10:06 -0500
Subject: [Freeswitch-users] Presence PUBLISH Not Updating After
	Softphone OffLine Then Available
In-Reply-To: <31FEE9072F104AE18B26812978FCA341@greyhawk.tonecommander.com>
References: <31FEE9072F104AE18B26812978FCA341@greyhawk.tonecommander.com>
Message-ID: <45963EB5-DFE5-43FA-AFCC-857E1E719E2D@jerris.com>

Try this again, I think I saw changes go in for this issue.

Mike

On Feb 5, 2010, at 2:38 PM, Jerry Richards wrote:

> I found a scenario where presence status is not distributed to subscribers.
> This is using the latest changes (as of Feb 03, 2010).  The scenario
> follows:
> 
> 1) Register two Bria softphones (A and B), which each have the other as a
> contact.
> 2) Set softphone B's presence status to 'Appear Offline'.
> 3) Softphone A correctly reflects contact B is offline.
> 4) Set softphone B's presence status to 'Available'.
> 5) ******* There is no change to contact B's status at softphone A *******
> 
> I posted a log at http://pastebin.freeswitch.org/12054.  At line 773, there
> is an error when FS is processing the PUBLISH from softphone B:
> 
> 773.2010-02-05 10:29:21.254221 [ERR] sofia_presence.c:674 DUMP PRESENCE SQL:
> 
> I did notice that after about 30 minutes, softphone B's status gets
> reflected at softphone A.



From mike at jerris.com  Tue Feb  9 13:12:38 2010
From: mike at jerris.com (Michael Jerris)
Date: Tue, 9 Feb 2010 16:12:38 -0500
Subject: [Freeswitch-users] Driving peripherals through Freeswitch?
In-Reply-To: <2srvm5945qgcno44oetn9ngii0u3aed73p@4ax.com>
References: <2srvm5945qgcno44oetn9ngii0u3aed73p@4ax.com>
Message-ID: <20F1477A-E98A-4BC9-904D-CB313D8E7B4C@jerris.com>

The possibilities are limitless, but requires someone to code a module to interface, or external scripts using the system api or some socket based application.

Mike

On Feb 8, 2010, at 6:07 AM, Fred-145 wrote:
> I don't know anything about this, but I was wondering if someone had
> successfully used a Freeswitch server to drive peripherals like
> switching on a heater by sending an SMS or calling an extension, etc.?
> 
> I'm thinking of tools like X10 to drive peripherals from a PC.
> 
> Has someone played with this kind of tool and could tell me what is
> technically possible?



From mike at jerris.com  Tue Feb  9 13:14:42 2010
From: mike at jerris.com (Michael Jerris)
Date: Tue, 9 Feb 2010 16:14:42 -0500
Subject: [Freeswitch-users] Dialplan search order
In-Reply-To: <798899361.20100208131909@yes.net.ua>
References: <798899361.20100208131909@yes.net.ua>
Message-ID: <E79C2B30-921E-4DC9-A045-D4B550F65B02@jerris.com>


On Feb 8, 2010, at 6:19 AM, Mike Tkachuk wrote:
> 
>  I'm using xml_curl for external dialplan fetch, but I like to
>  split static and dynamic parts of configuration, so for example,
>  leave call unloop logic, and voicemail extension in static xml file
>  while having all other parts dynamic. It will allow to avoid
>  unnecessary call to costly external source and also avoid xml parse
>  of content that is static.
> 
>  Currently FS first look in xml_curl and only after that falls back
>  to static files. Is that behavior possible to change, so FS will work
>  like that:
> 
>  1 - Look in static xml file and execute all extensions that have
>  'continue="true"'
>  2 - If previous step didn't stop on matching extension than look in
>  xml_curl or other source specified in dialplan param of sofia
>  config.
> 
>  Looks like <param name="dialplan" value="XML:/path_to_dialplan_default.xml,XML"/>
>  don't do the trick.

This is exactly how you do this.  What is not working about it?

Mike
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100209/70365a7c/attachment.html 

From jbrucehopkins at gmail.com  Tue Feb  9 13:15:05 2010
From: jbrucehopkins at gmail.com (Bruce Hopkins)
Date: Tue, 9 Feb 2010 21:15:05 +0000
Subject: [Freeswitch-users] g.722 --> SPEEX/16000 or SPEEX/32000
	transcoding issue
In-Reply-To: <03A2F6D0-754A-4067-A106-9D91B0D68A92@freeswitch.org>
References: <d03a10e01002091203qde4caabi1a7565929e60918f@mail.gmail.com>
	<03A2F6D0-754A-4067-A106-9D91B0D68A92@freeswitch.org>
Message-ID: <d03a10e01002091315n5fa9242eoe2a86c4edee1501d@mail.gmail.com>

OK, many thanks for the extremely swift response Brian.

I will try to get up and running as soon as I can with 1.0.5 and see if the
issue goes away.

thanks again
Bruce

On 9 February 2010 20:59, Brian West <brian at freeswitch.org> wrote:

> I would recommend you contact the FusionPBX project as 1.0.4 is no longer
> supported by our team as we are about to release 1.0.5 this week.
>
> /b
>
> On Feb 9, 2010, at 2:03 PM, Bruce Hopkins wrote:
>
> >
> > Using the FusionPBX ISO (FreeSWITCH 1.0.4) I find I cannot make a call
> which requires transcoding from g.722 (or other codec which declares 8kHz
> sampling rate in the SDP) to SPEEX/16000 or SPEEX/32000.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100209/9f32a266/attachment.html 

From robert.hadley at teotech.com  Tue Feb  9 13:14:57 2010
From: robert.hadley at teotech.com (Robert Hadley)
Date: Tue, 9 Feb 2010 13:14:57 -0800
Subject: [Freeswitch-users] Way to pass max-members from dialplan into
	conference?
Message-ID: <C7136E29C3FD472BB367FF412E282BED@greyhawk.tonecommander.com>

I have setting max-members=10 in conference.conf.xml working.  However, is
there are way to pass in the max-members=10 from the dialplan/default.xml to
mod_conference?  I tried using action application="set"
data="max-members=10" but it didn't work.  Also tried action
application="export" data="max-members=10" but it didn't work either.

 

>From default.xml:

    <extension name="nb_conferences">

      <condition field="destination_number" expression="^(30\d{2})$">

       <action application="answer"/>

<!-- tried set and export here to no avail -->

       <action application="conference" data="$1-${domain_name}@default"/>

 

Thanks,

Robert

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100209/e18f8b40/attachment.html 

From mike at jerris.com  Tue Feb  9 13:16:13 2010
From: mike at jerris.com (Michael Jerris)
Date: Tue, 9 Feb 2010 16:16:13 -0500
Subject: [Freeswitch-users] Video conferencing
In-Reply-To: <57499143.20100209185702@mail.ru>
References: <57499143.20100209185702@mail.ru>
Message-ID: <D9657862-D8A0-45C8-98E2-42814BE8D6A1@jerris.com>

Our video conference features "work" but the functionality is pretty limited.  We don;t have iframe detection and can not do any video transcoding, just video follow audio support.  This code needs some work for sure.

Mike

On Feb 9, 2010, at 11:57 AM, Sergey Kobzar wrote:

> Does anybody have a success story of implementing video conferencing?
> 
> I've spend some time with Goole and found that Asterisk has many
> limitations, hardware solutions are quite expensive. I played with FS
> without luck.



From brian at freeswitch.org  Tue Feb  9 13:18:33 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 9 Feb 2010 15:18:33 -0600
Subject: [Freeswitch-users] g.722 --> SPEEX/16000 or SPEEX/32000
	transcoding issue
In-Reply-To: <d03a10e01002091315n5fa9242eoe2a86c4edee1501d@mail.gmail.com>
References: <d03a10e01002091203qde4caabi1a7565929e60918f@mail.gmail.com>
	<03A2F6D0-754A-4067-A106-9D91B0D68A92@freeswitch.org>
	<d03a10e01002091315n5fa9242eoe2a86c4edee1501d@mail.gmail.com>
Message-ID: <F167991C-151B-4D40-BF27-80A4C3A44D1E@freeswitch.org>

What you're saying makes little or no sense to me even on 1.0.4, Can you pastebin your logs?

/b

On Feb 9, 2010, at 3:15 PM, Bruce Hopkins wrote:

> OK, many thanks for the extremely swift response Brian.
> 
> I will try to get up and running as soon as I can with 1.0.5 and see if the issue goes away.
> 
> thanks again
> Bruce



From mike at jerris.com  Tue Feb  9 13:25:35 2010
From: mike at jerris.com (Michael Jerris)
Date: Tue, 9 Feb 2010 16:25:35 -0500
Subject: [Freeswitch-users] No reINVITE when bridging two sessions from
	JavaScript with bypass_media_after_bridge=true
In-Reply-To: <8e9d67561002051202qeaf1551v8fdd29565f8c02fd@mail.gmail.com>
References: <8e9d67561002041147q6c135546u1f7e1887529a916d@mail.gmail.com>
	<3D739D7F-F5C6-4266-B04E-9413DCFDA6AF@freeswitch.org>
	<8e9d67561002050018q24e3a806kc8807b55268e759b@mail.gmail.com>
	<B66289FF-C2A9-4AC4-A56C-66E51B4726F4@freeswitch.org>
	<8e9d67561002051202qeaf1551v8fdd29565f8c02fd@mail.gmail.com>
Message-ID: <154214DA-5781-4FC1-81EA-B41392E6ACF6@jerris.com>

controlling multiple calls in a script like this is tricky, you need to use the first session to create the second one.  Why are you not just doing an originate to do all of this not even in a js file?  What exactly are you trying to accomplish

Mike

On Feb 5, 2010, at 3:02 PM, Timur Valishev wrote:

> I think we are on the right way) still does not work, but there is hope)
> 
> First of all, this script does not produce any reinvite either (even if replace bypass_media to bypass_media_after_bridge, or set bypass_media only on one channel):
> 
> <<<<<<<<<<<<<<<<<<<<<<<<<<<<<
> session = new Session("{bypass_media=true,ignore_early_media=true,hangup_after_bridge=true}user/1001");
> session2 = new Session("{bypass_media=true,ignore_early_media=true} user/1001");
> bridge(session, session2);
> >>>>>>>>>>>>>>>>>>>>>>>>>>>>>
> 
> BUT! if I run the following script:
> <<<<<<<<<<<<<<<<<<<<<<<<<<<<<
> session = new Session("{bypass_media=true,ignore_early_media=true,hangup_after_bridge=true} user/1001");
> session2 = new Session("{bypass_media=true,ignore_early_media=true}user/1001");
> >>>>>>>>>>>>>>>>>>>>>>>>>>>>>
> 
> And then manually type in the console
> uuid_media off <uuid of my session>
> 
> - then I get the reINVITE!
> 
> BUT! When I try to write it to the script:
> <<<<<<<<<<<<<<<<<<<<<<<<<<<<<
> session = new Session("{bypass_media=true,ignore_early_media=true,hangup_after_bridge=true}sofia/external/timwork at novion.ru");
> session2 = new Session("{bypass_media=true,ignore_early_media=true}sofia/external/timwork at novion.ru");
> bridge(session, session2);
> apiExecute('uuid_media off '+session.uuid); // <-- this line is not executed, because bridge hangs up untill BYE
> >>>>>>>>>>>>>>>>>>>>>>>>>>>>>
> 
> the last line is not executed, because bridge hangs up untill BYE
> 
> Then I've tried to do like this:
> <<<<<<<<<<<<<<<<<<<<<<<<<<<<<
> session = new Session("{bypass_media=true,ignore_early_media=true,hangup_after_bridge=true}user/1001");
> session2 = new Session("{bypass_media=true,ignore_early_media=true}user/1001");
> 
> session.setAutoHangup(false)
> session2.setAutoHangup(false)
> 
> apiExecute("uuid_bridge "+session.uuid+" "+session2.uuid);
> apiExecute('uuid_media off '+session.uuid);
> >>>>>>>>>>>>>>>>>>>>>>>>>>>>>
> 
> But sessions do not get bridged -( Even if I insert session.ready() after each call.
> 
> Any ideas on how to call the functions correctly to get the reINVITE?
> 
> Best regards,
> Timur Valishev
> 
> 2010/2/5 Brian West <brian at freeswitch.org>
> set it inside each of the {} for each session you create its not set after the fact the call is up already...  you're setting it too late.
> 
> you an also issue uuid_media off <uuid>
> 
> /b
> 
> On Feb 5, 2010, at 2:18 AM, Timur Valishev wrote:
> 
>> I've modified my script to make sure: <<<<<<<<<<<<<<<<<<<<<<<<<<<<,
>> session = new Session(
>> "{ignore_early_media=true,hangup_after_bridge=true}sofia/external/timwork at novion.ru"
>> );
>> session2 = new Session(
>> "{ignore_early_media=true}sofia/external/timwork at novion.ru"
>> );
>> session.setVariable('bypass_media', 'true');
>> session2.setVariable('bypass_media', 'true');
>> bridge(session, session2);
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100209/41bcb5e4/attachment-0001.html 

From jbrucehopkins at gmail.com  Tue Feb  9 13:55:13 2010
From: jbrucehopkins at gmail.com (Bruce Hopkins)
Date: Tue, 9 Feb 2010 21:55:13 +0000
Subject: [Freeswitch-users] g.722 --> SPEEX/16000 or SPEEX/32000
	transcoding issue
In-Reply-To: <F167991C-151B-4D40-BF27-80A4C3A44D1E@freeswitch.org>
References: <d03a10e01002091203qde4caabi1a7565929e60918f@mail.gmail.com>
	<03A2F6D0-754A-4067-A106-9D91B0D68A92@freeswitch.org>
	<d03a10e01002091315n5fa9242eoe2a86c4edee1501d@mail.gmail.com>
	<F167991C-151B-4D40-BF27-80A4C3A44D1E@freeswitch.org>
Message-ID: <d03a10e01002091355n1934627t3902e0d8af5c7dc3@mail.gmail.com>

Willdo,

To clarify in brief though, the scenario which occurs and causes the call to
fail is:

SIP client  1 (g.722 enabled only ) -----> INVITE (with SDP offer: Media
Attribute (a): rtpmap:9 g722/8000 ) -->FreeSWITCH

---> INVITE (with SDP offer including a bunch of codecs including rtpmap:
rtpmap:98 SPEEX/8000 but crucially *not* including SPEEX/16000 or
SPEEX/32000)

---> SIP client 2 (with only SPEEX/16000 or SPEEX/32000 enabled).

The second SIP client does not get offered a codec it can accept, so SIP
client 1 is sent a method 488 "Not Acceptable Here" message and the calling
party gets directed to the voicemail for the other SIP client.

By contrast, there is no problem calling SPEEX/32000 --> SPEEX/32000 or
calling SPEEX/16000 --> SPEEX/16000.

there is also no problem calling SPEEX/32000 --> g.722/8000.

I am wondering if the problem is that FreeSWITCH is interpreting g.722 as
being a narrowband (8kHz sample rate) codec, due to the historic anomaly of
it presenting g722/8000 in the SDP even though it in fact uses 16kHz
sampling, and for that reason not wanting to offer a 16kHz sample rate codec
to the second SIP client?

I suggest this as I also found trying to call alaw --> SPEEX/16000 does not
work, for example.



Here is the log file for the scenario which does not work (g.722 client
trying to call Speex wideband client).  Please let me know if a Wireshark
trace would be helpful.

2010-02-09 21:49:32.784387 [DEBUG] sofia.c:5224 0 acls to check for proxy
2010-02-09 21:49:32.784387 [DEBUG] sofia.c:5242 network ip is a proxy [0]
2010-02-09 21:49:32.784387 [DEBUG] sofia.c:5270 IP 192.168.10.131 Rejected
by acl "domains". Falling back to Digest auth.
2010-02-09 21:49:32.791074 [DEBUG] sofia.c:5224 0 acls to check for proxy
2010-02-09 21:49:32.791074 [DEBUG] sofia.c:5242 network ip is a proxy [0]
2010-02-09 21:49:32.791074 [DEBUG] sofia.c:5270 IP 192.168.10.131 Rejected
by acl "domains". Falling back to Digest auth.
2010-02-09 21:49:32.794074 [NOTICE] switch_channel.c:613 New Channel
sofia/internal/60002 at 192.168.10.30 [66610a8c-aec9-48ca-9b67-aafdcd3e3d08]
2010-02-09 21:49:32.794074 [DEBUG] sofia.c:3727 Channel sofia/internal/
60002 at 192.168.10.30 entering state [received][100]
2010-02-09 21:49:32.794074 [DEBUG] sofia.c:3738 Remote SDP:
v=0
o=- 5 2 IN IP4 192.168.10.131
s=CounterPath eyeBeam 1.5
c=IN IP4 192.168.10.131
t=0 0
m=audio 25592 RTP/AVP 9 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

2010-02-09 21:49:32.794074 [DEBUG] sofia_glue.c:3305 Audio Codec Compare
[G722:9:8000:20]/[G722:9:8000:20]
2010-02-09 21:49:32.795079 [DEBUG] sofia_glue.c:2143 Set Codec
sofia/internal/60002 at 192.168.10.30 G722/8000 20 ms 160 samples
2010-02-09 21:49:32.795079 [DEBUG] sofia_glue.c:3261 Set 2833 dtmf payload
to 101
2010-02-09 21:49:32.795079 [DEBUG] sofia.c:3885 (sofia/internal/
60002 at 192.168.10.30) State Change CS_NEW -> CS_INIT
2010-02-09 21:49:32.795079 [DEBUG] switch_core_session.c:999 Send signal
sofia/internal/60002 at 192.168.10.30 [BREAK]
2010-02-09 21:49:32.795079 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/60002 at 192.168.10.30) Running State Change CS_INIT
2010-02-09 21:49:32.795079 [DEBUG] switch_core_state_machine.c:338
(sofia/internal/60002 at 192.168.10.30) State INIT
2010-02-09 21:49:32.795079 [DEBUG] mod_sofia.c:83 sofia/internal/
60002 at 192.168.10.30 SOFIA INIT
2010-02-09 21:49:32.795079 [DEBUG] mod_sofia.c:111 (sofia/internal/
60002 at 192.168.10.30) State Change CS_INIT -> CS_ROUTING
2010-02-09 21:49:32.795079 [DEBUG] switch_core_session.c:999 Send signal
sofia/internal/60002 at 192.168.10.30 [BREAK]
2010-02-09 21:49:32.795079 [DEBUG] switch_core_state_machine.c:338
(sofia/internal/60002 at 192.168.10.30) State INIT going to sleep
2010-02-09 21:49:32.795079 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/60002 at 192.168.10.30) Running State Change CS_ROUTING
2010-02-09 21:49:32.795079 [DEBUG] switch_core_state_machine.c:341
(sofia/internal/60002 at 192.168.10.30) State ROUTING
2010-02-09 21:49:32.795079 [DEBUG] mod_sofia.c:132 sofia/internal/
60002 at 192.168.10.30 SOFIA ROUTING
2010-02-09 21:49:32.795079 [DEBUG] switch_core_state_machine.c:78
sofia/internal/60002 at 192.168.10.30 Standard ROUTING
2010-02-09 21:49:32.795079 [INFO] mod_dialplan_xml.c:408 Processing
60002->1001 in context default
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->unloop]
continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (PASS) [unloop]
${unroll_loops}(true) =~ /^true$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [unloop]
${sip_looped_call}() =~ /^true$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->tod_example]
continue=true
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (PASS) [tod_example]
${strftime(%w)}(2) =~ /^([1-5])$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [tod_example]
${strftime(%H%M)}(2149) =~ /^((09|1[0-7])[0-5][0-9]|1800)$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
[default->global-intercept] continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [global-intercept]
destination_number(1001) =~ /^\*886$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
[default->group-intercept] continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [group-intercept]
destination_number(1001) =~ /^\*8$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
[default->intercept-ext] continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [intercept-ext]
destination_number(1001) =~ /^\*\*(\d+)$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->redial]
continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [redial]
destination_number(1001) =~ /^\*870$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->global]
continue=true
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [global]
${network_addr}(192.168.10.131) =~ /^$/ break=never
Dialplan: sofia/internal/60002 at 192.168.10.30 ANTI-Action
set(use_profile=${cond(${acl(${network_addr} rfc1918)} == true ? nat :
default)})
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (PASS) [global]
${numbering_plan}() =~ /^$/ break=never
Dialplan: sofia/internal/60002 at 192.168.10.30 Action
set_user(default@${domain_name})

Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [global]
${call_debug}(false) =~ /^true$/ break=never
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [global]
${sip_has_crypto}() =~ /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/
break=never
Dialplan: sofia/internal/60002 at 192.168.10.30 Absolute Condition [global]
Dialplan: sofia/internal/60002 at 192.168.10.30 Action
db(insert/${domain_name}-spymap/${caller_id_number}/${uuid})
Dialplan: sofia/internal/60002 at 192.168.10.30 Action
db(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number})

Dialplan: sofia/internal/60002 at 192.168.10.30 Action
db(insert/${domain_name}-last_dial/global/${uuid})
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->snom-demo-2]
continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [snom-demo-2]
destination_number(1001) =~ /^\*9001$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->snom-demo-1]
continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [snom-demo-1]
destination_number(1001) =~ /^\*9000$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->eavesdrop]
continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [eavesdrop]
destination_number(1001) =~ /^\*88(.*)$|^\*0(.*)$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->eavesdrop]
continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [eavesdrop]
destination_number(1001) =~ /^\*779$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->call_privacy]
continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [call_privacy]
destination_number(1001) =~ /^\*67(\d+)$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->call_return]
continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [call_return]
destination_number(1001) =~ /^\*69$|^869$|^lcr$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->del-group]
continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [del-group]
destination_number(1001) =~ /^\*80(\d{2})$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->add-group]
continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [add-group]
destination_number(1001) =~ /^\*81(\d{2})$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
[default->call-group-simo] continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [call-group-simo]
destination_number(1001) =~ /^\*82(\d{2})$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
[default->call-group-order] continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [call-group-order]
destination_number(1001) =~ /^\*83(\d{2})$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
[default->extension-intercom] continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL)
[extension-intercom] destination_number(1001) =~ /^\*8(\d{4})$/
break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
[default->send_to_voicemail_5digits] continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL)
[send_to_voicemail_5digits] destination_number(1001) =~ /^\*99(\d{5})$/
break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
[default->send_to_voicemail_4digits] continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL)
[send_to_voicemail_4digits] destination_number(1001) =~ /^\*99(\d{4})$/
break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
[default->send_to_voicemail_3digits] continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL)
[send_to_voicemail_3digits] destination_number(1001) =~ /^\*99(\d{3})$/
break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->pizza_demo]
continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [pizza_demo]
destination_number(1001) =~ /^(pizza|74992)$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->2001]
continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [2001]
destination_number(1001) =~ /^2001$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->Call via
asterisk-pbx1] continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [Call via
asterisk-pbx1] destination_number(1001) =~ /269065/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->pizza_demo]
continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [pizza_demo]
destination_number(1001) =~ /^(pizza|74992)$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->5002]
continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [5002]
destination_number(1001) =~ /^5002$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->7002]
continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [7002]
destination_number(1001) =~ /^7002$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->DISA]
continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [DISA]
destination_number(1001) =~ /^\*(3472)$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->Recordings]
continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [Recordings]
destination_number(1001) =~ /^\*(732673)$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->7002.park]
continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [7002.park]
destination_number(1001) =~ /^\*7002$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
[default->group_dial_sales] continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [group_dial_sales]
destination_number(1001) =~ /^\*2000$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
[default->group_dial_support] continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL)
[group_dial_support] destination_number(1001) =~ /^\*2001$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
[default->group_dial_billing] continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL)
[group_dial_billing] destination_number(1001) =~ /^\*2002$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->vmain2]
continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [vmain2]
destination_number(1001) =~ /^vmain2$|^\*97$|^\*4000$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->vmain]
continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [vmain]
destination_number(1001) =~ /^vmain$|^\*98$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->sip_uri]
continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [sip_uri]
destination_number(1001) =~ /^sip:(.*)$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
[default->nb_conferences] continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [nb_conferences]
destination_number(1001) =~ /^\*(30\d{2})$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
[default->wb_conferences] continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [wb_conferences]
destination_number(1001) =~ /^\*(31\d{2})$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
[default->uwb_conferences] continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [uwb_conferences]
destination_number(1001) =~ /^\*(32\d{2})$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
[default->cdquality_conferences] continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL)
[cdquality_conferences] destination_number(1001) =~ /^\*(33\d{2})$/
break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
[default->freeswitch_public_conf_via_sip] continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL)
[freeswitch_public_conf_via_sip] destination_number(1001) =~
/^\*9(888|1616|3232)$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
[default->mad_boss_intercom] continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL)
[mad_boss_intercom] destination_number(1001) =~ /^\*0911$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
[default->mad_boss_intercom] continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL)
[mad_boss_intercom] destination_number(1001) =~ /^\*0912$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->mad_boss]
continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [mad_boss]
destination_number(1001) =~ /^\*0913$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->ivr_demo]
continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [ivr_demo]
destination_number(1001) =~ /^\*5000$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
[default->dynamic_conference] continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL)
[dynamic_conference] destination_number(1001) =~ /^\*5001$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
[default->rtp_multicast_page] continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL)
[rtp_multicast_page] destination_number(1001) =~ /^\*pagegroup$|^\*7243/
break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->park]
continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [park]
destination_number(1001) =~ /^\*5900$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->unpark]
continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [unpark]
destination_number(1001) =~ /^\*5901$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->park]
continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (PASS) [park]
source(mod_sofia) =~ /mod_sofia/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [park]
destination_number(1001) =~ /park\+(\d+)/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->unpark]
continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (PASS) [unpark]
source(mod_sofia) =~ /mod_sofia/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [unpark]
destination_number(1001) =~ /^parking$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->park]
continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (PASS) [park]
source(mod_sofia) =~ /mod_sofia/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [park]
destination_number(1001) =~ /callpark/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->unpark]
continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (PASS) [unpark]
source(mod_sofia) =~ /mod_sofia/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [unpark]
destination_number(1001) =~ /pickup/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->wait]
continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [wait]
destination_number(1001) =~ /^wait$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->fax_receive]
continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [fax_receive]
destination_number(1001) =~ /^\*9978$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->fax_transmit]
continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [fax_transmit]
destination_number(1001) =~ /^\*9979$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->ringback_180]
continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [ringback_180]
destination_number(1001) =~ /^\*9980$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
[default->ringback_183_uk_ring] continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL)
[ringback_183_uk_ring] destination_number(1001) =~ /^\*9981$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
[default->ringback_183_music_ring] continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL)
[ringback_183_music_ring] destination_number(1001) =~ /^\*9982$/
break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
[default->ringback_post_answer_uk_ring] continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL)
[ringback_post_answer_uk_ring] destination_number(1001) =~ /^\*9983$/
break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
[default->ringback_post_answer_music] continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL)
[ringback_post_answer_music] destination_number(1001) =~ /^\*9984$/
break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->ClueCon]
continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [ClueCon]
destination_number(1001) =~ /^\*9991$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->show_info]
continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [show_info]
destination_number(1001) =~ /^\*9992$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->video_record]
continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [video_record]
destination_number(1001) =~ /^\*9993$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
[default->video_playback] continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [video_playback]
destination_number(1001) =~ /^\*9994$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->delay_echo]
continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [delay_echo]
destination_number(1001) =~ /^\*9995$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->echo]
continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [echo]
destination_number(1001) =~ /^\*9996$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->milliwatt]
continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [milliwatt]
destination_number(1001) =~ /^\*9997$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->tone_stream]
continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [tone_stream]
destination_number(1001) =~ /^\*9998$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
[default->zrtp_enrollement] continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [zrtp_enrollement]
destination_number(1001) =~ /^\*9787$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->hold_music]
continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [hold_music]
destination_number(1001) =~ /^\*9999$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
[default->Local_Extension] continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (PASS) [Local_Extension]
destination_number(1001) =~ /(^\d{6}$|\d{5}$|^\d{4}$|^\d{3}$)/
break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 Action
set(dialed_extension=1001)
Dialplan: sofia/internal/60002 at 192.168.10.30 Action
export(dialed_extension=1001)
Dialplan: sofia/internal/60002 at 192.168.10.30 Action bind_meta_app(1 b s
execute_extension::dx XML features)
Dialplan: sofia/internal/60002 at 192.168.10.30 Action bind_meta_app(2 b s
record_session::/usr/local/freeswitch/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav)

Dialplan: sofia/internal/60002 at 192.168.10.30 Action bind_meta_app(3 b s
execute_extension::cf XML features)
Dialplan: sofia/internal/60002 at 192.168.10.30 Action set(ringback=${us-ring})

Dialplan: sofia/internal/60002 at 192.168.10.30 Action
set(transfer_ringback=local_stream://moh)
Dialplan: sofia/internal/60002 at 192.168.10.30 Action set(call_timeout=30)
Dialplan: sofia/internal/60002 at 192.168.10.30 Action
set(hangup_after_bridge=true)
Dialplan: sofia/internal/60002 at 192.168.10.30 Action
set(continue_on_fail=true)
Dialplan: sofia/internal/60002 at 192.168.10.30 Action
db(insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number})

Dialplan: sofia/internal/60002 at 192.168.10.30 Action
db(insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid})
Dialplan: sofia/internal/60002 at 192.168.10.30 Action
set(called_party_callgroup=${user_data(${dialed_extension}@${domain_name}
var callgroup)})
Dialplan: sofia/internal/60002 at 192.168.10.30 Action
db(insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid})
Dialplan: sofia/internal/60002 at 192.168.10.30 Action
bridge(user/${dialed_extension}@${domain_name})
Dialplan: sofia/internal/60002 at 192.168.10.30 Action answer()
Dialplan: sofia/internal/60002 at 192.168.10.30 Action sleep(1000)
Dialplan: sofia/internal/60002 at 192.168.10.30 Action voicemail(default
${domain_name} ${dialed_extension})
2010-02-09 21:49:32.798073 [DEBUG] switch_core_state_machine.c:122
(sofia/internal/60002 at 192.168.10.30) State Change CS_ROUTING -> CS_EXECUTE
2010-02-09 21:49:32.798073 [DEBUG] switch_core_session.c:999 Send signal
sofia/internal/60002 at 192.168.10.30 [BREAK]
2010-02-09 21:49:32.798073 [DEBUG] switch_core_state_machine.c:341
(sofia/internal/60002 at 192.168.10.30) State ROUTING going to sleep
2010-02-09 21:49:32.798073 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/60002 at 192.168.10.30) Running State Change CS_EXECUTE
2010-02-09 21:49:32.798073 [DEBUG] switch_core_state_machine.c:348
(sofia/internal/60002 at 192.168.10.30) State EXECUTE
2010-02-09 21:49:32.798073 [DEBUG] mod_sofia.c:181 sofia/internal/
60002 at 192.168.10.30 SOFIA EXECUTE
2010-02-09 21:49:32.798073 [DEBUG] switch_core_state_machine.c:159
sofia/internal/60002 at 192.168.10.30 Standard EXECUTE
EXECUTE sofia/internal/60002 at 192.168.10.30 set(use_profile=default)
2010-02-09 21:49:32.798073 [DEBUG] mod_dptools.c:768 sofia/internal/
60002 at 192.168.10.30 SET [use_profile]=[default]
EXECUTE sofia/internal/60002 at 192.168.10.30 set_user(default at 192.168.10.30)
EXECUTE sofia/internal/60002 at 192.168.10.30db(insert/192.168.10.30-spymap/60002/66610a8c-aec9-48ca-9b67-aafdcd3e3d08)
EXECUTE sofia/internal/60002 at 192.168.10.30db(insert/192.168.10.30-last_dial/60002/1001)
EXECUTE sofia/internal/60002 at 192.168.10.30db(insert/192.168.10.30-last_dial/global/66610a8c-aec9-48ca-9b67-aafdcd3e3d08)
EXECUTE sofia/internal/60002 at 192.168.10.30 set(dialed_extension=1001)
2010-02-09 21:49:32.857126 [DEBUG] mod_dptools.c:768 sofia/internal/
60002 at 192.168.10.30 SET [dialed_extension]=[1001]
EXECUTE sofia/internal/60002 at 192.168.10.30 export(dialed_extension=1001)
2010-02-09 21:49:32.858075 [DEBUG] mod_dptools.c:851 EXPORT
[dialed_extension]=[1001]
EXECUTE sofia/internal/60002 at 192.168.10.30 bind_meta_app(1 b s
execute_extension::dx XML features)
2010-02-09 21:49:32.858075 [INFO] switch_ivr_async.c:2174 Bound B-Leg: 1
execute_extension::dx XML features
EXECUTE sofia/internal/60002 at 192.168.10.30 bind_meta_app(2 b s
record_session::/usr/local/freeswitch/recordings/60002.2010-02-09-21-49-32.wav)
2010-02-09 21:49:32.858075 [INFO] switch_ivr_async.c:2174 Bound B-Leg: 2
record_session::/usr/local/freeswitch/recordings/60002.2010-02-09-21-49-32.wav
EXECUTE sofia/internal/60002 at 192.168.10.30 bind_meta_app(3 b s
execute_extension::cf XML features)
2010-02-09 21:49:32.859074 [INFO] switch_ivr_async.c:2174 Bound B-Leg: 3
execute_extension::cf XML features
EXECUTE sofia/internal/60002 at 192.168.10.30 set(ringback=%(2000, 4000, 440.0,
480.0))
2010-02-09 21:49:32.859074 [DEBUG] mod_dptools.c:768 sofia/internal/
60002 at 192.168.10.30 SET [ringback]=[%(2000, 4000, 440.0, 480.0)]
EXECUTE sofia/internal/60002 at 192.168.10.30set(transfer_ringback=local_stream://moh)
2010-02-09 21:49:32.859074 [DEBUG] mod_dptools.c:768 sofia/internal/
60002 at 192.168.10.30 SET [transfer_ringback]=[local_stream://moh]
EXECUTE sofia/internal/60002 at 192.168.10.30 set(call_timeout=30)
2010-02-09 21:49:32.860074 [DEBUG] mod_dptools.c:768 sofia/internal/
60002 at 192.168.10.30 SET [call_timeout]=[30]
EXECUTE sofia/internal/60002 at 192.168.10.30 set(hangup_after_bridge=true)
2010-02-09 21:49:32.860074 [DEBUG] mod_dptools.c:768 sofia/internal/
60002 at 192.168.10.30 SET [hangup_after_bridge]=[true]
EXECUTE sofia/internal/60002 at 192.168.10.30 set(continue_on_fail=true)
2010-02-09 21:49:32.860074 [DEBUG] mod_dptools.c:768 sofia/internal/
60002 at 192.168.10.30 SET [continue_on_fail]=[true]
EXECUTE sofia/internal/60002 at 192.168.10.30db(insert/192.168.10.30-call_return/1001/60002)
EXECUTE sofia/internal/60002 at 192.168.10.30db(insert/192.168.10.30-last_dial_ext/1001/66610a8c-aec9-48ca-9b67-aafdcd3e3d08)
EXECUTE sofia/internal/60002 at 192.168.10.30 set(called_party_callgroup=)
2010-02-09 21:49:32.870074 [DEBUG] mod_dptools.c:768 sofia/internal/
60002 at 192.168.10.30 SET [called_party_callgroup]=[UNDEF]
EXECUTE sofia/internal/60002 at 192.168.10.30db(insert/192.168.10.30-last_dial//66610a8c-aec9-48ca-9b67-aafdcd3e3d08)
EXECUTE sofia/internal/60002 at 192.168.10.30 bridge(user/1001 at 192.168.10.30)
2010-02-09 21:49:32.905073 [DEBUG] switch_ivr_originate.c:1735 variable
string 0 = [presence_id=1001 at 192.168.10.30]
2010-02-09 21:49:32.905073 [NOTICE] switch_channel.c:613 New Channel
sofia/internal/sip:1001 at 192.168.10.192:41080[df7cf235-f0e4-406a-83a5-dd2d681bb278]
2010-02-09 21:49:32.905073 [DEBUG] mod_sofia.c:3142 (sofia/internal/
sip:1001 at 192.168.10.192:41080) State Change CS_NEW -> CS_INIT
2010-02-09 21:49:32.905073 [DEBUG] switch_core_session.c:999 Send signal
sofia/internal/sip:1001 at 192.168.10.192:41080 [BREAK]
2010-02-09 21:49:32.906075 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/sip:1001 at 192.168.10.192:41080) Running State Change CS_INIT
2010-02-09 21:49:32.906075 [DEBUG] switch_core_state_machine.c:338
(sofia/internal/sip:1001 at 192.168.10.192:41080) State INIT
2010-02-09 21:49:32.906075 [DEBUG] mod_sofia.c:83 sofia/internal/
sip:1001 at 192.168.10.192:41080 SOFIA INIT
2010-02-09 21:49:32.907184 [DEBUG] mod_sofia.c:111 (sofia/internal/
sip:1001 at 192.168.10.192:41080) State Change CS_INIT -> CS_ROUTING
2010-02-09 21:49:32.907184 [DEBUG] switch_core_session.c:999 Send signal
sofia/internal/sip:1001 at 192.168.10.192:41080 [BREAK]
2010-02-09 21:49:32.907184 [DEBUG] sofia.c:3727 Channel sofia/internal/
sip:1001 at 192.168.10.192:41080 entering state [calling][0]
2010-02-09 21:49:32.907184 [DEBUG] switch_core_state_machine.c:338
(sofia/internal/sip:1001 at 192.168.10.192:41080) State INIT going to sleep
2010-02-09 21:49:32.907184 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/sip:1001 at 192.168.10.192:41080) Running State Change
CS_ROUTING
2010-02-09 21:49:32.907184 [DEBUG] switch_core_state_machine.c:341
(sofia/internal/sip:1001 at 192.168.10.192:41080) State ROUTING
2010-02-09 21:49:32.907184 [DEBUG] mod_sofia.c:132 sofia/internal/
sip:1001 at 192.168.10.192:41080 SOFIA ROUTING
2010-02-09 21:49:32.907184 [DEBUG] switch_ivr_originate.c:66
(sofia/internal/sip:1001 at 192.168.10.192:41080) State Change CS_ROUTING ->
CS_CONSUME_MEDIA
2010-02-09 21:49:32.907184 [DEBUG] switch_core_session.c:999 Send signal
sofia/internal/sip:1001 at 192.168.10.192:41080 [BREAK]
2010-02-09 21:49:32.907184 [DEBUG] switch_core_state_machine.c:341
(sofia/internal/sip:1001 at 192.168.10.192:41080) State ROUTING going to sleep
2010-02-09 21:49:32.907184 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/sip:1001 at 192.168.10.192:41080) Running State Change
CS_CONSUME_MEDIA
2010-02-09 21:49:33.023071 [DEBUG] switch_core_state_machine.c:360
(sofia/internal/sip:1001 at 192.168.10.192:41080) State CONSUME_MEDIA
2010-02-09 21:49:33.023071 [DEBUG] switch_core_state_machine.c:360
(sofia/internal/sip:1001 at 192.168.10.192:41080) State CONSUME_MEDIA going to
sleep
2010-02-09 21:49:33.329319 [DEBUG] sofia.c:3727 Channel sofia/internal/
sip:1001 at 192.168.10.192:41080 entering state [terminated][488]
2010-02-09 21:49:33.329319 [NOTICE] sofia.c:4331 Hangup sofia/internal/
sip:1001 at 192.168.10.192:41080 [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION]
2010-02-09 21:49:33.329319 [DEBUG] switch_channel.c:1912 Send signal
sofia/internal/sip:1001 at 192.168.10.192:41080 [KILL]
2010-02-09 21:49:33.329319 [DEBUG] switch_core_session.c:999 Send signal
sofia/internal/sip:1001 at 192.168.10.192:41080 [BREAK]
2010-02-09 21:49:33.329319 [DEBUG] switch_core_state_machine.c:459
sofia/internal/sip:1001 at 192.168.10.192:41080 thread mismatch skipping state
handler.
2010-02-09 21:49:33.329319 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/sip:1001 at 192.168.10.192:41080) Running State Change
CS_HANGUP
2010-02-09 21:49:33.330075 [DEBUG] switch_core_state_machine.c:488
(sofia/internal/sip:1001 at 192.168.10.192:41080) State HANGUP
2010-02-09 21:49:33.330075 [DEBUG] mod_sofia.c:352 sofia/internal/
sip:1001 at 192.168.10.192:41080 Overriding SIP cause 488 with 488 from the
other leg
2010-02-09 21:49:33.330075 [DEBUG] mod_sofia.c:358 Channel sofia/internal/
sip:1001 at 192.168.10.192:41080 hanging up, cause: INCOMPATIBLE_DESTINATION
2010-02-09 21:49:33.330075 [DEBUG] switch_core_state_machine.c:46
sofia/internal/sip:1001 at 192.168.10.192:41080 Standard HANGUP, cause:
INCOMPATIBLE_DESTINATION
2010-02-09 21:49:33.330075 [DEBUG] switch_core_state_machine.c:488
(sofia/internal/sip:1001 at 192.168.10.192:41080) State HANGUP going to sleep
2010-02-09 21:49:33.330075 [DEBUG] switch_core_state_machine.c:333
(sofia/internal/sip:1001 at 192.168.10.192:41080) State Change CS_HANGUP ->
CS_REPORTING
2010-02-09 21:49:33.330075 [DEBUG] switch_core_session.c:999 Send signal
sofia/internal/sip:1001 at 192.168.10.192:41080 [BREAK]
2010-02-09 21:49:33.330075 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/sip:1001 at 192.168.10.192:41080) Running State Change
CS_REPORTING
2010-02-09 21:49:33.330075 [DEBUG] switch_core_state_machine.c:579
(sofia/internal/sip:1001 at 192.168.10.192:41080) State REPORTING
2010-02-09 21:49:33.330075 [DEBUG] switch_core_state_machine.c:53
sofia/internal/sip:1001 at 192.168.10.192:41080 Standard REPORTING, cause:
INCOMPATIBLE_DESTINATION
2010-02-09 21:49:33.330075 [DEBUG] switch_core_state_machine.c:579
(sofia/internal/sip:1001 at 192.168.10.192:41080) State REPORTING going to
sleep
2010-02-09 21:49:33.330075 [DEBUG] switch_core_state_machine.c:327
(sofia/internal/sip:1001 at 192.168.10.192:41080) State Change CS_REPORTING ->
CS_DESTROY
2010-02-09 21:49:33.330075 [DEBUG] switch_core_session.c:999 Send signal
sofia/internal/sip:1001 at 192.168.10.192:41080 [BREAK]
2010-02-09 21:49:33.330075 [DEBUG] switch_core_session.c:1136 Session 2
(sofia/internal/sip:1001 at 192.168.10.192:41080) Locked, Waiting on external
entities
2010-02-09 21:49:33.331072 [DEBUG] switch_ivr_originate.c:3009 Originate
Resulted in Error Cause: 88 [INCOMPATIBLE_DESTINATION]
2010-02-09 21:49:33.331072 [NOTICE] switch_core_session.c:1154 Session 2
(sofia/internal/sip:1001 at 192.168.10.192:41080) Ended
2010-02-09 21:49:33.331072 [NOTICE] switch_core_session.c:1156 Close Channel
sofia/internal/sip:1001 at 192.168.10.192:41080 [CS_DESTROY]
2010-02-09 21:49:33.331072 [DEBUG] switch_core_state_machine.c:423
(sofia/internal/sip:1001 at 192.168.10.192:41080) Running State Change
CS_DESTROY
2010-02-09 21:49:33.331072 [DEBUG] switch_core_state_machine.c:434
(sofia/internal/sip:1001 at 192.168.10.192:41080) State DESTROY
2010-02-09 21:49:33.331072 [DEBUG] mod_sofia.c:293 sofia/internal/
sip:1001 at 192.168.10.192:41080 SOFIA DESTROY
2010-02-09 21:49:33.331072 [DEBUG] switch_core_state_machine.c:60
sofia/internal/sip:1001 at 192.168.10.192:41080 Standard DESTROY
2010-02-09 21:49:33.331072 [DEBUG] switch_core_state_machine.c:434
(sofia/internal/sip:1001 at 192.168.10.192:41080) State DESTROY going to sleep
2010-02-09 21:49:33.331072 [ERR] switch_ivr_originate.c:2249 Cannot create
outgoing channel of type [user] cause: [INCOMPATIBLE_DESTINATION]
2010-02-09 21:49:33.331072 [DEBUG] switch_ivr_originate.c:3009 Originate
Resulted in Error Cause: 88 [INCOMPATIBLE_DESTINATION]
2010-02-09 21:49:33.331072 [INFO] mod_dptools.c:2294 Originate Failed.
Cause: INCOMPATIBLE_DESTINATION
EXECUTE sofia/internal/60002 at 192.168.10.30 answer()
2010-02-09 21:49:33.332075 [DEBUG] mod_dptools.c:658 sofia/internal/
60002 at 192.168.10.30 receive message [ANSWER]
2010-02-09 21:49:33.332075 [DEBUG] sofia_glue.c:2381 AUDIO RTP
[sofia/internal/60002 at 192.168.10.30] 192.168.10.30 port 27634 ->
192.168.10.131 port 25592 codec: 9 ms: 20
2010-02-09 21:49:33.332075 [DEBUG] switch_rtp.c:1167 Starting timer [soft]
160 bytes per 20ms
2010-02-09 21:49:33.333080 [DEBUG] mod_sofia.c:571 Local SDP sofia/internal/
60002 at 192.168.10.30:
v=0
o=FreeSWITCH 1265704739 1265704740 IN IP4 192.168.10.30
s=FreeSWITCH
c=IN IP4 192.168.10.30
t=0 0
m=audio 27634 RTP/AVP 9 101
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

2010-02-09 21:49:33.333080 [DEBUG] sofia.c:3727 Channel sofia/internal/
60002 at 192.168.10.30 entering state [completed][200]
2010-02-09 21:49:33.333080 [DEBUG] switch_core_session.c:645 Send signal
sofia/internal/60002 at 192.168.10.30 [BREAK]
2010-02-09 21:49:33.333080 [NOTICE] mod_dptools.c:658 Channel
[sofia/internal/60002 at 192.168.10.30] has been answered
2010-02-09 21:49:33.333080 [DEBUG] switch_channel.c:182 sofia/internal/
60002 at 192.168.10.30 receive message [AUDIO_SYNC]
EXECUTE sofia/internal/60002 at 192.168.10.30 sleep(1000)
2010-02-09 21:49:33.334081 [DEBUG] switch_channel.c:182 sofia/internal/
60002 at 192.168.10.30 receive message [AUDIO_SYNC]
2010-02-09 21:49:33.395069 [DEBUG] switch_rtp.c:2004 Correct ip/port
confirmed.
2010-02-09 21:49:33.455069 [DEBUG] sofia.c:3727 Channel sofia/internal/
60002 at 192.168.10.30 entering state [ready][200]
EXECUTE sofia/internal/60002 at 192.168.10.30 voicemail(default 192.168.10.30
1001)
2010-02-09 21:49:34.335070 [DEBUG] mod_voicemail.c:730 [default] rwlock
2010-02-09 21:49:34.370083 [DEBUG] switch_channel.c:182 sofia/internal/
60002 at 192.168.10.30 receive message [AUDIO_SYNC]
2010-02-09 21:49:34.475070 [DEBUG] switch_ivr_play_say.c:118 No language
specified - Using [en]
2010-02-09 21:49:34.475070 [DEBUG] switch_ivr_play_say.c:273 Handle
play-file:[voicemail/vm-person.wav] (en:en)
2010-02-09 21:49:34.523070 [WARNING] switch_core_file.c:177 Sample rate
doesn't match
2010-02-09 21:49:34.524073 [DEBUG] switch_ivr_play_say.c:1154 Codec
Activated L16 at 16000hz 1 channels 20ms
2010-02-09 21:49:34.526072 [DEBUG] switch_core_io.c:652 sofia/internal/
60002 at 192.168.10.30 receive message [TRANSCODING_NECESSARY]
2010-02-09 21:49:35.875063 [DEBUG] switch_ivr_play_say.c:1446 done playing
file
2010-02-09 21:49:35.995062 [DEBUG] switch_ivr_play_say.c:273 Handle
say:[1001] (en:en)
2010-02-09 21:49:36.027062 [WARNING] switch_core_file.c:177 Sample rate
doesn't match
2010-02-09 21:49:36.027062 [DEBUG] switch_ivr_play_say.c:1154 Codec
Activated L16 at 16000hz 1 channels 20ms
2010-02-09 21:49:36.028065 [DEBUG] switch_core_io.c:652 sofia/internal/
60002 at 192.168.10.30 receive message [TRANSCODING_NECESSARY]
2010-02-09 21:49:36.455072 [DEBUG] switch_ivr_play_say.c:1446 done playing
file
2010-02-09 21:49:36.460061 [WARNING] switch_core_file.c:177 Sample rate
doesn't match
2010-02-09 21:49:36.460061 [DEBUG] switch_ivr_play_say.c:1154 Codec
Activated L16 at 16000hz 1 channels 20ms
2010-02-09 21:49:36.461074 [DEBUG] switch_core_io.c:652 sofia/internal/
60002 at 192.168.10.30 receive message [TRANSCODING_NECESSARY]
2010-02-09 21:49:36.995060 [DEBUG] switch_ivr_play_say.c:1446 done playing
file
2010-02-09 21:49:36.995060 [WARNING] switch_core_file.c:177 Sample rate
doesn't match
2010-02-09 21:49:36.995060 [DEBUG] switch_ivr_play_say.c:1154 Codec
Activated L16 at 16000hz 1 channels 20ms
2010-02-09 21:49:36.995060 [DEBUG] switch_core_io.c:652 sofia/internal/
60002 at 192.168.10.30 receive message [TRANSCODING_NECESSARY]
2010-02-09 21:49:37.535061 [DEBUG] switch_ivr_play_say.c:1446 done playing
file
2010-02-09 21:49:37.535061 [WARNING] switch_core_file.c:177 Sample rate
doesn't match
2010-02-09 21:49:37.535061 [DEBUG] switch_ivr_play_say.c:1154 Codec
Activated L16 at 16000hz 1 channels 20ms
2010-02-09 21:49:37.535061 [DEBUG] switch_core_io.c:652 sofia/internal/
60002 at 192.168.10.30 receive message [TRANSCODING_NECESSARY]
2010-02-09 21:49:37.995060 [DEBUG] switch_ivr_play_say.c:1446 done playing
file
2010-02-09 21:49:38.095060 [DEBUG] switch_ivr_play_say.c:273 Handle
play-file:[voicemail/vm-not_available.wav] (en:en)
2010-02-09 21:49:38.118060 [WARNING] switch_core_file.c:177 Sample rate
doesn't match
2010-02-09 21:49:38.119063 [DEBUG] switch_ivr_play_say.c:1154 Codec
Activated L16 at 16000hz 1 channels 20ms
2010-02-09 21:49:38.120062 [DEBUG] switch_core_io.c:652 sofia/internal/
60002 at 192.168.10.30 receive message [TRANSCODING_NECESSARY]
2010-02-09 21:49:39.075057 [DEBUG] switch_ivr_play_say.c:1446 done playing
file
2010-02-09 21:49:39.175057 [DEBUG] switch_ivr_play_say.c:118 No language
specified - Using [en]
2010-02-09 21:49:39.177086 [DEBUG] switch_ivr_play_say.c:273 Handle
play-file:[voicemail/vm-record_message.wav] (en:en)
2010-02-09 21:49:39.181967 [WARNING] switch_core_file.c:177 Sample rate
doesn't match
2010-02-09 21:49:39.181967 [DEBUG] switch_ivr_play_say.c:1154 Codec
Activated L16 at 16000hz 1 channels 20ms
2010-02-09 21:49:39.192068 [DEBUG] switch_core_io.c:652 sofia/internal/
60002 at 192.168.10.30 receive message [TRANSCODING_NECESSARY]
2010-02-09 21:49:40.999151 [NOTICE] sofia.c:329 Hangup sofia/internal/
60002 at 192.168.10.30 [CS_EXECUTE] [NORMAL_CLEARING]
2010-02-09 21:49:40.999151 [DEBUG] switch_channel.c:1912 Send signal
sofia/internal/60002 at 192.168.10.30 [KILL]
2010-02-09 21:49:40.999151 [DEBUG] switch_core_session.c:999 Send signal
sofia/internal/60002 at 192.168.10.30 [BREAK]
2010-02-09 21:49:40.999151 [DEBUG] switch_core_state_machine.c:459
sofia/internal/60002 at 192.168.10.30 thread mismatch skipping state handler.
2010-02-09 21:49:41.015049 [DEBUG] switch_ivr_play_say.c:1446 done playing
file
2010-02-09 21:49:41.114049 [DEBUG] switch_core_state_machine.c:348
(sofia/internal/60002 at 192.168.10.30) State EXECUTE going to sleep
2010-02-09 21:49:41.114049 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/60002 at 192.168.10.30) Running State Change CS_HANGUP
2010-02-09 21:49:41.114049 [DEBUG] switch_core_state_machine.c:488
(sofia/internal/60002 at 192.168.10.30) State HANGUP
2010-02-09 21:49:41.114049 [DEBUG] mod_sofia.c:352 sofia/internal/
60002 at 192.168.10.30 Overriding SIP cause 480 with 488 from the other leg
2010-02-09 21:49:41.114049 [DEBUG] mod_sofia.c:358 Channel sofia/internal/
60002 at 192.168.10.30 hanging up, cause: NORMAL_CLEARING
2010-02-09 21:49:41.114049 [DEBUG] switch_core_state_machine.c:46
sofia/internal/60002 at 192.168.10.30 Standard HANGUP, cause: NORMAL_CLEARING
2010-02-09 21:49:41.114049 [DEBUG] switch_core_state_machine.c:488
(sofia/internal/60002 at 192.168.10.30) State HANGUP going to sleep
2010-02-09 21:49:41.114049 [DEBUG] switch_core_state_machine.c:333
(sofia/internal/60002 at 192.168.10.30) State Change CS_HANGUP -> CS_REPORTING
2010-02-09 21:49:41.114049 [DEBUG] switch_core_session.c:999 Send signal
sofia/internal/60002 at 192.168.10.30 [BREAK]
2010-02-09 21:49:41.114049 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/60002 at 192.168.10.30) Running State Change CS_REPORTING
2010-02-09 21:49:41.114049 [DEBUG] switch_core_state_machine.c:579
(sofia/internal/60002 at 192.168.10.30) State REPORTING
2010-02-09 21:49:41.129049 [DEBUG] switch_core_state_machine.c:53
sofia/internal/60002 at 192.168.10.30 Standard REPORTING, cause:
NORMAL_CLEARING
2010-02-09 21:49:41.129049 [DEBUG] switch_core_state_machine.c:579
(sofia/internal/60002 at 192.168.10.30) State REPORTING going to sleep
2010-02-09 21:49:41.129049 [DEBUG] switch_core_state_machine.c:327
(sofia/internal/60002 at 192.168.10.30) State Change CS_REPORTING -> CS_DESTROY
2010-02-09 21:49:41.129049 [DEBUG] switch_core_session.c:999 Send signal
sofia/internal/60002 at 192.168.10.30 [BREAK]
2010-02-09 21:49:41.129049 [DEBUG] switch_core_session.c:1136 Session 1
(sofia/internal/60002 at 192.168.10.30) Locked, Waiting on external entities
2010-02-09 21:49:41.129049 [NOTICE] switch_core_session.c:1154 Session 1
(sofia/internal/60002 at 192.168.10.30) Ended
2010-02-09 21:49:41.129049 [NOTICE] switch_core_session.c:1156 Close Channel
sofia/internal/60002 at 192.168.10.30 [CS_DESTROY]
2010-02-09 21:49:41.129049 [DEBUG] switch_core_state_machine.c:423
(sofia/internal/60002 at 192.168.10.30) Running State Change CS_DESTROY
2010-02-09 21:49:41.129049 [DEBUG] switch_core_state_machine.c:434
(sofia/internal/60002 at 192.168.10.30) State DESTROY
2010-02-09 21:49:41.129049 [DEBUG] mod_sofia.c:293 sofia/internal/
60002 at 192.168.10.30 SOFIA DESTROY
2010-02-09 21:49:41.129049 [DEBUG] switch_core_state_machine.c:60
sofia/internal/60002 at 192.168.10.30 Standard DESTROY
2010-02-09 21:49:41.129049 [DEBUG] switch_core_state_machine.c:434
(sofia/internal/60002 at 192.168.10.30) State DESTROY going to sleep






On 9 February 2010 21:18, Brian West <brian at freeswitch.org> wrote:

> What you're saying makes little or no sense to me even on 1.0.4, Can you
> pastebin your logs?
>
> /b
>
> On Feb 9, 2010, at 3:15 PM, Bruce Hopkins wrote:
>
> > OK, many thanks for the extremely swift response Brian.
> >
> > I will try to get up and running as soon as I can with 1.0.5 and see if
> the issue goes away.
> >
> > thanks again
> > Bruce
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100209/f036726e/attachment-0001.html 

From brian at freeswitch.org  Tue Feb  9 14:00:33 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 9 Feb 2010 16:00:33 -0600
Subject: [Freeswitch-users] g.722 --> SPEEX/16000 or SPEEX/32000
	transcoding issue
In-Reply-To: <d03a10e01002091355n1934627t3902e0d8af5c7dc3@mail.gmail.com>
References: <d03a10e01002091203qde4caabi1a7565929e60918f@mail.gmail.com>
	<03A2F6D0-754A-4067-A106-9D91B0D68A92@freeswitch.org>
	<d03a10e01002091315n5fa9242eoe2a86c4edee1501d@mail.gmail.com>
	<F167991C-151B-4D40-BF27-80A4C3A44D1E@freeswitch.org>
	<d03a10e01002091355n1934627t3902e0d8af5c7dc3@mail.gmail.com>
Message-ID: <51A80F87-F2B9-477E-8D14-99C40E24413D@freeswitch.org>

You need to allow SPEEX at 8000h,SPEEX at 16000h,SPEEX at 32000h

/b

On Feb 9, 2010, at 3:55 PM, Bruce Hopkins wrote:

> Willdo,
> 
> To clarify in brief though, the scenario which occurs and causes the call to fail is:
> 
> SIP client  1 (g.722 enabled only ) -----> INVITE (with SDP offer: Media Attribute (a): rtpmap:9 g722/8000 ) -->FreeSWITCH
> 
> ---> INVITE (with SDP offer including a bunch of codecs including rtpmap: rtpmap:98 SPEEX/8000 but crucially not including SPEEX/16000 or SPEEX/32000)
> 
> ---> SIP client 2 (with only SPEEX/16000 or SPEEX/32000 enabled).
> 
> The second SIP client does not get offered a codec it can accept, so SIP client 1 is sent a method 488 "Not Acceptable Here" message and the calling party gets directed to the voicemail for the other SIP client.
> 
> By contrast, there is no problem calling SPEEX/32000 --> SPEEX/32000 or calling SPEEX/16000 --> SPEEX/16000.
> 
> there is also no problem calling SPEEX/32000 --> g.722/8000.
> 
> I am wondering if the problem is that FreeSWITCH is interpreting g.722 as being a narrowband (8kHz sample rate) codec, due to the historic anomaly of it presenting g722/8000 in the SDP even though it in fact uses 16kHz sampling, and for that reason not wanting to offer a 16kHz sample rate codec to the second SIP client?
> 
> I suggest this as I also found trying to call alaw --> SPEEX/16000 does not work, for example.



From jbrucehopkins at gmail.com  Tue Feb  9 14:04:51 2010
From: jbrucehopkins at gmail.com (Bruce Hopkins)
Date: Tue, 9 Feb 2010 22:04:51 +0000
Subject: [Freeswitch-users] g.722 --> SPEEX/16000 or SPEEX/32000
	transcoding issue
In-Reply-To: <d03a10e01002091355n1934627t3902e0d8af5c7dc3@mail.gmail.com>
References: <d03a10e01002091203qde4caabi1a7565929e60918f@mail.gmail.com>
	<03A2F6D0-754A-4067-A106-9D91B0D68A92@freeswitch.org>
	<d03a10e01002091315n5fa9242eoe2a86c4edee1501d@mail.gmail.com>
	<F167991C-151B-4D40-BF27-80A4C3A44D1E@freeswitch.org>
	<d03a10e01002091355n1934627t3902e0d8af5c7dc3@mail.gmail.com>
Message-ID: <d03a10e01002091404i5e7aefaam7235bef68828d42b@mail.gmail.com>

I think I see the problem:

In the log is the following:

2010-02-09 21:49:36.995060 [WARNING] switch_core_file.c:177 Sample rate
doesn't match

However, surely this is incorrect, as g.722 and SPEEX/16000 in fact both do
use the same sample rate of 16kHz

g.722 might not look like it from the SDP which announces g722/8000 - but
this is a historical error in the RFC.

Bruce

On 9 February 2010 21:55, Bruce Hopkins <jbrucehopkins at gmail.com> wrote:

> Willdo,
>
> To clarify in brief though, the scenario which occurs and causes the call
> to fail is:
>
> SIP client  1 (g.722 enabled only ) -----> INVITE (with SDP offer: Media
> Attribute (a): rtpmap:9 g722/8000 ) -->FreeSWITCH
>
> ---> INVITE (with SDP offer including a bunch of codecs including rtpmap:
> rtpmap:98 SPEEX/8000 but crucially *not* including SPEEX/16000 or
> SPEEX/32000)
>
> ---> SIP client 2 (with only SPEEX/16000 or SPEEX/32000 enabled).
>
> The second SIP client does not get offered a codec it can accept, so SIP
> client 1 is sent a method 488 "Not Acceptable Here" message and the calling
> party gets directed to the voicemail for the other SIP client.
>
> By contrast, there is no problem calling SPEEX/32000 --> SPEEX/32000 or
> calling SPEEX/16000 --> SPEEX/16000.
>
> there is also no problem calling SPEEX/32000 --> g.722/8000.
>
> I am wondering if the problem is that FreeSWITCH is interpreting g.722 as
> being a narrowband (8kHz sample rate) codec, due to the historic anomaly of
> it presenting g722/8000 in the SDP even though it in fact uses 16kHz
> sampling, and for that reason not wanting to offer a 16kHz sample rate codec
> to the second SIP client?
>
> I suggest this as I also found trying to call alaw --> SPEEX/16000 does not
> work, for example.
>
>
>
> Here is the log file for the scenario which does not work (g.722 client
> trying to call Speex wideband client).  Please let me know if a Wireshark
> trace would be helpful.
>
> 2010-02-09 21:49:32.784387 [DEBUG] sofia.c:5224 0 acls to check for proxy
> 2010-02-09 21:49:32.784387 [DEBUG] sofia.c:5242 network ip is a proxy [0]
> 2010-02-09 21:49:32.784387 [DEBUG] sofia.c:5270 IP 192.168.10.131 Rejected
> by acl "domains". Falling back to Digest auth.
> 2010-02-09 21:49:32.791074 [DEBUG] sofia.c:5224 0 acls to check for proxy
> 2010-02-09 21:49:32.791074 [DEBUG] sofia.c:5242 network ip is a proxy [0]
> 2010-02-09 21:49:32.791074 [DEBUG] sofia.c:5270 IP 192.168.10.131 Rejected
> by acl "domains". Falling back to Digest auth.
> 2010-02-09 21:49:32.794074 [NOTICE] switch_channel.c:613 New Channel
> sofia/internal/60002 at 192.168.10.30 [66610a8c-aec9-48ca-9b67-aafdcd3e3d08]
> 2010-02-09 21:49:32.794074 [DEBUG] sofia.c:3727 Channel sofia/internal/
> 60002 at 192.168.10.30 entering state [received][100]
> 2010-02-09 21:49:32.794074 [DEBUG] sofia.c:3738 Remote SDP:
> v=0
> o=- 5 2 IN IP4 192.168.10.131
> s=CounterPath eyeBeam 1.5
> c=IN IP4 192.168.10.131
> t=0 0
> m=audio 25592 RTP/AVP 9 101
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
>
> 2010-02-09 21:49:32.794074 [DEBUG] sofia_glue.c:3305 Audio Codec Compare
> [G722:9:8000:20]/[G722:9:8000:20]
> 2010-02-09 21:49:32.795079 [DEBUG] sofia_glue.c:2143 Set Codec
> sofia/internal/60002 at 192.168.10.30 G722/8000 20 ms 160 samples
> 2010-02-09 21:49:32.795079 [DEBUG] sofia_glue.c:3261 Set 2833 dtmf payload
> to 101
> 2010-02-09 21:49:32.795079 [DEBUG] sofia.c:3885 (sofia/internal/
> 60002 at 192.168.10.30) State Change CS_NEW -> CS_INIT
> 2010-02-09 21:49:32.795079 [DEBUG] switch_core_session.c:999 Send signal
> sofia/internal/60002 at 192.168.10.30 [BREAK]
> 2010-02-09 21:49:32.795079 [DEBUG] switch_core_state_machine.c:314
> (sofia/internal/60002 at 192.168.10.30) Running State Change CS_INIT
> 2010-02-09 21:49:32.795079 [DEBUG] switch_core_state_machine.c:338
> (sofia/internal/60002 at 192.168.10.30) State INIT
> 2010-02-09 21:49:32.795079 [DEBUG] mod_sofia.c:83 sofia/internal/
> 60002 at 192.168.10.30 SOFIA INIT
> 2010-02-09 21:49:32.795079 [DEBUG] mod_sofia.c:111 (sofia/internal/
> 60002 at 192.168.10.30) State Change CS_INIT -> CS_ROUTING
> 2010-02-09 21:49:32.795079 [DEBUG] switch_core_session.c:999 Send signal
> sofia/internal/60002 at 192.168.10.30 [BREAK]
> 2010-02-09 21:49:32.795079 [DEBUG] switch_core_state_machine.c:338
> (sofia/internal/60002 at 192.168.10.30) State INIT going to sleep
> 2010-02-09 21:49:32.795079 [DEBUG] switch_core_state_machine.c:314
> (sofia/internal/60002 at 192.168.10.30) Running State Change CS_ROUTING
> 2010-02-09 21:49:32.795079 [DEBUG] switch_core_state_machine.c:341
> (sofia/internal/60002 at 192.168.10.30) State ROUTING
> 2010-02-09 21:49:32.795079 [DEBUG] mod_sofia.c:132 sofia/internal/
> 60002 at 192.168.10.30 SOFIA ROUTING
> 2010-02-09 21:49:32.795079 [DEBUG] switch_core_state_machine.c:78
> sofia/internal/60002 at 192.168.10.30 Standard ROUTING
> 2010-02-09 21:49:32.795079 [INFO] mod_dialplan_xml.c:408 Processing
> 60002->1001 in context default
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->unloop]
> continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (PASS) [unloop]
> ${unroll_loops}(true) =~ /^true$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [unloop]
> ${sip_looped_call}() =~ /^true$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
> [default->tod_example] continue=true
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (PASS) [tod_example]
> ${strftime(%w)}(2) =~ /^([1-5])$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [tod_example]
> ${strftime(%H%M)}(2149) =~ /^((09|1[0-7])[0-5][0-9]|1800)$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
> [default->global-intercept] continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL)
> [global-intercept] destination_number(1001) =~ /^\*886$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
> [default->group-intercept] continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL)
> [group-intercept] destination_number(1001) =~ /^\*8$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
> [default->intercept-ext] continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [intercept-ext]
> destination_number(1001) =~ /^\*\*(\d+)$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->redial]
> continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [redial]
> destination_number(1001) =~ /^\*870$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->global]
> continue=true
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [global]
> ${network_addr}(192.168.10.131) =~ /^$/ break=never
> Dialplan: sofia/internal/60002 at 192.168.10.30 ANTI-Action
> set(use_profile=${cond(${acl(${network_addr} rfc1918)} == true ? nat :
> default)})
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (PASS) [global]
> ${numbering_plan}() =~ /^$/ break=never
> Dialplan: sofia/internal/60002 at 192.168.10.30 Action set_user(default@${domain_name})
>
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [global]
> ${call_debug}(false) =~ /^true$/ break=never
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [global]
> ${sip_has_crypto}() =~ /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/
> break=never
> Dialplan: sofia/internal/60002 at 192.168.10.30 Absolute Condition [global]
> Dialplan: sofia/internal/60002 at 192.168.10.30 Action
> db(insert/${domain_name}-spymap/${caller_id_number}/${uuid})
> Dialplan: sofia/internal/60002 at 192.168.10.30 Action
> db(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number})
>
> Dialplan: sofia/internal/60002 at 192.168.10.30 Action
> db(insert/${domain_name}-last_dial/global/${uuid})
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
> [default->snom-demo-2] continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [snom-demo-2]
> destination_number(1001) =~ /^\*9001$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
> [default->snom-demo-1] continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [snom-demo-1]
> destination_number(1001) =~ /^\*9000$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->eavesdrop]
> continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [eavesdrop]
> destination_number(1001) =~ /^\*88(.*)$|^\*0(.*)$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->eavesdrop]
> continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [eavesdrop]
> destination_number(1001) =~ /^\*779$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
> [default->call_privacy] continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [call_privacy]
> destination_number(1001) =~ /^\*67(\d+)$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
> [default->call_return] continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [call_return]
> destination_number(1001) =~ /^\*69$|^869$|^lcr$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->del-group]
> continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [del-group]
> destination_number(1001) =~ /^\*80(\d{2})$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->add-group]
> continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [add-group]
> destination_number(1001) =~ /^\*81(\d{2})$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
> [default->call-group-simo] continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL)
> [call-group-simo] destination_number(1001) =~ /^\*82(\d{2})$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
> [default->call-group-order] continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL)
> [call-group-order] destination_number(1001) =~ /^\*83(\d{2})$/
> break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
> [default->extension-intercom] continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL)
> [extension-intercom] destination_number(1001) =~ /^\*8(\d{4})$/
> break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
> [default->send_to_voicemail_5digits] continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL)
> [send_to_voicemail_5digits] destination_number(1001) =~ /^\*99(\d{5})$/
> break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
> [default->send_to_voicemail_4digits] continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL)
> [send_to_voicemail_4digits] destination_number(1001) =~ /^\*99(\d{4})$/
> break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
> [default->send_to_voicemail_3digits] continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL)
> [send_to_voicemail_3digits] destination_number(1001) =~ /^\*99(\d{3})$/
> break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->pizza_demo]
> continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [pizza_demo]
> destination_number(1001) =~ /^(pizza|74992)$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->2001]
> continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [2001]
> destination_number(1001) =~ /^2001$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->Call via
> asterisk-pbx1] continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [Call via
> asterisk-pbx1] destination_number(1001) =~ /269065/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->pizza_demo]
> continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [pizza_demo]
> destination_number(1001) =~ /^(pizza|74992)$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->5002]
> continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [5002]
> destination_number(1001) =~ /^5002$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->7002]
> continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [7002]
> destination_number(1001) =~ /^7002$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->DISA]
> continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [DISA]
> destination_number(1001) =~ /^\*(3472)$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->Recordings]
> continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [Recordings]
> destination_number(1001) =~ /^\*(732673)$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->7002.park]
> continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [7002.park]
> destination_number(1001) =~ /^\*7002$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
> [default->group_dial_sales] continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL)
> [group_dial_sales] destination_number(1001) =~ /^\*2000$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
> [default->group_dial_support] continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL)
> [group_dial_support] destination_number(1001) =~ /^\*2001$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
> [default->group_dial_billing] continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL)
> [group_dial_billing] destination_number(1001) =~ /^\*2002$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->vmain2]
> continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [vmain2]
> destination_number(1001) =~ /^vmain2$|^\*97$|^\*4000$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->vmain]
> continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [vmain]
> destination_number(1001) =~ /^vmain$|^\*98$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->sip_uri]
> continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [sip_uri]
> destination_number(1001) =~ /^sip:(.*)$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
> [default->nb_conferences] continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [nb_conferences]
> destination_number(1001) =~ /^\*(30\d{2})$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
> [default->wb_conferences] continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [wb_conferences]
> destination_number(1001) =~ /^\*(31\d{2})$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
> [default->uwb_conferences] continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL)
> [uwb_conferences] destination_number(1001) =~ /^\*(32\d{2})$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
> [default->cdquality_conferences] continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL)
> [cdquality_conferences] destination_number(1001) =~ /^\*(33\d{2})$/
> break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
> [default->freeswitch_public_conf_via_sip] continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL)
> [freeswitch_public_conf_via_sip] destination_number(1001) =~
> /^\*9(888|1616|3232)$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
> [default->mad_boss_intercom] continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL)
> [mad_boss_intercom] destination_number(1001) =~ /^\*0911$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
> [default->mad_boss_intercom] continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL)
> [mad_boss_intercom] destination_number(1001) =~ /^\*0912$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->mad_boss]
> continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [mad_boss]
> destination_number(1001) =~ /^\*0913$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->ivr_demo]
> continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [ivr_demo]
> destination_number(1001) =~ /^\*5000$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
> [default->dynamic_conference] continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL)
> [dynamic_conference] destination_number(1001) =~ /^\*5001$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
> [default->rtp_multicast_page] continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL)
> [rtp_multicast_page] destination_number(1001) =~ /^\*pagegroup$|^\*7243/
> break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->park]
> continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [park]
> destination_number(1001) =~ /^\*5900$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->unpark]
> continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [unpark]
> destination_number(1001) =~ /^\*5901$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->park]
> continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (PASS) [park]
> source(mod_sofia) =~ /mod_sofia/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [park]
> destination_number(1001) =~ /park\+(\d+)/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->unpark]
> continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (PASS) [unpark]
> source(mod_sofia) =~ /mod_sofia/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [unpark]
> destination_number(1001) =~ /^parking$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->park]
> continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (PASS) [park]
> source(mod_sofia) =~ /mod_sofia/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [park]
> destination_number(1001) =~ /callpark/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->unpark]
> continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (PASS) [unpark]
> source(mod_sofia) =~ /mod_sofia/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [unpark]
> destination_number(1001) =~ /pickup/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->wait]
> continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [wait]
> destination_number(1001) =~ /^wait$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
> [default->fax_receive] continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [fax_receive]
> destination_number(1001) =~ /^\*9978$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
> [default->fax_transmit] continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [fax_transmit]
> destination_number(1001) =~ /^\*9979$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
> [default->ringback_180] continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [ringback_180]
> destination_number(1001) =~ /^\*9980$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
> [default->ringback_183_uk_ring] continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL)
> [ringback_183_uk_ring] destination_number(1001) =~ /^\*9981$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
> [default->ringback_183_music_ring] continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL)
> [ringback_183_music_ring] destination_number(1001) =~ /^\*9982$/
> break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
> [default->ringback_post_answer_uk_ring] continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL)
> [ringback_post_answer_uk_ring] destination_number(1001) =~ /^\*9983$/
> break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
> [default->ringback_post_answer_music] continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL)
> [ringback_post_answer_music] destination_number(1001) =~ /^\*9984$/
> break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->ClueCon]
> continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [ClueCon]
> destination_number(1001) =~ /^\*9991$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->show_info]
> continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [show_info]
> destination_number(1001) =~ /^\*9992$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
> [default->video_record] continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [video_record]
> destination_number(1001) =~ /^\*9993$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
> [default->video_playback] continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [video_playback]
> destination_number(1001) =~ /^\*9994$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->delay_echo]
> continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [delay_echo]
> destination_number(1001) =~ /^\*9995$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->echo]
> continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [echo]
> destination_number(1001) =~ /^\*9996$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->milliwatt]
> continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [milliwatt]
> destination_number(1001) =~ /^\*9997$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
> [default->tone_stream] continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [tone_stream]
> destination_number(1001) =~ /^\*9998$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
> [default->zrtp_enrollement] continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL)
> [zrtp_enrollement] destination_number(1001) =~ /^\*9787$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->hold_music]
> continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [hold_music]
> destination_number(1001) =~ /^\*9999$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
> [default->Local_Extension] continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (PASS)
> [Local_Extension] destination_number(1001) =~
> /(^\d{6}$|\d{5}$|^\d{4}$|^\d{3}$)/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Action
> set(dialed_extension=1001)
> Dialplan: sofia/internal/60002 at 192.168.10.30 Action
> export(dialed_extension=1001)
> Dialplan: sofia/internal/60002 at 192.168.10.30 Action bind_meta_app(1 b s
> execute_extension::dx XML features)
> Dialplan: sofia/internal/60002 at 192.168.10.30 Action bind_meta_app(2 b s
> record_session::/usr/local/freeswitch/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav)
>
> Dialplan: sofia/internal/60002 at 192.168.10.30 Action bind_meta_app(3 b s
> execute_extension::cf XML features)
> Dialplan: sofia/internal/60002 at 192.168.10.30 Action
> set(ringback=${us-ring})
> Dialplan: sofia/internal/60002 at 192.168.10.30 Action
> set(transfer_ringback=local_stream://moh)
> Dialplan: sofia/internal/60002 at 192.168.10.30 Action set(call_timeout=30)
> Dialplan: sofia/internal/60002 at 192.168.10.30 Action
> set(hangup_after_bridge=true)
> Dialplan: sofia/internal/60002 at 192.168.10.30 Action
> set(continue_on_fail=true)
> Dialplan: sofia/internal/60002 at 192.168.10.30 Action
> db(insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number})
>
> Dialplan: sofia/internal/60002 at 192.168.10.30 Action
> db(insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid})
> Dialplan: sofia/internal/60002 at 192.168.10.30 Action
> set(called_party_callgroup=${user_data(${dialed_extension}@${domain_name}
> var callgroup)})
> Dialplan: sofia/internal/60002 at 192.168.10.30 Action
> db(insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid})
> Dialplan: sofia/internal/60002 at 192.168.10.30 Action
> bridge(user/${dialed_extension}@${domain_name})
> Dialplan: sofia/internal/60002 at 192.168.10.30 Action answer()
> Dialplan: sofia/internal/60002 at 192.168.10.30 Action sleep(1000)
> Dialplan: sofia/internal/60002 at 192.168.10.30 Action voicemail(default
> ${domain_name} ${dialed_extension})
> 2010-02-09 21:49:32.798073 [DEBUG] switch_core_state_machine.c:122
> (sofia/internal/60002 at 192.168.10.30) State Change CS_ROUTING -> CS_EXECUTE
> 2010-02-09 21:49:32.798073 [DEBUG] switch_core_session.c:999 Send signal
> sofia/internal/60002 at 192.168.10.30 [BREAK]
> 2010-02-09 21:49:32.798073 [DEBUG] switch_core_state_machine.c:341
> (sofia/internal/60002 at 192.168.10.30) State ROUTING going to sleep
> 2010-02-09 21:49:32.798073 [DEBUG] switch_core_state_machine.c:314
> (sofia/internal/60002 at 192.168.10.30) Running State Change CS_EXECUTE
> 2010-02-09 21:49:32.798073 [DEBUG] switch_core_state_machine.c:348
> (sofia/internal/60002 at 192.168.10.30) State EXECUTE
> 2010-02-09 21:49:32.798073 [DEBUG] mod_sofia.c:181 sofia/internal/
> 60002 at 192.168.10.30 SOFIA EXECUTE
> 2010-02-09 21:49:32.798073 [DEBUG] switch_core_state_machine.c:159
> sofia/internal/60002 at 192.168.10.30 Standard EXECUTE
> EXECUTE sofia/internal/60002 at 192.168.10.30 set(use_profile=default)
> 2010-02-09 21:49:32.798073 [DEBUG] mod_dptools.c:768 sofia/internal/
> 60002 at 192.168.10.30 SET [use_profile]=[default]
> EXECUTE sofia/internal/60002 at 192.168.10.30 set_user(default at 192.168.10.30)
> EXECUTE sofia/internal/60002 at 192.168.10.30db(insert/192.168.10.30-spymap/60002/66610a8c-aec9-48ca-9b67-aafdcd3e3d08)
> EXECUTE sofia/internal/60002 at 192.168.10.30db(insert/192.168.10.30-last_dial/60002/1001)
> EXECUTE sofia/internal/60002 at 192.168.10.30db(insert/192.168.10.30-last_dial/global/66610a8c-aec9-48ca-9b67-aafdcd3e3d08)
> EXECUTE sofia/internal/60002 at 192.168.10.30 set(dialed_extension=1001)
> 2010-02-09 21:49:32.857126 [DEBUG] mod_dptools.c:768 sofia/internal/
> 60002 at 192.168.10.30 SET [dialed_extension]=[1001]
> EXECUTE sofia/internal/60002 at 192.168.10.30 export(dialed_extension=1001)
> 2010-02-09 21:49:32.858075 [DEBUG] mod_dptools.c:851 EXPORT
> [dialed_extension]=[1001]
> EXECUTE sofia/internal/60002 at 192.168.10.30 bind_meta_app(1 b s
> execute_extension::dx XML features)
> 2010-02-09 21:49:32.858075 [INFO] switch_ivr_async.c:2174 Bound B-Leg: 1
> execute_extension::dx XML features
> EXECUTE sofia/internal/60002 at 192.168.10.30 bind_meta_app(2 b s
> record_session::/usr/local/freeswitch/recordings/60002.2010-02-09-21-49-32.wav)
> 2010-02-09 21:49:32.858075 [INFO] switch_ivr_async.c:2174 Bound B-Leg: 2
> record_session::/usr/local/freeswitch/recordings/60002.2010-02-09-21-49-32.wav
> EXECUTE sofia/internal/60002 at 192.168.10.30 bind_meta_app(3 b s
> execute_extension::cf XML features)
> 2010-02-09 21:49:32.859074 [INFO] switch_ivr_async.c:2174 Bound B-Leg: 3
> execute_extension::cf XML features
> EXECUTE sofia/internal/60002 at 192.168.10.30 set(ringback=%(2000, 4000,
> 440.0, 480.0))
> 2010-02-09 21:49:32.859074 [DEBUG] mod_dptools.c:768 sofia/internal/
> 60002 at 192.168.10.30 SET [ringback]=[%(2000, 4000, 440.0, 480.0)]
> EXECUTE sofia/internal/60002 at 192.168.10.30set(transfer_ringback=local_stream://moh)
> 2010-02-09 21:49:32.859074 [DEBUG] mod_dptools.c:768 sofia/internal/
> 60002 at 192.168.10.30 SET [transfer_ringback]=[local_stream://moh]
> EXECUTE sofia/internal/60002 at 192.168.10.30 set(call_timeout=30)
> 2010-02-09 21:49:32.860074 [DEBUG] mod_dptools.c:768 sofia/internal/
> 60002 at 192.168.10.30 SET [call_timeout]=[30]
> EXECUTE sofia/internal/60002 at 192.168.10.30 set(hangup_after_bridge=true)
> 2010-02-09 21:49:32.860074 [DEBUG] mod_dptools.c:768 sofia/internal/
> 60002 at 192.168.10.30 SET [hangup_after_bridge]=[true]
> EXECUTE sofia/internal/60002 at 192.168.10.30 set(continue_on_fail=true)
> 2010-02-09 21:49:32.860074 [DEBUG] mod_dptools.c:768 sofia/internal/
> 60002 at 192.168.10.30 SET [continue_on_fail]=[true]
> EXECUTE sofia/internal/60002 at 192.168.10.30db(insert/192.168.10.30-call_return/1001/60002)
> EXECUTE sofia/internal/60002 at 192.168.10.30db(insert/192.168.10.30-last_dial_ext/1001/66610a8c-aec9-48ca-9b67-aafdcd3e3d08)
> EXECUTE sofia/internal/60002 at 192.168.10.30 set(called_party_callgroup=)
> 2010-02-09 21:49:32.870074 [DEBUG] mod_dptools.c:768 sofia/internal/
> 60002 at 192.168.10.30 SET [called_party_callgroup]=[UNDEF]
> EXECUTE sofia/internal/60002 at 192.168.10.30db(insert/192.168.10.30-last_dial//66610a8c-aec9-48ca-9b67-aafdcd3e3d08)
> EXECUTE sofia/internal/60002 at 192.168.10.30 bridge(user/1001 at 192.168.10.30)
> 2010-02-09 21:49:32.905073 [DEBUG] switch_ivr_originate.c:1735 variable
> string 0 = [presence_id=1001 at 192.168.10.30]
> 2010-02-09 21:49:32.905073 [NOTICE] switch_channel.c:613 New Channel
> sofia/internal/sip:1001 at 192.168.10.192:41080[df7cf235-f0e4-406a-83a5-dd2d681bb278]
> 2010-02-09 21:49:32.905073 [DEBUG] mod_sofia.c:3142 (sofia/internal/
> sip:1001 at 192.168.10.192:41080) State Change CS_NEW -> CS_INIT
> 2010-02-09 21:49:32.905073 [DEBUG] switch_core_session.c:999 Send signal
> sofia/internal/sip:1001 at 192.168.10.192:41080 [BREAK]
> 2010-02-09 21:49:32.906075 [DEBUG] switch_core_state_machine.c:314
> (sofia/internal/sip:1001 at 192.168.10.192:41080) Running State Change
> CS_INIT
> 2010-02-09 21:49:32.906075 [DEBUG] switch_core_state_machine.c:338
> (sofia/internal/sip:1001 at 192.168.10.192:41080) State INIT
> 2010-02-09 21:49:32.906075 [DEBUG] mod_sofia.c:83 sofia/internal/
> sip:1001 at 192.168.10.192:41080 SOFIA INIT
> 2010-02-09 21:49:32.907184 [DEBUG] mod_sofia.c:111 (sofia/internal/
> sip:1001 at 192.168.10.192:41080) State Change CS_INIT -> CS_ROUTING
> 2010-02-09 21:49:32.907184 [DEBUG] switch_core_session.c:999 Send signal
> sofia/internal/sip:1001 at 192.168.10.192:41080 [BREAK]
> 2010-02-09 21:49:32.907184 [DEBUG] sofia.c:3727 Channel sofia/internal/
> sip:1001 at 192.168.10.192:41080 entering state [calling][0]
> 2010-02-09 21:49:32.907184 [DEBUG] switch_core_state_machine.c:338
> (sofia/internal/sip:1001 at 192.168.10.192:41080) State INIT going to sleep
> 2010-02-09 21:49:32.907184 [DEBUG] switch_core_state_machine.c:314
> (sofia/internal/sip:1001 at 192.168.10.192:41080) Running State Change
> CS_ROUTING
> 2010-02-09 21:49:32.907184 [DEBUG] switch_core_state_machine.c:341
> (sofia/internal/sip:1001 at 192.168.10.192:41080) State ROUTING
> 2010-02-09 21:49:32.907184 [DEBUG] mod_sofia.c:132 sofia/internal/
> sip:1001 at 192.168.10.192:41080 SOFIA ROUTING
> 2010-02-09 21:49:32.907184 [DEBUG] switch_ivr_originate.c:66
> (sofia/internal/sip:1001 at 192.168.10.192:41080) State Change CS_ROUTING ->
> CS_CONSUME_MEDIA
> 2010-02-09 21:49:32.907184 [DEBUG] switch_core_session.c:999 Send signal
> sofia/internal/sip:1001 at 192.168.10.192:41080 [BREAK]
> 2010-02-09 21:49:32.907184 [DEBUG] switch_core_state_machine.c:341
> (sofia/internal/sip:1001 at 192.168.10.192:41080) State ROUTING going to
> sleep
> 2010-02-09 21:49:32.907184 [DEBUG] switch_core_state_machine.c:314
> (sofia/internal/sip:1001 at 192.168.10.192:41080) Running State Change
> CS_CONSUME_MEDIA
> 2010-02-09 21:49:33.023071 [DEBUG] switch_core_state_machine.c:360
> (sofia/internal/sip:1001 at 192.168.10.192:41080) State CONSUME_MEDIA
> 2010-02-09 21:49:33.023071 [DEBUG] switch_core_state_machine.c:360
> (sofia/internal/sip:1001 at 192.168.10.192:41080) State CONSUME_MEDIA going
> to sleep
> 2010-02-09 21:49:33.329319 [DEBUG] sofia.c:3727 Channel sofia/internal/
> sip:1001 at 192.168.10.192:41080 entering state [terminated][488]
> 2010-02-09 21:49:33.329319 [NOTICE] sofia.c:4331 Hangup sofia/internal/
> sip:1001 at 192.168.10.192:41080 [CS_CONSUME_MEDIA]
> [INCOMPATIBLE_DESTINATION]
> 2010-02-09 21:49:33.329319 [DEBUG] switch_channel.c:1912 Send signal
> sofia/internal/sip:1001 at 192.168.10.192:41080 [KILL]
> 2010-02-09 21:49:33.329319 [DEBUG] switch_core_session.c:999 Send signal
> sofia/internal/sip:1001 at 192.168.10.192:41080 [BREAK]
> 2010-02-09 21:49:33.329319 [DEBUG] switch_core_state_machine.c:459
> sofia/internal/sip:1001 at 192.168.10.192:41080 thread mismatch skipping
> state handler.
> 2010-02-09 21:49:33.329319 [DEBUG] switch_core_state_machine.c:314
> (sofia/internal/sip:1001 at 192.168.10.192:41080) Running State Change
> CS_HANGUP
> 2010-02-09 21:49:33.330075 [DEBUG] switch_core_state_machine.c:488
> (sofia/internal/sip:1001 at 192.168.10.192:41080) State HANGUP
> 2010-02-09 21:49:33.330075 [DEBUG] mod_sofia.c:352 sofia/internal/
> sip:1001 at 192.168.10.192:41080 Overriding SIP cause 488 with 488 from the
> other leg
> 2010-02-09 21:49:33.330075 [DEBUG] mod_sofia.c:358 Channel sofia/internal/
> sip:1001 at 192.168.10.192:41080 hanging up, cause: INCOMPATIBLE_DESTINATION
> 2010-02-09 21:49:33.330075 [DEBUG] switch_core_state_machine.c:46
> sofia/internal/sip:1001 at 192.168.10.192:41080 Standard HANGUP, cause:
> INCOMPATIBLE_DESTINATION
> 2010-02-09 21:49:33.330075 [DEBUG] switch_core_state_machine.c:488
> (sofia/internal/sip:1001 at 192.168.10.192:41080) State HANGUP going to sleep
> 2010-02-09 21:49:33.330075 [DEBUG] switch_core_state_machine.c:333
> (sofia/internal/sip:1001 at 192.168.10.192:41080) State Change CS_HANGUP ->
> CS_REPORTING
> 2010-02-09 21:49:33.330075 [DEBUG] switch_core_session.c:999 Send signal
> sofia/internal/sip:1001 at 192.168.10.192:41080 [BREAK]
> 2010-02-09 21:49:33.330075 [DEBUG] switch_core_state_machine.c:314
> (sofia/internal/sip:1001 at 192.168.10.192:41080) Running State Change
> CS_REPORTING
> 2010-02-09 21:49:33.330075 [DEBUG] switch_core_state_machine.c:579
> (sofia/internal/sip:1001 at 192.168.10.192:41080) State REPORTING
> 2010-02-09 21:49:33.330075 [DEBUG] switch_core_state_machine.c:53
> sofia/internal/sip:1001 at 192.168.10.192:41080 Standard REPORTING, cause:
> INCOMPATIBLE_DESTINATION
> 2010-02-09 21:49:33.330075 [DEBUG] switch_core_state_machine.c:579
> (sofia/internal/sip:1001 at 192.168.10.192:41080) State REPORTING going to
> sleep
> 2010-02-09 21:49:33.330075 [DEBUG] switch_core_state_machine.c:327
> (sofia/internal/sip:1001 at 192.168.10.192:41080) State Change CS_REPORTING
> -> CS_DESTROY
> 2010-02-09 21:49:33.330075 [DEBUG] switch_core_session.c:999 Send signal
> sofia/internal/sip:1001 at 192.168.10.192:41080 [BREAK]
> 2010-02-09 21:49:33.330075 [DEBUG] switch_core_session.c:1136 Session 2
> (sofia/internal/sip:1001 at 192.168.10.192:41080) Locked, Waiting on external
> entities
> 2010-02-09 21:49:33.331072 [DEBUG] switch_ivr_originate.c:3009 Originate
> Resulted in Error Cause: 88 [INCOMPATIBLE_DESTINATION]
> 2010-02-09 21:49:33.331072 [NOTICE] switch_core_session.c:1154 Session 2
> (sofia/internal/sip:1001 at 192.168.10.192:41080) Ended
> 2010-02-09 21:49:33.331072 [NOTICE] switch_core_session.c:1156 Close
> Channel sofia/internal/sip:1001 at 192.168.10.192:41080 [CS_DESTROY]
> 2010-02-09 21:49:33.331072 [DEBUG] switch_core_state_machine.c:423
> (sofia/internal/sip:1001 at 192.168.10.192:41080) Running State Change
> CS_DESTROY
> 2010-02-09 21:49:33.331072 [DEBUG] switch_core_state_machine.c:434
> (sofia/internal/sip:1001 at 192.168.10.192:41080) State DESTROY
> 2010-02-09 21:49:33.331072 [DEBUG] mod_sofia.c:293 sofia/internal/
> sip:1001 at 192.168.10.192:41080 SOFIA DESTROY
> 2010-02-09 21:49:33.331072 [DEBUG] switch_core_state_machine.c:60
> sofia/internal/sip:1001 at 192.168.10.192:41080 Standard DESTROY
> 2010-02-09 21:49:33.331072 [DEBUG] switch_core_state_machine.c:434
> (sofia/internal/sip:1001 at 192.168.10.192:41080) State DESTROY going to
> sleep
> 2010-02-09 21:49:33.331072 [ERR] switch_ivr_originate.c:2249 Cannot create
> outgoing channel of type [user] cause: [INCOMPATIBLE_DESTINATION]
> 2010-02-09 21:49:33.331072 [DEBUG] switch_ivr_originate.c:3009 Originate
> Resulted in Error Cause: 88 [INCOMPATIBLE_DESTINATION]
> 2010-02-09 21:49:33.331072 [INFO] mod_dptools.c:2294 Originate Failed.
> Cause: INCOMPATIBLE_DESTINATION
> EXECUTE sofia/internal/60002 at 192.168.10.30 answer()
> 2010-02-09 21:49:33.332075 [DEBUG] mod_dptools.c:658 sofia/internal/
> 60002 at 192.168.10.30 receive message [ANSWER]
> 2010-02-09 21:49:33.332075 [DEBUG] sofia_glue.c:2381 AUDIO RTP
> [sofia/internal/60002 at 192.168.10.30] 192.168.10.30 port 27634 ->
> 192.168.10.131 port 25592 codec: 9 ms: 20
> 2010-02-09 21:49:33.332075 [DEBUG] switch_rtp.c:1167 Starting timer [soft]
> 160 bytes per 20ms
> 2010-02-09 21:49:33.333080 [DEBUG] mod_sofia.c:571 Local SDP
> sofia/internal/60002 at 192.168.10.30:
> v=0
> o=FreeSWITCH 1265704739 1265704740 IN IP4 192.168.10.30
> s=FreeSWITCH
> c=IN IP4 192.168.10.30
> t=0 0
> m=audio 27634 RTP/AVP 9 101
> a=rtpmap:9 G722/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
>
> 2010-02-09 21:49:33.333080 [DEBUG] sofia.c:3727 Channel sofia/internal/
> 60002 at 192.168.10.30 entering state [completed][200]
> 2010-02-09 21:49:33.333080 [DEBUG] switch_core_session.c:645 Send signal
> sofia/internal/60002 at 192.168.10.30 [BREAK]
> 2010-02-09 21:49:33.333080 [NOTICE] mod_dptools.c:658 Channel
> [sofia/internal/60002 at 192.168.10.30] has been answered
> 2010-02-09 21:49:33.333080 [DEBUG] switch_channel.c:182 sofia/internal/
> 60002 at 192.168.10.30 receive message [AUDIO_SYNC]
> EXECUTE sofia/internal/60002 at 192.168.10.30 sleep(1000)
> 2010-02-09 21:49:33.334081 [DEBUG] switch_channel.c:182 sofia/internal/
> 60002 at 192.168.10.30 receive message [AUDIO_SYNC]
> 2010-02-09 21:49:33.395069 [DEBUG] switch_rtp.c:2004 Correct ip/port
> confirmed.
> 2010-02-09 21:49:33.455069 [DEBUG] sofia.c:3727 Channel sofia/internal/
> 60002 at 192.168.10.30 entering state [ready][200]
> EXECUTE sofia/internal/60002 at 192.168.10.30 voicemail(default 192.168.10.30
> 1001)
> 2010-02-09 21:49:34.335070 [DEBUG] mod_voicemail.c:730 [default] rwlock
> 2010-02-09 21:49:34.370083 [DEBUG] switch_channel.c:182 sofia/internal/
> 60002 at 192.168.10.30 receive message [AUDIO_SYNC]
> 2010-02-09 21:49:34.475070 [DEBUG] switch_ivr_play_say.c:118 No language
> specified - Using [en]
> 2010-02-09 21:49:34.475070 [DEBUG] switch_ivr_play_say.c:273 Handle
> play-file:[voicemail/vm-person.wav] (en:en)
> 2010-02-09 21:49:34.523070 [WARNING] switch_core_file.c:177 Sample rate
> doesn't match
> 2010-02-09 21:49:34.524073 [DEBUG] switch_ivr_play_say.c:1154 Codec
> Activated L16 at 16000hz 1 channels 20ms
> 2010-02-09 21:49:34.526072 [DEBUG] switch_core_io.c:652 sofia/internal/
> 60002 at 192.168.10.30 receive message [TRANSCODING_NECESSARY]
> 2010-02-09 21:49:35.875063 [DEBUG] switch_ivr_play_say.c:1446 done playing
> file
> 2010-02-09 21:49:35.995062 [DEBUG] switch_ivr_play_say.c:273 Handle
> say:[1001] (en:en)
> 2010-02-09 21:49:36.027062 [WARNING] switch_core_file.c:177 Sample rate
> doesn't match
> 2010-02-09 21:49:36.027062 [DEBUG] switch_ivr_play_say.c:1154 Codec
> Activated L16 at 16000hz 1 channels 20ms
> 2010-02-09 21:49:36.028065 [DEBUG] switch_core_io.c:652 sofia/internal/
> 60002 at 192.168.10.30 receive message [TRANSCODING_NECESSARY]
> 2010-02-09 21:49:36.455072 [DEBUG] switch_ivr_play_say.c:1446 done playing
> file
> 2010-02-09 21:49:36.460061 [WARNING] switch_core_file.c:177 Sample rate
> doesn't match
> 2010-02-09 21:49:36.460061 [DEBUG] switch_ivr_play_say.c:1154 Codec
> Activated L16 at 16000hz 1 channels 20ms
> 2010-02-09 21:49:36.461074 [DEBUG] switch_core_io.c:652 sofia/internal/
> 60002 at 192.168.10.30 receive message [TRANSCODING_NECESSARY]
> 2010-02-09 21:49:36.995060 [DEBUG] switch_ivr_play_say.c:1446 done playing
> file
> 2010-02-09 21:49:36.995060 [WARNING] switch_core_file.c:177 Sample rate
> doesn't match
> 2010-02-09 21:49:36.995060 [DEBUG] switch_ivr_play_say.c:1154 Codec
> Activated L16 at 16000hz 1 channels 20ms
> 2010-02-09 21:49:36.995060 [DEBUG] switch_core_io.c:652 sofia/internal/
> 60002 at 192.168.10.30 receive message [TRANSCODING_NECESSARY]
> 2010-02-09 21:49:37.535061 [DEBUG] switch_ivr_play_say.c:1446 done playing
> file
> 2010-02-09 21:49:37.535061 [WARNING] switch_core_file.c:177 Sample rate
> doesn't match
> 2010-02-09 21:49:37.535061 [DEBUG] switch_ivr_play_say.c:1154 Codec
> Activated L16 at 16000hz 1 channels 20ms
> 2010-02-09 21:49:37.535061 [DEBUG] switch_core_io.c:652 sofia/internal/
> 60002 at 192.168.10.30 receive message [TRANSCODING_NECESSARY]
> 2010-02-09 21:49:37.995060 [DEBUG] switch_ivr_play_say.c:1446 done playing
> file
> 2010-02-09 21:49:38.095060 [DEBUG] switch_ivr_play_say.c:273 Handle
> play-file:[voicemail/vm-not_available.wav] (en:en)
> 2010-02-09 21:49:38.118060 [WARNING] switch_core_file.c:177 Sample rate
> doesn't match
> 2010-02-09 21:49:38.119063 [DEBUG] switch_ivr_play_say.c:1154 Codec
> Activated L16 at 16000hz 1 channels 20ms
> 2010-02-09 21:49:38.120062 [DEBUG] switch_core_io.c:652 sofia/internal/
> 60002 at 192.168.10.30 receive message [TRANSCODING_NECESSARY]
> 2010-02-09 21:49:39.075057 [DEBUG] switch_ivr_play_say.c:1446 done playing
> file
> 2010-02-09 21:49:39.175057 [DEBUG] switch_ivr_play_say.c:118 No language
> specified - Using [en]
> 2010-02-09 21:49:39.177086 [DEBUG] switch_ivr_play_say.c:273 Handle
> play-file:[voicemail/vm-record_message.wav] (en:en)
> 2010-02-09 21:49:39.181967 [WARNING] switch_core_file.c:177 Sample rate
> doesn't match
> 2010-02-09 21:49:39.181967 [DEBUG] switch_ivr_play_say.c:1154 Codec
> Activated L16 at 16000hz 1 channels 20ms
> 2010-02-09 21:49:39.192068 [DEBUG] switch_core_io.c:652 sofia/internal/
> 60002 at 192.168.10.30 receive message [TRANSCODING_NECESSARY]
> 2010-02-09 21:49:40.999151 [NOTICE] sofia.c:329 Hangup sofia/internal/
> 60002 at 192.168.10.30 [CS_EXECUTE] [NORMAL_CLEARING]
> 2010-02-09 21:49:40.999151 [DEBUG] switch_channel.c:1912 Send signal
> sofia/internal/60002 at 192.168.10.30 [KILL]
> 2010-02-09 21:49:40.999151 [DEBUG] switch_core_session.c:999 Send signal
> sofia/internal/60002 at 192.168.10.30 [BREAK]
> 2010-02-09 21:49:40.999151 [DEBUG] switch_core_state_machine.c:459
> sofia/internal/60002 at 192.168.10.30 thread mismatch skipping state handler.
> 2010-02-09 21:49:41.015049 [DEBUG] switch_ivr_play_say.c:1446 done playing
> file
> 2010-02-09 21:49:41.114049 [DEBUG] switch_core_state_machine.c:348
> (sofia/internal/60002 at 192.168.10.30) State EXECUTE going to sleep
> 2010-02-09 21:49:41.114049 [DEBUG] switch_core_state_machine.c:314
> (sofia/internal/60002 at 192.168.10.30) Running State Change CS_HANGUP
> 2010-02-09 21:49:41.114049 [DEBUG] switch_core_state_machine.c:488
> (sofia/internal/60002 at 192.168.10.30) State HANGUP
> 2010-02-09 21:49:41.114049 [DEBUG] mod_sofia.c:352 sofia/internal/
> 60002 at 192.168.10.30 Overriding SIP cause 480 with 488 from the other leg
> 2010-02-09 21:49:41.114049 [DEBUG] mod_sofia.c:358 Channel sofia/internal/
> 60002 at 192.168.10.30 hanging up, cause: NORMAL_CLEARING
> 2010-02-09 21:49:41.114049 [DEBUG] switch_core_state_machine.c:46
> sofia/internal/60002 at 192.168.10.30 Standard HANGUP, cause: NORMAL_CLEARING
> 2010-02-09 21:49:41.114049 [DEBUG] switch_core_state_machine.c:488
> (sofia/internal/60002 at 192.168.10.30) State HANGUP going to sleep
> 2010-02-09 21:49:41.114049 [DEBUG] switch_core_state_machine.c:333
> (sofia/internal/60002 at 192.168.10.30) State Change CS_HANGUP ->
> CS_REPORTING
> 2010-02-09 21:49:41.114049 [DEBUG] switch_core_session.c:999 Send signal
> sofia/internal/60002 at 192.168.10.30 [BREAK]
> 2010-02-09 21:49:41.114049 [DEBUG] switch_core_state_machine.c:314
> (sofia/internal/60002 at 192.168.10.30) Running State Change CS_REPORTING
> 2010-02-09 21:49:41.114049 [DEBUG] switch_core_state_machine.c:579
> (sofia/internal/60002 at 192.168.10.30) State REPORTING
> 2010-02-09 21:49:41.129049 [DEBUG] switch_core_state_machine.c:53
> sofia/internal/60002 at 192.168.10.30 Standard REPORTING, cause:
> NORMAL_CLEARING
> 2010-02-09 21:49:41.129049 [DEBUG] switch_core_state_machine.c:579
> (sofia/internal/60002 at 192.168.10.30) State REPORTING going to sleep
> 2010-02-09 21:49:41.129049 [DEBUG] switch_core_state_machine.c:327
> (sofia/internal/60002 at 192.168.10.30) State Change CS_REPORTING ->
> CS_DESTROY
> 2010-02-09 21:49:41.129049 [DEBUG] switch_core_session.c:999 Send signal
> sofia/internal/60002 at 192.168.10.30 [BREAK]
> 2010-02-09 21:49:41.129049 [DEBUG] switch_core_session.c:1136 Session 1
> (sofia/internal/60002 at 192.168.10.30) Locked, Waiting on external entities
> 2010-02-09 21:49:41.129049 [NOTICE] switch_core_session.c:1154 Session 1
> (sofia/internal/60002 at 192.168.10.30) Ended
> 2010-02-09 21:49:41.129049 [NOTICE] switch_core_session.c:1156 Close
> Channel sofia/internal/60002 at 192.168.10.30 [CS_DESTROY]
> 2010-02-09 21:49:41.129049 [DEBUG] switch_core_state_machine.c:423
> (sofia/internal/60002 at 192.168.10.30) Running State Change CS_DESTROY
> 2010-02-09 21:49:41.129049 [DEBUG] switch_core_state_machine.c:434
> (sofia/internal/60002 at 192.168.10.30) State DESTROY
> 2010-02-09 21:49:41.129049 [DEBUG] mod_sofia.c:293 sofia/internal/
> 60002 at 192.168.10.30 SOFIA DESTROY
> 2010-02-09 21:49:41.129049 [DEBUG] switch_core_state_machine.c:60
> sofia/internal/60002 at 192.168.10.30 Standard DESTROY
> 2010-02-09 21:49:41.129049 [DEBUG] switch_core_state_machine.c:434
> (sofia/internal/60002 at 192.168.10.30) State DESTROY going to sleep
>
>
>
>
>
>
>
> On 9 February 2010 21:18, Brian West <brian at freeswitch.org> wrote:
>
>> What you're saying makes little or no sense to me even on 1.0.4, Can you
>> pastebin your logs?
>>
>> /b
>>
>> On Feb 9, 2010, at 3:15 PM, Bruce Hopkins wrote:
>>
>> > OK, many thanks for the extremely swift response Brian.
>> >
>> > I will try to get up and running as soon as I can with 1.0.5 and see if
>> the issue goes away.
>> >
>> > thanks again
>> > Bruce
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100209/e2ab1b54/attachment-0001.html 

From brian at freeswitch.org  Tue Feb  9 14:08:28 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 9 Feb 2010 16:08:28 -0600
Subject: [Freeswitch-users] g.722 --> SPEEX/16000 or SPEEX/32000
	transcoding issue
In-Reply-To: <d03a10e01002091404i5e7aefaam7235bef68828d42b@mail.gmail.com>
References: <d03a10e01002091203qde4caabi1a7565929e60918f@mail.gmail.com>
	<03A2F6D0-754A-4067-A106-9D91B0D68A92@freeswitch.org>
	<d03a10e01002091315n5fa9242eoe2a86c4edee1501d@mail.gmail.com>
	<F167991C-151B-4D40-BF27-80A4C3A44D1E@freeswitch.org>
	<d03a10e01002091355n1934627t3902e0d8af5c7dc3@mail.gmail.com>
	<d03a10e01002091404i5e7aefaam7235bef68828d42b@mail.gmail.com>
Message-ID: <A97C0347-C9C5-4AFE-854E-10468A5E7711@freeswitch.org>

Nope we don't look at that... that isn't the problem.  The issue is you haven't allowed SPEEX at 16000h or SPEEX at 32000h

You can't just allow SPEEX... it can't figure out what you mean by just SPEEX.  See previous email.

/b

On Feb 9, 2010, at 4:04 PM, Bruce Hopkins wrote:

> g.722 might not look like it from the SDP which announces g722/8000 - but this is a historical error in the RFC.
> 
> Bruce



From brian at freeswitch.org  Tue Feb  9 14:10:33 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 9 Feb 2010 16:10:33 -0600
Subject: [Freeswitch-users] g.722 --> SPEEX/16000 or SPEEX/32000
	transcoding issue
In-Reply-To: <d03a10e01002091404i5e7aefaam7235bef68828d42b@mail.gmail.com>
References: <d03a10e01002091203qde4caabi1a7565929e60918f@mail.gmail.com>
	<03A2F6D0-754A-4067-A106-9D91B0D68A92@freeswitch.org>
	<d03a10e01002091315n5fa9242eoe2a86c4edee1501d@mail.gmail.com>
	<F167991C-151B-4D40-BF27-80A4C3A44D1E@freeswitch.org>
	<d03a10e01002091355n1934627t3902e0d8af5c7dc3@mail.gmail.com>
	<d03a10e01002091404i5e7aefaam7235bef68828d42b@mail.gmail.com>
Message-ID: <DBD23DC8-C9F3-432E-90E0-E4132D1C5C52@freeswitch.org>

This means you're playing a wav file that doesn't match so it has to resample it... Please open up vars.xml and read thru the codec docs there in the XML directly.  It will explain the codecs and how to allow them in various ptimes and rates as you need to do in your case.  Its got nothing to do with G722/8000 or anything to do with the file you're playing (which isn't matching the channel sample rate so it has to resample so at the very least it works.

Run the "file" util on your wav file and see what its real rate is.

/b

On Feb 9, 2010, at 4:04 PM, Bruce Hopkins wrote:

> 2010-02-09 21:49:36.995060 [WARNING] switch_core_file.c:177 Sample rate doesn't match



From jbrucehopkins at gmail.com  Tue Feb  9 14:13:06 2010
From: jbrucehopkins at gmail.com (Bruce Hopkins)
Date: Tue, 9 Feb 2010 22:13:06 +0000
Subject: [Freeswitch-users] g.722 --> SPEEX/16000 or SPEEX/32000
	transcoding issue
In-Reply-To: <51A80F87-F2B9-477E-8D14-99C40E24413D@freeswitch.org>
References: <d03a10e01002091203qde4caabi1a7565929e60918f@mail.gmail.com>
	<03A2F6D0-754A-4067-A106-9D91B0D68A92@freeswitch.org>
	<d03a10e01002091315n5fa9242eoe2a86c4edee1501d@mail.gmail.com>
	<F167991C-151B-4D40-BF27-80A4C3A44D1E@freeswitch.org>
	<d03a10e01002091355n1934627t3902e0d8af5c7dc3@mail.gmail.com>
	<51A80F87-F2B9-477E-8D14-99C40E24413D@freeswitch.org>
Message-ID: <d03a10e01002091413see99fe3ke52e0d3f91714216@mail.gmail.com>

Aha - I will try this now.  At the moment I only have SPEEX without anything
like @8000h or 16000h.

Thanks in advance, as I am sure you have spotted my noob error !

Bruce

On 9 February 2010 22:00, Brian West <brian at freeswitch.org> wrote:

> You need to allow SPEEX at 8000h,SPEEX at 16000h,SPEEX at 32000h
>
> /b
>
> On Feb 9, 2010, at 3:55 PM, Bruce Hopkins wrote:
>
> > Willdo,
> >
> > To clarify in brief though, the scenario which occurs and causes the call
> to fail is:
> >
> > SIP client  1 (g.722 enabled only ) -----> INVITE (with SDP offer: Media
> Attribute (a): rtpmap:9 g722/8000 ) -->FreeSWITCH
> >
> > ---> INVITE (with SDP offer including a bunch of codecs including rtpmap:
> rtpmap:98 SPEEX/8000 but crucially not including SPEEX/16000 or SPEEX/32000)
> >
> > ---> SIP client 2 (with only SPEEX/16000 or SPEEX/32000 enabled).
> >
> > The second SIP client does not get offered a codec it can accept, so SIP
> client 1 is sent a method 488 "Not Acceptable Here" message and the calling
> party gets directed to the voicemail for the other SIP client.
> >
> > By contrast, there is no problem calling SPEEX/32000 --> SPEEX/32000 or
> calling SPEEX/16000 --> SPEEX/16000.
> >
> > there is also no problem calling SPEEX/32000 --> g.722/8000.
> >
> > I am wondering if the problem is that FreeSWITCH is interpreting g.722 as
> being a narrowband (8kHz sample rate) codec, due to the historic anomaly of
> it presenting g722/8000 in the SDP even though it in fact uses 16kHz
> sampling, and for that reason not wanting to offer a 16kHz sample rate codec
> to the second SIP client?
> >
> > I suggest this as I also found trying to call alaw --> SPEEX/16000 does
> not work, for example.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100209/57a0d542/attachment.html 

From lon at kickasspixels.com  Tue Feb  9 14:17:30 2010
From: lon at kickasspixels.com (Lon Baker)
Date: Tue, 9 Feb 2010 14:17:30 -0800
Subject: [Freeswitch-users] T.38 Fax mode?
Message-ID: <5d3e0dc61002091417l32526087je432ee465f88ef46@mail.gmail.com>

Hi,

Is 1.0.5 going to support T.38 fax mode? It looks like its partially
implemented.

The comment leads me to think its not

/*
		   Here goes the T.38 SpanDSP initializing functions
		   T.38 will require a big effort as it needs a different approach
		   but the pieces are already in place
		*/

Lon


From jbrucehopkins at gmail.com  Tue Feb  9 14:26:24 2010
From: jbrucehopkins at gmail.com (Bruce Hopkins)
Date: Tue, 9 Feb 2010 22:26:24 +0000
Subject: [Freeswitch-users] g.722 --> SPEEX/16000 or SPEEX/32000
	transcoding issue
In-Reply-To: <d03a10e01002091413see99fe3ke52e0d3f91714216@mail.gmail.com>
References: <d03a10e01002091203qde4caabi1a7565929e60918f@mail.gmail.com>
	<03A2F6D0-754A-4067-A106-9D91B0D68A92@freeswitch.org>
	<d03a10e01002091315n5fa9242eoe2a86c4edee1501d@mail.gmail.com>
	<F167991C-151B-4D40-BF27-80A4C3A44D1E@freeswitch.org>
	<d03a10e01002091355n1934627t3902e0d8af5c7dc3@mail.gmail.com>
	<51A80F87-F2B9-477E-8D14-99C40E24413D@freeswitch.org>
	<d03a10e01002091413see99fe3ke52e0d3f91714216@mail.gmail.com>
Message-ID: <d03a10e01002091426q391ac6f9nb14cbf678e3fd840@mail.gmail.com>

Hi again Brian,

I can confirm that having followed your advice, it now works perfectly.

Many, many thanks for your extraordinarily quick and comprehensive help with
this.

Best wishes
Bruce

On 9 February 2010 22:13, Bruce Hopkins <jbrucehopkins at gmail.com> wrote:

> Aha - I will try this now.  At the moment I only have SPEEX without
> anything like @8000h or 16000h.
>
> Thanks in advance, as I am sure you have spotted my noob error !
>
> Bruce
>
>
> On 9 February 2010 22:00, Brian West <brian at freeswitch.org> wrote:
>
>> You need to allow SPEEX at 8000h,SPEEX at 16000h,SPEEX at 32000h
>>
>> /b
>>
>> On Feb 9, 2010, at 3:55 PM, Bruce Hopkins wrote:
>>
>> > Willdo,
>> >
>> > To clarify in brief though, the scenario which occurs and causes the
>> call to fail is:
>> >
>> > SIP client  1 (g.722 enabled only ) -----> INVITE (with SDP offer: Media
>> Attribute (a): rtpmap:9 g722/8000 ) -->FreeSWITCH
>> >
>> > ---> INVITE (with SDP offer including a bunch of codecs including
>> rtpmap: rtpmap:98 SPEEX/8000 but crucially not including SPEEX/16000 or
>> SPEEX/32000)
>> >
>> > ---> SIP client 2 (with only SPEEX/16000 or SPEEX/32000 enabled).
>> >
>> > The second SIP client does not get offered a codec it can accept, so SIP
>> client 1 is sent a method 488 "Not Acceptable Here" message and the calling
>> party gets directed to the voicemail for the other SIP client.
>> >
>> > By contrast, there is no problem calling SPEEX/32000 --> SPEEX/32000 or
>> calling SPEEX/16000 --> SPEEX/16000.
>> >
>> > there is also no problem calling SPEEX/32000 --> g.722/8000.
>> >
>> > I am wondering if the problem is that FreeSWITCH is interpreting g.722
>> as being a narrowband (8kHz sample rate) codec, due to the historic anomaly
>> of it presenting g722/8000 in the SDP even though it in fact uses 16kHz
>> sampling, and for that reason not wanting to offer a 16kHz sample rate codec
>> to the second SIP client?
>> >
>> > I suggest this as I also found trying to call alaw --> SPEEX/16000 does
>> not work, for example.
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100209/f5d948e2/attachment.html 

From sergey.kobzar at mail.ru  Tue Feb  9 14:44:22 2010
From: sergey.kobzar at mail.ru (Sergey Kobzar)
Date: Wed, 10 Feb 2010 00:44:22 +0200
Subject: [Freeswitch-users] Video conferencing
In-Reply-To: <D9657862-D8A0-45C8-98E2-42814BE8D6A1@jerris.com>
References: <57499143.20100209185702@mail.ru>
	<D9657862-D8A0-45C8-98E2-42814BE8D6A1@jerris.com>
Message-ID: <1899501996.20100210004422@mail.ru>

Which softphone do you use?

Do you know any other alternatives which works better?


Tuesday, February 9, 2010, 11:16:13 PM, Michael wrote:

> Our video conference features "work" but the functionality is
> pretty limited.  We don;t have iframe detection and can not do any
> video transcoding, just video follow audio support.  This code needs some work for sure.

> Mike

> On Feb 9, 2010, at 11:57 AM, Sergey Kobzar wrote:

>> Does anybody have a success story of implementing video conferencing?
>> 
>> I've spend some time with Goole and found that Asterisk has many
>> limitations, hardware solutions are quite expensive. I played with FS
>> without luck.


> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




-- 
Sergey



From rupa at rupa.com  Tue Feb  9 14:55:05 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Tue, 9 Feb 2010 16:55:05 -0600
Subject: [Freeswitch-users] UPnP Timeout
In-Reply-To: <89dbfdc31002090652h884bc2av82d53c6958fdc409@mail.gmail.com>
References: <89dbfdc31002090652h884bc2av82d53c6958fdc409@mail.gmail.com>
Message-ID: <b39273531002091455meb77162o2baefc3866efaed9@mail.gmail.com>

That is a different keep-alive.  I'm specifically talking about the
keep-alive packet that we get via upnp multicast.  Whenever we receive one
from the gateway we republish the nat mappings to.. um... keep them alive.
:)

On Tue, Feb 9, 2010 at 8:52 AM, Kim Culhan <w8hdkim at gmail.com> wrote:

> On Mon, February 8, 2010 11:21 pm, Troy Anderson wrote:
> > I have been using pfSense (1.2.3) and FS.  FS nicely uses UPnP to poke
> > holes in the firewall, but it seems that the holes close after a while.
>  I
> > cannot find any documentation in FS nor in pfSense as to what the timeout
> > is.  Is there a setting in FS to do some kind of keep-alive thing with
> > UPnP to keep, e.g. 5060, open?  Or is it already doing that and pfSense
> is
> > the issue?
>
> FS has provisions for keep-alive, see the bottom of the page for ping
> time value:
>
> http://wiki.freeswitch.org/wiki/SIP_Provider_Examples
>
> To watch the pf firewall hole timing you can install pftop from
> FreeBSD ports/sysutils
> which displays the filter states 'and more'.
>
> -kim
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
-Rupa
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100209/3ea876cd/attachment-0001.html 

From msc at freeswitch.org  Tue Feb  9 14:55:11 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 9 Feb 2010 14:55:11 -0800
Subject: [Freeswitch-users] Passing channel variables to mod_fifo and
	xml_cdr
In-Reply-To: <004301caa608$534747d0$f9d5d770$@net>
References: <004301caa608$534747d0$f9d5d770$@net>
Message-ID: <87f2f3b91002091455v6af079e1ie28ed6891d5ed628@mail.gmail.com>

On Thu, Feb 4, 2010 at 6:09 PM, Adam Ford <lists at redbonez.net> wrote:

>  When sending a call through mod_fifo I seem to be losing my custom
> channel variables that were assigned during prior processing of the call.
> In my example, I am trying to assign a unique identifier at the time the
> call enters my FreeSWITCH system in order to more easily tie the xml_cdr
> logs together.  This works great, until a call is processed through
> mod_fifo, which drops my custom channel variable in the calls that it
> generates.  Is it likely that I have something wrong with my config? Or does
> mod_fifo not support the passing of custom channel variables?
>
>
>
> The overall problem I am trying to solve is that mod_fifo generates a
> separate a-leg for every time it rings an agent.  If the agent answers, the
> a-leg log gets tied to the associated b-leg log with the uuids and I am able
> to see the entire call in xml_cdr.  However, if the agent rejects the call
> or doesn?t answer, the a-leg is abandoned with seemingly no association back
> to a b-leg log.  Anyone have a better suggestion for tying all these xml_cdr
> logs together for a full picture of the call?
>

Just curious - are you looking at this from the caller's perspective or the
agent's perspective? An unanswered/rejected call from FIFO to an agent
doesn't tell you very much. However, if you're trying to gather statistics
on an individual agent then I could see why you'd want to know how many FIFO
calls they failed to answer. As far as the "new" A leg not being tied back
to a B leg - Mike J is 100% correct: A FIFO call to an agent has absolutely
no correlation to any caller waiting in queue. (I suppose the only exception
to this rule would be if there was only one caller in the queue when the
FIFO called out to the agents.)

Like Mike said, FIFO is not ACD. FIFO is "get the caller to a human as
efficiently and quickly as possible." ACD is more of "connect the longest
waiting caller to the longest waiting agent, with possible exceptions for
skills, etc." Check out Andrew Thompson's SpiceCSM for a possible solution
to your scenario.

http://www.opencsm.org/wiki/index.php/SpiceCSM_Community_Edition

-MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100209/a3382a29/attachment.html 

From mike at jerris.com  Tue Feb  9 15:03:43 2010
From: mike at jerris.com (Michael Jerris)
Date: Tue, 9 Feb 2010 18:03:43 -0500
Subject: [Freeswitch-users] Way to pass max-members from dialplan into
	conference?
In-Reply-To: <C7136E29C3FD472BB367FF412E282BED@greyhawk.tonecommander.com>
References: <C7136E29C3FD472BB367FF412E282BED@greyhawk.tonecommander.com>
Message-ID: <57F0B3AE-BB20-4F27-9CF5-7A9AE77E9738@jerris.com>

We don 't have this right now where you can set a var, but you could add it with one line around line 5265, just set confierence->max_members there based on the result of switch_channel_get_variable.  This would let you set max_members when you first create a conference.  If you would like it to be re-set with any caller, you could do it later down.

Mike

On Feb 9, 2010, at 4:14 PM, Robert Hadley wrote:

> I have setting max-members=10 in conference.conf.xml working.  However, is there are way to pass in the max-members=10 from the dialplan/default.xml to mod_conference?  I tried using action application=?set? data=?max-members=10? but it didn?t work.  Also tried action application=?export? data=?max-members=10? but it didn?t work either.
>  
> From default.xml:
>     <extension name="nb_conferences">
>       <condition field="destination_number" expression="^(30\d{2})$">
>        <action application="answer"/>
> <!-- tried set and export here to no avail -->
>        <action application="conference" data="$1-${domain_name}@default"/>
>  
> Thanks,
> Robert
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100209/5f87f89a/attachment.html 

From mike at jerris.com  Tue Feb  9 15:04:02 2010
From: mike at jerris.com (Michael Jerris)
Date: Tue, 9 Feb 2010 18:04:02 -0500
Subject: [Freeswitch-users] T.38 Fax mode?
In-Reply-To: <5d3e0dc61002091417l32526087je432ee465f88ef46@mail.gmail.com>
References: <5d3e0dc61002091417l32526087je432ee465f88ef46@mail.gmail.com>
Message-ID: <34A64C2E-88A5-48C2-818D-85242321BCA4@jerris.com>

no

On Feb 9, 2010, at 5:17 PM, Lon Baker wrote:

> Hi,
> 
> Is 1.0.5 going to support T.38 fax mode? It looks like its partially
> implemented.
> 
> The comment leads me to think its not
> 
> /*
> 		   Here goes the T.38 SpanDSP initializing functions
> 		   T.38 will require a big effort as it needs a different approach
> 		   but the pieces are already in place
> 		*/



From mike at jerris.com  Tue Feb  9 15:05:03 2010
From: mike at jerris.com (Michael Jerris)
Date: Tue, 9 Feb 2010 18:05:03 -0500
Subject: [Freeswitch-users] Video conferencing
In-Reply-To: <1899501996.20100210004422@mail.ru>
References: <57499143.20100209185702@mail.ru>
	<D9657862-D8A0-45C8-98E2-42814BE8D6A1@jerris.com>
	<1899501996.20100210004422@mail.ru>
Message-ID: <A50D4526-42A7-44B2-99E6-6876303D3358@jerris.com>

they all suck, I have used hard devices and several different softphones, none of which I found usable or acceptable.

Mike

On Feb 9, 2010, at 5:44 PM, Sergey Kobzar wrote:

> Which softphone do you use?
> 
> Do you know any other alternatives which works better?
> 
> 
> Tuesday, February 9, 2010, 11:16:13 PM, Michael wrote:
> 
>> Our video conference features "work" but the functionality is
>> pretty limited.  We don;t have iframe detection and can not do any
>> video transcoding, just video follow audio support.  This code needs some work for sure.
> 
>> Mike



From mike at jerris.com  Tue Feb  9 15:07:20 2010
From: mike at jerris.com (Michael Jerris)
Date: Tue, 9 Feb 2010 18:07:20 -0500
Subject: [Freeswitch-users] Where are new sounds_dir and recordings_dir
	variables set?
In-Reply-To: <28788FA27BE74B9FA6BFFE52BED11B60@greyhawk.tonecommander.com>
References: <28788FA27BE74B9FA6BFFE52BED11B60@greyhawk.tonecommander.com>
Message-ID: <17F2ACB0-0344-4EC1-BB1F-21F3BF4C0892@jerris.com>

These are set at configure time based on the prefix.  When FreeSWITCH starts, it sets global vars (before the config pre-processing) that should work just like the old hard-coded values and how base_dir works with /sounds added to it

Mike

On Feb 9, 2010, at 2:28 PM, Robert Hadley wrote:

>  
> The XML conf files have been recently modified to replace ?$${base_dir}/sounds? with $${sounds_dir}.  The same replacement was done for $${base_dir}/recordings and $${recordings_dir}.  In vars.xml, the X-PRE-PROCESS cmd to setting the old variable sound_prefix was removed but set commands were not added for the new variables sounds_dir and recordings_dir.  How is FS finding determining the value of $${sounds_dir} and $${recordings_dir} in the XML files?
>  
> I ask because I have 2 cloned FS servers where one can find sound files in the IVR when an invalid extension is entered, but the other does not and is using the wrong path to search for sound files.
>  
> Server 2 Fails:
> 2010-02-09 11:11:16.735745 [DEBUG] switch_ivr_menu.c:381 digits '9999'
> 2010-02-09 11:11:16.735745 [DEBUG] switch_ivr_menu.c:475 action regex [9999] [/^((5[34][8901][0-9])||(10[0-5][0-9])||(30\d{2}))$/] [0]
> 2010-02-09 11:11:16.735745 [DEBUG] switch_ivr_menu.c:565 IVR menu 'teo_ivr' caught invalid input '9999'
> 2010-02-09 11:11:16.735745 [ERR] mod_sndfile.c:194 Error Opening File [/opt/teoswitch/sounds/ivr/ivr-that_was_an_invalid_entry.wav] [System error : No such file or directory.]
>  
> Server 1 Works:
> 2010-02-09 11:17:25.085660 [DEBUG] switch_ivr_menu.c:378 digits '9999'
> 2010-02-09 11:17:25.085660 [DEBUG] switch_ivr_menu.c:472 action regex [9999] [/^((5[34][8901][0-9])||(10[0-5][0-9])||(30\d{2}))$/] [0]
> 2010-02-09 11:17:25.085660 [DEBUG] switch_ivr_menu.c:562 IVR menu 'teo_ivr' caught invalid input '9999'
> 2010-02-09 11:17:25.085660 [DEBUG] switch_ivr_play_say.c:1162 Codec Activated L16 at 8000hz 1 channels 20ms
> 2010-02-09 11:17:25.085660 [DEBUG] switch_core_io.c:652 sofia/internal/1045 at 192.168.72.141:5060 receive message [TRANSCODING_NECESSARY]
> 2010-02-09 11:17:26.765470 [DEBUG] switch_ivr_play_say.c:1454 done playing file
>  
> I have compared the conf folders and they are nearly identical.  The only difference I know of is Server 2 was originally compiled with mod_flite enabled but it is not loaded at runtime.  Any suggestions why one server can find sound files but the other looks in the wrong path?
>  
> Thanks,
> Robert
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100209/d7eb530a/attachment-0001.html 

From msc at freeswitch.org  Tue Feb  9 15:09:16 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 9 Feb 2010 15:09:16 -0800
Subject: [Freeswitch-users] Way to pass max-members from dialplan into
	conference?
In-Reply-To: <C7136E29C3FD472BB367FF412E282BED@greyhawk.tonecommander.com>
References: <C7136E29C3FD472BB367FF412E282BED@greyhawk.tonecommander.com>
Message-ID: <87f2f3b91002091509w56eacdedt6a291d9b98838521@mail.gmail.com>

On Tue, Feb 9, 2010 at 1:14 PM, Robert Hadley <robert.hadley at teotech.com>wrote:

>  I have setting max-members=10 in conference.conf.xml working.  However,
> is there are way to pass in the max-members=10 from the dialplan/default.xml
> to mod_conference?  I tried using action application=?set?
> data=?max-members=10? but it didn?t work.  Also tried action
> application=?export? data=?max-members=10? but it didn?t work either.
>
>
>
> From default.xml:
>
>     <extension name="nb_conferences">
>
>       <condition field="destination_number" expression="^(30\d{2})$">
>
>        <action application="answer"/>
>
> <!-- tried set and export here to no avail -->
>
>        <action application="conference" data="$1-${domain_name}@default"/>
>
>
>
> Thanks,
>
> Robert
>

Robert,

I checked with Brian and also took a look inside mod_conference.c. I didn't
see any way that you could override the conference params that are contained
in the conference profile. So you'll either need to make a new profile or
join the big leagues and start trying out mod_xml_curl. :)

-MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100209/80c1f79f/attachment.html 

From msc at freeswitch.org  Tue Feb  9 15:12:45 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 9 Feb 2010 15:12:45 -0800
Subject: [Freeswitch-users] UPnP Timeout
In-Reply-To: <b39273531002091455meb77162o2baefc3866efaed9@mail.gmail.com>
References: <89dbfdc31002090652h884bc2av82d53c6958fdc409@mail.gmail.com>
	<b39273531002091455meb77162o2baefc3866efaed9@mail.gmail.com>
Message-ID: <87f2f3b91002091512n6b66c233xe2436376c347e992@mail.gmail.com>

On Tue, Feb 9, 2010 at 2:55 PM, Rupa Schomaker <rupa at rupa.com> wrote:

> That is a different keep-alive.  I'm specifically talking about the
> keep-alive packet that we get via upnp multicast.  Whenever we receive one
> from the gateway we republish the nat mappings to.. um... keep them alive.
> :)
>
>
Ah.. I see: keep them alive. And all this time I thought it was to keep the
NAT mappings from dying! :P
-MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100209/566e46a5/attachment.html 

From mike at jerris.com  Tue Feb  9 15:17:34 2010
From: mike at jerris.com (Michael Jerris)
Date: Tue, 9 Feb 2010 18:17:34 -0500
Subject: [Freeswitch-users] Bridging sessions from two separate lua
	scripts; uuid_bridge? intercept?
In-Reply-To: <eb7797d71002080737k3f8fa5c9m8bf3ef2fcf42881e@mail.gmail.com>
References: <eb7797d71002071831w27301c5dva767b82b38c2ce6f@mail.gmail.com>
	<C23AA196-E851-4D5B-B1B5-2C2F32F6BB68@freeswitch.org>
	<eb7797d71002071929pb53e1adue2e91e6bdb598251@mail.gmail.com>
	<eb7797d71002080737k3f8fa5c9m8bf3ef2fcf42881e@mail.gmail.com>
Message-ID: <D753EDBD-6ED7-43C2-B5F5-A899E3C3D3E2@jerris.com>

1.4?  how does the future look, report back?

http://files-sync.freeswitch.org/windows_installer/freepbx_svn.exe

I think this has latest FreeSWITCH in it to, Carlos, can you confirm that?

Mike

On Feb 8, 2010, at 10:37 AM, Adam Wilt wrote:

> One other thing I should mention. I'm running FreeSWITCH version 1.4 (build 14460) in Windows. 
> Brian suggested I upgrade to the build in the http://files-sync.freeswitch.org/windows_installer/  folder, but it turned out to be the exact same build I already had. I'd love to try upgrade to 1.5 in case this problem has been fixed already.
> 
> 
> On Sun, Feb 7, 2010 at 10:29 PM, Adam Wilt <wiltingtree at gmail.com> wrote:
> Thanks Michael for the reply.
> Here's the pastebin link: http://pastebin.freeswitch.org/12084
> 
> 
> On Sun, Feb 7, 2010 at 9:50 PM, Michael S Collins <msc at freeswitch.org> wrote:
> Pastebin a debug log so we can see what is happening when the script runs. 
> 
> -MC
> 
> Sent from my iPhone
> 
> On Feb 7, 2010, at 8:31 PM, Adam Wilt <wiltingtree at gmail.com> wrote:
> 
>> Hi. I have two sessions running in two separate Lua scripts, and I want to bridge them so that the bridged call is being controlled by the first (a-leg) script.
>> If I simply use uuid_bridge, I get no error but the calls don't bridge.
>> I've tried intercept, but I don't understand how it should be used; nothing I try seems to work.
>> Here's what I have:
>> 
>> function bridge_calls(session,api,b_leg_uuid, call_len)
>>   session:setAutoHangup(false)
>>   session:execute("sched_hangup","+" .. tostring(call_len) .. " " .. tostring(session.uuid))
>>   session:execute("set","continue_on_fail=true")
>>   api:executeString("intercept -bleg " .. tostring(b_leg_uuid))
>>   api:executeString("uuid_bridge " .. tostring(session.uuid) .. " " .. tostring(b_leg_uuid))
>> end
>> 
>> I'd really appreciate any help.
>> 
>> Thanks,
>> Adam
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100209/9a147cf7/attachment.html 

From sergey.kobzar at mail.ru  Tue Feb  9 15:31:45 2010
From: sergey.kobzar at mail.ru (Sergey Kobzar)
Date: Wed, 10 Feb 2010 01:31:45 +0200
Subject: [Freeswitch-users] Video conferencing
In-Reply-To: <A50D4526-42A7-44B2-99E6-6876303D3358@jerris.com>
References: <57499143.20100209185702@mail.ru>
	<D9657862-D8A0-45C8-98E2-42814BE8D6A1@jerris.com>
	<1899501996.20100210004422@mail.ru>
	<A50D4526-42A7-44B2-99E6-6876303D3358@jerris.com>
Message-ID: <794626665.20100210013145@mail.ru>

I've found only this: http://code.google.com/p/openmeetings/

But didn't try yet. Also it is not exactly what I want.


Wednesday, February 10, 2010, 1:05:03 AM, Michael wrote:

> they all suck, I have used hard devices and several different
> softphones, none of which I found usable or acceptable.

> Mike

> On Feb 9, 2010, at 5:44 PM, Sergey Kobzar wrote:

>> Which softphone do you use?
>> 
>> Do you know any other alternatives which works better?
>> 
>> 
>> Tuesday, February 9, 2010, 11:16:13 PM, Michael wrote:
>> 
>>> Our video conference features "work" but the functionality is
>>> pretty limited.  We don;t have iframe detection and can not do any
>>> video transcoding, just video follow audio support.  This code needs some work for sure.
>> 
>>> Mike


> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




-- 
Sergey



From kristian.kielhofner at gmail.com  Tue Feb  9 15:36:10 2010
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Tue, 9 Feb 2010 18:36:10 -0500
Subject: [Freeswitch-users] T.38 Fax mode?
In-Reply-To: <5d3e0dc61002091417l32526087je432ee465f88ef46@mail.gmail.com>
References: <5d3e0dc61002091417l32526087je432ee465f88ef46@mail.gmail.com>
Message-ID: <2d9149cd1002091536p266f8b13x74224a3fb46f1af4@mail.gmail.com>

Pass-through with proxy media/bypass media works well...

On Tue, Feb 9, 2010 at 5:17 PM, Lon Baker <lon at kickasspixels.com> wrote:
> Hi,
>
> Is 1.0.5 going to support T.38 fax mode? It looks like its partially
> implemented.
>
> The comment leads me to think its not
>
> /*
> ? ? ? ? ? ? ? ? ? Here goes the T.38 SpanDSP initializing functions
> ? ? ? ? ? ? ? ? ? T.38 will require a big effort as it needs a different approach
> ? ? ? ? ? ? ? ? ? but the pieces are already in place
> ? ? ? ? ? ? ? ?*/
>
> Lon
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com


From anthony.minessale at gmail.com  Tue Feb  9 15:57:58 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 9 Feb 2010 17:57:58 -0600
Subject: [Freeswitch-users] Presence PUBLISH Not Updating After
	Softphone OffLine Then Available
In-Reply-To: <45963EB5-DFE5-43FA-AFCC-857E1E719E2D@jerris.com>
References: <31FEE9072F104AE18B26812978FCA341@greyhawk.tonecommander.com>
	<45963EB5-DFE5-43FA-AFCC-857E1E719E2D@jerris.com>
Message-ID: <191c3a031002091557jfc9b486lb832ea538883bce9@mail.gmail.com>

he means update to trunk first then try it again obviously.


On Tue, Feb 9, 2010 at 3:10 PM, Michael Jerris <mike at jerris.com> wrote:

> Try this again, I think I saw changes go in for this issue.
>
> Mike
>
> On Feb 5, 2010, at 2:38 PM, Jerry Richards wrote:
>
> > I found a scenario where presence status is not distributed to
> subscribers.
> > This is using the latest changes (as of Feb 03, 2010).  The scenario
> > follows:
> >
> > 1) Register two Bria softphones (A and B), which each have the other as a
> > contact.
> > 2) Set softphone B's presence status to 'Appear Offline'.
> > 3) Softphone A correctly reflects contact B is offline.
> > 4) Set softphone B's presence status to 'Available'.
> > 5) ******* There is no change to contact B's status at softphone A
> *******
> >
> > I posted a log at http://pastebin.freeswitch.org/12054.  At line 773,
> there
> > is an error when FS is processing the PUBLISH from softphone B:
> >
> > 773.2010-02-05 10:29:21.254221 [ERR] sofia_presence.c:674 DUMP PRESENCE
> SQL:
> >
> > I did notice that after about 30 minutes, softphone B's status gets
> > reflected at softphone A.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100209/dd8ca2b5/attachment-0001.html 

From pjintheusa at gmail.com  Tue Feb  9 16:01:09 2010
From: pjintheusa at gmail.com (Phillip Jones)
Date: Tue, 9 Feb 2010 19:01:09 -0500
Subject: [Freeswitch-users] Last call: buy the devs dinner!
In-Reply-To: <4B71885F.5090908@gmx.net>
References: <87f2f3b91002090735g1c6c69eby5274f6dfa9127fc5@mail.gmail.com>
	<4B71885F.5090908@gmx.net>
Message-ID: <367751821002091601u5c4a275bs6f2894a087b3ffcf@mail.gmail.com>

>> Thanks for the great work you all have done so far.

I second that.

On Tue, Feb 9, 2010 at 11:07 AM, Peter P GMX <Prometheus001 at gmx.net> wrote:

> Hello Michael,
>
> just hit the paypal button. Enjoy your dinner! I think it's not just
> dinner, it will be also 50% work I think (discussing about issues and
> new features etc.) which brings additional benefits to the copmmunity.
> Thanks for the great work you all have done so far.
>
> Best regards
> Peter
>
> Michael Collins schrieb:
> > Hey all,
> >
> > Thanks so much for the donations that have come in already! We
> > appreciate your generosity. The dev team really wants to release 1.0.5
> > but they're kinda hungry! :) Please hit the PayPal button on the main
> > freeswitch.org <http://freeswitch.org> page to drop a few dollars in
> > the hat. Also, keep in mind that we have the "extended family" of
> > developers all here so it's not just Tony, Mike, and Brian. Let's all
> > pitch in and have a great dinner for them.
> >
> > Thanks!
> > -Michael
> > ------------------------------------------------------------------------
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100209/ce5f6b89/attachment.html 

From peder at networkoblivion.com  Tue Feb  9 16:26:01 2010
From: peder at networkoblivion.com (Peder)
Date: Tue, 9 Feb 2010 18:26:01 -0600
Subject: [Freeswitch-users] T.38 Fax mode?
In-Reply-To: <2d9149cd1002091536p266f8b13x74224a3fb46f1af4@mail.gmail.com>
References: <5d3e0dc61002091417l32526087je432ee465f88ef46@mail.gmail.com>
	<2d9149cd1002091536p266f8b13x74224a3fb46f1af4@mail.gmail.com>
Message-ID: <008201caa9e7$9fccf770$df66e650$@com>

Do you find any difference in success rate between using proxy instead of
bypass for T.38?  I wouldn't think it would really matter, I am just
curious.

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Kristian
Kielhofner
Sent: Tuesday, February 09, 2010 5:36 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] T.38 Fax mode?

Pass-through with proxy media/bypass media works well...

On Tue, Feb 9, 2010 at 5:17 PM, Lon Baker <lon at kickasspixels.com> wrote:
> Hi,
>
> Is 1.0.5 going to support T.38 fax mode? It looks like its partially
> implemented.
>
> The comment leads me to think its not
>
> /*
> ? ? ? ? ? ? ? ? ? Here goes the T.38 SpanDSP initializing functions
> ? ? ? ? ? ? ? ? ? T.38 will require a big effort as it needs a different
approach
> ? ? ? ? ? ? ? ? ? but the pieces are already in place
> ? ? ? ? ? ? ? ?*/
>
> Lon
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



From kristian.kielhofner at gmail.com  Tue Feb  9 17:21:30 2010
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Tue, 9 Feb 2010 20:21:30 -0500
Subject: [Freeswitch-users] T.38 Fax mode?
In-Reply-To: <008201caa9e7$9fccf770$df66e650$@com>
References: <5d3e0dc61002091417l32526087je432ee465f88ef46@mail.gmail.com>
	<2d9149cd1002091536p266f8b13x74224a3fb46f1af4@mail.gmail.com>
	<008201caa9e7$9fccf770$df66e650$@com>
Message-ID: <2d9149cd1002091721o4f100970k568acef52644f823@mail.gmail.com>

We almost always use proxy media because our T.38 endpoints are behind NAT.

On Tue, Feb 9, 2010 at 7:26 PM, Peder <peder at networkoblivion.com> wrote:
> Do you find any difference in success rate between using proxy instead of
> bypass for T.38? ?I wouldn't think it would really matter, I am just
> curious.
>

-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com


From jingwei.yang at gmail.com  Tue Feb  9 18:19:21 2010
From: jingwei.yang at gmail.com (Jingwei Yang)
Date: Wed, 10 Feb 2010 10:19:21 +0800
Subject: [Freeswitch-users]  How to record the call upon successful bridge
Message-ID: <13529f9d1002091819y2b363f50vc314d1b6197503b2@mail.gmail.com>

Hi,

I'm using uuid_bridge to bridge two calls. May I know how to start recording
only when the bridge succeeds?

Thanks,
-Jingwei
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100210/3aafdd07/attachment.html 

From frank at carmickle.com  Tue Feb  9 18:46:39 2010
From: frank at carmickle.com (Frank Carmickle)
Date: Tue, 9 Feb 2010 21:46:39 -0500
Subject: [Freeswitch-users] ACL question and js error
In-Reply-To: <cfe871e3305675257b8c74ed4fb5a11b.squirrel@www.webcontracts.co.uk>
References: <ea05d45dca9be2d979fb386f960bc077.squirrel@www.webcontracts.co.uk>
	<20100207145907.GF31942@base.carmickle.com>
	<cfe871e3305675257b8c74ed4fb5a11b.squirrel@www.webcontracts.co.uk>
Message-ID: <20100210024638.GN31942@base.carmickle.com>

On Tue, Feb 09, Matthew Law wrote:
> On Sun, February 7, 2010 2:59 pm, Frank Carmickle wrote:
> > On Sun, Feb 07, Matthew Law wrote:
> >> After some more experiments I have a working replacement for the
> >> asterisk
> >> box we were using before, which is great.
> >>
> >> I had problems getting incoming calls to work. Changing the entry in
> >> acl.conf.xml from:
> >>
> >> <list name="domains" default="deny">
> >>    <node type="allow" domain="$${domain}"/>
> >> </list>
> >>
> >> to:
> >>
> >> <list name="domains" default="allow">
> >>    <node type="allow" domain="$${domain}"/>
> >> </list>
> >>
> >> and reloading xml works but this gets reverted every time FS starts up.
> >> I've scanned the wiki docs and can't see anything pertaining to that.
> >> Why/where is this happening and how do I make it the default?  Actually,
> >> the question should probably be is it sensible to do that? - the box is
> >> out on the internet and I really only want to take incoming calls from
> >> voiptalk.org, but I can't find a list of IPs on their site which I could
> >> create an acl from...
> >
> > This is what gateway definitions are for in sofia.
> 
> I'm still struggling with this.  How where do I tell sofia to allow
> incoming connections from this gateway?
> 
> Here's my sip_profiles/external/voiptalk.org.xml with the sensitive stuff
> removed:
> 
> <include>
>         <gateway name="voiptalk">
>                 <param name="username" value="XXXXXXX" />
>                 <param name="realm" value="voiptalk.org" />
>                 <param name="from-user" value="XXXXXXX"/>
>                 <param name="from-domain" value="voiptalk.org"/>
>                 <param name="password" value="XXXXXXX" />
>                 <param name="extension" value="voiptalk-incoming"/>
>                 <param name="proxy" value="voiptalk.org"/>
>                 <param name="register-proxy" value="voiptalk.org"/>
>                 <param name="expire-seconds" value="600"/>
>                 <param name="register" value="true"/>
>                 <param name="ping" value="25"/>
>         </gateway>
> </include>
> 
> Do I need to add something to this file or maybe sofia.conf.xml to allow
> connections from this domain?  Most everything else is working now, just
> banging my head on this.

After doing a 
sofia profile $profile rescan reloadxml

it still doesn't work?  Are you sure it isn't hitting the dialplan and failing?  I have never used

<param name="extension" value="voiptalk-incoming"/>

I usually leave it commented and then match on the  destination in the dialplan.

<condition field="destination_number" expression="^(4124134655|4128484655|19734226137)$">

HTH
--FC


From andrew at hijacked.us  Tue Feb  9 18:56:07 2010
From: andrew at hijacked.us (Andrew Thompson)
Date: Tue, 9 Feb 2010 21:56:07 -0500
Subject: [Freeswitch-users] Passing channel variables to mod_fifo and
	xml_cdr
In-Reply-To: <87f2f3b91002091455v6af079e1ie28ed6891d5ed628@mail.gmail.com>
References: <004301caa608$534747d0$f9d5d770$@net>
	<87f2f3b91002091455v6af079e1ie28ed6891d5ed628@mail.gmail.com>
Message-ID: <20100210025607.GE27785@hijacked.us>

On Tue, Feb 09, 2010 at 02:55:11PM -0800, Michael Collins wrote:
> On Thu, Feb 4, 2010 at 6:09 PM, Adam Ford <lists at redbonez.net> wrote:
> 
> >  When sending a call through mod_fifo I seem to be losing my custom
> > channel variables that were assigned during prior processing of the call.
> > In my example, I am trying to assign a unique identifier at the time the
> > call enters my FreeSWITCH system in order to more easily tie the xml_cdr
> > logs together.  This works great, until a call is processed through
> > mod_fifo, which drops my custom channel variable in the calls that it
> > generates.  Is it likely that I have something wrong with my config? Or does
> > mod_fifo not support the passing of custom channel variables?
> >
> >
> >
> > The overall problem I am trying to solve is that mod_fifo generates a
> > separate a-leg for every time it rings an agent.  If the agent answers, the
> > a-leg log gets tied to the associated b-leg log with the uuids and I am able
> > to see the entire call in xml_cdr.  However, if the agent rejects the call
> > or doesn?t answer, the a-leg is abandoned with seemingly no association back
> > to a b-leg log.  Anyone have a better suggestion for tying all these xml_cdr
> > logs together for a full picture of the call?
> >
> 
> Just curious - are you looking at this from the caller's perspective or the
> agent's perspective? An unanswered/rejected call from FIFO to an agent
> doesn't tell you very much. However, if you're trying to gather statistics
> on an individual agent then I could see why you'd want to know how many FIFO
> calls they failed to answer. As far as the "new" A leg not being tied back
> to a B leg - Mike J is 100% correct: A FIFO call to an agent has absolutely
> no correlation to any caller waiting in queue. (I suppose the only exception
> to this rule would be if there was only one caller in the queue when the
> FIFO called out to the agents.)
> 
> Like Mike said, FIFO is not ACD. FIFO is "get the caller to a human as
> efficiently and quickly as possible." ACD is more of "connect the longest
> waiting caller to the longest waiting agent, with possible exceptions for
> skills, etc." Check out Andrew Thompson's SpiceCSM for a possible solution
> to your scenario.
> 
> http://www.opencsm.org/wiki/index.php/SpiceCSM_Community_Edition
>

Actually http://github.com/Vagabond/OpenACD is the place to go now
(although its pretty short on documentation right now). I actually
deployed it last friday and its been running fairly well since then.
After I get a few remaining TODOs out of the way I'll cut a 1.0 RC1 and
do a real announcement (and write some documentation).

On a related note, OpenACD tracks agent ringouts and ring cancels so you
can see if an agent is being lazy and bouncing calls.

Andrew


From spiritonly at gmail.com  Tue Feb  9 19:14:46 2010
From: spiritonly at gmail.com (=?UTF-8?B?5p2o5rGf6aqF?=)
Date: Wed, 10 Feb 2010 11:14:46 +0800
Subject: [Freeswitch-users] How to exchange media when I developed new
	endpoint module?
Message-ID: <93b0f8ce1002091914q581f12fvd1de01e2181f0f3a@mail.gmail.com>

Hi,
    I am developping a new endpoint module, now I can make an inbound call
and execute IVR.
When I make an outbound call and bridge the inbound leg and outbound leg, I
receive remote alerting and pickup remote phone but there isn't
any voice exchange.
    So how to exchange media next?
----------------------------------------------------------------------
gtalk:    spiritonly at gmail.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100210/f743acb0/attachment.html 

From jmesquita at freeswitch.org  Tue Feb  9 19:32:28 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Wed, 10 Feb 2010 01:32:28 -0200
Subject: [Freeswitch-users] How to exchange media when I developed new
	endpoint module?
In-Reply-To: <93b0f8ce1002091914q581f12fvd1de01e2181f0f3a@mail.gmail.com>
References: <93b0f8ce1002091914q581f12fvd1de01e2181f0f3a@mail.gmail.com>
Message-ID: <ebbc2d551002091932u52dbf1b6jfecf2dfdb43b8a7c@mail.gmail.com>

You should look at read_frame and write_frame implementations of other
endpoint modules.

This should pretty much tell you how things work...

Jo?o Mesquita


On Wed, Feb 10, 2010 at 1:14 AM, ??? <spiritonly at gmail.com> wrote:

> Hi,
>     I am developping a new endpoint module, now I can make an inbound call
> and execute IVR.
> When I make an outbound call and bridge the inbound leg and outbound leg, I
> receive remote alerting and pickup remote phone but there isn't
> any voice exchange.
>     So how to exchange media next?
> ----------------------------------------------------------------------
> gtalk:    spiritonly at gmail.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100210/86caf75f/attachment-0001.html 

From brian at freeswitch.org  Tue Feb  9 19:44:26 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 9 Feb 2010 21:44:26 -0600
Subject: [Freeswitch-users] How to exchange media when I developed new
	endpoint module?
In-Reply-To: <ebbc2d551002091932u52dbf1b6jfecf2dfdb43b8a7c@mail.gmail.com>
References: <93b0f8ce1002091914q581f12fvd1de01e2181f0f3a@mail.gmail.com>
	<ebbc2d551002091932u52dbf1b6jfecf2dfdb43b8a7c@mail.gmail.com>
Message-ID: <4021C303-1AC3-48F2-A17B-63C02C1C54D4@freeswitch.org>

But the bigger question is what protocol are you doing that you have to create your own endpoint module?  

/b

On Feb 9, 2010, at 9:32 PM, Jo?o Mesquita wrote:

> You should look at read_frame and write_frame implementations of other endpoint modules.
> 
> This should pretty much tell you how things work...
> 
> Jo?o Mesquita



From intralanman at freeswitch.org  Tue Feb  9 20:00:22 2010
From: intralanman at freeswitch.org (Raymond Chandler)
Date: Tue, 9 Feb 2010 23:00:22 -0500
Subject: [Freeswitch-users] How to record the call upon successful bridge
In-Reply-To: <13529f9d1002091819y2b363f50vc314d1b6197503b2@mail.gmail.com>
References: <13529f9d1002091819y2b363f50vc314d1b6197503b2@mail.gmail.com>
Message-ID: <E66ACC04-AFFA-4052-8707-C583ACEDDF7B@freeswitch.org>


On Feb 9, 2010, at 9:19 PM, Jingwei Yang wrote:

> Hi,
> 
> I'm using uuid_bridge to bridge two calls. May I know how to start recording only when the bridge succeeds?
> 

try setting api_hangup_hook to session record

-Ray

From intralanman at freeswitch.org  Tue Feb  9 20:01:03 2010
From: intralanman at freeswitch.org (Raymond Chandler)
Date: Tue, 9 Feb 2010 23:01:03 -0500
Subject: [Freeswitch-users] How to record the call upon successful bridge
In-Reply-To: <13529f9d1002091819y2b363f50vc314d1b6197503b2@mail.gmail.com>
References: <13529f9d1002091819y2b363f50vc314d1b6197503b2@mail.gmail.com>
Message-ID: <257401E3-B32F-4699-96F6-415BA2ADDE88@freeswitch.org>

errr... .execute_on_answer might be better ;-)
-Ray 

On Feb 9, 2010, at 9:19 PM, Jingwei Yang wrote:

> Hi,
> 
> I'm using uuid_bridge to bridge two calls. May I know how to start recording only when the bridge succeeds?
> 
> Thanks,
> -Jingwei
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From mike at jerris.com  Tue Feb  9 20:09:26 2010
From: mike at jerris.com (Michael Jerris)
Date: Tue, 9 Feb 2010 23:09:26 -0500
Subject: [Freeswitch-users] Announcement: FreeSWITCH Add Broadsoft SCA
	Support
In-Reply-To: <767B677E-0D19-43A9-89DE-D96EC4A715EC@freeswitch.org>
References: <87f2f3b91001111742k691966cfs82515db77c13d4d4@mail.gmail.com>
	<d361ca7b1002082110h4e3c60d0p88412f753444ec3d@mail.gmail.com>
	<016d01caa98e$f6df25f0$e49d71d0$@com>
	<d361ca7b1002090817h4aa3c13dka4d4db8b4adc7606@mail.gmail.com>
	<1C0A1BEE-3480-4A33-916F-52067AF5B9C9@freeswitch.org>
	<d361ca7b1002090838j6c42dcd2of8affb5f281900fd@mail.gmail.com>
	<767B677E-0D19-43A9-89DE-D96EC4A715EC@freeswitch.org>
Message-ID: <8D062B6E-4691-4BA6-B038-0EC6E8642DDA@jerris.com>

will it work with multi-domain if you have the aliases for all the domains?

Mike

On Feb 9, 2010, at 12:02 PM, Brian West wrote:

>     <!--<param name="dbname" value="share_presence"/>-->
>     <!--<param name="presence-hosts" value="$${domain}"/>-->
> 
> On Feb 9, 2010, at 10:38 AM, Yehavi Bourvine wrote:
> 
>> What do you mean by "bonding them"?
>>  
>> 2010/2/9 Brian West <brian at freeswitch.org>
>> It will work across profiles if you bond them.  :P

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100209/3deca515/attachment.html 

From mike at jerris.com  Tue Feb  9 20:13:59 2010
From: mike at jerris.com (Michael Jerris)
Date: Tue, 9 Feb 2010 23:13:59 -0500
Subject: [Freeswitch-users] Any Known Dual-Registration Issue?
In-Reply-To: <035001caa9c8$2dca81c0$895f8540$@com>
References: <5AD33F8B1F0848CF9C3A314A60039366@greyhawk.tonecommander.com>
	<035001caa9c8$2dca81c0$895f8540$@com>
Message-ID: <E54E66C0-20C4-4472-B746-29CE498837E8@jerris.com>

As a note, this can cause issues too if you have multiple legitimate registrations from the same ip and port, beware unless you understand the consequences.

Mike

On Feb 9, 2010, at 3:40 PM, Peder wrote:

> What kind of phones?  If you have multiple registartion, this can happen
> sometimes if you reboot a phone.  Crappy phones, like Grandstream, don't
> un-register when you reboot and then when they come back up, they register
> again and thus two registrations until the lifetime of the registration ends
> and it gets flushed.  Changing the multiple-registration to contact can help
> as I believe that uses port and source IP as part of the registration info:
> 
> <param name="multiple-registrations" value="contact"/>



From pablosaro at gmail.com  Tue Feb  9 20:23:35 2010
From: pablosaro at gmail.com (Pablo Hernan Saro)
Date: Wed, 10 Feb 2010 01:23:35 -0300
Subject: [Freeswitch-users] Question regarding testing IVRs
Message-ID: <247f8101002092023t3f30a600o1a9f43771c879e61@mail.gmail.com>

Hi there,

I was wondering if anyone knows about an automated tool for testing IVR
systems... Or should I use SIPP for this purpose?
I will really appreciate your inputs.
Regards

Pablo
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100210/e28d8555/attachment.html 

From brian at freeswitch.org  Tue Feb  9 20:30:06 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 9 Feb 2010 22:30:06 -0600
Subject: [Freeswitch-users] Announcement: FreeSWITCH Add Broadsoft SCA
	Support
In-Reply-To: <8D062B6E-4691-4BA6-B038-0EC6E8642DDA@jerris.com>
References: <87f2f3b91001111742k691966cfs82515db77c13d4d4@mail.gmail.com>
	<d361ca7b1002082110h4e3c60d0p88412f753444ec3d@mail.gmail.com>
	<016d01caa98e$f6df25f0$e49d71d0$@com>
	<d361ca7b1002090817h4aa3c13dka4d4db8b4adc7606@mail.gmail.com>
	<1C0A1BEE-3480-4A33-916F-52067AF5B9C9@freeswitch.org>
	<d361ca7b1002090838j6c42dcd2of8affb5f281900fd@mail.gmail.com>
	<767B677E-0D19-43A9-89DE-D96EC4A715EC@freeswitch.org>
	<8D062B6E-4691-4BA6-B038-0EC6E8642DDA@jerris.com>
Message-ID: <9D328485-D4C3-4B14-808A-CB63BBF06DC1@freeswitch.org>

In theory it should.  I haven't tested that but it should work the same.

/b

On Feb 9, 2010, at 10:09 PM, Michael Jerris wrote:

> will it work with multi-domain if you have the aliases for all the domains?
> 
> Mike



From brian at freeswitch.org  Tue Feb  9 20:30:52 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 9 Feb 2010 22:30:52 -0600
Subject: [Freeswitch-users] Question regarding testing IVRs
In-Reply-To: <247f8101002092023t3f30a600o1a9f43771c879e61@mail.gmail.com>
References: <247f8101002092023t3f30a600o1a9f43771c879e61@mail.gmail.com>
Message-ID: <857B3E7B-0ECA-4B6D-A72E-C0258BCFF24E@freeswitch.org>

Nope.  Unless you know exactly what you're doing its a waste of time.

/b

On Feb 9, 2010, at 10:23 PM, Pablo Hernan Saro wrote:

> Hi there,
> 
> I was wondering if anyone knows about an automated tool for testing IVR systems... Or should I use SIPP for this purpose?
> I will really appreciate your inputs.
> Regards
> 
> Pablo



From mike at jerris.com  Tue Feb  9 20:56:46 2010
From: mike at jerris.com (Michael Jerris)
Date: Tue, 9 Feb 2010 23:56:46 -0500
Subject: [Freeswitch-users] Way to pass max-members from dialplan into
	conference?
In-Reply-To: <87f2f3b91002091509w56eacdedt6a291d9b98838521@mail.gmail.com>
References: <C7136E29C3FD472BB367FF412E282BED@greyhawk.tonecommander.com>
	<87f2f3b91002091509w56eacdedt6a291d9b98838521@mail.gmail.com>
Message-ID: <8965B012-D22D-4155-B76B-CC026DA9D884@jerris.com>

or:

svn commit -m"mod_conference add conference_max_members channel variable that can be set on the first channel calling a conference to override the profiles max-members param"
Sending        mod_conference/mod_conference.c
Transmitting file data .
Committed revision 16597.

compile tested, for your building pleasure.

Mike

On Feb 9, 2010, at 6:09 PM, Michael Collins wrote:

> 
> 
> On Tue, Feb 9, 2010 at 1:14 PM, Robert Hadley <robert.hadley at teotech.com> wrote:
> I have setting max-members=10 in conference.conf.xml working.  However, is there are way to pass in the max-members=10 from the dialplan/default.xml to mod_conference?  I tried using action application=?set? data=?max-members=10? but it didn?t work.  Also tried action application=?export? data=?max-members=10? but it didn?t work either.
> 
>  
> From default.xml:
> 
>     <extension name="nb_conferences">
> 
>       <condition field="destination_number" expression="^(30\d{2})$">
> 
>        <action application="answer"/>
> 
> <!-- tried set and export here to no avail -->
> 
>        <action application="conference" data="$1-${domain_name}@default"/>
> 
>  
> Thanks,
> 
> Robert
> 
> 
> Robert,
> 
> I checked with Brian and also took a look inside mod_conference.c. I didn't see any way that you could override the conference params that are contained in the conference profile. So you'll either need to make a new profile or join the big leagues and start trying out mod_xml_curl. :)
> 
> -MC

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100209/23a0b9f3/attachment.html 

From nagalenoj at gmail.com  Tue Feb  9 21:57:30 2010
From: nagalenoj at gmail.com (Nagalenoj H.)
Date: Wed, 10 Feb 2010 11:27:30 +0530
Subject: [Freeswitch-users] Play music to A leg.
In-Reply-To: <87f2f3b91002090737l4efa8en29bde9457fef7ea5@mail.gmail.com>
References: <fe67a3341002090519r2cc6e8edm7702b8d012fc2be3@mail.gmail.com>
	<87f2f3b91002090737l4efa8en29bde9457fef7ea5@mail.gmail.com>
Message-ID: <fe67a3341002092157x7484f3d2id6678510bbec66cc@mail.gmail.com>

Because, I want to get some digits before bridging the legs. I've tried
group_confirm_key, but it accepts only one digit, I need multiple digits, so
I can't use.
I've also tried group_confirm_file, but when I do originate for multiple
extensions, I want this script to work based on the answered extension.

So, I've originated and processed the events to do my job.

How do I play some music to A leg?

On Tue, Feb 9, 2010 at 9:07 PM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Tue, Feb 9, 2010 at 5:19 AM, Nagalenoj H. <nagalenoj at gmail.com> wrote:
>
>> Dear friends,
>>   In event socket, I'm originating a call to a number from A leg and till
>> the person answers the call, I would want to play some music to the A leg,
>> till I bridge these A leg and originated call.
>>
>>   I don't want to use bridge, in which I could use ringback.
>>
> You don't want to use bridge because... why?
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Regards,
Nagalenoj H.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100210/23fd3b23/attachment-0001.html 

From rm at callrica.co.za  Tue Feb  9 23:31:22 2010
From: rm at callrica.co.za (Roly Maz)
Date: Wed, 10 Feb 2010 09:31:22 +0200
Subject: [Freeswitch-users] Can't access event socket from 0.0.0.0
Message-ID: <00f501caaa23$26068820$72139860$@co.za>

 

I am trying to execute commands over telnet from my XP box  to my Debian FS
box. I have modified the event_socket.conf.xml so that the Listen IP is
0.0.0.0, as per wiki

 

When i try telnet i get a Connect Failed. 

 

If i run netstat on the FS box it shows 127.0.0.1 is listening on port 8021?
It also shows my FS static IP 10.0.18.244 is listening on 5060, which is
correct.

 

I am using the default conf on Freeswitch version 1.0.trunk 16590M  Am i
missing a setting?

 

Any pointers would be much appreciated

 

Roly

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100210/92540339/attachment.html 

From tim at novion.ru  Wed Feb 10 00:35:06 2010
From: tim at novion.ru (Timur Valishev)
Date: Wed, 10 Feb 2010 11:35:06 +0300
Subject: [Freeswitch-users] No reINVITE when bridging two sessions from
	JavaScript with bypass_media_after_bridge=true
In-Reply-To: <154214DA-5781-4FC1-81EA-B41392E6ACF6@jerris.com>
References: <8e9d67561002041147q6c135546u1f7e1887529a916d@mail.gmail.com>
	<3D739D7F-F5C6-4266-B04E-9413DCFDA6AF@freeswitch.org>
	<8e9d67561002050018q24e3a806kc8807b55268e759b@mail.gmail.com>
	<B66289FF-C2A9-4AC4-A56C-66E51B4726F4@freeswitch.org>
	<8e9d67561002051202qeaf1551v8fdd29565f8c02fd@mail.gmail.com>
	<154214DA-5781-4FC1-81EA-B41392E6ACF6@jerris.com>
Message-ID: <8e9d67561002100035n2d14c74dj83f784713c59d542@mail.gmail.com>

Dear Mike,

I'm trying to build a kind of complicated callback. I will be happy if
you suggest alternative way to do it!

Scenario is:

0. Get command over the socket to initiate connection, get A-number,
B-number, route preference, Caller ID option (incognito/normal)
1. Call billing stored procedure to determine maximum call duration
2. Reply over the socket (or better through database?) that connection
is in progrees (to display it on the user GUI)
3. Initiate connection to A-number
4. Upon connection to A, say welcome, start calling B, play ringback tone
5. Upon dialling B, call billing stored procedure to report that
session state changed and report the status to user GUI (over the
socket or through the database?)
5.1 If there was error during connection - speak the reason to end
user (e.g. "Number busy" or "Timeout expired" etc.)
6. Join peers in bypass media mode
6.1 Wait for various commands over the socket - e.g. transfer the
call, put on hold, join to conference etc. User will have GUI for that
operations.
7. If B hangs up, call billing stored procedure to finalize session
and calculate the cost of the call. Cost is to be calculated only by
B-leg length. Speak to A the cost of the call, say thanks and hang up.
7.1 If A hangs up, just call billing and terminate.

Best regards,
Timur Valishev

2010/2/10 Michael Jerris <mike at jerris.com>:
> controlling multiple calls in a script like this is tricky, you need to use
> the first session to create the second one. ?Why are you not just doing an
> originate to do all of this not even in a js file? ?What exactly are you
> trying to accomplish
> Mike
> On Feb 5, 2010, at 3:02 PM, Timur Valishev wrote:
>
> I think we are on the right way) still does not work, but there is hope)
> First of all, this script does not produce any reinvite either (even if
> replace?bypass_media to?bypass_media_after_bridge, or set?bypass_media only
> on one channel):
> <<<<<<<<<<<<<<<<<<<<<<<<<<<<<
> session = new
> Session("{bypass_media=true,ignore_early_media=true,hangup_after_bridge=true}user/1001");
> session2 = new
> Session("{bypass_media=true,ignore_early_media=true}?user/1001");
> bridge(session, session2);
>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>
> BUT! if I run the following script:
> <<<<<<<<<<<<<<<<<<<<<<<<<<<<<
> session = new
> Session("{bypass_media=true,ignore_early_media=true,hangup_after_bridge=true}?user/1001");
> session2 = new
> Session("{bypass_media=true,ignore_early_media=true}user/1001");
>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>
> And then manually type in the console
> uuid_media off <uuid of my session>
> - then I get the reINVITE!
> BUT! When I try to write it to the script:
> <<<<<<<<<<<<<<<<<<<<<<<<<<<<<
> session = new
> Session("{bypass_media=true,ignore_early_media=true,hangup_after_bridge=true}sofia/external/timwork at novion.ru");
> session2 = new
> Session("{bypass_media=true,ignore_early_media=true}sofia/external/timwork at novion.ru");
> bridge(session, session2);
> apiExecute('uuid_media off '+session.uuid); // <-- this line is not
> executed, because bridge hangs up untill BYE
>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>
> the last line is not executed, because bridge hangs up untill BYE
> Then I've tried to do like this:
> <<<<<<<<<<<<<<<<<<<<<<<<<<<<<
> session = new
> Session("{bypass_media=true,ignore_early_media=true,hangup_after_bridge=true}user/1001");
> session2 = new
> Session("{bypass_media=true,ignore_early_media=true}user/1001");
> session.setAutoHangup(false)
> session2.setAutoHangup(false)
> apiExecute("uuid_bridge "+session.uuid+" "+session2.uuid);
> apiExecute('uuid_media off '+session.uuid);
>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>
> But sessions do not get bridged -( Even if I insert session.ready() after
> each call.
> Any ideas on how to call the functions correctly to get the reINVITE?
> Best regards,
> Timur Valishev
> 2010/2/5 Brian West <brian at freeswitch.org>
>>
>> set it inside each of the {} for each session you create its not set after
>> the fact the call is up already... ?you're setting it too late.
>> you an also issue uuid_media off <uuid>
>> /b
>> On Feb 5, 2010, at 2:18 AM, Timur Valishev wrote:
>>
>> I've?modified my script to make sure: <<<<<<<<<<<<<<<<<<<<<<<<<<<<,
>> session = new Session(
>>
>> "{ignore_early_media=true,hangup_after_bridge=true}sofia/external/timwork at novion.ru"
>> );
>> session2 = new Session(
>> "{ignore_early_media=true}sofia/external/timwork at novion.ru"
>> );
>> session.setVariable('bypass_media', 'true');
>> session2.setVariable('bypass_media', 'true');
>> bridge(session, session2);
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From jingwei.yang at gmail.com  Wed Feb 10 00:39:41 2010
From: jingwei.yang at gmail.com (Jingwei Yang)
Date: Wed, 10 Feb 2010 16:39:41 +0800
Subject: [Freeswitch-users] How to record the call upon successful bridge
In-Reply-To: <257401E3-B32F-4699-96F6-415BA2ADDE88@freeswitch.org>
References: <13529f9d1002091819y2b363f50vc314d1b6197503b2@mail.gmail.com>
	<257401E3-B32F-4699-96F6-415BA2ADDE88@freeswitch.org>
Message-ID: <13529f9d1002100039q43775563r49b11136ddda0a0@mail.gmail.com>

Hi Ray,

Thanks a lot for the replies. Allow me to elaborate a little bit about my
situation.

1. client A calls in and parks at Fifo myq.
2. FS connets Agent B to an extension (via originate
skypiax/ANY/jingwei.yang 33333)
3. uuid_bridge client A and agent B

I believe the spice I can add is in the extension 33333. Here's how I define
it.

    <extension name="agent_hold">
      <condition field="destination_number" expression="^33333$">
        <action application="set" data="*execute_on_answer*=record_session
/tmp/33333.wav"/>
        <action application="answer"/>
        <action application="sleep" data="2000"/>
        <action application="info"/>
        <action application="playback"
data="ivr/ivr-hold_connect_call.wav"/>
      </condition>
    </extension>

But the wav file didn't get generated at all. Please advise whether the
above is in correct usage.

I've also tried bridge_pre_execute_bleg_app and bridge_pre_execute_bleg_data

        <action application="set"
data="bridge_pre_execute_bleg_app=record_session"/>
        <action application="set"
data="bridge_pre_execute_bleg_data=/tmp/33333.wav"/>

The audio file didn't appear either.

The only successful method is by using record_session directly like this:

        <action application="record_session" data="/tmp/33333.wav"/>

However, in this form, the wav file includes the waiting music, which is not
ideal.

Thanks and best regards,
-Jingwei

On Wed, Feb 10, 2010 at 12:01 PM, Raymond Chandler <
intralanman at freeswitch.org> wrote:

> errr... .execute_on_answer might be better ;-)
> -Ray
>
> On Feb 9, 2010, at 9:19 PM, Jingwei Yang wrote:
>
> > Hi,
> >
> > I'm using uuid_bridge to bridge two calls. May I know how to start
> recording only when the bridge succeeds?
> >
> > Thanks,
> > -Jingwei
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100210/1493da48/attachment.html 

From xanlich at gmail.com  Wed Feb 10 01:17:30 2010
From: xanlich at gmail.com (Chia-Yen Wu)
Date: Wed, 10 Feb 2010 17:17:30 +0800
Subject: [Freeswitch-users] Lua script hangup detect
Message-ID: <314dc3f81002100117h6b1bc49cna989624a1ae4e0e2@mail.gmail.com>

Hello,
i tried to use Lua script to replace xml macro in dialplan,
but I found out that Lua wont terminate if client hangup,
,so the session is still on but client is already hangup,
is there a way to avoid this ?
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100210/39f27e2f/attachment.html 

From kond at nstel.ru  Wed Feb 10 01:40:14 2010
From: kond at nstel.ru (Nikolay Kondratyev)
Date: Wed, 10 Feb 2010 12:40:14 +0300
Subject: [Freeswitch-users] can't see sip messages in the log file
Message-ID: <20100210094014.C746F11F81@mail.nstel.ru>

Hi all,

I have a little problem with FreeSWITCH Version 1.0.5-20100209-0400
(16587M).

I'd like to see SIP messages in the log file. and I tried 

sofia profile internal siptrace on

sofia loglevel all 9

console loglevel 9

but alas i can only see what SDP is used..

The configuration is default with the exception for some new local
extensions and mod_h323 compiled in.

 Am I missing something?

 

Can anybody please advise how to include sip messages into the log file?

(I thought that "sofia profile internal siptrace on" should be enough, but
alas..)

 

Thanks in advance,

Nikolay.

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100210/4c31d3cf/attachment-0001.html 

From jason at jasonjgw.net  Wed Feb 10 02:02:43 2010
From: jason at jasonjgw.net (Jason White)
Date: Wed, 10 Feb 2010 21:02:43 +1100
Subject: [Freeswitch-users] can't see sip messages in the log file
In-Reply-To: <20100210094014.C746F11F81@mail.nstel.ru>
References: <20100210094014.C746F11F81@mail.nstel.ru>
Message-ID: <20100210100243.GA16435@jdc.jasonjgw.net>

Nikolay Kondratyev <kond at nstel.ru> wrote:
> Can anybody please advise how to include sip messages into the log file?

<param name="sip-trace" value="yes"/>
in the SIP profile you want to trace, then
sofia profile profile-name restart reloadxml
or restarting FreeSWITCH should do it.



From rm at callrica.co.za  Wed Feb 10 02:12:27 2010
From: rm at callrica.co.za (Roly Maz)
Date: Wed, 10 Feb 2010 12:12:27 +0200
Subject: [Freeswitch-users] Can't access event socket from 0.0.0.0
In-Reply-To: <00f501caaa23$26068820$72139860$@co.za>
References: <00f501caaa23$26068820$72139860$@co.za>
Message-ID: <012601caaa39$a79ca840$f6d5f8c0$@co.za>

I have figured this out...newbie Linux error -I was making my changes in the
usr/src/freeswitch folder - i should have been using the
usr/local/freeswitch folder.  

 

Duh!

 

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Roly Maz
Sent: 10 February 2010 09:31 AM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Can't access event socket from 0.0.0.0

 

 

I am trying to execute commands over telnet from my XP box  to my Debian FS
box. I have modified the event_socket.conf.xml so that the Listen IP is
0.0.0.0, as per wiki

 

When i try telnet i get a Connect Failed. 

 

If i run netstat on the FS box it shows 127.0.0.1 is listening on port 8021?
It also shows my FS static IP 10.0.18.244 is listening on 5060, which is
correct.

 

I am using the default conf on Freeswitch version 1.0.trunk 16590M  Am i
missing a setting?

 

Any pointers would be much appreciated

 

Roly

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100210/779f53e4/attachment.html 

From jingwei.yang at gmail.com  Wed Feb 10 02:38:18 2010
From: jingwei.yang at gmail.com (Jingwei Yang)
Date: Wed, 10 Feb 2010 18:38:18 +0800
Subject: [Freeswitch-users] Can't access event socket from 0.0.0.0
In-Reply-To: <012601caaa39$a79ca840$f6d5f8c0$@co.za>
References: <00f501caaa23$26068820$72139860$@co.za>
	<012601caaa39$a79ca840$f6d5f8c0$@co.za>
Message-ID: <13529f9d1002100238p55bd97c2mb5b64f3a6d3c9521@mail.gmail.com>

Totally understandable. I've made the same mistake before :)

On Wed, Feb 10, 2010 at 6:12 PM, Roly Maz <rm at callrica.co.za> wrote:

>  I have figured this out...newbie Linux error ?I was making my changes in
> the usr/src/freeswitch folder ? i should have been using the
> usr/local/freeswitch folder.
>
>
>
> Duh!
>
>
>
>
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Roly Maz
> *Sent:* 10 February 2010 09:31 AM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* [Freeswitch-users] Can't access event socket from 0.0.0.0
>
>
>
>
>
> I am trying to execute commands over telnet from my XP box  to my Debian FS
> box. I have modified the event_socket.conf.xml so that the Listen IP is
> 0.0.0.0, as per wiki
>
>
>
> When i try telnet i get a Connect Failed.
>
>
>
> If i run netstat on the FS box it shows 127.0.0.1 is listening on port
> 8021? It also shows my FS static IP 10.0.18.244 is listening on 5060, which
> is correct.
>
>
>
> I am using the default conf on Freeswitch version 1.0.trunk 16590M  Am i
> missing a setting?
>
>
>
> Any pointers would be much appreciated
>
>
>
> Roly
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100210/985b5365/attachment.html 

From vmaruani at interwise.com  Wed Feb 10 02:44:24 2010
From: vmaruani at interwise.com (Victor Maruani)
Date: Wed, 10 Feb 2010 12:44:24 +0200
Subject: [Freeswitch-users] Bypass-media and REFER method
In-Reply-To: <B816EF85C732094BA2016B2112F3DC2B0402C777@isr-brass.interwise.com>
References: <B816EF85C732094BA2016B2112F3DC2B0402C777@isr-brass.interwise.com>
Message-ID: <B816EF85C732094BA2016B2112F3DC2B040C5CA6@isr-brass.interwise.com>

Hi,

 

I can't have a blind transfer work properly if I use bypass-media=true.

 

My first message may have been unclear, here I added excerpt from the
dialplan:

 

                <!--- ext MyIVR -->

                 <extension name="Extension MyIVR" continue="false">

                   <condition field="destination_number"
expression="^337$">

                                   <action application="set"
data="bypass_media=true"/>

                                   <action application="bridge"
data="sofia/internal/337 at 10.10.5.51"/>

                   </condition>

                 </extension>

 

 

                <!--- ext MyServ -->

                 <extension name="Extension MyServ " continue="false">

                   <condition field="destination_number"
expression="^3341$">

                                                <action
application="set" data="bypass_media=true"/>

                                   <action application="bridge"
data="sofia/internal/3341 at 10.10.5.48"/>

                   </condition>

                 </extension>   

 

The connection to MyIVR works. Then it sends Refer-to (3341 at ...) which
fails as I described it in the previous mail.

 

I would like to know if the feature has been validated and if I'm
missing something in the configuration.

 

Any help would be very appreciated.

 

Thanks!

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Victor Maruani
Sent: Sunday, February 07, 2010 5:01 PM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Bypass-media and REFER method

 

Hi,

 

I'm trying to do a POC using FS, the goal is to have FS handle REFERs
containing proprietary data. 

I want to have some logic on top of FS and also use the fail over
mechanism.

in short, I have something like this:

(third party) A side  --- FS ---- B side (IVR server)

 

the IVR the sends a REFER to FS. I don't want A to deal with it. 

now say B refers to C, it would be considered as a "group" C1, C2 ... to
which I want FS to failover.

only when one has answered should A be updated (REINVITE) and B notified
and disconnected.

if all fails I would expect B to be notified of the failure and proceed
as I wish without "losing" A.

 

from what I've read FS should be OK for the job but I have a couple
issues:

 

1 ) I have some issues getting FS handle a REFER while in bypass-media
mode.

(I tried with the release and some revisions including latest)

first when I bridge A and B everything is fine and media is bypassed.

When B sends REFER to C:

-           FS immediately NOTIFY B of success and send a reinvite to A
with SDP containing its own media IP/port.

-          then it does INVITE C with A's SDP.

-          B gets disconnected. A is not updated with C's sdp.

so at this point A sends RTP to FS and C sends RTP to A. ...

 

I basically have one extension for B: (set bypass-media and bridge to B)

and another extension to C which does the same actions.

what do you think I do wrong? 

 

 

2 ) how can I catch the REFER and set variables from it? (like ref-by or
ref-to)

in the dial plan I do catch the INVITE sent to C, but how to do it with
the REFER itself?

 

 

thanks for your help!

 

 

Best Regards,

Victor.

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100210/15e25966/attachment-0001.html 

From kond at nstel.ru  Wed Feb 10 02:49:05 2010
From: kond at nstel.ru (Nikolay Kondratyev)
Date: Wed, 10 Feb 2010 13:49:05 +0300
Subject: [Freeswitch-users] h323 - sip call is not working
Message-ID: <20100210104911.A632A11F49@mail.nstel.ru>

Hi all,

I compiled FreeSWITCH Version 1.0.5-20100209-0400 (16587M) with mod_h323.

When I call from h323 (Avaya IPOffice) to local fs extention (x-lite) I hear
ring back, but when x-lite picks up, he hears silence, while IPOffice user
continues to hear ringback.

The log is at the http://pastebin.freeswitch.org/12091

My configuration is almost default, several local extentions added, and
h323.conf from http://wiki.freeswitch.org/wiki/Mod_h323

5840 - user at IPOffice

2853 - x-lite registered at FS

IPOffice ip address: 172.23.14.2

FS ip address 172.23.22.49

 

Can anybody please advise how to solve that?

Is it a configuration or a software problem?

 

Thanks in advance,

Nikolay.

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100210/c4517f38/attachment.html 

From nazim.agabekov at gmail.com  Wed Feb 10 02:58:29 2010
From: nazim.agabekov at gmail.com (Nazim Agabekov)
Date: Wed, 10 Feb 2010 14:58:29 +0400
Subject: [Freeswitch-users] Lua script hangup detect
In-Reply-To: <314dc3f81002100117h6b1bc49cna989624a1ae4e0e2@mail.gmail.com>
References: <314dc3f81002100117h6b1bc49cna989624a1ae4e0e2@mail.gmail.com>
Message-ID: <4B729155.7010708@gmail.com>

Hello,
Can you pastebin your script?

http://pastebin.freeswitch.org

On 02/10/2010 01:17 PM, Chia-Yen Wu wrote:
> Hello, 
> i tried to use Lua script to replace xml macro in dialplan,
> but I found out that Lua wont terminate if client hangup,
> ,so the session is still on but client is already hangup,
> is there a way to avoid this ?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100210/1f577801/attachment.html 

From kond at nstel.ru  Wed Feb 10 03:00:08 2010
From: kond at nstel.ru (Nikolay Kondratyev)
Date: Wed, 10 Feb 2010 14:00:08 +0300
Subject: [Freeswitch-users] can't see sip messages in the log file
In-Reply-To: <20100210100243.GA16435@jdc.jasonjgw.net>
Message-ID: <20100210110008.25BFD12292@mail.nstel.ru>

Jason, thanks for the reply.
Isn't "sofia profile internal siptrace on" a command line equivalent of
<param name="sip-trace" value="yes"/>?

Any way I tried it, but with the same result.
I still don't see SIP.

Thanks and regards,
Nikolay.

> > Can anybody please advise how to include sip messages into the log file?
> 
> <param name="sip-trace" value="yes"/>
> in the SIP profile you want to trace, then
> sofia profile profile-name restart reloadxml
> or restarting FreeSWITCH should do it.



From lakindia89 at gmail.com  Wed Feb 10 03:07:31 2010
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Wed, 10 Feb 2010 16:37:31 +0530
Subject: [Freeswitch-users] How to kill multiple UUIDs.
Message-ID: <7d79b3931002100307h4d370a22l6f04cafa5bf5806d@mail.gmail.com>

Hi all,

My situation is
    A called to 1005 -- Which executes an ESL program.
    Now from the program I will made the parallel call using "api originate
[origination_uuid=uuid1]user/1006,[origination_uuid2]user/1001 &park()".
    UUID's are obtained from create_uuid.

    I'll then wait for the api to return, to check whether the call is
answered or rejected by the other end.
    But while I'm waiting, if A hangup the call, I just want to kill the
calls that are originated by my program.
    So I taught of using api_hang_up_hook and I set that variable to
uuid_kill uuid1 uuid2.
    But it only killed the uuid1.

Is there any other ways to kill multiple uuid's??
please help?
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100210/e374f564/attachment.html 

From kond at nstel.ru  Wed Feb 10 03:08:01 2010
From: kond at nstel.ru (Nikolay Kondratyev)
Date: Wed, 10 Feb 2010 14:08:01 +0300
Subject: [Freeswitch-users] h323 - sip call is not working
In-Reply-To: <20100210104911.A632A11F49@mail.nstel.ru>
Message-ID: <20100210110812.01E6512112@mail.nstel.ru>

I forgot to add, that from tcpdump trace (and from log) one can see that FS
does not send answer to IPOffice.

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Nikolay
Kondratyev
Sent: Wednesday, February 10, 2010 1:49 PM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] h323 - sip call is not working

 

Hi all,

I compiled FreeSWITCH Version 1.0.5-20100209-0400 (16587M) with mod_h323.

When I call from h323 (Avaya IPOffice) to local fs extention (x-lite) I hear
ring back, but when x-lite picks up, he hears silence, while IPOffice user
continues to hear ringback.

The log is at the http://pastebin.freeswitch.org/12091

My configuration is almost default, several local extentions added, and
h323.conf from http://wiki.freeswitch.org/wiki/Mod_h323

5840 - user at IPOffice

2853 - x-lite registered at FS

IPOffice ip address: 172.23.14.2

FS ip address 172.23.22.49

 

Can anybody please advise how to solve that?

Is it a configuration or a software problem?

 

Thanks in advance,

Nikolay.

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100210/bdcf3ad6/attachment.html 

From nazim.agabekov at gmail.com  Wed Feb 10 03:28:19 2010
From: nazim.agabekov at gmail.com (Nazim Agabekov)
Date: Wed, 10 Feb 2010 15:28:19 +0400
Subject: [Freeswitch-users] Lua script hangup detect
In-Reply-To: <314dc3f81002100117h6b1bc49cna989624a1ae4e0e2@mail.gmail.com>
References: <314dc3f81002100117h6b1bc49cna989624a1ae4e0e2@mail.gmail.com>
Message-ID: <4B729853.1090206@gmail.com>

Generally you could avoid this by checking session status. Below is an
example.

function selectAge (langId)

   session:flushDigits()

   digits = ""
   while ("" == digits) do

      if not session:ready() then return nil end

      digits = session:playAndGetDigits(2, 3, 1, 5000, "#",
              wav_base .. langId .. "/" .. age_prompt_wav,
              wav_base .. langId .. "/" .. age_incorrect_wav,
              "\\d+");

      log("info", "Got dtmf: ".. digits .."\n");
      if "" ~= digits then
         if (age_max < tonumber(digits) or age_min > tonumber(digits) ) then
             log("info", "Incorrect age: ".. digits ..".. retrying\n");
             session:streamFile (wav_base .. langId .. "/" ..
age_incorrect_wav)
             digits = ""
         end
      end

   end

   log("info", "Got age dtmf: ".. digits .."\n");
   return tonumber(digits)
end



On 02/10/2010 01:17 PM, Chia-Yen Wu wrote:
> Hello, 
> i tried to use Lua script to replace xml macro in dialplan,
> but I found out that Lua wont terminate if client hangup,
> ,so the session is still on but client is already hangup,
> is there a way to avoid this ?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100210/a8705c42/attachment-0001.html 

From brian at freeswitch.org  Wed Feb 10 06:24:45 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 10 Feb 2010 08:24:45 -0600
Subject: [Freeswitch-users] Lua script hangup detect
In-Reply-To: <4B729853.1090206@gmail.com>
References: <314dc3f81002100117h6b1bc49cna989624a1ae4e0e2@mail.gmail.com>
	<4B729853.1090206@gmail.com>
Message-ID: <3302DFBC-8D82-460C-BC9A-11D309E50012@freeswitch.org>

Just wrap it in a while(session:ready() == true)

/b



On Feb 10, 2010, at 5:28 AM, Nazim Agabekov wrote:

> session:ready()



From kond at nstel.ru  Wed Feb 10 06:25:04 2010
From: kond at nstel.ru (Nikolay Kondratyev)
Date: Wed, 10 Feb 2010 17:25:04 +0300
Subject: [Freeswitch-users] can't see sip messages in the log file
In-Reply-To: <20100210110008.25BFD12292@mail.nstel.ru>
Message-ID: <20100210142456.2DA9A11F4A@mail.nstel.ru>

After some experiments i clarified this question.
SIP messages go into the freeswitch log when:
<param name="sip-trace" value="yes"/> ("sofia profile profile-name siptrace
on" is cli equivalent for this parameter)
AND
Sofia tracelevel is set to info (sofia tracelevel info). 

The default sofia tracelevel is 'console'. That's why I did not see sip
messages in the log after turning on just "siptrace".

Thanks and regards,
Nikolay.

> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-
> users-bounces at lists.freeswitch.org] On Behalf Of Nikolay Kondratyev
> Sent: Wednesday, February 10, 2010 2:00 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] can't see sip messages in the log file
> 
> Jason, thanks for the reply.
> Isn't "sofia profile internal siptrace on" a command line equivalent of
> <param name="sip-trace" value="yes"/>?
> 
> Any way I tried it, but with the same result.
> I still don't see SIP.
> 
> Thanks and regards,
> Nikolay.
> 
> > > Can anybody please advise how to include sip messages into the log
> file?
> >
> > <param name="sip-trace" value="yes"/>
> > in the SIP profile you want to trace, then
> > sofia profile profile-name restart reloadxml
> > or restarting FreeSWITCH should do it.
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From anthony.minessale at gmail.com  Wed Feb 10 06:45:36 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 10 Feb 2010 08:45:36 -0600
Subject: [Freeswitch-users] Bypass-media and REFER method
In-Reply-To: <B816EF85C732094BA2016B2112F3DC2B040C5CA6@isr-brass.interwise.com>
References: <B816EF85C732094BA2016B2112F3DC2B0402C777@isr-brass.interwise.com>
	<B816EF85C732094BA2016B2112F3DC2B040C5CA6@isr-brass.interwise.com>
Message-ID: <191c3a031002100645n5d827949xdefd24143f10759@mail.gmail.com>

update to latest trunk and reproduce your problem with full debug enabled.

sofia profile internal siptrace on
console loglevel debug


On Wed, Feb 10, 2010 at 4:44 AM, Victor Maruani <vmaruani at interwise.com>wrote:

>  Hi,
>
>
>
> I can't have a blind transfer work properly if I use bypass-media=true.
>
>
>
> My first message may have been unclear, here I added excerpt from the
> dialplan:
>
>
>
>                 <!--- ext MyIVR -->
>
>                  <extension name="Extension MyIVR" continue="false">
>
>                    <condition field="destination_number"
> expression="^337$">
>
>                                    <action application="set"
> data="bypass_media=true"/>
>
>                                    <action application="bridge"
> data="sofia/internal/337 at 10.10.5.51"/>
>
>                    </condition>
>
>                  </extension>
>
>
>
>
>
>                 <!--- ext MyServ -->
>
>                  <extension name="Extension MyServ " continue="false">
>
>                    <condition field="destination_number"
> expression="^3341$">
>
>                                                 <action application="set"
> data="bypass_media=true"/>
>
>                                    <action application="bridge"
> data="sofia/internal/3341 at 10.10.5.48"/>
>
>                    </condition>
>
>                  </extension>
>
>
>
> The connection to MyIVR works. Then it sends Refer-to (3341 at ...) which
> fails as I described it in the previous mail.
>
>
>
> I would like to know if the feature has been validated and if I'm missing
> something in the configuration.
>
>
>
> Any help would be very appreciated.
>
>
>
> Thanks!
>
>
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Victor
> Maruani
> *Sent:* Sunday, February 07, 2010 5:01 PM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* [Freeswitch-users] Bypass-media and REFER method
>
>
>
> Hi,
>
>
>
> I'm trying to do a POC using FS, the goal is to have FS handle REFERs
> containing proprietary data.
>
> I want to have some logic on top of FS and also use the fail over
> mechanism.
>
> in short, I have something like this:
>
> (third party) A side  --- FS ---- B side (IVR server)
>
>
>
> the IVR the sends a REFER to FS. I don't want A to deal with it.
>
> now say B refers to C, it would be considered as a "group" C1, C2 ... to
> which I want FS to failover.
>
> only when one has answered should A be updated (REINVITE) and B notified
> and disconnected.
>
> if all fails I would expect B to be notified of the failure and proceed as
> I wish without "losing" A.
>
>
>
> from what I've read FS should be OK for the job but I have a couple issues:
>
>
>
> 1 ) I have some issues getting FS handle a REFER while in bypass-media
> mode.
>
> (I tried with the release and some revisions including latest)
>
> first when I bridge A and B everything is fine and media is bypassed.
>
> When B sends REFER to C:
>
> -           FS immediately NOTIFY B of success and send a reinvite to A
> with SDP containing its own media IP/port.
>
> -          then it does INVITE C with A's SDP.
>
> -          B gets disconnected. A is not updated with C's sdp.
>
> so at this point A sends RTP to FS and C sends RTP to A. ?
>
>
>
> I basically have one extension for B: (set bypass-media and bridge to B)
>
> and another extension to C which does the same actions.
>
> what do you think I do wrong?
>
>
>
>
>
> 2 ) how can I catch the REFER and set variables from it? (like ref-by or
> ref-to)
>
> in the dial plan I do catch the INVITE sent to C, but how to do it with the
> REFER itself?
>
>
>
>
>
> thanks for your help!
>
>
>
>
>
> Best Regards,
>
> Victor.
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100210/a0487c3e/attachment.html 

From rupa at rupa.com  Wed Feb 10 06:50:00 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Wed, 10 Feb 2010 08:50:00 -0600
Subject: [Freeswitch-users] can't see sip messages in the log file
In-Reply-To: <20100210142456.2DA9A11F4A@mail.nstel.ru>
References: <20100210110008.25BFD12292@mail.nstel.ru>
	<20100210142456.2DA9A11F4A@mail.nstel.ru>
Message-ID: <b39273531002100650j2ff8652cu733256c88e5b7aea@mail.gmail.com>

Thanks.  I've added that to the sofia help/completion.

On Wed, Feb 10, 2010 at 8:25 AM, Nikolay Kondratyev <kond at nstel.ru> wrote:

> After some experiments i clarified this question.
> SIP messages go into the freeswitch log when:
> <param name="sip-trace" value="yes"/> ("sofia profile profile-name siptrace
> on" is cli equivalent for this parameter)
> AND
> Sofia tracelevel is set to info (sofia tracelevel info).
>
> The default sofia tracelevel is 'console'. That's why I did not see sip
> messages in the log after turning on just "siptrace".
>
> Thanks and regards,
> Nikolay.
>
> > -----Original Message-----
> > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-
> > users-bounces at lists.freeswitch.org] On Behalf Of Nikolay Kondratyev
> > Sent: Wednesday, February 10, 2010 2:00 PM
> > To: freeswitch-users at lists.freeswitch.org
> > Subject: Re: [Freeswitch-users] can't see sip messages in the log file
> >
> > Jason, thanks for the reply.
> > Isn't "sofia profile internal siptrace on" a command line equivalent of
> > <param name="sip-trace" value="yes"/>?
> >
> > Any way I tried it, but with the same result.
> > I still don't see SIP.
> >
> > Thanks and regards,
> > Nikolay.
> >
> > > > Can anybody please advise how to include sip messages into the log
> > file?
> > >
> > > <param name="sip-trace" value="yes"/>
> > > in the SIP profile you want to trace, then
> > > sofia profile profile-name restart reloadxml
> > > or restarting FreeSWITCH should do it.
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
-Rupa
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100210/553704c1/attachment-0001.html 

From msc at freeswitch.org  Wed Feb 10 07:02:02 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 10 Feb 2010 07:02:02 -0800
Subject: [Freeswitch-users] Play music to A leg.
In-Reply-To: <fe67a3341002092157x7484f3d2id6678510bbec66cc@mail.gmail.com>
References: <fe67a3341002090519r2cc6e8edm7702b8d012fc2be3@mail.gmail.com>
	<87f2f3b91002090737l4efa8en29bde9457fef7ea5@mail.gmail.com>
	<fe67a3341002092157x7484f3d2id6678510bbec66cc@mail.gmail.com>
Message-ID: <87f2f3b91002100702u7581589ua866d2492a008b23@mail.gmail.com>

On Tue, Feb 9, 2010 at 9:57 PM, Nagalenoj H. <nagalenoj at gmail.com> wrote:

> Because, I want to get some digits before bridging the legs. I've tried
> group_confirm_key, but it accepts only one digit, I need multiple digits, so
> I can't use.
> I've also tried group_confirm_file, but when I do originate for multiple
> extensions, I want this script to work based on the answered extension.
>
> So, I've originated and processed the events to do my job.
>
> How do I play some music to A leg?
>
> I might be missing something, but couldn't you just park the call ("A leg")
until you connect to the other party ("B leg") and then uuid_bridge at
whatever point you want?
-MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100210/0936b48b/attachment.html 

From msc at freeswitch.org  Wed Feb 10 07:09:44 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 10 Feb 2010 07:09:44 -0800
Subject: [Freeswitch-users] How to record the call upon successful bridge
In-Reply-To: <13529f9d1002100039q43775563r49b11136ddda0a0@mail.gmail.com>
References: <13529f9d1002091819y2b363f50vc314d1b6197503b2@mail.gmail.com>
	<257401E3-B32F-4699-96F6-415BA2ADDE88@freeswitch.org>
	<13529f9d1002100039q43775563r49b11136ddda0a0@mail.gmail.com>
Message-ID: <87f2f3b91002100709m32d975b5rf16823e2b02bd8c2@mail.gmail.com>

Make sure that you are on the latest SVN trunk and retest. Pastebin the
complete debug log from start to finish of the call.
-MC

On Wed, Feb 10, 2010 at 12:39 AM, Jingwei Yang <jingwei.yang at gmail.com>wrote:

> Hi Ray,
>
> Thanks a lot for the replies. Allow me to elaborate a little bit about my
> situation.
>
> 1. client A calls in and parks at Fifo myq.
> 2. FS connets Agent B to an extension (via originate
> skypiax/ANY/jingwei.yang 33333)
> 3. uuid_bridge client A and agent B
>
> I believe the spice I can add is in the extension 33333. Here's how I
> define it.
>
>     <extension name="agent_hold">
>       <condition field="destination_number" expression="^33333$">
>         <action application="set" data="*execute_on_answer*=record_session
> /tmp/33333.wav"/>
>         <action application="answer"/>
>         <action application="sleep" data="2000"/>
>         <action application="info"/>
>         <action application="playback"
> data="ivr/ivr-hold_connect_call.wav"/>
>       </condition>
>     </extension>
>
> But the wav file didn't get generated at all. Please advise whether the
> above is in correct usage.
>
> I've also tried bridge_pre_execute_bleg_app and
> bridge_pre_execute_bleg_data
>
>         <action application="set"
> data="bridge_pre_execute_bleg_app=record_session"/>
>         <action application="set"
> data="bridge_pre_execute_bleg_data=/tmp/33333.wav"/>
>
> The audio file didn't appear either.
>
> The only successful method is by using record_session directly like this:
>
>         <action application="record_session" data="/tmp/33333.wav"/>
>
> However, in this form, the wav file includes the waiting music, which is
> not ideal.
>
> Thanks and best regards,
> -Jingwei
>
>
> On Wed, Feb 10, 2010 at 12:01 PM, Raymond Chandler <
> intralanman at freeswitch.org> wrote:
>
>> errr... .execute_on_answer might be better ;-)
>> -Ray
>>
>> On Feb 9, 2010, at 9:19 PM, Jingwei Yang wrote:
>>
>> > Hi,
>> >
>> > I'm using uuid_bridge to bridge two calls. May I know how to start
>> recording only when the bridge succeeds?
>> >
>> > Thanks,
>> > -Jingwei
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100210/e8d54d30/attachment.html 

From mike at jerris.com  Wed Feb 10 07:14:22 2010
From: mike at jerris.com (Michael Jerris)
Date: Wed, 10 Feb 2010 10:14:22 -0500
Subject: [Freeswitch-users] No reINVITE when bridging two sessions from
	JavaScript with bypass_media_after_bridge=true
In-Reply-To: <8e9d67561002100035n2d14c74dj83f784713c59d542@mail.gmail.com>
References: <8e9d67561002041147q6c135546u1f7e1887529a916d@mail.gmail.com>
	<3D739D7F-F5C6-4266-B04E-9413DCFDA6AF@freeswitch.org>
	<8e9d67561002050018q24e3a806kc8807b55268e759b@mail.gmail.com>
	<B66289FF-C2A9-4AC4-A56C-66E51B4726F4@freeswitch.org>
	<8e9d67561002051202qeaf1551v8fdd29565f8c02fd@mail.gmail.com>
	<154214DA-5781-4FC1-81EA-B41392E6ACF6@jerris.com>
	<8e9d67561002100035n2d14c74dj83f784713c59d542@mail.gmail.com>
Message-ID: <E789278F-9A2F-462A-823D-8361F327E684@jerris.com>

I would suggest using scripts in between the bridge attempts when you need to do logic, but to do most of this in dialplan where you can.  so through 3, you can handle however you were, then drip A back to dialplan to bridge the call (continue_on_fail, hangup_after_bridge=false), then after bridge, call another script if you need to do more advanced logic.


On Feb 10, 2010, at 3:35 AM, Timur Valishev wrote:

> Dear Mike,
> 
> I'm trying to build a kind of complicated callback. I will be happy if
> you suggest alternative way to do it!
> 
> Scenario is:
> 
> 0. Get command over the socket to initiate connection, get A-number,
> B-number, route preference, Caller ID option (incognito/normal)
> 1. Call billing stored procedure to determine maximum call duration
> 2. Reply over the socket (or better through database?) that connection
> is in progrees (to display it on the user GUI)
> 3. Initiate connection to A-number
> 4. Upon connection to A, say welcome, start calling B, play ringback tone
> 5. Upon dialling B, call billing stored procedure to report that
> session state changed and report the status to user GUI (over the
> socket or through the database?)
> 5.1 If there was error during connection - speak the reason to end
> user (e.g. "Number busy" or "Timeout expired" etc.)
> 6. Join peers in bypass media mode
> 6.1 Wait for various commands over the socket - e.g. transfer the
> call, put on hold, join to conference etc. User will have GUI for that
> operations.
> 7. If B hangs up, call billing stored procedure to finalize session
> and calculate the cost of the call. Cost is to be calculated only by
> B-leg length. Speak to A the cost of the call, say thanks and hang up.
> 7.1 If A hangs up, just call billing and terminate.
> 
> Best regards,
> Timur Valishev
> 
> 2010/2/10 Michael Jerris <mike at jerris.com>:
>> controlling multiple calls in a script like this is tricky, you need to use
>> the first session to create the second one.  Why are you not just doing an
>> originate to do all of this not even in a js file?  What exactly are you
>> trying to accomplish
>> Mike



From mike at jerris.com  Wed Feb 10 07:16:09 2010
From: mike at jerris.com (Michael Jerris)
Date: Wed, 10 Feb 2010 10:16:09 -0500
Subject: [Freeswitch-users] How to kill multiple UUIDs.
In-Reply-To: <7d79b3931002100307h4d370a22l6f04cafa5bf5806d@mail.gmail.com>
References: <7d79b3931002100307h4d370a22l6f04cafa5bf5806d@mail.gmail.com>
Message-ID: <A35CD010-E86F-4B7E-A28B-2A9000B8E009@jerris.com>

you can api hangup hook to call

lua multi_kill.lua uuid1 uuid2 uuid?.

and then write the trivial lua script for that.

Mike

On Feb 10, 2010, at 6:07 AM, lakshmanan ganapathy wrote:

> Hi all,
> 
> My situation is
>     A called to 1005 -- Which executes an ESL program.
>     Now from the program I will made the parallel call using "api originate [origination_uuid=uuid1]user/1006,[origination_uuid2]user/1001 &park()".
>     UUID's are obtained from create_uuid.
> 
>     I'll then wait for the api to return, to check whether the call is answered or rejected by the other end.
>     But while I'm waiting, if A hangup the call, I just want to kill the calls that are originated by my program.
>     So I taught of using api_hang_up_hook and I set that variable to uuid_kill uuid1 uuid2.
>     But it only killed the uuid1.
> 
> Is there any other ways to kill multiple uuid's??
> please help?



From mike at jerris.com  Wed Feb 10 07:28:34 2010
From: mike at jerris.com (Michael Jerris)
Date: Wed, 10 Feb 2010 10:28:34 -0500
Subject: [Freeswitch-users] UPnP Timeout
In-Reply-To: <b39273531002091455meb77162o2baefc3866efaed9@mail.gmail.com>
References: <89dbfdc31002090652h884bc2av82d53c6958fdc409@mail.gmail.com>
	<b39273531002091455meb77162o2baefc3866efaed9@mail.gmail.com>
Message-ID: <4BB6DD8A-995D-4A36-AE92-1C15E5F70876@jerris.com>

Rupa-

	Can you offer up a tcpdump command he could run on the box that would catch all of this for later diagnosis if it happens again?

On Feb 9, 2010, at 5:55 PM, Rupa Schomaker wrote:

> That is a different keep-alive.  I'm specifically talking about the keep-alive packet that we get via upnp multicast.  Whenever we receive one from the gateway we republish the nat mappings to.. um... keep them alive. :)
> 
> On Tue, Feb 9, 2010 at 8:52 AM, Kim Culhan <w8hdkim at gmail.com> wrote:
> On Mon, February 8, 2010 11:21 pm, Troy Anderson wrote:
> > I have been using pfSense (1.2.3) and FS.  FS nicely uses UPnP to poke
> > holes in the firewall, but it seems that the holes close after a while.  I
> > cannot find any documentation in FS nor in pfSense as to what the timeout
> > is.  Is there a setting in FS to do some kind of keep-alive thing with
> > UPnP to keep, e.g. 5060, open?  Or is it already doing that and pfSense is
> > the issue?
> 
> FS has provisions for keep-alive, see the bottom of the page for ping
> time value:
> 
> http://wiki.freeswitch.org/wiki/SIP_Provider_Examples
> 
> To watch the pf firewall hole timing you can install pftop from
> FreeBSD ports/sysutils
> which displays the filter states 'and more'.
> 
> -kim
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> -- 
> -Rupa
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100210/f96cffe0/attachment.html 

From rupa at rupa.com  Wed Feb 10 07:53:44 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Wed, 10 Feb 2010 09:53:44 -0600
Subject: [Freeswitch-users] UPnP Timeout
In-Reply-To: <4BB6DD8A-995D-4A36-AE92-1C15E5F70876@jerris.com>
References: <89dbfdc31002090652h884bc2av82d53c6958fdc409@mail.gmail.com>
	<b39273531002091455meb77162o2baefc3866efaed9@mail.gmail.com>
	<4BB6DD8A-995D-4A36-AE92-1C15E5F70876@jerris.com>
Message-ID: <b39273531002100753n77132cbai8e4a61ada8736179@mail.gmail.com>

sure,

for upnp:

tcpdump -w trace.pcap 'host 239.255.255.250'

for nat-pmp:

tcpdump -w trace.pcap 'host 224.0.0.1'

I updated the wiki in the troubleshooting section.

On Wed, Feb 10, 2010 at 9:28 AM, Michael Jerris <mike at jerris.com> wrote:

> Rupa-
>
> Can you offer up a tcpdump command he could run on the box that would catch
> all of this for later diagnosis if it happens again?
>
> On Feb 9, 2010, at 5:55 PM, Rupa Schomaker wrote:
>
> That is a different keep-alive.  I'm specifically talking about the
> keep-alive packet that we get via upnp multicast.  Whenever we receive one
> from the gateway we republish the nat mappings to.. um... keep them alive.
> :)
>
> On Tue, Feb 9, 2010 at 8:52 AM, Kim Culhan <w8hdkim at gmail.com> wrote:
>
>> On Mon, February 8, 2010 11:21 pm, Troy Anderson wrote:
>> > I have been using pfSense (1.2.3) and FS.  FS nicely uses UPnP to poke
>> > holes in the firewall, but it seems that the holes close after a while.
>>  I
>> > cannot find any documentation in FS nor in pfSense as to what the
>> timeout
>> > is.  Is there a setting in FS to do some kind of keep-alive thing with
>> > UPnP to keep, e.g. 5060, open?  Or is it already doing that and pfSense
>> is
>> > the issue?
>>
>> FS has provisions for keep-alive, see the bottom of the page for ping
>> time value:
>>
>> http://wiki.freeswitch.org/wiki/SIP_Provider_Examples
>>
>> To watch the pf firewall hole timing you can install pftop from
>> FreeBSD ports/sysutils
>> which displays the filter states 'and more'.
>>
>> -kim
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> -Rupa
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100210/80ceb8ac/attachment-0001.html 

From anthony.minessale at gmail.com  Wed Feb 10 07:59:43 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 10 Feb 2010 09:59:43 -0600
Subject: [Freeswitch-users] How to kill multiple UUIDs.
In-Reply-To: <A35CD010-E86F-4B7E-A28B-2A9000B8E009@jerris.com>
References: <7d79b3931002100307h4d370a22l6f04cafa5bf5806d@mail.gmail.com>
	<A35CD010-E86F-4B7E-A28B-2A9000B8E009@jerris.com>
Message-ID: <191c3a031002100759v2fef0f48na5afccf46bb08311@mail.gmail.com>

or you can set a common var like foo=bar on all the chans and do

hupall normal_clearing foo bar


On Wed, Feb 10, 2010 at 9:16 AM, Michael Jerris <mike at jerris.com> wrote:

> you can api hangup hook to call
>
> lua multi_kill.lua uuid1 uuid2 uuid?.
>
> and then write the trivial lua script for that.
>
> Mike
>
> On Feb 10, 2010, at 6:07 AM, lakshmanan ganapathy wrote:
>
> > Hi all,
> >
> > My situation is
> >     A called to 1005 -- Which executes an ESL program.
> >     Now from the program I will made the parallel call using "api
> originate [origination_uuid=uuid1]user/1006,[origination_uuid2]user/1001
> &park()".
> >     UUID's are obtained from create_uuid.
> >
> >     I'll then wait for the api to return, to check whether the call is
> answered or rejected by the other end.
> >     But while I'm waiting, if A hangup the call, I just want to kill the
> calls that are originated by my program.
> >     So I taught of using api_hang_up_hook and I set that variable to
> uuid_kill uuid1 uuid2.
> >     But it only killed the uuid1.
> >
> > Is there any other ways to kill multiple uuid's??
> > please help?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100210/efde6758/attachment.html 

From xanlich at gmail.com  Wed Feb 10 08:00:39 2010
From: xanlich at gmail.com (Chia-Yen Wu)
Date: Thu, 11 Feb 2010 00:00:39 +0800
Subject: [Freeswitch-users] Lua script hangup detect
In-Reply-To: <4B729155.7010708@gmail.com>
References: <314dc3f81002100117h6b1bc49cna989624a1ae4e0e2@mail.gmail.com>
	<4B729155.7010708@gmail.com>
Message-ID: <314dc3f81002100800o41fd6561w394d5e83e69d3d80@mail.gmail.com>

thx for reply, but shouldnt Lua script terminate when client hangs up?

maybe it will stuck in a few situation before it reach to session:status
check point.

like GetDigit or something else? (sorry, I dont have FS right now, need test
it tomorrow)

2010/2/10 Nazim Agabekov <nazim.agabekov at gmail.com>

>  Hello,
> Can you pastebin your script?
>
> http://pastebin.freeswitch.org
>
>
> On 02/10/2010 01:17 PM, Chia-Yen Wu wrote:
>
> Hello,
>  i tried to use Lua script to replace xml macro in dialplan,
>  but I found out that Lua wont terminate if client hangup,
> ,so the session is still on but client is already hangup,
> is there a way to avoid this ?
>
>
> _______________________________________________
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/0e329d98/attachment.html 

From jerry.richards at teotech.com  Wed Feb 10 08:13:26 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Wed, 10 Feb 2010 08:13:26 -0800
Subject: [Freeswitch-users] Any Known Dual-Registration Issue?
In-Reply-To: <035001caa9c8$2dca81c0$895f8540$@com>
References: <5AD33F8B1F0848CF9C3A314A60039366@greyhawk.tonecommander.com>
	<035001caa9c8$2dca81c0$895f8540$@com>
Message-ID: <3017421995484996A379AADB394D48CB@greyhawk.tonecommander.com>

I am using Bria Pro softphones, but it is possible that the PC was
disconnected before a restart.  Any phone does not necessarily send
unregister, for example if there is a power outage or it is disconnected
from the network.  Anyway, I'll just try setting multiple-registrations to
contact as you suggest.

Thanks,
Jerry
 

-----Original Message-----
From: Peder [mailto:peder at networkoblivion.com] 
Sent: Tuesday, February 09, 2010 12:41 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Any Known Dual-Registration Issue?

What kind of phones?  If you have multiple registartion, this can happen
sometimes if you reboot a phone.  Crappy phones, like Grandstream, don't
un-register when you reboot and then when they come back up, they register
again and thus two registrations until the lifetime of the registration ends
and it gets flushed.  Changing the multiple-registration to contact can help
as I believe that uses port and source IP as part of the registration info:

<param name="multiple-registrations" value="contact"/>



-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jerry
Richards
Sent: Tuesday, February 09, 2010 2:27 PM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Any Known Dual-Registration Issue?

I've noticed that sometimes my phones end up with two registrations with two
Call-IDs at Freeswitch.  Is there any known bug that would cause this?  I've
seen it on different phone models, so I'm thinking there is some timing
issue with Freeswitch.

Best Regards,
Jerry


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org






From nazim.agabekov at gmail.com  Wed Feb 10 08:34:01 2010
From: nazim.agabekov at gmail.com (Nazim Agabekov)
Date: Wed, 10 Feb 2010 20:34:01 +0400
Subject: [Freeswitch-users] Lua script hangup detect
In-Reply-To: <314dc3f81002100800o41fd6561w394d5e83e69d3d80@mail.gmail.com>
References: <314dc3f81002100117h6b1bc49cna989624a1ae4e0e2@mail.gmail.com>	<4B729155.7010708@gmail.com>
	<314dc3f81002100800o41fd6561w394d5e83e69d3d80@mail.gmail.com>
Message-ID: <4B72DFF9.4040402@gmail.com>

Lua script is not terminating immediately on hangup.
This behavior allows user to finalize the script nicely (free
dynamically allocated resources, update logs, e.t.c)
> maybe it will stuck in a few situation before it reach to
> session:status check point.
>
> like GetDigit or something else? (sorry, I dont have FS right now,
> need test it tomorrow)
I've never encountered such a problem. Usually GetDigit-like functions
have timeout parameter, so they don't block forever.
Just check the session status often and it will work like a charm ;


On 02/10/2010 08:00 PM, Chia-Yen Wu wrote:
> thx for reply, but shouldnt Lua script terminate when client hangs up?
>
> maybe it will stuck in a few situation before it reach to
> session:status check point.
>
> like GetDigit or something else? (sorry, I dont have FS right now,
> need test it tomorrow)
>
> 2010/2/10 Nazim Agabekov <nazim.agabekov at gmail.com
> <mailto:nazim.agabekov at gmail.com>>
>
>     Hello,
>     Can you pastebin your script?
>
>     http://pastebin.freeswitch.org
>
>
>     On 02/10/2010 01:17 PM, Chia-Yen Wu wrote:
>>     Hello, 
>>     i tried to use Lua script to replace xml macro in dialplan,
>>     but I found out that Lua wont terminate if client hangup,
>>     ,so the session is still on but client is already hangup,
>>     is there a way to avoid this ?
>>
>>
>>     _______________________________________________
>>     FreeSWITCH-users mailing list
>>     FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     http://www.freeswitch.org
>>       
>
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100210/d0c8e3eb/attachment.html 

From vmaruani at interwise.com  Wed Feb 10 09:01:47 2010
From: vmaruani at interwise.com (Victor Maruani)
Date: Wed, 10 Feb 2010 19:01:47 +0200
Subject: [Freeswitch-users] Bypass-media and REFER method
In-Reply-To: <191c3a031002100645n5d827949xdefd24143f10759@mail.gmail.com>
References: <B816EF85C732094BA2016B2112F3DC2B0402C777@isr-brass.interwise.com><B816EF85C732094BA2016B2112F3DC2B040C5CA6@isr-brass.interwise.com>
	<191c3a031002100645n5d827949xdefd24143f10759@mail.gmail.com>
Message-ID: <B816EF85C732094BA2016B2112F3DC2B040C6061@isr-brass.interwise.com>

Hi,

 

Logs are on pb 12099

I hope this helps.

Reproduced with revision 16599.

 

A-side (10.10.5.19) is an x-lite registered with extension 1002

B (.5.51) refers to C (.5.48) none are registered.

 

Please refer to previous emails for details of dialplan and what I try
to do...

Let me know if you need more info 

 

Thanks!

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Anthony Minessale
Sent: Wednesday, February 10, 2010 4:46 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Bypass-media and REFER method

 

update to latest trunk and reproduce your problem with full debug
enabled.

sofia profile internal siptrace on
console loglevel debug



On Wed, Feb 10, 2010 at 4:44 AM, Victor Maruani <vmaruani at interwise.com>
wrote:

Hi,

 

I can't have a blind transfer work properly if I use bypass-media=true.

 

My first message may have been unclear, here I added excerpt from the
dialplan:

 

                <!--- ext MyIVR -->

                 <extension name="Extension MyIVR" continue="false">

                   <condition field="destination_number"
expression="^337$">

                                   <action application="set"
data="bypass_media=true"/>

                                   <action application="bridge"
data="sofia/internal/337 at 10.10.5.51"/>

                   </condition>

                 </extension>

 

 

                <!--- ext MyServ -->

                 <extension name="Extension MyServ " continue="false">

                   <condition field="destination_number"
expression="^3341$">

                                                <action
application="set" data="bypass_media=true"/>

                                   <action application="bridge"
data="sofia/internal/3341 at 10.10.5.48"/>

                   </condition>

                 </extension>   

 

The connection to MyIVR works. Then it sends Refer-to (3341 at ...) which
fails as I described it in the previous mail.

 

I would like to know if the feature has been validated and if I'm
missing something in the configuration.

 

Any help would be very appreciated.

 

Thanks!

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Victor Maruani
Sent: Sunday, February 07, 2010 5:01 PM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Bypass-media and REFER method

 

Hi,

 

I'm trying to do a POC using FS, the goal is to have FS handle REFERs
containing proprietary data. 

I want to have some logic on top of FS and also use the fail over
mechanism.

in short, I have something like this:

(third party) A side  --- FS ---- B side (IVR server)

 

the IVR the sends a REFER to FS. I don't want A to deal with it. 

now say B refers to C, it would be considered as a "group" C1, C2 ... to
which I want FS to failover.

only when one has answered should A be updated (REINVITE) and B notified
and disconnected.

if all fails I would expect B to be notified of the failure and proceed
as I wish without "losing" A.

 

from what I've read FS should be OK for the job but I have a couple
issues:

 

1 ) I have some issues getting FS handle a REFER while in bypass-media
mode.

(I tried with the release and some revisions including latest)

first when I bridge A and B everything is fine and media is bypassed.

When B sends REFER to C:

-           FS immediately NOTIFY B of success and send a reinvite to A
with SDP containing its own media IP/port.

-          then it does INVITE C with A's SDP.

-          B gets disconnected. A is not updated with C's sdp.

so at this point A sends RTP to FS and C sends RTP to A. ...

 

I basically have one extension for B: (set bypass-media and bridge to B)

and another extension to C which does the same actions.

what do you think I do wrong? 

 

 

2 ) how can I catch the REFER and set variables from it? (like ref-by or
ref-to)

in the dial plan I do catch the INVITE sent to C, but how to do it with
the REFER itself?

 

 

thanks for your help!

 

 

Best Regards,

Victor.

 


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
<mailto:MSN%3Aanthony_minessale at hotmail.com> 
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
<mailto:PAYPAL%3Aanthony.minessale at gmail.com> 
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:+19193869900

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100210/6ccad925/attachment-0001.html 

From red.rain.seven at gmail.com  Wed Feb 10 09:16:51 2010
From: red.rain.seven at gmail.com (Henry Huang)
Date: Thu, 11 Feb 2010 01:16:51 +0800
Subject: [Freeswitch-users] How to kill multiple UUIDs.
In-Reply-To: <191c3a031002100759v2fef0f48na5afccf46bb08311@mail.gmail.com>
References: <7d79b3931002100307h4d370a22l6f04cafa5bf5806d@mail.gmail.com>
	<A35CD010-E86F-4B7E-A28B-2A9000B8E009@jerris.com>
	<191c3a031002100759v2fef0f48na5afccf46bb08311@mail.gmail.com>
Message-ID: <59ad9ca11002100916l6982c88r22e1b45af55b80a1@mail.gmail.com>

is there an example of setting common var for multiple channels?
or just do it normally like set var=foo before all the bridged call I would
like to hang up at the same time?
and the var name can be anything I want?


Henry

On Wed, Feb 10, 2010 at 11:59 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> or you can set a common var like foo=bar on all the chans and do
>
> hupall normal_clearing foo bar
>
>
>
> On Wed, Feb 10, 2010 at 9:16 AM, Michael Jerris <mike at jerris.com> wrote:
>
>> you can api hangup hook to call
>>
>> lua multi_kill.lua uuid1 uuid2 uuid?.
>>
>> and then write the trivial lua script for that.
>>
>> Mike
>>
>> On Feb 10, 2010, at 6:07 AM, lakshmanan ganapathy wrote:
>>
>> > Hi all,
>> >
>> > My situation is
>> >     A called to 1005 -- Which executes an ESL program.
>> >     Now from the program I will made the parallel call using "api
>> originate [origination_uuid=uuid1]user/1006,[origination_uuid2]user/1001
>> &park()".
>> >     UUID's are obtained from create_uuid.
>> >
>> >     I'll then wait for the api to return, to check whether the call is
>> answered or rejected by the other end.
>> >     But while I'm waiting, if A hangup the call, I just want to kill the
>> calls that are originated by my program.
>> >     So I taught of using api_hang_up_hook and I set that variable to
>> uuid_kill uuid1 uuid2.
>> >     But it only killed the uuid1.
>> >
>> > Is there any other ways to kill multiple uuid's??
>> > please help?
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/620ba28a/attachment.html 

From nicolas at medularis.com  Wed Feb 10 09:32:08 2010
From: nicolas at medularis.com (Nicolas Brenner)
Date: Wed, 10 Feb 2010 14:32:08 -0300
Subject: [Freeswitch-users] Lua script hangup detect
In-Reply-To: <4B72DFF9.4040402@gmail.com>
References: <314dc3f81002100117h6b1bc49cna989624a1ae4e0e2@mail.gmail.com>
	<4B729155.7010708@gmail.com>
	<314dc3f81002100800o41fd6561w394d5e83e69d3d80@mail.gmail.com>
	<4B72DFF9.4040402@gmail.com>
Message-ID: <1b46b4e81002100932x7af486ekd4b05f4750cace20@mail.gmail.com>

On Wed, Feb 10, 2010 at 1:34 PM, Nazim Agabekov <nazim.agabekov at gmail.com>wrote:

>  Lua script is not terminating immediately on hangup.
>


When does it terminate then? Will the script terminate when it finishes
running or does it need some special instruction?



> This behavior allows user to finalize the script nicely (free dynamically
> allocated resources, update logs, e.t.c)
>
>  maybe it will stuck in a few situation before it reach to session:status
> check point.
>
>  like GetDigit or something else? (sorry, I dont have FS right now, need
> test it tomorrow)
>
> I've never encountered such a problem. Usually GetDigit-like functions have
> timeout parameter, so they don't block forever.
> Just check the session status often and it will work like a charm ;
>
>
>
> On 02/10/2010 08:00 PM, Chia-Yen Wu wrote:
>
> thx for reply, but shouldnt Lua script terminate when client hangs up?
>
>  maybe it will stuck in a few situation before it reach to session:status
> check point.
>
>  like GetDigit or something else? (sorry, I dont have FS right now, need
> test it tomorrow)
>
> 2010/2/10 Nazim Agabekov <nazim.agabekov at gmail.com>
>
>> Hello,
>> Can you pastebin your script?
>>
>> http://pastebin.freeswitch.org
>>
>>
>> On 02/10/2010 01:17 PM, Chia-Yen Wu wrote:
>>
>>  Hello,
>>  i tried to use Lua script to replace xml macro in dialplan,
>>  but I found out that Lua wont terminate if client hangup,
>> ,so the session is still on but client is already hangup,
>> is there a way to avoid this ?
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100210/40e6ff7e/attachment.html 

From nazim.agabekov at gmail.com  Wed Feb 10 09:53:44 2010
From: nazim.agabekov at gmail.com (Nazim Agabekov)
Date: Wed, 10 Feb 2010 21:53:44 +0400
Subject: [Freeswitch-users] Lua script hangup detect
In-Reply-To: <1b46b4e81002100932x7af486ekd4b05f4750cace20@mail.gmail.com>
References: <314dc3f81002100117h6b1bc49cna989624a1ae4e0e2@mail.gmail.com>	<4B729155.7010708@gmail.com>	<314dc3f81002100800o41fd6561w394d5e83e69d3d80@mail.gmail.com>	<4B72DFF9.4040402@gmail.com>
	<1b46b4e81002100932x7af486ekd4b05f4750cace20@mail.gmail.com>
Message-ID: <4B72F2A8.4070503@gmail.com>

I think it continues to run until it finishes. I'll check it tomorrow on
my test system.

On 02/10/2010 09:32 PM, Nicolas Brenner wrote:
>
> On Wed, Feb 10, 2010 at 1:34 PM, Nazim Agabekov
> <nazim.agabekov at gmail.com <mailto:nazim.agabekov at gmail.com>> wrote:
>
>     Lua script is not terminating immediately on hangup.
>
>
>
> When does it terminate then? Will the script terminate when it
> finishes running or does it need some special instruction?
>
>  
>
>     This behavior allows user to finalize the script nicely (free
>     dynamically allocated resources, update logs, e.t.c)
>
>>     maybe it will stuck in a few situation before it reach to
>>     session:status check point.
>>
>>     like GetDigit or something else? (sorry, I dont have FS right
>>     now, need test it tomorrow)
>     I've never encountered such a problem. Usually GetDigit-like
>     functions have timeout parameter, so they don't block forever.
>     Just check the session status often and it will work like a charm ;
>
>
>
>     On 02/10/2010 08:00 PM, Chia-Yen Wu wrote:
>>     thx for reply, but shouldnt Lua script terminate when client
>>     hangs up?
>>
>>     maybe it will stuck in a few situation before it reach to
>>     session:status check point.
>>
>>     like GetDigit or something else? (sorry, I dont have FS right
>>     now, need test it tomorrow)
>>
>>     2010/2/10 Nazim Agabekov <nazim.agabekov at gmail.com
>>     <mailto:nazim.agabekov at gmail.com>>
>>
>>         Hello,
>>         Can you pastebin your script?
>>
>>         http://pastebin.freeswitch.org
>>
>>
>>         On 02/10/2010 01:17 PM, Chia-Yen Wu wrote:
>>>         Hello, 
>>>         i tried to use Lua script to replace xml macro in dialplan,
>>>         but I found out that Lua wont terminate if client hangup,
>>>         ,so the session is still on but client is already hangup,
>>>         is there a way to avoid this ?
>>>
>>>
>>>         _______________________________________________
>>>         FreeSWITCH-users mailing list
>>>         FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>         http://www.freeswitch.org
>>>           
>>
>>
>>         _______________________________________________
>>         FreeSWITCH-users mailing list
>>         FreeSWITCH-users at lists.freeswitch.org
>>         <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>         http://www.freeswitch.org
>>
>>
>>
>>     _______________________________________________
>>     FreeSWITCH-users mailing list
>>     FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     http://www.freeswitch.org
>>       
>
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100210/03aa76a0/attachment-0001.html 

From Prometheus001 at gmx.net  Wed Feb 10 10:44:23 2010
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Wed, 10 Feb 2010 19:44:23 +0100
Subject: [Freeswitch-users] Skypiax latency
Message-ID: <4B72FE87.4000401@gmx.net>

Hello,

I have a problem with latency and mod_skypiax

Skype=>SIP is always fine (~0.3sec)
SIP => Skype is always bad (~2-4 sec)
I would expect that latency in both directions should be the same.

Anybody has discovered this before and has a solution?

The scenario is as follows:

SIP-UA => FreeswitchServer=>Freeswitch/SkypIaxServer=>Skype
Both freeswitch servers are in the same LAN, so latency should be low.

Best regards
Peter


From robert.hadley at teotech.com  Wed Feb 10 10:50:15 2010
From: robert.hadley at teotech.com (Robert Hadley)
Date: Wed, 10 Feb 2010 10:50:15 -0800
Subject: [Freeswitch-users] demo_ivr cannot find sound files via relative
	paths
Message-ID: <78030B9E01DD48F8B024A97035F96282@greyhawk.tonecommander.com>

Hi,

 

It appears a recent change (possibly the new sounds_dir variable or the new
ivr_menu folder?) may have broken relative sound file paths in the IVR.  I
built a today's trunk version and installed to the default location. Using
the default conf files the demo_ivr cannot find files based on the relative
paths specified in ivr_menus/demo_ivr.xml.

 

[root at TEO-UCM-T2 conf]# cat ivr_menus/demo_ivr.xml

<include>

  <!-- demo IVR setup -->

  <!-- demo IVR, Main Menu -->

  <menu name="demo_ivr"

      greet-long="phrase:demo_ivr_main_menu"

      greet-short="phrase:demo_ivr_main_menu_short"

      invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"

      exit-sound="voicemail/vm-goodbye.wav"

 

 

2010-02-10 10:32:15.300337 [DEBUG] switch_ivr_menu.c:414 Executing IVR menu
demo_ivr

2010-02-10 10:32:15.300337 [DEBUG] switch_ivr_play_say.c:118 No language
specified - Using [en]

2010-02-10 10:32:15.300337 [DEBUG] switch_ivr_play_say.c:273 Handle
play-file:[ivr/ivr-welcome_to_freeswitch.wav] (en:en)

2010-02-10 10:32:15.300337 [DEBUG] switch_ivr_play_say.c:1158 Codec
Activated L16 at 8000hz 1 channels 20ms

2010-02-10 10:32:17.922147 [DEBUG] switch_ivr_play_say.c:1450 done playing
file

2010-02-10 10:32:18.042149 [DEBUG] switch_ivr_play_say.c:273 Handle
play-file:[ivr/ivr-this_ivr_will_let_you_test_features.wav] (en:en)

2010-02-10 10:32:18.042149 [DEBUG] switch_ivr_play_say.c:1158 Codec
Activated L16 at 8000hz 1 channels 20ms

2010-02-10 10:32:19.962158 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF 2:800

2010-02-10 10:32:19.962158 [DEBUG] switch_ivr_play_say.c:1450 done playing
file

2010-02-10 10:32:20.082156 [DEBUG] switch_ivr_menu.c:329 waiting for 3/4
digits t/o 2000

2010-02-10 10:32:20.120617 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF 2:400

2010-02-10 10:32:20.442158 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF 2:800

2010-02-10 10:32:20.682162 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF 2:800

2010-02-10 10:32:20.682162 [DEBUG] switch_ivr_menu.c:376 digits '2222'

2010-02-10 10:32:20.682162 [DEBUG] switch_ivr_menu.c:470 action regex [2222]
[/^(10[01][0-9])$/] [0]

2010-02-10 10:32:20.682162 [DEBUG] switch_ivr_menu.c:560 IVR menu 'demo_ivr'
caught invalid input '2222'

2010-02-10 10:32:20.682162 [ERR] mod_sndfile.c:194 Error Opening File
[/usr/local/freeswitch/sounds/ivr/ivr-that_was_an_invalid_entry.wav] [System
error : No such file or directory.]

2010-02-10 10:32:21.701505 [DEBUG] switch_ivr_play_say.c:118 No language
specified - Using [en]

2010-02-10 10:32:21.701505 [DEBUG] switch_ivr_play_say.c:273 Handle
play-file:[silence_stream://1000] (en:en)

2010-02-10 10:32:21.701505 [DEBUG] switch_ivr_play_say.c:1158 Codec
Activated L16 at 8000hz 1 channels 20ms

2010-02-10 10:32:22.700352 [DEBUG] switch_ivr_play_say.c:1450 done playing
file

 

Regards,

Robert

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100210/4e17a0df/attachment.html 

From woof at iwoof.org  Wed Feb 10 11:11:32 2010
From: woof at iwoof.org (Andy Spitzer)
Date: Wed, 10 Feb 2010 14:11:32 -0500
Subject: [Freeswitch-users] Question regarding testing IVRs
In-Reply-To: <247f8101002092023t3f30a600o1a9f43771c879e61@mail.gmail.com>
References: <247f8101002092023t3f30a600o1a9f43771c879e61@mail.gmail.com>
Message-ID: <op.u7xd9iyog0dksq@knights-macbook.fuzzy>

Woof!

On Tue, 09 Feb 2010 23:23:35 -0500, Pablo Hernan Saro  
<pablosaro at gmail.com> wrote:

> I was wondering if anyone knows about an automated tool for testing IVR  
> systems

There are the expensive systems out there you can spend $$$ on (e.g.  
Empirix Hammer), or you can just code up some scripts that drive a second  
instance of FreeSWITCH to place calls, dial digits, etc.  I prefer the  
latter, especially for simple load testing.  If you instrument your IVR so  
that these scripts can externally determine what state each call is in  
(often just by polling the log files), then this can be an excellent tool  
for more complicated testing scenarios as well.

But beware...writing a decent IVR test script (for any platform, be it  
home brew or commercial), can end up being about an order of magnitude  
more complicated than the building the original system under test.  This  
is because while an IVR system can just wait for something to happen or  
timeout, the test system should track response times, statistics, vary  
it's input but still be within certain ranges, query the same database the  
IVR is using to know about account numbers, state, etc.

For many cases, just automating simple calls to generate load, and then  
having a person call in and exercise the complicated bits of the IVR while  
under load will get you pretty far.

--Woof!


From tculjaga at gmail.com  Wed Feb 10 11:13:51 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Wed, 10 Feb 2010 20:13:51 +0100
Subject: [Freeswitch-users] h323 - sip call is not working
In-Reply-To: <20100210104911.A632A11F49@mail.nstel.ru>
References: <20100210104911.A632A11F49@mail.nstel.ru>
Message-ID: <65d96fc81002101113p44fe6f43oba72c3a2e13785cc@mail.gmail.com>

On Wed, Feb 10, 2010 at 11:49 AM, Nikolay Kondratyev <kond at nstel.ru> wrote:

>  Hi all,
>
> I compiled FreeSWITCH Version 1.0.5-20100209-0400 (16587M) with mod_h323.
>
> When I call from h323 (Avaya IPOffice) to local fs extention (x-lite) I
> hear ring back, but when x-lite picks up, he hears silence, while IPOffice
> user continues to hear ringback.
>
> The log is at the http://pastebin.freeswitch.org/12091
>
> My configuration is almost default, several local extentions added, and
> h323.conf from http://wiki.freeswitch.org/wiki/Mod_h323
>
> 5840 ? user at IPOffice
>
> 2853 ? x-lite registered at FS
>
> IPOffice ip address: 172.23.14.2
>
> FS ip address 172.23.22.49
>
>
>
> Can anybody please advise how to solve that?
>
> Is it a configuration or a software problem?
>
>
>

Please can you send me the tcpdump as well (not filtered). IPOffice i known
to have a "broken" H323 stack.
did you try to play with tunneling and h245 in setup settings as well ?

It looks like there is some h245 negotiation still pending but cant see that
from the logs.


> Thanks in advance,
>
> Nikolay.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100210/eec71392/attachment.html 

From anthony.minessale at gmail.com  Wed Feb 10 11:29:26 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 10 Feb 2010 13:29:26 -0600
Subject: [Freeswitch-users] How to kill multiple UUIDs.
In-Reply-To: <59ad9ca11002100916l6982c88r22e1b45af55b80a1@mail.gmail.com>
References: <7d79b3931002100307h4d370a22l6f04cafa5bf5806d@mail.gmail.com>
	<A35CD010-E86F-4B7E-A28B-2A9000B8E009@jerris.com>
	<191c3a031002100759v2fef0f48na5afccf46bb08311@mail.gmail.com>
	<59ad9ca11002100916l6982c88r22e1b45af55b80a1@mail.gmail.com>
Message-ID: <191c3a031002101129h1103c2e7l6100b2d41807fddf@mail.gmail.com>

if you put a var in the leading {} it will get set on all channels

{foo=bar}sofia/internal/test at server.com


On Wed, Feb 10, 2010 at 11:16 AM, Henry Huang <red.rain.seven at gmail.com>wrote:

> is there an example of setting common var for multiple channels?
> or just do it normally like set var=foo before all the bridged call I would
> like to hang up at the same time?
> and the var name can be anything I want?
>
>
> Henry
>
>
> On Wed, Feb 10, 2010 at 11:59 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> or you can set a common var like foo=bar on all the chans and do
>>
>> hupall normal_clearing foo bar
>>
>>
>>
>> On Wed, Feb 10, 2010 at 9:16 AM, Michael Jerris <mike at jerris.com> wrote:
>>
>>> you can api hangup hook to call
>>>
>>> lua multi_kill.lua uuid1 uuid2 uuid?.
>>>
>>> and then write the trivial lua script for that.
>>>
>>> Mike
>>>
>>> On Feb 10, 2010, at 6:07 AM, lakshmanan ganapathy wrote:
>>>
>>> > Hi all,
>>> >
>>> > My situation is
>>> >     A called to 1005 -- Which executes an ESL program.
>>> >     Now from the program I will made the parallel call using "api
>>> originate [origination_uuid=uuid1]user/1006,[origination_uuid2]user/1001
>>> &park()".
>>> >     UUID's are obtained from create_uuid.
>>> >
>>> >     I'll then wait for the api to return, to check whether the call is
>>> answered or rejected by the other end.
>>> >     But while I'm waiting, if A hangup the call, I just want to kill
>>> the calls that are originated by my program.
>>> >     So I taught of using api_hang_up_hook and I set that variable to
>>> uuid_kill uuid1 uuid2.
>>> >     But it only killed the uuid1.
>>> >
>>> > Is there any other ways to kill multiple uuid's??
>>> > please help?
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100210/97461df1/attachment-0001.html 

From peter.olsson at visionutveckling.se  Wed Feb 10 11:42:55 2010
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Wed, 10 Feb 2010 20:42:55 +0100
Subject: [Freeswitch-users] h323 - sip call is not working
In-Reply-To: <65d96fc81002101113p44fe6f43oba72c3a2e13785cc@mail.gmail.com>
References: <20100210104911.A632A11F49@mail.nstel.ru>,
	<65d96fc81002101113p44fe6f43oba72c3a2e13785cc@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C5576CE4923@cooper>

I've been running both h323 and SIP between FS and Avaya IPO for some time. No problems at all. :)

But make sure to enable h323 fast start and disable "direct media path" in the IPO, if I remember correctly these where the only two parameters that made any real difference for me.

But I do recommenf to use SIP, since it's much better supported by FS.

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Tihomir Culjaga [tculjaga at gmail.com]
Skickat: den 10 februari 2010 20:13
Till: freeswitch-users at lists.freeswitch.org
?mne: Re: [Freeswitch-users] h323 - sip call is not working

On Wed, Feb 10, 2010 at 11:49 AM, Nikolay Kondratyev <kond at nstel.ru<mailto:kond at nstel.ru>> wrote:
Hi all,
I compiled FreeSWITCH Version 1.0.5-20100209-0400 (16587M) with mod_h323.
When I call from h323 (Avaya IPOffice) to local fs extention (x-lite) I hear ring back, but when x-lite picks up, he hears silence, while IPOffice user continues to hear ringback.
The log is at the http://pastebin.freeswitch.org/12091
My configuration is almost default, several local extentions added, and h323.conf from http://wiki.freeswitch.org/wiki/Mod_h323
5840 ? user at IPOffice
2853 ? x-lite registered at FS
IPOffice ip address: 172.23.14.2
FS ip address 172.23.22.49

Can anybody please advise how to solve that?
Is it a configuration or a software problem?


Please can you send me the tcpdump as well (not filtered). IPOffice i known to have a "broken" H323 stack.
did you try to play with tunneling and h245 in setup settings as well ?

It looks like there is some h245 negotiation still pending but cant see that from the logs.

Thanks in advance,
Nikolay.

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


!DSPAM:4b73075e32931909716500!


From tculjaga at gmail.com  Wed Feb 10 12:27:28 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Wed, 10 Feb 2010 21:27:28 +0100
Subject: [Freeswitch-users] h323 - sip call is not working
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C5576CE4923@cooper>
References: <20100210104911.A632A11F49@mail.nstel.ru>
	<65d96fc81002101113p44fe6f43oba72c3a2e13785cc@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C5576CE4923@cooper>
Message-ID: <65d96fc81002101227n3febe3bdgb0c36d767fa5be8e@mail.gmail.com>

On Wed, Feb 10, 2010 at 8:42 PM, Peter Olsson <
peter.olsson at visionutveckling.se> wrote:

> I've been running both h323 and SIP between FS and Avaya IPO for some time.
> No problems at all. :)
>
> But make sure to enable h323 fast start and disable "direct media path" in
> the IPO, if I remember correctly these where the only two parameters that
> made any real difference for me.
>
> But I do recommenf to use SIP, since it's much better supported by FS.
>
> /Peter
>
>
Peter, what H323plus version are you using ?

did you noticed q931 release cause is not mapped H323 => SIP correctly ?
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100210/191ba117/attachment.html 

From matt at webcontracts.co.uk  Wed Feb 10 15:03:48 2010
From: matt at webcontracts.co.uk (Matthew Law)
Date: Wed, 10 Feb 2010 23:03:48 -0000
Subject: [Freeswitch-users] ACL question and js error
In-Reply-To: <20100210024638.GN31942@base.carmickle.com>
References: <ea05d45dca9be2d979fb386f960bc077.squirrel@www.webcontracts.co.uk>
	<20100207145907.GF31942@base.carmickle.com>
	<cfe871e3305675257b8c74ed4fb5a11b.squirrel@www.webcontracts.co.uk>
	<20100210024638.GN31942@base.carmickle.com>
Message-ID: <2112b95ba7e53b541a9f5aad82b77f96.squirrel@www.webcontracts.co.uk>


On Wed, February 10, 2010 2:46 am, Frank Carmickle wrote:
> After doing a
> sofia profile $profile rescan reloadxml
>
> it still doesn't work?  Are you sure it isn't hitting the dialplan and
> failing?  I have never used
>
> <param name="extension" value="voiptalk-incoming"/>
>
> I usually leave it commented and then match on the  destination in the
> dialplan.
>
> <condition field="destination_number"
> expression="^(4124134655|4128484655|19734226137)$">
>
> HTH
> --FC

It was working (there was another issue but that is now fixed too).  I was
not reloading the sofia config properly.  You live and learn.

Thanks,

Matt.



From ederwander at gmail.com  Wed Feb 10 15:43:04 2010
From: ederwander at gmail.com (Eder Souza)
Date: Wed, 10 Feb 2010 21:43:04 -0200
Subject: [Freeswitch-users] Pause during dialing ?
Message-ID: <622bedea1002101543h1c58dd05nb1f81c8d751a600f@mail.gmail.com>

Hi list!!

How i can make calls with Pause in string outgoing ??


here examples in asterisk

exten => _1201.,1,Dial(SIP/eder at eder,60,TtrD(ww891w${EXTEN:4}))

OR

exten => _8X.,1,Dial(Zap/g1/ww0w${EXTEN:1})


In Asterisk exist the flag "w", but im FS how make this ??


thx

Eng Eder de Souza
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100210/973caa7b/attachment.html 

From mayamatakeshi at gmail.com  Wed Feb 10 15:45:09 2010
From: mayamatakeshi at gmail.com (mayamatakeshi)
Date: Thu, 11 Feb 2010 08:45:09 +0900
Subject: [Freeswitch-users] How to kill multiple UUIDs.
In-Reply-To: <191c3a031002101129h1103c2e7l6100b2d41807fddf@mail.gmail.com>
References: <7d79b3931002100307h4d370a22l6f04cafa5bf5806d@mail.gmail.com>
	<A35CD010-E86F-4B7E-A28B-2A9000B8E009@jerris.com>
	<191c3a031002100759v2fef0f48na5afccf46bb08311@mail.gmail.com>
	<59ad9ca11002100916l6982c88r22e1b45af55b80a1@mail.gmail.com>
	<191c3a031002101129h1103c2e7l6100b2d41807fddf@mail.gmail.com>
Message-ID: <15b9404e1002101545g421280adl93fb7ec036d05e9b@mail.gmail.com>

On Thu, Feb 11, 2010 at 4:29 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> if you put a var in the leading {} it will get set on all channels
>
> {foo=bar}sofia/internal/test at server.com


I can see that when I use command bridge, CHANNEL_ORIGINATE shows up with
variable Other-Leg-Unique-ID set to the UID of the channel executing bridge.
To be consistent, could I use this same name when calling originate to
create LegB for a parked LegA? Would it clash with anything?


>
>
> On Wed, Feb 10, 2010 at 11:16 AM, Henry Huang <red.rain.seven at gmail.com>wrote:
>
>> is there an example of setting common var for multiple channels?
>> or just do it normally like set var=foo before all the bridged call I
>> would like to hang up at the same time?
>> and the var name can be anything I want?
>>
>>
>> Henry
>>
>>
>> On Wed, Feb 10, 2010 at 11:59 PM, Anthony Minessale <
>> anthony.minessale at gmail.com> wrote:
>>
>>> or you can set a common var like foo=bar on all the chans and do
>>>
>>> hupall normal_clearing foo bar
>>>
>>>
>>>
>>> On Wed, Feb 10, 2010 at 9:16 AM, Michael Jerris <mike at jerris.com> wrote:
>>>
>>>> you can api hangup hook to call
>>>>
>>>> lua multi_kill.lua uuid1 uuid2 uuid?.
>>>>
>>>> and then write the trivial lua script for that.
>>>>
>>>> Mike
>>>>
>>>> On Feb 10, 2010, at 6:07 AM, lakshmanan ganapathy wrote:
>>>>
>>>> > Hi all,
>>>> >
>>>> > My situation is
>>>> >     A called to 1005 -- Which executes an ESL program.
>>>> >     Now from the program I will made the parallel call using "api
>>>> originate [origination_uuid=uuid1]user/1006,[origination_uuid2]user/1001
>>>> &park()".
>>>> >     UUID's are obtained from create_uuid.
>>>> >
>>>> >     I'll then wait for the api to return, to check whether the call is
>>>> answered or rejected by the other end.
>>>> >     But while I'm waiting, if A hangup the call, I just want to kill
>>>> the calls that are originated by my program.
>>>> >     So I taught of using api_hang_up_hook and I set that variable to
>>>> uuid_kill uuid1 uuid2.
>>>> >     But it only killed the uuid1.
>>>> >
>>>> > Is there any other ways to kill multiple uuid's??
>>>> > please help?
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>> iax:guest at conference.freeswitch.org/888
>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> pstn:+19193869900
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/31ebe7f6/attachment.html 

From mayamatakeshi at gmail.com  Wed Feb 10 15:46:56 2010
From: mayamatakeshi at gmail.com (mayamatakeshi)
Date: Thu, 11 Feb 2010 08:46:56 +0900
Subject: [Freeswitch-users] How to kill multiple UUIDs.
In-Reply-To: <15b9404e1002101545g421280adl93fb7ec036d05e9b@mail.gmail.com>
References: <7d79b3931002100307h4d370a22l6f04cafa5bf5806d@mail.gmail.com>
	<A35CD010-E86F-4B7E-A28B-2A9000B8E009@jerris.com>
	<191c3a031002100759v2fef0f48na5afccf46bb08311@mail.gmail.com>
	<59ad9ca11002100916l6982c88r22e1b45af55b80a1@mail.gmail.com>
	<191c3a031002101129h1103c2e7l6100b2d41807fddf@mail.gmail.com>
	<15b9404e1002101545g421280adl93fb7ec036d05e9b@mail.gmail.com>
Message-ID: <15b9404e1002101546r66ffe4b6ye5842ac5a6590f3a@mail.gmail.com>

On Thu, Feb 11, 2010 at 8:45 AM, mayamatakeshi <mayamatakeshi at gmail.com>wrote:

>
>
> On Thu, Feb 11, 2010 at 4:29 AM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> if you put a var in the leading {} it will get set on all channels
>>
>> {foo=bar}sofia/internal/test at server.com
>
>
> I can see that when I use command bridge, CHANNEL_ORIGINATE shows up with
> variable Other-Leg-Unique-ID set to the UID of the channel executing bridge.
> To be consistent, could I use this same name when calling originate to
> create LegB for a parked LegA? Would it clash with anything?
>

Ah, nevermind. As soon as I hit the send button I realized the variable will
be named:
variable_Other-Leg-Unique-ID.


>
>
>>
>>
>> On Wed, Feb 10, 2010 at 11:16 AM, Henry Huang <red.rain.seven at gmail.com>wrote:
>>
>>> is there an example of setting common var for multiple channels?
>>> or just do it normally like set var=foo before all the bridged call I
>>> would like to hang up at the same time?
>>> and the var name can be anything I want?
>>>
>>>
>>> Henry
>>>
>>>
>>> On Wed, Feb 10, 2010 at 11:59 PM, Anthony Minessale <
>>> anthony.minessale at gmail.com> wrote:
>>>
>>>> or you can set a common var like foo=bar on all the chans and do
>>>>
>>>> hupall normal_clearing foo bar
>>>>
>>>>
>>>>
>>>> On Wed, Feb 10, 2010 at 9:16 AM, Michael Jerris <mike at jerris.com>wrote:
>>>>
>>>>> you can api hangup hook to call
>>>>>
>>>>> lua multi_kill.lua uuid1 uuid2 uuid?.
>>>>>
>>>>> and then write the trivial lua script for that.
>>>>>
>>>>> Mike
>>>>>
>>>>> On Feb 10, 2010, at 6:07 AM, lakshmanan ganapathy wrote:
>>>>>
>>>>> > Hi all,
>>>>> >
>>>>> > My situation is
>>>>> >     A called to 1005 -- Which executes an ESL program.
>>>>> >     Now from the program I will made the parallel call using "api
>>>>> originate [origination_uuid=uuid1]user/1006,[origination_uuid2]user/1001
>>>>> &park()".
>>>>> >     UUID's are obtained from create_uuid.
>>>>> >
>>>>> >     I'll then wait for the api to return, to check whether the call
>>>>> is answered or rejected by the other end.
>>>>> >     But while I'm waiting, if A hangup the call, I just want to kill
>>>>> the calls that are originated by my program.
>>>>> >     So I taught of using api_hang_up_hook and I set that variable to
>>>>> uuid_kill uuid1 uuid2.
>>>>> >     But it only killed the uuid1.
>>>>> >
>>>>> > Is there any other ways to kill multiple uuid's??
>>>>> > please help?
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> Anthony Minessale II
>>>>
>>>> FreeSWITCH http://www.freeswitch.org/
>>>> ClueCon http://www.cluecon.com/
>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>
>>>> AIM: anthm
>>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>> IRC: irc.freenode.net #freeswitch
>>>>
>>>> FreeSWITCH Developer Conference
>>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>>> iax:guest at conference.freeswitch.org/888
>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>> pstn:+19193869900
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/7fa156b0/attachment-0001.html 

From mike at jerris.com  Wed Feb 10 17:13:18 2010
From: mike at jerris.com (Michael Jerris)
Date: Wed, 10 Feb 2010 20:13:18 -0500
Subject: [Freeswitch-users] Any Known Dual-Registration Issue?
In-Reply-To: <3017421995484996A379AADB394D48CB@greyhawk.tonecommander.com>
References: <5AD33F8B1F0848CF9C3A314A60039366@greyhawk.tonecommander.com>
	<035001caa9c8$2dca81c0$895f8540$@com>
	<3017421995484996A379AADB394D48CB@greyhawk.tonecommander.com>
Message-ID: <6F52CA14-07F0-41D3-B137-8623A9F09675@jerris.com>

This is correct, so registrations have a timeout.  If your timeout is long, and you allow multiple registrations, it will keep trying to call the bad registration until timeout.  Our middle-ground setting is to have a setting that specifies only one reg per ip/port.  Anything beyond that is just not possible with the protocol, at least when using udp.  If you are using tcp for the registrations it *might* clear that down properly in these cases.  So you have 3 ways to address this.  Only allow 1 registration per user, only allow 1 registration per contact, allow all the registrations and expect this issue until timeout (set at the length you can tollerate), and a possible 4th of tcp may address this.

Mike


On Feb 10, 2010, at 11:13 AM, Jerry Richards wrote:

> I am using Bria Pro softphones, but it is possible that the PC was
> disconnected before a restart.  Any phone does not necessarily send
> unregister, for example if there is a power outage or it is disconnected
> from the network.  Anyway, I'll just try setting multiple-registrations to
> contact as you suggest.
> 
> Thanks,
> Jerry
> 



From mcampbellsmith at gmail.com  Wed Feb 10 18:24:23 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Thu, 11 Feb 2010 13:24:23 +1100
Subject: [Freeswitch-users] Forbidden using UDP, works with TCP/TLS
Message-ID: <33c87fa31002101824h21e3bf7lb13b68e8d4fd36a2@mail.gmail.com>

Hi!

I had a user registered using TLS transport. That was working fine but
I want to change the ATA over to use UDP instead.

All I thought I should have to do was to change the transport and
ports used to register in the ATA (SPA3102).  However, when I do this,
FS responds with Forbidden.

When I change the settings back to use TCP or TLS, registration is successful.

What would cause FS to respond with forbidden?  I do not change the
username/password fields in either case.

Thanks


From jingwei.yang at gmail.com  Wed Feb 10 19:44:14 2010
From: jingwei.yang at gmail.com (Jingwei Yang)
Date: Thu, 11 Feb 2010 11:44:14 +0800
Subject: [Freeswitch-users] How to record the call upon successful bridge
In-Reply-To: <87f2f3b91002100709m32d975b5rf16823e2b02bd8c2@mail.gmail.com>
References: <13529f9d1002091819y2b363f50vc314d1b6197503b2@mail.gmail.com>
	<257401E3-B32F-4699-96F6-415BA2ADDE88@freeswitch.org>
	<13529f9d1002100039q43775563r49b11136ddda0a0@mail.gmail.com>
	<87f2f3b91002100709m32d975b5rf16823e2b02bd8c2@mail.gmail.com>
Message-ID: <13529f9d1002101944s5a7b9454ya11209763e2cc62@mail.gmail.com>

Hi Michael,

Thanks for the reply. I've upgraded FS to the latest revision 16600 and here
are the log details.

Using execute_on_answer: http://pastebin.freeswitch.org/12109
Using bridge_pre_execute_bleg_app and bridge_pre_execute_bleg_data:
http://pastebin.freeswitch.org/12110

Thanks and best regards,
-Jingwei

On Wed, Feb 10, 2010 at 11:09 PM, Michael Collins <msc at freeswitch.org>wrote:

> Make sure that you are on the latest SVN trunk and retest. Pastebin the
> complete debug log from start to finish of the call.
> -MC
>
>
> On Wed, Feb 10, 2010 at 12:39 AM, Jingwei Yang <jingwei.yang at gmail.com>wrote:
>
>> Hi Ray,
>>
>> Thanks a lot for the replies. Allow me to elaborate a little bit about my
>> situation.
>>
>> 1. client A calls in and parks at Fifo myq.
>> 2. FS connets Agent B to an extension (via originate
>> skypiax/ANY/jingwei.yang 33333)
>> 3. uuid_bridge client A and agent B
>>
>> I believe the spice I can add is in the extension 33333. Here's how I
>> define it.
>>
>>     <extension name="agent_hold">
>>       <condition field="destination_number" expression="^33333$">
>>         <action application="set" data="*execute_on_answer*=record_session
>> /tmp/33333.wav"/>
>>         <action application="answer"/>
>>         <action application="sleep" data="2000"/>
>>         <action application="info"/>
>>         <action application="playback"
>> data="ivr/ivr-hold_connect_call.wav"/>
>>       </condition>
>>     </extension>
>>
>> But the wav file didn't get generated at all. Please advise whether the
>> above is in correct usage.
>>
>> I've also tried bridge_pre_execute_bleg_app and
>> bridge_pre_execute_bleg_data
>>
>>         <action application="set"
>> data="bridge_pre_execute_bleg_app=record_session"/>
>>         <action application="set"
>> data="bridge_pre_execute_bleg_data=/tmp/33333.wav"/>
>>
>> The audio file didn't appear either.
>>
>> The only successful method is by using record_session directly like this:
>>
>>         <action application="record_session" data="/tmp/33333.wav"/>
>>
>> However, in this form, the wav file includes the waiting music, which is
>> not ideal.
>>
>> Thanks and best regards,
>>  -Jingwei
>>
>>
>> On Wed, Feb 10, 2010 at 12:01 PM, Raymond Chandler <
>> intralanman at freeswitch.org> wrote:
>>
>>> errr... .execute_on_answer might be better ;-)
>>> -Ray
>>>
>>> On Feb 9, 2010, at 9:19 PM, Jingwei Yang wrote:
>>>
>>> > Hi,
>>> >
>>> > I'm using uuid_bridge to bridge two calls. May I know how to start
>>> recording only when the bridge succeeds?
>>> >
>>> > Thanks,
>>> > -Jingwei
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/7f276b7b/attachment.html 

From msc at freeswitch.org  Wed Feb 10 20:57:49 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 10 Feb 2010 20:57:49 -0800
Subject: [Freeswitch-users] demo_ivr cannot find sound files via
	relative paths
In-Reply-To: <78030B9E01DD48F8B024A97035F96282@greyhawk.tonecommander.com>
References: <78030B9E01DD48F8B024A97035F96282@greyhawk.tonecommander.com>
Message-ID: <87f2f3b91002102057t7b039221kf3975553414880cf@mail.gmail.com>

If I read this log correctly it failed to find the "invalid entry" file but
it did find the phrases just fine. Can you confirm the presence of this
file:
/usr/local/freeswitch/sounds/en/us/callie/ivr/8000/ivr-that_was_an_invalid_entry.wav

(It looks like this call is at 8kHz so that's where I'm assuming FS is
looking to find the sound file...)

-MC

On Wed, Feb 10, 2010 at 10:50 AM, Robert Hadley
<robert.hadley at teotech.com>wrote:

>  Hi,
>
>
>
> It appears a recent change (possibly the new sounds_dir variable or the new
> ivr_menu folder?) may have broken relative sound file paths in the IVR.  I
> built a today?s trunk version and installed to the default location. Using
> the default conf files the demo_ivr cannot find files based on the relative
> paths specified in ivr_menus/demo_ivr.xml.
>
>
>
> [root at TEO-UCM-T2 conf]# cat ivr_menus/demo_ivr.xml
>
> <include>
>
>   <!-- demo IVR setup -->
>
>   <!-- demo IVR, Main Menu -->
>
>   <menu name="demo_ivr"
>
>       greet-long="phrase:demo_ivr_main_menu"
>
>       greet-short="phrase:demo_ivr_main_menu_short"
>
>       invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"
>
>       exit-sound="voicemail/vm-goodbye.wav"
>
>
>
>
>
> 2010-02-10 10:32:15.300337 [DEBUG] switch_ivr_menu.c:414 Executing IVR menu
> demo_ivr
>
> 2010-02-10 10:32:15.300337 [DEBUG] switch_ivr_play_say.c:118 No language
> specified - Using [en]
>
> 2010-02-10 10:32:15.300337 [DEBUG] switch_ivr_play_say.c:273 Handle
> play-file:[ivr/ivr-welcome_to_freeswitch.wav] (en:en)
>
> 2010-02-10 10:32:15.300337 [DEBUG] switch_ivr_play_say.c:1158 Codec
> Activated L16 at 8000hz 1 channels 20ms
>
> 2010-02-10 10:32:17.922147 [DEBUG] switch_ivr_play_say.c:1450 done playing
> file
>
> 2010-02-10 10:32:18.042149 [DEBUG] switch_ivr_play_say.c:273 Handle
> play-file:[ivr/ivr-this_ivr_will_let_you_test_features.wav] (en:en)
>
> 2010-02-10 10:32:18.042149 [DEBUG] switch_ivr_play_say.c:1158 Codec
> Activated L16 at 8000hz 1 channels 20ms
>
> 2010-02-10 10:32:19.962158 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF 2:800
>
> 2010-02-10 10:32:19.962158 [DEBUG] switch_ivr_play_say.c:1450 done playing
> file
>
> 2010-02-10 10:32:20.082156 [DEBUG] switch_ivr_menu.c:329 waiting for 3/4
> digits t/o 2000
>
> 2010-02-10 10:32:20.120617 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF 2:400
>
> 2010-02-10 10:32:20.442158 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF 2:800
>
> 2010-02-10 10:32:20.682162 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF 2:800
>
> 2010-02-10 10:32:20.682162 [DEBUG] switch_ivr_menu.c:376 digits '2222'
>
> 2010-02-10 10:32:20.682162 [DEBUG] switch_ivr_menu.c:470 action regex
> [2222] [/^(10[01][0-9])$/] [0]
>
> 2010-02-10 10:32:20.682162 [DEBUG] switch_ivr_menu.c:560 IVR menu
> 'demo_ivr' caught invalid input '2222'
>
> 2010-02-10 10:32:20.682162 [ERR] mod_sndfile.c:194 Error Opening File
> [/usr/local/freeswitch/sounds/ivr/ivr-that_was_an_invalid_entry.wav] [System
> error : No such file or directory.]
>
> 2010-02-10 10:32:21.701505 [DEBUG] switch_ivr_play_say.c:118 No language
> specified - Using [en]
>
> 2010-02-10 10:32:21.701505 [DEBUG] switch_ivr_play_say.c:273 Handle
> play-file:[silence_stream://1000] (en:en)
>
> 2010-02-10 10:32:21.701505 [DEBUG] switch_ivr_play_say.c:1158 Codec
> Activated L16 at 8000hz 1 channels 20ms
>
> 2010-02-10 10:32:22.700352 [DEBUG] switch_ivr_play_say.c:1450 done playing
> file
>
>
>
> Regards,
>
> Robert
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100210/61638b2e/attachment-0001.html 

From msc at freeswitch.org  Wed Feb 10 21:18:42 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 10 Feb 2010 21:18:42 -0800
Subject: [Freeswitch-users] Interesting article on OSS telephony
Message-ID: <87f2f3b91002102118x27df10c4qc7fc8d1fad2b7b92@mail.gmail.com>

Check out the article on TMCnet.com, linked-to here:
http://www.freeswitch.org/node/238

-Michael
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100210/e3c9608b/attachment.html 

From mike at jerris.com  Wed Feb 10 21:21:06 2010
From: mike at jerris.com (Michael Jerris)
Date: Thu, 11 Feb 2010 00:21:06 -0500
Subject: [Freeswitch-users] Lua script hangup detect
In-Reply-To: <4B72F2A8.4070503@gmail.com>
References: <314dc3f81002100117h6b1bc49cna989624a1ae4e0e2@mail.gmail.com>	<4B729155.7010708@gmail.com>	<314dc3f81002100800o41fd6561w394d5e83e69d3d80@mail.gmail.com>	<4B72DFF9.4040402@gmail.com>
	<1b46b4e81002100932x7af486ekd4b05f4750cace20@mail.gmail.com>
	<4B72F2A8.4070503@gmail.com>
Message-ID: <4CA5432B-5F6E-438E-B6EF-3277847E7922@jerris.com>

lua runs until you finish your script.  You need to exit when you are supposed to.  We have session::ready() for you to test this.

Mike

On Feb 10, 2010, at 12:53 PM, Nazim Agabekov wrote:

> I think it continues to run until it finishes. I'll check it tomorrow on my test system.
> 
> On 02/10/2010 09:32 PM, Nicolas Brenner wrote:
>> 
>> 
>> On Wed, Feb 10, 2010 at 1:34 PM, Nazim Agabekov <nazim.agabekov at gmail.com> wrote:
>> Lua script is not terminating immediately on hangup.
>> 
>> 
>> When does it terminate then? Will the script terminate when it finishes running or does it need some special instruction?
>> 
>>  
>> This behavior allows user to finalize the script nicely (free dynamically allocated resources, update logs, e.t.c)
>> 
>>> maybe it will stuck in a few situation before it reach to session:status check point.
>>> 
>>> like GetDigit or something else? (sorry, I dont have FS right now, need test it tomorrow)
>> I've never encountered such a problem. Usually GetDigit-like functions have timeout parameter, so they don't block forever.
>> Just check the session status often and it will work like a charm ;
>> 
>> 
>> 
>> On 02/10/2010 08:00 PM, Chia-Yen Wu wrote:
>>> 
>>> thx for reply, but shouldnt Lua script terminate when client hangs up?
>>> 
>>> maybe it will stuck in a few situation before it reach to session:status check point.
>>> 
>>> like GetDigit or something else? (sorry, I dont have FS right now, need test it tomorrow)
>>> 
>>> 2010/2/10 Nazim Agabekov <nazim.agabekov at gmail.com>
>>> Hello,
>>> Can you pastebin your script?
>>> 
>>> http://pastebin.freeswitch.org
>>> 
>>> 
>>> On 02/10/2010 01:17 PM, Chia-Yen Wu wrote:
>>>> Hello, 
>>>> i tried to use Lua script to replace xml macro in dialplan,
>>>> but I found out that Lua wont terminate if client hangup,
>>>> ,so the session is still on but client is already hangup,
>>>> is there a way to avoid this ?
>>>> 
>>>> 
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>   
>>> 
>>> 
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>> 
>>> 
>>> 
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>   
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>   
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/13253628/attachment.html 

From paul at apcl.us  Wed Feb 10 18:12:11 2010
From: paul at apcl.us (Paul Levin)
Date: Wed, 10 Feb 2010 21:12:11 -0500
Subject: [Freeswitch-users] is there a command to get a list of registered
	accounts?
Message-ID: <4B73677B.6020406@apcl.us>

An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100210/18b7c23c/attachment.html 

From paul at apcl.us  Wed Feb 10 18:15:34 2010
From: paul at apcl.us (Paul Levin)
Date: Wed, 10 Feb 2010 21:15:34 -0500
Subject: [Freeswitch-users] how to set From address for voicemail emails?
Message-ID: <4B736846.1040908@apcl.us>

An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100210/ba1e3c35/attachment.html 

From mike at jerris.com  Wed Feb 10 21:25:08 2010
From: mike at jerris.com (Michael Jerris)
Date: Thu, 11 Feb 2010 00:25:08 -0500
Subject: [Freeswitch-users] Forbidden using UDP, works with TCP/TLS
In-Reply-To: <33c87fa31002101824h21e3bf7lb13b68e8d4fd36a2@mail.gmail.com>
References: <33c87fa31002101824h21e3bf7lb13b68e8d4fd36a2@mail.gmail.com>
Message-ID: <2EF27EFC-1634-46A3-B438-2EE99BEF58F1@jerris.com>

Look at the FreeSWITCH debug logs and compare the differences, they should tell you why.

Mike

On Feb 10, 2010, at 9:24 PM, Mark Campbell-Smith wrote:

> Hi!
> 
> I had a user registered using TLS transport. That was working fine but
> I want to change the ATA over to use UDP instead.
> 
> All I thought I should have to do was to change the transport and
> ports used to register in the ATA (SPA3102).  However, when I do this,
> FS responds with Forbidden.
> 
> When I change the settings back to use TCP or TLS, registration is successful.
> 
> What would cause FS to respond with forbidden?  I do not change the
> username/password fields in either case.



From mcampbellsmith at gmail.com  Wed Feb 10 21:29:31 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Thu, 11 Feb 2010 16:29:31 +1100
Subject: [Freeswitch-users] is there a command to get a list of
	registered accounts?
In-Reply-To: <4B73677B.6020406@apcl.us>
References: <4B73677B.6020406@apcl.us>
Message-ID: <33c87fa31002102129r5d692dc5pfb414b80b47be221@mail.gmail.com>

sofia status profile xxx (where xxx is usually internal or external)

On Thu, Feb 11, 2010 at 1:12 PM, Paul Levin <paul at apcl.us> wrote:
> At the FreeSwitch console is there a command I can enter to get a list of
> all sip accounts that are currently registered?
> ??? Thanks,
> ??? Paul
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From mike at jerris.com  Wed Feb 10 21:33:39 2010
From: mike at jerris.com (Michael Jerris)
Date: Thu, 11 Feb 2010 00:33:39 -0500
Subject: [Freeswitch-users] demo_ivr cannot find sound files via
	relative paths
In-Reply-To: <87f2f3b91002102057t7b039221kf3975553414880cf@mail.gmail.com>
References: <78030B9E01DD48F8B024A97035F96282@greyhawk.tonecommander.com>
	<87f2f3b91002102057t7b039221kf3975553414880cf@mail.gmail.com>
Message-ID: <E4BC77C3-8271-4F30-B6F8-6822E4747BC3@jerris.com>

did you make any changes to the default configs?  what is in your vars.xml related to sounds?  the relative paths were always relative to that dir, did you make any changes to the sounds prefix ?

Mike

On Feb 10, 2010, at 11:57 PM, Michael Collins wrote:

> If I read this log correctly it failed to find the "invalid entry" file but it did find the phrases just fine. Can you confirm the presence of this file:
> /usr/local/freeswitch/sounds/en/us/callie/ivr/8000/ivr-that_was_an_invalid_entry.wav
> 
> (It looks like this call is at 8kHz so that's where I'm assuming FS is looking to find the sound file...)
> 
> -MC
> 
> On Wed, Feb 10, 2010 at 10:50 AM, Robert Hadley <robert.hadley at teotech.com> wrote:
> Hi,
> 
>  
> It appears a recent change (possibly the new sounds_dir variable or the new ivr_menu folder?) may have broken relative sound file paths in the IVR.  I built a today?s trunk version and installed to the default location. Using the default conf files the demo_ivr cannot find files based on the relative paths specified in ivr_menus/demo_ivr.xml.
> 
>  
> [root at TEO-UCM-T2 conf]# cat ivr_menus/demo_ivr.xml
> 
> <include>
> 
>   <!-- demo IVR setup -->
> 
>   <!-- demo IVR, Main Menu -->
> 
>   <menu name="demo_ivr"
> 
>       greet-long="phrase:demo_ivr_main_menu"
> 
>       greet-short="phrase:demo_ivr_main_menu_short"
> 
>       invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"
> 
>       exit-sound="voicemail/vm-goodbye.wav"
> 
>  
>  
> 2010-02-10 10:32:15.300337 [DEBUG] switch_ivr_menu.c:414 Executing IVR menu demo_ivr
> 
> 2010-02-10 10:32:15.300337 [DEBUG] switch_ivr_play_say.c:118 No language specified - Using [en]
> 
> 2010-02-10 10:32:15.300337 [DEBUG] switch_ivr_play_say.c:273 Handle play-file:[ivr/ivr-welcome_to_freeswitch.wav] (en:en)
> 
> 2010-02-10 10:32:15.300337 [DEBUG] switch_ivr_play_say.c:1158 Codec Activated L16 at 8000hz 1 channels 20ms
> 
> 2010-02-10 10:32:17.922147 [DEBUG] switch_ivr_play_say.c:1450 done playing file
> 
> 2010-02-10 10:32:18.042149 [DEBUG] switch_ivr_play_say.c:273 Handle play-file:[ivr/ivr-this_ivr_will_let_you_test_features.wav] (en:en)
> 
> 2010-02-10 10:32:18.042149 [DEBUG] switch_ivr_play_say.c:1158 Codec Activated L16 at 8000hz 1 channels 20ms
> 
> 2010-02-10 10:32:19.962158 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF 2:800
> 
> 2010-02-10 10:32:19.962158 [DEBUG] switch_ivr_play_say.c:1450 done playing file
> 
> 2010-02-10 10:32:20.082156 [DEBUG] switch_ivr_menu.c:329 waiting for 3/4 digits t/o 2000
> 
> 2010-02-10 10:32:20.120617 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF 2:400
> 
> 2010-02-10 10:32:20.442158 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF 2:800
> 
> 2010-02-10 10:32:20.682162 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF 2:800
> 
> 2010-02-10 10:32:20.682162 [DEBUG] switch_ivr_menu.c:376 digits '2222'
> 
> 2010-02-10 10:32:20.682162 [DEBUG] switch_ivr_menu.c:470 action regex [2222] [/^(10[01][0-9])$/] [0]
> 
> 2010-02-10 10:32:20.682162 [DEBUG] switch_ivr_menu.c:560 IVR menu 'demo_ivr' caught invalid input '2222'
> 
> 2010-02-10 10:32:20.682162 [ERR] mod_sndfile.c:194 Error Opening File [/usr/local/freeswitch/sounds/ivr/ivr-that_was_an_invalid_entry.wav] [System error : No such file or directory.]
> 
> 2010-02-10 10:32:21.701505 [DEBUG] switch_ivr_play_say.c:118 No language specified - Using [en]
> 
> 2010-02-10 10:32:21.701505 [DEBUG] switch_ivr_play_say.c:273 Handle play-file:[silence_stream://1000] (en:en)
> 
> 2010-02-10 10:32:21.701505 [DEBUG] switch_ivr_play_say.c:1158 Codec Activated L16 at 8000hz 1 channels 20ms
> 
> 2010-02-10 10:32:22.700352 [DEBUG] switch_ivr_play_say.c:1450 done playing file
> 
>  
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/0a233797/attachment-0001.html 

From mike at jerris.com  Wed Feb 10 21:34:18 2010
From: mike at jerris.com (Michael Jerris)
Date: Thu, 11 Feb 2010 00:34:18 -0500
Subject: [Freeswitch-users] is there a command to get a list of
	registered accounts?
In-Reply-To: <4B73677B.6020406@apcl.us>
References: <4B73677B.6020406@apcl.us>
Message-ID: <1FD15E7C-1942-4C3A-9D64-36CAB61C49E5@jerris.com>

sofia status profile <profile_name>

On Feb 10, 2010, at 9:12 PM, Paul Levin wrote:
> At the FreeSwitch console is there a command I can enter to get a list of all sip accounts that are currently registered?

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/15c013d3/attachment.html 

From mike at jerris.com  Wed Feb 10 21:36:47 2010
From: mike at jerris.com (Michael Jerris)
Date: Thu, 11 Feb 2010 00:36:47 -0500
Subject: [Freeswitch-users] how to set From address for voicemail emails?
In-Reply-To: <4B736846.1040908@apcl.us>
References: <4B736846.1040908@apcl.us>
Message-ID: <7471DDD5-0A73-4F18-AF90-BA8534EBC9E7@jerris.com>

This is the difference between what is sent to the mail server in the mime content, and what is passed as MAIL FROM:  to the smtp server.  The latter is controlled by that param, the former is in the template.

Mike

On Feb 10, 2010, at 9:15 PM, Paul Levin wrote:

> I am running FreeSwitch on Windows.  I have msmtp setup and voice mail emails are being sent. 
> 
> I have msmtp configured to set a "From" address of me at mydomain.com, but when FreeSwitch sends an email with a voice mail message from Alice, the From address of the email is Alice at sipServerDomain.com.  According to the Mod voicemail document (http://wiki.freeswitch.org/wiki/Mod_voicemail) the email_email-from parameter should control this, but I tried setting it in conf\autoload_configs\directory.conf.xml (as per that document) and also in Bob.xml (the account getting the voicemail).  Neither place changed the value being used.  
> 
> How do I get this changed?

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/1b75d3e5/attachment.html 

From mailinglist at fribert.dk  Wed Feb 10 21:45:45 2010
From: mailinglist at fribert.dk (mailinglist)
Date: Thu, 11 Feb 2010 06:45:45 +0100
Subject: [Freeswitch-users] Need help setting up a feature
Message-ID: <4B73A799020000E100000470@mail.fribert.dk>

Sorry for the repost, but the previous thread just died :-)
 
I'm trying to get the possibility of transfering an incoming call from one extension to another, and give the possibility of turning it into a conference.
I don't have a 'transfer' button.
I do have an 'R' button on the Siemens handsets, and a 'Flash' button on the Sipura. The 'Flash' button gives me a new dialtone, gives the caller MOH, and then I can dial the new extension, and transfer the call, but not create a conference.
But the Siemens handset does not have a 'flash', and pressing the R doesn't do anything.
 
It might be two different features 'transfer' and 'conference'...

But I thought that using the bind_meta_app would accomplish both.
 
It's on an incoming call from the outside.
So the situation:
The Public folder has an entry that matches the dialed number, and does a transfer to 8202.
Then the dialplan matches the 8202 with a group, and the phone rings.
Somebody picks it up, finds out that it needs to be transferred to another extension, or transferred to a conference with a second extension.
How do I construct that?
 
 
 
Best regards
Fribse
 
/Running Freeswitch on pfSense running on a VMWare ESXi 4.0
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/89c5612d/attachment.html 

From ustcorporation at yahoo.com  Wed Feb 10 23:30:19 2010
From: ustcorporation at yahoo.com (teldev)
Date: Wed, 10 Feb 2010 23:30:19 -0800 (PST)
Subject: [Freeswitch-users] Seeking Advice on SIP Phones like Aastra
In-Reply-To: <9DFA265F-8632-466B-8910-1E2F9D956908@jerris.com>
References: <9DFA265F-8632-466B-8910-1E2F9D956908@jerris.com>
Message-ID: <1265873419114-4553224.post@n2.nabble.com>


Hello Mike,

"Can you post for all to see some idea of how these applications work,
lanagages used, some samples so we can see if we can get some interest in
this?"

"I am not sure the phone gui has much to do with FreeSWITCH at all, other
than pulling a little data from the databases (again, ODBC highly
recommended)" 

-->  The idea would be to create a phone GUI on Aastra 6739i that would
enable the touchscreen to do something like what is shown in these demos:

http://www.youtube.com/watch?v=xZDHibW1gUs
http://www.youtube.com/watch?v=Y58TPNCOTZI

Perform FreeSWITCH actions: answer calls, place calls, setup conferences/add
participants, etc.
Display FreeSWITCH states: voice mail waiting count, extensions that are
online/offline, call log, etc. 
Introduce new actions: dial By name directory with pictures, visual setup of
conferences, etc.  

Since the Aastra 6739i, SNOM 870, and some others have built-in XML browsers
it seems like they could be programmed to interact well with FreeSWITCH and
various third party systems such as online feeds like weather reports, news,
stock prices.  Aastra is taking the IPhone approach of letting developers
create the apps.  They even have a similar slogan "There's an App for That"
versus Apple's "There's an app for just about anything". 

http://www.ucstrategies.com/unified-communications-expert-views/aastra-phone-theres-an-app-for-that.aspx

I don't know to what extent FreeSWITCH interacts with this phone out of the
box, it reportedly works well with Asterisk though.  I'm sure FS will likely
understand certain button presses like hold, mute, speaker, transfer, etc. 
I will be ordering one within a few days. 

A few quesions for the FS community:

Does this project have merit and fulfill a need?  (I have yet to use FS as a
PBX)

Can anyone shed some light on what features on the phone will work and what
I've mentioned will require custom development?

Any ideas on how to architect an integration that would leverage what FS
already does?  Currently our IVR work uses Javascript/SpiderMonkey to call
our own web service for database reads/writes.  For this project, we'd need
to interact more directly with FS and learn more about it's internals.  

teldev

-- 
View this message in context: http://n2.nabble.com/Re-Seeking-Advice-on-SIP-Phones-like-Aastra-tp4543930p4553224.html
Sent from the freeswitch-users mailing list archive at Nabble.com.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100210/cb7e6d7b/attachment.html 

From nazim.agabekov at gmail.com  Wed Feb 10 23:45:02 2010
From: nazim.agabekov at gmail.com (Nazim Agabekov)
Date: Thu, 11 Feb 2010 11:45:02 +0400
Subject: [Freeswitch-users] Lua script hangup detect
In-Reply-To: <4CA5432B-5F6E-438E-B6EF-3277847E7922@jerris.com>
References: <314dc3f81002100117h6b1bc49cna989624a1ae4e0e2@mail.gmail.com>	<4B729155.7010708@gmail.com>	<314dc3f81002100800o41fd6561w394d5e83e69d3d80@mail.gmail.com>	<4B72DFF9.4040402@gmail.com>	<1b46b4e81002100932x7af486ekd4b05f4750cace20@mail.gmail.com>	<4B72F2A8.4070503@gmail.com>
	<4CA5432B-5F6E-438E-B6EF-3277847E7922@jerris.com>
Message-ID: <4B73B57E.9040901@gmail.com>

Thanks Mike! Mod_Lua is great. I had experience with a lot of
proprietary IVR systems in the past.
FreeSWITCH and Mod_lua + luasql beats them all. Functionality is really
impressive.

On 02/11/2010 09:21 AM, Michael Jerris wrote:
> lua runs until you finish your script.  You need to exit when you are
> supposed to.  We have session::ready() for you to test this.
>
> Mike
>
> On Feb 10, 2010, at 12:53 PM, Nazim Agabekov wrote:
>
>> I think it continues to run until it finishes. I'll check it tomorrow
>> on my test system.
>>
>> On 02/10/2010 09:32 PM, Nicolas Brenner wrote:
>>>
>>> On Wed, Feb 10, 2010 at 1:34 PM, Nazim Agabekov
>>> <nazim.agabekov at gmail.com <mailto:nazim.agabekov at gmail.com>> wrote:
>>>
>>>     Lua script is not terminating immediately on hangup.
>>>
>>>
>>>
>>> When does it terminate then? Will the script terminate when it
>>> finishes running or does it need some special instruction?
>>>
>>>  
>>>
>>>     This behavior allows user to finalize the script nicely (free
>>>     dynamically allocated resources, update logs, e.t.c)
>>>
>>>>     maybe it will stuck in a few situation before it reach to
>>>>     session:status check point.
>>>>
>>>>     like GetDigit or something else? (sorry, I dont have FS right
>>>>     now, need test it tomorrow)
>>>     I've never encountered such a problem. Usually GetDigit-like
>>>     functions have timeout parameter, so they don't block forever.
>>>     Just check the session status often and it will work like a charm ;
>>>
>>>
>>>
>>>     On 02/10/2010 08:00 PM, Chia-Yen Wu wrote:
>>>>     thx for reply, but shouldnt Lua script terminate when client
>>>>     hangs up?
>>>>
>>>>     maybe it will stuck in a few situation before it reach to
>>>>     session:status check point.
>>>>
>>>>     like GetDigit or something else? (sorry, I dont have FS right
>>>>     now, need test it tomorrow)
>>>>
>>>>     2010/2/10 Nazim Agabekov <nazim.agabekov at gmail.com
>>>>     <mailto:nazim.agabekov at gmail.com>>
>>>>
>>>>         Hello,
>>>>         Can you pastebin your script?
>>>>
>>>>         http://pastebin.freeswitch.org
>>>>         <http://pastebin.freeswitch.org/>
>>>>
>>>>
>>>>         On 02/10/2010 01:17 PM, Chia-Yen Wu wrote:
>>>>>         Hello, 
>>>>>         i tried to use Lua script to replace xml macro in dialplan,
>>>>>         but I found out that Lua wont terminate if client hangup,
>>>>>         ,so the session is still on but client is already hangup,
>>>>>         is there a way to avoid this ?
>>>>>
>>>>>
>>>>>         _______________________________________________
>>>>>         FreeSWITCH-users mailing list
>>>>>         FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>         http://www.freeswitch.org <http://www.freeswitch.org/>
>>>>>           
>>>>
>>>>
>>>>         _______________________________________________
>>>>         FreeSWITCH-users mailing list
>>>>         FreeSWITCH-users at lists.freeswitch.org
>>>>         <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>         http://www.freeswitch.org <http://www.freeswitch.org/>
>>>>
>>>>
>>>>
>>>>     _______________________________________________
>>>>     FreeSWITCH-users mailing list
>>>>     FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>     http://www.freeswitch.org <http://www.freeswitch.org/>
>>>>       
>>>
>>>
>>>     _______________________________________________
>>>     FreeSWITCH-users mailing list
>>>     FreeSWITCH-users at lists.freeswitch.org
>>>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>     http://www.freeswitch.org <http://www.freeswitch.org/>
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>   
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/32984796/attachment-0001.html 

From peter.olsson at visionutveckling.se  Wed Feb 10 23:50:31 2010
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Thu, 11 Feb 2010 08:50:31 +0100
Subject: [Freeswitch-users] h323 - sip call is not working
In-Reply-To: <65d96fc81002101227n3febe3bdgb0c36d767fa5be8e@mail.gmail.com>
References: <20100210104911.A632A11F49@mail.nstel.ru>
	<65d96fc81002101113p44fe6f43oba72c3a2e13785cc@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C5576CE4923@cooper>
	<65d96fc81002101227n3febe3bdgb0c36d767fa5be8e@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C5577127F67@cooper>

I'm using the opal module, since mod_h323 is not available in Windows yet. We're not so dependant of all the correct release codes etc, we use it mostly for ASR and voicemail applications, but I do know there have been discussions about this on the mailing list. My recommendation is to use SIP to the IPO instead (mostly beacuse SIP is much better supported by FS), we're using that more and more.

/Peter

Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Tihomir Culjaga
Skickat: den 10 februari 2010 21:27
Till: freeswitch-users at lists.freeswitch.org
?mne: Re: [Freeswitch-users] h323 - sip call is not working


On Wed, Feb 10, 2010 at 8:42 PM, Peter Olsson <peter.olsson at visionutveckling.se<mailto:peter.olsson at visionutveckling.se>> wrote:
I've been running both h323 and SIP between FS and Avaya IPO for some time. No problems at all. :)

But make sure to enable h323 fast start and disable "direct media path" in the IPO, if I remember correctly these where the only two parameters that made any real difference for me.

But I do recommenf to use SIP, since it's much better supported by FS.

/Peter

Peter, what H323plus version are you using ?

did you noticed q931 release cause is not mapped H323 => SIP correctly ?

!DSPAM:4b73196232931389031035!
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/905e4fcb/attachment.html 

From ustcorporation at yahoo.com  Thu Feb 11 00:01:47 2010
From: ustcorporation at yahoo.com (teldev)
Date: Thu, 11 Feb 2010 00:01:47 -0800 (PST)
Subject: [Freeswitch-users] Seeking Advice on SIP Phones like Aastra
In-Reply-To: <9DFA265F-8632-466B-8910-1E2F9D956908@jerris.com>
References: <9DFA265F-8632-466B-8910-1E2F9D956908@jerris.com>
Message-ID: <1265875307968-4553322.post@n2.nabble.com>


Hello Mike,

Last post lost all formatting when I used Nabble, this is readable now.

"Can you post for all to see some idea of how these applications work,
lanagages used, some samples so we can see if we can get some interest in
this?"

"I am not sure the phone gui has much to do with FreeSWITCH at all, other
than pulling a little data from the databases (again, ODBC highly
recommended)" 

-->  The idea would be to create a phone GUI on Aastra 6739i that would
enable the touchscreen to do something like what is shown in these demos:

http://www.youtube.com/watch?v=xZDHibW1gUs
http://www.youtube.com/watch?v=Y58TPNCOTZI

Perform FreeSWITCH actions: answer calls, place calls, setup conferences/add
participants, etc.
Display FreeSWITCH states: voice mail waiting count, extensions that are
online/offline, call log, etc. 
Introduce new actions: dial By name directory with pictures, visual setup of
conferences, etc.  

Since the Aastra 6739i, SNOM 870, and some others have built-in XML browsers
it seems like they could be programmed to interact well with FreeSWITCH and
various third party systems such as online feeds like weather reports, news,
stock prices.  Aastra is taking the IPhone approach of letting developers
create the apps.  They even have a similar slogan "There's an App for That"
versus Apple's "There's an app for just about anything". 

http://www.ucstrategies.com/unified-communications-expert-views/aastra-phone-theres-an-app-for-that.aspx

I don't know to what extent FreeSWITCH interacts with this phone out of the
box, it reportedly works well with Asterisk though.  I'm sure FS will likely
understand certain button presses like hold, mute, speaker, transfer, etc. 
I will be ordering one within a few days. 

A few quesions for the FS community:

Does this project have merit and fulfill a need?  (I have yet to use FS as a
PBX)

Can anyone shed some light on what features on the phone will work and what
I've mentioned will require custom development?

Any ideas on how to architect an integration that would leverage what FS
already does?  Currently our IVR work uses Javascript/SpiderMonkey to call
our own web service for database reads/writes.  For this project, we'd need
to interact more directly with FS and learn more about it's internals.  
-- 
View this message in context: http://n2.nabble.com/Re-Seeking-Advice-on-SIP-Phones-like-Aastra-tp4543930p4553322.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From kond at nstel.ru  Thu Feb 11 00:04:31 2010
From: kond at nstel.ru (Nikolay Kondratyev)
Date: Thu, 11 Feb 2010 11:04:31 +0300
Subject: [Freeswitch-users] h323 - sip call is not working
In-Reply-To: <65d96fc81002101113p44fe6f43oba72c3a2e13785cc@mail.gmail.com>
Message-ID: <20100211080431.DD68E11F34@mail.nstel.ru>

Tihomir,

I've just sent the tcpdump to your address (because I think mail list will
not accept 300K attachement).

Thanks and regards,

Nikolay.

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Tihomir
Culjaga
Sent: Wednesday, February 10, 2010 10:14 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] h323 - sip call is not working

 

 

On Wed, Feb 10, 2010 at 11:49 AM, Nikolay Kondratyev <kond at nstel.ru> wrote:

Hi all,

I compiled FreeSWITCH Version 1.0.5-20100209-0400 (16587M) with mod_h323.

When I call from h323 (Avaya IPOffice) to local fs extention (x-lite) I hear
ring back, but when x-lite picks up, he hears silence, while IPOffice user
continues to hear ringback.

The log is at the http://pastebin.freeswitch.org/12091

My configuration is almost default, several local extentions added, and
h323.conf from http://wiki.freeswitch.org/wiki/Mod_h323

5840 - user at IPOffice

2853 - x-lite registered at FS

IPOffice ip address: 172.23.14.2

FS ip address 172.23.22.49

 

Can anybody please advise how to solve that?

Is it a configuration or a software problem?

 


Please can you send me the tcpdump as well (not filtered). IPOffice i known
to have a "broken" H323 stack.
did you try to play with tunneling and h245 in setup settings as well ?

It looks like there is some h245 negotiation still pending but cant see that
from the logs.
 

Thanks in advance,

Nikolay.


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/4ac225cd/attachment.html 

From kond at nstel.ru  Thu Feb 11 00:08:18 2010
From: kond at nstel.ru (Nikolay Kondratyev)
Date: Thu, 11 Feb 2010 11:08:18 +0300
Subject: [Freeswitch-users] can't see sip messages in the log file
In-Reply-To: <b39273531002100650j2ff8652cu733256c88e5b7aea@mail.gmail.com>
Message-ID: <20100211080818.D469D1210F@mail.nstel.ru>

What is the version number where "sofia help" will show tracelevel? 

Thanks and regards,

Nikolay.

 

Thanks.  I've added that to the sofia help/completion.

On Wed, Feb 10, 2010 at 8:25 AM, Nikolay Kondratyev <kond at nstel.ru> wrote:

After some experiments i clarified this question.
SIP messages go into the freeswitch log when:
<param name="sip-trace" value="yes"/> ("sofia profile profile-name siptrace
on" is cli equivalent for this parameter)
AND
Sofia tracelevel is set to info (sofia tracelevel info).

The default sofia tracelevel is 'console'. That's why I did not see sip
messages in the log after turning on just "siptrace".

Thanks and regards,
Nikolay.


> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-
> users-bounces at lists.freeswitch.org] On Behalf Of Nikolay Kondratyev
> Sent: Wednesday, February 10, 2010 2:00 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] can't see sip messages in the log file
>
> Jason, thanks for the reply.
> Isn't "sofia profile internal siptrace on" a command line equivalent of
> <param name="sip-trace" value="yes"/>?
>
> Any way I tried it, but with the same result.
> I still don't see SIP.
>
> Thanks and regards,
> Nikolay.
>
> > > Can anybody please advise how to include sip messages into the log
> file?
> >
> > <param name="sip-trace" value="yes"/>
> > in the SIP profile you want to trace, then
> > sofia profile profile-name restart reloadxml
> > or restarting FreeSWITCH should do it.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




-- 
-Rupa

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/a32a0fe5/attachment-0001.html 

From kond at nstel.ru  Thu Feb 11 00:18:17 2010
From: kond at nstel.ru (Nikolay Kondratyev)
Date: Thu, 11 Feb 2010 11:18:17 +0300
Subject: [Freeswitch-users] h323 - sip call is not working
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C5576CE4923@cooper>
Message-ID: <20100211081817.BAB611200B@mail.nstel.ru>

> But I do recommenf to use SIP, since it's much better supported by FS.
> 
> /Peter
But SIP is poorly supported by IPO.
Thanks and regards,
Nikolay.



From mike at jerris.com  Thu Feb 11 00:36:16 2010
From: mike at jerris.com (Michael Jerris)
Date: Thu, 11 Feb 2010 03:36:16 -0500
Subject: [Freeswitch-users] Lua script hangup detect
In-Reply-To: <4B73B57E.9040901@gmail.com>
References: <314dc3f81002100117h6b1bc49cna989624a1ae4e0e2@mail.gmail.com>	<4B729155.7010708@gmail.com>	<314dc3f81002100800o41fd6561w394d5e83e69d3d80@mail.gmail.com>	<4B72DFF9.4040402@gmail.com>	<1b46b4e81002100932x7af486ekd4b05f4750cace20@mail.gmail.com>	<4B72F2A8.4070503@gmail.com>
	<4CA5432B-5F6E-438E-B6EF-3277847E7922@jerris.com>
	<4B73B57E.9040901@gmail.com>
Message-ID: <D8FFF1FB-7B2D-49BC-940C-445EBF838BE3@jerris.com>

luasql, now with free memory leaks, while supplies last (don't use mysql with lua? or really ever)

Mike

On Feb 11, 2010, at 2:45 AM, Nazim Agabekov wrote:

> Thanks Mike! Mod_Lua is great. I had experience with a lot of proprietary IVR systems in the past.
> FreeSWITCH and Mod_lua + luasql beats them all. Functionality is really impressive.
> 
> On 02/11/2010 09:21 AM, Michael Jerris wrote:
>> 
>> lua runs until you finish your script.  You need to exit when you are supposed to.  We have session::ready() for you to test this.
>> 
>> Mike
>> 
>> On Feb 10, 2010, at 12:53 PM, Nazim Agabekov wrote:
>> 
>>> I think it continues to run until it finishes. I'll check it tomorrow on my test system.
>>> 



From mike at jerris.com  Thu Feb 11 00:36:55 2010
From: mike at jerris.com (Michael Jerris)
Date: Thu, 11 Feb 2010 03:36:55 -0500
Subject: [Freeswitch-users] Seeking Advice on SIP Phones like Aastra
In-Reply-To: <1265873419114-4553224.post@n2.nabble.com>
References: <9DFA265F-8632-466B-8910-1E2F9D956908@jerris.com>
	<1265873419114-4553224.post@n2.nabble.com>
Message-ID: <F1451E5C-CE20-49C9-878D-B3B4ABD3BD78@jerris.com>

parse error, line too long

On Feb 11, 2010, at 2:30 AM, teldev wrote:

> Hello Mike, "Can you post for all to see some idea of how these applications work, lanagages used, some samples so we can see if we can get some interest in this?" "I am not sure the phone gui has much to do with FreeSWITCH at all, other than pulling a little data from the databases (again, ODBC highly recommended)" --> The idea would be to create a phone GUI on Aastra 6739i that would enable the touchscreen to do something like what is shown in these demos: http://www.youtube.com/watch?v=xZDHibW1gUs http://www.youtube.com/watch?v=Y58TPNCOTZI Perform FreeSWITCH actions: answer calls, place calls, setup conferences/add participants, etc. Display FreeSWITCH states: voice mail waiting count, extensions that are online/offline, call log, etc. Introduce new actions: dial By name directory with pictures, visual setup of conferences, etc. Since the Aastra 6739i, SNOM 870, and some others have built-in XML browsers it seems like they could be programmed to interact well with FreeSWITCH and various third party systems such as online feeds like weather reports, news, stock prices. Aastra is taking the IPhone approach of letting developers create the apps. They even have a similar slogan "There's an App for That" versus Apple's "There's an app for just about anything". http://www.ucstrategies.com/unified-communications-expert-views/aastra-phone-theres-an-app-for-that.aspx I don't know to what extent FreeSWITCH interacts with this phone out of the box, it reportedly works well with Asterisk though. I'm sure FS will likely understand certain button presses like hold, mute, speaker, transfer, etc. I will be ordering one within a few days. A few quesions for the FS community: Does this project have merit and fulfill a need? (I have yet to use FS as a PBX) Can anyone shed some light on what features on the phone will work and what I've mentioned will require custom development? Any ideas on how to architect an integration that would leverage what FS already does? Currently our IVR work uses Javascript/SpiderMonkey to call our own web service for database reads/writes. For this project, we'd need to interact more directly with FS and learn more about it's internals. teldev 
> View this message in context: Re: Seeking Advice on SIP Phones like Aastra
> Sent from the freeswitch-users mailing list archive at Nabble.com.
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/d19684fb/attachment.html 

From mike at jerris.com  Thu Feb 11 00:43:18 2010
From: mike at jerris.com (Michael Jerris)
Date: Thu, 11 Feb 2010 03:43:18 -0500
Subject: [Freeswitch-users] Seeking Advice on SIP Phones like Aastra
In-Reply-To: <1265875307968-4553322.post@n2.nabble.com>
References: <9DFA265F-8632-466B-8910-1E2F9D956908@jerris.com>
	<1265875307968-4553322.post@n2.nabble.com>
Message-ID: <5AFFA858-A491-467E-9AEA-F9F9BBEAD2AD@jerris.com>


On Feb 11, 2010, at 3:01 AM, teldev wrote:

> 
> Hello Mike,
> 
> Last post lost all formatting when I used Nabble, this is readable now.
> 
> "Can you post for all to see some idea of how these applications work,
> lanagages used, some samples so we can see if we can get some interest in
> this?"
> 
> "I am not sure the phone gui has much to do with FreeSWITCH at all, other
> than pulling a little data from the databases (again, ODBC highly
> recommended)" 
> 
> -->  The idea would be to create a phone GUI on Aastra 6739i that would
> enable the touchscreen to do something like what is shown in these demos:
> 
> http://www.youtube.com/watch?v=xZDHibW1gUs
> http://www.youtube.com/watch?v=Y58TPNCOTZI
> 
> Perform FreeSWITCH actions: answer calls, place calls, setup conferences/add
> participants, etc.
> Display FreeSWITCH states: voice mail waiting count, extensions that are
> online/offline, call log, etc. 
> Introduce new actions: dial By name directory with pictures, visual setup of
> conferences, etc.  
> 
> Since the Aastra 6739i, SNOM 870, and some others have built-in XML browsers
> it seems like they could be programmed to interact well with FreeSWITCH and
> various third party systems such as online feeds like weather reports, news,
> stock prices.  Aastra is taking the IPhone approach of letting developers
> create the apps.  They even have a similar slogan "There's an App for That"
> versus Apple's "There's an app for just about anything". 

Slogan ripoff FAIL

> 
> http://www.ucstrategies.com/unified-communications-expert-views/aastra-phone-theres-an-app-for-that.aspx
> 
> I don't know to what extent FreeSWITCH interacts with this phone out of the
> box, it reportedly works well with Asterisk though.  I'm sure FS will likely
> understand certain button presses like hold, mute, speaker, transfer, etc. 
> I will be ordering one within a few days. 
> 
> A few quesions for the FS community:
> 
> Does this project have merit and fulfill a need?  (I have yet to use FS as a
> PBX)
> 

Sure.

> Can anyone shed some light on what features on the phone will work and what
> I've mentioned will require custom development?

Thats a hard question, short answer, all the normal sip stuff, and a bunch of the abnormal stuff too.

> Any ideas on how to architect an integration that would leverage what FS
> already does?  Currently our IVR work uses Javascript/SpiderMonkey to call
> our own web service for database reads/writes.  For this project, we'd need
> to interact more directly with FS and learn more about it's internals.

I am not sure this is totally true, its probably pretty trivial database work for everything you need.

Mike






From mike at jerris.com  Thu Feb 11 00:45:37 2010
From: mike at jerris.com (Michael Jerris)
Date: Thu, 11 Feb 2010 03:45:37 -0500
Subject: [Freeswitch-users] can't see sip messages in the log file
In-Reply-To: <20100211080818.D469D1210F@mail.nstel.ru>
References: <20100211080818.D469D1210F@mail.nstel.ru>
Message-ID: <8A8A10A3-19CE-4C41-8EF2-D0630B9AE7B4@jerris.com>


Revision
16599
Author
rupa
Date
2010-02-10 08:49:32 -0600 (Wed, 10 Feb 2010)
Log Message

document tracelevel, add completion for tracelevel
Modified Paths

freeswitch/trunk/src/mod/endpoints/mod_sofia/mod_sofia.c

On Feb 11, 2010, at 3:08 AM, Nikolay Kondratyev wrote:

> What is the version number where ?sofia help? will show tracelevel?
> Thanks and regards,
> Nikolay.
>  
> Thanks.  I've added that to the sofia help/completion.
> 
> On Wed, Feb 10, 2010 at 8:25 AM, Nikolay Kondratyev <kond at nstel.ru> wrote:
> After some experiments i clarified this question.
> SIP messages go into the freeswitch log when:
> <param name="sip-trace" value="yes"/> ("sofia profile profile-name siptrace
> on" is cli equivalent for this parameter)
> AND
> Sofia tracelevel is set to info (sofia tracelevel info).
> 
> The default sofia tracelevel is 'console'. That's why I did not see sip
> messages in the log after turning on just "siptrace".
> 
> Thanks and regards,
> Nikolay.
> 
> > -----Original Message-----
> > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-
> > users-bounces at lists.freeswitch.org] On Behalf Of Nikolay Kondratyev
> > Sent: Wednesday, February 10, 2010 2:00 PM
> > To: freeswitch-users at lists.freeswitch.org
> > Subject: Re: [Freeswitch-users] can't see sip messages in the log file
> >
> > Jason, thanks for the reply.
> > Isn't "sofia profile internal siptrace on" a command line equivalent of
> > <param name="sip-trace" value="yes"/>?
> >
> > Any way I tried it, but with the same result.
> > I still don't see SIP.
> >
> > Thanks and regards,
> > Nikolay.
> >
> > > > Can anybody please advise how to include sip messages into the log
> > file?
> > >
> > > <param name="sip-trace" value="yes"/>
> > > in the SIP profile you want to trace, then
> > > sofia profile profile-name restart reloadxml
> > > or restarting FreeSWITCH should do it.
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> -- 
> -Rupa
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/8c89cd95/attachment-0001.html 

From nazim.agabekov at gmail.com  Thu Feb 11 00:56:32 2010
From: nazim.agabekov at gmail.com (Nazim Agabekov)
Date: Thu, 11 Feb 2010 12:56:32 +0400
Subject: [Freeswitch-users] Lua script hangup detect
In-Reply-To: <D8FFF1FB-7B2D-49BC-940C-445EBF838BE3@jerris.com>
References: <314dc3f81002100117h6b1bc49cna989624a1ae4e0e2@mail.gmail.com>	<4B729155.7010708@gmail.com>	<314dc3f81002100800o41fd6561w394d5e83e69d3d80@mail.gmail.com>	<4B72DFF9.4040402@gmail.com>	<1b46b4e81002100932x7af486ekd4b05f4750cace20@mail.gmail.com>	<4B72F2A8.4070503@gmail.com>	<4CA5432B-5F6E-438E-B6EF-3277847E7922@jerris.com>	<4B73B57E.9040901@gmail.com>
	<D8FFF1FB-7B2D-49BC-940C-445EBF838BE3@jerris.com>
Message-ID: <4B73C640.20700@gmail.com>

I'm using luasql with ODBC MySQL driver in production. I've never tried
to use luasql with "native" mysql driver, but ODBC one works great.

On 02/11/2010 12:36 PM, Michael Jerris wrote:
> luasql, now with free memory leaks, while supplies last (don't use mysql with lua? or really ever)
>
> Mike
>
> On Feb 11, 2010, at 2:45 AM, Nazim Agabekov wrote:
>
>   
>> Thanks Mike! Mod_Lua is great. I had experience with a lot of proprietary IVR systems in the past.
>> FreeSWITCH and Mod_lua + luasql beats them all. Functionality is really impressive.
>>
>> On 02/11/2010 09:21 AM, Michael Jerris wrote:
>>     
>>> lua runs until you finish your script.  You need to exit when you are supposed to.  We have session::ready() for you to test this.
>>>
>>> Mike
>>>
>>> On Feb 10, 2010, at 12:53 PM, Nazim Agabekov wrote:
>>>
>>>       
>>>> I think it continues to run until it finishes. I'll check it tomorrow on my test system.
>>>>
>>>>         
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   



From max.bridgewater at gmail.com  Thu Feb 11 01:03:59 2010
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Thu, 11 Feb 2010 01:03:59 -0800
Subject: [Freeswitch-users] Freeswitch and G729
Message-ID: <c55d46951002110103y458b4b15r33df3e403d5eeef2@mail.gmail.com>

Hi,

Some quick questions related to the upcomming Freeswitch G729 support:

1) When can it be tried?
2) Does it support lower bit rate extensions such as D, F, H, I, C?

Thanks,
Max.


From jingwei.yang at gmail.com  Thu Feb 11 01:30:44 2010
From: jingwei.yang at gmail.com (Jingwei Yang)
Date: Thu, 11 Feb 2010 17:30:44 +0800
Subject: [Freeswitch-users] Is it possible to specify music on hold with
	uuid_hold?
Message-ID: <13529f9d1002110130v335d8177m74f791577b558ca@mail.gmail.com>

Hello,

Is it possible to specify a customized music file when the caller is put on
hold by uuid_hold?

Maybe something like this? uuid_hold {music_on_hold=/tmp/moh.wav}
<caller_uuid>

Regards,
-Jingwei
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/cf729028/attachment.html 

From mcampbellsmith at gmail.com  Thu Feb 11 01:31:08 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Thu, 11 Feb 2010 20:31:08 +1100
Subject: [Freeswitch-users] Forbidden using UDP, works with TCP/TLS
In-Reply-To: <2EF27EFC-1634-46A3-B438-2EE99BEF58F1@jerris.com>
References: <33c87fa31002101824h21e3bf7lb13b68e8d4fd36a2@mail.gmail.com>
	<2EF27EFC-1634-46A3-B438-2EE99BEF58F1@jerris.com>
Message-ID: <33c87fa31002110131w3cf6ad00n802669856085a879@mail.gmail.com>

I don't see debug logs (fsctl debug_level 9 or loglevel 9) during the
registration process.

All I see is the sip messages when the sip trace is activated (403 Forbidden)

Is there other debugging that I can enable?

On Thu, Feb 11, 2010 at 4:25 PM, Michael Jerris <mike at jerris.com> wrote:
> Look at the FreeSWITCH debug logs and compare the differences, they should tell you why.
>
> Mike
>
> On Feb 10, 2010, at 9:24 PM, Mark Campbell-Smith wrote:
>
>> Hi!
>>
>> I had a user registered using TLS transport. That was working fine but
>> I want to change the ATA over to use UDP instead.
>>
>> All I thought I should have to do was to change the transport and
>> ports used to register in the ATA (SPA3102). ?However, when I do this,
>> FS responds with Forbidden.
>>
>> When I change the settings back to use TCP or TLS, registration is successful.
>>
>> What would cause FS to respond with forbidden? ?I do not change the
>> username/password fields in either case.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From mike at jerris.com  Thu Feb 11 01:57:55 2010
From: mike at jerris.com (Michael Jerris)
Date: Thu, 11 Feb 2010 04:57:55 -0500
Subject: [Freeswitch-users] Forbidden using UDP, works with TCP/TLS
In-Reply-To: <33c87fa31002110131w3cf6ad00n802669856085a879@mail.gmail.com>
References: <33c87fa31002101824h21e3bf7lb13b68e8d4fd36a2@mail.gmail.com>
	<2EF27EFC-1634-46A3-B438-2EE99BEF58F1@jerris.com>
	<33c87fa31002110131w3cf6ad00n802669856085a879@mail.gmail.com>
Message-ID: <CA21231C-108B-4445-A3B9-BF8D94FFE898@jerris.com>

you can crank up the sofia loglevel as well

Mike

On Feb 11, 2010, at 4:31 AM, Mark Campbell-Smith wrote:

> I don't see debug logs (fsctl debug_level 9 or loglevel 9) during the
> registration process.
> 
> All I see is the sip messages when the sip trace is activated (403 Forbidden)
> 
> Is there other debugging that I can enable?
> 
> On Thu, Feb 11, 2010 at 4:25 PM, Michael Jerris <mike at jerris.com> wrote:
>> Look at the FreeSWITCH debug logs and compare the differences, they should tell you why.
>> 
>> Mike
>> 
>> On Feb 10, 2010, at 9:24 PM, Mark Campbell-Smith wrote:
>> 
>>> Hi!
>>> 
>>> I had a user registered using TLS transport. That was working fine but
>>> I want to change the ATA over to use UDP instead.
>>> 
>>> All I thought I should have to do was to change the transport and
>>> ports used to register in the ATA (SPA3102).  However, when I do this,
>>> FS responds with Forbidden.
>>> 
>>> When I change the settings back to use TCP or TLS, registration is successful.
>>> 
>>> What would cause FS to respond with forbidden?  I do not change the
>>> username/password fields in either case.
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From nagalenoj at gmail.com  Thu Feb 11 02:11:49 2010
From: nagalenoj at gmail.com (Nagalenoj H.)
Date: Thu, 11 Feb 2010 15:41:49 +0530
Subject: [Freeswitch-users] Play music to A leg.
In-Reply-To: <87f2f3b91002100702u7581589ua866d2492a008b23@mail.gmail.com>
References: <fe67a3341002090519r2cc6e8edm7702b8d012fc2be3@mail.gmail.com>
	<87f2f3b91002090737l4efa8en29bde9457fef7ea5@mail.gmail.com>
	<fe67a3341002092157x7484f3d2id6678510bbec66cc@mail.gmail.com>
	<87f2f3b91002100702u7581589ua866d2492a008b23@mail.gmail.com>
Message-ID: <fe67a3341002110211w60390e1cm2d12fadd1e802778@mail.gmail.com>

But My scenario is,
   After I get the call from X.
   I answer the call in some scenarios and won't answer the call. So, this
leg can either be answered or unanswered.
   I originate a call to another number.
   After getting some digits from this originated leg.
   I do uuid_bridge of these 2 legs.

I want to play some file[ringback] to leg A before bridging to B.

On Wed, Feb 10, 2010 at 8:32 PM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Tue, Feb 9, 2010 at 9:57 PM, Nagalenoj H. <nagalenoj at gmail.com> wrote:
>
>> Because, I want to get some digits before bridging the legs. I've tried
>> group_confirm_key, but it accepts only one digit, I need multiple digits, so
>> I can't use.
>> I've also tried group_confirm_file, but when I do originate for multiple
>> extensions, I want this script to work based on the answered extension.
>>
>> So, I've originated and processed the events to do my job.
>>
>> How do I play some music to A leg?
>>
>> I might be missing something, but couldn't you just park the call ("A
> leg") until you connect to the other party ("B leg") and then uuid_bridge at
> whatever point you want?
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Regards,
Nagalenoj H.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/04bf0402/attachment.html 

From mcampbellsmith at gmail.com  Thu Feb 11 02:13:53 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Thu, 11 Feb 2010 21:13:53 +1100
Subject: [Freeswitch-users] Forbidden using UDP, works with TCP/TLS
In-Reply-To: <CA21231C-108B-4445-A3B9-BF8D94FFE898@jerris.com>
References: <33c87fa31002101824h21e3bf7lb13b68e8d4fd36a2@mail.gmail.com>
	<2EF27EFC-1634-46A3-B438-2EE99BEF58F1@jerris.com>
	<33c87fa31002110131w3cf6ad00n802669856085a879@mail.gmail.com>
	<CA21231C-108B-4445-A3B9-BF8D94FFE898@jerris.com>
Message-ID: <33c87fa31002110213q1de0eebfsb950d8725ea9aba@mail.gmail.com>

ah thats true... The trace is not too readable to me, but may give
some insight to someone that can read the sofia logs....


recv 752 bytes from udp/[121.xxx.xxx.xxx]:5060 at 10:10:37.803288:
   ------------------------------------------------------------------------
   REGISTER sip:mydns.dyndns.org SIP/2.0
   Via: SIP/2.0/UDP 121.xxx.xxx.xxx:5060;branch=z9hG4bK-9052c91f
   From: 2000 <sip:2000 at mydns.dyndns.org>;tag=7a9dbbbfa691136do0
   To: 2000 <sip:2000 at mydns.dyndns.org>
   Call-ID: 610db38-dd3b511f at 121.xxx.xxx.xxx
   CSeq: 32330 REGISTER
   Contact: 2000 <sip:2000 at 121.xxx.xxx.xxx:5060>;expires=900
   Authorization: Digest username="2000", realm="mydns.dyndns.org",
nonce="b3298cfe-16f5-11df-9734-e77ba7a22ac3",
uri="sip:mydns.dyndns.org",
response="724fca542ce08d3f12b9ba1043bebb0c", algorithm=MD5,
qop="1225e2f1"
   Max-Forwards: 70
   User-Agent: Linksys/SPA3102-5.1.10(GW)
   Supported: x-sipura
   Supported: replaces
   Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
   Content-Length: 0


   ------------------------------------------------------------------------
tport_deliver(0x97cde80): msg 0x98297e8 (752 bytes) from
udp/121.xxx.xxx.xxx:5060/sip next=(nil)
nta: received REGISTER sip:mydns.dyndns.org SIP/2.0 (CSeq 32330)
nta: REGISTER (32330) going to a default leg
nua: nua_stack_process_request: entering
nua: nh_create: entering
nua: nh_create_handle: entering
nua: nua_stack_set_params: entering
soa_clone(static::0x97cc698, 0x9794808, 0x981cb70) called
soa_set_params(static::0x9758ba8, ...) called
nua(0x981cb70): event i_register 100 Trying
nua: nua_application_event: entering
nua: nua_respond: entering
nua(0x981cb70): sent signal r_respond
nua: nua_handle_destroy: entering
nua(0x981cb70): sent signal r_destroy
nua: nua_handle_magic: entering
nua: nua_handle_destroy: entering
nua(0x981cb70): recv signal r_respond 403 Forbidden
nua: nua_stack_set_params: entering
soa_set_params(static::0x9758ba8, ...) called
tport_tsend(0x97cde80) tpn = UDP/121.xxx.xxx.xxx:5060
tport_resolve addrinfo = 121.xxx.xxx.xxx:5060
tport_by_addrinfo(0x97cde80): not found by name UDP/121.xxx.xxx.xxx:5060
tport_vsend(0x97cde80): 495 bytes of 495 to udp/121.xxx.xxx.xxx:5060
tport_vsend returned 495
send 495 bytes to udp/[121.xxx.xxx.xxx]:5060 at 10:10:37.812955:
   ------------------------------------------------------------------------
   SIP/2.0 403 Forbidden

On Thu, Feb 11, 2010 at 8:57 PM, Michael Jerris <mike at jerris.com> wrote:
> you can crank up the sofia loglevel as well
>
> Mike
>
> On Feb 11, 2010, at 4:31 AM, Mark Campbell-Smith wrote:
>
>> I don't see debug logs (fsctl debug_level 9 or loglevel 9) during the
>> registration process.
>>
>> All I see is the sip messages when the sip trace is activated (403 Forbidden)
>>
>> Is there other debugging that I can enable?
>>
>> On Thu, Feb 11, 2010 at 4:25 PM, Michael Jerris <mike at jerris.com> wrote:
>>> Look at the FreeSWITCH debug logs and compare the differences, they should tell you why.
>>>
>>> Mike
>>>
>>> On Feb 10, 2010, at 9:24 PM, Mark Campbell-Smith wrote:
>>>
>>>> Hi!
>>>>
>>>> I had a user registered using TLS transport. That was working fine but
>>>> I want to change the ATA over to use UDP instead.
>>>>
>>>> All I thought I should have to do was to change the transport and
>>>> ports used to register in the ATA (SPA3102). ?However, when I do this,
>>>> FS responds with Forbidden.
>>>>
>>>> When I change the settings back to use TCP or TLS, registration is successful.
>>>>
>>>> What would cause FS to respond with forbidden? ?I do not change the
>>>> username/password fields in either case.
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From jingwei.yang at gmail.com  Thu Feb 11 02:33:10 2010
From: jingwei.yang at gmail.com (Jingwei Yang)
Date: Thu, 11 Feb 2010 18:33:10 +0800
Subject: [Freeswitch-users]  Is it possible to repeat music in playback
Message-ID: <13529f9d1002110233g57c8a1a5j3cdd38e22fa6a13d@mail.gmail.com>

Hello,

I've defined a very simple dialplan like the one below and when the caller
is connected to this plan, I hope to keep the call alive and repeat the
music set by playback. How am I able to achieve this?


    <extension name="agent_hold">
      <condition field="destination_number" expression="^33333$">
        <action application="answer"/>
        <action application="playback" data="customized/test.wav"/>
      </condition>
    </extension>


Thanks,
-Jingwei
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/d911bda1/attachment.html 

From tculjaga at gmail.com  Thu Feb 11 03:34:22 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Thu, 11 Feb 2010 12:34:22 +0100
Subject: [Freeswitch-users] h323 - sip call is not working
In-Reply-To: <20100211081817.BAB611200B@mail.nstel.ru>
References: <549CFEF87AEDE841A38E9D15EAB4C04C5576CE4923@cooper>
	<20100211081817.BAB611200B@mail.nstel.ru>
Message-ID: <65d96fc81002110334o1ec01bddp521528cab618acfc@mail.gmail.com>

Nikolay, you are sending slow start with tunneling=true ?!?!

It is not gong to work :)

Please can you set fast start instead?


Your call failed because there was no mediaControll channel negotiated at
all... actually the call had to be aborted because wrong signaling .. but
anyhow.

Please on your IPO use FastStart with h245Tunneling=true... also, same
settings on FS side as well (exclude h245 in setup as well).




Frame 13 (277 bytes on wire, 277 bytes captured)
Ethernet II, Src: AlliedTe_22:9b:4a (00:00:cd:22:9b:4a), Dst:
Vmware_67:33:a7 (00:0c:29:67:33:a7)
Internet Protocol, Src: 172.23.14.2 (172.23.14.2), Dst: 172.23.22.49
(172.23.22.49)
Transmission Control Protocol, Src Port: oirtgsvc (4141), Dst Port:
h323hostcall (1720), Seq: 1, Ack: 1, Len: 223
TPKT, Version: 3, Length: 223
Q.931
    Protocol discriminator: Q.931
    Call reference value length: 2
    Call reference flag: Message sent from originating side
    Call reference value: 0012
    Message type: SETUP (0x05)
    Bearer capability
    Display  'Gornak Alexandr>2853'
    Calling party number: '5840'
    Called party number: '2853'
    User-user
H.225.0 CS
    H323-UserInformation
        h323-uu-pdu
            h323-message-body: setup (0)
                setup
            h4501SupplementaryService: 1 item
           * 1... .... h245Tunneling: True*







On Thu, Feb 11, 2010 at 9:18 AM, Nikolay Kondratyev <kond at nstel.ru> wrote:

> > But I do recommenf to use SIP, since it's much better supported by FS.
> >
> > /Peter
> But SIP is poorly supported by IPO.
> Thanks and regards,
> Nikolay.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/03009b45/attachment-0001.html 

From dist.lists at gmail.com  Thu Feb 11 04:08:19 2010
From: dist.lists at gmail.com (Hristo Trendev)
Date: Thu, 11 Feb 2010 14:08:19 +0200
Subject: [Freeswitch-users] Skypiax latency
In-Reply-To: <4B72FE87.4000401@gmx.net>
References: <4B72FE87.4000401@gmx.net>
Message-ID: <2a73afe1002110408r7f72cabdwf6a02ba27a314065@mail.gmail.com>

Hi Peter,
Take a look at http://jira.freeswitch.org/browse/MODSKYPIAX-29 and
http://jira.freeswitch.org/browse/MODLANG-130. I haven't tested the
trick in MODLANG-130 with skypiax yet, but if you use script to handle
the calls this may help.

BR,
Hristo


On 2/10/10, Peter P GMX <Prometheus001 at gmx.net> wrote:
> Hello,
>
> I have a problem with latency and mod_skypiax
>
> Skype=>SIP is always fine (~0.3sec)
> SIP => Skype is always bad (~2-4 sec)
> I would expect that latency in both directions should be the same.
>
> Anybody has discovered this before and has a solution?
>
> The scenario is as follows:
>
> SIP-UA => FreeswitchServer=>Freeswitch/SkypIaxServer=>Skype
> Both freeswitch servers are in the same LAN, so latency should be low.
>
> Best regards
> Peter
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From paul at apcl.us  Thu Feb 11 05:10:53 2010
From: paul at apcl.us (Paul Levin)
Date: Thu, 11 Feb 2010 08:10:53 -0500
Subject: [Freeswitch-users] how to set From address for voicemail emails?
In-Reply-To: <7471DDD5-0A73-4F18-AF90-BA8534EBC9E7@jerris.com>
References: <4B736846.1040908@apcl.us>
	<7471DDD5-0A73-4F18-AF90-BA8534EBC9E7@jerris.com>
Message-ID: <4B7401DD.6050408@apcl.us>

An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/283fbabf/attachment.html 

From brian at freeswitch.org  Thu Feb 11 05:22:03 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 11 Feb 2010 07:22:03 -0600
Subject: [Freeswitch-users] Freeswitch and G729
In-Reply-To: <c55d46951002110103y458b4b15r33df3e403d5eeef2@mail.gmail.com>
References: <c55d46951002110103y458b4b15r33df3e403d5eeef2@mail.gmail.com>
Message-ID: <0737B9F4-25A7-4044-BC50-97388844C739@freeswitch.org>


On Feb 11, 2010, at 3:03 AM, Max Bridgewater wrote:

> Hi,
> 
> Some quick questions related to the upcomming Freeswitch G729 support:
> 
> 1) When can it be tried?

SOON!

> 2) Does it support lower bit rate extensions such as D, F, H, I, C?

A and B.

> 
> Thanks,
> Max.



From brian at freeswitch.org  Thu Feb 11 05:22:31 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 11 Feb 2010 07:22:31 -0600
Subject: [Freeswitch-users] Is it possible to specify music on hold with
	uuid_hold?
In-Reply-To: <13529f9d1002110130v335d8177m74f791577b558ca@mail.gmail.com>
References: <13529f9d1002110130v335d8177m74f791577b558ca@mail.gmail.com>
Message-ID: <8FF07098-581B-47FB-A039-63A58041E5BF@freeswitch.org>

uuid_hold isn't for that.  It sends the hold indication to the far end... not the near end.

/b

On Feb 11, 2010, at 3:30 AM, Jingwei Yang wrote:

> Hello,
> 
> Is it possible to specify a customized music file when the caller is put on hold by uuid_hold?
> 
> Maybe something like this? uuid_hold {music_on_hold=/tmp/moh.wav} <caller_uuid>
> 
> Regards,
> -Jingwei



From brian at freeswitch.org  Thu Feb 11 05:26:21 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 11 Feb 2010 07:26:21 -0600
Subject: [Freeswitch-users] Is it possible to repeat music in playback
In-Reply-To: <13529f9d1002110233g57c8a1a5j3cdd38e22fa6a13d@mail.gmail.com>
References: <13529f9d1002110233g57c8a1a5j3cdd38e22fa6a13d@mail.gmail.com>
Message-ID: <2FBDB302-2B54-40CC-B694-332854219774@freeswitch.org>

Why not just use Fifo to hold them?  Or Park the agent and send the session a message to play music?  You then have options to define loop count.

http://wiki.freeswitch.org/wiki/Event_Socket#execute

/b

On Feb 11, 2010, at 4:33 AM, Jingwei Yang wrote:

> Hello,
> 
> I've defined a very simple dialplan like the one below and when the caller is connected to this plan, I hope to keep the call alive and repeat the music set by playback. How am I able to achieve this?
> 
> 
>     <extension name="agent_hold">
>       <condition field="destination_number" expression="^33333$">
>         <action application="answer"/>
>         <action application="playback" data="customized/test.wav"/>
>       </condition>
>     </extension>
> 
> 
> Thanks,
> -Jingwei



From brian at freeswitch.org  Thu Feb 11 05:26:59 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 11 Feb 2010 07:26:59 -0600
Subject: [Freeswitch-users] Lua script hangup detect
In-Reply-To: <4B73C640.20700@gmail.com>
References: <314dc3f81002100117h6b1bc49cna989624a1ae4e0e2@mail.gmail.com>	<4B729155.7010708@gmail.com>	<314dc3f81002100800o41fd6561w394d5e83e69d3d80@mail.gmail.com>	<4B72DFF9.4040402@gmail.com>	<1b46b4e81002100932x7af486ekd4b05f4750cace20@mail.gmail.com>	<4B72F2A8.4070503@gmail.com>	<4CA5432B-5F6E-438E-B6EF-3277847E7922@jerris.com>	<4B73B57E.9040901@gmail.com>
	<D8FFF1FB-7B2D-49BC-940C-445EBF838BE3@jerris.com>
	<4B73C640.20700@gmail.com>
Message-ID: <FFD1A126-5CC0-4D02-87CE-B3EBA553FC4A@freeswitch.org>

Yes ODBC doesn't seem to leak... while the native one leaks like a sieve.

/b

On Feb 11, 2010, at 2:56 AM, Nazim Agabekov wrote:

> I'm using luasql with ODBC MySQL driver in production. I've never tried
> to use luasql with "native" mysql driver, but ODBC one works great.



From rupa at rupa.com  Thu Feb 11 05:28:22 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Thu, 11 Feb 2010 07:28:22 -0600
Subject: [Freeswitch-users] Need help setting up a feature
In-Reply-To: <4B73A799020000E100000470@mail.fribert.dk>
References: <4B73A799020000E100000470@mail.fribert.dk>
Message-ID: <b39273531002110528p42218d07q95e97485139315f1@mail.gmail.com>

My Siemens A580 has options for controlling the R key.  It seems that you
can either have it setup for transfer or as a hook flash.  Default is as a
transfer key.

I haven't succeeded in getting it to work for transfer and it is wayyyyy
down low on my list of things to do with the phone.

>From the web UI:

 Call Transfer   Use the R key to initiate call transfer with the SIP Refer
method.: Yes    No  Transfer Call by On-Hook: Yes    No  Derive target
address: from SIP URL    from SIP contact header  Find target addr.
automatically: Yes    No
  Hook Flash (R-key)     R key settings are disabled because the R key is
being used for call transfer.

On Wed, Feb 10, 2010 at 11:45 PM, mailinglist <mailinglist at fribert.dk>wrote:

>  Sorry for the repost, but the previous thread just died :-)
>
> I'm trying to get the possibility of transfering an incoming call from one
> extension to another, and give the possibility of turning it into a
> conference.
> I don't have a 'transfer' button.
> I do have an 'R' button on the Siemens handsets, and a 'Flash' button on
> the Sipura. The 'Flash' button gives me a new dialtone, gives the caller
> MOH, and then I can dial the new extension, and transfer the call, but not
> create a conference.
> But the Siemens handset does not have a 'flash', and pressing the R doesn't
> do anything.
>
> It might be two different features 'transfer' and 'conference'...
>
> But I thought that using the bind_meta_app would accomplish both.
>
> It's on an incoming call from the outside.
> So the situation:
> The Public folder has an entry that matches the dialed number, and does a
> transfer to 8202.
> Then the dialplan matches the 8202 with a group, and the phone rings.
> Somebody picks it up, finds out that it needs to be transferred to another
> extension, or transferred to a conference with a second extension.
> How do I construct that?
>
>
>
> Best regards
> Fribse
>
> /Running Freeswitch on pfSense running on a VMWare ESXi 4.0
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/28955317/attachment.html 

From moizchinoy at gmail.com  Thu Feb 11 05:32:55 2010
From: moizchinoy at gmail.com (Moiz Chinoy)
Date: Thu, 11 Feb 2010 17:32:55 +0400
Subject: [Freeswitch-users] Is it necessary to call hangup....
Message-ID: <29b888f81002110532o3136d376g655c4b7a6dcbf598@mail.gmail.com>

Hello,

Is it necessary to call hangup in dialplan after say playing a file
through playback application.

-- 
Regards,
Moiz Chinoy.


From brian at freeswitch.org  Thu Feb 11 05:36:53 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 11 Feb 2010 07:36:53 -0600
Subject: [Freeswitch-users] Is it necessary to call hangup....
In-Reply-To: <29b888f81002110532o3136d376g655c4b7a6dcbf598@mail.gmail.com>
References: <29b888f81002110532o3136d376g655c4b7a6dcbf598@mail.gmail.com>
Message-ID: <7F8C140A-01BD-45A9-B71A-D3036BE10B3E@freeswitch.org>

If the dialplan runs out of stuff to do it'll hang up on you anyway.

/b

On Feb 11, 2010, at 7:32 AM, Moiz Chinoy wrote:

> Hello,
> 
> Is it necessary to call hangup in dialplan after say playing a file
> through playback application.
> 
> -- 
> Regards,
> Moiz Chinoy.



From wessels147 at gmail.com  Thu Feb 11 05:40:31 2010
From: wessels147 at gmail.com (wessels)
Date: Thu, 11 Feb 2010 14:40:31 +0100
Subject: [Freeswitch-users] multiple isdn msn on an extension with a single
	sip account
Message-ID: <a58c5fe71002110540q4e8eabb9qd8c57fc765d69bca@mail.gmail.com>

Hi,

I'm evaluating the replacement of an ISDN pbx. One of the things I
can't figure out is a suitable replacement for the msn numbers on a
phone with a single sip account.
The current ISDN phones can be programmed to recognize up to eight
different msn numbers. The user of the phone can see and hear which
msn number was called.
So I've got a BRI or PRI , with multiple msn, feed that into
freeswitch and simple extensions like the linksys spa922. I want the
extension to ring on more than one msn but I want to be able to see
which msn was originally called in the display. Can this be setup
using freeswitch and a simple phone??? Any directions would be most
helpful.

Thank you,

Wessel s


From jingwei.yang at gmail.com  Thu Feb 11 05:54:42 2010
From: jingwei.yang at gmail.com (Jingwei Yang)
Date: Thu, 11 Feb 2010 21:54:42 +0800
Subject: [Freeswitch-users] Is it possible to specify music on hold with
	uuid_hold?
In-Reply-To: <8FF07098-581B-47FB-A039-63A58041E5BF@freeswitch.org>
References: <13529f9d1002110130v335d8177m74f791577b558ca@mail.gmail.com>
	<8FF07098-581B-47FB-A039-63A58041E5BF@freeswitch.org>
Message-ID: <13529f9d1002110554u65431499hc3fa877252023577@mail.gmail.com>

Sorry Brian, I don't quite understand your answer. What is the far end and
what is the near end? In my case, I bridge client A to agent B. While
they're talking, I use uuid_hold to put client A on hold. Then A hears the
default music. After a while, uuid_hold off A and the conversation between A
and B resumes. uuid_hold looks perfect for my situation except I'm not able
to change the default music.

Regards,
-Jingwei

On Thu, Feb 11, 2010 at 9:22 PM, Brian West <brian at freeswitch.org> wrote:

> uuid_hold isn't for that.  It sends the hold indication to the far end...
> not the near end.
>
> /b
>
> On Feb 11, 2010, at 3:30 AM, Jingwei Yang wrote:
>
> > Hello,
> >
> > Is it possible to specify a customized music file when the caller is put
> on hold by uuid_hold?
> >
> > Maybe something like this? uuid_hold {music_on_hold=/tmp/moh.wav}
> <caller_uuid>
> >
> > Regards,
> > -Jingwei
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/79efae51/attachment.html 

From brian at freeswitch.org  Thu Feb 11 05:59:53 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 11 Feb 2010 07:59:53 -0600
Subject: [Freeswitch-users] Is it possible to specify music on hold with
	uuid_hold?
In-Reply-To: <13529f9d1002110554u65431499hc3fa877252023577@mail.gmail.com>
References: <13529f9d1002110130v335d8177m74f791577b558ca@mail.gmail.com>
	<8FF07098-581B-47FB-A039-63A58041E5BF@freeswitch.org>
	<13529f9d1002110554u65431499hc3fa877252023577@mail.gmail.com>
Message-ID: <E37048AD-5A46-42D5-B724-6C2344AD8A2B@freeswitch.org>

uuid_setvar the variable hold_music on the opposite UUID you're holding... uuid_hold isn't doing exactly what you think it is.  ;)

/b

On Feb 11, 2010, at 7:54 AM, Jingwei Yang wrote:

> Sorry Brian, I don't quite understand your answer. What is the far end and what is the near end? In my case, I bridge client A to agent B. While they're talking, I use uuid_hold to put client A on hold. Then A hears the default music. After a while, uuid_hold off A and the conversation between A and B resumes. uuid_hold looks perfect for my situation except I'm not able to change the default music.
> 
> Regards,
> -Jingwei



From anthony.minessale at gmail.com  Thu Feb 11 06:39:15 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 11 Feb 2010 08:39:15 -0600
Subject: [Freeswitch-users] Is it possible to repeat music in playback
In-Reply-To: <2FBDB302-2B54-40CC-B694-332854219774@freeswitch.org>
References: <13529f9d1002110233g57c8a1a5j3cdd38e22fa6a13d@mail.gmail.com>
	<2FBDB302-2B54-40CC-B694-332854219774@freeswitch.org>
Message-ID: <191c3a031002110639h3cf14efdvf7715b973fe3caed@mail.gmail.com>

or try endless_playback app


On Thu, Feb 11, 2010 at 7:26 AM, Brian West <brian at freeswitch.org> wrote:

> Why not just use Fifo to hold them?  Or Park the agent and send the session
> a message to play music?  You then have options to define loop count.
>
> http://wiki.freeswitch.org/wiki/Event_Socket#execute
>
> /b
>
> On Feb 11, 2010, at 4:33 AM, Jingwei Yang wrote:
>
> > Hello,
> >
> > I've defined a very simple dialplan like the one below and when the
> caller is connected to this plan, I hope to keep the call alive and repeat
> the music set by playback. How am I able to achieve this?
> >
> >
> >     <extension name="agent_hold">
> >       <condition field="destination_number" expression="^33333$">
> >         <action application="answer"/>
> >         <action application="playback" data="customized/test.wav"/>
> >       </condition>
> >     </extension>
> >
> >
> > Thanks,
> > -Jingwei
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/e7642b87/attachment.html 

From kond at nstel.ru  Thu Feb 11 07:18:18 2010
From: kond at nstel.ru (Nikolay Kondratyev)
Date: Thu, 11 Feb 2010 18:18:18 +0300
Subject: [Freeswitch-users] h323 - sip call is not working
In-Reply-To: <65d96fc81002110334o1ec01bddp521528cab618acfc@mail.gmail.com>
Message-ID: <20100211151819.95FC711F60@mail.nstel.ru>

Tihomir,

Thanks for help.

I enabled fast start on IPO and I can hear voice now. But the ringback tone
and voice appears to be wheezy, but I will investigate that tomorrow.

Thanks again.

Nikolay.

 

 

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Tihomir
Culjaga
Sent: Thursday, February 11, 2010 2:34 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] h323 - sip call is not working

 

Nikolay, you are sending slow start with tunneling=true ?!?!

It is not gong to work :)

Please can you set fast start instead?


Your call failed because there was no mediaControll channel negotiated at
all... actually the call had to be aborted because wrong signaling .. but
anyhow.

Please on your IPO use FastStart with h245Tunneling=true... also, same
settings on FS side as well (exclude h245 in setup as well).




Frame 13 (277 bytes on wire, 277 bytes captured)
Ethernet II, Src: AlliedTe_22:9b:4a (00:00:cd:22:9b:4a), Dst:
Vmware_67:33:a7 (00:0c:29:67:33:a7)
Internet Protocol, Src: 172.23.14.2 (172.23.14.2), Dst: 172.23.22.49
(172.23.22.49)
Transmission Control Protocol, Src Port: oirtgsvc (4141), Dst Port:
h323hostcall (1720), Seq: 1, Ack: 1, Len: 223
TPKT, Version: 3, Length: 223
Q.931
    Protocol discriminator: Q.931
    Call reference value length: 2
    Call reference flag: Message sent from originating side
    Call reference value: 0012
    Message type: SETUP (0x05)
    Bearer capability
    Display  'Gornak Alexandr>2853'
    Calling party number: '5840'
    Called party number: '2853'
    User-user
H.225.0 CS
    H323-UserInformation
        h323-uu-pdu
            h323-message-body: setup (0)
                setup
            h4501SupplementaryService: 1 item
            1... .... h245Tunneling: True








On Thu, Feb 11, 2010 at 9:18 AM, Nikolay Kondratyev <kond at nstel.ru> wrote:

> But I do recommenf to use SIP, since it's much better supported by FS.
>
> /Peter

But SIP is poorly supported by IPO.
Thanks and regards,

Nikolay.


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org

http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/fc3f9420/attachment-0001.html 

From ederwander at gmail.com  Thu Feb 11 07:24:03 2010
From: ederwander at gmail.com (Eder Souza)
Date: Thu, 11 Feb 2010 13:24:03 -0200
Subject: [Freeswitch-users] call die after 30 seconds in lua script
Message-ID: <622bedea1002110724p4757a5a4r42a9da7330f10498@mail.gmail.com>

Hi list

im testing the script welcome.lua from
http://wiki.freeswitch.org/wiki/Lua_Welcome_IVR_Example

but when do a transfer my call drop in 30 seconds just in script lua!!

callig one ramal (ex: 12345) direct in x-lite the call works !!

When make a session:execute("transfer","12345") in the script lua the call
drop after 30 seconds!!


why ??


Att,


Eng Eder de Souza
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/1bd4d037/attachment.html 

From msc at freeswitch.org  Thu Feb 11 07:54:03 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 11 Feb 2010 07:54:03 -0800
Subject: [Freeswitch-users] Play music to A leg.
In-Reply-To: <fe67a3341002110211w60390e1cm2d12fadd1e802778@mail.gmail.com>
References: <fe67a3341002090519r2cc6e8edm7702b8d012fc2be3@mail.gmail.com>
	<87f2f3b91002090737l4efa8en29bde9457fef7ea5@mail.gmail.com>
	<fe67a3341002092157x7484f3d2id6678510bbec66cc@mail.gmail.com>
	<87f2f3b91002100702u7581589ua866d2492a008b23@mail.gmail.com>
	<fe67a3341002110211w60390e1cm2d12fadd1e802778@mail.gmail.com>
Message-ID: <87f2f3b91002110754laf9d01dxa94a36bc1b8c83df@mail.gmail.com>

Hehe, this is getting more and more complicated. You may want to consider
using the event socket and have your call control be done from a more 3rd
party-ish perspective. If you've got all these different scenarios it might
be better to let an external script do all the work.

http://wiki.freeswitch.org/wiki/Event_Socket

-MC

On Thu, Feb 11, 2010 at 2:11 AM, Nagalenoj H. <nagalenoj at gmail.com> wrote:

> But My scenario is,
>    After I get the call from X.
>    I answer the call in some scenarios and won't answer the call. So, this
> leg can either be answered or unanswered.
>    I originate a call to another number.
>    After getting some digits from this originated leg.
>    I do uuid_bridge of these 2 legs.
>
> I want to play some file[ringback] to leg A before bridging to B.
>
> On Wed, Feb 10, 2010 at 8:32 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>>
>>
>> On Tue, Feb 9, 2010 at 9:57 PM, Nagalenoj H. <nagalenoj at gmail.com> wrote:
>>
>>> Because, I want to get some digits before bridging the legs. I've tried
>>> group_confirm_key, but it accepts only one digit, I need multiple digits, so
>>> I can't use.
>>> I've also tried group_confirm_file, but when I do originate for multiple
>>> extensions, I want this script to work based on the answered extension.
>>>
>>> So, I've originated and processed the events to do my job.
>>>
>>> How do I play some music to A leg?
>>>
>>> I might be missing something, but couldn't you just park the call ("A
>> leg") until you connect to the other party ("B leg") and then uuid_bridge at
>> whatever point you want?
>> -MC
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Regards,
> Nagalenoj H.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/06c6dac0/attachment.html 

From msc at freeswitch.org  Thu Feb 11 08:03:24 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 11 Feb 2010 08:03:24 -0800
Subject: [Freeswitch-users] call die after 30 seconds in lua script
In-Reply-To: <622bedea1002110724p4757a5a4r42a9da7330f10498@mail.gmail.com>
References: <622bedea1002110724p4757a5a4r42a9da7330f10498@mail.gmail.com>
Message-ID: <87f2f3b91002110803u52b84246qfa529177a0e37e44@mail.gmail.com>

On Thu, Feb 11, 2010 at 7:24 AM, Eder Souza <ederwander at gmail.com> wrote:

> Hi list
>
> im testing the script welcome.lua from
> http://wiki.freeswitch.org/wiki/Lua_Welcome_IVR_Example
>
> but when do a transfer my call drop in 30 seconds just in script lua!!
>
> callig one ramal (ex: 12345) direct in x-lite the call works !!
>
> When make a session:execute("transfer","12345") in the script lua the call
> drop after 30 seconds!!
>
>
> why ??
>
>
What does the FS console debug log say right before the disconnect?
-MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/9c664943/attachment.html 

From anthony.minessale at gmail.com  Thu Feb 11 08:04:34 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 11 Feb 2010 10:04:34 -0600
Subject: [Freeswitch-users] Play music to A leg.
In-Reply-To: <87f2f3b91002110754laf9d01dxa94a36bc1b8c83df@mail.gmail.com>
References: <fe67a3341002090519r2cc6e8edm7702b8d012fc2be3@mail.gmail.com>
	<87f2f3b91002090737l4efa8en29bde9457fef7ea5@mail.gmail.com>
	<fe67a3341002092157x7484f3d2id6678510bbec66cc@mail.gmail.com>
	<87f2f3b91002100702u7581589ua866d2492a008b23@mail.gmail.com>
	<fe67a3341002110211w60390e1cm2d12fadd1e802778@mail.gmail.com>
	<87f2f3b91002110754laf9d01dxa94a36bc1b8c83df@mail.gmail.com>
Message-ID: <191c3a031002110804m4c18e7e0y317f44272b42c8b4@mail.gmail.com>

group_confirm_key in execute mode can execute a lua script instead that can
read as many digits as you want and parse the results.


On Thu, Feb 11, 2010 at 9:54 AM, Michael Collins <msc at freeswitch.org> wrote:

> Hehe, this is getting more and more complicated. You may want to consider
> using the event socket and have your call control be done from a more 3rd
> party-ish perspective. If you've got all these different scenarios it might
> be better to let an external script do all the work.
>
> http://wiki.freeswitch.org/wiki/Event_Socket
>
> -MC
>
>
> On Thu, Feb 11, 2010 at 2:11 AM, Nagalenoj H. <nagalenoj at gmail.com> wrote:
>
>> But My scenario is,
>>    After I get the call from X.
>>    I answer the call in some scenarios and won't answer the call. So, this
>> leg can either be answered or unanswered.
>>    I originate a call to another number.
>>    After getting some digits from this originated leg.
>>    I do uuid_bridge of these 2 legs.
>>
>> I want to play some file[ringback] to leg A before bridging to B.
>>
>> On Wed, Feb 10, 2010 at 8:32 PM, Michael Collins <msc at freeswitch.org>wrote:
>>
>>>
>>>
>>> On Tue, Feb 9, 2010 at 9:57 PM, Nagalenoj H. <nagalenoj at gmail.com>wrote:
>>>
>>>> Because, I want to get some digits before bridging the legs. I've tried
>>>> group_confirm_key, but it accepts only one digit, I need multiple digits, so
>>>> I can't use.
>>>> I've also tried group_confirm_file, but when I do originate for multiple
>>>> extensions, I want this script to work based on the answered extension.
>>>>
>>>> So, I've originated and processed the events to do my job.
>>>>
>>>> How do I play some music to A leg?
>>>>
>>>> I might be missing something, but couldn't you just park the call ("A
>>> leg") until you connect to the other party ("B leg") and then uuid_bridge at
>>> whatever point you want?
>>> -MC
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Regards,
>> Nagalenoj H.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/64c2458f/attachment.html 

From ederwander at gmail.com  Thu Feb 11 08:37:06 2010
From: ederwander at gmail.com (Eder Souza)
Date: Thu, 11 Feb 2010 14:37:06 -0200
Subject: [Freeswitch-users] call die after 30 seconds in lua script
In-Reply-To: <87f2f3b91002110803u52b84246qfa529177a0e37e44@mail.gmail.com>
References: <622bedea1002110724p4757a5a4r42a9da7330f10498@mail.gmail.com>
	<87f2f3b91002110803u52b84246qfa529177a0e37e44@mail.gmail.com>
Message-ID: <622bedea1002110837h43bf7ab2k756ba703663fe667@mail.gmail.com>

Hi Michael

resolved i think lol lol

2010-02-11 13:02:12.566609 [NOTICE] sofia.c:322 Hangup
sofia/internal/eder at ip [CS_EXECUTE] [NORMAL_CLEARING]
2010-02-11 13:02:12.566609 [DEBUG] switch_channel.c:1683 Send signal
sofia/internal/eder at ip [KILL]
2010-02-11 13:02:12.566609 [DEBUG] switch_core_session.c:932 Send signal
sofia/internal/eder at ip [BREAK]
2010-02-11 13:02:12.572884 [DEBUG] switch_ivr_play_say.c:1391 done playing
file
2010-02-11 13:02:12.572884 [DEBUG] switch_core_state_machine.c:490 (
sofia/internal/eder at ip) State EXECUTE going to sleep
2010-02-11 13:02:12.572884 [DEBUG] switch_core_state_machine.c:397 (
sofia/internal/eder at ip) Running State Change CS_HANGUP
2010-02-11 13:02:12.572884 [DEBUG] switch_core_state_machine.c:433 (
sofia/internal/eder at ip) State HANGUP
2010-02-11 13:02:12.572884 [DEBUG] mod_sofia.c:338 Channel
sofia/internal/eder at ip hanging up, cause: NORMAL_CLEARING
2010-02-11 13:02:12.572884 [DEBUG] switch_core_state_machine.c:46
sofia/internal/eder at ip Standard HANGUP, cause: NORMAL_CLEARING
2010-02-11 13:02:12.572884 [DEBUG] switch_core_state_machine.c:433 (
sofia/internal/eder at ip) State HANGUP going to sleep
2010-02-11 13:02:12.572884 [DEBUG] switch_core_state_machine.c:475 (
sofia/internal/eder at ip) State Change CS_HANGUP -> CS_REPORTING
2010-02-11 13:02:12.572884 [DEBUG] switch_core_session.c:932 Send signal
sofia/internal/eder at ip [BREAK]
2010-02-11 13:02:12.572884 [DEBUG] switch_core_state_machine.c:397 (
sofia/internal/eder at ip) Running State Change CS_REPORTING
2010-02-11 13:02:12.572884 [DEBUG] switch_core_state_machine.c:607 (
sofia/internal/eder at ip) State REPORTING
2010-02-11 13:02:12.572884 [DEBUG] switch_core_state_machine.c:53
sofia/internal/eder at ip Standard REPORTING, cause: NORMAL_CLEARING
2010-02-11 13:02:12.572884 [DEBUG] switch_core_state_machine.c:607 (
sofia/internal/eder at ip) State REPORTING going to sleep
2010-02-11 13:02:12.572884 [DEBUG] switch_core_state_machine.c:410 (
sofia/internal/eder at ip) State Change CS_REPORTING -> CS_DESTROY
2010-02-11 13:02:12.572884 [DEBUG] switch_core_session.c:1066 Session 55 (
sofia/internal/eder at ip) Locked, Waiting on external entities
2010-02-11 13:02:12.572884 [NOTICE] switch_core_session.c:1084 Session 55 (
sofia/internal/eder at ip) Ended
2010-02-11 13:02:12.572884 [NOTICE] switch_core_session.c:1086 Close Channel
sofia/internal/eder at ip [CS_DESTROY]
2010-02-11 13:02:12.572884 [DEBUG] switch_core_state_machine.c:559 (
sofia/internal/eder at ip) State DESTROY
2010-02-11 13:02:12.572884 [DEBUG] mod_sofia.c:255
sofia/internal/eder at ipSOFIA DESTROY
2010-02-11 13:02:12.572884 [DEBUG] switch_core_state_machine.c:60
sofia/internal/eder at ip Standard DESTROY
2010-02-11 13:02:12.572884 [DEBUG] switch_core_state_machine.c:559 (
sofia/internal/eder at ip) State DESTROY going to sleep

im note this error when i transfer my calls to hold_music (MOH) extension!!

when set "answer" the call drop after 30 seconds removing the flag "answer"
my transfer dont die see:

in my lua script i remove "session:answer();"
here

--session:answer(); --"""comented line work now""
session:setAutoHangup(false)
digito = "hua"
digito = session:playAndGetDigits(1, 1, 1, 10000, "#",
"/usr/local/freeswitch/sounds/en/us/callie/conference/8000/conf-pin.wav",
"", "\\d+");
        freeswitch.consoleLog("info", "Got dtmf: ".. digito .."\n");
     end
     if (digito == "5") then
         freeswitch.consoleLog("info", "tecla digitada: ".. digito .."\n");
         session:execute("transfer","9000");

.
.
.


in my /freeswitch/conf/dialplan/default.xml i remove "<action
application="answer"/>" see line for answer coment

<extension name="teste_eder">
      <condition field="destination_number" expression="^9000$">
          <!-- <action application="answer"/> """"comented line work
now"""-->
          <action application="playback" data="$${hold_music}"/>
      </condition>
    </extension>




Att,


Eng Eder de Souza

On Thu, Feb 11, 2010 at 2:03 PM, Michael Collins <msc at freeswitch.org> wrote:

>
>
>   On Thu, Feb 11, 2010 at 7:24 AM, Eder Souza <ederwander at gmail.com>wrote:
>
>> Hi list
>>
>> im testing the script welcome.lua from
>> http://wiki.freeswitch.org/wiki/Lua_Welcome_IVR_Example
>>
>> but when do a transfer my call drop in 30 seconds just in script lua!!
>>
>> callig one ramal (ex: 12345) direct in x-lite the call works !!
>>
>> When make a session:execute("transfer","12345") in the script lua the call
>> drop after 30 seconds!!
>>
>>
>> why ??
>>
>>
> What does the FS console debug log say right before the disconnect?
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/a0feec0c/attachment-0001.html 

From jmesquita at freeswitch.org  Thu Feb 11 08:49:58 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Thu, 11 Feb 2010 14:49:58 -0200
Subject: [Freeswitch-users] Is it necessary to call hangup....
In-Reply-To: <7F8C140A-01BD-45A9-B71A-D3036BE10B3E@freeswitch.org>
References: <29b888f81002110532o3136d376g655c4b7a6dcbf598@mail.gmail.com>
	<7F8C140A-01BD-45A9-B71A-D3036BE10B3E@freeswitch.org>
Message-ID: <ebbc2d551002110849r11f6e586p4e8d90274233ac8a@mail.gmail.com>

Just a side question that relates to your answer Brian. When we run out of
stuff to do on the dialplan, we return 404 (if the leg is SIP of course)?

JM


On Thu, Feb 11, 2010 at 11:36 AM, Brian West <brian at freeswitch.org> wrote:

> If the dialplan runs out of stuff to do it'll hang up on you anyway.
>
> /b
>
> On Feb 11, 2010, at 7:32 AM, Moiz Chinoy wrote:
>
> > Hello,
> >
> > Is it necessary to call hangup in dialplan after say playing a file
> > through playback application.
> >
> > --
> > Regards,
> > Moiz Chinoy.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/199886ae/attachment.html 

From ivdreg at gmail.com  Thu Feb 11 08:51:36 2010
From: ivdreg at gmail.com (ivdreg ivdreg)
Date: Thu, 11 Feb 2010 18:51:36 +0200
Subject: [Freeswitch-users] Help on: park_timeout variable
Message-ID: <b437a0e1002110851v274b8cbah54d920a5e59a809b@mail.gmail.com>

Hi All,

After updating to current SVN from 1.0.4 I have a problem when caller party
hangs up a call. I have 3 seconds timeout before B leg disconnects. I think
that this is caused by  code in switch_ivr_bridge.c in function static
switch_status_t audio_bridge_on_exchange_media(switch_core_session_t
*session)
......

        if (switch_channel_get_state(channel) == CS_EXCHANGE_MEDIA) {
                switch_channel_set_variable(channel, "park_timeout", "3");
                switch_channel_set_state(channel, CS_PARK);
        }
......

This happens even if I set park_after_bridge=false variable.
Is anybody has this problem ?

Thanks
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/80155a37/attachment.html 

From robert.hadley at teotech.com  Thu Feb 11 08:53:12 2010
From: robert.hadley at teotech.com (Robert Hadley)
Date: Thu, 11 Feb 2010 08:53:12 -0800
Subject: [Freeswitch-users] demo_ivr cannot find sound files viarelative
	paths
In-Reply-To: <E4BC77C3-8271-4F30-B6F8-6822E4747BC3@jerris.com>
References: <78030B9E01DD48F8B024A97035F96282@greyhawk.tonecommander.com><87f2f3b91002102057t7b039221kf3975553414880cf@mail.gmail.com>
	<E4BC77C3-8271-4F30-B6F8-6822E4747BC3@jerris.com>
Message-ID: <54726F0FE98C44398AFD34C3685CC47A@greyhawk.tonecommander.com>

That was it.  Thanks Mike.

 

This test build was trunk and default configs.  The problem is in the trunk
vars.xml this line has been removed.  I added this line back to vars.xml and
the demo_ivr started working again.

 

  <X-PRE-PROCESS cmd="set"
data="sound_prefix=$${base_dir}/sounds/en/us/callie"/>

 

The IVR source must still require sound_prefix to be set (and does not use
the new sounds_dir variable).

 

Thanks again,

Robert

 

  _____  

From: Michael Jerris [mailto:mike at jerris.com] 
Sent: Wednesday, February 10, 2010 9:34 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] demo_ivr cannot find sound files viarelative
paths

 

did you make any changes to the default configs?  what is in your vars.xml
related to sounds?  the relative paths were always relative to that dir, did
you make any changes to the sounds prefix ?

 

Mike

 

On Feb 10, 2010, at 11:57 PM, Michael Collins wrote:





If I read this log correctly it failed to find the "invalid entry" file but
it did find the phrases just fine. Can you confirm the presence of this
file:
/usr/local/freeswitch/sounds/en/us/callie/ivr/8000/ivr-that_was_an_invalid_e
ntry.wav

(It looks like this call is at 8kHz so that's where I'm assuming FS is
looking to find the sound file...)

-MC

On Wed, Feb 10, 2010 at 10:50 AM, Robert Hadley <robert.hadley at teotech.com>
wrote:



Hi,

 

It appears a recent change (possibly the new sounds_dir variable or the new
ivr_menu folder?) may have broken relative sound file paths in the IVR.  I
built a today's trunk version and installed to the default location. Using
the default conf files the demo_ivr cannot find files based on the relative
paths specified in ivr_menus/demo_ivr.xml.

 

[root at TEO-UCM-T2 conf]# cat ivr_menus/demo_ivr.xml

<include>

  <!-- demo IVR setup -->

  <!-- demo IVR, Main Menu -->

  <menu name="demo_ivr"

      greet-long="phrase:demo_ivr_main_menu"

      greet-short="phrase:demo_ivr_main_menu_short"

      invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"

      exit-sound="voicemail/vm-goodbye.wav"

 

 

2010-02-10 10:32:15.300337 [DEBUG] switch_ivr_menu.c:414 Executing IVR menu
demo_ivr

2010-02-10 10:32:15.300337 [DEBUG] switch_ivr_play_say.c:118 No language
specified - Using [en]

2010-02-10 10:32:15.300337 [DEBUG] switch_ivr_play_say.c:273 Handle
play-file:[ivr/ivr-welcome_to_freeswitch.wav] (en:en)

2010-02-10 10:32:15.300337 [DEBUG] switch_ivr_play_say.c:1158 Codec
Activated L16 at 8000hz 1 channels 20ms

2010-02-10 10:32:17.922147 [DEBUG] switch_ivr_play_say.c:1450 done playing
file

2010-02-10 10:32:18.042149 [DEBUG] switch_ivr_play_say.c:273 Handle
play-file:[ivr/ivr-this_ivr_will_let_you_test_features.wav] (en:en)

2010-02-10 10:32:18.042149 [DEBUG] switch_ivr_play_say.c:1158 Codec
Activated L16 at 8000hz 1 channels 20ms

2010-02-10 10:32:19.962158 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF 2:800

2010-02-10 10:32:19.962158 [DEBUG] switch_ivr_play_say.c:1450 done playing
file

2010-02-10 10:32:20.082156 [DEBUG] switch_ivr_menu.c:329 waiting for 3/4
digits t/o 2000

2010-02-10 10:32:20.120617 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF 2:400

2010-02-10 10:32:20.442158 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF 2:800

2010-02-10 10:32:20.682162 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF 2:800

2010-02-10 10:32:20.682162 [DEBUG] switch_ivr_menu.c:376 digits '2222'

2010-02-10 10:32:20.682162 [DEBUG] switch_ivr_menu.c:470 action regex [2222]
[/^(10[01][0-9])$/] [0]

2010-02-10 10:32:20.682162 [DEBUG] switch_ivr_menu.c:560 IVR menu 'demo_ivr'
caught invalid input '2222'

2010-02-10 10:32:20.682162 [ERR] mod_sndfile.c:194 Error Opening File
[/usr/local/freeswitch/sounds/ivr/ivr-that_was_an_invalid_entry.wav] [System
error : No such file or directory.]

2010-02-10 10:32:21.701505 [DEBUG] switch_ivr_play_say.c:118 No language
specified - Using [en]

2010-02-10 10:32:21.701505 [DEBUG] switch_ivr_play_say.c:273 Handle
play-file:[silence_stream://1000] (en:en)

2010-02-10 10:32:21.701505 [DEBUG] switch_ivr_play_say.c:1158 Codec
Activated L16 at 8000hz 1 channels 20ms

2010-02-10 10:32:22.700352 [DEBUG] switch_ivr_play_say.c:1450 done playing
file

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/b1616c63/attachment-0001.html 

From tculjaga at gmail.com  Thu Feb 11 10:13:50 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Thu, 11 Feb 2010 19:13:50 +0100
Subject: [Freeswitch-users] h323 - sip call is not working
In-Reply-To: <20100211151819.95FC711F60@mail.nstel.ru>
References: <65d96fc81002110334o1ec01bddp521528cab618acfc@mail.gmail.com>
	<20100211151819.95FC711F60@mail.nstel.ru>
Message-ID: <65d96fc81002111013r38d938b0t3ffc1a49b7ff5b92@mail.gmail.com>

On Thu, Feb 11, 2010 at 4:18 PM, Nikolay Kondratyev <kond at nstel.ru> wrote:

>  Tihomir,
>
> Thanks for help.
>
> I enabled fast start on IPO and I can hear voice now. But the ringback tone
> and voice appears to be wheezy, but I will investigate that tomorrow.
>

set framing time for your codec to 30ms in IPO, also play with  PI in
alerting... set it to 2.



> Thanks again.
>
> Nikolay.
>
>
>
>
>
>
>   ------------------------------
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Tihomir
> Culjaga
> *Sent:* Thursday, February 11, 2010 2:34 PM
>
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] h323 - sip call is not working
>
>
>
> Nikolay, you are sending slow start with tunneling=true ?!?!
>
>
> It is not gong to work :)
>
> Please can you set fast start instead?
>
>
> Your call failed because there was no mediaControll channel negotiated at
> all... actually the call had to be aborted because wrong signaling .. but
> anyhow.
>
> Please on your IPO use FastStart with h245Tunneling=true... also, same
> settings on FS side as well (exclude h245 in setup as well).
>
>
>
>
> Frame 13 (277 bytes on wire, 277 bytes captured)
> Ethernet II, Src: AlliedTe_22:9b:4a (00:00:cd:22:9b:4a), Dst:
> Vmware_67:33:a7 (00:0c:29:67:33:a7)
> Internet Protocol, Src: 172.23.14.2 (172.23.14.2), Dst: 172.23.22.49
> (172.23.22.49)
> Transmission Control Protocol, Src Port: oirtgsvc (4141), Dst Port:
> h323hostcall (1720), Seq: 1, Ack: 1, Len: 223
> TPKT, Version: 3, Length: 223
> Q.931
>     Protocol discriminator: Q.931
>     Call reference value length: 2
>     Call reference flag: Message sent from originating side
>     Call reference value: 0012
>     Message type: SETUP (0x05)
>     Bearer capability
>     Display  'Gornak Alexandr>2853'
>     Calling party number: '5840'
>     Called party number: '2853'
>     User-user
> H.225.0 CS
>     H323-UserInformation
>         h323-uu-pdu
>             h323-message-body: setup (0)
>                 setup
>             h4501SupplementaryService: 1 item
>            * 1... .... h245Tunneling: True*
>
>
>
>
>
>
>  On Thu, Feb 11, 2010 at 9:18 AM, Nikolay Kondratyev <kond at nstel.ru>
> wrote:
>
> > But I do recommenf to use SIP, since it's much better supported by FS.
> >
> > /Peter
>
> But SIP is poorly supported by IPO.
> Thanks and regards,
>
> Nikolay.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/bc426167/attachment.html 

From paul at apcl.us  Thu Feb 11 10:43:09 2010
From: paul at apcl.us (Paul Levin)
Date: Thu, 11 Feb 2010 13:43:09 -0500
Subject: [Freeswitch-users] is there a command to get a list
 of	registered accounts?
In-Reply-To: <33c87fa31002102129r5d692dc5pfb414b80b47be221@mail.gmail.com>
References: <4B73677B.6020406@apcl.us>
	<33c87fa31002102129r5d692dc5pfb414b80b47be221@mail.gmail.com>
Message-ID: <4B744FBD.1090700@apcl.us>

An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/db3b0238/attachment.html 

From troy at tlainvestments.com  Thu Feb 11 11:11:18 2010
From: troy at tlainvestments.com (Troy Anderson)
Date: Thu, 11 Feb 2010 12:11:18 -0700
Subject: [Freeswitch-users] Calls being parked on DTMF
Message-ID: <4663A209-CA71-4389-8ED9-C574CD939711@tlainvestments.com>

This is a strange one.  I make a call using an FXO analog line (mod_openzap).  During the call, I dial 200 and FS parks the call.  This is a Sangoma card using wanpipe.  Is there some kind of setting in there where it interprets DTMF?

Is there a way to see what OpenZAP is writing "ending bridge by request from write function"?

Thanks for any help on this!
-Troy

Here's where it happens (phone number is redacted).

2010-02-11 11:49:48.455727 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF 2:1680
2010-02-11 11:49:48.466015 [DEBUG] zap_io.c:1870 1:1 GENERATE DTMF [2]
2010-02-11 11:49:48.895733 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF 0:1280
2010-02-11 11:49:48.905726 [DEBUG] zap_io.c:1870 1:1 GENERATE DTMF [0]
2010-02-11 11:49:49.455734 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF 0:1360
2010-02-11 11:49:49.465771 [DEBUG] zap_io.c:1870 1:1 GENERATE DTMF [0]
2010-02-11 11:49:49.706363 [DEBUG] switch_ivr_bridge.c:469 OpenZAP/1:1/602xxxxxxx ending bridge by request from write function
2010-02-11 11:49:49.706363 [DEBUG] switch_core_session.c:638 Send signal sofia/internal/105 at 10.0.1.202 [BREAK]
2010-02-11 11:49:49.706363 [DEBUG] switch_ivr_bridge.c:562 BRIDGE THREAD DONE [sofia/internal/105 at 10.0.1.202]
2010-02-11 11:49:49.706363 [DEBUG] switch_ivr_bridge.c:582 Send signal OpenZAP/1:1/602xxxxxxx [BREAK]
2010-02-11 11:49:49.716359 [DEBUG] switch_core_session.c:638 Send signal OpenZAP/1:1/602xxxxxxx [BREAK]
2010-02-11 11:49:49.716359 [DEBUG] switch_ivr_bridge.c:562 BRIDGE THREAD DONE [OpenZAP/1:1/602xxxxxxx]
2010-02-11 11:49:49.716359 [DEBUG] switch_ivr_bridge.c:582 Send signal sofia/internal/105 at 10.0.1.202 [BREAK]
2010-02-11 11:49:49.716359 [DEBUG] switch_ivr_bridge.c:640 (OpenZAP/1:1/602xxxxxxx) State Change CS_EXCHANGE_MEDIA -> CS_PARK
2010-02-11 11:49:49.716359 [DEBUG] switch_core_session.c:1018 Send signal OpenZAP/1:1/602xxxxxxx [BREAK]
2010-02-11 11:49:49.716359 [DEBUG] switch_core_state_machine.c:351 (OpenZAP/1:1/602xxxxxxx) State EXCHANGE_MEDIA going to sleep
2010-02-11 11:49:49.716359 [DEBUG] switch_core_state_machine.c:314 (OpenZAP/1:1/602xxxxxxx) Running State Change CS_PARK
2010-02-11 11:49:49.716359 [DEBUG] switch_core_state_machine.c:357 (OpenZAP/1:1/602xxxxxxx) State PARK
2010-02-11 11:49:49.716359 [DEBUG] switch_core_state_machine.c:206 OpenZAP/1:1/602xxxxxxx Standard PARK





From anthony.minessale at gmail.com  Thu Feb 11 11:42:46 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 11 Feb 2010 13:42:46 -0600
Subject: [Freeswitch-users] Help on: park_timeout variable
In-Reply-To: <b437a0e1002110851v274b8cbah54d920a5e59a809b@mail.gmail.com>
References: <b437a0e1002110851v274b8cbah54d920a5e59a809b@mail.gmail.com>
Message-ID: <191c3a031002111142n625ca0d4n641378242cd69b87@mail.gmail.com>

This is what happens in a b leg, it only happens when you transfer a call.
This is by design to give the other phone a chance to kill the leg.  This is
not really a problem persae.


On Thu, Feb 11, 2010 at 10:51 AM, ivdreg ivdreg <ivdreg at gmail.com> wrote:

> Hi All,
>
> After updating to current SVN from 1.0.4 I have a problem when caller party
> hangs up a call. I have 3 seconds timeout before B leg disconnects. I think
> that this is caused by  code in switch_ivr_bridge.c in function static
> switch_status_t audio_bridge_on_exchange_media(switch_core_session_t
> *session)
> ......
>
>         if (switch_channel_get_state(channel) == CS_EXCHANGE_MEDIA) {
>                 switch_channel_set_variable(channel, "park_timeout", "3");
>                 switch_channel_set_state(channel, CS_PARK);
>         }
> ......
>
> This happens even if I set park_after_bridge=false variable.
> Is anybody has this problem ?
>
> Thanks
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/f3731dff/attachment.html 

From anthony.minessale at gmail.com  Thu Feb 11 11:52:49 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 11 Feb 2010 13:52:49 -0600
Subject: [Freeswitch-users] Calls being parked on DTMF
In-Reply-To: <4663A209-CA71-4389-8ED9-C574CD939711@tlainvestments.com>
References: <4663A209-CA71-4389-8ED9-C574CD939711@tlainvestments.com>
Message-ID: <191c3a031002111152r1a56d46dl905d9c9c63e598d9@mail.gmail.com>

try lastest


On Thu, Feb 11, 2010 at 1:11 PM, Troy Anderson <troy at tlainvestments.com>wrote:

> This is a strange one.  I make a call using an FXO analog line
> (mod_openzap).  During the call, I dial 200 and FS parks the call.  This is
> a Sangoma card using wanpipe.  Is there some kind of setting in there where
> it interprets DTMF?
>
> Is there a way to see what OpenZAP is writing "ending bridge by request
> from write function"?
>
> Thanks for any help on this!
> -Troy
>
> Here's where it happens (phone number is redacted).
>
> 2010-02-11 11:49:48.455727 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF 2:1680
> 2010-02-11 11:49:48.466015 [DEBUG] zap_io.c:1870 1:1 GENERATE DTMF [2]
> 2010-02-11 11:49:48.895733 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF 0:1280
> 2010-02-11 11:49:48.905726 [DEBUG] zap_io.c:1870 1:1 GENERATE DTMF [0]
> 2010-02-11 11:49:49.455734 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF 0:1360
> 2010-02-11 11:49:49.465771 [DEBUG] zap_io.c:1870 1:1 GENERATE DTMF [0]
> 2010-02-11 11:49:49.706363 [DEBUG] switch_ivr_bridge.c:469
> OpenZAP/1:1/602xxxxxxx ending bridge by request from write function
> 2010-02-11 11:49:49.706363 [DEBUG] switch_core_session.c:638 Send signal
> sofia/internal/105 at 10.0.1.202 [BREAK]
> 2010-02-11 11:49:49.706363 [DEBUG] switch_ivr_bridge.c:562 BRIDGE THREAD
> DONE [sofia/internal/105 at 10.0.1.202]
> 2010-02-11 11:49:49.706363 [DEBUG] switch_ivr_bridge.c:582 Send signal
> OpenZAP/1:1/602xxxxxxx [BREAK]
> 2010-02-11 11:49:49.716359 [DEBUG] switch_core_session.c:638 Send signal
> OpenZAP/1:1/602xxxxxxx [BREAK]
> 2010-02-11 11:49:49.716359 [DEBUG] switch_ivr_bridge.c:562 BRIDGE THREAD
> DONE [OpenZAP/1:1/602xxxxxxx]
> 2010-02-11 11:49:49.716359 [DEBUG] switch_ivr_bridge.c:582 Send signal
> sofia/internal/105 at 10.0.1.202 [BREAK]
> 2010-02-11 11:49:49.716359 [DEBUG] switch_ivr_bridge.c:640
> (OpenZAP/1:1/602xxxxxxx) State Change CS_EXCHANGE_MEDIA -> CS_PARK
> 2010-02-11 11:49:49.716359 [DEBUG] switch_core_session.c:1018 Send signal
> OpenZAP/1:1/602xxxxxxx [BREAK]
> 2010-02-11 11:49:49.716359 [DEBUG] switch_core_state_machine.c:351
> (OpenZAP/1:1/602xxxxxxx) State EXCHANGE_MEDIA going to sleep
> 2010-02-11 11:49:49.716359 [DEBUG] switch_core_state_machine.c:314
> (OpenZAP/1:1/602xxxxxxx) Running State Change CS_PARK
> 2010-02-11 11:49:49.716359 [DEBUG] switch_core_state_machine.c:357
> (OpenZAP/1:1/602xxxxxxx) State PARK
> 2010-02-11 11:49:49.716359 [DEBUG] switch_core_state_machine.c:206
> OpenZAP/1:1/602xxxxxxx Standard PARK
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/ebafb4b4/attachment-0001.html 

From anthony.minessale at gmail.com  Thu Feb 11 11:53:06 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 11 Feb 2010 13:53:06 -0600
Subject: [Freeswitch-users] Calls being parked on DTMF
In-Reply-To: <191c3a031002111152r1a56d46dl905d9c9c63e598d9@mail.gmail.com>
References: <4663A209-CA71-4389-8ED9-C574CD939711@tlainvestments.com>
	<191c3a031002111152r1a56d46dl905d9c9c63e598d9@mail.gmail.com>
Message-ID: <191c3a031002111153h2f98f657n9fd4e71f47846710@mail.gmail.com>

err latest

On Thu, Feb 11, 2010 at 1:52 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> try lastest
>
>
>
> On Thu, Feb 11, 2010 at 1:11 PM, Troy Anderson <troy at tlainvestments.com>wrote:
>
>> This is a strange one.  I make a call using an FXO analog line
>> (mod_openzap).  During the call, I dial 200 and FS parks the call.  This is
>> a Sangoma card using wanpipe.  Is there some kind of setting in there where
>> it interprets DTMF?
>>
>> Is there a way to see what OpenZAP is writing "ending bridge by request
>> from write function"?
>>
>> Thanks for any help on this!
>> -Troy
>>
>> Here's where it happens (phone number is redacted).
>>
>> 2010-02-11 11:49:48.455727 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF 2:1680
>> 2010-02-11 11:49:48.466015 [DEBUG] zap_io.c:1870 1:1 GENERATE DTMF [2]
>> 2010-02-11 11:49:48.895733 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF 0:1280
>> 2010-02-11 11:49:48.905726 [DEBUG] zap_io.c:1870 1:1 GENERATE DTMF [0]
>> 2010-02-11 11:49:49.455734 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF 0:1360
>> 2010-02-11 11:49:49.465771 [DEBUG] zap_io.c:1870 1:1 GENERATE DTMF [0]
>> 2010-02-11 11:49:49.706363 [DEBUG] switch_ivr_bridge.c:469
>> OpenZAP/1:1/602xxxxxxx ending bridge by request from write function
>> 2010-02-11 11:49:49.706363 [DEBUG] switch_core_session.c:638 Send signal
>> sofia/internal/105 at 10.0.1.202 [BREAK]
>> 2010-02-11 11:49:49.706363 [DEBUG] switch_ivr_bridge.c:562 BRIDGE THREAD
>> DONE [sofia/internal/105 at 10.0.1.202]
>> 2010-02-11 11:49:49.706363 [DEBUG] switch_ivr_bridge.c:582 Send signal
>> OpenZAP/1:1/602xxxxxxx [BREAK]
>> 2010-02-11 11:49:49.716359 [DEBUG] switch_core_session.c:638 Send signal
>> OpenZAP/1:1/602xxxxxxx [BREAK]
>> 2010-02-11 11:49:49.716359 [DEBUG] switch_ivr_bridge.c:562 BRIDGE THREAD
>> DONE [OpenZAP/1:1/602xxxxxxx]
>> 2010-02-11 11:49:49.716359 [DEBUG] switch_ivr_bridge.c:582 Send signal
>> sofia/internal/105 at 10.0.1.202 [BREAK]
>> 2010-02-11 11:49:49.716359 [DEBUG] switch_ivr_bridge.c:640
>> (OpenZAP/1:1/602xxxxxxx) State Change CS_EXCHANGE_MEDIA -> CS_PARK
>> 2010-02-11 11:49:49.716359 [DEBUG] switch_core_session.c:1018 Send signal
>> OpenZAP/1:1/602xxxxxxx [BREAK]
>> 2010-02-11 11:49:49.716359 [DEBUG] switch_core_state_machine.c:351
>> (OpenZAP/1:1/602xxxxxxx) State EXCHANGE_MEDIA going to sleep
>> 2010-02-11 11:49:49.716359 [DEBUG] switch_core_state_machine.c:314
>> (OpenZAP/1:1/602xxxxxxx) Running State Change CS_PARK
>> 2010-02-11 11:49:49.716359 [DEBUG] switch_core_state_machine.c:357
>> (OpenZAP/1:1/602xxxxxxx) State PARK
>> 2010-02-11 11:49:49.716359 [DEBUG] switch_core_state_machine.c:206
>> OpenZAP/1:1/602xxxxxxx Standard PARK
>>
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/4e7f6f0a/attachment.html 

From anthony.minessale at gmail.com  Thu Feb 11 11:53:42 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 11 Feb 2010 13:53:42 -0600
Subject: [Freeswitch-users] Help on: park_timeout variable
In-Reply-To: <191c3a031002111142n625ca0d4n641378242cd69b87@mail.gmail.com>
References: <b437a0e1002110851v274b8cbah54d920a5e59a809b@mail.gmail.com>
	<191c3a031002111142n625ca0d4n641378242cd69b87@mail.gmail.com>
Message-ID: <191c3a031002111153n655103b4ub46d67f9094d6f57@mail.gmail.com>

actually, i see a small buglet there, try trunk.


On Thu, Feb 11, 2010 at 1:42 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> This is what happens in a b leg, it only happens when you transfer a call.
> This is by design to give the other phone a chance to kill the leg.  This is
> not really a problem persae.
>
>
> On Thu, Feb 11, 2010 at 10:51 AM, ivdreg ivdreg <ivdreg at gmail.com> wrote:
>
>> Hi All,
>>
>> After updating to current SVN from 1.0.4 I have a problem when caller
>> party hangs up a call. I have 3 seconds timeout before B leg disconnects. I
>> think that this is caused by  code in switch_ivr_bridge.c in function static
>> switch_status_t audio_bridge_on_exchange_media(switch_core_session_t
>> *session)
>> ......
>>
>>         if (switch_channel_get_state(channel) == CS_EXCHANGE_MEDIA) {
>>                 switch_channel_set_variable(channel, "park_timeout", "3");
>>                 switch_channel_set_state(channel, CS_PARK);
>>         }
>> ......
>>
>> This happens even if I set park_after_bridge=false variable.
>> Is anybody has this problem ?
>>
>> Thanks
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/5dd3c584/attachment.html 

From Prometheus001 at gmx.net  Thu Feb 11 11:53:34 2010
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Thu, 11 Feb 2010 20:53:34 +0100
Subject: [Freeswitch-users] Skypiax latency
In-Reply-To: <2a73afe1002110408r7f72cabdwf6a02ba27a314065@mail.gmail.com>
References: <4B72FE87.4000401@gmx.net>
	<2a73afe1002110408r7f72cabdwf6a02ba27a314065@mail.gmail.com>
Message-ID: <4B74603E.3060208@gmx.net>

Hello Hristo,

I am using xm-curl, so xml is processed. But your hints really helped me
to tie the problem down.

As described, my call scenario is as follows:
SIP-UA => FreeswitchServer=>Freeswitch/SkypIaxServer=>Skype

When I call from Skype to SIP and

    * let the SIP phone ringing for 1 sec, I receive a ~1,5 sec delay
      from SIP to Skype
    * let the SIP phone ringing for 2 sec, I receive a ~2,5 sec delay
      from SIP to Skype
    * let the SIP phone ringing for 3 sec, I receive a ~3,5 sec delay
      from SIP to Skype

So - the longer the SIP phone rings - the longer is the delay.

I also tested the other direction:
If I call from SIP to Skype the behaviour is exactly the same: The
longer the Skype phone rings - the longer is the delay from SIP to Skype.

This may be somehow related to MODSKYPIAX-29, but is a different
scenario. Should I open a new Jira for this or should I attach my
findings to MODSKYPIAX-29?

Best regards
Peter



Hristo Trendev schrieb:
> Hi Peter,
> Take a look at http://jira.freeswitch.org/browse/MODSKYPIAX-29 and
> http://jira.freeswitch.org/browse/MODLANG-130. I haven't tested the
> trick in MODLANG-130 with skypiax yet, but if you use script to handle
> the calls this may help.
>
> BR,
> Hristo
>
>
> On 2/10/10, Peter P GMX <Prometheus001 at gmx.net> wrote:
>   
>> Hello,
>>
>> I have a problem with latency and mod_skypiax
>>
>> Skype=>SIP is always fine (~0.3sec)
>> SIP => Skype is always bad (~2-4 sec)
>> I would expect that latency in both directions should be the same.
>>
>> Anybody has discovered this before and has a solution?
>>
>> The scenario is as follows:
>>
>> SIP-UA => FreeswitchServer=>Freeswitch/SkypIaxServer=>Skype
>> Both freeswitch servers are in the same LAN, so latency should be low.
>>
>> Best regards
>> Peter
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>     
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>   


From gmaruzz at celliax.org  Thu Feb 11 11:58:40 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Thu, 11 Feb 2010 20:58:40 +0100
Subject: [Freeswitch-users] Skypiax latency
In-Reply-To: <4B74603E.3060208@gmx.net>
References: <4B72FE87.4000401@gmx.net>
	<2a73afe1002110408r7f72cabdwf6a02ba27a314065@mail.gmail.com> 
	<4B74603E.3060208@gmx.net>
Message-ID: <7b197bef1002111158w6eeda53as77451e93ff1f6e04@mail.gmail.com>

pick that phone!

just jokin

open a new Jira

-giovanni

On Thu, Feb 11, 2010 at 8:53 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
> Hello Hristo,
>
> I am using xm-curl, so xml is processed. But your hints really helped me
> to tie the problem down.
>
> As described, my call scenario is as follows:
> SIP-UA => FreeswitchServer=>Freeswitch/SkypIaxServer=>Skype
>
> When I call from Skype to SIP and
>
> ? ?* let the SIP phone ringing for 1 sec, I receive a ~1,5 sec delay
> ? ? ?from SIP to Skype
> ? ?* let the SIP phone ringing for 2 sec, I receive a ~2,5 sec delay
> ? ? ?from SIP to Skype
> ? ?* let the SIP phone ringing for 3 sec, I receive a ~3,5 sec delay
> ? ? ?from SIP to Skype
>
> So - the longer the SIP phone rings - the longer is the delay.
>
> I also tested the other direction:
> If I call from SIP to Skype the behaviour is exactly the same: The
> longer the Skype phone rings - the longer is the delay from SIP to Skype.
>
> This may be somehow related to MODSKYPIAX-29, but is a different
> scenario. Should I open a new Jira for this or should I attach my
> findings to MODSKYPIAX-29?
>
> Best regards
> Peter
>
>
>
> Hristo Trendev schrieb:
>> Hi Peter,
>> Take a look at http://jira.freeswitch.org/browse/MODSKYPIAX-29 and
>> http://jira.freeswitch.org/browse/MODLANG-130. I haven't tested the
>> trick in MODLANG-130 with skypiax yet, but if you use script to handle
>> the calls this may help.
>>
>> BR,
>> Hristo
>>
>>
>> On 2/10/10, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>
>>> Hello,
>>>
>>> I have a problem with latency and mod_skypiax
>>>
>>> Skype=>SIP is always fine (~0.3sec)
>>> SIP => Skype is always bad (~2-4 sec)
>>> I would expect that latency in both directions should be the same.
>>>
>>> Anybody has discovered this before and has a solution?
>>>
>>> The scenario is as follows:
>>>
>>> SIP-UA => FreeswitchServer=>Freeswitch/SkypIaxServer=>Skype
>>> Both freeswitch servers are in the same LAN, so latency should be low.
>>>
>>> Best regards
>>> Peter
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From maxim.tsvetov at gmail.com  Thu Feb 11 12:09:09 2010
From: maxim.tsvetov at gmail.com (Maxim Tsvetov)
Date: Thu, 11 Feb 2010 23:09:09 +0300
Subject: [Freeswitch-users] DTMF problem
Message-ID: <89c9bbf81002111209v6e2f36b3x86a628c5e1390599@mail.gmail.com>

Hello everybody

Please help!
I'm trying to setup connection between Cisco 2811 and FS (Win2003) using
SIP.
Everything working correctly and I can make calls both ways. The only
problem is
when I'm calling from PSTN to FS over Cisco. It doesn't recognize DTMF.

I use G711 a-law (PCMA) codec and inband DTMF.

Regards,
Maxim
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/f37ad059/attachment.html 

From brian at freeswitch.org  Thu Feb 11 12:22:35 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 11 Feb 2010 14:22:35 -0600
Subject: [Freeswitch-users] DTMF problem
In-Reply-To: <89c9bbf81002111209v6e2f36b3x86a628c5e1390599@mail.gmail.com>
References: <89c9bbf81002111209v6e2f36b3x86a628c5e1390599@mail.gmail.com>
Message-ID: <F960CC45-C9D2-4150-A108-C0E8B230284D@freeswitch.org>

Add 'dtmf-relay rtp-nte' and 'dtmf-interworkingrtp-nte' to your voice peer.

/b

On Feb 11, 2010, at 2:09 PM, Maxim Tsvetov wrote:

> Hello everybody
> 
> Please help!
> I'm trying to setup connection between Cisco 2811 and FS (Win2003) using SIP.
> Everything working correctly and I can make calls both ways. The only problem is
> when I'm calling from PSTN to FS over Cisco. It doesn't recognize DTMF.
> 
> I use G711 a-law (PCMA) codec and inband DTMF.
> 
> Regards,
> Maxim
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From troy at tlainvestments.com  Thu Feb 11 12:35:09 2010
From: troy at tlainvestments.com (Troy Anderson)
Date: Thu, 11 Feb 2010 13:35:09 -0700
Subject: [Freeswitch-users] Calls being parked on DTMF
In-Reply-To: <191c3a031002111153h2f98f657n9fd4e71f47846710@mail.gmail.com>
References: <4663A209-CA71-4389-8ED9-C574CD939711@tlainvestments.com>
	<191c3a031002111152r1a56d46dl905d9c9c63e598d9@mail.gmail.com>
	<191c3a031002111153h2f98f657n9fd4e71f47846710@mail.gmail.com>
Message-ID: <D97A70D0-9196-41CB-9430-C6EAA2213D13@tlainvestments.com>

The test was from 16605 (from this morning). I just tried it with latest (16608) and the issue persists.

Thanks! 

-Troy

 
Troy Anderson
President, Lead Software Architect
e troy at chronostelecom.com
o 480.522.2115
c 602.327.1729

On Feb 11, 2010, at 12:53 PM, Anthony Minessale wrote:

> err latest
> 
> On Thu, Feb 11, 2010 at 1:52 PM, Anthony Minessale <anthony.minessale at gmail.com> wrote:
> try lastest
> 
> 
> 
> On Thu, Feb 11, 2010 at 1:11 PM, Troy Anderson <troy at tlainvestments.com> wrote:
> This is a strange one.  I make a call using an FXO analog line (mod_openzap).  During the call, I dial 200 and FS parks the call.  This is a Sangoma card using wanpipe.  Is there some kind of setting in there where it interprets DTMF?
> 
> Is there a way to see what OpenZAP is writing "ending bridge by request from write function"?
> 
> Thanks for any help on this!
> -Troy
> 
> Here's where it happens (phone number is redacted).
> 
> 2010-02-11 11:49:48.455727 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF 2:1680
> 2010-02-11 11:49:48.466015 [DEBUG] zap_io.c:1870 1:1 GENERATE DTMF [2]
> 2010-02-11 11:49:48.895733 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF 0:1280
> 2010-02-11 11:49:48.905726 [DEBUG] zap_io.c:1870 1:1 GENERATE DTMF [0]
> 2010-02-11 11:49:49.455734 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF 0:1360
> 2010-02-11 11:49:49.465771 [DEBUG] zap_io.c:1870 1:1 GENERATE DTMF [0]
> 2010-02-11 11:49:49.706363 [DEBUG] switch_ivr_bridge.c:469 OpenZAP/1:1/602xxxxxxx ending bridge by request from write function
> 2010-02-11 11:49:49.706363 [DEBUG] switch_core_session.c:638 Send signal sofia/internal/105 at 10.0.1.202 [BREAK]
> 2010-02-11 11:49:49.706363 [DEBUG] switch_ivr_bridge.c:562 BRIDGE THREAD DONE [sofia/internal/105 at 10.0.1.202]
> 2010-02-11 11:49:49.706363 [DEBUG] switch_ivr_bridge.c:582 Send signal OpenZAP/1:1/602xxxxxxx [BREAK]
> 2010-02-11 11:49:49.716359 [DEBUG] switch_core_session.c:638 Send signal OpenZAP/1:1/602xxxxxxx [BREAK]
> 2010-02-11 11:49:49.716359 [DEBUG] switch_ivr_bridge.c:562 BRIDGE THREAD DONE [OpenZAP/1:1/602xxxxxxx]
> 2010-02-11 11:49:49.716359 [DEBUG] switch_ivr_bridge.c:582 Send signal sofia/internal/105 at 10.0.1.202 [BREAK]
> 2010-02-11 11:49:49.716359 [DEBUG] switch_ivr_bridge.c:640 (OpenZAP/1:1/602xxxxxxx) State Change CS_EXCHANGE_MEDIA -> CS_PARK
> 2010-02-11 11:49:49.716359 [DEBUG] switch_core_session.c:1018 Send signal OpenZAP/1:1/602xxxxxxx [BREAK]
> 2010-02-11 11:49:49.716359 [DEBUG] switch_core_state_machine.c:351 (OpenZAP/1:1/602xxxxxxx) State EXCHANGE_MEDIA going to sleep
> 2010-02-11 11:49:49.716359 [DEBUG] switch_core_state_machine.c:314 (OpenZAP/1:1/602xxxxxxx) Running State Change CS_PARK
> 2010-02-11 11:49:49.716359 [DEBUG] switch_core_state_machine.c:357 (OpenZAP/1:1/602xxxxxxx) State PARK
> 2010-02-11 11:49:49.716359 [DEBUG] switch_core_state_machine.c:206 OpenZAP/1:1/602xxxxxxx Standard PARK
> 
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/4d60cfcb/attachment.html 

From anthony.minessale at gmail.com  Thu Feb 11 12:40:14 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 11 Feb 2010 14:40:14 -0600
Subject: [Freeswitch-users] Is it necessary to call hangup....
In-Reply-To: <ebbc2d551002110849r11f6e586p4e8d90274233ac8a@mail.gmail.com>
References: <29b888f81002110532o3136d376g655c4b7a6dcbf598@mail.gmail.com>
	<7F8C140A-01BD-45A9-B71A-D3036BE10B3E@freeswitch.org>
	<ebbc2d551002110849r11f6e586p4e8d90274233ac8a@mail.gmail.com>
Message-ID: <191c3a031002111240o55a7c36ib25531198e9845a8@mail.gmail.com>

Only returns 404 if it did nothing in dp

On Feb 11, 2010 10:57 AM, "Jo?o Mesquita" <jmesquita at freeswitch.org> wrote:

Just a side question that relates to your answer Brian. When we run out of
stuff to do on the dialplan, we return 404 (if the leg is SIP of course)?

JM




On Thu, Feb 11, 2010 at 11:36 AM, Brian West <brian at freeswitch.org> wrote:
>
> If the dialplan ru...

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/9bcb2c1c/attachment.html 

From mike at jerris.com  Thu Feb 11 12:42:25 2010
From: mike at jerris.com (Michael Jerris)
Date: Thu, 11 Feb 2010 14:42:25 -0600
Subject: [Freeswitch-users] demo_ivr cannot find sound files viarelative
	paths
In-Reply-To: <54726F0FE98C44398AFD34C3685CC47A@greyhawk.tonecommander.com>
References: <78030B9E01DD48F8B024A97035F96282@greyhawk.tonecommander.com><87f2f3b91002102057t7b039221kf3975553414880cf@mail.gmail.com>
	<E4BC77C3-8271-4F30-B6F8-6822E4747BC3@jerris.com>
	<54726F0FE98C44398AFD34C3685CC47A@greyhawk.tonecommander.com>
Message-ID: <438A7FC8-D68A-4D5D-AB1B-404F4C9A8AA9@jerris.com>

Just looked back at this and I am completely wrong.  Sounds prefix  
used to be set correctly in vars.xml and I remove that and set it to  
just the sounds dir in the core.  This is incorrect.  I think we will  
make the core set a default to the sounds dir and allow it to be  
overridden in vars.xml and restore that line in the default config.   
Expect a patch on this shortly.

Mike

On 2010-02-11, at 10:53 AM, "Robertre Hadley"  
<robert.hadley at teotech.com> wrote:

> That was it.  Thanks Mike.
>
> This test build was trunk and default configs.  The problem is in  
> the trunk vars.xml this line has been removed.  I added this line  
> back to vars.xml and the demo_ivr started working again.
>
>   <X-PRE-PROCESS cmd="set" data="sound_prefix=$${base_dir}/sounds/en/ 
> us/callie"/>
>
> The IVR source must still require sound_prefix to be set (and does  
> not use the new sounds_dir variable).
>
> Thanks again,
> Robert
>
> From: Michael Jerris [mailto:mike at jerris.com]
> Sent: Wednesday, February 10, 2010 9:34 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] demo_ivr cannot find sound files  
> viarelative paths
>
> did you make any changes to the default configs?  what is in your  
> vars.xml related to sounds?  the relative paths were always relative  
> to that dir, did you make any changes to the sounds prefix ?
>
> Mike
>
> On Feb 10, 2010, at 11:57 PM, Michael Collins wrote:
>
>
> If I read this log correctly it failed to find the "invalid entry"  
> file but it did find the phrases just fine. Can you confirm the  
> presence of this file:
> /usr/local/freeswitch/sounds/en/us/callie/ivr/8000/ivr- 
> that_was_an_invalid_entry.wav
>
> (It looks like this call is at 8kHz so that's where I'm assuming FS  
> is looking to find the sound file...)
>
> -MC
>
> On Wed, Feb 10, 2010 at 10:50 AM, Robert  Hadley <robert.hadley at teotech.com 
> > wrote:
>
> Hi,
>
> It appears a recent change (possibly the new sounds_dir variable or  
> the new ivr_menu folder?) may have broken relative sound file paths  
> in the IVR.  I built a today?s trunk version and installed to the de 
> fault location. Using the default conf files the demo_ivr cannot fin 
> d files based on the relative paths specified in ivr_menus/demo_ivr. 
> xml.
>
> [root at TEO-UCM-T2 conf]# cat ivr_menus/demo_ivr.xml
> <include>
>   <!-- demo IVR setup -->
>   <!-- demo IVR, Main Menu -->
>   <menu name="demo_ivr"
>       greet-long="phrase:demo_ivr_main_menu"
>       greet-short="phrase:demo_ivr_main_menu_short"
>       invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"
>       exit-sound="voicemail/vm-goodbye.wav"
>
>
> 2010-02-10 10:32:15.300337 [DEBUG] switch_ivr_menu.c:414 Executing  
> IVR menu demo_ivr
> 2010-02-10 10:32:15.300337 [DEBUG] switch_ivr_play_say.c:118 No  
> language specified - Using [en]
> 2010-02-10 10:32:15.300337 [DEBUG] switch_ivr_play_say.c:273 Handle  
> play-file:[ivr/ivr-welcome_to_freeswitch.wav] (en:en)
> 2010-02-10 10:32:15.300337 [DEBUG] switch_ivr_play_say.c:1158 Codec  
> Activated L16 at 8000hz 1 channels 20ms
> 2010-02-10 10:32:17.922147 [DEBUG] switch_ivr_play_say.c:1450 done  
> playing file
> 2010-02-10 10:32:18.042149 [DEBUG] switch_ivr_play_say.c:273 Handle  
> play-file:[ivr/ivr-this_ivr_will_let_you_test_features.wav] (en:en)
> 2010-02-10 10:32:18.042149 [DEBUG] switch_ivr_play_say.c:1158 Codec  
> Activated L16 at 8000hz 1 channels 20ms
> 2010-02-10 10:32:19.962158 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF  
> 2:800
> 2010-02-10 10:32:19.962158 [DEBUG] switch_ivr_play_say.c:1450 done  
> playing file
> 2010-02-10 10:32:20.082156 [DEBUG] switch_ivr_menu.c:329 waiting for  
> 3/4 digits t/o 2000
> 2010-02-10 10:32:20.120617 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF  
> 2:400
> 2010-02-10 10:32:20.442158 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF  
> 2:800
> 2010-02-10 10:32:20.682162 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF  
> 2:800
> 2010-02-10 10:32:20.682162 [DEBUG] switch_ivr_menu.c:376 digits '2222'
> 2010-02-10 10:32:20.682162 [DEBUG] switch_ivr_menu.c:470 action  
> regex [2222] [/^(10[01][0-9])$/] [0]
> 2010-02-10 10:32:20.682162 [DEBUG] switch_ivr_menu.c:560 IVR menu  
> 'demo_ivr' caught invalid input '2222'
> 2010-02-10 10:32:20.682162 [ERR] mod_sndfile.c:194 Error Opening  
> File [/usr/local/freeswitch/sounds/ivr/ivr- 
> that_was_an_invalid_entry.wav] [System error : No such file or  
> directory.]
> 2010-02-10 10:32:21.701505 [DEBUG] switch_ivr_play_say.c:118 No  
> language specified - Using [en]
> 2010-02-10 10:32:21.701505 [DEBUG] switch_ivr_play_say.c:273 Handle  
> play-file:[silence_stream://1000] (en:en)
> 2010-02-10 10:32:21.701505 [DEBUG] switch_ivr_play_say.c:1158 Codec  
> Activated L16 at 8000hz 1 channels 20ms
> 2010-02-10 10:32:22.700352 [DEBUG] switch_ivr_play_say.c:1450 done  
> playing file
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/8c69271b/attachment-0001.html 

From anthony.minessale at gmail.com  Thu Feb 11 12:46:46 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 11 Feb 2010 14:46:46 -0600
Subject: [Freeswitch-users] Calls being parked on DTMF
In-Reply-To: <D97A70D0-9196-41CB-9430-C6EAA2213D13@tlainvestments.com>
References: <4663A209-CA71-4389-8ED9-C574CD939711@tlainvestments.com>
	<191c3a031002111152r1a56d46dl905d9c9c63e598d9@mail.gmail.com>
	<191c3a031002111153h2f98f657n9fd4e71f47846710@mail.gmail.com>
	<D97A70D0-9196-41CB-9430-C6EAA2213D13@tlainvestments.com>
Message-ID: <191c3a031002111246h1ca4e0bo18549e150a29836e@mail.gmail.com>

I am skeptical,

Did you actually rebuild and restart FS after updating.

get a console trace with debug level please.

console loglevel debug



On Thu, Feb 11, 2010 at 2:35 PM, Troy Anderson <troy at tlainvestments.com>wrote:

> The test was from 16605 (from this morning). I just tried it with latest
> (16608) and the issue persists.
>
> Thanks!
>
> -Troy
>
>    <http://www.chronostelecom.com>
>
> Troy Anderson
>
> President, Lead Software Architect
>
> e troy at chronostelecom.com
>
> o 480.522.2115
>
> c 602.327.1729
>
> On Feb 11, 2010, at 12:53 PM, Anthony Minessale wrote:
>
> err latest
>
> On Thu, Feb 11, 2010 at 1:52 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> try lastest
>>
>>
>>
>> On Thu, Feb 11, 2010 at 1:11 PM, Troy Anderson <troy at tlainvestments.com>wrote:
>>
>>> This is a strange one.  I make a call using an FXO analog line
>>> (mod_openzap).  During the call, I dial 200 and FS parks the call.  This is
>>> a Sangoma card using wanpipe.  Is there some kind of setting in there where
>>> it interprets DTMF?
>>>
>>> Is there a way to see what OpenZAP is writing "ending bridge by request
>>> from write function"?
>>>
>>> Thanks for any help on this!
>>> -Troy
>>>
>>> Here's where it happens (phone number is redacted).
>>>
>>> 2010-02-11 11:49:48.455727 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF 2:1680
>>> 2010-02-11 11:49:48.466015 [DEBUG] zap_io.c:1870 1:1 GENERATE DTMF [2]
>>> 2010-02-11 11:49:48.895733 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF 0:1280
>>> 2010-02-11 11:49:48.905726 [DEBUG] zap_io.c:1870 1:1 GENERATE DTMF [0]
>>> 2010-02-11 11:49:49.455734 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF 0:1360
>>> 2010-02-11 11:49:49.465771 [DEBUG] zap_io.c:1870 1:1 GENERATE DTMF [0]
>>> 2010-02-11 11:49:49.706363 [DEBUG] switch_ivr_bridge.c:469
>>> OpenZAP/1:1/602xxxxxxx ending bridge by request from write function
>>> 2010-02-11 11:49:49.706363 [DEBUG] switch_core_session.c:638 Send signal
>>> sofia/internal/105 at 10.0.1.202 [BREAK]
>>> 2010-02-11 11:49:49.706363 [DEBUG] switch_ivr_bridge.c:562 BRIDGE THREAD
>>> DONE [sofia/internal/105 at 10.0.1.202]
>>> 2010-02-11 11:49:49.706363 [DEBUG] switch_ivr_bridge.c:582 Send signal
>>> OpenZAP/1:1/602xxxxxxx [BREAK]
>>> 2010-02-11 11:49:49.716359 [DEBUG] switch_core_session.c:638 Send signal
>>> OpenZAP/1:1/602xxxxxxx [BREAK]
>>> 2010-02-11 11:49:49.716359 [DEBUG] switch_ivr_bridge.c:562 BRIDGE THREAD
>>> DONE [OpenZAP/1:1/602xxxxxxx]
>>> 2010-02-11 11:49:49.716359 [DEBUG] switch_ivr_bridge.c:582 Send signal
>>> sofia/internal/105 at 10.0.1.202 [BREAK]
>>> 2010-02-11 11:49:49.716359 [DEBUG] switch_ivr_bridge.c:640
>>> (OpenZAP/1:1/602xxxxxxx) State Change CS_EXCHANGE_MEDIA -> CS_PARK
>>> 2010-02-11 11:49:49.716359 [DEBUG] switch_core_session.c:1018 Send signal
>>> OpenZAP/1:1/602xxxxxxx [BREAK]
>>> 2010-02-11 11:49:49.716359 [DEBUG] switch_core_state_machine.c:351
>>> (OpenZAP/1:1/602xxxxxxx) State EXCHANGE_MEDIA going to sleep
>>> 2010-02-11 11:49:49.716359 [DEBUG] switch_core_state_machine.c:314
>>> (OpenZAP/1:1/602xxxxxxx) Running State Change CS_PARK
>>> 2010-02-11 11:49:49.716359 [DEBUG] switch_core_state_machine.c:357
>>> (OpenZAP/1:1/602xxxxxxx) State PARK
>>> 2010-02-11 11:49:49.716359 [DEBUG] switch_core_state_machine.c:206
>>> OpenZAP/1:1/602xxxxxxx Standard PARK
>>>
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/d84a1eaf/attachment.html 

From anthony.minessale at gmail.com  Thu Feb 11 12:49:57 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 11 Feb 2010 14:49:57 -0600
Subject: [Freeswitch-users] Lua script hangup detect
In-Reply-To: <FFD1A126-5CC0-4D02-87CE-B3EBA553FC4A@freeswitch.org>
References: <314dc3f81002100117h6b1bc49cna989624a1ae4e0e2@mail.gmail.com>
	<314dc3f81002100800o41fd6561w394d5e83e69d3d80@mail.gmail.com>
	<4B72DFF9.4040402@gmail.com>
	<1b46b4e81002100932x7af486ekd4b05f4750cace20@mail.gmail.com>
	<4B72F2A8.4070503@gmail.com>
	<4CA5432B-5F6E-438E-B6EF-3277847E7922@jerris.com>
	<4B73B57E.9040901@gmail.com>
	<D8FFF1FB-7B2D-49BC-940C-445EBF838BE3@jerris.com>
	<4B73C640.20700@gmail.com>
	<FFD1A126-5CC0-4D02-87CE-B3EBA553FC4A@freeswitch.org>
Message-ID: <191c3a031002111249sd655896s5da75e61736b238d@mail.gmail.com>

The script cannot end if you are stuck in a while loop that never exits.
The same thing is true in any script in any language.


On Thu, Feb 11, 2010 at 7:26 AM, Brian West <brian at freeswitch.org> wrote:

> Yes ODBC doesn't seem to leak... while the native one leaks like a sieve.
>
> /b
>
> On Feb 11, 2010, at 2:56 AM, Nazim Agabekov wrote:
>
> > I'm using luasql with ODBC MySQL driver in production. I've never tried
> > to use luasql with "native" mysql driver, but ODBC one works great.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/242fd618/attachment.html 

From maxim.tsvetov at gmail.com  Thu Feb 11 13:12:33 2010
From: maxim.tsvetov at gmail.com (maxim.tsvetov)
Date: Thu, 11 Feb 2010 13:12:33 -0800 (PST)
Subject: [Freeswitch-users] DTMF problem
In-Reply-To: <F960CC45-C9D2-4150-A108-C0E8B230284D@freeswitch.org>
References: <89c9bbf81002111209v6e2f36b3x86a628c5e1390599@mail.gmail.com>
	<F960CC45-C9D2-4150-A108-C0E8B230284D@freeswitch.org>
Message-ID: <1265922753047-4557446.post@n2.nabble.com>


I already added "dtmf-relay rtp-nte" and this doesn't work.

Also I don't have "dtmf-interworking rtp-nte" command in Cisco.
-- 
View this message in context: http://n2.nabble.com/DTMF-problem-tp4557122p4557446.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From troy at tlainvestments.com  Thu Feb 11 13:14:48 2010
From: troy at tlainvestments.com (Troy Anderson)
Date: Thu, 11 Feb 2010 14:14:48 -0700
Subject: [Freeswitch-users] Calls being parked on DTMF
In-Reply-To: <4C603407-C72A-4D93-9451-771A8634BD54@tlainvestments.com>
References: <4663A209-CA71-4389-8ED9-C574CD939711@tlainvestments.com>
	<191c3a031002111152r1a56d46dl905d9c9c63e598d9@mail.gmail.com>
	<191c3a031002111153h2f98f657n9fd4e71f47846710@mail.gmail.com>
	<D97A70D0-9196-41CB-9430-C6EAA2213D13@tlainvestments.com>
	<191c3a031002111246h1ca4e0bo18549e150a29836e@mail.gmail.com>
	<4C603407-C72A-4D93-9451-771A8634BD54@tlainvestments.com>
Message-ID: <87de771b1002111314n4ecc3681m20cd3cbaa5f77def@mail.gmail.com>

I did a fs stop, make, make install, fs start. ?As we speak, I'm doing
a make sure, etc. And will report back.

-Troy

On Thursday, February 11, 2010, Troy Anderson <admin at tlainvestments.com> wrote:
> I did a fs stop, make, make install, fs start. ?As we speak, I'm doing a make sure, etc. And will report back.
> -Troy
> --?I'm probably driving while typing this, so pardon the typos! --
> On Feb 11, 2010, at 1:46 PM, Anthony Minessale <anthony.minessale at gmail.com> wrote:
>
> I am skeptical,
>
> Did you actually rebuild and restart FS after updating.
>
> get a console trace with debug level please.
>
> console loglevel debug
>
>
>
> On Thu, Feb 11, 2010 at 2:35 PM, Troy Anderson <troy at tlainvestments.com> wrote:
> The test was from 16605 (from this morning). I just tried it with latest (16608) and the issue persists.
>
> Thanks!
> -Troy
>
>
>
> 	
> 		??<http://www.chronostelecom.com>
> 	
> 	
> 		Troy Anderson
> 		President, Lead Software Architect
> 		e troy at chronostelecom.com
> 		o 480.522.2115
> 		c 602.327.1729
> 		
> 	
>
>
>
>
> On Feb 11, 2010, at 12:53 PM, Anthony Minessale wrote:
> err latest
>
> On Thu, Feb 11, 2010 at 1:52 PM, Anthony Minessale <anthony.minessale at gmail.com> wrote:
>
> try lastest
>
>
> On Thu, Feb 11, 2010 at 1:11 PM, Troy Anderson <troy at tlainvestments.com> wrote:
>
> This is a strange one. ?I make a call using an FXO analog line (mod_openzap). ?During the call, I dial 200 and FS parks the call. ?This is a Sangoma card using wanpipe. ?Is there some kind of setting in there where it interprets DTMF?
>
> Is there a way to see what OpenZAP is writing "ending bridge by request from write function"?
>
> Thanks for any help on this!
> -Troy
>
> Here's where it happens (phone number is redacted).
>
> 2010-02-11 11:49:48.455727 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF 2:1680
> 2010-02-11 11:49:48.466015 [DEBUG] zap_io.c:1870 1:1 GENERATE DTMF [2]
> 2010-02-11 11:49:48.895733 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF 0:1280
> 2010-02-11 11:49:48.905726 [DEBUG] zap_io.c:1870 1:1 GENERATE DTMF [0]
> 2010-02-11 11:49:49.455734 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF 0:1360
> 2010-02-11 11:49:49.465771 [DEBUG] zap_io.c:1870 1:1 GENERATE DTMF [0]
> 2010-02-11 11:49:49.706363 [DEBUG] switch_ivr_bridge.c:469 OpenZAP/1:1/602xxxxxxx ending bridge by request from write function
> 2010-02-11 11:49:49.706363 [DEBUG] switch_core_session.c:638 Send signal sofia/internal/105 at 10.0.1.202 [BREAK]
> 2010-02-11 11:49:49.706363 [DEBUG] switch_ivr_bridge.c:562 BRIDGE THREAD DONE [sofia/internal/105 at 10.0.1.202]
> 2010-02-11 11:49:49.706363 [DEBUG] switch_ivr_bridge.c:582 Send signal OpenZAP/1:1/602xxxxxxx [BREAK]
> 2010-02-11 11:49:49.716359 [DEBUG] switch_core_session.c:638 Send signal OpenZAP/1:1/602xxxxxxx [BREAK]
> 2010-02-11 11:49:49.716359 [DEBUG] switch_ivr_bridge.c:562 BRIDGE THREAD DONE [OpenZAP/1:1/602xxxxxxx]
> 2010-02-11 11:49:49.716359 [DEBUG] switch_ivr_bridge.c:582 Send signal sofia/internal/105 at 10.0.1.202 [BREAK]
> 2010-02-11 11:49:49.716359 [DEBUG] switch_ivr_bridge.c:640 (OpenZAP/1:1/602xxxxxxx) State Change CS_EXCHANGE_MEDIA -> CS_PARK
> 2010-02-11 11:49:49.716359 [DEBUG] switch_core_session.c:1018 Send signal OpenZAP/1:1/602xxxxxxx [BREAK]
> 2010-02-11 11:49:49.716359 [DEBUG] switch_core_state_machine.c:351 (OpenZ

-- 
Troy Anderson
Chronos Consulting
6501 E. Greenway Pkwy #103/422
Scottsdale, AZ 85254
(480) 922-5380 (office)
(602) 327-1729 (cell)

--


From brian at freeswitch.org  Thu Feb 11 13:19:44 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 11 Feb 2010 15:19:44 -0600
Subject: [Freeswitch-users] Calls being parked on DTMF
In-Reply-To: <87de771b1002111314n4ecc3681m20cd3cbaa5f77def@mail.gmail.com>
References: <4663A209-CA71-4389-8ED9-C574CD939711@tlainvestments.com>
	<191c3a031002111152r1a56d46dl905d9c9c63e598d9@mail.gmail.com>
	<191c3a031002111153h2f98f657n9fd4e71f47846710@mail.gmail.com>
	<D97A70D0-9196-41CB-9430-C6EAA2213D13@tlainvestments.com>
	<191c3a031002111246h1ca4e0bo18549e150a29836e@mail.gmail.com>
	<4C603407-C72A-4D93-9451-771A8634BD54@tlainvestments.com>
	<87de771b1002111314n4ecc3681m20cd3cbaa5f77def@mail.gmail.com>
Message-ID: <530A2EC6-3585-418A-B375-720E973F68A0@freeswitch.org>

"make current" is the best thing you can do.

/b

On Feb 11, 2010, at 3:14 PM, Troy Anderson wrote:

> I did a fs stop, make, make install, fs start.  As we speak, I'm doing
> a make sure, etc. And will report back.
> 
> -Troy



From mike at van.lammeren.net  Thu Feb 11 13:51:20 2010
From: mike at van.lammeren.net (Mike van Lammeren)
Date: Thu, 11 Feb 2010 16:51:20 -0500
Subject: [Freeswitch-users] Lua script hangup detect
In-Reply-To: <FFD1A126-5CC0-4D02-87CE-B3EBA553FC4A@freeswitch.org>
References: <314dc3f81002100117h6b1bc49cna989624a1ae4e0e2@mail.gmail.com>
	<314dc3f81002100800o41fd6561w394d5e83e69d3d80@mail.gmail.com>
	<4B72DFF9.4040402@gmail.com>
	<1b46b4e81002100932x7af486ekd4b05f4750cace20@mail.gmail.com>
	<4B72F2A8.4070503@gmail.com>
	<4CA5432B-5F6E-438E-B6EF-3277847E7922@jerris.com>
	<4B73B57E.9040901@gmail.com>
	<D8FFF1FB-7B2D-49BC-940C-445EBF838BE3@jerris.com>
	<4B73C640.20700@gmail.com>
	<FFD1A126-5CC0-4D02-87CE-B3EBA553FC4A@freeswitch.org>
Message-ID: <5d2828f1002111351s5f0cdff2odb6b35fa9be9eb32@mail.gmail.com>

D'oh!

I am currently working on a project that uses Lua and the native MySQL
driver, so as soon as I read this comment, I decided that I had better do a
bit of research.

I wrote a Lua test script that makes 10 queries against a MySQL database,
then ran it repeatedly.

My results show that "leaks like a sieve" is quite correct. To me, it
doesn't look like any memory is released, ever. After only 5000 queries or
so, the memory allocated for FreeSWITCH balloons from 15 Mb to 50 Mb, and
never goes back down.

Thanks, Brian, for the heads up!

Mike van Lammeren


On Thu, Feb 11, 2010 at 8:26 AM, Brian West <brian at freeswitch.org> wrote:

> Yes ODBC doesn't seem to leak... while the native one leaks like a sieve.
>
> /b
>
> On Feb 11, 2010, at 2:56 AM, Nazim Agabekov wrote:
>
> > I'm using luasql with ODBC MySQL driver in production. I've never tried
> > to use luasql with "native" mysql driver, but ODBC one works great.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/a78bb120/attachment.html 

From mike at jerris.com  Thu Feb 11 14:22:46 2010
From: mike at jerris.com (Michael Jerris)
Date: Thu, 11 Feb 2010 17:22:46 -0500
Subject: [Freeswitch-users] Forbidden using UDP, works with TCP/TLS
In-Reply-To: <33c87fa31002110213q1de0eebfsb950d8725ea9aba@mail.gmail.com>
References: <33c87fa31002101824h21e3bf7lb13b68e8d4fd36a2@mail.gmail.com>
	<2EF27EFC-1634-46A3-B438-2EE99BEF58F1@jerris.com>
	<33c87fa31002110131w3cf6ad00n802669856085a879@mail.gmail.com>
	<CA21231C-108B-4445-A3B9-BF8D94FFE898@jerris.com>
	<33c87fa31002110213q1de0eebfsb950d8725ea9aba@mail.gmail.com>
Message-ID: <0A905834-850F-4D1F-AEC5-D3A87CE4141B@jerris.com>

how is this different from the working one?

Mike

On Feb 11, 2010, at 5:13 AM, Mark Campbell-Smith wrote:

> ah thats true... The trace is not too readable to me, but may give
> some insight to someone that can read the sofia logs....
> 
> 
> recv 752 bytes from udp/[121.xxx.xxx.xxx]:5060 at 10:10:37.803288:
>   ------------------------------------------------------------------------
>   REGISTER sip:mydns.dyndns.org SIP/2.0
>   Via: SIP/2.0/UDP 121.xxx.xxx.xxx:5060;branch=z9hG4bK-9052c91f
>   From: 2000 <sip:2000 at mydns.dyndns.org>;tag=7a9dbbbfa691136do0
>   To: 2000 <sip:2000 at mydns.dyndns.org>
>   Call-ID: 610db38-dd3b511f at 121.xxx.xxx.xxx
>   CSeq: 32330 REGISTER
>   Contact: 2000 <sip:2000 at 121.xxx.xxx.xxx:5060>;expires=900
>   Authorization: Digest username="2000", realm="mydns.dyndns.org",
> nonce="b3298cfe-16f5-11df-9734-e77ba7a22ac3",
> uri="sip:mydns.dyndns.org",
> response="724fca542ce08d3f12b9ba1043bebb0c", algorithm=MD5,
> qop="1225e2f1"
>   Max-Forwards: 70
>   User-Agent: Linksys/SPA3102-5.1.10(GW)
>   Supported: x-sipura
>   Supported: replaces
>   Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
>   Content-Length: 0
> 
> 
>   ------------------------------------------------------------------------
> tport_deliver(0x97cde80): msg 0x98297e8 (752 bytes) from
> udp/121.xxx.xxx.xxx:5060/sip next=(nil)
> nta: received REGISTER sip:mydns.dyndns.org SIP/2.0 (CSeq 32330)
> nta: REGISTER (32330) going to a default leg
> nua: nua_stack_process_request: entering
> nua: nh_create: entering
> nua: nh_create_handle: entering
> nua: nua_stack_set_params: entering
> soa_clone(static::0x97cc698, 0x9794808, 0x981cb70) called
> soa_set_params(static::0x9758ba8, ...) called
> nua(0x981cb70): event i_register 100 Trying
> nua: nua_application_event: entering
> nua: nua_respond: entering
> nua(0x981cb70): sent signal r_respond
> nua: nua_handle_destroy: entering
> nua(0x981cb70): sent signal r_destroy
> nua: nua_handle_magic: entering
> nua: nua_handle_destroy: entering
> nua(0x981cb70): recv signal r_respond 403 Forbidden
> nua: nua_stack_set_params: entering
> soa_set_params(static::0x9758ba8, ...) called
> tport_tsend(0x97cde80) tpn = UDP/121.xxx.xxx.xxx:5060
> tport_resolve addrinfo = 121.xxx.xxx.xxx:5060
> tport_by_addrinfo(0x97cde80): not found by name UDP/121.xxx.xxx.xxx:5060
> tport_vsend(0x97cde80): 495 bytes of 495 to udp/121.xxx.xxx.xxx:5060
> tport_vsend returned 495
> send 495 bytes to udp/[121.xxx.xxx.xxx]:5060 at 10:10:37.812955:
>   ------------------------------------------------------------------------
>   SIP/2.0 403 Forbidden
> 
> On Thu, Feb 11, 2010 at 8:57 PM, Michael Jerris <mike at jerris.com> wrote:
>> you can crank up the sofia loglevel as well
>> 
>> Mike
>> 
>> On Feb 11, 2010, at 4:31 AM, Mark Campbell-Smith wrote:
>> 
>>> I don't see debug logs (fsctl debug_level 9 or loglevel 9) during the
>>> registration process.
>>> 
>>> All I see is the sip messages when the sip trace is activated (403 Forbidden)
>>> 
>>> Is there other debugging that I can enable?
>>> 
>>> On Thu, Feb 11, 2010 at 4:25 PM, Michael Jerris <mike at jerris.com> wrote:
>>>> Look at the FreeSWITCH debug logs and compare the differences, they should tell you why.
>>>> 
>>>> Mike



From mike at jerris.com  Thu Feb 11 14:34:30 2010
From: mike at jerris.com (Michael Jerris)
Date: Thu, 11 Feb 2010 17:34:30 -0500
Subject: [Freeswitch-users] how to set From address for voicemail emails?
In-Reply-To: <4B7401DD.6050408@apcl.us>
References: <4B736846.1040908@apcl.us>
	<7471DDD5-0A73-4F18-AF90-BA8534EBC9E7@jerris.com>
	<4B7401DD.6050408@apcl.us>
Message-ID: <E9393F2E-4EBC-410D-8E8A-A3D947DABE26@jerris.com>

what your looking for is in the template files in voicemail, they are in the conf directory with a tpl file extension

http://svn.freeswitch.org/svn/freeswitch/trunk/conf/voicemail.tpl

Mike


On Feb 11, 2010, at 8:10 AM, Paul Levin wrote:

> Mike,
>     Thank you very much for the reply.  But you are talking way over my head.  Can you please tell me very specifically what file(s) and value(s) I need to set/change in order to set the From address on voice mail emails?
>     Thanks,
>     Paul
> 
> 
> Michael Jerris wrote:
>> 
>> This is the difference between what is sent to the mail server in the mime content, and what is passed as MAIL FROM:  to the smtp server.  The latter is controlled by that param, the former is in the template.
>> 
>> Mike
>> 
>> On Feb 10, 2010, at 9:15 PM, Paul Levin wrote:
>> 
>>> I am running FreeSwitch on Windows.  I have msmtp setup and voice mail emails are being sent. 
>>> 
>>> I have msmtp configured to set a "From" address of me at mydomain.com, but when FreeSwitch sends an email with a voice mail message from Alice, the From address of the email is Alice at sipServerDomain.com.  According to the Mod voicemail document (http://wiki.freeswitch.org/wiki/Mod_voicemail) the email_email-from parameter should control this, but I tried setting it in conf\autoload_configs\directory.conf.xml (as per that document) and also in Bob.xml (the account getting the voicemail).  Neither place changed the value being used.  
>>> 
>>> How do I get this changed?
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>   
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/8701a210/attachment.html 

From paul at apcl.us  Thu Feb 11 14:42:08 2010
From: paul at apcl.us (Paul Levin)
Date: Thu, 11 Feb 2010 17:42:08 -0500
Subject: [Freeswitch-users] how to set From address for voicemail emails?
	(repost)
Message-ID: <4B7487C0.2010200@apcl.us>

An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/cf871ff2/attachment.html 

From joel.sisko at iconverged.com  Thu Feb 11 14:48:50 2010
From: joel.sisko at iconverged.com (joel.sisko at iconverged.com)
Date: Thu, 11 Feb 2010 16:48:50 -0600 (CST)
Subject: [Freeswitch-users] Mod_Conference capacity....
In-Reply-To: <268119402.962161265928492648.JavaMail.root@mail-2.01.com>
Message-ID: <715035894.962251265928530648.JavaMail.root@mail-2.01.com>




Mod_Conference 

Group, I am looking for anyone?s input (based on your own experience) on how many individual three party conference?s and the biggest single conference (total number of listeners with just 1 speaker) FreeSwitch can handle using the mod_conference application? Just looking for rough numbers if I were to use a dual processor quad core system with 12GB of RAM. 

I understand that transcoding and other factors create limits but I am just looking for same raw numbers that I should be able to obtain if the moon and stars were to align correctly. 

Thanks for the input in advance. 

Joel
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/e40795a0/attachment-0001.html 

From paul at apcl.us  Thu Feb 11 14:48:19 2010
From: paul at apcl.us (Paul Levin)
Date: Thu, 11 Feb 2010 17:48:19 -0500
Subject: [Freeswitch-users] how to set From address for voicemail emails?
In-Reply-To: <E9393F2E-4EBC-410D-8E8A-A3D947DABE26@jerris.com>
References: <4B736846.1040908@apcl.us>
	<7471DDD5-0A73-4F18-AF90-BA8534EBC9E7@jerris.com>	<4B7401DD.6050408@apcl.us>
	<E9393F2E-4EBC-410D-8E8A-A3D947DABE26@jerris.com>
Message-ID: <4B748933.7000100@apcl.us>

An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/dc8a1f02/attachment.html 

From paul at apcl.us  Thu Feb 11 14:48:55 2010
From: paul at apcl.us (Paul Levin)
Date: Thu, 11 Feb 2010 17:48:55 -0500
Subject: [Freeswitch-users] how to set From address for voicemail
 emails? (repost)
In-Reply-To: <4B7487C0.2010200@apcl.us>
References: <4B7487C0.2010200@apcl.us>
Message-ID: <4B748957.5070206@apcl.us>

An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/95b70ce3/attachment.html 

From anthony.minessale at gmail.com  Thu Feb 11 15:01:52 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 11 Feb 2010 17:01:52 -0600
Subject: [Freeswitch-users] Bypass-media and REFER method
In-Reply-To: <B816EF85C732094BA2016B2112F3DC2B040C6061@isr-brass.interwise.com>
References: <B816EF85C732094BA2016B2112F3DC2B0402C777@isr-brass.interwise.com>
	<B816EF85C732094BA2016B2112F3DC2B040C5CA6@isr-brass.interwise.com>
	<191c3a031002100645n5d827949xdefd24143f10759@mail.gmail.com>
	<B816EF85C732094BA2016B2112F3DC2B040C6061@isr-brass.interwise.com>
Message-ID: <191c3a031002111501y7e5917b8g4faae4b8752e6fc2@mail.gmail.com>

Where do you think it's failing?
that log shows it get refer, go back to dp, invite to 3341 and then the
other end hangs up at some point.


On Wed, Feb 10, 2010 at 11:01 AM, Victor Maruani <vmaruani at interwise.com>wrote:

>  Hi,
>
>
>
> Logs are on pb 12099
>
> I hope this helps.
>
> Reproduced with revision 16599.
>
>
>
> A-side (10.10.5.19) is an x-lite registered with extension 1002
>
> B (.5.51) refers to C (.5.48) none are registered.
>
>
>
> Please refer to previous emails for details of dialplan and what I try to
> do?
>
> Let me know if you need more info
>
>
>
> Thanks!
>
>
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony
> Minessale
> *Sent:* Wednesday, February 10, 2010 4:46 PM
>
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Bypass-media and REFER method
>
>
>
> update to latest trunk and reproduce your problem with full debug enabled.
>
> sofia profile internal siptrace on
> console loglevel debug
>
>  On Wed, Feb 10, 2010 at 4:44 AM, Victor Maruani <vmaruani at interwise.com>
> wrote:
>
> Hi,
>
>
>
> I can't have a blind transfer work properly if I use bypass-media=true.
>
>
>
> My first message may have been unclear, here I added excerpt from the
> dialplan:
>
>
>
>                 <!--- ext MyIVR -->
>
>                  <extension name="Extension MyIVR" continue="false">
>
>                    <condition field="destination_number"
> expression="^337$">
>
>                                    <action application="set"
> data="bypass_media=true"/>
>
>                                    <action application="bridge"
> data="sofia/internal/337 at 10.10.5.51"/>
>
>                    </condition>
>
>                  </extension>
>
>
>
>
>
>                 <!--- ext MyServ -->
>
>                  <extension name="Extension MyServ " continue="false">
>
>                    <condition field="destination_number"
> expression="^3341$">
>
>                                                 <action application="set"
> data="bypass_media=true"/>
>
>                                    <action application="bridge"
> data="sofia/internal/3341 at 10.10.5.48"/>
>
>                    </condition>
>
>                  </extension>
>
>
>
> The connection to MyIVR works. Then it sends Refer-to (3341 at ...) which
> fails as I described it in the previous mail.
>
>
>
> I would like to know if the feature has been validated and if I'm missing
> something in the configuration.
>
>
>
> Any help would be very appreciated.
>
>
>
> Thanks!
>
>
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Victor
> Maruani
> *Sent:* Sunday, February 07, 2010 5:01 PM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* [Freeswitch-users] Bypass-media and REFER method
>
>
>
> Hi,
>
>
>
> I'm trying to do a POC using FS, the goal is to have FS handle REFERs
> containing proprietary data.
>
> I want to have some logic on top of FS and also use the fail over
> mechanism.
>
> in short, I have something like this:
>
> (third party) A side  --- FS ---- B side (IVR server)
>
>
>
> the IVR the sends a REFER to FS. I don't want A to deal with it.
>
> now say B refers to C, it would be considered as a "group" C1, C2 ... to
> which I want FS to failover.
>
> only when one has answered should A be updated (REINVITE) and B notified
> and disconnected.
>
> if all fails I would expect B to be notified of the failure and proceed as
> I wish without "losing" A.
>
>
>
> from what I've read FS should be OK for the job but I have a couple issues:
>
>
>
> 1 ) I have some issues getting FS handle a REFER while in bypass-media
> mode.
>
> (I tried with the release and some revisions including latest)
>
> first when I bridge A and B everything is fine and media is bypassed.
>
> When B sends REFER to C:
>
> -           FS immediately NOTIFY B of success and send a reinvite to A
> with SDP containing its own media IP/port.
>
> -          then it does INVITE C with A's SDP.
>
> -          B gets disconnected. A is not updated with C's sdp.
>
> so at this point A sends RTP to FS and C sends RTP to A. ?
>
>
>
> I basically have one extension for B: (set bypass-media and bridge to B)
>
> and another extension to C which does the same actions.
>
> what do you think I do wrong?
>
>
>
>
>
> 2 ) how can I catch the REFER and set variables from it? (like ref-by or
> ref-to)
>
> in the dial plan I do catch the INVITE sent to C, but how to do it with the
> REFER itself?
>
>
>
>
>
> thanks for your help!
>
>
>
>
>
> Best Regards,
>
> Victor.
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/0e7b5e70/attachment-0001.html 

From gavin.henry at gmail.com  Thu Feb 11 15:22:49 2010
From: gavin.henry at gmail.com (Gavin Henry)
Date: Thu, 11 Feb 2010 23:22:49 +0000
Subject: [Freeswitch-users] Mod_Conference capacity....
In-Reply-To: <715035894.962251265928530648.JavaMail.root@mail-2.01.com>
References: <268119402.962161265928492648.JavaMail.root@mail-2.01.com>
	<715035894.962251265928530648.JavaMail.root@mail-2.01.com>
Message-ID: <13ca621c1002111522v7fe55a62w84043ed986155b28@mail.gmail.com>

Why don't use script a test or use sipp and then dial in yourself to listen?

Cheers.

On 11/02/2010, joel.sisko at iconverged.com <joel.sisko at iconverged.com> wrote:
>
>
>
> Mod_Conference
>
> Group, I am looking for anyone?s input (based on your own experience) on how
> many individual three party conference?s and the biggest single conference
> (total number of listeners with just 1 speaker) FreeSwitch can handle using
> the mod_conference application? Just looking for rough numbers if I were to
> use a dual processor quad core system with 12GB of RAM.
>
> I understand that transcoding and other factors create limits but I am just
> looking for same raw numbers that I should be able to obtain if the moon and
> stars were to align correctly.
>
> Thanks for the input in advance.
>
> Joel

-- 
Sent from my mobile device

http://www.suretecsystems.com/services/openldap/
http://www.suretectelecom.com


From brian at freeswitch.org  Thu Feb 11 15:47:46 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 11 Feb 2010 17:47:46 -0600
Subject: [Freeswitch-users] Mod_Conference capacity....
In-Reply-To: <13ca621c1002111522v7fe55a62w84043ed986155b28@mail.gmail.com>
References: <268119402.962161265928492648.JavaMail.root@mail-2.01.com>
	<715035894.962251265928530648.JavaMail.root@mail-2.01.com>
	<13ca621c1002111522v7fe55a62w84043ed986155b28@mail.gmail.com>
Message-ID: <E7571BC2-135B-4743-B941-1A84968A4395@freeswitch.org>

Not an optimal test scenario unless you know wtf you're doing!

/b

On Feb 11, 2010, at 5:22 PM, Gavin Henry wrote:

> Why don't use script a test or use sipp and then dial in yourself to listen?
> 
> Cheers.



From joel.sisko at iconverged.com  Thu Feb 11 16:05:58 2010
From: joel.sisko at iconverged.com (Joel Sisko)
Date: Thu, 11 Feb 2010 18:05:58 -0600 (CST)
Subject: [Freeswitch-users] Mod_Conference capacity....
In-Reply-To: <895275167.980621265932897183.JavaMail.root@mail-2.01.com>
Message-ID: <530719564.981411265933158149.JavaMail.root@mail-2.01.com>

Brian/Gavin thanks for the input. But I agree with Brian, if were that easy I would have done it prior to the post.

Just looking to find out what some of the communities success has been to see if this is a path we should go down for a conference solution platform.

Joel
----- Original Message -----
From: "Brian West" <brian at freeswitch.org>
To: freeswitch-users at lists.freeswitch.org
Sent: Thursday, February 11, 2010 3:47:46 PM GMT -08:00 US/Canada Pacific
Subject: Re: [Freeswitch-users] Mod_Conference capacity....

Not an optimal test scenario unless you know wtf you're doing!

/b

On Feb 11, 2010, at 5:22 PM, Gavin Henry wrote:

> Why don't use script a test or use sipp and then dial in yourself to listen?
> 
> Cheers.


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


From ustcorporation at yahoo.com  Thu Feb 11 19:12:07 2010
From: ustcorporation at yahoo.com (teldev)
Date: Thu, 11 Feb 2010 19:12:07 -0800 (PST)
Subject: [Freeswitch-users] Problem installing latest Wanpipe for Sangoma
 A104DE under Centos 5.3 32-bit
Message-ID: <1265944327228-4559052.post@n2.nabble.com>


On the step "Compiling API Development Utilities ...Failed" received message
"Error:  Failed to compile WANPIPE API Tools  !!!"




-- 
View this message in context: http://n2.nabble.com/Problem-installing-latest-Wanpipe-for-Sangoma-A104DE-under-Centos-5-3-32-bit-tp4559052p4559052.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From brian at freeswitch.org  Thu Feb 11 19:17:43 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 11 Feb 2010 21:17:43 -0600
Subject: [Freeswitch-users] Problem installing latest Wanpipe for
	Sangoma A104DE under Centos 5.3 32-bit
In-Reply-To: <1265944327228-4559052.post@n2.nabble.com>
References: <1265944327228-4559052.post@n2.nabble.com>
Message-ID: <93BAA7CC-0930-4D87-88D7-D0D9A3C2EAE5@freeswitch.org>

I would recommend you contact Sangoma for assistance.

/b

On Feb 11, 2010, at 9:12 PM, teldev wrote:

> 
> On the step "Compiling API Development Utilities ...Failed" received message
> "Error:  Failed to compile WANPIPE API Tools  !!!"
> 



From lakindia89 at gmail.com  Thu Feb 11 20:18:25 2010
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Fri, 12 Feb 2010 09:48:25 +0530
Subject: [Freeswitch-users] Play music to A leg.
In-Reply-To: <191c3a031002110804m4c18e7e0y317f44272b42c8b4@mail.gmail.com>
References: <fe67a3341002090519r2cc6e8edm7702b8d012fc2be3@mail.gmail.com>
	<87f2f3b91002090737l4efa8en29bde9457fef7ea5@mail.gmail.com>
	<fe67a3341002092157x7484f3d2id6678510bbec66cc@mail.gmail.com>
	<87f2f3b91002100702u7581589ua866d2492a008b23@mail.gmail.com>
	<fe67a3341002110211w60390e1cm2d12fadd1e802778@mail.gmail.com>
	<87f2f3b91002110754laf9d01dxa94a36bc1b8c83df@mail.gmail.com>
	<191c3a031002110804m4c18e7e0y317f44272b42c8b4@mail.gmail.com>
Message-ID: <7d79b3931002112018y2512954cuca83ce21ff3406fe@mail.gmail.com>

Dear Antony,
In bridge if we are making parallel calls, then group_confirm_key in execute
mode will execute for all the extensions, and whomsoever finishes the script
first, will be bridged.

But I think nagalenoj need to execute the script for the extension which
answers the call first, not for all the extension.!!!.

>From nanalenoj's post
            " but when I do originate for multiple extensions, I want this
script to work based on the answered extension."


On Thu, Feb 11, 2010 at 9:34 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> group_confirm_key in execute mode can execute a lua script instead that can
> read as many digits as you want and parse the results.
>
>
>
> On Thu, Feb 11, 2010 at 9:54 AM, Michael Collins <msc at freeswitch.org>wrote:
>
>> Hehe, this is getting more and more complicated. You may want to consider
>> using the event socket and have your call control be done from a more 3rd
>> party-ish perspective. If you've got all these different scenarios it might
>> be better to let an external script do all the work.
>>
>> http://wiki.freeswitch.org/wiki/Event_Socket
>>
>> -MC
>>
>>
>> On Thu, Feb 11, 2010 at 2:11 AM, Nagalenoj H. <nagalenoj at gmail.com>wrote:
>>
>>> But My scenario is,
>>>    After I get the call from X.
>>>    I answer the call in some scenarios and won't answer the call. So,
>>> this leg can either be answered or unanswered.
>>>    I originate a call to another number.
>>>    After getting some digits from this originated leg.
>>>    I do uuid_bridge of these 2 legs.
>>>
>>> I want to play some file[ringback] to leg A before bridging to B.
>>>
>>> On Wed, Feb 10, 2010 at 8:32 PM, Michael Collins <msc at freeswitch.org>wrote:
>>>
>>>>
>>>>
>>>> On Tue, Feb 9, 2010 at 9:57 PM, Nagalenoj H. <nagalenoj at gmail.com>wrote:
>>>>
>>>>> Because, I want to get some digits before bridging the legs. I've tried
>>>>> group_confirm_key, but it accepts only one digit, I need multiple digits, so
>>>>> I can't use.
>>>>> I've also tried group_confirm_file, but when I do originate for
>>>>> multiple extensions, I want this script to work based on the answered
>>>>> extension.
>>>>>
>>>>> So, I've originated and processed the events to do my job.
>>>>>
>>>>> How do I play some music to A leg?
>>>>>
>>>>> I might be missing something, but couldn't you just park the call ("A
>>>> leg") until you connect to the other party ("B leg") and then uuid_bridge at
>>>> whatever point you want?
>>>> -MC
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> Regards,
>>> Nagalenoj H.
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100212/3116590a/attachment.html 

From lakindia89 at gmail.com  Thu Feb 11 20:21:57 2010
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Fri, 12 Feb 2010 09:51:57 +0530
Subject: [Freeswitch-users] How to kill multiple UUIDs.
In-Reply-To: <191c3a031002100759v2fef0f48na5afccf46bb08311@mail.gmail.com>
References: <7d79b3931002100307h4d370a22l6f04cafa5bf5806d@mail.gmail.com>
	<A35CD010-E86F-4B7E-A28B-2A9000B8E009@jerris.com>
	<191c3a031002100759v2fef0f48na5afccf46bb08311@mail.gmail.com>
Message-ID: <7d79b3931002112021y21ad551do50f3913352aa1855@mail.gmail.com>

Thanks antony, that works.

Thank you all very much!!

On Wed, Feb 10, 2010 at 9:29 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> or you can set a common var like foo=bar on all the chans and do
>
> hupall normal_clearing foo bar
>
>
>
> On Wed, Feb 10, 2010 at 9:16 AM, Michael Jerris <mike at jerris.com> wrote:
>
>> you can api hangup hook to call
>>
>> lua multi_kill.lua uuid1 uuid2 uuid?.
>>
>> and then write the trivial lua script for that.
>>
>> Mike
>>
>> On Feb 10, 2010, at 6:07 AM, lakshmanan ganapathy wrote:
>>
>> > Hi all,
>> >
>> > My situation is
>> >     A called to 1005 -- Which executes an ESL program.
>> >     Now from the program I will made the parallel call using "api
>> originate [origination_uuid=uuid1]user/1006,[origination_uuid2]user/1001
>> &park()".
>> >     UUID's are obtained from create_uuid.
>> >
>> >     I'll then wait for the api to return, to check whether the call is
>> answered or rejected by the other end.
>> >     But while I'm waiting, if A hangup the call, I just want to kill the
>> calls that are originated by my program.
>> >     So I taught of using api_hang_up_hook and I set that variable to
>> uuid_kill uuid1 uuid2.
>> >     But it only killed the uuid1.
>> >
>> > Is there any other ways to kill multiple uuid's??
>> > please help?
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100212/10476717/attachment-0001.html 

From jingwei.yang at gmail.com  Thu Feb 11 21:56:55 2010
From: jingwei.yang at gmail.com (Jingwei Yang)
Date: Fri, 12 Feb 2010 13:56:55 +0800
Subject: [Freeswitch-users] Is it possible to repeat music in playback
In-Reply-To: <191c3a031002110639h3cf14efdvf7715b973fe3caed@mail.gmail.com>
References: <13529f9d1002110233g57c8a1a5j3cdd38e22fa6a13d@mail.gmail.com>
	<2FBDB302-2B54-40CC-B694-332854219774@freeswitch.org>
	<191c3a031002110639h3cf14efdvf7715b973fe3caed@mail.gmail.com>
Message-ID: <13529f9d1002112156u11603033u698236c86e0d9abb@mail.gmail.com>

Thanks Anthony, exactly what I want.

On Thu, Feb 11, 2010 at 10:39 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> or try endless_playback app
>
>
>
> On Thu, Feb 11, 2010 at 7:26 AM, Brian West <brian at freeswitch.org> wrote:
>
>> Why not just use Fifo to hold them?  Or Park the agent and send the
>> session a message to play music?  You then have options to define loop
>> count.
>>
>> http://wiki.freeswitch.org/wiki/Event_Socket#execute
>>
>> /b
>>
>> On Feb 11, 2010, at 4:33 AM, Jingwei Yang wrote:
>>
>> > Hello,
>> >
>> > I've defined a very simple dialplan like the one below and when the
>> caller is connected to this plan, I hope to keep the call alive and repeat
>> the music set by playback. How am I able to achieve this?
>> >
>> >
>> >     <extension name="agent_hold">
>> >       <condition field="destination_number" expression="^33333$">
>> >         <action application="answer"/>
>> >         <action application="playback" data="customized/test.wav"/>
>> >       </condition>
>> >     </extension>
>> >
>> >
>> > Thanks,
>> > -Jingwei
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100212/f027575c/attachment.html 

From jingwei.yang at gmail.com  Fri Feb 12 00:44:56 2010
From: jingwei.yang at gmail.com (Jingwei Yang)
Date: Fri, 12 Feb 2010 16:44:56 +0800
Subject: [Freeswitch-users] Is it possible to specify music on hold with
	uuid_hold?
In-Reply-To: <E37048AD-5A46-42D5-B724-6C2344AD8A2B@freeswitch.org>
References: <13529f9d1002110130v335d8177m74f791577b558ca@mail.gmail.com>
	<8FF07098-581B-47FB-A039-63A58041E5BF@freeswitch.org>
	<13529f9d1002110554u65431499hc3fa877252023577@mail.gmail.com>
	<E37048AD-5A46-42D5-B724-6C2344AD8A2B@freeswitch.org>
Message-ID: <13529f9d1002120044j712590ch11456b6b98c86d72@mail.gmail.com>

Thanks Brian. I think I understand what you mean about holding the far end
now. When client A and agent B are in a call and I try to uuid_hold B,
client A is in fact put on hold. This looks a bit strange but the good thing
is I'm able to let A hear the customized hold music by defining it in the
dialplan that B is connected to before bridged to A. Thanks!

On Thu, Feb 11, 2010 at 9:59 PM, Brian West <brian at freeswitch.org> wrote:

> uuid_setvar the variable hold_music on the opposite UUID you're holding...
> uuid_hold isn't doing exactly what you think it is.  ;)
>
> /b
>
> On Feb 11, 2010, at 7:54 AM, Jingwei Yang wrote:
>
> > Sorry Brian, I don't quite understand your answer. What is the far end
> and what is the near end? In my case, I bridge client A to agent B. While
> they're talking, I use uuid_hold to put client A on hold. Then A hears the
> default music. After a while, uuid_hold off A and the conversation between A
> and B resumes. uuid_hold looks perfect for my situation except I'm not able
> to change the default music.
> >
> > Regards,
> > -Jingwei
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100212/f828733c/attachment.html 

From codecomplete at free.fr  Fri Feb 12 02:17:47 2010
From: codecomplete at free.fr (Fred-145)
Date: Fri, 12 Feb 2010 11:17:47 +0100
Subject: [Freeswitch-users] Driving peripherals through Freeswitch?
References: <2srvm5945qgcno44oetn9ngii0u3aed73p@4ax.com>
	<2srvm5945qgcno44oetn9ngii0u3aed73p-e09XROE/p8c@public.gmane.org>
	<20F1477A-E98A-4BC9-904D-CB313D8E7B4C@jerris.com>
Message-ID: <slaan5p2j1b0njesrn6u6eokdvno4dns0i@4ax.com>

On Tue, 9 Feb 2010 16:12:38 -0500, Michael Jerris
<mike at jerris.com> wrote:
>The possibilities are limitless, but requires someone to code a module to interface, or external scripts using the system api or some socket based application.

Thanks for the feedback. I'll see what I can find.



From ivdreg at gmail.com  Fri Feb 12 02:20:54 2010
From: ivdreg at gmail.com (ivdreg ivdreg)
Date: Fri, 12 Feb 2010 12:20:54 +0200
Subject: [Freeswitch-users] Help on: park_timeout variable
In-Reply-To: <191c3a031002111153n655103b4ub46d67f9094d6f57@mail.gmail.com>
References: <b437a0e1002110851v274b8cbah54d920a5e59a809b@mail.gmail.com>
	<191c3a031002111142n625ca0d4n641378242cd69b87@mail.gmail.com>
	<191c3a031002111153n655103b4ub46d67f9094d6f57@mail.gmail.com>
Message-ID: <b437a0e1002120220i2b4a9a04jdafdd0ae09102bf@mail.gmail.com>

Thanks Anthony. Everything is fine now.

2010/2/11 Anthony Minessale <anthony.minessale at gmail.com>

> actually, i see a small buglet there, try trunk.
>
>
>
> On Thu, Feb 11, 2010 at 1:42 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> This is what happens in a b leg, it only happens when you transfer a
>> call.  This is by design to give the other phone a chance to kill the leg.
>> This is not really a problem persae.
>>
>>
>> On Thu, Feb 11, 2010 at 10:51 AM, ivdreg ivdreg <ivdreg at gmail.com> wrote:
>>
>>> Hi All,
>>>
>>> After updating to current SVN from 1.0.4 I have a problem when caller
>>> party hangs up a call. I have 3 seconds timeout before B leg disconnects. I
>>> think that this is caused by  code in switch_ivr_bridge.c in function static
>>> switch_status_t audio_bridge_on_exchange_media(switch_core_session_t
>>> *session)
>>> ......
>>>
>>>         if (switch_channel_get_state(channel) == CS_EXCHANGE_MEDIA) {
>>>                 switch_channel_set_variable(channel, "park_timeout",
>>> "3");
>>>                 switch_channel_set_state(channel, CS_PARK);
>>>         }
>>> ......
>>>
>>> This happens even if I set park_after_bridge=false variable.
>>> Is anybody has this problem ?
>>>
>>> Thanks
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100212/710985fd/attachment.html 

From codecomplete at free.fr  Fri Feb 12 02:20:25 2010
From: codecomplete at free.fr (Fred-145)
Date: Fri, 12 Feb 2010 11:20:25 +0100
Subject: [Freeswitch-users] FS based Softphone?
References: <e90351e71002081039y7c7723a6x7181d9fdd208bdde@mail.gmail.com>
Message-ID: <tpaan5t7uldeqk8ud10k4pja31hi2jv7tt@4ax.com>

On Mon, 8 Feb 2010 10:39:46 -0800, Christian Jensen
<christian at officepools.com> wrote:
> If not, what is the best softphone for use with
> a mostly windows but some ubuntu and mac environment?

Looks like ZoIPer is the only softphone available for the three OS's

http://www.zoiper.com/



From lakindia89 at gmail.com  Fri Feb 12 03:23:32 2010
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Fri, 12 Feb 2010 16:53:32 +0530
Subject: [Freeswitch-users] Is it possible to repeat music in playback
In-Reply-To: <191c3a031002110639h3cf14efdvf7715b973fe3caed@mail.gmail.com>
References: <13529f9d1002110233g57c8a1a5j3cdd38e22fa6a13d@mail.gmail.com>
	<2FBDB302-2B54-40CC-B694-332854219774@freeswitch.org>
	<191c3a031002110639h3cf14efdvf7715b973fe3caed@mail.gmail.com>
Message-ID: <7d79b3931002120323x7a2520fcr1610af69ba4ca51@mail.gmail.com>

Hi antony,
Is there any way to stop the endless_playback??
I tried with break. But it didn't worked!!


On Thu, Feb 11, 2010 at 8:09 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> or try endless_playback app
>
>
>
> On Thu, Feb 11, 2010 at 7:26 AM, Brian West <brian at freeswitch.org> wrote:
>
>> Why not just use Fifo to hold them?  Or Park the agent and send the
>> session a message to play music?  You then have options to define loop
>> count.
>>
>> http://wiki.freeswitch.org/wiki/Event_Socket#execute
>>
>> /b
>>
>> On Feb 11, 2010, at 4:33 AM, Jingwei Yang wrote:
>>
>> > Hello,
>> >
>> > I've defined a very simple dialplan like the one below and when the
>> caller is connected to this plan, I hope to keep the call alive and repeat
>> the music set by playback. How am I able to achieve this?
>> >
>> >
>> >     <extension name="agent_hold">
>> >       <condition field="destination_number" expression="^33333$">
>> >         <action application="answer"/>
>> >         <action application="playback" data="customized/test.wav"/>
>> >       </condition>
>> >     </extension>
>> >
>> >
>> > Thanks,
>> > -Jingwei
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100212/9346b79b/attachment.html 

From kond at nstel.ru  Fri Feb 12 03:24:27 2010
From: kond at nstel.ru (Nikolay Kondratyev)
Date: Fri, 12 Feb 2010 14:24:27 +0300
Subject: [Freeswitch-users] h323 - sip call is not working
In-Reply-To: <65d96fc81002111013r38d938b0t3ffc1a49b7ff5b92@mail.gmail.com>
Message-ID: <20100212112427.0FC1F11F9E@mail.nstel.ru>

Tihomir, 

Thanks for the reply.

 

I was quite surprised to see 30ms packetization time. and first thought it's
a typo.

But now I see (if I'm not mistaken) that FS really sends open logical
channel with Alaw:30ms when opening logical channel during fast start (see
attached file).

This parameter is "display only" on my IPO. 

Is it possible to change it on the FS side? I think it should be somewhere
in mod_h323 configuration. But I did not found any mod_h323 configuration
parameters except ip addr. and port.

 

By the way, I know that one can use different packetization times for the
same codec, but I've never heard, that somebody really uses 30 ms for
G711Alaw. Always 20ms.

 

Thanks ans regards,

Nikolay.

 

 

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Tihomir
Culjaga
Sent: Thursday, February 11, 2010 9:14 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] h323 - sip call is not working

 

 

On Thu, Feb 11, 2010 at 4:18 PM, Nikolay Kondratyev <kond at nstel.ru> wrote:

Tihomir,

Thanks for help.

I enabled fast start on IPO and I can hear voice now. But the ringback tone
and voice appears to be wheezy, but I will investigate that tomorrow.


set framing time for your codec to 30ms in IPO, also play with  PI in
alerting... set it to 2.

 

Thanks again.

Nikolay.

 

 

 


  _____  


From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Tihomir
Culjaga
Sent: Thursday, February 11, 2010 2:34 PM


To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] h323 - sip call is not working

 

Nikolay, you are sending slow start with tunneling=true ?!?!



It is not gong to work :)

Please can you set fast start instead?


Your call failed because there was no mediaControll channel negotiated at
all... actually the call had to be aborted because wrong signaling .. but
anyhow.

Please on your IPO use FastStart with h245Tunneling=true... also, same
settings on FS side as well (exclude h245 in setup as well).




Frame 13 (277 bytes on wire, 277 bytes captured)
Ethernet II, Src: AlliedTe_22:9b:4a (00:00:cd:22:9b:4a), Dst:
Vmware_67:33:a7 (00:0c:29:67:33:a7)
Internet Protocol, Src: 172.23.14.2 (172.23.14.2), Dst: 172.23.22.49
(172.23.22.49)
Transmission Control Protocol, Src Port: oirtgsvc (4141), Dst Port:
h323hostcall (1720), Seq: 1, Ack: 1, Len: 223
TPKT, Version: 3, Length: 223
Q.931
    Protocol discriminator: Q.931
    Call reference value length: 2
    Call reference flag: Message sent from originating side
    Call reference value: 0012
    Message type: SETUP (0x05)
    Bearer capability
    Display  'Gornak Alexandr>2853'
    Calling party number: '5840'
    Called party number: '2853'
    User-user
H.225.0 CS
    H323-UserInformation
        h323-uu-pdu
            h323-message-body: setup (0)
                setup
            h4501SupplementaryService: 1 item
            1... .... h245Tunneling: True







On Thu, Feb 11, 2010 at 9:18 AM, Nikolay Kondratyev <kond at nstel.ru> wrote:

> But I do recommenf to use SIP, since it's much better supported by FS.
>
> /Peter

But SIP is poorly supported by IPO.
Thanks and regards,

Nikolay.


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org

http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

 


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100212/403d779e/attachment-0001.html 
-------------- next part --------------
An embedded and charset-unspecified text was scrubbed...
Name: alaw30ms.txt
Url: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100212/403d779e/attachment-0001.txt 

From vetali100 at gmail.com  Fri Feb 12 01:33:27 2010
From: vetali100 at gmail.com (Vitalii Colosov)
Date: Fri, 12 Feb 2010 11:33:27 +0200
Subject: [Freeswitch-users] Other party does not hear voice when connecting
	with X-Lite
Message-ID: <a858e7691002120133q5e4c806fx309517e4fdec45c5@mail.gmail.com>

Hi,

I have installed FS default configuration and setup an external gateway for
international calls.
When I call to international phone using YATE windows client, both parties
able to hear voice... erevything is OK.

But when I connect using X-Lite, it connects but other party cannot hear me.
I can hear him well...

Could you please hint in which direction should I look?

Thank you,
Vitalii Colosov
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100212/5ac6d9ec/attachment.html 

From bottleman at icf.org.ru  Fri Feb 12 04:07:39 2010
From: bottleman at icf.org.ru (Georgiewskiy Yuriy)
Date: Fri, 12 Feb 2010 15:07:39 +0300 (MSK)
Subject: [Freeswitch-users] h323 - sip call is not working
In-Reply-To: <20100212112427.0FC1F11F9E@mail.nstel.ru>
References: <20100212112427.0FC1F11F9E@mail.nstel.ru>
Message-ID: <alpine.DEB.2.00.1002121506130.27140@srv.icf.org.ru>

On 2010-02-12 14:24 +0300, Nikolay Kondratyev wrote freeswitch-users at lists....:

NK>Tihomir, 
NK>
NK>Thanks for the reply.
NK>
NK> 
NK>
NK>I was quite surprised to see 30ms packetization time. and first thought it's
NK>a typo.
NK>
NK>But now I see (if I'm not mistaken) that FS really sends open logical
NK>channel with Alaw:30ms when opening logical channel during fast start (see
NK>attached file).
NK>
NK>This parameter is "display only" on my IPO. 
NK>
NK>Is it possible to change it on the FS side? I think it should be somewhere
NK>in mod_h323 configuration. But I did not found any mod_h323 configuration
NK>parameters except ip addr. and port.

It's not implemented at this time.

NK>By the way, I know that one can use different packetization times for the
NK>same codec, but I've never heard, that somebody really uses 30 ms for
NK>G711Alaw. Always 20ms.
NK>
NK> 
NK>
NK>Thanks ans regards,
NK>
NK>Nikolay.
NK>
NK> 
NK>
NK> 
NK>
NK> 
NK>
NK>  _____  
NK>
NK>From: freeswitch-users-bounces at lists.freeswitch.org
NK>[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Tihomir
NK>Culjaga
NK>Sent: Thursday, February 11, 2010 9:14 PM
NK>To: freeswitch-users at lists.freeswitch.org
NK>Subject: Re: [Freeswitch-users] h323 - sip call is not working
NK>
NK> 
NK>
NK> 
NK>
NK>On Thu, Feb 11, 2010 at 4:18 PM, Nikolay Kondratyev <kond at nstel.ru> wrote:
NK>
NK>Tihomir,
NK>
NK>Thanks for help.
NK>
NK>I enabled fast start on IPO and I can hear voice now. But the ringback tone
NK>and voice appears to be wheezy, but I will investigate that tomorrow.
NK>
NK>
NK>set framing time for your codec to 30ms in IPO, also play with  PI in
NK>alerting... set it to 2.
NK>
NK> 
NK>
NK>Thanks again.
NK>
NK>Nikolay.
NK>
NK> 
NK>
NK> 
NK>
NK> 
NK>
NK>
NK>  _____  
NK>
NK>
NK>From: freeswitch-users-bounces at lists.freeswitch.org
NK>[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Tihomir
NK>Culjaga
NK>Sent: Thursday, February 11, 2010 2:34 PM
NK>
NK>
NK>To: freeswitch-users at lists.freeswitch.org
NK>Subject: Re: [Freeswitch-users] h323 - sip call is not working
NK>
NK> 
NK>
NK>Nikolay, you are sending slow start with tunneling=true ?!?!
NK>
NK>
NK>
NK>It is not gong to work :)
NK>
NK>Please can you set fast start instead?
NK>
NK>
NK>Your call failed because there was no mediaControll channel negotiated at
NK>all... actually the call had to be aborted because wrong signaling .. but
NK>anyhow.
NK>
NK>Please on your IPO use FastStart with h245Tunneling=true... also, same
NK>settings on FS side as well (exclude h245 in setup as well).
NK>
NK>
NK>
NK>
NK>Frame 13 (277 bytes on wire, 277 bytes captured)
NK>Ethernet II, Src: AlliedTe_22:9b:4a (00:00:cd:22:9b:4a), Dst:
NK>Vmware_67:33:a7 (00:0c:29:67:33:a7)
NK>Internet Protocol, Src: 172.23.14.2 (172.23.14.2), Dst: 172.23.22.49
NK>(172.23.22.49)
NK>Transmission Control Protocol, Src Port: oirtgsvc (4141), Dst Port:
NK>h323hostcall (1720), Seq: 1, Ack: 1, Len: 223
NK>TPKT, Version: 3, Length: 223
NK>Q.931
NK>    Protocol discriminator: Q.931
NK>    Call reference value length: 2
NK>    Call reference flag: Message sent from originating side
NK>    Call reference value: 0012
NK>    Message type: SETUP (0x05)
NK>    Bearer capability
NK>    Display  'Gornak Alexandr>2853'
NK>    Calling party number: '5840'
NK>    Called party number: '2853'
NK>    User-user
NK>H.225.0 CS
NK>    H323-UserInformation
NK>        h323-uu-pdu
NK>            h323-message-body: setup (0)
NK>                setup
NK>            h4501SupplementaryService: 1 item
NK>            1... .... h245Tunneling: True
NK>
NK>
NK>
NK>
NK>
NK>
NK>
NK>On Thu, Feb 11, 2010 at 9:18 AM, Nikolay Kondratyev <kond at nstel.ru> wrote:
NK>
NK>> But I do recommenf to use SIP, since it's much better supported by FS.
NK>>
NK>> /Peter
NK>
NK>But SIP is poorly supported by IPO.
NK>Thanks and regards,
NK>
NK>Nikolay.
NK>
NK>
NK>_______________________________________________
NK>FreeSWITCH-users mailing list
NK>FreeSWITCH-users at lists.freeswitch.org
NK>
NK>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
NK>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
NK>http://www.freeswitch.org
NK>
NK> 
NK>
NK>
NK>_______________________________________________
NK>FreeSWITCH-users mailing list
NK>FreeSWITCH-users at lists.freeswitch.org
NK>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
NK>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
NK>http://www.freeswitch.org
NK>
NK> 
NK>
NK>

C ?????????                       With Best Regards
???????????? ????.                Georgiewskiy Yuriy
+7 4872 711666                    +7 4872 711666
???? +7 4872 711143               fax +7 4872 711143
???????? ??? "?? ?? ??????"       IT Service Ltd
http://nkoort.ru                  http://nkoort.ru
JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
YG129-RIPE                        YG129-RIPE

From tculjaga at gmail.com  Fri Feb 12 04:47:08 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Fri, 12 Feb 2010 13:47:08 +0100
Subject: [Freeswitch-users] h323 - sip call is not working
In-Reply-To: <alpine.DEB.2.00.1002121506130.27140@srv.icf.org.ru>
References: <20100212112427.0FC1F11F9E@mail.nstel.ru>
	<alpine.DEB.2.00.1002121506130.27140@srv.icf.org.ru>
Message-ID: <65d96fc81002120447j2010d4fbha771f30b0fc42e6b@mail.gmail.com>

.

> NK>
> NK>This parameter is "display only" on my IPO.
> NK>
> NK>Is it possible to change it on the FS side? I think it should be
> somewhere
>

It is deep in H323plus... The problem is that IPO should stick to the
protocol and adjust the framing size accordingly... Anyhow, yes this is one
of the issues that makes trigger a different call flow on the remote side.
In my scenario, when cisco PGW receives a setup with different framing size,
it triggers a different behaviour.... PGW starts to negotiate framing using
Facility OLC.

On Avaya S8700 the behavior is clean ... it accepts it.

Anyhow, i'm looking for a way to make this config available in the
h323.conf.xml


T.


> NK>in mod_h323 configuration. But I did not found any mod_h323
> configuration
> NK>parameters except ip addr. and port.
>
> It's not implemented at this time.
>
> NK>By the way, I know that one can use different packetization times for
> the
> NK>same codec, but I've never heard, that somebody really uses 30 ms for
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100212/661c9d2a/attachment.html 

From brian at freeswitch.org  Fri Feb 12 06:29:06 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 12 Feb 2010 08:29:06 -0600
Subject: [Freeswitch-users] h323 - sip call is not working
In-Reply-To: <20100212112427.0FC1F11F9E@mail.nstel.ru>
References: <20100212112427.0FC1F11F9E@mail.nstel.ru>
Message-ID: <26C4E111-8329-48F8-A8DA-081B851A9514@freeswitch.org>

This is a rather broad assumption.  I have seen 40ms, 60ms and even 80ms in the wild.  It all depends on what you want to do.  It lowers overhead and increases efficiency on the wire.

/b

On Feb 12, 2010, at 5:24 AM, Nikolay Kondratyev wrote:

> By the way, I know that one can use different packetization times for the same codec, but I?ve never heard, that somebody really uses 30 ms for G711Alaw. Always 20ms.
>  

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100212/d60a20d1/attachment.html 

From brian at freeswitch.org  Fri Feb 12 06:30:28 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 12 Feb 2010 08:30:28 -0600
Subject: [Freeswitch-users] Other party does not hear voice when
	connecting with X-Lite
In-Reply-To: <a858e7691002120133q5e4c806fx309517e4fdec45c5@mail.gmail.com>
References: <a858e7691002120133q5e4c806fx309517e4fdec45c5@mail.gmail.com>
Message-ID: <2F81A680-6A5C-4E3F-9590-FFF6AC4EA80D@freeswitch.org>

is x-lite behind nat with the freeswitch box?  If so you'll need to disable the discover global IP so that it doesn't try to hair pin thru your NAT router.... Most nat routers won't work correctly trying to do that.

/b

On Feb 12, 2010, at 3:33 AM, Vitalii Colosov wrote:

>  
> But when I connect using X-Lite, it connects but other party cannot hear me.
> I can hear him well...



From brian at freeswitch.org  Fri Feb 12 06:31:17 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 12 Feb 2010 08:31:17 -0600
Subject: [Freeswitch-users] FS based Softphone?
In-Reply-To: <tpaan5t7uldeqk8ud10k4pja31hi2jv7tt@4ax.com>
References: <e90351e71002081039y7c7723a6x7181d9fdd208bdde@mail.gmail.com>
	<tpaan5t7uldeqk8ud10k4pja31hi2jv7tt@4ax.com>
Message-ID: <1941DDEC-0B9D-4A9E-87E5-12E70178A0B0@freeswitch.org>

LIES LIES LIES... FSComm runs on all three OS's heck even FreeSWITCH on the command line runs on all three.  :P

/b
PS: Zoiper doesn't do 32kHz or 48kHz voip.

On Feb 12, 2010, at 4:20 AM, Fred-145 wrote:

> Looks like ZoIPer is the only softphone available for the three OS's
> 
> http://www.zoiper.com/

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100212/48e7ec85/attachment.html 

From Prometheus001 at gmx.net  Fri Feb 12 06:37:51 2010
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Fri, 12 Feb 2010 15:37:51 +0100
Subject: [Freeswitch-users] FS based Softphone?
In-Reply-To: <tpaan5t7uldeqk8ud10k4pja31hi2jv7tt@4ax.com>
References: <e90351e71002081039y7c7723a6x7181d9fdd208bdde@mail.gmail.com>
	<tpaan5t7uldeqk8ud10k4pja31hi2jv7tt@4ax.com>
Message-ID: <4B7567BF.3010609@gmx.net>

I even got TLS/SRTP working with Zoiper Bizz.

Best regards
Peter

Fred-145 schrieb:
> On Mon, 8 Feb 2010 10:39:46 -0800, Christian Jensen
> <christian at officepools.com> wrote:
>   
>> If not, what is the best softphone for use with
>> a mostly windows but some ubuntu and mac environment?
>>     
>
> Looks like ZoIPer is the only softphone available for the three OS's
>
> http://www.zoiper.com/
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>   


From edpimentl at gmail.com  Fri Feb 12 06:49:45 2010
From: edpimentl at gmail.com (EdPimentl)
Date: Fri, 12 Feb 2010 09:49:45 -0500
Subject: [Freeswitch-users] FS based Softphone?
In-Reply-To: <4B7567BF.3010609@gmx.net>
References: <e90351e71002081039y7c7723a6x7181d9fdd208bdde@mail.gmail.com> 
	<tpaan5t7uldeqk8ud10k4pja31hi2jv7tt@4ax.com> <4B7567BF.3010609@gmx.net>
Message-ID: <9dc4a1671002120649p45940f7bi85b041740a4fbe1@mail.gmail.com>

Why not take the positive approach and if there are features, you want
FScomm to support, then develop it, or put up a bounty for it.
This would be a more productive and welcome alternative.

-E
http://vCardCloud.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100212/f6edb375/attachment.html 

From brian at freeswitch.org  Fri Feb 12 06:55:00 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 12 Feb 2010 08:55:00 -0600
Subject: [Freeswitch-users] FS based Softphone?
In-Reply-To: <9dc4a1671002120649p45940f7bi85b041740a4fbe1@mail.gmail.com>
References: <e90351e71002081039y7c7723a6x7181d9fdd208bdde@mail.gmail.com>
	<tpaan5t7uldeqk8ud10k4pja31hi2jv7tt@4ax.com>
	<4B7567BF.3010609@gmx.net>
	<9dc4a1671002120649p45940f7bi85b041740a4fbe1@mail.gmail.com>
Message-ID: <0C4E0EED-F33C-4F25-897E-7FCFB8FF35C7@freeswitch.org>

Excellent idea.  ;) BTW it does everything FreeSWITCH can do already.  :P

/b

On Feb 12, 2010, at 8:49 AM, EdPimentl wrote:

> Why not take the positive approach and if there are features, you want FScomm to support, then develop it, or put up a bounty for it.
> This would be a more productive and welcome alternative.
> 
> -E
> http://vCardCloud.com

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100212/1d1d203e/attachment-0001.html 

From jmesquita at freeswitch.org  Fri Feb 12 07:10:45 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Fri, 12 Feb 2010 13:10:45 -0200
Subject: [Freeswitch-users] FS based Softphone?
In-Reply-To: <0C4E0EED-F33C-4F25-897E-7FCFB8FF35C7@freeswitch.org>
References: <e90351e71002081039y7c7723a6x7181d9fdd208bdde@mail.gmail.com>
	<tpaan5t7uldeqk8ud10k4pja31hi2jv7tt@4ax.com>
	<4B7567BF.3010609@gmx.net>
	<9dc4a1671002120649p45940f7bi85b041740a4fbe1@mail.gmail.com>
	<0C4E0EED-F33C-4F25-897E-7FCFB8FF35C7@freeswitch.org>
Message-ID: <ebbc2d551002120710l57ac8dfer83b731dd9ea7c747@mail.gmail.com>

And some extra work and goodies will be available this weekend.


Jo?o Mesquita
FSComm Developer


On Fri, Feb 12, 2010 at 12:55 PM, Brian West <brian at freeswitch.org> wrote:

> Excellent idea.  ;) BTW it does everything FreeSWITCH can do already.  :P
>
> /b
>
> On Feb 12, 2010, at 8:49 AM, EdPimentl wrote:
>
> Why not take the positive approach and if there are features, you want
> FScomm to support, then develop it, or put up a bounty for it.
> This would be a more productive and welcome alternative.
>
> -E
> http://vCardCloud.com
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100212/f1c07c40/attachment.html 

From woodydickson at gmail.com  Fri Feb 12 07:32:30 2010
From: woodydickson at gmail.com (Woody Dickson)
Date: Fri, 12 Feb 2010 07:32:30 -0800
Subject: [Freeswitch-users] transmission status of channel
Message-ID: <a44688a1002120732v7d95a6f9x2d4bd7ed68de59eb@mail.gmail.com>

Hi,

Is there anyway to obtain the %packet lost, latency, and jitter info
for each channel?

Any idea how to obtain those information?

thx,
Woody


From joel.sisko at iconverged.com  Fri Feb 12 08:33:45 2010
From: joel.sisko at iconverged.com (Joel Sisko)
Date: Fri, 12 Feb 2010 10:33:45 -0600 (CST)
Subject: [Freeswitch-users] Mod_Conference capacity....
In-Reply-To: <530719564.981411265933158149.JavaMail.root@mail-2.01.com>
Message-ID: <1804026806.1068641265992425257.JavaMail.root@mail-2.01.com>

Brian/Gavin do you know of any resource/link that can give an indication on what we could be expected of conference capacity? We are trying to determine a few different platforms we can possible use for conferencing and doing a little preliminary homework.

Kindest regards,

Joel


----- Original Message -----
From: "Joel Sisko" <joel.sisko at iconverged.com>
To: freeswitch-users at lists.freeswitch.org
Sent: Thursday, February 11, 2010 4:05:58 PM GMT -08:00 US/Canada Pacific
Subject: Re: [Freeswitch-users] Mod_Conference capacity....

Brian/Gavin thanks for the input. But I agree with Brian, if were that easy I would have done it prior to the post.

Just looking to find out what some of the communities success has been to see if this is a path we should go down for a conference solution platform.

Joel
----- Original Message -----
From: "Brian West" <brian at freeswitch.org>
To: freeswitch-users at lists.freeswitch.org
Sent: Thursday, February 11, 2010 3:47:46 PM GMT -08:00 US/Canada Pacific
Subject: Re: [Freeswitch-users] Mod_Conference capacity....

Not an optimal test scenario unless you know wtf you're doing!

/b

On Feb 11, 2010, at 5:22 PM, Gavin Henry wrote:

> Why don't use script a test or use sipp and then dial in yourself to listen?
> 
> Cheers.


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


From anthony.minessale at gmail.com  Fri Feb 12 08:56:31 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 12 Feb 2010 10:56:31 -0600
Subject: [Freeswitch-users] Mod_Conference capacity....
In-Reply-To: <1804026806.1068641265992425257.JavaMail.root@mail-2.01.com>
References: <530719564.981411265933158149.JavaMail.root@mail-2.01.com>
	<1804026806.1068641265992425257.JavaMail.root@mail-2.01.com>
Message-ID: <191c3a031002120856q93e52cep14ee96f1e7ed6d1@mail.gmail.com>

consider commercial support from consulting at freeswitch.org


On Fri, Feb 12, 2010 at 10:33 AM, Joel Sisko <joel.sisko at iconverged.com>wrote:

> Brian/Gavin do you know of any resource/link that can give an indication on
> what we could be expected of conference capacity? We are trying to determine
> a few different platforms we can possible use for conferencing and doing a
> little preliminary homework.
>
> Kindest regards,
>
> Joel
>
>
> ----- Original Message -----
> From: "Joel Sisko" <joel.sisko at iconverged.com>
> To: freeswitch-users at lists.freeswitch.org
> Sent: Thursday, February 11, 2010 4:05:58 PM GMT -08:00 US/Canada Pacific
> Subject: Re: [Freeswitch-users] Mod_Conference capacity....
>
> Brian/Gavin thanks for the input. But I agree with Brian, if were that easy
> I would have done it prior to the post.
>
> Just looking to find out what some of the communities success has been to
> see if this is a path we should go down for a conference solution platform.
>
> Joel
> ----- Original Message -----
> From: "Brian West" <brian at freeswitch.org>
> To: freeswitch-users at lists.freeswitch.org
> Sent: Thursday, February 11, 2010 3:47:46 PM GMT -08:00 US/Canada Pacific
> Subject: Re: [Freeswitch-users] Mod_Conference capacity....
>
> Not an optimal test scenario unless you know wtf you're doing!
>
> /b
>
> On Feb 11, 2010, at 5:22 PM, Gavin Henry wrote:
>
> > Why don't use script a test or use sipp and then dial in yourself to
> listen?
> >
> > Cheers.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100212/8f065cfd/attachment.html 

From anthony.minessale at gmail.com  Fri Feb 12 09:07:03 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 12 Feb 2010 11:07:03 -0600
Subject: [Freeswitch-users] Play music to A leg.
In-Reply-To: <7d79b3931002112018y2512954cuca83ce21ff3406fe@mail.gmail.com>
References: <fe67a3341002090519r2cc6e8edm7702b8d012fc2be3@mail.gmail.com>
	<87f2f3b91002090737l4efa8en29bde9457fef7ea5@mail.gmail.com>
	<fe67a3341002092157x7484f3d2id6678510bbec66cc@mail.gmail.com>
	<87f2f3b91002100702u7581589ua866d2492a008b23@mail.gmail.com>
	<fe67a3341002110211w60390e1cm2d12fadd1e802778@mail.gmail.com>
	<87f2f3b91002110754laf9d01dxa94a36bc1b8c83df@mail.gmail.com>
	<191c3a031002110804m4c18e7e0y317f44272b42c8b4@mail.gmail.com>
	<7d79b3931002112018y2512954cuca83ce21ff3406fe@mail.gmail.com>
Message-ID: <191c3a031002120907l28fbdf2dgab5df7dd1b5a2f76@mail.gmail.com>

the script executes for everyone and gives them a chance to dial multiple
digits to test for, this is what he asked for, instead of 1 digit dial
multiple digits.  you set the correct string as a variable on the channel
and everybody runs the script and whoever dials the right digits wins the
rest will be hungup on.


On Thu, Feb 11, 2010 at 10:18 PM, lakshmanan ganapathy <lakindia89 at gmail.com
> wrote:

> Dear Antony,
> In bridge if we are making parallel calls, then group_confirm_key in
> execute mode will execute for all the extensions, and whomsoever finishes
> the script first, will be bridged.
>
> But I think nagalenoj need to execute the script for the extension which
> answers the call first, not for all the extension.!!!.
>
> From nanalenoj's post
>
>             " but when I do originate for multiple extensions, I want this
> script to work based on the answered extension."
>
>
> On Thu, Feb 11, 2010 at 9:34 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> group_confirm_key in execute mode can execute a lua script instead that
>> can read as many digits as you want and parse the results.
>>
>>
>>
>> On Thu, Feb 11, 2010 at 9:54 AM, Michael Collins <msc at freeswitch.org>wrote:
>>
>>> Hehe, this is getting more and more complicated. You may want to consider
>>> using the event socket and have your call control be done from a more 3rd
>>> party-ish perspective. If you've got all these different scenarios it might
>>> be better to let an external script do all the work.
>>>
>>> http://wiki.freeswitch.org/wiki/Event_Socket
>>>
>>> -MC
>>>
>>>
>>> On Thu, Feb 11, 2010 at 2:11 AM, Nagalenoj H. <nagalenoj at gmail.com>wrote:
>>>
>>>> But My scenario is,
>>>>    After I get the call from X.
>>>>    I answer the call in some scenarios and won't answer the call. So,
>>>> this leg can either be answered or unanswered.
>>>>    I originate a call to another number.
>>>>    After getting some digits from this originated leg.
>>>>    I do uuid_bridge of these 2 legs.
>>>>
>>>> I want to play some file[ringback] to leg A before bridging to B.
>>>>
>>>> On Wed, Feb 10, 2010 at 8:32 PM, Michael Collins <msc at freeswitch.org>wrote:
>>>>
>>>>>
>>>>>
>>>>> On Tue, Feb 9, 2010 at 9:57 PM, Nagalenoj H. <nagalenoj at gmail.com>wrote:
>>>>>
>>>>>> Because, I want to get some digits before bridging the legs. I've
>>>>>> tried group_confirm_key, but it accepts only one digit, I need multiple
>>>>>> digits, so I can't use.
>>>>>> I've also tried group_confirm_file, but when I do originate for
>>>>>> multiple extensions, I want this script to work based on the answered
>>>>>> extension.
>>>>>>
>>>>>> So, I've originated and processed the events to do my job.
>>>>>>
>>>>>> How do I play some music to A leg?
>>>>>>
>>>>>> I might be missing something, but couldn't you just park the call ("A
>>>>> leg") until you connect to the other party ("B leg") and then uuid_bridge at
>>>>> whatever point you want?
>>>>> -MC
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>>
>>>> --
>>>> Regards,
>>>> Nagalenoj H.
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100212/7f207adc/attachment-0001.html 

From msc at freeswitch.org  Fri Feb 12 09:22:01 2010
From: msc at freeswitch.org (Michael S Collins)
Date: Fri, 12 Feb 2010 10:22:01 -0700
Subject: [Freeswitch-users] transmission status of channel
In-Reply-To: <a44688a1002120732v7d95a6f9x2d4bd7ed68de59eb@mail.gmail.com>
References: <a44688a1002120732v7d95a6f9x2d4bd7ed68de59eb@mail.gmail.com>
Message-ID: <3028D6AC-46F2-4291-BAA8-E49C9347917B@freeswitch.org>

Check in the XML cdrs. I'm on a plan right now so I can't easily point  
you to a specific wiki page. :)

-MC

Sent from my iPhone

On Feb 12, 2010, at 8:32 AM, Woody Dickson <woodydickson at gmail.com>  
wrote:

> Hi,
>
> Is there anyway to obtain the %packet lost, latency, and jitter info
> for each channel?
>
> Any idea how to obtain those information?
>
> thx,
> Woody
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org


From joel.sisko at iconverged.com  Fri Feb 12 09:30:45 2010
From: joel.sisko at iconverged.com (Joel Sisko)
Date: Fri, 12 Feb 2010 11:30:45 -0600 (CST)
Subject: [Freeswitch-users] Mod_Conference capacity....
In-Reply-To: <191c3a031002120856q93e52cep14ee96f1e7ed6d1@mail.gmail.com>
Message-ID: <324899354.1084631265995845519.JavaMail.root@mail-2.01.com>

So if I understand correctly that there are no numbers to be considered from anyone at this point in regards to capacity as a general guideline? 

Kindest regards, 

Joel 
----- Original Message ----- 
From: "Anthony Minessale" <anthony.minessale at gmail.com> 
To: freeswitch-users at lists.freeswitch.org 
Sent: Friday, February 12, 2010 8:56:31 AM GMT -08:00 US/Canada Pacific 
Subject: Re: [Freeswitch-users] Mod_Conference capacity.... 

consider commercial support from consulting at freeswitch.org 



On Fri, Feb 12, 2010 at 10:33 AM, Joel Sisko < joel.sisko at iconverged.com > wrote: 


Brian/Gavin do you know of any resource/link that can give an indication on what we could be expected of conference capacity? We are trying to determine a few different platforms we can possible use for conferencing and doing a little preliminary homework. 

Kindest regards, 

Joel 



----- Original Message ----- 
From: "Joel Sisko" < joel.sisko at iconverged.com > 
To: freeswitch-users at lists.freeswitch.org 



Sent: Thursday, February 11, 2010 4:05:58 PM GMT -08:00 US/Canada Pacific 
Subject: Re: [Freeswitch-users] Mod_Conference capacity.... 

Brian/Gavin thanks for the input. But I agree with Brian, if were that easy I would have done it prior to the post. 

Just looking to find out what some of the communities success has been to see if this is a path we should go down for a conference solution platform. 

Joel 
----- Original Message ----- 
From: "Brian West" < brian at freeswitch.org > 
To: freeswitch-users at lists.freeswitch.org 
Sent: Thursday, February 11, 2010 3:47:46 PM GMT -08:00 US/Canada Pacific 
Subject: Re: [Freeswitch-users] Mod_Conference capacity.... 

Not an optimal test scenario unless you know wtf you're doing! 

/b 

On Feb 11, 2010, at 5:22 PM, Gavin Henry wrote: 

> Why don't use script a test or use sipp and then dial in yourself to listen? 
> 
> Cheers. 


_______________________________________________ 
FreeSWITCH-users mailing list 
FreeSWITCH-users at lists.freeswitch.org 
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users 
UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users 
http://www.freeswitch.org 

_______________________________________________ 
FreeSWITCH-users mailing list 
FreeSWITCH-users at lists.freeswitch.org 
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users 
UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users 
http://www.freeswitch.org 

_______________________________________________ 
FreeSWITCH-users mailing list 
FreeSWITCH-users at lists.freeswitch.org 
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users 
UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users 
http://www.freeswitch.org 



-- 
Anthony Minessale II 

FreeSWITCH http://www.freeswitch.org/ 
ClueCon http://www.cluecon.com/ 
Twitter: http://twitter.com/FreeSWITCH_wire 

AIM: anthm 
MSN:anthony_minessale at hotmail.com 
GTALK/JABBER/ PAYPAL:anthony.minessale at gmail.com 
IRC: irc.freenode.net #freeswitch 

FreeSWITCH Developer Conference 
sip:888 at conference.freeswitch.org 
iax:guest at conference.freeswitch.org/888 
googletalk:conf+888 at conference.freeswitch.org 
pstn:+19193869900 

_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org 
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100212/f499aaff/attachment.html 

From anthony.minessale at gmail.com  Fri Feb 12 09:44:24 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 12 Feb 2010 11:44:24 -0600
Subject: [Freeswitch-users] Mod_Conference capacity....
In-Reply-To: <324899354.1084631265995845519.JavaMail.root@mail-2.01.com>
References: <191c3a031002120856q93e52cep14ee96f1e7ed6d1@mail.gmail.com>
	<324899354.1084631265995845519.JavaMail.root@mail-2.01.com>
Message-ID: <191c3a031002120944u346a3f7cq779f5c9e0ae31522@mail.gmail.com>

Producing benchmark numbers for an opensource project is a cardinal sin.
I can safely tell you that it is "many" and very competitive with anything
else you will encounter as long as you use a modern multi-core 64bit
machine.

As I said there is commercial support available which is customary for
anyone needing assistance setting up a company.


On Fri, Feb 12, 2010 at 11:30 AM, Joel Sisko <joel.sisko at iconverged.com>wrote:

> So if I understand correctly that there are no numbers to be considered
> from anyone at this point in regards to capacity as a general guideline?
>
>
> Kindest regards,
>
> Joel
> ----- Original Message -----
> From: "Anthony Minessale" <anthony.minessale at gmail.com>
> To: freeswitch-users at lists.freeswitch.org
> Sent: Friday, February 12, 2010 8:56:31 AM GMT -08:00 US/Canada Pacific
> Subject: Re: [Freeswitch-users] Mod_Conference capacity....
>
> consider commercial support from consulting at freeswitch.org
>
>
> On Fri, Feb 12, 2010 at 10:33 AM, Joel Sisko <joel.sisko at iconverged.com>wrote:
>
>> Brian/Gavin do you know of any resource/link that can give an indication
>> on what we could be expected of conference capacity? We are trying to
>> determine a few different platforms we can possible use for conferencing and
>> doing a little preliminary homework.
>>
>> Kindest regards,
>>
>> Joel
>>
>>
>> ----- Original Message -----
>> From: "Joel Sisko" <joel.sisko at iconverged.com>
>> To: freeswitch-users at lists.freeswitch.org
>> Sent: Thursday, February 11, 2010 4:05:58 PM GMT -08:00 US/Canada Pacific
>> Subject: Re: [Freeswitch-users] Mod_Conference capacity....
>>
>> Brian/Gavin thanks for the input. But I agree with Brian, if were that
>> easy I would have done it prior to the post.
>>
>> Just looking to find out what some of the communities success has been to
>> see if this is a path we should go down for a conference solution platform.
>>
>> Joel
>> ----- Original Message -----
>> From: "Brian West" <brian at freeswitch.org>
>> To: freeswitch-users at lists.freeswitch.org
>> Sent: Thursday, February 11, 2010 3:47:46 PM GMT -08:00 US/Canada Pacific
>> Subject: Re: [Freeswitch-users] Mod_Conference capacity....
>>
>> Not an optimal test scenario unless you know wtf you're doing!
>>
>> /b
>>
>> On Feb 11, 2010, at 5:22 PM, Gavin Henry wrote:
>>
>> > Why don't use script a test or use sipp and then dial in yourself to
>> listen?
>> >
>> > Cheers.
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________ FreeSWITCH-users mailing
> list FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100212/a8baa250/attachment-0001.html 

From max.bridgewater at gmail.com  Fri Feb 12 09:51:30 2010
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Fri, 12 Feb 2010 09:51:30 -0800
Subject: [Freeswitch-users] Choppy connection in one direction
Message-ID: <c55d46951002120951o324c16bwd8b0da5c02b30721@mail.gmail.com>

Hi Gents,

What can be wrong with my settings? Pleas help. I have a simple
Portech VoIP-GSM gateway in an African country where i try to
terminate calls. My setup is:

Skype<->Skypiax<->Portech Gateway<->GSM Phone

I have this setup in place right now. Calling from Europe, people can
understand me down there without a glitch. But I can barely hear what
they say.

I replaced the setup with: Skype  <->Skypiax<-> Skype. No change in result.

Then I tried with the Howler G729 codec:
Setup 1:  Xlite <-> Freeswitch <->Portech <->GSM Phone: No change in result.
Setup 2:  GSM Phone <-> Voip.MS <-> Freeswitch <->Portech <->GSM
Phone: No change in result.

In all these cases, the communication is perfect in one direction but
very choppy in the orther.
The only thing that works is direct Skype to Skype. And it works
perfectly, suggesting that the connection is probably not the issue.
The speed i could measure is 140kbs/30kbs.

And assuming the the internet connection is the issue, i would have
hoped that if Skype<->Skype works, then Skype<->Skypiax<->Skype would
also work.

Any idea?

Max.


From rupa at rupa.com  Fri Feb 12 09:52:13 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Fri, 12 Feb 2010 11:52:13 -0600
Subject: [Freeswitch-users] transmission status of channel
In-Reply-To: <3028D6AC-46F2-4291-BAA8-E49C9347917B@freeswitch.org>
References: <a44688a1002120732v7d95a6f9x2d4bd7ed68de59eb@mail.gmail.com>
	<3028D6AC-46F2-4291-BAA8-E49C9347917B@freeswitch.org>
Message-ID: <b39273531002120952u698ab5d1x799696b052ced9a9@mail.gmail.com>

I'm pretty sure that info doesn't exist.  Don't we need RTCP (plus
infrastructure for measuring) for this?

On Fri, Feb 12, 2010 at 11:22 AM, Michael S Collins <msc at freeswitch.org>wrote:

> Check in the XML cdrs. I'm on a plan right now so I can't easily point
> you to a specific wiki page. :)
>
> -MC
>
> Sent from my iPhone
>
> On Feb 12, 2010, at 8:32 AM, Woody Dickson <woodydickson at gmail.com>
> wrote:
>
> > Hi,
> >
> > Is there anyway to obtain the %packet lost, latency, and jitter info
> > for each channel?
> >
> > Any idea how to obtain those information?
> >
> > thx,
> > Woody
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> > users
> > http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
-Rupa
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100212/ce583762/attachment.html 

From tim at novion.ru  Fri Feb 12 10:10:23 2010
From: tim at novion.ru (Timur Valishev)
Date: Fri, 12 Feb 2010 21:10:23 +0300
Subject: [Freeswitch-users] transmission status of channel
In-Reply-To: <b39273531002120952u698ab5d1x799696b052ced9a9@mail.gmail.com>
References: <a44688a1002120732v7d95a6f9x2d4bd7ed68de59eb@mail.gmail.com>
	<3028D6AC-46F2-4291-BAA8-E49C9347917B@freeswitch.org>
	<b39273531002120952u698ab5d1x799696b052ced9a9@mail.gmail.com>
Message-ID: <8e9d67561002121010p37b16aceyb06cda0420e1aa99@mail.gmail.com>

It would be very nice if FS pass RTCP information to channel vars...

Best regards,
Timur Valishev

2010/2/12 Rupa Schomaker <rupa at rupa.com>:
> I'm pretty sure that info doesn't exist. ?Don't we need RTCP (plus
> infrastructure for measuring) for this?
>
> On Fri, Feb 12, 2010 at 11:22 AM, Michael S Collins <msc at freeswitch.org>
> wrote:
>>
>> Check in the XML cdrs. I'm on a plan right now so I can't easily point
>> you to a specific wiki page. :)
>>
>> -MC
>>
>> Sent from my iPhone
>>
>> On Feb 12, 2010, at 8:32 AM, Woody Dickson <woodydickson at gmail.com>
>> wrote:
>>
>> > Hi,
>> >
>> > Is there anyway to obtain the %packet lost, latency, and jitter info
>> > for each channel?
>> >
>> > Any idea how to obtain those information?
>> >
>> > thx,
>> > Woody
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>> > users
>> > http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From joel.sisko at iconverged.com  Fri Feb 12 10:25:25 2010
From: joel.sisko at iconverged.com (joel.sisko at iconverged.com)
Date: Fri, 12 Feb 2010 12:25:25 -0600 (CST)
Subject: [Freeswitch-users] Mod_Conference capacity....
In-Reply-To: <892897350.1100311265999066927.JavaMail.root@mail-2.01.com>
Message-ID: <1235523542.1100531265999125823.JavaMail.root@mail-2.01.com>


Anthony thanks for the input. I am not looking for benchmark and understand the reasons why it does not make sense to do so since application usage and hardware will effect that benchmark. 

Looking for some successes by the group that they can share that would lead me to believe that using FreeSwitch would be worth the time and money to invest over our current conference solution. As an example, I can state that I have seen Yate used with 200 people in a single conference (all G711) work flawlessly, so does 200 count as "many" from your perspective or will FreeSwitch do that in its sleep? 

Thanks for the help. 

Joel 


----- Original Message ----- 
From: "Anthony Minessale" <anthony.minessale at gmail.com> 
To: freeswitch-users at lists.freeswitch.org 
Sent: Friday, February 12, 2010 9:44:24 AM GMT -08:00 US/Canada Pacific 
Subject: Re: [Freeswitch-users] Mod_Conference capacity.... 

Producing benchmark numbers for an opensource project is a cardinal sin. 
I can safely tell you that it is "many" and very competitive with anything else you will encounter as long as you use a modern multi-core 64bit machine. 

As I said there is commercial support available which is customary for anyone needing assistance setting up a company. 



On Fri, Feb 12, 2010 at 11:30 AM, Joel Sisko < joel.sisko at iconverged.com > wrote: 




So if I understand correctly that there are no numbers to be considered from anyone at this point in regards to capacity as a general guideline? 


Kindest regards, 

Joel 
----- Original Message ----- 

From: "Anthony Minessale" < anthony.minessale at gmail.com > 
To: freeswitch-users at lists.freeswitch.org 



Sent: Friday, February 12, 2010 8:56:31 AM GMT -08:00 US/Canada Pacific 
Subject: Re: [Freeswitch-users] Mod_Conference capacity.... 

consider commercial support from consulting at freeswitch.org 



On Fri, Feb 12, 2010 at 10:33 AM, Joel Sisko < joel.sisko at iconverged.com > wrote: 


Brian/Gavin do you know of any resource/link that can give an indication on what we could be expected of conference capacity? We are trying to determine a few different platforms we can possible use for conferencing and doing a little preliminary homework. 

Kindest regards, 

Joel 



----- Original Message ----- 
From: "Joel Sisko" < joel.sisko at iconverged.com > 
To: freeswitch-users at lists.freeswitch.org 



Sent: Thursday, February 11, 2010 4:05:58 PM GMT -08:00 US/Canada Pacific 
Subject: Re: [Freeswitch-users] Mod_Conference capacity.... 

Brian/Gavin thanks for the input. But I agree with Brian, if were that easy I would have done it prior to the post. 

Just looking to find out what some of the communities success has been to see if this is a path we should go down for a conference solution platform. 

Joel 
----- Original Message ----- 
From: "Brian West" < brian at freeswitch.org > 
To: freeswitch-users at lists.freeswitch.org 
Sent: Thursday, February 11, 2010 3:47:46 PM GMT -08:00 US/Canada Pacific 
Subject: Re: [Freeswitch-users] Mod_Conference capacity.... 

Not an optimal test scenario unless you know wtf you're doing! 

/b 

On Feb 11, 2010, at 5:22 PM, Gavin Henry wrote: 

> Why don't use script a test or use sipp and then dial in yourself to listen? 
> 
> Cheers. 


_______________________________________________ 
FreeSWITCH-users mailing list 
FreeSWITCH-users at lists.freeswitch.org 
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users 
UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users 
http://www.freeswitch.org 

_______________________________________________ 
FreeSWITCH-users mailing list 
FreeSWITCH-users at lists.freeswitch.org 
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users 
UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users 
http://www.freeswitch.org 

_______________________________________________ 
FreeSWITCH-users mailing list 
FreeSWITCH-users at lists.freeswitch.org 
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users 
UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users 
http://www.freeswitch.org 



-- 
Anthony Minessale II 

FreeSWITCH http://www.freeswitch.org/ 
ClueCon http://www.cluecon.com/ 
Twitter: http://twitter.com/FreeSWITCH_wire 

AIM: anthm 
MSN:anthony_minessale at hotmail.com 
GTALK/JABBER/ PAYPAL:anthony.minessale at gmail.com 
IRC: irc.freenode.net #freeswitch 

FreeSWITCH Developer Conference 
sip:888 at conference.freeswitch.org 
iax:guest at conference.freeswitch.org/888 
googletalk:conf+888 at conference.freeswitch.org 
pstn:+19193869900 

_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org 
_______________________________________________ 
FreeSWITCH-users mailing list 
FreeSWITCH-users at lists.freeswitch.org 
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users 
UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users 
http://www.freeswitch.org 




-- 
Anthony Minessale II 

FreeSWITCH http://www.freeswitch.org/ 
ClueCon http://www.cluecon.com/ 
Twitter: http://twitter.com/FreeSWITCH_wire 

AIM: anthm 
MSN:anthony_minessale at hotmail.com 
GTALK/JABBER/ PAYPAL:anthony.minessale at gmail.com 
IRC: irc.freenode.net #freeswitch 

FreeSWITCH Developer Conference 
sip:888 at conference.freeswitch.org 
iax:guest at conference.freeswitch.org/888 
googletalk:conf+888 at conference.freeswitch.org 
pstn:+19193869900 

_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org 
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100212/57a30aad/attachment-0001.html 

From gmaruzz at celliax.org  Fri Feb 12 11:10:06 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Fri, 12 Feb 2010 20:10:06 +0100
Subject: [Freeswitch-users] Choppy connection in one direction
In-Reply-To: <c55d46951002120951o324c16bwd8b0da5c02b30721@mail.gmail.com>
References: <c55d46951002120951o324c16bwd8b0da5c02b30721@mail.gmail.com>
Message-ID: <7b197bef1002121110v31bcbf0bmaf46c3b54cd6645a@mail.gmail.com>

it do not seems to be a problem of connection, seems a problem of
audio levels, I mean volumes...

Have I understood correctly?

-gm

On Fri, Feb 12, 2010 at 6:51 PM, Max Bridgewater
<max.bridgewater at gmail.com> wrote:
> Hi Gents,
>
> What can be wrong with my settings? Pleas help. I have a simple
> Portech VoIP-GSM gateway in an African country where i try to
> terminate calls. My setup is:
>
> Skype<->Skypiax<->Portech Gateway<->GSM Phone
>
> I have this setup in place right now. Calling from Europe, people can
> understand me down there without a glitch. But I can barely hear what
> they say.
>
> I replaced the setup with: Skype ?<->Skypiax<-> Skype. No change in result.
>
> Then I tried with the Howler G729 codec:
> Setup 1: ?Xlite <-> Freeswitch <->Portech <->GSM Phone: No change in result.
> Setup 2: ?GSM Phone <-> Voip.MS <-> Freeswitch <->Portech <->GSM
> Phone: No change in result.
>
> In all these cases, the communication is perfect in one direction but
> very choppy in the orther.
> The only thing that works is direct Skype to Skype. And it works
> perfectly, suggesting that the connection is probably not the issue.
> The speed i could measure is 140kbs/30kbs.
>
> And assuming the the internet connection is the issue, i would have
> hoped that if Skype<->Skype works, then Skype<->Skypiax<->Skype would
> also work.
>
> Any idea?
>
> Max.
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From max.bridgewater at gmail.com  Fri Feb 12 11:17:39 2010
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Fri, 12 Feb 2010 11:17:39 -0800
Subject: [Freeswitch-users] Choppy connection in one direction
In-Reply-To: <7b197bef1002121110v31bcbf0bmaf46c3b54cd6645a@mail.gmail.com>
References: <c55d46951002120951o324c16bwd8b0da5c02b30721@mail.gmail.com>
	<7b197bef1002121110v31bcbf0bmaf46c3b54cd6645a@mail.gmail.com>
Message-ID: <c55d46951002121117m67e24aa2y37c393161d318a93@mail.gmail.com>

Well, i can hear the other side but not understand. The sound is
complely choppy; sounding more like an alien trying to talk to me!

On Fri, Feb 12, 2010 at 11:10 AM, Giovanni Maruzzelli
<gmaruzz at celliax.org> wrote:
> it do not seems to be a problem of connection, seems a problem of
> audio levels, I mean volumes...
>
> Have I understood correctly?
>
> -gm
>
> On Fri, Feb 12, 2010 at 6:51 PM, Max Bridgewater
> <max.bridgewater at gmail.com> wrote:
>> Hi Gents,
>>
>> What can be wrong with my settings? Pleas help. I have a simple
>> Portech VoIP-GSM gateway in an African country where i try to
>> terminate calls. My setup is:
>>
>> Skype<->Skypiax<->Portech Gateway<->GSM Phone
>>
>> I have this setup in place right now. Calling from Europe, people can
>> understand me down there without a glitch. But I can barely hear what
>> they say.
>>
>> I replaced the setup with: Skype ?<->Skypiax<-> Skype. No change in result.
>>
>> Then I tried with the Howler G729 codec:
>> Setup 1: ?Xlite <-> Freeswitch <->Portech <->GSM Phone: No change in result.
>> Setup 2: ?GSM Phone <-> Voip.MS <-> Freeswitch <->Portech <->GSM
>> Phone: No change in result.
>>
>> In all these cases, the communication is perfect in one direction but
>> very choppy in the orther.
>> The only thing that works is direct Skype to Skype. And it works
>> perfectly, suggesting that the connection is probably not the issue.
>> The speed i could measure is 140kbs/30kbs.
>>
>> And assuming the the internet connection is the issue, i would have
>> hoped that if Skype<->Skype works, then Skype<->Skypiax<->Skype would
>> also work.
>>
>> Any idea?
>>
>> Max.
>>
>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>


From sos at sokhapkin.dyndns.org  Fri Feb 12 11:22:18 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Fri, 12 Feb 2010 14:22:18 -0500
Subject: [Freeswitch-users] bypass_media bug?
Message-ID: <201002121422.18544.sos@sokhapkin.dyndns.org>

Simple dialplan:

    <extension name="103">
      <condition field="destination_number" expression="^103$">
        <action application="set" data="continue_on_fail=true"/>
        <action application="set" data="bypass_media=true"/>
        <action application="bridge" data="sofia/cwu/103 at 192.168.1.254"/>
        <action application="bridge" data="sofia/cwu/104 at 192.168.1.254"/>
      </condition>
    </extension>

103 at 192.168.1.254 returns 183 early media and then "480 temporary 
unavailable", 104 at 192.168.1.254 answers the call (echo test).

When 104 answers, Freeswitch returns to caller in SDP media port from "183", 
but not media port from 104's "200 OK"

Is it a bug or expected behavior? If expected - is there a variable to control 
the behavior? Everything works OK if I replace bypass_media=true with 
bypass_media_after_bridge=true, but sending reinvites is not acceptable to 
me.


From gmaruzz at celliax.org  Fri Feb 12 11:40:07 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Fri, 12 Feb 2010 20:40:07 +0100
Subject: [Freeswitch-users] Choppy connection in one direction
In-Reply-To: <c55d46951002121117m67e24aa2y37c393161d318a93@mail.gmail.com>
References: <c55d46951002120951o324c16bwd8b0da5c02b30721@mail.gmail.com> 
	<7b197bef1002121110v31bcbf0bmaf46c3b54cd6645a@mail.gmail.com> 
	<c55d46951002121117m67e24aa2y37c393161d318a93@mail.gmail.com>
Message-ID: <7b197bef1002121140n1814c09fr70293e46ada06175@mail.gmail.com>

Please, don't talk to aliens!

And you have this issue in both cases, when you use skypiax and when
you don't use it?

I mean, you have this issue any times you put FS in the middle, huh?




On Fri, Feb 12, 2010 at 8:17 PM, Max Bridgewater
<max.bridgewater at gmail.com> wrote:
> Well, i can hear the other side but not understand. The sound is
> complely choppy; sounding more like an alien trying to talk to me!
>
> On Fri, Feb 12, 2010 at 11:10 AM, Giovanni Maruzzelli
> <gmaruzz at celliax.org> wrote:
>> it do not seems to be a problem of connection, seems a problem of
>> audio levels, I mean volumes...
>>
>> Have I understood correctly?
>>
>> -gm
>>
>> On Fri, Feb 12, 2010 at 6:51 PM, Max Bridgewater
>> <max.bridgewater at gmail.com> wrote:
>>> Hi Gents,
>>>
>>> What can be wrong with my settings? Pleas help. I have a simple
>>> Portech VoIP-GSM gateway in an African country where i try to
>>> terminate calls. My setup is:
>>>
>>> Skype<->Skypiax<->Portech Gateway<->GSM Phone
>>>
>>> I have this setup in place right now. Calling from Europe, people can
>>> understand me down there without a glitch. But I can barely hear what
>>> they say.
>>>
>>> I replaced the setup with: Skype ?<->Skypiax<-> Skype. No change in result.
>>>
>>> Then I tried with the Howler G729 codec:
>>> Setup 1: ?Xlite <-> Freeswitch <->Portech <->GSM Phone: No change in result.
>>> Setup 2: ?GSM Phone <-> Voip.MS <-> Freeswitch <->Portech <->GSM
>>> Phone: No change in result.
>>>
>>> In all these cases, the communication is perfect in one direction but
>>> very choppy in the orther.
>>> The only thing that works is direct Skype to Skype. And it works
>>> perfectly, suggesting that the connection is probably not the issue.
>>> The speed i could measure is 140kbs/30kbs.
>>>
>>> And assuming the the internet connection is the issue, i would have
>>> hoped that if Skype<->Skype works, then Skype<->Skypiax<->Skype would
>>> also work.
>>>
>>> Any idea?
>>>
>>> Max.
>>>
>>
>>
>>
>> --
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> Cell : +39-347-2665618
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From max.bridgewater at gmail.com  Fri Feb 12 12:01:54 2010
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Fri, 12 Feb 2010 12:01:54 -0800
Subject: [Freeswitch-users] Choppy connection in one direction
In-Reply-To: <7b197bef1002121140n1814c09fr70293e46ada06175@mail.gmail.com>
References: <c55d46951002120951o324c16bwd8b0da5c02b30721@mail.gmail.com>
	<7b197bef1002121110v31bcbf0bmaf46c3b54cd6645a@mail.gmail.com>
	<c55d46951002121117m67e24aa2y37c393161d318a93@mail.gmail.com>
	<7b197bef1002121140n1814c09fr70293e46ada06175@mail.gmail.com>
Message-ID: <c55d46951002121201x4a23565fxa44ec15e39c9a969@mail.gmail.com>

On Fri, Feb 12, 2010 at 11:40 AM, Giovanni Maruzzelli
<gmaruzz at celliax.org> wrote:
> Please, don't talk to aliens!
Why not? I could understand what they say, i would be at least curious
to see them.

>
> I mean, you have this issue any times you put FS in the middle, huh?

Yeah, it seems this appears only when FS is in the mix. Direct
connection to the Portech gateway is possible, but without G729 (or
something similar) on my softphones, it definitely sounds crappy.

Max.


From chrisg.lists at gmail.com  Fri Feb 12 12:14:26 2010
From: chrisg.lists at gmail.com (Chris Graham)
Date: Fri, 12 Feb 2010 22:14:26 +0200
Subject: [Freeswitch-users] Mod_IAX
Message-ID: <b8bb96081002121214y1b9ba3a3x6bcab6c53bce30c9@mail.gmail.com>

Hi All,

I have only recently gotten into freeswitch being a asterisk guy for 5 years
give or take. I love the freeswitch XML way of doing things. My question is
why was mod_iax dropped? Is opal the replacement? On the Wiki is says its
beta?

Thanks for the clarity.
Chris G.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100212/d4cfb18e/attachment.html 

From anthony.minessale at gmail.com  Fri Feb 12 12:37:00 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 12 Feb 2010 14:37:00 -0600
Subject: [Freeswitch-users] Mod_IAX
In-Reply-To: <b8bb96081002121214y1b9ba3a3x6bcab6c53bce30c9@mail.gmail.com>
References: <b8bb96081002121214y1b9ba3a3x6bcab6c53bce30c9@mail.gmail.com>
Message-ID: <191c3a031002121237y5de4af7fxd971164c79e29d20@mail.gmail.com>

mod_iax fell into disuse and nobody volunteered when a call to maintain it
was put on the community.
The short answer is that the IAX2 protocol changed somewhere over time in a
way that made the IAX library we were using crash unexpectedly.  We don't
have the resources to debug the 3rd party code in that IAX lib so we dropped
it to protect FreeSWITCH from unwanted crashes.


On Fri, Feb 12, 2010 at 2:14 PM, Chris Graham <chrisg.lists at gmail.com>wrote:

> Hi All,
>
> I have only recently gotten into freeswitch being a asterisk guy for 5
> years give or take. I love the freeswitch XML way of doing things. My
> question is why was mod_iax dropped? Is opal the replacement? On the Wiki is
> says its beta?
>
> Thanks for the clarity.
> Chris G.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100212/a29d8e32/attachment.html 

From peder at networkoblivion.com  Fri Feb 12 12:40:59 2010
From: peder at networkoblivion.com (Peder)
Date: Fri, 12 Feb 2010 14:40:59 -0600
Subject: [Freeswitch-users] Mod_IAX
In-Reply-To: <b8bb96081002121214y1b9ba3a3x6bcab6c53bce30c9@mail.gmail.com>
References: <b8bb96081002121214y1b9ba3a3x6bcab6c53bce30c9@mail.gmail.com>
Message-ID: <08dd01caac23$af4b9740$0de2c5c0$@com>

Lack of support.  Nobody wanted to take over development of it.

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Chris
Graham
Sent: Friday, February 12, 2010 2:14 PM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Mod_IAX

 

Hi All,

I have only recently gotten into freeswitch being a asterisk guy for 5 years
give or take. I love the freeswitch XML way of doing things. My question is
why was mod_iax dropped? Is opal the replacement? On the Wiki is says its
beta?

Thanks for the clarity.
Chris G.

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100212/d2061c32/attachment-0001.html 

From tculjaga at gmail.com  Fri Feb 12 12:46:11 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Fri, 12 Feb 2010 21:46:11 +0100
Subject: [Freeswitch-users] h323 - sip call is not working
In-Reply-To: <26C4E111-8329-48F8-A8DA-081B851A9514@freeswitch.org>
References: <20100212112427.0FC1F11F9E@mail.nstel.ru>
	<26C4E111-8329-48F8-A8DA-081B851A9514@freeswitch.org>
Message-ID: <65d96fc81002121246r48e867abp3c11f7f72a0ee906@mail.gmail.com>

On Fri, Feb 12, 2010 at 3:29 PM, Brian West <brian at freeswitch.org> wrote:

> This is a rather broad assumption.  I have seen 40ms, 60ms and even 80ms in
> the wild.  It all depends on what you want to do.  It lowers overhead and
> increases efficiency on the wire.
>
> /b
>
> On Feb 12, 2010, at 5:24 AM, Nikolay Kondratyev wrote:
>
> By the way, I know that one can use different packetization times for the
> same codec, but I?ve never heard, that somebody really uses 30 ms for
> G711Alaw. Always 20ms.
>
>
>
everything above 60 ms is a nonsense ... and ugly :) It screws your voice
quality not even thinking VBD (voice band data) over that line :).

Anyhow, Nikolay, your problem is broken IPO h323 stack and the know avaya
"flexibility" when interoping with other vendor equipments. Here IPO is
unable to negotiate a different framing size than the default and sadly this
is the core of the problem.

Please, can you send me two tcpdump captures of calls between IPO and FS:

   1. a capture with fast start & h245tunneling=true
   2. a captire with fast start & h245tunelling=true + h245inSetup


I just want to be sure of something.

T.



>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100212/82cdafaa/attachment.html 

From tculjaga at gmail.com  Fri Feb 12 14:07:59 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Fri, 12 Feb 2010 23:07:59 +0100
Subject: [Freeswitch-users] Mod_Conference capacity....
In-Reply-To: <1235523542.1100531265999125823.JavaMail.root@mail-2.01.com>
References: <892897350.1100311265999066927.JavaMail.root@mail-2.01.com>
	<1235523542.1100531265999125823.JavaMail.root@mail-2.01.com>
Message-ID: <65d96fc81002121407l5f4cbd3cw8af46337a13520c2@mail.gmail.com>

On Fri, Feb 12, 2010 at 7:25 PM, <joel.sisko at iconverged.com> wrote:

> Anthony thanks for the input. I am not looking for benchmark and understand
> the reasons why it does not make sense to do so since application usage and
> hardware will effect that benchmark.
>
> Looking for some successes by the group that they can share that would lead
> me to believe that using FreeSwitch would be worth the time and money to
> invest over our current conference solution. As an example, I can state that
> I have seen Yate used with 200 people in a single conference (all G711) work
> flawlessly, so does 200 count as "many" from your perspective or will
> FreeSwitch do that in its sleep?
>
>
Again, what is your problem with SIPP ?

I can say i used it wery well and reached incredible numbers ... I didn't
try conference but i did other things:

1. calls to FS that answered the call and played some prompts (wav -
transcoding needed) - 200 CPS and 2000 sim calls (limited!)  - a quad core
64bit proc (unfortunately AMD but :P)
2. calls to FS acting as an AS returning 302 messages - 480 CPS including
LDAP BD lookup in the background - a dualcore 64bit proc intel

There is not rule of thumb that can tell you what performance can you get.
It realy depends of the application you are running, the dialplan you are
using,  what HW (RAM, CPU, HDD/ramDISK...) are you running FS on...



The only way you kan know all of this is hit the limit of your platform
yourself (SIPP is an excelent tool for that, just play with it for a day or
two).... or pay someone to make this test for you within your enviorment.

Chears!
T.




> Thanks for the help.
>
>
> Joel
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100212/fd09cf01/attachment.html 

From gavin.henry at gmail.com  Fri Feb 12 14:25:19 2010
From: gavin.henry at gmail.com (Gavin Henry)
Date: Fri, 12 Feb 2010 22:25:19 +0000
Subject: [Freeswitch-users] Mod_Conference capacity....
In-Reply-To: <E7571BC2-135B-4743-B941-1A84968A4395@freeswitch.org>
References: <268119402.962161265928492648.JavaMail.root@mail-2.01.com>
	<715035894.962251265928530648.JavaMail.root@mail-2.01.com>
	<13ca621c1002111522v7fe55a62w84043ed986155b28@mail.gmail.com>
	<E7571BC2-135B-4743-B941-1A84968A4395@freeswitch.org>
Message-ID: <13ca621c1002121425s79d5e814h651aa7ce674e242a@mail.gmail.com>

Exactly, but he could setup his expected users via sipp and increase
participants whilst he is dialled in and wait until the audio gets
bad. A rough and ready test but give you some figures.

There's plenty of info if you search on the asterisks lists etc. Just
a thought, and yes you need to know wtf you're doing :-)

On 11/02/2010, Brian West <brian at freeswitch.org> wrote:
> Not an optimal test scenario unless you know wtf you're doing!
>
> /b
>
> On Feb 11, 2010, at 5:22 PM, Gavin Henry wrote:
>
>> Why don't use script a test or use sipp and then dial in yourself to
>> listen?
>>
>> Cheers.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>

-- 
Sent from my mobile device

http://www.suretecsystems.com/services/openldap/
http://www.suretectelecom.com


From gavin.henry at gmail.com  Fri Feb 12 14:36:56 2010
From: gavin.henry at gmail.com (Gavin Henry)
Date: Fri, 12 Feb 2010 22:36:56 +0000
Subject: [Freeswitch-users] Mod_Conference capacity....
In-Reply-To: <191c3a031002120944u346a3f7cq779f5c9e0ae31522@mail.gmail.com>
References: <191c3a031002120856q93e52cep14ee96f1e7ed6d1@mail.gmail.com>
	<324899354.1084631265995845519.JavaMail.root@mail-2.01.com>
	<191c3a031002120944u346a3f7cq779f5c9e0ae31522@mail.gmail.com>
Message-ID: <13ca621c1002121436n9b44dcexffeaafe998d4c88a@mail.gmail.com>

Hi,

I think this is unfair to automatically point a user to commercial
support. Its like holding your users hostage until they pay for info.

I'm willing to write a page on the wiki with info on the recommended
tools to use etc. with links to the commercial support resource list
if they can't be bothered to do the setup themselves or want the
experts paid for guidance.

It just reads like a bit of an insult. Have a question? Well you must
be stupid and need to pay for help immediately.

He may have even written the page for the docs team if he was
encouraged first with a few pointers on how to do the tests and
written them up with results for others to find with a big YMMV
warning.

Just my thoughts from experiences as the Doc team lead for the
OpenLDAP project and dealing with lots of users asking questions too.

But some users never read docs and there a very good amount of high
quality pages in the wiki. Others, if helped, come back and
contribute, but you may have lost this one.

Gav.

On 12/02/2010, Anthony Minessale <anthony.minessale at gmail.com> wrote:
> Producing benchmark numbers for an opensource project is a cardinal sin.
> I can safely tell you that it is "many" and very competitive with anything
> else you will encounter as long as you use a modern multi-core 64bit
> machine.
>
> As I said there is commercial support available which is customary for
> anyone needing assistance setting up a company.
>
>
> On Fri, Feb 12, 2010 at 11:30 AM, Joel Sisko
> <joel.sisko at iconverged.com>wrote:
>
>> So if I understand correctly that there are no numbers to be considered
>> from anyone at this point in regards to capacity as a general guideline?
>>
>>
>> Kindest regards,
>>
>> Joel
>> ----- Original Message -----
>> From: "Anthony Minessale" <anthony.minessale at gmail.com>
>> To: freeswitch-users at lists.freeswitch.org
>> Sent: Friday, February 12, 2010 8:56:31 AM GMT -08:00 US/Canada Pacific
>> Subject: Re: [Freeswitch-users] Mod_Conference capacity....
>>
>> consider commercial support from consulting at freeswitch.org
>>
>>
>> On Fri, Feb 12, 2010 at 10:33 AM, Joel Sisko
>> <joel.sisko at iconverged.com>wrote:
>>
>>> Brian/Gavin do you know of any resource/link that can give an indication
>>> on what we could be expected of conference capacity? We are trying to
>>> determine a few different platforms we can possible use for conferencing
>>> and
>>> doing a little preliminary homework.
>>>
>>> Kindest regards,
>>>
>>> Joel
>>>
>>>
>>> ----- Original Message -----
>>> From: "Joel Sisko" <joel.sisko at iconverged.com>
>>> To: freeswitch-users at lists.freeswitch.org
>>> Sent: Thursday, February 11, 2010 4:05:58 PM GMT -08:00 US/Canada Pacific
>>> Subject: Re: [Freeswitch-users] Mod_Conference capacity....
>>>
>>> Brian/Gavin thanks for the input. But I agree with Brian, if were that
>>> easy I would have done it prior to the post.
>>>
>>> Just looking to find out what some of the communities success has been to
>>> see if this is a path we should go down for a conference solution
>>> platform.
>>>
>>> Joel
>>> ----- Original Message -----
>>> From: "Brian West" <brian at freeswitch.org>
>>> To: freeswitch-users at lists.freeswitch.org
>>> Sent: Thursday, February 11, 2010 3:47:46 PM GMT -08:00 US/Canada Pacific
>>> Subject: Re: [Freeswitch-users] Mod_Conference capacity....
>>>
>>> Not an optimal test scenario unless you know wtf you're doing!
>>>
>>> /b
>>>
>>> On Feb 11, 2010, at 5:22 PM, Gavin Henry wrote:
>>>
>>> > Why don't use script a test or use sipp and then dial in yourself to
>>> listen?
>>> >
>>> > Cheers.
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>> _______________________________________________ FreeSWITCH-users mailing
>> list FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>

-- 
Sent from my mobile device

http://www.suretecsystems.com/services/openldap/
http://www.suretectelecom.com


From joseph.puchalski at personalcyberspace.com  Fri Feb 12 15:35:04 2010
From: joseph.puchalski at personalcyberspace.com (Joseph Puchalski)
Date: Fri, 12 Feb 2010 23:35:04 +0000
Subject: [Freeswitch-users] How to set outbound caller id info for multiple
	users/extensions
Message-ID: <093DD565390C1E4FB15D7B383E86BB05AF05AB@Goose.personalcyberspace.net>

I'm having problems setting different outbound caller id info for different extensions/users.

I've set up a small system with two active users. I set up my users by copying and modifying existing entries from the dialplan files that come with freeSWITCH

Here are my two extensions:

<include>
  <user id="5859">
    <params>
      <param name="password" value="xxxxxx"/>
      <param name="vm-password" value="xxxxxx"/>
    </params>
    <variables>
      <variable name="toll_allow" value="domestic,international,local"/>
      <variable name="accountcode" value="5859"/>
      <variable name="user_context" value="default"/>
      <variable name="effective_caller_id_name" value="5859"/>
      <variable name="effective_caller_id_number" value="5859"/>
      <variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/>
      <variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/>
      <variable name="callgroup" value="techsupport"/>
    </variables>
  </user>
</include>

<include>
  <user id="5515">
    <params>
      <param name="password" value="xxxxxxx"/>
      <param name="vm-password" value="xxxxxxx"/>
    </params>
    <variables>
      <variable name="toll_allow" value="domestic,international,local"/>
      <variable name="accountcode" value="5515"/>
      <variable name="user_context" value="default"/>
      <variable name="effective_caller_id_name" value="5515"/>
      <variable name="effective_caller_id_number" value="5515"/>
      <variable name="outbound_caller_id_name" value="Jane Doe"/>
      <variable name="outbound_caller_id_number" value="5558395515"/>
      <variable name="callgroup" value="techsupport"/>
    </variables>
  </user>
</include>

These extensions are in files named 5859.xml and 5515.xml respectively.

I'm using a SIP trunk from Vitelity (in and out) with two DIDs corresponding to the extensions above.

Inbound and outbound calling work as needed with one exception: Calls originating from user/extension 5515 go out with the caller ID of extension/user 5859.

Extension 5859 was the first that I created.

Where should I be setting the outbound caller id number for my second extension? I've been trying to track this down in the available documentation but have been unable to do so.

I apologize ahead of time if this is answered somewhere obvious that I've missed.

Thanks for any help.

Joe (FreeSWITCH newbie)
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100212/c836abc9/attachment-0001.html 

From mcampbellsmith at gmail.com  Fri Feb 12 16:29:25 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Sat, 13 Feb 2010 11:29:25 +1100
Subject: [Freeswitch-users] Forbidden using UDP, works with TCP/TLS
In-Reply-To: <0A905834-850F-4D1F-AEC5-D3A87CE4141B@jerris.com>
References: <33c87fa31002101824h21e3bf7lb13b68e8d4fd36a2@mail.gmail.com>
	<2EF27EFC-1634-46A3-B438-2EE99BEF58F1@jerris.com>
	<33c87fa31002110131w3cf6ad00n802669856085a879@mail.gmail.com>
	<CA21231C-108B-4445-A3B9-BF8D94FFE898@jerris.com>
	<33c87fa31002110213q1de0eebfsb950d8725ea9aba@mail.gmail.com>
	<0A905834-850F-4D1F-AEC5-D3A87CE4141B@jerris.com>
Message-ID: <33c87fa31002121629k74f26807i131841aa4252461b@mail.gmail.com>

This is the working one with TLS... does this shed any light on why
this user can not register using UDP?

   ------------------------------------------------------------------------
   REGISTER sip:mydns.dyndns.org:442 SIP/2.0
   Via: SIP/2.0/TLS 121.xxx.xxx.xxx:10371;branch=z9hG4bK-9465a0f9;rport
   From: 2000 <sip:2000 at mydns.dyndns.org:442>;tag=34ac954c7d2abf51o0
   To: 2000 <sip:2000 at mydns.dyndns.org:442>
   Call-ID: 8657c383-fea70d0a at 10.0.0.1
   CSeq: 31055 REGISTER
   Max-Forwards: 70
   Authorization: Digest
username="2000",realm="mydns.dyndns.org",nonce="5000f49a-1836-11df-985e-e77ba7a22ac3",uri="sip:mydns.dyndns.org:442",algorithm=MD5,response="2bad1d5fadbb0465b0a513352db0292b",qop=auth,nc=00000001,cnonce="6951ee10"
   Contact: 2000 <sip:2000 at 121.xxx.xxx.xxx:10371;transport=tls>;expires=1800
   User-Agent: Linksys/SPA3102-5.1.10(GW)
   Content-Length: 0
   Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
   Supported: x-sipura, replaces

   ------------------------------------------------------------------------
tport_deliver(0xb6d07cd8): msg 0xb6da3cc0 (795 bytes) from
tls/121.xxx.xxx.xxx:10371/sips next=(nil)
nta: received REGISTER sip:mydns.dyndns.org:442 SIP/2.0 (CSeq 31055)
nta: REGISTER (31055) going to a default leg
nua: nua_stack_process_request: entering
nua: nh_create: entering
nua: nh_create_handle: entering
nua: nua_stack_set_params: entering
soa_clone(static::0x97cc698, 0x9794808, 0xb6d83680) called
soa_set_params(static::0xb6d77020, ...) called
nua(0xb6d83680): event i_register 100 Trying
nua: nua_application_event: entering
nua: nua_respond: entering
nua(0xb6d83680): sent signal r_respond
nua: nua_handle_destroy: entering
nua(0xb6d83680): sent signal r_destroy
nua: nua_handle_magic: entering
nua: nua_handle_destroy: entering
tport(0xb6d07cd8): reset timer
nua(0xb6d83680): recv signal r_respond 200 OK
nua: nua_stack_set_params: entering
soa_set_params(static::0xb6d77020, ...) called
tport_tsend(0xb6d07cd8) tpn = TLS/121.xxx.xxx.xxx:10371
tport_tls_writevec: vec 0xb6d2eb80 0xb6da4270 92 (92)
tport_tls_writevec: vec 0xb6d2eb80 0xb6ded80a 76 (76)
tport_tls_writevec: vec 0xb6d2eb80 0xb6da42cc 69 (69)
tport_tls_writevec: vec 0xb6d2eb80 0xb6ded889 59 (59)
tport_tls_writevec: vec 0xb6d2eb80 0xb6da4311 329 (329)
tport_vsend(0xb6d07cd8): 625 bytes of 625 to tls/121.xxx.xxx.xxx:10371
tport_vsend returned 625
send 625 bytes to tls/[121.xxx.xxx.xxx]:10371 at 00:25:39.795356:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK

On Fri, Feb 12, 2010 at 9:22 AM, Michael Jerris <mike at jerris.com> wrote:
> how is this different from the working one?
>
> Mike
>
> On Feb 11, 2010, at 5:13 AM, Mark Campbell-Smith wrote:
>
>> ah thats true... The trace is not too readable to me, but may give
>> some insight to someone that can read the sofia logs....
>>
>>
>> recv 752 bytes from udp/[121.xxx.xxx.xxx]:5060 at 10:10:37.803288:
>> ? ------------------------------------------------------------------------
>> ? REGISTER sip:mydns.dyndns.org SIP/2.0
>> ? Via: SIP/2.0/UDP 121.xxx.xxx.xxx:5060;branch=z9hG4bK-9052c91f
>> ? From: 2000 <sip:2000 at mydns.dyndns.org>;tag=7a9dbbbfa691136do0
>> ? To: 2000 <sip:2000 at mydns.dyndns.org>
>> ? Call-ID: 610db38-dd3b511f at 121.xxx.xxx.xxx
>> ? CSeq: 32330 REGISTER
>> ? Contact: 2000 <sip:2000 at 121.xxx.xxx.xxx:5060>;expires=900
>> ? Authorization: Digest username="2000", realm="mydns.dyndns.org",
>> nonce="b3298cfe-16f5-11df-9734-e77ba7a22ac3",
>> uri="sip:mydns.dyndns.org",
>> response="724fca542ce08d3f12b9ba1043bebb0c", algorithm=MD5,
>> qop="1225e2f1"
>> ? Max-Forwards: 70
>> ? User-Agent: Linksys/SPA3102-5.1.10(GW)
>> ? Supported: x-sipura
>> ? Supported: replaces
>> ? Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
>> ? Content-Length: 0
>>
>>
>> ? ------------------------------------------------------------------------
>> tport_deliver(0x97cde80): msg 0x98297e8 (752 bytes) from
>> udp/121.xxx.xxx.xxx:5060/sip next=(nil)
>> nta: received REGISTER sip:mydns.dyndns.org SIP/2.0 (CSeq 32330)
>> nta: REGISTER (32330) going to a default leg
>> nua: nua_stack_process_request: entering
>> nua: nh_create: entering
>> nua: nh_create_handle: entering
>> nua: nua_stack_set_params: entering
>> soa_clone(static::0x97cc698, 0x9794808, 0x981cb70) called
>> soa_set_params(static::0x9758ba8, ...) called
>> nua(0x981cb70): event i_register 100 Trying
>> nua: nua_application_event: entering
>> nua: nua_respond: entering
>> nua(0x981cb70): sent signal r_respond
>> nua: nua_handle_destroy: entering
>> nua(0x981cb70): sent signal r_destroy
>> nua: nua_handle_magic: entering
>> nua: nua_handle_destroy: entering
>> nua(0x981cb70): recv signal r_respond 403 Forbidden
>> nua: nua_stack_set_params: entering
>> soa_set_params(static::0x9758ba8, ...) called
>> tport_tsend(0x97cde80) tpn = UDP/121.xxx.xxx.xxx:5060
>> tport_resolve addrinfo = 121.xxx.xxx.xxx:5060
>> tport_by_addrinfo(0x97cde80): not found by name UDP/121.xxx.xxx.xxx:5060
>> tport_vsend(0x97cde80): 495 bytes of 495 to udp/121.xxx.xxx.xxx:5060
>> tport_vsend returned 495
>> send 495 bytes to udp/[121.xxx.xxx.xxx]:5060 at 10:10:37.812955:
>> ? ------------------------------------------------------------------------
>> ? SIP/2.0 403 Forbidden
>>
>> On Thu, Feb 11, 2010 at 8:57 PM, Michael Jerris <mike at jerris.com> wrote:
>>> you can crank up the sofia loglevel as well
>>>
>>> Mike
>>>
>>> On Feb 11, 2010, at 4:31 AM, Mark Campbell-Smith wrote:
>>>
>>>> I don't see debug logs (fsctl debug_level 9 or loglevel 9) during the
>>>> registration process.
>>>>
>>>> All I see is the sip messages when the sip trace is activated (403 Forbidden)
>>>>
>>>> Is there other debugging that I can enable?
>>>>
>>>> On Thu, Feb 11, 2010 at 4:25 PM, Michael Jerris <mike at jerris.com> wrote:
>>>>> Look at the FreeSWITCH debug logs and compare the differences, they should tell you why.
>>>>>
>>>>> Mike
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From wiltingtree at gmail.com  Fri Feb 12 19:04:01 2010
From: wiltingtree at gmail.com (Adam Wilt)
Date: Fri, 12 Feb 2010 22:04:01 -0500
Subject: [Freeswitch-users] Bridging sessions from two separate lua
	scripts; uuid_bridge? intercept?
In-Reply-To: <D753EDBD-6ED7-43C2-B5F5-A899E3C3D3E2@jerris.com>
References: <eb7797d71002071831w27301c5dva767b82b38c2ce6f@mail.gmail.com>
	<C23AA196-E851-4D5B-B1B5-2C2F32F6BB68@freeswitch.org>
	<eb7797d71002071929pb53e1adue2e91e6bdb598251@mail.gmail.com>
	<eb7797d71002080737k3f8fa5c9m8bf3ef2fcf42881e@mail.gmail.com>
	<D753EDBD-6ED7-43C2-B5F5-A899E3C3D3E2@jerris.com>
Message-ID: <eb7797d71002121904v5bdfac87w2684c461ca1770cc@mail.gmail.com>

Thanks again for the help Michael.

I'm now upgraded to version 1.5, but I'm still getting the same
problem. When I try to bridge sessions from two separate lua scripts,
both sessions hang up on me. I think maybe I don't understand how
"intercept" works.
Anyway, I posted the debug trace here:

                      http://pastebin.freeswitch.org/12121

And I also put together a small example which exhibits the problem.
The first script is started by an inbound call and starts the second
script. The second script places an outbound call and tries to bridge
the two sessions together:

Inbound script:  http://pastebin.freeswitch.org/12122
Outbound script: http://pastebin.freeswitch.org/12123

Thanks,
Adam




On 2/9/10, Michael Jerris <mike at jerris.com> wrote:
> 1.4?  how does the future look, report back?
>
> http://files-sync.freeswitch.org/windows_installer/freepbx_svn.exe
>
> I think this has latest FreeSWITCH in it to, Carlos, can you confirm that?
>
> Mike
>
> On Feb 8, 2010, at 10:37 AM, Adam Wilt wrote:
>
>> One other thing I should mention. I'm running FreeSWITCH version 1.4
>> (build 14460) in Windows.
>> Brian suggested I upgrade to the build in the
>> http://files-sync.freeswitch.org/windows_installer/  folder, but it turned
>> out to be the exact same build I already had. I'd love to try upgrade to
>> 1.5 in case this problem has been fixed already.
>>
>>
>> On Sun, Feb 7, 2010 at 10:29 PM, Adam Wilt <wiltingtree at gmail.com> wrote:
>> Thanks Michael for the reply.
>> Here's the pastebin link: http://pastebin.freeswitch.org/12084
>>
>>
>> On Sun, Feb 7, 2010 at 9:50 PM, Michael S Collins <msc at freeswitch.org>
>> wrote:
>> Pastebin a debug log so we can see what is happening when the script runs.
>>
>>
>> -MC
>>
>> Sent from my iPhone
>>
>> On Feb 7, 2010, at 8:31 PM, Adam Wilt <wiltingtree at gmail.com> wrote:
>>
>>> Hi. I have two sessions running in two separate Lua scripts, and I want
>>> to bridge them so that the bridged call is being controlled by the first
>>> (a-leg) script.
>>> If I simply use uuid_bridge, I get no error but the calls don't bridge.
>>> I've tried intercept, but I don't understand how it should be used;
>>> nothing I try seems to work.
>>> Here's what I have:
>>>
>>> function bridge_calls(session,api,b_leg_uuid, call_len)
>>>   session:setAutoHangup(false)
>>>   session:execute("sched_hangup","+" .. tostring(call_len) .. " " ..
>>> tostring(session.uuid))
>>>   session:execute("set","continue_on_fail=true")
>>>   api:executeString("intercept -bleg " .. tostring(b_leg_uuid))
>>>   api:executeString("uuid_bridge " .. tostring(session.uuid) .. " " ..
>>> tostring(b_leg_uuid))
>>> end
>>>
>>> I'd really appreciate any help.
>>>
>>> Thanks,
>>> Adam
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>


From jason at jasonjgw.net  Fri Feb 12 19:08:12 2010
From: jason at jasonjgw.net (Jason White)
Date: Sat, 13 Feb 2010 14:08:12 +1100
Subject: [Freeswitch-users] FreeSWITCH Debian packages - build issues
Message-ID: <20100213030812.GA19108@jdc.jasonjgw.net>

Has anyone successfully built the Debian packages recently from the source
repository?

The problem I'm experiencing is that openzap is specified to be built, but it
is never actually compiled. Consequently, the packages can't be created (the
process fails due to the missing mod_openzap.so file).

I don't need openzap; I can easily comment it out, but I also think the
supplied package files should work as is.

first step: confirm whether my experience under Debian Sid is shared by others
using different versions of Debian or Ubuntu.



From woodydickson at gmail.com  Fri Feb 12 23:14:48 2010
From: woodydickson at gmail.com (Woody Dickson)
Date: Fri, 12 Feb 2010 23:14:48 -0800
Subject: [Freeswitch-users] transmission status of channel
In-Reply-To: <8e9d67561002121010p37b16aceyb06cda0420e1aa99@mail.gmail.com>
References: <a44688a1002120732v7d95a6f9x2d4bd7ed68de59eb@mail.gmail.com>
	<3028D6AC-46F2-4291-BAA8-E49C9347917B@freeswitch.org>
	<b39273531002120952u698ab5d1x799696b052ced9a9@mail.gmail.com>
	<8e9d67561002121010p37b16aceyb06cda0420e1aa99@mail.gmail.com>
Message-ID: <a44688a1002122314t4060f29ehce920077ee90704b@mail.gmail.com>

Does anyone know where to find those RTCP info from the core rtp stack?

On Fri, Feb 12, 2010 at 10:10 AM, Timur Valishev <tim at novion.ru> wrote:
> It would be very nice if FS pass RTCP information to channel vars...
>
> Best regards,
> Timur Valishev
>
> 2010/2/12 Rupa Schomaker <rupa at rupa.com>:
>> I'm pretty sure that info doesn't exist. ?Don't we need RTCP (plus
>> infrastructure for measuring) for this?
>>
>> On Fri, Feb 12, 2010 at 11:22 AM, Michael S Collins <msc at freeswitch.org>
>> wrote:
>>>
>>> Check in the XML cdrs. I'm on a plan right now so I can't easily point
>>> you to a specific wiki page. :)
>>>
>>> -MC
>>>
>>> Sent from my iPhone
>>>
>>> On Feb 12, 2010, at 8:32 AM, Woody Dickson <woodydickson at gmail.com>
>>> wrote:
>>>
>>> > Hi,
>>> >
>>> > Is there anyway to obtain the %packet lost, latency, and jitter info
>>> > for each channel?
>>> >
>>> > Any idea how to obtain those information?
>>> >
>>> > thx,
>>> > Woody
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>>> > users
>>> > http://www.freeswitch.org
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>>
>> --
>> -Rupa
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From mailinglist at fribert.dk  Sat Feb 13 00:54:30 2010
From: mailinglist at fribert.dk (mailinglist)
Date: Sat, 13 Feb 2010 09:54:30 +0100
Subject: [Freeswitch-users] Svar: Re:  Need help setting up a feature
In-Reply-To: <b39273531002110528p42218d07q95e97485139315f1@mail.gmail.com>
References: <4B73A799020000E100000470@mail.fribert.dk>
	<b39273531002110528p42218d07q95e97485139315f1@mail.gmail.com>
Message-ID: <4B7676D6020000E100000481@mail.fribert.dk>

Hi Rupa
 
I've got similar settings here, but I can't make them work, nothing happens when I press R:
 
The settings are
DTMP over VoIP connections: Send settings [ ]Auto   [X] Audio  [ ] RFC 2833 [X] SIP Info
 
Call Transfer
Use the R key to initiate call     ( ) Yes   (X) No
transfer with the SIP Refer Method
 
Transfer Call by On-Hook        (X) Yes   ( ) No
 
Derive Target address             ( ) from SIP URL   (X)   from SIP Contact Header
 
Find Target address automatically    ( ) Yes   (X) No
 
Hook Flash  (R-key)
 
Application Type:   dtmf-relay
Application Signal: 16
 
That's all the settings, I had to set 'send settings' from auto to audio+SIP Info otherwise the R-Flash settings were disabled.

 
But all in all, I think a *1 or something during the call would be a better method, because I should be able to make that work on al types of phones, right?
 
 
Best regards
Fribse
 
/Running Freeswitch on pfSense running on a VMWare ESXi 4.0

>>> 11-02-2010 kl. 14:28 skrev Rupa Schomaker <rupa at rupa.com> i meddelelsen <b39273531002110528p42218d07q95e97485139315f1 at mail.gmail.com>:

My Siemens A580 has options for controlling the R key. It seems that you can either have it setup for transfer or as a hook flash. Default is as a transfer key.

I haven't succeeded in getting it to work for transfer and it is wayyyyy down low on my list of things to do with the phone.

>From the web UI:


Call Transfer
Use the R key to initiate call transfer with the SIP Refer method.:Yes No 
Transfer Call by On-Hook:Yes No 
Derive target address:from SIP URL from SIP contact header 
Find target addr. automatically:Yes No 

Hook Flash (R-key)
R key settings are disabled because the R key is being used for call transfer. 

On Wed, Feb 10, 2010 at 11:45 PM, mailinglist <mailinglist at fribert.dk> wrote:


Sorry for the repost, but the previous thread just died :-)
I'm trying to get the possibility of transfering an incoming call from one extension to another, and give the possibility of turning it into a conference.
I don't have a 'transfer' button.
I do have an 'R' button on the Siemens handsets, and a 'Flash' button on the Sipura. The 'Flash' button gives me a new dialtone, gives the caller MOH, and then I can dial the new extension, and transfer the call, but not create a conference.
But the Siemens handset does not have a 'flash', and pressing the R doesn't do anything.
It might be two different features 'transfer' and 'conference'...

But I thought that using the bind_meta_app would accomplish both.
It's on an incoming call from the outside.
So the situation:
The Public folder has an entry that matches the dialed number, and does a transfer to 8202.
Then the dialplan matches the 8202 with a group, and the phone rings.
Somebody picks it up, finds out that it needs to be transferred to another extension, or transferred to a conference with a second extension.
How do I construct that?
Best regards
Fribse
/Running Freeswitch on pfSense running on a VMWare ESXi 4.0

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org





-- 
-Rupa
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100213/2df30d3b/attachment-0001.html 

From errotan at gmail.com  Sat Feb 13 00:57:11 2010
From: errotan at gmail.com (=?iso-8859-1?q?Pusk=E1s_Zsolt?=)
Date: Sat, 13 Feb 2010 09:57:11 +0100
Subject: [Freeswitch-users] FreeSWITCH Debian packages - build issues
In-Reply-To: <20100213030812.GA19108@jdc.jasonjgw.net>
References: <20100213030812.GA19108@jdc.jasonjgw.net>
Message-ID: <201002130957.11633.errotan@gmail.com>

2010. febru?r 13. 04.08.12 Jason White d?tummal ezt ?rta:
> Has anyone successfully built the Debian packages recently from the source
> repository?
> 
> The problem I'm experiencing is that openzap is specified to be built, but
>  it is never actually compiled. Consequently, the packages can't be created
>  (the process fails due to the missing mod_openzap.so file).
> 
> I don't need openzap; I can easily comment it out, but I also think the
> supplied package files should work as is.
> 
> first step: confirm whether my experience under Debian Sid is shared by
>  others using different versions of Debian or Ubuntu.
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 

Openzap compiles without errors on Debian "testing" amd64 for me.
Svn version: 16625 


From jason at jasonjgw.net  Sat Feb 13 01:09:07 2010
From: jason at jasonjgw.net (Jason White)
Date: Sat, 13 Feb 2010 20:09:07 +1100
Subject: [Freeswitch-users] FreeSWITCH Debian packages - build issues
In-Reply-To: <201002130957.11633.errotan@gmail.com>
References: <20100213030812.GA19108@jdc.jasonjgw.net>
	<201002130957.11633.errotan@gmail.com>
Message-ID: <20100213090907.GA29452@jdc.jasonjgw.net>

Pusk?s Zsolt <errotan at gmail.com> wrote:
 
> Openzap compiles without errors on Debian "testing" amd64 for me.
> Svn version: 16625 

Thanks. Was that with the package build?

It seems I have local modifications here that I'd forgotten about. I'll
investigate further.



From errotan at gmail.com  Sat Feb 13 02:09:35 2010
From: errotan at gmail.com (=?iso-8859-1?q?Pusk=E1s_Zsolt?=)
Date: Sat, 13 Feb 2010 11:09:35 +0100
Subject: [Freeswitch-users] FreeSWITCH Debian packages - build issues
In-Reply-To: <20100213090907.GA29452@jdc.jasonjgw.net>
References: <20100213030812.GA19108@jdc.jasonjgw.net>
	<201002130957.11633.errotan@gmail.com>
	<20100213090907.GA29452@jdc.jasonjgw.net>
Message-ID: <201002131109.35877.errotan@gmail.com>

2010. febru?r 13. 10.09.07 Jason White d?tummal ezt ?rta:
> Pusk?s Zsolt <errotan at gmail.com> wrote:
> > Openzap compiles without errors on Debian "testing" amd64 for me.
> > Svn version: 16625
> 
> Thanks. Was that with the package build?
> 
> It seems I have local modifications here that I'd forgotten about. I'll
> investigate further.
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 

I just compiled fs using defaults (just uncommented the openzap line in 
modues.conf). I don't know how to build a package for that but I can try if 
you got some instructions how to do that for testing purposes.


From vetali100 at gmail.com  Sat Feb 13 02:47:51 2010
From: vetali100 at gmail.com (Vitalii Colosov)
Date: Sat, 13 Feb 2010 12:47:51 +0200
Subject: [Freeswitch-users] Other party does not hear voice when
	connecting with X-Lite
In-Reply-To: <2F81A680-6A5C-4E3F-9590-FFF6AC4EA80D@freeswitch.org>
References: <a858e7691002120133q5e4c806fx309517e4fdec45c5@mail.gmail.com>
	<2F81A680-6A5C-4E3F-9590-FFF6AC4EA80D@freeswitch.org>
Message-ID: <a858e7691002130247s439177acg23d6a41de3a9d21@mail.gmail.com>

Thanks for the advice!

Yes, x-lite and FS are behind the nat.

I changed x-lite settings from "Discover global IP" to "Use Local IP
adress", but this did not solve the problem.

I would like to add that when x-lite is configured directly to international
gateway sip proxy, it is working fine.
Only when it is connected via FS, voice is missing...

There is definetely something wrong with FS configuration...

Do you know if I can enable some debug level that will provide me some
useful information about voice part...?

Thank you!

Vitalie



2010/2/12 Brian West <brian at freeswitch.org>

> is x-lite behind nat with the freeswitch box?  If so you'll need to disable
> the discover global IP so that it doesn't try to hair pin thru your NAT
> router.... Most nat routers won't work correctly trying to do that.
>
> /b
>
> On Feb 12, 2010, at 3:33 AM, Vitalii Colosov wrote:
>
> >
> > But when I connect using X-Lite, it connects but other party cannot hear
> me.
> > I can hear him well...
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100213/41cf968c/attachment.html 

From tculjaga at gmail.com  Sat Feb 13 03:27:11 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Sat, 13 Feb 2010 12:27:11 +0100
Subject: [Freeswitch-users] Mod_Conference capacity....
In-Reply-To: <13ca621c1002121436n9b44dcexffeaafe998d4c88a@mail.gmail.com>
References: <191c3a031002120856q93e52cep14ee96f1e7ed6d1@mail.gmail.com>
	<324899354.1084631265995845519.JavaMail.root@mail-2.01.com>
	<191c3a031002120944u346a3f7cq779f5c9e0ae31522@mail.gmail.com>
	<13ca621c1002121436n9b44dcexffeaafe998d4c88a@mail.gmail.com>
Message-ID: <65d96fc81002130327q66d8227ao29ae46f8c7ee556a@mail.gmail.com>

On Fri, Feb 12, 2010 at 11:36 PM, Gavin Henry <gavin.henry at gmail.com> wrote:

> Hi,
>
> I think this is unfair to automatically point a user to commercial
> support. Its like holding your users hostage until they pay for info.
>
>
well, the point is that every application is different and nobody can say
what the performance for this or that HW exactly is. This is not like
holding a hostage it is more like "we need to play on your existing platform
run some tests so we can come out with some real benchmark"... this is what
i can read from Anthony's e-mail.


I'm willing to write a page on the wiki with info on the recommended
> tools to use etc. with links to the commercial support resource list
> if they can't be bothered to do the setup themselves or want the
> experts paid for guidance.
>
>

yes, SIPP together with nmon and wireshark and Adobe Audition are the right
tools (at least thats what i'm using...) for such benchmarking.


   - SIPP to generate traffic load,
   - nmon to get some real stats on the machine
   - wireshark to quickly check the jitter on your test call and extract the
   voice stream (sniffing has to be done on a mirrored switch port)
   - adobe autition to perform voice quality check on the extracted voice
   stream




> It just reads like a bit of an insult. Have a question? Well you must
> be stupid and need to pay for help immediately.
>
>
dont think so, "you must learn how to use all these tools"... if you don't
want to do it, there is another option but it is not free :(



> He may have even written the page for the docs team if he was
> encouraged first with a few pointers on how to do the tests and
> written them up with results for others to find with a big YMMV
> warning.
>

well ... i think the time is always the issue + nobody likes documenting
things :(


>
> Just my thoughts from experiences as the Doc team lead for the
> OpenLDAP project and dealing with lots of users asking questions too.
>
> But some users never read docs and there a very good amount of high
> quality pages in the wiki. Others, if helped, come back and
> contribute, but you may have lost this one.
>
>
I cannot say anything here except you are right and i agree what you are
saying about documenting things ...



> Gav.
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100213/ea836fdf/attachment.html 

From mcampbellsmith at gmail.com  Sat Feb 13 03:43:41 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Sat, 13 Feb 2010 22:43:41 +1100
Subject: [Freeswitch-users] Forbidden using UDP, works with TCP/TLS
In-Reply-To: <33c87fa31002121629k74f26807i131841aa4252461b@mail.gmail.com>
References: <33c87fa31002101824h21e3bf7lb13b68e8d4fd36a2@mail.gmail.com>
	<2EF27EFC-1634-46A3-B438-2EE99BEF58F1@jerris.com>
	<33c87fa31002110131w3cf6ad00n802669856085a879@mail.gmail.com>
	<CA21231C-108B-4445-A3B9-BF8D94FFE898@jerris.com>
	<33c87fa31002110213q1de0eebfsb950d8725ea9aba@mail.gmail.com>
	<0A905834-850F-4D1F-AEC5-D3A87CE4141B@jerris.com>
	<33c87fa31002121629k74f26807i131841aa4252461b@mail.gmail.com>
Message-ID: <33c87fa31002130343k7a01e9abt9c33360399f246f8@mail.gmail.com>

More testing. The device registers successfully to my SIP provider
directly using UDP - why would FS be rejecting the registration
request?



On Sat, Feb 13, 2010 at 11:29 AM, Mark Campbell-Smith
<mcampbellsmith at gmail.com> wrote:
> This is the working one with TLS... does this shed any light on why
> this user can not register using UDP?
>
> ? ------------------------------------------------------------------------
> ? REGISTER sip:mydns.dyndns.org:442 SIP/2.0
> ? Via: SIP/2.0/TLS 121.xxx.xxx.xxx:10371;branch=z9hG4bK-9465a0f9;rport
> ? From: 2000 <sip:2000 at mydns.dyndns.org:442>;tag=34ac954c7d2abf51o0
> ? To: 2000 <sip:2000 at mydns.dyndns.org:442>
> ? Call-ID: 8657c383-fea70d0a at 10.0.0.1
> ? CSeq: 31055 REGISTER
> ? Max-Forwards: 70
> ? Authorization: Digest
> username="2000",realm="mydns.dyndns.org",nonce="5000f49a-1836-11df-985e-e77ba7a22ac3",uri="sip:mydns.dyndns.org:442",algorithm=MD5,response="2bad1d5fadbb0465b0a513352db0292b",qop=auth,nc=00000001,cnonce="6951ee10"
> ? Contact: 2000 <sip:2000 at 121.xxx.xxx.xxx:10371;transport=tls>;expires=1800
> ? User-Agent: Linksys/SPA3102-5.1.10(GW)
> ? Content-Length: 0
> ? Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
> ? Supported: x-sipura, replaces
>
> ? ------------------------------------------------------------------------
> tport_deliver(0xb6d07cd8): msg 0xb6da3cc0 (795 bytes) from
> tls/121.xxx.xxx.xxx:10371/sips next=(nil)
> nta: received REGISTER sip:mydns.dyndns.org:442 SIP/2.0 (CSeq 31055)
> nta: REGISTER (31055) going to a default leg
> nua: nua_stack_process_request: entering
> nua: nh_create: entering
> nua: nh_create_handle: entering
> nua: nua_stack_set_params: entering
> soa_clone(static::0x97cc698, 0x9794808, 0xb6d83680) called
> soa_set_params(static::0xb6d77020, ...) called
> nua(0xb6d83680): event i_register 100 Trying
> nua: nua_application_event: entering
> nua: nua_respond: entering
> nua(0xb6d83680): sent signal r_respond
> nua: nua_handle_destroy: entering
> nua(0xb6d83680): sent signal r_destroy
> nua: nua_handle_magic: entering
> nua: nua_handle_destroy: entering
> tport(0xb6d07cd8): reset timer
> nua(0xb6d83680): recv signal r_respond 200 OK
> nua: nua_stack_set_params: entering
> soa_set_params(static::0xb6d77020, ...) called
> tport_tsend(0xb6d07cd8) tpn = TLS/121.xxx.xxx.xxx:10371
> tport_tls_writevec: vec 0xb6d2eb80 0xb6da4270 92 (92)
> tport_tls_writevec: vec 0xb6d2eb80 0xb6ded80a 76 (76)
> tport_tls_writevec: vec 0xb6d2eb80 0xb6da42cc 69 (69)
> tport_tls_writevec: vec 0xb6d2eb80 0xb6ded889 59 (59)
> tport_tls_writevec: vec 0xb6d2eb80 0xb6da4311 329 (329)
> tport_vsend(0xb6d07cd8): 625 bytes of 625 to tls/121.xxx.xxx.xxx:10371
> tport_vsend returned 625
> send 625 bytes to tls/[121.xxx.xxx.xxx]:10371 at 00:25:39.795356:
> ? ------------------------------------------------------------------------
> ? SIP/2.0 200 OK
>
> On Fri, Feb 12, 2010 at 9:22 AM, Michael Jerris <mike at jerris.com> wrote:
>> how is this different from the working one?
>>
>> Mike
>>
>> On Feb 11, 2010, at 5:13 AM, Mark Campbell-Smith wrote:
>>
>>> ah thats true... The trace is not too readable to me, but may give
>>> some insight to someone that can read the sofia logs....
>>>
>>>
>>> recv 752 bytes from udp/[121.xxx.xxx.xxx]:5060 at 10:10:37.803288:
>>> ? ------------------------------------------------------------------------
>>> ? REGISTER sip:mydns.dyndns.org SIP/2.0
>>> ? Via: SIP/2.0/UDP 121.xxx.xxx.xxx:5060;branch=z9hG4bK-9052c91f
>>> ? From: 2000 <sip:2000 at mydns.dyndns.org>;tag=7a9dbbbfa691136do0
>>> ? To: 2000 <sip:2000 at mydns.dyndns.org>
>>> ? Call-ID: 610db38-dd3b511f at 121.xxx.xxx.xxx
>>> ? CSeq: 32330 REGISTER
>>> ? Contact: 2000 <sip:2000 at 121.xxx.xxx.xxx:5060>;expires=900
>>> ? Authorization: Digest username="2000", realm="mydns.dyndns.org",
>>> nonce="b3298cfe-16f5-11df-9734-e77ba7a22ac3",
>>> uri="sip:mydns.dyndns.org",
>>> response="724fca542ce08d3f12b9ba1043bebb0c", algorithm=MD5,
>>> qop="1225e2f1"
>>> ? Max-Forwards: 70
>>> ? User-Agent: Linksys/SPA3102-5.1.10(GW)
>>> ? Supported: x-sipura
>>> ? Supported: replaces
>>> ? Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
>>> ? Content-Length: 0
>>>
>>>
>>> ? ------------------------------------------------------------------------
>>> tport_deliver(0x97cde80): msg 0x98297e8 (752 bytes) from
>>> udp/121.xxx.xxx.xxx:5060/sip next=(nil)
>>> nta: received REGISTER sip:mydns.dyndns.org SIP/2.0 (CSeq 32330)
>>> nta: REGISTER (32330) going to a default leg
>>> nua: nua_stack_process_request: entering
>>> nua: nh_create: entering
>>> nua: nh_create_handle: entering
>>> nua: nua_stack_set_params: entering
>>> soa_clone(static::0x97cc698, 0x9794808, 0x981cb70) called
>>> soa_set_params(static::0x9758ba8, ...) called
>>> nua(0x981cb70): event i_register 100 Trying
>>> nua: nua_application_event: entering
>>> nua: nua_respond: entering
>>> nua(0x981cb70): sent signal r_respond
>>> nua: nua_handle_destroy: entering
>>> nua(0x981cb70): sent signal r_destroy
>>> nua: nua_handle_magic: entering
>>> nua: nua_handle_destroy: entering
>>> nua(0x981cb70): recv signal r_respond 403 Forbidden
>>> nua: nua_stack_set_params: entering
>>> soa_set_params(static::0x9758ba8, ...) called
>>> tport_tsend(0x97cde80) tpn = UDP/121.xxx.xxx.xxx:5060
>>> tport_resolve addrinfo = 121.xxx.xxx.xxx:5060
>>> tport_by_addrinfo(0x97cde80): not found by name UDP/121.xxx.xxx.xxx:5060
>>> tport_vsend(0x97cde80): 495 bytes of 495 to udp/121.xxx.xxx.xxx:5060
>>> tport_vsend returned 495
>>> send 495 bytes to udp/[121.xxx.xxx.xxx]:5060 at 10:10:37.812955:
>>> ? ------------------------------------------------------------------------
>>> ? SIP/2.0 403 Forbidden
>>>
>>> On Thu, Feb 11, 2010 at 8:57 PM, Michael Jerris <mike at jerris.com> wrote:
>>>> you can crank up the sofia loglevel as well
>>>>
>>>> Mike
>>>>
>>>> On Feb 11, 2010, at 4:31 AM, Mark Campbell-Smith wrote:
>>>>
>>>>> I don't see debug logs (fsctl debug_level 9 or loglevel 9) during the
>>>>> registration process.
>>>>>
>>>>> All I see is the sip messages when the sip trace is activated (403 Forbidden)
>>>>>
>>>>> Is there other debugging that I can enable?
>>>>>
>>>>> On Thu, Feb 11, 2010 at 4:25 PM, Michael Jerris <mike at jerris.com> wrote:
>>>>>> Look at the FreeSWITCH debug logs and compare the differences, they should tell you why.
>>>>>>
>>>>>> Mike
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>


From vetali100 at gmail.com  Sat Feb 13 03:46:12 2010
From: vetali100 at gmail.com (Vitalii Colosov)
Date: Sat, 13 Feb 2010 13:46:12 +0200
Subject: [Freeswitch-users] Other party does not hear voice when
	connecting with X-Lite
In-Reply-To: <a858e7691002130247s439177acg23d6a41de3a9d21@mail.gmail.com>
References: <a858e7691002120133q5e4c806fx309517e4fdec45c5@mail.gmail.com>
	<2F81A680-6A5C-4E3F-9590-FFF6AC4EA80D@freeswitch.org>
	<a858e7691002130247s439177acg23d6a41de3a9d21@mail.gmail.com>
Message-ID: <a858e7691002130346y2018746o2a71a084a3fd12ee@mail.gmail.com>

I found the following differences when using YATE and X-Lite clients in FS
log:

YATE client (working)
[DEBUG] sofia_glue.c:2528 AUDIO RTP [sofia/internal/
1000 at sip.myprovider123.com] 10.244.47.100 port 22894 ->
<<<my.global.ip.adress>>> port 20080 codec: 0 ms: 30

X-Lite client (not working)
[DEBUG] sofia_glue.c:2528 AUDIO RTP [sofia/internal/
1000 at sip.myprovider123.com] 10.244.47.100 port 23952 -> 192.168.2.10
(<<<my.LOCAL.ip.adress behind the router>>>) port 46370 codec: 0 ms: 20


It looks like FS is trying to send MEDIA part to my local ip adress instead
of global...And Yate sends correct, to global IP...

Any ideas?

Thank

2010/2/13 Vitalii Colosov <vetali100 at gmail.com>

> Thanks for the advice!
>
> Yes, x-lite and FS are behind the nat.
>
> I changed x-lite settings from "Discover global IP" to "Use Local IP
> adress", but this did not solve the problem.
>
> I would like to add that when x-lite is configured directly to
> international gateway sip proxy, it is working fine.
> Only when it is connected via FS, voice is missing...
>
>  There is definetely something wrong with FS configuration...
>
> Do you know if I can enable some debug level that will provide me some
> useful information about voice part...?
>
> Thank you!
>
> Vitalie
>
>
>
> 2010/2/12 Brian West <brian at freeswitch.org>
>
> is x-lite behind nat with the freeswitch box?  If so you'll need to disable
>> the discover global IP so that it doesn't try to hair pin thru your NAT
>> router.... Most nat routers won't work correctly trying to do that.
>>
>> /b
>>
>> On Feb 12, 2010, at 3:33 AM, Vitalii Colosov wrote:
>>
>> >
>> > But when I connect using X-Lite, it connects but other party cannot hear
>> me.
>> > I can hear him well...
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100213/c608ff02/attachment.html 

From rupa at rupa.com  Sat Feb 13 06:23:19 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Sat, 13 Feb 2010 08:23:19 -0600
Subject: [Freeswitch-users] Svar: Re: Need help setting up a feature
In-Reply-To: <4B7676D6020000E100000481@mail.fribert.dk>
References: <4B73A799020000E100000470@mail.fribert.dk>
	<b39273531002110528p42218d07q95e97485139315f1@mail.gmail.com>
	<4B7676D6020000E100000481@mail.fribert.dk>
Message-ID: <b39273531002130623s4f4a8617r6b4ad0f3ebad1836@mail.gmail.com>

Ok, how to transfer a call on my siemens a580.  It is more complex than it
needs to.

While call is up with partya, press Menu and then select External call.
Place call to party b, talk to party b and tell them you are transfering
Press the flash (R) key to initiate the transfer.

This works for me with the settings I have down below.  With the settings
you have listed in your paste, pressing the flash key will just send a INFO
packet with the key set to "16".  This is probably not what you want.

On Sat, Feb 13, 2010 at 2:54 AM, mailinglist <mailinglist at fribert.dk> wrote:

>  Hi Rupa
>
> I've got similar settings here, but I can't make them work, nothing happens
> when I press R:
>
> The settings are
> DTMP over VoIP connections: Send settings [ ]Auto   [X] Audio  [ ] RFC 2833
> [X] SIP Info
>
> Call Transfer
> Use the R key to initiate call     ( ) Yes   (X) No
> transfer with the SIP Refer Method
>
> Transfer Call by On-Hook        (X) Yes   ( ) No
>
> Derive Target address             ( ) from SIP URL   (X)   from SIP Contact
> Header
>
> Find Target address automatically    ( ) Yes   (X) No
>
> Hook Flash  (R-key)
>
> Application Type:   dtmf-relay
> Application Signal: 16
>
> That's all the settings, I had to set 'send settings' from auto to
> audio+SIP Info otherwise the R-Flash settings were disabled.
>
>
> But all in all, I think a *1 or something during the call would be a better
> method, because I should be able to make that work on al types of phones,
> right?
>
>
>  Best regards
> Fribse
>
> /Running Freeswitch on pfSense running on a VMWare ESXi 4.0
>
> >>> 11-02-2010 kl. 14:28 skrev Rupa Schomaker <rupa at rupa.com> i
> meddelelsen <b39273531002110528p42218d07q95e97485139315f1 at mail.gmail.com>:
>   My Siemens A580 has options for controlling the R key. It seems that you
> can either have it setup for transfer or as a hook flash. Default is as a
> transfer key.
>
> I haven't succeeded in getting it to work for transfer and it is wayyyyy
> down low on my list of things to do with the phone.
>
> From the web UI:
>
>    Call Transfer  Use the R key to initiate call transfer with the SIP
> Refer method.: Yes No  Transfer Call by On-Hook: Yes No  Derive target
> address: from SIP URL from SIP contact header  Find target addr.
> automatically: Yes No
>   Hook Flash (R-key)  R key settings are disabled because the R key is
> being used for call transfer.
>
> On Wed, Feb 10, 2010 at 11:45 PM, mailinglist <mailinglist at fribert.dk>wrote:
>
>>  Sorry for the repost, but the previous thread just died :-)
>>  I'm trying to get the possibility of transfering an incoming call from
>> one extension to another, and give the possibility of turning it into a
>> conference.
>> I don't have a 'transfer' button.
>> I do have an 'R' button on the Siemens handsets, and a 'Flash' button on
>> the Sipura. The 'Flash' button gives me a new dialtone, gives the caller
>> MOH, and then I can dial the new extension, and transfer the call, but not
>> create a conference.
>> But the Siemens handset does not have a 'flash', and pressing the R
>> doesn't do anything.
>>  It might be two different features 'transfer' and 'conference'...
>>
>> But I thought that using the bind_meta_app would accomplish both.
>>  It's on an incoming call from the outside.
>> So the situation:
>> The Public folder has an entry that matches the dialed number, and does a
>> transfer to 8202.
>> Then the dialplan matches the 8202 with a group, and the phone rings.
>> Somebody picks it up, finds out that it needs to be transferred to another
>> extension, or transferred to a conference with a second extension.
>> How do I construct that?
>>   Best regards
>> Fribse
>>  /Running Freeswitch on pfSense running on a VMWare ESXi 4.0
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100213/8a8d5e2f/attachment.html 

From anthony.minessale at gmail.com  Sat Feb 13 07:07:33 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 13 Feb 2010 09:07:33 -0600
Subject: [Freeswitch-users] How to set outbound caller id info for
	multiple users/extensions
In-Reply-To: <093DD565390C1E4FB15D7B383E86BB05AF05AB@Goose.personalcyberspace.net>
References: <093DD565390C1E4FB15D7B383E86BB05AF05AB@Goose.personalcyberspace.net>
Message-ID: <191c3a031002130707v5f1d5ee0lb9e791e9a0aab956@mail.gmail.com>

It should be covered on the wiki http://wiki.freeswitch.org

On Feb 12, 2010 6:23 PM, "Joseph Puchalski" <
joseph.puchalski at personalcyberspace.com> wrote:

 I'm having problems setting different outbound caller id info for different
extensions/users.



I've set up a small system with two active users. I set up my users by
copying and modifying existing entries from the dialplan files that come
with freeSWITCH



Here are my two extensions:



<include>

  <user id="5859">

    <params>

      <param name="password" value="xxxxxx"/>

      <param name="vm-password" value="xxxxxx"/>

    </params>

    <variables>

      <variable name="toll_allow" value="domestic,international,local"/>

      <variable name="accountcode" value="5859"/>

      <variable name="user_context" value="default"/>

      <variable name="effective_caller_id_name" value="5859"/>

      <variable name="effective_caller_id_number" value="5859"/>

      <variable name="outbound_caller_id_name"
value="$${outbound_caller_name}"/>

      <variable name="outbound_caller_id_number"
value="$${outbound_caller_id}"/>

      <variable name="callgroup" value="techsupport"/>

    </variables>

  </user>

</include>



<include>

  <user id="5515">

    <params>

      <param name="password" value="xxxxxxx"/>

      <param name="vm-password" value="xxxxxxx"/>

    </params>

    <variables>

      <variable name="toll_allow" value="domestic,international,local"/>

      <variable name="accountcode" value="5515"/>

      <variable name="user_context" value="default"/>

      <variable name="effective_caller_id_name" value="5515"/>

      <variable name="effective_caller_id_number" value="5515"/>

      <variable name="outbound_caller_id_name" value="Jane Doe"/>

      <variable name="outbound_caller_id_number" value="5558395515"/>

      <variable name="callgroup" value="techsupport"/>

    </variables>

  </user>

</include>



These extensions are in files named 5859.xml and 5515.xml respectively.



I'm using a SIP trunk from Vitelity (in and out) with two DIDs corresponding
to the extensions above.



Inbound and outbound calling work as needed with one exception: Calls
originating from user/extension 5515 go out with the caller ID of
extension/user 5859.



Extension 5859 was the first that I created.



Where should I be setting the outbound caller id number for my second
extension? I've been trying to track this down in the available
documentation but have been unable to do so.



I apologize ahead of time if this is answered somewhere obvious that I've
missed.



Thanks for any help.



Joe (FreeSWITCH newbie)

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100213/d7e7160a/attachment-0001.html 

From brian at freeswitch.org  Sat Feb 13 07:08:53 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 13 Feb 2010 09:08:53 -0600
Subject: [Freeswitch-users] Forbidden using UDP, works with TCP/TLS
In-Reply-To: <33c87fa31002130343k7a01e9abt9c33360399f246f8@mail.gmail.com>
References: <33c87fa31002101824h21e3bf7lb13b68e8d4fd36a2@mail.gmail.com>
	<2EF27EFC-1634-46A3-B438-2EE99BEF58F1@jerris.com>
	<33c87fa31002110131w3cf6ad00n802669856085a879@mail.gmail.com>
	<CA21231C-108B-4445-A3B9-BF8D94FFE898@jerris.com>
	<33c87fa31002110213q1de0eebfsb950d8725ea9aba@mail.gmail.com>
	<0A905834-850F-4D1F-AEC5-D3A87CE4141B@jerris.com>
	<33c87fa31002121629k74f26807i131841aa4252461b@mail.gmail.com>
	<33c87fa31002130343k7a01e9abt9c33360399f246f8@mail.gmail.com>
Message-ID: <C37AB6F3-F712-41B6-9D6D-7A8AE3C6CABB@freeswitch.org>

Can't tell since you keep cutting the lines required to figure this out... FULL console log with FULL sip trace.  You always cut out what I wanna see.  Just get a pcap.

/b

On Feb 13, 2010, at 5:43 AM, Mark Campbell-Smith wrote:

> More testing. The device registers successfully to my SIP provider
> directly using UDP - why would FS be rejecting the registration
> request?



From brian at freeswitch.org  Sat Feb 13 07:09:35 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 13 Feb 2010 09:09:35 -0600
Subject: [Freeswitch-users] Other party does not hear voice when
	connecting with X-Lite
In-Reply-To: <a858e7691002130346y2018746o2a71a084a3fd12ee@mail.gmail.com>
References: <a858e7691002120133q5e4c806fx309517e4fdec45c5@mail.gmail.com>
	<2F81A680-6A5C-4E3F-9590-FFF6AC4EA80D@freeswitch.org>
	<a858e7691002130247s439177acg23d6a41de3a9d21@mail.gmail.com>
	<a858e7691002130346y2018746o2a71a084a3fd12ee@mail.gmail.com>
Message-ID: <D160C09B-1623-4A0A-BB14-1A1CC04A9276@freeswitch.org>

That should work either way then...are you trying to do this all on the same machine?

/b

On Feb 13, 2010, at 5:46 AM, Vitalii Colosov wrote:

> It looks like FS is trying to send MEDIA part to my local ip adress instead of global...And Yate sends correct, to global IP...
> 
> Any ideas?
> 
> Thank 



From brian at freeswitch.org  Sat Feb 13 07:11:05 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 13 Feb 2010 09:11:05 -0600
Subject: [Freeswitch-users] transmission status of channel
In-Reply-To: <a44688a1002122314t4060f29ehce920077ee90704b@mail.gmail.com>
References: <a44688a1002120732v7d95a6f9x2d4bd7ed68de59eb@mail.gmail.com>
	<3028D6AC-46F2-4291-BAA8-E49C9347917B@freeswitch.org>
	<b39273531002120952u698ab5d1x799696b052ced9a9@mail.gmail.com>
	<8e9d67561002121010p37b16aceyb06cda0420e1aa99@mail.gmail.com>
	<a44688a1002122314t4060f29ehce920077ee90704b@mail.gmail.com>
Message-ID: <2FA9500D-5990-4B8E-8DBF-0966DEDC42EA@freeswitch.org>

FreeSWITCH doesn't support RTCP yet.

/b

On Feb 13, 2010, at 1:14 AM, Woody Dickson wrote:

> Does anyone know where to find those RTCP info from the core rtp stack?
> 
> On Fri, Feb 12, 2010 at 10:10 AM, Timur Valishev <tim at novion.ru> wrote:
>> It would be very nice if FS pass RTCP information to channel vars...
>> 
>> Best regards,
>> Timur Valishev

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100213/6e5a8d36/attachment.html 

From brian at freeswitch.org  Sat Feb 13 07:14:26 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 13 Feb 2010 09:14:26 -0600
Subject: [Freeswitch-users] Bridging sessions from two separate lua
	scripts; uuid_bridge? intercept?
In-Reply-To: <eb7797d71002121904v5bdfac87w2684c461ca1770cc@mail.gmail.com>
References: <eb7797d71002071831w27301c5dva767b82b38c2ce6f@mail.gmail.com>
	<C23AA196-E851-4D5B-B1B5-2C2F32F6BB68@freeswitch.org>
	<eb7797d71002071929pb53e1adue2e91e6bdb598251@mail.gmail.com>
	<eb7797d71002080737k3f8fa5c9m8bf3ef2fcf42881e@mail.gmail.com>
	<D753EDBD-6ED7-43C2-B5F5-A899E3C3D3E2@jerris.com>
	<eb7797d71002121904v5bdfac87w2684c461ca1770cc@mail.gmail.com>
Message-ID: <BD307995-B481-4F2A-B072-21AEA34BBEAC@freeswitch.org>

http://wiki.freeswitch.org/wiki/Mod_lua#session:setAutoHangup

But I don't get why you're doing this in such a convoluted way... I'm still not clear why you're going at the task in this manner.  Can you clearly outline what exactly you're trying to accomplish?

/b

On Feb 12, 2010, at 9:04 PM, Adam Wilt wrote:

> Thanks again for the help Michael.
> 
> I'm now upgraded to version 1.5, but I'm still getting the same
> problem. When I try to bridge sessions from two separate lua scripts,
> both sessions hang up on me. I think maybe I don't understand how
> "intercept" works.
> Anyway, I posted the debug trace here:
> 
>                      http://pastebin.freeswitch.org/12121
> 
> And I also put together a small example which exhibits the problem.
> The first script is started by an inbound call and starts the second
> script. The second script places an outbound call and tries to bridge
> the two sessions together:
> 
> Inbound script:  http://pastebin.freeswitch.org/12122
> Outbound script: http://pastebin.freeswitch.org/12123
> 
> Thanks,
> Adam

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100213/31a38f36/attachment.html 

From anthony.minessale at gmail.com  Sat Feb 13 07:14:40 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 13 Feb 2010 09:14:40 -0600
Subject: [Freeswitch-users] Bridging sessions from two separate lua
	scripts; uuid_bridge? intercept?
In-Reply-To: <eb7797d71002121904v5bdfac87w2684c461ca1770cc@mail.gmail.com>
References: <eb7797d71002071831w27301c5dva767b82b38c2ce6f@mail.gmail.com>
	<C23AA196-E851-4D5B-B1B5-2C2F32F6BB68@freeswitch.org>
	<eb7797d71002071929pb53e1adue2e91e6bdb598251@mail.gmail.com>
	<eb7797d71002080737k3f8fa5c9m8bf3ef2fcf42881e@mail.gmail.com>
	<D753EDBD-6ED7-43C2-B5F5-A899E3C3D3E2@jerris.com>
	<eb7797d71002121904v5bdfac87w2684c461ca1770cc@mail.gmail.com>
Message-ID: <191c3a031002130714x2d876657p9c90d768d28a4496@mail.gmail.com>

One or the other of intercept or uuid_bridge not both.

Either uuid_bridge uuid1 uuid2 or
Execute intercept on session1 with uuid2
Or uuid_transfer to intercept:uuid2 inline

When you uuid_bridge you must exit the script.

When you intercept the call will block until the bridge is over unless its
bypass media otherwise exit the script

If either session was created inside the script use session:setAutoHangup(0)
on them first.

On Feb 12, 2010 9:11 PM, "Adam Wilt" <wiltingtree at gmail.com> wrote:

Thanks again for the help Michael.

I'm now upgraded to version 1.5, but I'm still getting the same
problem. When I try to bridge sessions from two separate lua scripts,
both sessions hang up on me. I think maybe I don't understand how
"intercept" works.
Anyway, I posted the debug trace here:

                     http://pastebin.freeswitch.org/12121

And I also put together a small example which exhibits the problem.
The first script is started by an inbound call and starts the second
script. The second script places an outbound call and tries to bridge
the two sessions together:

Inbound script:  http://pastebin.freeswitch.org/12122
Outbound script: http://pastebin.freeswitch.org/12123

Thanks,
Adam





On 2/9/10, Michael Jerris <mike at jerris.com> wrote:
> 1.4? how does the future look, report back...
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100213/74e4f824/attachment.html 

From anthony.minessale at gmail.com  Sat Feb 13 07:15:30 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 13 Feb 2010 09:15:30 -0600
Subject: [Freeswitch-users] transmission status of channel
In-Reply-To: <a44688a1002122314t4060f29ehce920077ee90704b@mail.gmail.com>
References: <a44688a1002120732v7d95a6f9x2d4bd7ed68de59eb@mail.gmail.com>
	<3028D6AC-46F2-4291-BAA8-E49C9347917B@freeswitch.org>
	<b39273531002120952u698ab5d1x799696b052ced9a9@mail.gmail.com>
	<8e9d67561002121010p37b16aceyb06cda0420e1aa99@mail.gmail.com>
	<a44688a1002122314t4060f29ehce920077ee90704b@mail.gmail.com>
Message-ID: <191c3a031002130715w46e2dc7eo284314498a4b53fa@mail.gmail.com>

Yes in the patch that would be written when an appropriate bounty is
collected.

On Feb 13, 2010 1:22 AM, "Woody Dickson" <woodydickson at gmail.com> wrote:

Does anyone know where to find those RTCP info from the core rtp stack?


On Fri, Feb 12, 2010 at 10:10 AM, Timur Valishev <tim at novion.ru> wrote:
> It would be very nice if ...
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100213/0cbef68c/attachment.html 

From brian at freeswitch.org  Sat Feb 13 07:17:01 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 13 Feb 2010 09:17:01 -0600
Subject: [Freeswitch-users] How to set outbound caller id info for
	multiple users/extensions
In-Reply-To: <191c3a031002130707v5f1d5ee0lb9e791e9a0aab956@mail.gmail.com>
References: <093DD565390C1E4FB15D7B383E86BB05AF05AB@Goose.personalcyberspace.net>
	<191c3a031002130707v5f1d5ee0lb9e791e9a0aab956@mail.gmail.com>
Message-ID: <3FF62C88-2423-43F0-B8A3-C64EF4BC80AC@freeswitch.org>

I also have to point out their is no such official variable for "outbound_caller_id_name" or "outbound_caller_id_number", Those are just made up variables I used in the default config.

01_example.com.xml

You'll notice I use these lines.

      <action application="set" data="effective_caller_id_number=${outbound_caller_id_number}"/>
      <action application="set" data="effective_caller_id_name=${outbound_caller_id_name}"/>

Its just a way to set the users default outbound caller ID .

/b


On Feb 13, 2010, at 9:07 AM, Anthony Minessale wrote:

> It should be covered on the wiki http://wiki.freeswitch.org
> 
> 
>> On Feb 12, 2010 6:23 PM, "Joseph Puchalski" <joseph.puchalski at personalcyberspace.com> wrote:
>> 
>> I'm having problems setting different outbound caller id info for different extensions/users.
>> 
>>  
>> I've set up a small system with two active users. I set up my users by copying and modifying existing entries from the dialplan files that come with freeSWITCH
>> 
>>  
>> Here are my two extensions:
>> 
>>  
>> <include>
>> 
>>   <user id="5859">
>> 
>>     <params>
>> 
>>       <param name="password" value="xxxxxx"/>
>> 
>>       <param name="vm-password" value="xxxxxx"/>
>> 
>>     </params>
>> 
>>     <variables>
>> 
>>       <variable name="toll_allow" value="domestic,international,local"/>
>> 
>>       <variable name="accountcode" value="5859"/>
>> 
>>       <variable name="user_context" value="default"/>
>> 
>>       <variable name="effective_caller_id_name" value="5859"/>
>> 
>>       <variable name="effective_caller_id_number" value="5859"/>
>> 
>>       <variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/>
>> 
>>       <variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/>
>> 
>>       <variable name="callgroup" value="techsupport"/>
>> 
>>     </variables>
>> 
>>   </user>
>> 
>> </include>
>> 
>>  
>> <include>
>> 
>>   <user id="5515">
>> 
>>     <params>
>> 
>>       <param name="password" value="xxxxxxx"/>
>> 
>>       <param name="vm-password" value="xxxxxxx"/>
>> 
>>     </params>
>> 
>>     <variables>
>> 
>>       <variable name="toll_allow" value="domestic,international,local"/>
>> 
>>       <variable name="accountcode" value="5515"/>
>> 
>>       <variable name="user_context" value="default"/>
>> 
>>       <variable name="effective_caller_id_name" value="5515"/>
>> 
>>       <variable name="effective_caller_id_number" value="5515"/>
>> 
>>       <variable name="outbound_caller_id_name" value="Jane Doe"/>
>> 
>>       <variable name="outbound_caller_id_number" value="5558395515"/>
>> 
>>       <variable name="callgroup" value="techsupport"/>
>> 
>>     </variables>
>> 
>>   </user>
>> 
>> </include>
>> 
>>  
>> These extensions are in files named 5859.xml and 5515.xml respectively.
>> 
>>  
>> I'm using a SIP trunk from Vitelity (in and out) with two DIDs corresponding to the extensions above.
>> 
>>  
>> Inbound and outbound calling work as needed with one exception: Calls originating from user/extension 5515 go out with the caller ID of extension/user 5859.
>> 
>>  
>> Extension 5859 was the first that I created.
>> 
>>  
>> Where should I be setting the outbound caller id number for my second extension? I've been trying to track this down in the available documentation but have been unable to do so.
>> 
>>  
>> I apologize ahead of time if this is answered somewhere obvious that I've missed.
>> 
>>  
>> Thanks for any help.
>> 
>>  
>> Joe (FreeSWITCH newbie)
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100213/52f12681/attachment-0001.html 

From anthony.minessale at gmail.com  Sat Feb 13 07:22:38 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 13 Feb 2010 09:22:38 -0600
Subject: [Freeswitch-users] Mod_Conference capacity....
In-Reply-To: <65d96fc81002130327q66d8227ao29ae46f8c7ee556a@mail.gmail.com>
References: <191c3a031002120856q93e52cep14ee96f1e7ed6d1@mail.gmail.com>
	<324899354.1084631265995845519.JavaMail.root@mail-2.01.com>
	<191c3a031002120944u346a3f7cq779f5c9e0ae31522@mail.gmail.com>
	<13ca621c1002121436n9b44dcexffeaafe998d4c88a@mail.gmail.com>
	<65d96fc81002130327q66d8227ao29ae46f8c7ee556a@mail.gmail.com>
Message-ID: <191c3a031002130722v5c282c00s69f0b51b91e42520@mail.gmail.com>

Look through the archives of this list and count the answers to questions I
provide daily for the last several years.  It will be a clear pattern.

People persistantly asking about load testing and benchmark numbers will
always get the same response: we only support it commerically.

We learned from experience the dangers of entertaining such questioning and
its our policy to not do so over our free public forum.

On Feb 13, 2010 5:34 AM, "Tihomir Culjaga" <tculjaga at gmail.com> wrote:



On Fri, Feb 12, 2010 at 11:36 PM, Gavin Henry <gavin.henry at gmail.com> wrote:
>
> Hi,
>
> I think thi...

well, the point is that every application is different and nobody can say
what the performance for this or that HW exactly is. This is not like
holding a hostage it is more like "we need to play on your existing platform
run some tests so we can come out with some real benchmark"... this is what
i can read from Anthony's e-mail.


> I'm willing to write a page on the wiki with info on the recommended
> tools to use etc. with link...


yes, SIPP together with nmon and wireshark and Adobe Audition are the right
tools (at least thats what i'm using...) for such benchmarking.


   - SIPP to generate traffic load,
   - nmon to get some real stats on the machine
   - wireshark to quickly check the jitter on your test call and extract the
   voice stream (sniffing has to be done on a mirrored switch port)
   - adobe autition to perform voice quality check on the extracted voice
   stream





>
> It just reads like a bit of an insult. Have a question? Well you must
> be stupid and need to p...

dont think so, "you must learn how to use all these tools"... if you don't
want to do it, there is another option but it is not free :(




>
> He may have even written the page for the docs team if he was
> encouraged first with a few poi...

well ... i think the time is always the issue + nobody likes documenting
things :(



>
>
> Just my thoughts from experiences as the Doc team lead for the
> OpenLDAP project and dealing...

I cannot say anything here except you are right and i agree what you are
saying about documenting things ...



> Gav.
>
>

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100213/281f69d6/attachment.html 

From scottferri09 at gmail.com  Sat Feb 13 07:52:32 2010
From: scottferri09 at gmail.com (Scott Fernandez)
Date: Sat, 13 Feb 2010 21:22:32 +0530
Subject: [Freeswitch-users] Establishing a Call from .Net based
	application
In-Reply-To: <367751821002010737o4ca83476j62c96cb60c8a456a@mail.gmail.com>
References: <874941.17255.qm@web33502.mail.mud.yahoo.com>
	<df97b9c51001310545n54195cadr9faa3db46917ca9e@mail.gmail.com>
	<367751821002010737o4ca83476j62c96cb60c8a456a@mail.gmail.com>
Message-ID: <df97b9c51002130752h48aae119ja80797ab8a3f5b2d@mail.gmail.com>

Hi,

I've done the following things in order for my .NET application to talk to
FS.

1. Per the link http://wiki.freeswitch.org/wiki/Mod_managed , I've enabled
the Mod_managed module in the mentioned location in FS on CentOS 5.2.

2. There is a configuration settings required to Map the "DLL" to ".so"
object in CentOS.
Now, the question is which DLL and .so file to be made available and where?

3. Do we need to include the AppDemo.class in .NET C# classes that we have
built? If so. how do we initiate the call and get the status of that
particular call with the help of AppDemo.class?. Can I have any specific
code for this?

All I need is to initiate a call from .NET application and then it should
talk to mod_managed module and establish a call. Secondly, I need to know
the status of the call such as Ringing, Active, Hangup etc.

How do we achieve this with mod_managed?. Any sample coding that we can get
for this scenario?

Thanks for your help as always :)

Regards,
Scott


On Mon, Feb 1, 2010 at 9:07 PM, Phillip Jones <pjintheusa at gmail.com> wrote:

> As mentioned http://wiki.freeswitch.org/wiki/Mod_managed should give you
> every thing you need to get mod_managed set up.
>
> Then in the source take a look at demo.csx and particularity AppDemo class.
>
> That should get you started.
>
>
>
> On Sun, Jan 31, 2010 at 8:45 AM, Scott Fernandez <scottferri09 at gmail.com>wrote:
>
>> Hi,
>>
>> Thx for the information. Can I have some detailed steps to configure
>> mod_managed class call control and how do we write the API commands in .Net
>> applications?
>>
>> In addition, how do we get the current STATE of the call when I use
>> webapi?. Because it is required for me to route the call to the user upon it
>> is answered or disconnect  it.
>>
>> Thanks,
>> Scott
>>
>> On Wed, Jan 20, 2010 at 8:47 PM, Diego Toro <dftoro at yahoo.com> wrote:
>>
>>> Hi, the answer is yes, you can to use mod_managed wich offer C# managed
>>> class to call control http://wiki.freeswitch.org/wiki/Mod_managed. Or
>>> using managed ESL (libs/esl/managed) which offer C# managed class to receive
>>> and send events and commands to FreeSwitch.
>>>
>>> Diego Toro
>>> http://lacarretade.blogspot.com/
>>>
>>>
>>> --- On Wed, 1/20/10, Scott Fernandez <scottferri09 at gmail.com> wrote:
>>>
>>> > From: Scott Fernandez <scottferri09 at gmail.com>
>>> > Subject: Re: [Freeswitch-users] Establishing a Call from .Net based
>>> application
>>> > To: freeswitch-users at lists.freeswitch.org
>>> > Date: Wednesday, January 20, 2010, 2:17 AM
>>>  > Thanks Dome. Will try it out and get back to
>>> > you if I come across any issues.
>>> >
>>> > Regards,
>>> > Scott.
>>> >
>>> > On Wed, Jan 20, 2010 at 11:02 AM,
>>> > Dome Charoenyost <dome at tel.co.th>
>>> > wrote:
>>> >
>>> > Please try http://wiki.freeswitch.org/wiki/Webapi
>>> >
>>> >
>>> > you can create class and map to webapi.
>>> >
>>> >
>>> >
>>> > Dome C.
>>> >
>>> >
>>> >
>>> > 2010/1/19 Scott Fernandez <scottferri09 at gmail.com>:
>>> >
>>> > > Hi,
>>> >
>>> > >
>>> >
>>> > > Is there any API modules available for me to initiate
>>> > a call from .Net based
>>> >
>>> > > application?.
>>> >
>>> > >
>>> >
>>> > > The idea is to include the API modules if any with the
>>> > .NET base classes so
>>> >
>>> > > that the API commands will be made available on it. I
>>> > know it is doable when
>>> >
>>> > > I use socket programming in .NET in which Telnet
>>> > session is created.
>>> >
>>> > > However, this would potentially hamper the performance
>>> > of the application
>>> >
>>> > > because of multiple sessions that will be created for
>>> > each call.
>>> >
>>> > >
>>> >
>>> > > Other than that, Is there any Freeswitch API modules
>>> > (like plug-ins)
>>> >
>>> > > available in order to include it into the .Net classes
>>> > and start building
>>> >
>>> > > the customized application?
>>> >
>>> > >
>>> >
>>> > > Any help from any one is highly appreciated.
>>> >
>>> > >
>>> >
>>> > > Thanks,
>>> >
>>> > > Scott
>>> >
>>> > >
>>> >
>>> > >
>>> > _______________________________________________
>>> >
>>> > > FreeSWITCH-users mailing list
>>> >
>>> > > FreeSWITCH-users at lists.freeswitch.org
>>> >
>>> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >
>>> > > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >
>>> > > http://www.freeswitch.org
>>> >
>>> > >
>>> >
>>> > >
>>> >
>>> >
>>> >
>>> > _______________________________________________
>>> >
>>> > FreeSWITCH-users mailing list
>>> >
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> >
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >
>>> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >
>>> > http://www.freeswitch.org
>>> >
>>> >
>>> >
>>> >
>>> > -----Inline Attachment Follows-----
>>>  >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>>
>>>
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100213/77549daa/attachment.html 

From dftoro at yahoo.com  Sat Feb 13 09:05:59 2010
From: dftoro at yahoo.com (Diego Toro)
Date: Sat, 13 Feb 2010 09:05:59 -0800 (PST)
Subject: [Freeswitch-users] Establishing a Call from .Net based
	application
In-Reply-To: <df97b9c51002130752h48aae119ja80797ab8a3f5b2d@mail.gmail.com>
Message-ID: <494815.23786.qm@web33508.mail.mud.yahoo.com>


> 1. Per the link http://wiki.freeswitch.org/wiki/Mod_managed
>  , I've enabled the Mod_managed module in the mentioned
> location in FS on CentOS 5.2.

mod_managed is supported on CentOs with Mono.


3. Do we need to include the AppDemo.class in .NET C#
> classes that we have built? If so. how do we initiate the
> call and get the status of that particular call with the
> help of AppDemo.class?. Can I have any specific code for
> this?

AppDemo is only a example about how to use mod_managed like application.

a. You should implement IAppPlugin interface so FreeSwitch brings call control to your C# class through mod_managed.

Simple Example:

using FreeSWITCH;
using FreeSWITCH.Native;

namespace BITS.Ivr.Bp.Server 
{
 public class CIV_BPFSProcess : FreeSWITCH.IAppPlugin
{
  public void Run(AppContext context)
  {
   //answer call
   context.Session.Answer();
   //sleep 2 seconds
   context.Session.sleep(2000, 1);
   //hangup call
   context.Session.Hangup("NORMAL_CLEARING");
  }
 }
}

b. Add next extension to a dialplan

<include>
  <extension name="test">
    <condition field="destination_number" expression="^445(\d+)$" >
    <action application="managed" data="CCU_BPFSProcess"/>
 </condition>	
  </extension>
</include>

c. Call for example to 445100 from a softphone


Diego Toro 
http://lacarretade.blogspot.com/


--- On Sat, 2/13/10, Scott Fernandez <scottferri09 at gmail.com> wrote:

> From: Scott Fernandez <scottferri09 at gmail.com>
> Subject: Re: [Freeswitch-users] Establishing a Call from .Net based application
> To: freeswitch-users at lists.freeswitch.org
> Date: Saturday, February 13, 2010, 10:52 AM
> Hi,
> 
> I've done the following things in order for my .NET
> application to talk to FS.
> 
> 1. Per the link http://wiki.freeswitch.org/wiki/Mod_managed
>  , I've enabled the Mod_managed module in the mentioned
> location in FS on CentOS 5.2.
> 
> 2. There is a configuration settings required to Map the
> "DLL" to ".so" object in CentOS.
> Now, the question is which DLL and .so file to be made
> available and where?
> 
> 
> 3. Do we need to include the AppDemo.class in .NET C#
> classes that we have built? If so. how do we initiate the
> call and get the status of that particular call with the
> help of AppDemo.class?. Can I have any specific code for
> this?
> 
> 
> All I need is to initiate a call from .NET application and
> then it should talk to mod_managed module and establish a
> call. Secondly, I need to know the status of the call such
> as Ringing, Active, Hangup etc. 
> 
> How do we achieve this with mod_managed?. Any sample coding
> that we can get for this scenario?
> 
> 
> Thanks for your help as always :)
> 
> Regards,
> Scott
> 
> 
> On Mon, Feb 1, 2010 at 9:07 PM,
> Phillip Jones <pjintheusa at gmail.com>
> wrote:
> 
> As mentioned http://wiki.freeswitch.org/wiki/Mod_managed
> should give you every thing you need to get mod_managed set
> up. 
> 
> 
> Then in the source take a look at demo.csx and
> particularity AppDemo class.
> 
> 
> That should get you started.
> 
> 
> On Sun, Jan 31, 2010 at 8:45 AM,
> Scott Fernandez <scottferri09 at gmail.com>
> wrote:
> 
> 
> Hi,
> ?
> Thx for the information. Can I have some detailed
> steps to configure mod_managed class call control and how do
> we write the API commands in .Net applications?
> ?
> In addition, how do we get the current STATE of the
> call when I use webapi?. Because it is required for me to
> route the call to the user upon it is answered or
> disconnect? it.
> ?
> Thanks,
> Scott
> 
> 
> On Wed, Jan 20, 2010 at 8:47 PM,
> Diego Toro <dftoro at yahoo.com>
> wrote:
> 
> Hi, the answer is yes, you can to
> use mod_managed wich offer C# managed class to call control
> http://wiki.freeswitch.org/wiki/Mod_managed.
> Or using managed ESL (libs/esl/managed) which offer C#
> managed class to receive and send events and commands to
> FreeSwitch.
> 
> 
> 
> 
> Diego Toro
> http://lacarretade.blogspot.com/
> 
> 
> --- On Wed, 1/20/10, Scott Fernandez <scottferri09 at gmail.com>
> wrote:
> 
> 
> 
> 
> > From: Scott Fernandez <scottferri09 at gmail.com>
> 
> > Subject: Re: [Freeswitch-users] Establishing a
> Call from .Net based application
> > To: freeswitch-users at lists.freeswitch.org
> 
> 
> > Date: Wednesday, January 20, 2010, 2:17 AM
> 
> 
> 
> 
> > Thanks Dome. Will try it out and get back to
> > you if I come across any issues.
> >
> > Regards,
> > Scott.
> >
> > On Wed, Jan 20, 2010 at 11:02 AM,
> > Dome Charoenyost <dome at tel.co.th>
> 
> 
> 
> > wrote:
> >
> > Please try http://wiki.freeswitch.org/wiki/Webapi
> >
> >
> > you can create class and map to webapi.
> >
> 
> 
> 
> >
> >
> > Dome C.
> >
> >
> >
> > 2010/1/19 Scott Fernandez <scottferri09 at gmail.com>:
> >
> > > Hi,
> >
> > >
> 
> 
> >
> > > Is there any API modules available for me to
> initiate
> 
> > a call from .Net based
> >
> > > application?.
> >
> > >
> >
> > > The idea is to include the API modules if any
> with the
> > .NET base classes so
> >
> > > that the API commands will be made available on
> it. I
> 
> 
> 
> > know it is doable when
> >
> > > I use socket programming in .NET in which Telnet
> > session is created.
> >
> > > However, this would potentially hamper the
> performance
> > of the application
> 
> 
> 
> >
> > > because of multiple sessions that will be created
> for
> > each call.
> >
> > >
> >
> > > Other than that, Is there any Freeswitch API
> modules
> > (like plug-ins)
> >
> 
> 
> 
> > > available in order to include it into the .Net
> classes
> > and start building
> >
> > > the customized application?
> >
> > >
> >
> > > Any help from any one is highly appreciated.
> 
> 
> 
> >
> > >
> >
> > > Thanks,
> >
> > > Scott
> >
> > >
> >
> > >
> > _______________________________________________
> >
> > > FreeSWITCH-users mailing list
> 
> 
> 
> >
> > > FreeSWITCH-users at lists.freeswitch.org
> >
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> 
> 
> 
> >
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >
> > > http://www.freeswitch.org
> 
> 
> 
> >
> > >
> >
> > >
> >
> >
> >
> > _______________________________________________
> >
> > FreeSWITCH-users mailing list
> >
> > FreeSWITCH-users at lists.freeswitch.org
> 
> 
> 
> >
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 
> 
> 
> >
> > http://www.freeswitch.org
> >
> >
> >
> >
> > -----Inline Attachment Follows-----
> 
> 
> 
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> 
> 
> 
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 
> 
> 
> > http://www.freeswitch.org
> >
> 
> 
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> 
> 
> 
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 
> 
> 
> http://www.freeswitch.org
> 
> 
> 
> _______________________________________________
> 
> FreeSWITCH-users mailing list
> 
> FreeSWITCH-users at lists.freeswitch.org
> 
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> 
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 
> http://www.freeswitch.org
> 
> 
> 
> 
> 
> _______________________________________________
> 
> FreeSWITCH-users mailing list
> 
> FreeSWITCH-users at lists.freeswitch.org
> 
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> 
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 
> http://www.freeswitch.org
> 
> 
> 
> 
> 
> -----Inline Attachment Follows-----
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 


      


From wiltingtree at gmail.com  Sat Feb 13 10:18:17 2010
From: wiltingtree at gmail.com (Adam Wilt)
Date: Sat, 13 Feb 2010 13:18:17 -0500
Subject: [Freeswitch-users] Bridging sessions from two separate lua
	scripts; uuid_bridge? intercept?
In-Reply-To: <191c3a031002130714x2d876657p9c90d768d28a4496@mail.gmail.com>
References: <eb7797d71002071831w27301c5dva767b82b38c2ce6f@mail.gmail.com>
	<C23AA196-E851-4D5B-B1B5-2C2F32F6BB68@freeswitch.org>
	<eb7797d71002071929pb53e1adue2e91e6bdb598251@mail.gmail.com>
	<eb7797d71002080737k3f8fa5c9m8bf3ef2fcf42881e@mail.gmail.com>
	<D753EDBD-6ED7-43C2-B5F5-A899E3C3D3E2@jerris.com>
	<eb7797d71002121904v5bdfac87w2684c461ca1770cc@mail.gmail.com>
	<191c3a031002130714x2d876657p9c90d768d28a4496@mail.gmail.com>
Message-ID: <eb7797d71002131018r56e86adnc054a3eaa2c05a96@mail.gmail.com>

Thanks Anthony and Brian.

setAutoHangup was specified in the script calling intercept, but not the
other script. When I add it to the other script, neither side hangs up, but
the call is not bridged. Neither party can hear the other party. From your
description, intercept sounds like what I want to do.

I made sure bypass media was off. Does the session being intercepted have to
be in a certain state for the intercept to work? In my example, a prompt is
being played during the time it is intercepted, but I tried having it sleep
instead with the same result.

The reason I'm doing it this way is because both parties have to go through
a bunch of different states before they are allowed to speak to each other.
I tried controlling both legs from the same script previously, but sometimes
one session would block waiting for the other session.

Thanks,
Adam


On Sat, Feb 13, 2010 at 10:14 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> One or the other of intercept or uuid_bridge not both.
>
> Either uuid_bridge uuid1 uuid2 or
> Execute intercept on session1 with uuid2
> Or uuid_transfer to intercept:uuid2 inline
>
> When you uuid_bridge you must exit the script.
>
> When you intercept the call will block until the bridge is over unless its
> bypass media otherwise exit the script
>
> If either session was created inside the script use
> session:setAutoHangup(0) on them first.
>
> On Feb 12, 2010 9:11 PM, "Adam Wilt" <wiltingtree at gmail.com> wrote:
>
> Thanks again for the help Michael.
>
> I'm now upgraded to version 1.5, but I'm still getting the same
> problem. When I try to bridge sessions from two separate lua scripts,
> both sessions hang up on me. I think maybe I don't understand how
> "intercept" works.
> Anyway, I posted the debug trace here:
>
>                      http://pastebin.freeswitch.org/12121
>
> And I also put together a small example which exhibits the problem.
> The first script is started by an inbound call and starts the second
> script. The second script places an outbound call and tries to bridge
> the two sessions together:
>
> Inbound script:  http://pastebin.freeswitch.org/12122
> Outbound script: http://pastebin.freeswitch.org/12123
>
> Thanks,
> Adam
>
>
>
>
>
> On 2/9/10, Michael Jerris <mike at jerris.com> wrote:
> > 1.4? how does the future look, report back...
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100213/6dcabba2/attachment.html 

From gavin.henry at gmail.com  Sat Feb 13 11:20:21 2010
From: gavin.henry at gmail.com (Gavin Henry)
Date: Sat, 13 Feb 2010 19:20:21 +0000
Subject: [Freeswitch-users] Mod_Conference capacity....
In-Reply-To: <191c3a031002130722v5c282c00s69f0b51b91e42520@mail.gmail.com>
References: <191c3a031002120856q93e52cep14ee96f1e7ed6d1@mail.gmail.com>
	<324899354.1084631265995845519.JavaMail.root@mail-2.01.com>
	<191c3a031002120944u346a3f7cq779f5c9e0ae31522@mail.gmail.com>
	<13ca621c1002121436n9b44dcexffeaafe998d4c88a@mail.gmail.com>
	<65d96fc81002130327q66d8227ao29ae46f8c7ee556a@mail.gmail.com>
	<191c3a031002130722v5c282c00s69f0b51b91e42520@mail.gmail.com>
Message-ID: <13ca621c1002131120u482f6420xea1f7cad534cbe1a@mail.gmail.com>

On 13 February 2010 15:22, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> Look through the archives of this list and count the answers to questions I
> provide daily for the last several years.? It will be a clear pattern.
>
> People persistantly asking about load testing and benchmark numbers will
> always get the same response: we only support it commerically.
>
> We learned from experience the dangers of entertaining such questioning and
> its our policy to not do so over our free public forum.

OK, thanks for the info. If this is now the case these three links
should be updated to reflect the
FreeSWITCH projects stand point:

http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations
http://wiki.freeswitch.org/wiki/Long_term_testing_chevymanjosh
http://wiki.freeswitch.org/wiki/FreeSwitch_FAQ#Q:_How_many_concurrent_calls_can_it_support.3F__Any_benchmarks.3F

or a new FAQ entry created etc.

Thanks,

Gavin.


-- 
http://www.suretecsystems.com/services/openldap/
http://www.suretectelecom.com


From wiltingtree at gmail.com  Sat Feb 13 11:32:41 2010
From: wiltingtree at gmail.com (Adam Wilt)
Date: Sat, 13 Feb 2010 14:32:41 -0500
Subject: [Freeswitch-users] Bridging sessions from two separate lua
	scripts; uuid_bridge? intercept?
In-Reply-To: <eb7797d71002131018r56e86adnc054a3eaa2c05a96@mail.gmail.com>
References: <eb7797d71002071831w27301c5dva767b82b38c2ce6f@mail.gmail.com>
	<C23AA196-E851-4D5B-B1B5-2C2F32F6BB68@freeswitch.org>
	<eb7797d71002071929pb53e1adue2e91e6bdb598251@mail.gmail.com>
	<eb7797d71002080737k3f8fa5c9m8bf3ef2fcf42881e@mail.gmail.com>
	<D753EDBD-6ED7-43C2-B5F5-A899E3C3D3E2@jerris.com>
	<eb7797d71002121904v5bdfac87w2684c461ca1770cc@mail.gmail.com>
	<191c3a031002130714x2d876657p9c90d768d28a4496@mail.gmail.com>
	<eb7797d71002131018r56e86adnc054a3eaa2c05a96@mail.gmail.com>
Message-ID: <eb7797d71002131132x2da4d2a0g54b8b8faf19520be@mail.gmail.com>

Here's a new debug log which shows what happens using only intercept and
having setAutoHangup(false) on both sides:

           http://pastebin.freeswitch.org/12124

<http://pastebin.freeswitch.org/12124>And here's my updated bridge function:

function bridge_calls(session,api,b_leg_uuid, call_len)
  freeswitch.consoleLog("info","A leg - Bridging " .. tostring(b_leg_uuid)
.. " with " .. tostring(session.uuid) .. "\n")
  session:setAutoHangup(false)
  session:execute("set","continue_on_fail=true")
  api:executeString("uuid_media " .. tostring(b_leg_uuid))
  api:executeString("uuid_media " .. tostring(session.uuid))
  api:executeString("intercept -bleg " .. tostring(b_leg_uuid))
end


Thanks,
Adam


On Sat, Feb 13, 2010 at 1:18 PM, Adam Wilt <wiltingtree at gmail.com> wrote:

> Thanks Anthony and Brian.
>
> setAutoHangup was specified in the script calling intercept, but not the
> other script. When I add it to the other script, neither side hangs up, but
> the call is not bridged. Neither party can hear the other party. From your
> description, intercept sounds like what I want to do.
>
> I made sure bypass media was off. Does the session being intercepted have
> to be in a certain state for the intercept to work? In my example, a prompt
> is being played during the time it is intercepted, but I tried having it
> sleep instead with the same result.
>
> The reason I'm doing it this way is because both parties have to go through
> a bunch of different states before they are allowed to speak to each other.
> I tried controlling both legs from the same script previously, but sometimes
> one session would block waiting for the other session.
>
> Thanks,
> Adam
>
>
> On Sat, Feb 13, 2010 at 10:14 AM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> One or the other of intercept or uuid_bridge not both.
>>
>> Either uuid_bridge uuid1 uuid2 or
>> Execute intercept on session1 with uuid2
>> Or uuid_transfer to intercept:uuid2 inline
>>
>> When you uuid_bridge you must exit the script.
>>
>> When you intercept the call will block until the bridge is over unless its
>> bypass media otherwise exit the script
>>
>> If either session was created inside the script use
>> session:setAutoHangup(0) on them first.
>>
>> On Feb 12, 2010 9:11 PM, "Adam Wilt" <wiltingtree at gmail.com> wrote:
>>
>> Thanks again for the help Michael.
>>
>> I'm now upgraded to version 1.5, but I'm still getting the same
>> problem. When I try to bridge sessions from two separate lua scripts,
>> both sessions hang up on me. I think maybe I don't understand how
>> "intercept" works.
>> Anyway, I posted the debug trace here:
>>
>>                      http://pastebin.freeswitch.org/12121
>>
>> And I also put together a small example which exhibits the problem.
>> The first script is started by an inbound call and starts the second
>> script. The second script places an outbound call and tries to bridge
>> the two sessions together:
>>
>> Inbound script:  http://pastebin.freeswitch.org/12122
>> Outbound script: http://pastebin.freeswitch.org/12123
>>
>> Thanks,
>> Adam
>>
>>
>>
>>
>>
>> On 2/9/10, Michael Jerris <mike at jerris.com> wrote:
>> > 1.4? how does the future look, report back...
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100213/04625cbe/attachment.html 

From mgg at giagnocavo.net  Sat Feb 13 11:45:03 2010
From: mgg at giagnocavo.net (Michael Giagnocavo)
Date: Sat, 13 Feb 2010 14:45:03 -0500
Subject: [Freeswitch-users] Establishing a Call from .Net
	based	application
In-Reply-To: <df97b9c51002130752h48aae119ja80797ab8a3f5b2d@mail.gmail.com>
References: <874941.17255.qm@web33502.mail.mud.yahoo.com>
	<df97b9c51001310545n54195cadr9faa3db46917ca9e@mail.gmail.com>
	<367751821002010737o4ca83476j62c96cb60c8a456a@mail.gmail.com>
	<df97b9c51002130752h48aae119ja80797ab8a3f5b2d@mail.gmail.com>
Message-ID: <6E8D2069C08AA84A83D336E996AE4C67032C9D5346@mse17be1.mse17.exchange.ms>


2. There is a configuration settings required to Map the "DLL" to ".so" object in CentOS.
Now, the question is which DLL and .so file to be made available and where?

"
If you are experiencing NullReferenceExceptions with any plugin run through the dialplan, make sure you have included the appropriate entry in your dllmap<http://mono-project.com/Config_DllMap> configuration:
 <dllmap dll="mod_managed.dll" target="mod_managed.so" os="!windows"/>
"
mod_managed.so will be in your freeswitch mod directory.

All I need is to initiate a call from .NET application and then it should talk to mod_managed module and establish a call. Secondly, I need to know the status of the call such as Ringing, Active, Hangup etc.

To initiate a call, try ManagedSession.Originate.
-Michael
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100213/44ac713a/attachment-0001.html 

From brian at freeswitch.org  Sat Feb 13 11:54:10 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 13 Feb 2010 13:54:10 -0600
Subject: [Freeswitch-users] Mod_Conference capacity....
In-Reply-To: <13ca621c1002131120u482f6420xea1f7cad534cbe1a@mail.gmail.com>
References: <191c3a031002120856q93e52cep14ee96f1e7ed6d1@mail.gmail.com>
	<324899354.1084631265995845519.JavaMail.root@mail-2.01.com>
	<191c3a031002120944u346a3f7cq779f5c9e0ae31522@mail.gmail.com>
	<13ca621c1002121436n9b44dcexffeaafe998d4c88a@mail.gmail.com>
	<65d96fc81002130327q66d8227ao29ae46f8c7ee556a@mail.gmail.com>
	<191c3a031002130722v5c282c00s69f0b51b91e42520@mail.gmail.com>
	<13ca621c1002131120u482f6420xea1f7cad534cbe1a@mail.gmail.com>
Message-ID: <E2A03ABF-69BF-4F85-9BB9-7CEAA7738E64@freeswitch.org>

This is the FAQ entry.  It applies to EVERYTHING related to load testing.

/b

On Feb 13, 2010, at 1:20 PM, Gavin Henry wrote:

> http://wiki.freeswitch.org/wiki/FreeSwitch_FAQ#Q:_How_many_concurrent_calls_can_it_support.3F__Any_benchmarks.3F

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100213/b8d90379/attachment.html 

From mbsip at gazeta.pl  Sat Feb 13 12:03:07 2010
From: mbsip at gazeta.pl (mbsip)
Date: Sat, 13 Feb 2010 21:03:07 +0100
Subject: [Freeswitch-users]  mod_python fetching data from mysql for VM
Message-ID: <28f27f5d1002131203j132c6230sa260358611a16247@mail.gmail.com>

Hello,

I am trying to use mod_python to fetch data from Mysql db (through
ODBC) and execute voicemail application.
Below a part of my script:

db=MySQLdb.connect("localhost","root","","test")
       Cursor=db.cursor()
       sql = "SELECT email FROM VM WHERE called_num=%s" % the_dest
       Cursor.execute(sql)
       while (1):
               Results = Cursor.fetchone()
               if Results == None:
                       break
               consoleLog("debug", "Found email " + Results[0] +"\n")
               the_recipient = Results[0]
       db.close()

Now i have email address corresponding with called number. The
question is how to use it for voicemail application?
So it also means how to omit all /directory/default/....xml, where
there are all VM parameters set and use fetched data.

       session.answer()
       session.execute("voicemail", "default ${domain} " + the_dest)

Is this possible or should I start all VM app in python from the scratch?

Thanks,
Maciej


From anthony.minessale at gmail.com  Sat Feb 13 12:39:48 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 13 Feb 2010 14:39:48 -0600
Subject: [Freeswitch-users] Bridging sessions from two separate lua
	scripts; uuid_bridge? intercept?
In-Reply-To: <eb7797d71002131132x2da4d2a0g54b8b8faf19520be@mail.gmail.com>
References: <eb7797d71002071831w27301c5dva767b82b38c2ce6f@mail.gmail.com>
	<C23AA196-E851-4D5B-B1B5-2C2F32F6BB68@freeswitch.org>
	<eb7797d71002071929pb53e1adue2e91e6bdb598251@mail.gmail.com>
	<eb7797d71002080737k3f8fa5c9m8bf3ef2fcf42881e@mail.gmail.com>
	<D753EDBD-6ED7-43C2-B5F5-A899E3C3D3E2@jerris.com>
	<eb7797d71002121904v5bdfac87w2684c461ca1770cc@mail.gmail.com>
	<191c3a031002130714x2d876657p9c90d768d28a4496@mail.gmail.com>
	<eb7797d71002131018r56e86adnc054a3eaa2c05a96@mail.gmail.com>
	<eb7797d71002131132x2da4d2a0g54b8b8faf19520be@mail.gmail.com>
Message-ID: <191c3a031002131239v8d0a8e2xd42725a4682b54b2@mail.gmail.com>

Sigh

On Feb 13, 2010 1:38 PM, "Adam Wilt" <wiltingtree at gmail.com> wrote:

Here's a new debug log which shows what happens using only intercept and
having setAutoHangup(false) on both sides:

           http://pastebin.freeswitch.org/12124

<http://pastebin.freeswitch.org/12124>And here's my updated bridge function:

> function bridge_calls(session,api,b_leg_uuid, call_len)
  freeswitch.consoleLog("info","A leg - Bridging " .. tostring(b_leg_uuid)
.. " with " .. tostring(session.uuid) .. "\n")
  session:setAutoHangup(false)


>   session:execute("set","continue_on_fail=true")
  api:executeString("uuid_media " .. tostring(b_leg_uuid))
  api:executeString("uuid_media " .. tostring(session.uuid))


>   api:executeString("intercept -bleg " .. tostring(b_leg_uuid))
end


Thanks,
Adam




On Sat, Feb 13, 2010 at 1:18 PM, Adam Wilt <wiltingtree at gmail.com> wrote:
>
> Thanks Anthony and ...

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100213/093ad23b/attachment.html 

From sos at sokhapkin.dyndns.org  Sat Feb 13 12:52:12 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Sat, 13 Feb 2010 15:52:12 -0500
Subject: [Freeswitch-users] bypass_media bug?
In-Reply-To: <201002121422.18544.sos@sokhapkin.dyndns.org>
References: <201002121422.18544.sos@sokhapkin.dyndns.org>
Message-ID: <201002131552.12289.sos@sokhapkin.dyndns.org>

More precisely, FS returns to A leg caller SDP from first 183, but not from 
final 200 OK.

On Friday 12 February 2010, Sergey Okhapkin wrote:
> Simple dialplan:
>
>     <extension name="103">
>       <condition field="destination_number" expression="^103$">
>         <action application="set" data="continue_on_fail=true"/>
>         <action application="set" data="bypass_media=true"/>
>         <action application="bridge" data="sofia/cwu/103 at 192.168.1.254"/>
>         <action application="bridge" data="sofia/cwu/104 at 192.168.1.254"/>
>       </condition>
>     </extension>
>
> 103 at 192.168.1.254 returns 183 early media and then "480 temporary
> unavailable", 104 at 192.168.1.254 answers the call (echo test).
>
> When 104 answers, Freeswitch returns to caller in SDP media port from
> "183", but not media port from 104's "200 OK"
>
> Is it a bug or expected behavior? If expected - is there a variable to
> control the behavior? Everything works OK if I replace bypass_media=true
> with bypass_media_after_bridge=true, but sending reinvites is not
> acceptable to me.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From paul at apcl.us  Sat Feb 13 12:59:05 2010
From: paul at apcl.us (Paul Levin)
Date: Sat, 13 Feb 2010 15:59:05 -0500
Subject: [Freeswitch-users] can vm settings go in
	conf\directory\default\default.xml?
Message-ID: <4B771299.8000001@apcl.us>

An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100213/46ebc760/attachment.html 

From wiltingtree at gmail.com  Sat Feb 13 13:03:19 2010
From: wiltingtree at gmail.com (Adam Wilt)
Date: Sat, 13 Feb 2010 16:03:19 -0500
Subject: [Freeswitch-users] Bridging sessions from two separate lua
	scripts; uuid_bridge? intercept?
In-Reply-To: <191c3a031002131239v8d0a8e2xd42725a4682b54b2@mail.gmail.com>
References: <eb7797d71002071831w27301c5dva767b82b38c2ce6f@mail.gmail.com>
	<C23AA196-E851-4D5B-B1B5-2C2F32F6BB68@freeswitch.org>
	<eb7797d71002071929pb53e1adue2e91e6bdb598251@mail.gmail.com>
	<eb7797d71002080737k3f8fa5c9m8bf3ef2fcf42881e@mail.gmail.com>
	<D753EDBD-6ED7-43C2-B5F5-A899E3C3D3E2@jerris.com>
	<eb7797d71002121904v5bdfac87w2684c461ca1770cc@mail.gmail.com>
	<191c3a031002130714x2d876657p9c90d768d28a4496@mail.gmail.com>
	<eb7797d71002131018r56e86adnc054a3eaa2c05a96@mail.gmail.com>
	<eb7797d71002131132x2da4d2a0g54b8b8faf19520be@mail.gmail.com>
	<191c3a031002131239v8d0a8e2xd42725a4682b54b2@mail.gmail.com>
Message-ID: <eb7797d71002131303q19b657dv91b7ae08ab5b34cf@mail.gmail.com>

Does "sigh" mean that it's a problem with FreeSWITCH, or perhaps that I did
something stupid?



On Sat, Feb 13, 2010 at 3:39 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> Sigh
>
> On Feb 13, 2010 1:38 PM, "Adam Wilt" <wiltingtree at gmail.com> wrote:
>
> Here's a new debug log which shows what happens using only intercept and
> having setAutoHangup(false) on both sides:
>
>            http://pastebin.freeswitch.org/12124
>
> <http://pastebin.freeswitch.org/12124>And here's my updated bridge
> function:
>
> > function bridge_calls(session,api,b_leg_uuid, call_len)
>   freeswitch.consoleLog("info","A leg - Bridging " .. tostring(b_leg_uuid)
> .. " with " .. tostring(session.uuid) .. "\n")
>   session:setAutoHangup(false)
>
>
> >   session:execute("set","continue_on_fail=true")
>   api:executeString("uuid_media " .. tostring(b_leg_uuid))
>   api:executeString("uuid_media " .. tostring(session.uuid))
>
>
> >   api:executeString("intercept -bleg " .. tostring(b_leg_uuid))
> end
>
>
> Thanks,
> Adam
>
>
>
>
> On Sat, Feb 13, 2010 at 1:18 PM, Adam Wilt <wiltingtree at gmail.com> wrote:
> >
> > Thanks Anthony and ...
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100213/7c50c0b2/attachment.html 

From mike at jerris.com  Sat Feb 13 13:21:04 2010
From: mike at jerris.com (Michael Jerris)
Date: Sat, 13 Feb 2010 16:21:04 -0500
Subject: [Freeswitch-users] mod_python fetching data from mysql for VM
In-Reply-To: <28f27f5d1002131203j132c6230sa260358611a16247@mail.gmail.com>
References: <28f27f5d1002131203j132c6230sa260358611a16247@mail.gmail.com>
Message-ID: <8E4B95E9-3914-4FEE-9C8C-9A5B4C5E2AB2@jerris.com>

Can you describe what your trying to accomplish, I don't understand what the goal is.  What feature are you looking for that does not already exist in mod_voiceamil.

Mike

On Feb 13, 2010, at 3:03 PM, mbsip wrote:

> Hello,
> 
> I am trying to use mod_python to fetch data from Mysql db (through
> ODBC) and execute voicemail application.
> Below a part of my script:
> 
> db=MySQLdb.connect("localhost","root","","test")
>       Cursor=db.cursor()
>       sql = "SELECT email FROM VM WHERE called_num=%s" % the_dest
>       Cursor.execute(sql)
>       while (1):
>               Results = Cursor.fetchone()
>               if Results == None:
>                       break
>               consoleLog("debug", "Found email " + Results[0] +"\n")
>               the_recipient = Results[0]
>       db.close()
> 
> Now i have email address corresponding with called number. The
> question is how to use it for voicemail application?
> So it also means how to omit all /directory/default/....xml, where
> there are all VM parameters set and use fetched data.
> 
>       session.answer()
>       session.execute("voicemail", "default ${domain} " + the_dest)
> 
> Is this possible or should I start all VM app in python from the scratch?



From mbsip at gazeta.pl  Sat Feb 13 13:23:02 2010
From: mbsip at gazeta.pl (mbsip)
Date: Sat, 13 Feb 2010 22:23:02 +0100
Subject: [Freeswitch-users] can vm settings go in
	conf\directory\default\default.xml?
In-Reply-To: <4B771299.8000001@apcl.us>
References: <4B771299.8000001@apcl.us>
Message-ID: <28f27f5d1002131323n86f03e9wdb60b784559b291f@mail.gmail.com>

Have you tried doing the same with
/usr/local/freeswitch/conf/directory/default.xml ?

Maciej

2010/2/13 Paul Levin <paul at apcl.us>:
> I'm running FS on Windows (in case that matters here).
>
> In conf\directory\default\Bob.xml I have the settings:
>
> ??? ? <param name="vm-email-all-messages" value="true"/>
> ??? ? <param name="vm-keep-local-after-email" value="false"/>
> ??? ? <param name="vm-attach-file" value="true"/>
>
> in addition to other vm- setting that are specific to Bob.? When a voice
> mail is left for Bob, an email is sent to the configured email address.? It
> is working well.? When the email is sent, I can see in the console the
> lines:
>
> 2010-02-10 15:41:36.949484 [DEBUG]
> switch_utils.c:633 Emailed file [C:\WINDOWS\TEMP\mail.12658416960810]
> to [bob at domain.com]
>
> 2010-02-10 15:41:36.949484 [DEBUG]
> mod_voicemail.c:2541 Sending message to bob at domain.com
>
> I then move those 3 <param> lines into the default\default.xml file.? Now
> when a voice mail is left for Bob, an email is not sent and those debug
> lines do not appear on the console.
>
> I don't mind keeping those 3 lines in each user file, but I'm expecting to
> have about 10,000 users and its kinda silly to repeat those lines in each
> user's file.? Can't they go in the default.xml file (and have it work)?
>
> ??? Thanks,
> ??? Paul
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From gavin.henry at gmail.com  Sat Feb 13 13:26:05 2010
From: gavin.henry at gmail.com (Gavin Henry)
Date: Sat, 13 Feb 2010 21:26:05 +0000
Subject: [Freeswitch-users] Mod_Conference capacity....
In-Reply-To: <E2A03ABF-69BF-4F85-9BB9-7CEAA7738E64@freeswitch.org>
References: <191c3a031002120856q93e52cep14ee96f1e7ed6d1@mail.gmail.com>
	<324899354.1084631265995845519.JavaMail.root@mail-2.01.com>
	<191c3a031002120944u346a3f7cq779f5c9e0ae31522@mail.gmail.com>
	<13ca621c1002121436n9b44dcexffeaafe998d4c88a@mail.gmail.com>
	<65d96fc81002130327q66d8227ao29ae46f8c7ee556a@mail.gmail.com>
	<191c3a031002130722v5c282c00s69f0b51b91e42520@mail.gmail.com>
	<13ca621c1002131120u482f6420xea1f7cad534cbe1a@mail.gmail.com>
	<E2A03ABF-69BF-4F85-9BB9-7CEAA7738E64@freeswitch.org>
Message-ID: <13ca621c1002131326p38eb73bcs444f57b26fdfe381@mail.gmail.com>

On 13 February 2010 19:54, Brian West <brian at freeswitch.org> wrote:
> This is the FAQ entry. ?It applies to EVERYTHING related to load testing.
> /b
> On Feb 13, 2010, at 1:20 PM, Gavin Henry wrote:
>
> http://wiki.freeswitch.org/wiki/FreeSwitch_FAQ#Q:_How_many_concurrent_calls_can_it_support.3F__Any_benchmarks.3F
>

Hi Brian,

OK, then It should clearly say what Anthony said or something like it:

"Please do not ask this question on the mailing lists as you will
always get the same official response from the FreeSWITCH project; "we
only perform benchmarking and confirm these results per FreeSWITCH
deployment, as each deployment will result in varying figures.
Commercial support is available from the project for this task. The
project has learned from experience the dangers of entertaining such
questions and its policy is to not do so over the free public forum."

Thanks.



-- 
http://www.suretecsystems.com/services/openldap/
http://www.suretectelecom.com


From mcampbellsmith at gmail.com  Sat Feb 13 13:36:14 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Sun, 14 Feb 2010 08:36:14 +1100
Subject: [Freeswitch-users] Forbidden using UDP, works with TCP/TLS
In-Reply-To: <C37AB6F3-F712-41B6-9D6D-7A8AE3C6CABB@freeswitch.org>
References: <33c87fa31002101824h21e3bf7lb13b68e8d4fd36a2@mail.gmail.com>
	<2EF27EFC-1634-46A3-B438-2EE99BEF58F1@jerris.com>
	<33c87fa31002110131w3cf6ad00n802669856085a879@mail.gmail.com>
	<CA21231C-108B-4445-A3B9-BF8D94FFE898@jerris.com>
	<33c87fa31002110213q1de0eebfsb950d8725ea9aba@mail.gmail.com>
	<0A905834-850F-4D1F-AEC5-D3A87CE4141B@jerris.com>
	<33c87fa31002121629k74f26807i131841aa4252461b@mail.gmail.com>
	<33c87fa31002130343k7a01e9abt9c33360399f246f8@mail.gmail.com>
	<C37AB6F3-F712-41B6-9D6D-7A8AE3C6CABB@freeswitch.org>
Message-ID: <33c87fa31002131336j7b05db7aof6ba6e3439aebc41@mail.gmail.com>

Thanks Brian.

The full log is pasted here http://pastebin.freeswitch.org/12133



On Sun, Feb 14, 2010 at 2:08 AM, Brian West <brian at freeswitch.org> wrote:
> Can't tell since you keep cutting the lines required to figure this out... FULL console log with FULL sip trace. ?You always cut out what I wanna see. ?Just get a pcap.
>
> /b
>
> On Feb 13, 2010, at 5:43 AM, Mark Campbell-Smith wrote:
>
>> More testing. The device registers successfully to my SIP provider
>> directly using UDP - why would FS be rejecting the registration
>> request?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From mbsip at gazeta.pl  Sat Feb 13 13:54:11 2010
From: mbsip at gazeta.pl (mbsip)
Date: Sat, 13 Feb 2010 22:54:11 +0100
Subject: [Freeswitch-users] mod_python fetching data from mysql for VM
In-Reply-To: <8E4B95E9-3914-4FEE-9C8C-9A5B4C5E2AB2@jerris.com>
References: <28f27f5d1002131203j132c6230sa260358611a16247@mail.gmail.com>
	<8E4B95E9-3914-4FEE-9C8C-9A5B4C5E2AB2@jerris.com>
Message-ID: <28f27f5d1002131354k23a0eaf5jc51b05106c68d719@mail.gmail.com>

Thx for prompt reply.

The main task is to be able to use Mysql db in conjunction with VM
(but not only voicemail_msgs, voicemail_prefs).

Lets imagine sb is calling 1000 and wants to record the message.
According to mod_voicemail settings message should be sent to some
email address.
But the information about user 1000 and his settings like email
address, passwd, quota should be fetched from Mysql db, not from
directory/default/1000.xml.
That's why I am using <action application="python" data="test"/> in my
dialplan to work with python script which in turn should do the magic.
The script should be able to gather all necessery data about user 1000
(like email address in shown example) and use them in VM.

So the problem is how to modify the script to force voicemail app to
use data from DB.
Currently  session.execute("voicemail", "default ${domain} " +
the_dest) is still using .xml files.

Thx,
Maciej.


2010/2/13 Michael Jerris <mike at jerris.com>:
> Can you describe what your trying to accomplish, I don't understand what the goal is. ?What feature are you looking for that does not already exist in mod_voiceamil.
>
> Mike
>
> On Feb 13, 2010, at 3:03 PM, mbsip wrote:
>
>> Hello,
>>
>> I am trying to use mod_python to fetch data from Mysql db (through
>> ODBC) and execute voicemail application.
>> Below a part of my script:
>>
>> db=MySQLdb.connect("localhost","root","","test")
>> ? ? ? Cursor=db.cursor()
>> ? ? ? sql = "SELECT email FROM VM WHERE called_num=%s" % the_dest
>> ? ? ? Cursor.execute(sql)
>> ? ? ? while (1):
>> ? ? ? ? ? ? ? Results = Cursor.fetchone()
>> ? ? ? ? ? ? ? if Results == None:
>> ? ? ? ? ? ? ? ? ? ? ? break
>> ? ? ? ? ? ? ? consoleLog("debug", "Found email " + Results[0] +"\n")
>> ? ? ? ? ? ? ? the_recipient = Results[0]
>> ? ? ? db.close()
>>
>> Now i have email address corresponding with called number. The
>> question is how to use it for voicemail application?
>> So it also means how to omit all /directory/default/....xml, where
>> there are all VM parameters set and use fetched data.
>>
>> ? ? ? session.answer()
>> ? ? ? session.execute("voicemail", "default ${domain} " + the_dest)
>>
>> Is this possible or should I start all VM app in python from the scratch?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From mbsip at gazeta.pl  Sat Feb 13 13:59:13 2010
From: mbsip at gazeta.pl (mbsip)
Date: Sat, 13 Feb 2010 22:59:13 +0100
Subject: [Freeswitch-users] mod_python fetching data from mysql for VM
In-Reply-To: <28f27f5d1002131354k23a0eaf5jc51b05106c68d719@mail.gmail.com>
References: <28f27f5d1002131203j132c6230sa260358611a16247@mail.gmail.com>
	<8E4B95E9-3914-4FEE-9C8C-9A5B4C5E2AB2@jerris.com>
	<28f27f5d1002131354k23a0eaf5jc51b05106c68d719@mail.gmail.com>
Message-ID: <28f27f5d1002131359h164e4077x326f646ff2367ad9@mail.gmail.com>

There is a lack of connection between fatched data and voicemail and I
dont know how to achieve it.

Thx,
Maciej.


2010/2/13 mbsip <mbsip at gazeta.pl>:
> Thx for prompt reply.
>
> The main task is to be able to use Mysql db in conjunction with VM
> (but not only voicemail_msgs, voicemail_prefs).
>
> Lets imagine sb is calling 1000 and wants to record the message.
> According to mod_voicemail settings message should be sent to some
> email address.
> But the information about user 1000 and his settings like email
> address, passwd, quota should be fetched from Mysql db, not from
> directory/default/1000.xml.
> That's why I am using <action application="python" data="test"/> in my
> dialplan to work with python script which in turn should do the magic.
> The script should be able to gather all necessery data about user 1000
> (like email address in shown example) and use them in VM.
>
> So the problem is how to modify the script to force voicemail app to
> use data from DB.
> Currently ?session.execute("voicemail", "default ${domain} " +
> the_dest) is still using .xml files.
>
> Thx,
> Maciej.
>
>
> 2010/2/13 Michael Jerris <mike at jerris.com>:
>> Can you describe what your trying to accomplish, I don't understand what the goal is. ?What feature are you looking for that does not already exist in mod_voiceamil.
>>
>> Mike
>>
>> On Feb 13, 2010, at 3:03 PM, mbsip wrote:
>>
>>> Hello,
>>>
>>> I am trying to use mod_python to fetch data from Mysql db (through
>>> ODBC) and execute voicemail application.
>>> Below a part of my script:
>>>
>>> db=MySQLdb.connect("localhost","root","","test")
>>> ? ? ? Cursor=db.cursor()
>>> ? ? ? sql = "SELECT email FROM VM WHERE called_num=%s" % the_dest
>>> ? ? ? Cursor.execute(sql)
>>> ? ? ? while (1):
>>> ? ? ? ? ? ? ? Results = Cursor.fetchone()
>>> ? ? ? ? ? ? ? if Results == None:
>>> ? ? ? ? ? ? ? ? ? ? ? break
>>> ? ? ? ? ? ? ? consoleLog("debug", "Found email " + Results[0] +"\n")
>>> ? ? ? ? ? ? ? the_recipient = Results[0]
>>> ? ? ? db.close()
>>>
>>> Now i have email address corresponding with called number. The
>>> question is how to use it for voicemail application?
>>> So it also means how to omit all /directory/default/....xml, where
>>> there are all VM parameters set and use fetched data.
>>>
>>> ? ? ? session.answer()
>>> ? ? ? session.execute("voicemail", "default ${domain} " + the_dest)
>>>
>>> Is this possible or should I start all VM app in python from the scratch?
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>


From jmesquita at freeswitch.org  Sat Feb 13 14:00:05 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Sat, 13 Feb 2010 19:00:05 -0300
Subject: [Freeswitch-users] mod_python fetching data from mysql for VM
In-Reply-To: <28f27f5d1002131203j132c6230sa260358611a16247@mail.gmail.com>
References: <28f27f5d1002131203j132c6230sa260358611a16247@mail.gmail.com>
Message-ID: <ebbc2d551002131400k7fbfca7fm4b72fe545ae37e22@mail.gmail.com>

Not sure what you are asking since you provided the answer yourself...

Yes it is possible and that is the way to do it.

JM


On Sat, Feb 13, 2010 at 5:03 PM, mbsip <mbsip at gazeta.pl> wrote:

> Hello,
>
> I am trying to use mod_python to fetch data from Mysql db (through
> ODBC) and execute voicemail application.
> Below a part of my script:
>
> db=MySQLdb.connect("localhost","root","","test")
>       Cursor=db.cursor()
>       sql = "SELECT email FROM VM WHERE called_num=%s" % the_dest
>       Cursor.execute(sql)
>       while (1):
>               Results = Cursor.fetchone()
>               if Results == None:
>                       break
>               consoleLog("debug", "Found email " + Results[0] +"\n")
>               the_recipient = Results[0]
>       db.close()
>
> Now i have email address corresponding with called number. The
> question is how to use it for voicemail application?
> So it also means how to omit all /directory/default/....xml, where
> there are all VM parameters set and use fetched data.
>
>       session.answer()
>       session.execute("voicemail", "default ${domain} " + the_dest)
>
> Is this possible or should I start all VM app in python from the scratch?
>
> Thanks,
> Maciej
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100213/8fc53e30/attachment.html 

From jmesquita at freeswitch.org  Sat Feb 13 14:02:51 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Sat, 13 Feb 2010 19:02:51 -0300
Subject: [Freeswitch-users] Mod_Conference capacity....
In-Reply-To: <13ca621c1002131326p38eb73bcs444f57b26fdfe381@mail.gmail.com>
References: <191c3a031002120856q93e52cep14ee96f1e7ed6d1@mail.gmail.com>
	<324899354.1084631265995845519.JavaMail.root@mail-2.01.com>
	<191c3a031002120944u346a3f7cq779f5c9e0ae31522@mail.gmail.com>
	<13ca621c1002121436n9b44dcexffeaafe998d4c88a@mail.gmail.com>
	<65d96fc81002130327q66d8227ao29ae46f8c7ee556a@mail.gmail.com>
	<191c3a031002130722v5c282c00s69f0b51b91e42520@mail.gmail.com>
	<13ca621c1002131120u482f6420xea1f7cad534cbe1a@mail.gmail.com>
	<E2A03ABF-69BF-4F85-9BB9-7CEAA7738E64@freeswitch.org>
	<13ca621c1002131326p38eb73bcs444f57b26fdfe381@mail.gmail.com>
Message-ID: <ebbc2d551002131402o66f6d641p13e0c7dcef29c8be@mail.gmail.com>

The wiki is public, you know?

JM


On Sat, Feb 13, 2010 at 6:26 PM, Gavin Henry <gavin.henry at gmail.com> wrote:

> On 13 February 2010 19:54, Brian West <brian at freeswitch.org> wrote:
> > This is the FAQ entry.  It applies to EVERYTHING related to load testing.
> > /b
> > On Feb 13, 2010, at 1:20 PM, Gavin Henry wrote:
> >
> >
> http://wiki.freeswitch.org/wiki/FreeSwitch_FAQ#Q:_How_many_concurrent_calls_can_it_support.3F__Any_benchmarks.3F
> >
>
> Hi Brian,
>
> OK, then It should clearly say what Anthony said or something like it:
>
> "Please do not ask this question on the mailing lists as you will
> always get the same official response from the FreeSWITCH project; "we
> only perform benchmarking and confirm these results per FreeSWITCH
> deployment, as each deployment will result in varying figures.
> Commercial support is available from the project for this task. The
> project has learned from experience the dangers of entertaining such
> questions and its policy is to not do so over the free public forum."
>
> Thanks.
>
>
>
> --
> http://www.suretecsystems.com/services/openldap/
> http://www.suretectelecom.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100213/0d9a77ae/attachment-0001.html 

From peder at networkoblivion.com  Sat Feb 13 14:07:45 2010
From: peder at networkoblivion.com (Peder)
Date: Sat, 13 Feb 2010 16:07:45 -0600
Subject: [Freeswitch-users] Mod_Conference capacity....
In-Reply-To: <13ca621c1002131326p38eb73bcs444f57b26fdfe381@mail.gmail.com>
References: <191c3a031002120856q93e52cep14ee96f1e7ed6d1@mail.gmail.com>	<324899354.1084631265995845519.JavaMail.root@mail-2.01.com>	<191c3a031002120944u346a3f7cq779f5c9e0ae31522@mail.gmail.com>	<13ca621c1002121436n9b44dcexffeaafe998d4c88a@mail.gmail.com>	<65d96fc81002130327q66d8227ao29ae46f8c7ee556a@mail.gmail.com>	<191c3a031002130722v5c282c00s69f0b51b91e42520@mail.gmail.com>	<13ca621c1002131120u482f6420xea1f7cad534cbe1a@mail.gmail.com>	<E2A03ABF-69BF-4F85-9BB9-7CEAA7738E64@freeswitch.org>
	<13ca621c1002131326p38eb73bcs444f57b26fdfe381@mail.gmail.com>
Message-ID: <0ad201caacf8$f8c35c20$ea4a1460$@com>

Instead of complaining about this, why don't you benchmark it yourself on
your hardware and post your results on the wiki for others to see in the
future?  Anybody can edit and add info to the wiki.  

By the way, if you stay on the list for any length of time you will see that
about once a week, someone says "I want to do load testing, please tell me
how", or "I did a load test and didn't get x calls per second, why".  I'm
not even involved in the development of this project and it gets old seeing
that same question over and over again.  Really, the standard answer should
be "there is no software limit, it depends on your hardware".  If you want
to know the limits of your setup, test it and see what it is.  


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Gavin
Henry
Sent: Saturday, February 13, 2010 3:26 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Mod_Conference capacity....

On 13 February 2010 19:54, Brian West <brian at freeswitch.org> wrote:
> This is the FAQ entry. ?It applies to EVERYTHING related to load testing.
> /b
> On Feb 13, 2010, at 1:20 PM, Gavin Henry wrote:
>
>
http://wiki.freeswitch.org/wiki/FreeSwitch_FAQ#Q:_How_many_concurrent_calls_
can_it_support.3F__Any_benchmarks.3F
>

Hi Brian,

OK, then It should clearly say what Anthony said or something like it:

"Please do not ask this question on the mailing lists as you will
always get the same official response from the FreeSWITCH project; "we
only perform benchmarking and confirm these results per FreeSWITCH
deployment, as each deployment will result in varying figures.
Commercial support is available from the project for this task. The
project has learned from experience the dangers of entertaining such
questions and its policy is to not do so over the free public forum."

Thanks.



-- 
http://www.suretecsystems.com/services/openldap/
http://www.suretectelecom.com

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



From jmesquita at freeswitch.org  Sat Feb 13 14:20:00 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Sat, 13 Feb 2010 19:20:00 -0300
Subject: [Freeswitch-users] mod_python fetching data from mysql for VM
In-Reply-To: <28f27f5d1002131359h164e4077x326f646ff2367ad9@mail.gmail.com>
References: <28f27f5d1002131203j132c6230sa260358611a16247@mail.gmail.com>
	<8E4B95E9-3914-4FEE-9C8C-9A5B4C5E2AB2@jerris.com>
	<28f27f5d1002131354k23a0eaf5jc51b05106c68d719@mail.gmail.com>
	<28f27f5d1002131359h164e4077x326f646ff2367ad9@mail.gmail.com>
Message-ID: <ebbc2d551002131420v58cf95f2lfc9356f2377728b6@mail.gmail.com>

Maciej,

Take a look at the xml_hooks we have on mod_python. Might do the trick for
you.

http://svn.freeswitch.org/svn/freeswitch/trunk/src/mod/languages/mod_python/python_example.py

JM


On Sat, Feb 13, 2010 at 6:59 PM, mbsip <mbsip at gazeta.pl> wrote:

> There is a lack of connection between fatched data and voicemail and I
> dont know how to achieve it.
>
> Thx,
> Maciej.
>
>
> 2010/2/13 mbsip <mbsip at gazeta.pl>:
> > Thx for prompt reply.
> >
> > The main task is to be able to use Mysql db in conjunction with VM
> > (but not only voicemail_msgs, voicemail_prefs).
> >
> > Lets imagine sb is calling 1000 and wants to record the message.
> > According to mod_voicemail settings message should be sent to some
> > email address.
> > But the information about user 1000 and his settings like email
> > address, passwd, quota should be fetched from Mysql db, not from
> > directory/default/1000.xml.
> > That's why I am using <action application="python" data="test"/> in my
> > dialplan to work with python script which in turn should do the magic.
> > The script should be able to gather all necessery data about user 1000
> > (like email address in shown example) and use them in VM.
> >
> > So the problem is how to modify the script to force voicemail app to
> > use data from DB.
> > Currently  session.execute("voicemail", "default ${domain} " +
> > the_dest) is still using .xml files.
> >
> > Thx,
> > Maciej.
> >
> >
> > 2010/2/13 Michael Jerris <mike at jerris.com>:
> >> Can you describe what your trying to accomplish, I don't understand what
> the goal is.  What feature are you looking for that does not already exist
> in mod_voiceamil.
> >>
> >> Mike
> >>
> >> On Feb 13, 2010, at 3:03 PM, mbsip wrote:
> >>
> >>> Hello,
> >>>
> >>> I am trying to use mod_python to fetch data from Mysql db (through
> >>> ODBC) and execute voicemail application.
> >>> Below a part of my script:
> >>>
> >>> db=MySQLdb.connect("localhost","root","","test")
> >>>       Cursor=db.cursor()
> >>>       sql = "SELECT email FROM VM WHERE called_num=%s" % the_dest
> >>>       Cursor.execute(sql)
> >>>       while (1):
> >>>               Results = Cursor.fetchone()
> >>>               if Results == None:
> >>>                       break
> >>>               consoleLog("debug", "Found email " + Results[0] +"\n")
> >>>               the_recipient = Results[0]
> >>>       db.close()
> >>>
> >>> Now i have email address corresponding with called number. The
> >>> question is how to use it for voicemail application?
> >>> So it also means how to omit all /directory/default/....xml, where
> >>> there are all VM parameters set and use fetched data.
> >>>
> >>>       session.answer()
> >>>       session.execute("voicemail", "default ${domain} " + the_dest)
> >>>
> >>> Is this possible or should I start all VM app in python from the
> scratch?
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100213/6bcb289f/attachment.html 

From gavin.henry at gmail.com  Sat Feb 13 14:33:40 2010
From: gavin.henry at gmail.com (Gavin Henry)
Date: Sat, 13 Feb 2010 22:33:40 +0000
Subject: [Freeswitch-users] Mod_Conference capacity....
In-Reply-To: <ebbc2d551002131402o66f6d641p13e0c7dcef29c8be@mail.gmail.com>
References: <191c3a031002120856q93e52cep14ee96f1e7ed6d1@mail.gmail.com>
	<324899354.1084631265995845519.JavaMail.root@mail-2.01.com>
	<191c3a031002120944u346a3f7cq779f5c9e0ae31522@mail.gmail.com>
	<13ca621c1002121436n9b44dcexffeaafe998d4c88a@mail.gmail.com>
	<65d96fc81002130327q66d8227ao29ae46f8c7ee556a@mail.gmail.com>
	<191c3a031002130722v5c282c00s69f0b51b91e42520@mail.gmail.com>
	<13ca621c1002131120u482f6420xea1f7cad534cbe1a@mail.gmail.com>
	<E2A03ABF-69BF-4F85-9BB9-7CEAA7738E64@freeswitch.org>
	<13ca621c1002131326p38eb73bcs444f57b26fdfe381@mail.gmail.com>
	<ebbc2d551002131402o66f6d641p13e0c7dcef29c8be@mail.gmail.com>
Message-ID: <13ca621c1002131433w7963879bqcab11c278387356d@mail.gmail.com>

2010/2/13 Jo?o Mesquita <jmesquita at freeswitch.org>:
> The wiki is public, you know?
>

Of course I know. I don't represent the project so couldn't obviously
add that so it was in anyway official. I'm not trying to be picky here
only make things clear in the docs for when the next person comes
along and the FreeSWITCH team can just answer an email with the link
to the FAQ on the wiki.

-- 
http://www.suretecsystems.com/services/openldap/
http://www.suretectelecom.com


From paul at apcl.us  Sat Feb 13 14:35:23 2010
From: paul at apcl.us (Paul Levin)
Date: Sat, 13 Feb 2010 17:35:23 -0500
Subject: [Freeswitch-users] can vm settings go in
 conf\directory\default\default.xml?
Message-ID: <4B77292B.6080207@apcl.us>

An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100213/6f607f72/attachment.html 

From gavin.henry at gmail.com  Sat Feb 13 14:38:06 2010
From: gavin.henry at gmail.com (Gavin Henry)
Date: Sat, 13 Feb 2010 22:38:06 +0000
Subject: [Freeswitch-users] Mod_Conference capacity....
In-Reply-To: <0ad201caacf8$f8c35c20$ea4a1460$@com>
References: <191c3a031002120856q93e52cep14ee96f1e7ed6d1@mail.gmail.com>
	<324899354.1084631265995845519.JavaMail.root@mail-2.01.com>
	<191c3a031002120944u346a3f7cq779f5c9e0ae31522@mail.gmail.com>
	<13ca621c1002121436n9b44dcexffeaafe998d4c88a@mail.gmail.com>
	<65d96fc81002130327q66d8227ao29ae46f8c7ee556a@mail.gmail.com>
	<191c3a031002130722v5c282c00s69f0b51b91e42520@mail.gmail.com>
	<13ca621c1002131120u482f6420xea1f7cad534cbe1a@mail.gmail.com>
	<E2A03ABF-69BF-4F85-9BB9-7CEAA7738E64@freeswitch.org>
	<13ca621c1002131326p38eb73bcs444f57b26fdfe381@mail.gmail.com>
	<0ad201caacf8$f8c35c20$ea4a1460$@com>
Message-ID: <13ca621c1002131438t78864be4y2ef6e0f19b14f991@mail.gmail.com>

On 13 February 2010 22:07, Peder <peder at networkoblivion.com> wrote:
> Instead of complaining about this, why don't you benchmark it yourself on
> your hardware and post your results on the wiki for others to see in the
> future? ?Anybody can edit and add info to the wiki.
>
> By the way, if you stay on the list for any length of time you will see that
> about once a week, someone says "I want to do load testing, please tell me
> how", or "I did a load test and didn't get x calls per second, why". ?I'm
> not even involved in the development of this project and it gets old seeing
> that same question over and over again. ?Really, the standard answer should
> be "there is no software limit, it depends on your hardware". ?If you want
> to know the limits of your setup, test it and see what it is.

Did you read any of this thread at all? The OP asked about techniques
to benchmark his kit, and any rough
figures. The projects answer was to hire them for some commercial
support. So anything I post on the wiki is not
official. My suggestion is merely to put the official project stance
on any benchmark questions or post to the
wiki some tips on how to run your own tests with a link to getting
them verified by the team via the commercial
support.

In no way am I complaining, just trying to get this cleared up for the
next time someone asks. That's the whole
point of docs and why I enjoy writing them. Write them once and refer
people to them, fixing doc bugs along the
way.

Gavin.



-- 
http://www.suretecsystems.com/services/openldap/
http://www.suretectelecom.com


From mbsip at gazeta.pl  Sat Feb 13 14:38:17 2010
From: mbsip at gazeta.pl (mbsip)
Date: Sat, 13 Feb 2010 23:38:17 +0100
Subject: [Freeswitch-users] mod_python fetching data from mysql for VM
In-Reply-To: <ebbc2d551002131420v58cf95f2lfc9356f2377728b6@mail.gmail.com>
References: <28f27f5d1002131203j132c6230sa260358611a16247@mail.gmail.com>
	<8E4B95E9-3914-4FEE-9C8C-9A5B4C5E2AB2@jerris.com>
	<28f27f5d1002131354k23a0eaf5jc51b05106c68d719@mail.gmail.com>
	<28f27f5d1002131359h164e4077x326f646ff2367ad9@mail.gmail.com>
	<ebbc2d551002131420v58cf95f2lfc9356f2377728b6@mail.gmail.com>
Message-ID: <28f27f5d1002131438l58718bbat304407f49397c965@mail.gmail.com>

Jo?o,

Thanks for hint, because i don't know how the db fetched data could be
used with voicemail.
I am about to ready it carefully :P

Thanks,
Maciej

2010/2/13 Jo?o Mesquita <jmesquita at freeswitch.org>:
> Maciej,
>
> Take a look at the xml_hooks we have on mod_python. Might do the trick for
> you.
>
> http://svn.freeswitch.org/svn/freeswitch/trunk/src/mod/languages/mod_python/python_example.py
>
> JM
>
>
> On Sat, Feb 13, 2010 at 6:59 PM, mbsip <mbsip at gazeta.pl> wrote:
>>
>> There is a lack of connection between fatched data and voicemail and I
>> dont know how to achieve it.
>>
>> Thx,
>> Maciej.
>>
>>
>> 2010/2/13 mbsip <mbsip at gazeta.pl>:
>> > Thx for prompt reply.
>> >
>> > The main task is to be able to use Mysql db in conjunction with VM
>> > (but not only voicemail_msgs, voicemail_prefs).
>> >
>> > Lets imagine sb is calling 1000 and wants to record the message.
>> > According to mod_voicemail settings message should be sent to some
>> > email address.
>> > But the information about user 1000 and his settings like email
>> > address, passwd, quota should be fetched from Mysql db, not from
>> > directory/default/1000.xml.
>> > That's why I am using <action application="python" data="test"/> in my
>> > dialplan to work with python script which in turn should do the magic.
>> > The script should be able to gather all necessery data about user 1000
>> > (like email address in shown example) and use them in VM.
>> >
>> > So the problem is how to modify the script to force voicemail app to
>> > use data from DB.
>> > Currently ?session.execute("voicemail", "default ${domain} " +
>> > the_dest) is still using .xml files.
>> >
>> > Thx,
>> > Maciej.
>> >
>> >
>> > 2010/2/13 Michael Jerris <mike at jerris.com>:
>> >> Can you describe what your trying to accomplish, I don't understand
>> >> what the goal is. ?What feature are you looking for that does not already
>> >> exist in mod_voiceamil.
>> >>
>> >> Mike
>> >>
>> >> On Feb 13, 2010, at 3:03 PM, mbsip wrote:
>> >>
>> >>> Hello,
>> >>>
>> >>> I am trying to use mod_python to fetch data from Mysql db (through
>> >>> ODBC) and execute voicemail application.
>> >>> Below a part of my script:
>> >>>
>> >>> db=MySQLdb.connect("localhost","root","","test")
>> >>> ? ? ? Cursor=db.cursor()
>> >>> ? ? ? sql = "SELECT email FROM VM WHERE called_num=%s" % the_dest
>> >>> ? ? ? Cursor.execute(sql)
>> >>> ? ? ? while (1):
>> >>> ? ? ? ? ? ? ? Results = Cursor.fetchone()
>> >>> ? ? ? ? ? ? ? if Results == None:
>> >>> ? ? ? ? ? ? ? ? ? ? ? break
>> >>> ? ? ? ? ? ? ? consoleLog("debug", "Found email " + Results[0] +"\n")
>> >>> ? ? ? ? ? ? ? the_recipient = Results[0]
>> >>> ? ? ? db.close()
>> >>>
>> >>> Now i have email address corresponding with called number. The
>> >>> question is how to use it for voicemail application?
>> >>> So it also means how to omit all /directory/default/....xml, where
>> >>> there are all VM parameters set and use fetched data.
>> >>>
>> >>> ? ? ? session.answer()
>> >>> ? ? ? session.execute("voicemail", "default ${domain} " + the_dest)
>> >>>
>> >>> Is this possible or should I start all VM app in python from the
>> >>> scratch?
>> >>
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From dftoro at yahoo.com  Sat Feb 13 14:42:34 2010
From: dftoro at yahoo.com (Diego Toro)
Date: Sat, 13 Feb 2010 14:42:34 -0800 (PST)
Subject: [Freeswitch-users] signaling information SIP INFO
Message-ID: <650320.69699.qm@web33503.mail.mud.yahoo.com>

Hi,

I need send SIP INFO message where the body of the SIP message consists of signaling information to a gateway.

Example:

INFO sip:7007471000 at example.com SIP/2.0 
Via: SIP/2.0/UDP alice.uk.example.com:5060 
From: <sip:7007471234 at alice.uk.example.com>;tag=d3f423d 
To: <sip:7007471000 at example.com>;tag=8942 
Call-ID: 312352 at myphone 
CSeq: 5 INFO 
Content-Length: 24 
Content-Type: application/dtmf-relay 

Signal=16
Duration=160 

 
How I can do that with FreeSwitch events ?


Thank you

Diego Toro
http://lacarretade.blogspot.com/


      


From mbsip at gazeta.pl  Sat Feb 13 14:44:33 2010
From: mbsip at gazeta.pl (mbsip)
Date: Sat, 13 Feb 2010 23:44:33 +0100
Subject: [Freeswitch-users] can vm settings go in
	conf\directory\default\default.xml?
In-Reply-To: <4B77292B.6080207@apcl.us>
References: <4B77292B.6080207@apcl.us>
Message-ID: <28f27f5d1002131444h36df1ba7g7438dd685d8f4281@mail.gmail.com>

So now I am with You Paul. I have the same thoughs and problem :P

Thanks,
Maciej.

2010/2/13 Paul Levin <paul at apcl.us>:
> Thank you for the reply Maciej.
>
> Looks like I made an error in my first email, so let me repeat the problem.
> If I put the lines:
>
> <param name="vm-email-all-messages" value="true"/>
> <param name="vm-keep-local-after-email" value="false"/>
> <param name="vm-attach-file" value="true"/>
>
> into?? conf/directory/default/Bob.xml?? then emails for voice mail are sent.
>
> If I remove those three lines from Bob.xml and put them into
> conf/directory/default.xml?? then emails are not sent.
>
> I would have thought that putting those lines in
> conf/directory/default.xml? would remove the requirement to have them in
> Bob.xml.? No?
>
> ??? Thanks,
> ??? Paul
>
>
>
> From:
> mbsip <mbsip at gazeta.pl>
> Date:
> Sat, 13 Feb 2010 22:23:02 +0100
>
>> Have you tried doing the same with
>> /usr/local/freeswitch/conf/directory/default.xml ?
>
>> Maciej
>
> 2010/2/13 Paul Levin <paul at apcl.us>:
>
>> I'm running FS on Windows (in case that matters here).
>>
>> In conf\directory\default\Bob.xml I have the settings:
>>
>> ??? ? <param name="vm-email-all-messages" value="true"/>
>> ??? ? <param name="vm-keep-local-after-email" value="false"/>
>> ??? ? <param name="vm-attach-file" value="true"/>
>>
>> in addition to other vm- setting that are specific to Bob.? When a voice
>> mail is left for Bob, an email is sent to the configured email address.
>> It
>> is working well.? When the email is sent, I can see in the console the
>> lines:
>>
>> 2010-02-10 15:41:36.949484 [DEBUG]
>> switch_utils.c:633 Emailed file [C:\WINDOWS\TEMP\mail.12658416960810]
>> to [bob at domain.com]
>>
>> 2010-02-10 15:41:36.949484 [DEBUG]
>> mod_voicemail.c:2541 Sending message to bob at domain.com
>>
>> I then move those 3 <param> lines into the default\default.xml file.? Now
>> when a voice mail is left for Bob, an email is not sent and those debug
>> lines do not appear on the console.
>>
>> I don't mind keeping those 3 lines in each user file, but I'm expecting to
>> have about 10,000 users and its kinda silly to repeat those lines in each
>> user's file.? Can't they go in the default.xml file (and have it work)?
>>
>> ??? Thanks,
>> ??? Paul
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From anthony.minessale at gmail.com  Sat Feb 13 14:48:53 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 13 Feb 2010 16:48:53 -0600
Subject: [Freeswitch-users] Mod_Conference capacity....
In-Reply-To: <13ca621c1002131438t78864be4y2ef6e0f19b14f991@mail.gmail.com>
References: <191c3a031002120856q93e52cep14ee96f1e7ed6d1@mail.gmail.com>
	<191c3a031002120944u346a3f7cq779f5c9e0ae31522@mail.gmail.com>
	<13ca621c1002121436n9b44dcexffeaafe998d4c88a@mail.gmail.com>
	<65d96fc81002130327q66d8227ao29ae46f8c7ee556a@mail.gmail.com>
	<191c3a031002130722v5c282c00s69f0b51b91e42520@mail.gmail.com>
	<13ca621c1002131120u482f6420xea1f7cad534cbe1a@mail.gmail.com>
	<E2A03ABF-69BF-4F85-9BB9-7CEAA7738E64@freeswitch.org>
	<13ca621c1002131326p38eb73bcs444f57b26fdfe381@mail.gmail.com>
	<0ad201caacf8$f8c35c20$ea4a1460$@com>
	<13ca621c1002131438t78864be4y2ef6e0f19b14f991@mail.gmail.com>
Message-ID: <191c3a031002131448x59e42e22q6f931dd54834c0a7@mail.gmail.com>

Helping with docs, now there is a topic we all groove on.

On Feb 13, 2010 4:43 PM, "Gavin Henry" <gavin.henry at gmail.com> wrote:

On 13 February 2010 22:07, Peder <peder at networkoblivion.com> wrote:
> Instead of complaining about t...
Did you read any of this thread at all? The OP asked about techniques
to benchmark his kit, and any rough
figures. The projects answer was to hire them for some commercial
support. So anything I post on the wiki is not
official. My suggestion is merely to put the official project stance
on any benchmark questions or post to the
wiki some tips on how to run your own tests with a link to getting
them verified by the team via the commercial
support.

In no way am I complaining, just trying to get this cleared up for the
next time someone asks. That's the whole
point of docs and why I enjoy writing them. Write them once and refer
people to them, fixing doc bugs along the
way.

Gavin.




-- 
http://www.suretecsystems.com/services/openldap/
http://www.suretectelecom.com

_____________...
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100213/cc675829/attachment.html 

From jason at jasonjgw.net  Sat Feb 13 14:53:20 2010
From: jason at jasonjgw.net (Jason White)
Date: Sun, 14 Feb 2010 09:53:20 +1100
Subject: [Freeswitch-users] FreeSWITCH Debian packages - build issues
In-Reply-To: <201002131109.35877.errotan@gmail.com>
References: <20100213030812.GA19108@jdc.jasonjgw.net>
	<201002130957.11633.errotan@gmail.com>
	<20100213090907.GA29452@jdc.jasonjgw.net>
	<201002131109.35877.errotan@gmail.com>
Message-ID: <20100213225320.GA4990@jdc.jasonjgw.net>

Pusk?s Zsolt <errotan at gmail.com> wrote:
 
> I just compiled fs using defaults (just uncommented the openzap line in 
> modues.conf). I don't know how to build a package for that but I can try if 
> you got some instructions how to do that for testing purposes.

The instructions are on the FreeSWITCH wiki for building Debian packages.

The problem appears not to be OpenZap; it's Memcache not finding libpthreads
during its ./configure step and failing at that point.



From rupa at rupa.com  Sat Feb 13 15:03:10 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Sat, 13 Feb 2010 17:03:10 -0600
Subject: [Freeswitch-users] FreeSWITCH Debian packages - build issues
In-Reply-To: <20100213225320.GA4990@jdc.jasonjgw.net>
References: <20100213030812.GA19108@jdc.jasonjgw.net>
	<201002130957.11633.errotan@gmail.com>
	<20100213090907.GA29452@jdc.jasonjgw.net>
	<201002131109.35877.errotan@gmail.com>
	<20100213225320.GA4990@jdc.jasonjgw.net>
Message-ID: <b39273531002131503q1fa89c27n1d7a0406967ed558@mail.gmail.com>

hmmmm... memcache builds for me but I don't do the deb builds, just
./configure (blah blah) && make.  Are we just missing a dependency or is it
configure missing something that is new/different in sid but not testing or
stable?

On Sat, Feb 13, 2010 at 4:53 PM, Jason White <jason at jasonjgw.net> wrote:

> Pusk?s Zsolt <errotan at gmail.com> wrote:
>
> > I just compiled fs using defaults (just uncommented the openzap line in
> > modues.conf). I don't know how to build a package for that but I can try
> if
> > you got some instructions how to do that for testing purposes.
>
> The instructions are on the FreeSWITCH wiki for building Debian packages.
>
> The problem appears not to be OpenZap; it's Memcache not finding
> libpthreads
> during its ./configure step and failing at that point.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
-Rupa
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100213/be19723e/attachment.html 

From brian at freeswitch.org  Sat Feb 13 15:08:28 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 13 Feb 2010 17:08:28 -0600
Subject: [Freeswitch-users] Mod_Conference capacity....
In-Reply-To: <0ad201caacf8$f8c35c20$ea4a1460$@com>
References: <191c3a031002120856q93e52cep14ee96f1e7ed6d1@mail.gmail.com>	<324899354.1084631265995845519.JavaMail.root@mail-2.01.com>	<191c3a031002120944u346a3f7cq779f5c9e0ae31522@mail.gmail.com>	<13ca621c1002121436n9b44dcexffeaafe998d4c88a@mail.gmail.com>	<65d96fc81002130327q66d8227ao29ae46f8c7ee556a@mail.gmail.com>	<191c3a031002130722v5c282c00s69f0b51b91e42520@mail.gmail.com>	<13ca621c1002131120u482f6420xea1f7cad534cbe1a@mail.gmail.com>	<E2A03ABF-69BF-4F85-9BB9-7CEAA7738E64@freeswitch.org>
	<13ca621c1002131326p38eb73bcs444f57b26fdfe381@mail.gmail.com>
	<0ad201caacf8$f8c35c20$ea4a1460$@com>
Message-ID: <8ED2A793-9F68-4064-8593-6D142E099769@freeswitch.org>

We have install guides and yet people still have problems installing FreeSWITCH.  shrug!  More docs are a plus.

/b

On Feb 13, 2010, at 4:07 PM, Peder wrote:

> Instead of complaining about this, why don't you benchmark it yourself on
> your hardware and post your results on the wiki for others to see in the
> future?  Anybody can edit and add info to the wiki.  



From brian at freeswitch.org  Sat Feb 13 15:11:31 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 13 Feb 2010 17:11:31 -0600
Subject: [Freeswitch-users] Forbidden using UDP, works with TCP/TLS
In-Reply-To: <33c87fa31002131336j7b05db7aof6ba6e3439aebc41@mail.gmail.com>
References: <33c87fa31002101824h21e3bf7lb13b68e8d4fd36a2@mail.gmail.com>
	<2EF27EFC-1634-46A3-B438-2EE99BEF58F1@jerris.com>
	<33c87fa31002110131w3cf6ad00n802669856085a879@mail.gmail.com>
	<CA21231C-108B-4445-A3B9-BF8D94FFE898@jerris.com>
	<33c87fa31002110213q1de0eebfsb950d8725ea9aba@mail.gmail.com>
	<0A905834-850F-4D1F-AEC5-D3A87CE4141B@jerris.com>
	<33c87fa31002121629k74f26807i131841aa4252461b@mail.gmail.com>
	<33c87fa31002130343k7a01e9abt9c33360399f246f8@mail.gmail.com>
	<C37AB6F3-F712-41B6-9D6D-7A8AE3C6CABB@freeswitch.org>
	<33c87fa31002131336j7b05db7aof6ba6e3439aebc41@mail.gmail.com>
Message-ID: <9D702362-BB5F-4C0F-9EE7-DABA3B926C36@freeswitch.org>

You know you could have obscured the first part of the IP and not the LAST... kinda removes the ability to tell WHO sent what.  

From that log I guess your password is wrong.

/b

On Feb 13, 2010, at 3:36 PM, Mark Campbell-Smith wrote:

> Thanks Brian.
> 
> The full log is pasted here http://pastebin.freeswitch.org/12133
> 
> 
> 
> On Sun, Feb 14, 2010 at 2:08 AM, Brian West <brian at freeswitch.org> wrote:
>> Can't tell since you keep cutting the lines required to figure this out... FULL console log with FULL sip trace.  You always cut out what I wanna see.  Just get a pcap.
>> 
>> /b
>> 
>> On Feb 13, 2010, at 5:43 AM, Mark Campbell-Smith wrote:
>> 
>>> More testing. The device registers successfully to my SIP provider
>>> directly using UDP - why would FS be rejecting the registration
>>> request?
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From brian at freeswitch.org  Sat Feb 13 15:12:58 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 13 Feb 2010 17:12:58 -0600
Subject: [Freeswitch-users] signaling information SIP INFO
In-Reply-To: <650320.69699.qm@web33503.mail.mud.yahoo.com>
References: <650320.69699.qm@web33503.mail.mud.yahoo.com>
Message-ID: <B2999AB0-F1BF-462F-8891-BB3E9A641B4C@freeswitch.org>

Read "case SWITCH_EVENT_SEND_INFO:" in mod_sofia.c you'll see how.

/b

On Feb 13, 2010, at 4:42 PM, Diego Toro wrote:

> Hi,
> 
> I need send SIP INFO message where the body of the SIP message consists of signaling information to a gateway.
> 
> Example:
> 
> INFO sip:7007471000 at example.com SIP/2.0 
> Via: SIP/2.0/UDP alice.uk.example.com:5060 
> From: <sip:7007471234 at alice.uk.example.com>;tag=d3f423d 
> To: <sip:7007471000 at example.com>;tag=8942 
> Call-ID: 312352 at myphone 
> CSeq: 5 INFO 
> Content-Length: 24 
> Content-Type: application/dtmf-relay 
> 
> Signal=16
> Duration=160 
> 
> 
> How I can do that with FreeSwitch events ?
> 
> 
> Thank you
> 
> Diego Toro
> http://lacarretade.blogspot.com/
> 
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From brian at freeswitch.org  Sat Feb 13 15:14:07 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 13 Feb 2010 17:14:07 -0600
Subject: [Freeswitch-users] bypass_media bug?
In-Reply-To: <201002131552.12289.sos@sokhapkin.dyndns.org>
References: <201002121422.18544.sos@sokhapkin.dyndns.org>
	<201002131552.12289.sos@sokhapkin.dyndns.org>
Message-ID: <5DBF5E29-BAD3-4B8A-B94E-A0986C39551D@freeswitch.org>

Don't you have a jira on this already?  If not open it.. and put the details in... but I think you already have one.

/b

On Feb 13, 2010, at 2:52 PM, Sergey Okhapkin wrote:

> More precisely, FS returns to A leg caller SDP from first 183, but not from 
> final 200 OK.



From brian at freeswitch.org  Sat Feb 13 15:20:34 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 13 Feb 2010 17:20:34 -0600
Subject: [Freeswitch-users] Bridging sessions from two separate lua
	scripts; uuid_bridge? intercept?
In-Reply-To: <eb7797d71002131303q19b657dv91b7ae08ab5b34cf@mail.gmail.com>
References: <eb7797d71002071831w27301c5dva767b82b38c2ce6f@mail.gmail.com>
	<C23AA196-E851-4D5B-B1B5-2C2F32F6BB68@freeswitch.org>
	<eb7797d71002071929pb53e1adue2e91e6bdb598251@mail.gmail.com>
	<eb7797d71002080737k3f8fa5c9m8bf3ef2fcf42881e@mail.gmail.com>
	<D753EDBD-6ED7-43C2-B5F5-A899E3C3D3E2@jerris.com>
	<eb7797d71002121904v5bdfac87w2684c461ca1770cc@mail.gmail.com>
	<191c3a031002130714x2d876657p9c90d768d28a4496@mail.gmail.com>
	<eb7797d71002131018r56e86adnc054a3eaa2c05a96@mail.gmail.com>
	<eb7797d71002131132x2da4d2a0g54b8b8faf19520be@mail.gmail.com>
	<191c3a031002131239v8d0a8e2xd42725a4682b54b2@mail.gmail.com>
	<eb7797d71002131303q19b657dv91b7ae08ab5b34cf@mail.gmail.com>
Message-ID: <AEB40341-C9EB-4DA4-B139-916BC59472AF@freeswitch.org>

You need to stop trying to do everything manually.  You know FreeSWITCH does a lot of things for you so you can just work with the call but you seem to be trying WAY too hard to accomplish your task.  Let FreeSWITCH do the hard work for you... 

api:executeString("intercept -bleg " .. tostring(b_leg_uuid))

This line is 100% WRONG.

Intercept is an application NOT an API call.

Just do session:execute("intercept", "");, Those uuid_media calls are pointless just set the bypass_media_after_bridge variable if you want to go bypass after the bridge.  

I would give you my extended you're trying too hard speech but I think you get the picture.

/b

On Feb 13, 2010, at 3:03 PM, Adam Wilt wrote:

> Does "sigh" mean that it's a problem with FreeSWITCH, or perhaps that I did something stupid?



From lawwton at gmail.com  Sat Feb 13 15:24:13 2010
From: lawwton at gmail.com (Alfredo Quiroga-Villamil)
Date: Sat, 13 Feb 2010 18:24:13 -0500
Subject: [Freeswitch-users] Mod_Conference capacity....
In-Reply-To: <191c3a031002131448x59e42e22q6f931dd54834c0a7@mail.gmail.com>
References: <191c3a031002120856q93e52cep14ee96f1e7ed6d1@mail.gmail.com>
	<13ca621c1002121436n9b44dcexffeaafe998d4c88a@mail.gmail.com>
	<65d96fc81002130327q66d8227ao29ae46f8c7ee556a@mail.gmail.com>
	<191c3a031002130722v5c282c00s69f0b51b91e42520@mail.gmail.com>
	<13ca621c1002131120u482f6420xea1f7cad534cbe1a@mail.gmail.com>
	<E2A03ABF-69BF-4F85-9BB9-7CEAA7738E64@freeswitch.org>
	<13ca621c1002131326p38eb73bcs444f57b26fdfe381@mail.gmail.com>
	<0ad201caacf8$f8c35c20$ea4a1460$@com>
	<13ca621c1002131438t78864be4y2ef6e0f19b14f991@mail.gmail.com>
	<191c3a031002131448x59e42e22q6f931dd54834c0a7@mail.gmail.com>
Message-ID: <5fe6fa8f1002131524k61d18d9ej906b1c52c551065a@mail.gmail.com>

One thing that seems to help a lot is to have a "Success Stories"
page. A lot of the previous emails emphasize the point of benchmarking
FS and hitting it hard with a suite of tools available to do just
that. That all makes sense. The problem is that for a lot of
newcomers, benchmarking doesn't really make a lot of sense right away.
In more details ... the process would be something like this:

a) I need to use a system that's able to do certain things for a
conference for instance (happens to be my case). It needs to be
flexible and do a bunch of things.

b) FS seems like the best candidate out there "today".

c) Get a whatever server and install FS on it. Nice the thing runs, it
seems flexible, good performance, I can't believe my own eyes. Good
product.

d) Let me build an application around it now. Here is an ESL, WEB API,
cool. The sky is the limit.

e) Application is now built.

f) Let's benchmark ... Ok, let me now buy what I think the best server
will be and it'll cost $2000.00.

g) FS is amazing, I am getting these many concurrent calls, active
sessions, things are great for this X system that I bought. I will
scale up and get 5 more of the same type of server.

Now the question is, was that the right choice to make? Having access
to a place where others post their results help you make a better
determination on what kind of platforms to run your system on. Someone
with whom I've had a nice exchange of words in the past threw
something out there that was really helpful ... "Modern machine with
dual core, 64 bit for best results"; if my memory serves me right.
This is the kind of information that would be needed to then go out
and buy the server/s and then do the actual benchmarking and correctly
engineer/architect the platform. If I see that some of the success
stories are using Servers X with Foo CPU/s and bar MEM, then that
would be for me a good indication of the direction or areas to
investigate, go into.

If I then want top of the line tuning, excellent benchmarking and many
more things, yeah having the commercial support and having to pay for
it it's def. not a problem. After all a lot of work and hours go into
developing the product and we are getting it free. That part is
perfectly understood and supported as well as the development of some
more specific features.

Alfredo

On Sat, Feb 13, 2010 at 5:48 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> Helping with docs, now there is a topic we all groove on.
>
> On Feb 13, 2010 4:43 PM, "Gavin Henry" <gavin.henry at gmail.com> wrote:
>
> On 13 February 2010 22:07, Peder <peder at networkoblivion.com> wrote:
>> Instead of complaining about t...
>
> Did you read any of this thread at all? The OP asked about techniques
> to benchmark his kit, and any rough
> figures. The projects answer was to hire them for some commercial
> support. So anything I post on the wiki is not
> official. My suggestion is merely to put the official project stance
> on any benchmark questions or post to the
> wiki some tips on how to run your own tests with a link to getting
> them verified by the team via the commercial
> support.
>
> In no way am I complaining, just trying to get this cleared up for the
> next time someone asks. That's the whole
> point of docs and why I enjoy writing them. Write them once and refer
> people to them, fixing doc bugs along the
> way.
>
> Gavin.
>
>
>
> --
> http://www.suretecsystems.com/services/openldap/
> http://www.suretectelecom.com
>
> _____________...
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From ederwander at gmail.com  Sat Feb 13 15:37:36 2010
From: ederwander at gmail.com (Eder Souza)
Date: Sat, 13 Feb 2010 21:37:36 -0200
Subject: [Freeswitch-users] Send DTMF after call bridge
Message-ID: <622bedea1002131537l314bb5a4g9ed51bafd323c728@mail.gmail.com>

Hi

How i can send DTMF digits after call bridge ??

Example::

<action application="bridge" data="*openzap*/2/1/1831"/>


After my ramal 1831 answer i want send DTMF's how i can make this??


Att,


Eng Eder de Souza
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100213/2bfec53d/attachment.html 

From gavin.henry at gmail.com  Sat Feb 13 15:45:45 2010
From: gavin.henry at gmail.com (Gavin Henry)
Date: Sat, 13 Feb 2010 23:45:45 +0000
Subject: [Freeswitch-users] Mod_Conference capacity....
In-Reply-To: <191c3a031002131448x59e42e22q6f931dd54834c0a7@mail.gmail.com>
References: <191c3a031002120856q93e52cep14ee96f1e7ed6d1@mail.gmail.com>
	<13ca621c1002121436n9b44dcexffeaafe998d4c88a@mail.gmail.com>
	<65d96fc81002130327q66d8227ao29ae46f8c7ee556a@mail.gmail.com>
	<191c3a031002130722v5c282c00s69f0b51b91e42520@mail.gmail.com>
	<13ca621c1002131120u482f6420xea1f7cad534cbe1a@mail.gmail.com>
	<E2A03ABF-69BF-4F85-9BB9-7CEAA7738E64@freeswitch.org>
	<13ca621c1002131326p38eb73bcs444f57b26fdfe381@mail.gmail.com>
	<0ad201caacf8$f8c35c20$ea4a1460$@com>
	<13ca621c1002131438t78864be4y2ef6e0f19b14f991@mail.gmail.com>
	<191c3a031002131448x59e42e22q6f931dd54834c0a7@mail.gmail.com>
Message-ID: <13ca621c1002131545r486c3aeaobea7716af03e0a2a@mail.gmail.com>

On 13 February 2010 22:48, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> Helping with docs, now there is a topic we all groove on.

Agreed! When I find something that isn't there I'll add it or improve it!

Gavin.

-- 
http://www.suretecsystems.com/services/openldap/
http://www.suretectelecom.com


From jmesquita at freeswitch.org  Sat Feb 13 15:49:05 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Sat, 13 Feb 2010 21:49:05 -0200
Subject: [Freeswitch-users] Send DTMF after call bridge
In-Reply-To: <622bedea1002131537l314bb5a4g9ed51bafd323c728@mail.gmail.com>
References: <622bedea1002131537l314bb5a4g9ed51bafd323c728@mail.gmail.com>
Message-ID: <ebbc2d551002131549w43313062w9a63ea20bdf3383a@mail.gmail.com>

Eder, acho que ? isso que vc precisa.

http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_queue_dtmf

Abra?os,
Jo?o Mesquita


On Sat, Feb 13, 2010 at 9:37 PM, Eder Souza <ederwander at gmail.com> wrote:

> Hi
>
> How i can send DTMF digits after call bridge ??
>
> Example::
>
> <action application="bridge" data="*openzap*/2/1/1831"/>
>
>
> After my ramal 1831 answer i want send DTMF's how i can make this??
>
>
> Att,
>
>
> Eng Eder de Souza
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100213/2c9b6dfe/attachment.html 

From ederwander at gmail.com  Sat Feb 13 16:05:57 2010
From: ederwander at gmail.com (Eder Souza)
Date: Sat, 13 Feb 2010 22:05:57 -0200
Subject: [Freeswitch-users] Send DTMF after call bridge
In-Reply-To: <ebbc2d551002131549w43313062w9a63ea20bdf3383a@mail.gmail.com>
References: <622bedea1002131537l314bb5a4g9ed51bafd323c728@mail.gmail.com>
	<ebbc2d551002131549w43313062w9a63ea20bdf3383a@mail.gmail.com>
Message-ID: <622bedea1002131605p7c328ee1v993cbc38fd6da9dc@mail.gmail.com>

*Jo?o Thank you very much *

*brazilian ?  LOl LOl*

*
*

*Muito Obrigado irei testar *

*
*

*Eng Eder de Souza*


2010/2/13 Jo?o Mesquita <jmesquita at freeswitch.org>

> Eder, acho que ? isso que vc precisa.
>
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_queue_dtmf
>
> Abra?os,
> Jo?o Mesquita
>
>
> On Sat, Feb 13, 2010 at 9:37 PM, Eder Souza <ederwander at gmail.com> wrote:
>
>> Hi
>>
>> How i can send DTMF digits after call bridge ??
>>
>> Example::
>>
>> <action application="bridge" data="*openzap*/2/1/1831"/>
>>
>>
>> After my ramal 1831 answer i want send DTMF's how i can make this??
>>
>>
>> Att,
>>
>>
>> Eng Eder de Souza
>>
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100213/3c51a611/attachment.html 

From gmaruzz at celliax.org  Sat Feb 13 19:35:01 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Sun, 14 Feb 2010 04:35:01 +0100
Subject: [Freeswitch-users] mod_skypiax (skype endpoint) new timing
Message-ID: <7b197bef1002131935j3fdf80c4w4050f10bea105d49@mail.gmail.com>

Hello FreeSWITCHers,

I've just committed on svn16640 new timing for mod_skypiax, and I
would like if you guys give it a test in the various use cases.

ciao for now,

-giovanni

-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From jason at jasonjgw.net  Sat Feb 13 20:20:33 2010
From: jason at jasonjgw.net (Jason White)
Date: Sun, 14 Feb 2010 15:20:33 +1100
Subject: [Freeswitch-users] FreeSWITCH Debian packages - build issues
In-Reply-To: <20100213225320.GA4990@jdc.jasonjgw.net>
References: <20100213030812.GA19108@jdc.jasonjgw.net>
	<201002130957.11633.errotan@gmail.com>
	<20100213090907.GA29452@jdc.jasonjgw.net>
	<201002131109.35877.errotan@gmail.com>
	<20100213225320.GA4990@jdc.jasonjgw.net>
Message-ID: <20100214042033.GA19822@jdc.jasonjgw.net>

Just to close this thread for now, FreeSWITCH builds correctly if I remove the
memcache module from the Debian package files.

Maybe when memcache in FreeSWITCH is updated to libmemcache 0.37 (which is in
Debian unstable currently) the autoconf problem, which I'm not inclined to
track down myself at the moement as I don't use memcache, will go away.



From lakindia89 at gmail.com  Sun Feb 14 01:29:04 2010
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Sun, 14 Feb 2010 14:59:04 +0530
Subject: [Freeswitch-users] Is it possible to repeat music in playback
In-Reply-To: <7d79b3931002120323x7a2520fcr1610af69ba4ca51@mail.gmail.com>
References: <13529f9d1002110233g57c8a1a5j3cdd38e22fa6a13d@mail.gmail.com>
	<2FBDB302-2B54-40CC-B694-332854219774@freeswitch.org>
	<191c3a031002110639h3cf14efdvf7715b973fe3caed@mail.gmail.com>
	<7d79b3931002120323x7a2520fcr1610af69ba4ca51@mail.gmail.com>
Message-ID: <7d79b3931002140129k6c9655c8o9a6956966bb22b70@mail.gmail.com>

Hi all,
Any update on this. How to stop an endless_playback???

On Fri, Feb 12, 2010 at 4:53 PM, lakshmanan ganapathy
<lakindia89 at gmail.com>wrote:

> Hi antony,
> Is there any way to stop the endless_playback??
> I tried with break. But it didn't worked!!
>
>
>
> On Thu, Feb 11, 2010 at 8:09 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> or try endless_playback app
>>
>>
>>
>> On Thu, Feb 11, 2010 at 7:26 AM, Brian West <brian at freeswitch.org> wrote:
>>
>>> Why not just use Fifo to hold them?  Or Park the agent and send the
>>> session a message to play music?  You then have options to define loop
>>> count.
>>>
>>> http://wiki.freeswitch.org/wiki/Event_Socket#execute
>>>
>>> /b
>>>
>>> On Feb 11, 2010, at 4:33 AM, Jingwei Yang wrote:
>>>
>>> > Hello,
>>> >
>>> > I've defined a very simple dialplan like the one below and when the
>>> caller is connected to this plan, I hope to keep the call alive and repeat
>>> the music set by playback. How am I able to achieve this?
>>> >
>>> >
>>> >     <extension name="agent_hold">
>>> >       <condition field="destination_number" expression="^33333$">
>>> >         <action application="answer"/>
>>> >         <action application="playback" data="customized/test.wav"/>
>>> >       </condition>
>>> >     </extension>
>>> >
>>> >
>>> > Thanks,
>>> > -Jingwei
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100214/04662e4a/attachment-0001.html 

From vmaruani at interwise.com  Sun Feb 14 01:33:29 2010
From: vmaruani at interwise.com (Victor Maruani)
Date: Sun, 14 Feb 2010 11:33:29 +0200
Subject: [Freeswitch-users] Bypass-media and REFER method
In-Reply-To: <191c3a031002111501y7e5917b8g4faae4b8752e6fc2@mail.gmail.com>
References: <B816EF85C732094BA2016B2112F3DC2B0402C777@isr-brass.interwise.com><B816EF85C732094BA2016B2112F3DC2B040C5CA6@isr-brass.interwise.com><191c3a031002100645n5d827949xdefd24143f10759@mail.gmail.com><B816EF85C732094BA2016B2112F3DC2B040C6061@isr-brass.interwise.com>
	<191c3a031002111501y7e5917b8g4faae4b8752e6fc2@mail.gmail.com>
Message-ID: <B816EF85C732094BA2016B2112F3DC2B040C693B@isr-brass.interwise.com>

Hi,

 

I would say it fails in 2 points:

First in the fact that a "NOTIFY 200 OK" (line 1070) is sent right after
FS gets the REFER.

 

Then in the REINVITE (line 1094) sent to A (10.10.5.19)  just after this
NOTIFY,

This REINVITE contains the SDP of the FS (10.10.5.92) causing the A side
to send media to FS. There will be no REINVITE with SDP of C
(10.10.5.48)

 

But as you say, just afterwards, the REFER action is actually done and C
is invited by FS with the SDP of A.

 

Conclusion :

1)

B is notified of success just after it sent the REFER and is
disconnected.

B may be notified of every step of the connection to C (100 trying...
200 OK) when these actually happen.

What if C is down? Can't FS notify a failure? (didn't test that.)

 

2)

'A' gets to send media to FS Because of a REINVITE which disconnect him
from B (we are in bypass media mode) . 

during the process of REFER, A should be still connected to B from a
media perspective.

The REINVITE is not done at the right time with the right params.

Here, a pseudo bridge (on way voice)  is established when C gets the
INVITE and is sending media to A.

 

A can hear C but C can't hear A after the REFER.

If C was down, A would be "lost" in FS...

 

I believe the correct behavior would be:

B sends REFER.

FS INVITE C 

C replies 100, 180... 200 and FS notifies B in accordance.

Once C has sent 200 OK with its SDP. B is disconnected and A is updated
(REINVITE) with C's SDP.

 

Please share your thoughts, I still don't know if it's a bug or if I
configured something wrong although I don't think so.

Hasn't anyone done that before?

 

Thank you.

 

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Anthony Minessale
Sent: Friday, February 12, 2010 1:02 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Bypass-media and REFER method

 

Where do you think it's failing?
that log shows it get refer, go back to dp, invite to 3341 and then the
other end hangs up at some point.



On Wed, Feb 10, 2010 at 11:01 AM, Victor Maruani
<vmaruani at interwise.com> wrote:

Hi,

 

Logs are on pb 12099

I hope this helps.

Reproduced with revision 16599.

 

A-side (10.10.5.19) is an x-lite registered with extension 1002

B (.5.51) refers to C (.5.48) none are registered.

 

Please refer to previous emails for details of dialplan and what I try
to do...

Let me know if you need more info 

 

Thanks!

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Anthony Minessale
Sent: Wednesday, February 10, 2010 4:46 PM


To: freeswitch-users at lists.freeswitch.org

Subject: Re: [Freeswitch-users] Bypass-media and REFER method

 

update to latest trunk and reproduce your problem with full debug
enabled.

sofia profile internal siptrace on
console loglevel debug

On Wed, Feb 10, 2010 at 4:44 AM, Victor Maruani <vmaruani at interwise.com>
wrote:

Hi,

 

I can't have a blind transfer work properly if I use bypass-media=true.

 

My first message may have been unclear, here I added excerpt from the
dialplan:

 

                <!--- ext MyIVR -->

                 <extension name="Extension MyIVR" continue="false">

                   <condition field="destination_number"
expression="^337$">

                                   <action application="set"
data="bypass_media=true"/>

                                   <action application="bridge"
data="sofia/internal/337 at 10.10.5.51"/>

                   </condition>

                 </extension>

 

 

                <!--- ext MyServ -->

                 <extension name="Extension MyServ " continue="false">

                   <condition field="destination_number"
expression="^3341$">

                                                <action
application="set" data="bypass_media=true"/>

                                   <action application="bridge"
data="sofia/internal/3341 at 10.10.5.48"/>

                   </condition>

                 </extension>   

 

The connection to MyIVR works. Then it sends Refer-to (3341 at ...) which
fails as I described it in the previous mail.

 

I would like to know if the feature has been validated and if I'm
missing something in the configuration.

 

Any help would be very appreciated.

 

Thanks!

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Victor Maruani
Sent: Sunday, February 07, 2010 5:01 PM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Bypass-media and REFER method

 

Hi,

 

I'm trying to do a POC using FS, the goal is to have FS handle REFERs
containing proprietary data. 

I want to have some logic on top of FS and also use the fail over
mechanism.

in short, I have something like this:

(third party) A side  --- FS ---- B side (IVR server)

 

the IVR the sends a REFER to FS. I don't want A to deal with it. 

now say B refers to C, it would be considered as a "group" C1, C2 ... to
which I want FS to failover.

only when one has answered should A be updated (REINVITE) and B notified
and disconnected.

if all fails I would expect B to be notified of the failure and proceed
as I wish without "losing" A.

 

from what I've read FS should be OK for the job but I have a couple
issues:

 

1 ) I have some issues getting FS handle a REFER while in bypass-media
mode.

(I tried with the release and some revisions including latest)

first when I bridge A and B everything is fine and media is bypassed.

When B sends REFER to C:

-           FS immediately NOTIFY B of success and send a reinvite to A
with SDP containing its own media IP/port.

-          then it does INVITE C with A's SDP.

-          B gets disconnected. A is not updated with C's sdp.

so at this point A sends RTP to FS and C sends RTP to A. ...

 

I basically have one extension for B: (set bypass-media and bridge to B)

and another extension to C which does the same actions.

what do you think I do wrong? 

 

 

2 ) how can I catch the REFER and set variables from it? (like ref-by or
ref-to)

in the dial plan I do catch the INVITE sent to C, but how to do it with
the REFER itself?

 

 

thanks for your help!

 

 

Best Regards,

Victor.

 


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
<mailto:MSN%3Aanthony_minessale at hotmail.com> 
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
<mailto:PAYPAL%3Aanthony.minessale at gmail.com> 
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:+19193869900


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
<mailto:MSN%3Aanthony_minessale at hotmail.com> 
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
<mailto:PAYPAL%3Aanthony.minessale at gmail.com> 
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:+19193869900

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100214/35b3de6d/attachment-0001.html 

From vetali100 at gmail.com  Sun Feb 14 02:06:34 2010
From: vetali100 at gmail.com (Vitalii Colosov)
Date: Sun, 14 Feb 2010 12:06:34 +0200
Subject: [Freeswitch-users] Other party does not hear voice when
	connecting with X-Lite
In-Reply-To: <D160C09B-1623-4A0A-BB14-1A1CC04A9276@freeswitch.org>
References: <a858e7691002120133q5e4c806fx309517e4fdec45c5@mail.gmail.com>
	<2F81A680-6A5C-4E3F-9590-FFF6AC4EA80D@freeswitch.org>
	<a858e7691002130247s439177acg23d6a41de3a9d21@mail.gmail.com>
	<a858e7691002130346y2018746o2a71a084a3fd12ee@mail.gmail.com>
	<D160C09B-1623-4A0A-BB14-1A1CC04A9276@freeswitch.org>
Message-ID: <a858e7691002140206q2292e055t3511f98ffa6a22fb@mail.gmail.com>

No, it is done on the different PCs...

Sorry, when I started the topic, I have described the problem how it is
visible from PC of my friend.
Then I tried to reproduce the same on my own PC, and you are right...I was
not able to hear anything as well, not only both party wasn't.
Also, from my PC I was NOT able to hear guitar on test number "9999".

This log reflects the call from my PC.
SIP header sent by XLITE was :

       INVITE sip:9999 at sip.voipsler.com <sip%3A9999 at sip.voipsler.com>SIP/2.0

       Via: SIP/2.0/UDP 192.168.2.10:8342<<<LOCAL IP>>>
;branch=z9hG4bK-d8754z-2f03fe469803ff0f-1---d8754z-;rport

       Contact: <sip:1000 at 192.168.2.10:8342<<<LOCAL IP>>>>




*Yesterday I put STUN server* at the XLITE settings and started to hear
guitar on "9999", and SIP header has changed.

       INVITE sip:1001 at sip.voipsler.com <sip%3A1001 at sip.voipsler.com>SIP/2.0

       Via: SIP/2.0/UDP 192.168.2.10:8216<<<LOCAL
IP>>>;branch=z9hG4bK-d8754z-1758da08c07c1e37-1---d8754z-;rport


       Contact: <sip:1000@<<<*my-global-ip-adress*>>>:8216>

BUT I am still NOT able to hear anything on my PC...

All ports are open on my PC and on FS server.


I tried to use the following option, but no luck: <param
name="aggressive-nat-detection" value="true"/>

http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#NATing_.5Bapply-nat-acl.2C_aggressive-nat-detection.5D


                           "This will enable NAT mode if the network IP/port
from which the request was received differs from the IP/Port combination in
the SIP Via: header, or if the Via: header contains the received parameter
(regardless of what it contains.) "


Do you know what else can I try?
Thank you,
Vitalii



2010/2/13 Brian West <brian at freeswitch.org>

> That should work either way then...are you trying to do this all on the
> same machine?
>
> /b
>
> On Feb 13, 2010, at 5:46 AM, Vitalii Colosov wrote:
>
> > It looks like FS is trying to send MEDIA part to my local ip adress
> instead of global...And Yate sends correct, to global IP...
> >
> > Any ideas?
> >
> > Thank
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100214/e95cf2ff/attachment.html 

From mbsip at gazeta.pl  Sun Feb 14 06:52:03 2010
From: mbsip at gazeta.pl (mbsip)
Date: Sun, 14 Feb 2010 15:52:03 +0100
Subject: [Freeswitch-users] mod_python fetching data from mysql for VM
In-Reply-To: <28f27f5d1002131438l58718bbat304407f49397c965@mail.gmail.com>
References: <28f27f5d1002131203j132c6230sa260358611a16247@mail.gmail.com>
	<8E4B95E9-3914-4FEE-9C8C-9A5B4C5E2AB2@jerris.com>
	<28f27f5d1002131354k23a0eaf5jc51b05106c68d719@mail.gmail.com>
	<28f27f5d1002131359h164e4077x326f646ff2367ad9@mail.gmail.com>
	<ebbc2d551002131420v58cf95f2lfc9356f2377728b6@mail.gmail.com>
	<28f27f5d1002131438l58718bbat304407f49397c965@mail.gmail.com>
Message-ID: <28f27f5d1002140652q656ab6deh40109b223d4f2f4a@mail.gmail.com>

Hi.

Please correct me if my approach is okay.
1. in python.conf.xml
    <param name="xml-handler-script" value="testscript"/>
    <param name="xml-handler-bindings" value="directory"/>
2. in dialplan
    <extension name="public_did_11digits">
      <condition field="destination_number" expression="^(\d{11})$">
        <action application="answer"/>
        <action application="voicemail" data="default $${domain} $1"/>
      </condition>
    </extension>
3.testscript.py (as for now only static entries)
def xml_fetch(params):

        xml = '''
<?xml version="1.0" encoding="UTF-8" standalone="no"?>
<document type="freeswitch/xml">
  <section name="directory">
    <domain name="mydomainHERE">
      <params>
        <param name="dial-string"
value="{presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
      </params>
      <groups>
        <group name="default">
         <users>
          <user id="12345678901">
            <params>
               <param name="vm-mailto" value="email at aaa.com"/
            </params>
          </user>
        </group>
       </users>
      </groups>
    </domain>
  </section>
</document>
'''

        return xml


Unfortunately aforemetnioned configuration does not work at all and
produce following errors:
2010-02-14 17:31:16.878878 [DEBUG] sofia.c:4110 Channel
sofia/internal/100 at 10.10.10.10 entering state [completed][200]
2010-02-14 17:31:16.878878 [DEBUG] switch_core_session.c:638 Send
signal sofia/internal/100 at 10.10.10.10 [BREAK]
2010-02-14 17:31:16.878878 [NOTICE] mod_dptools.c:715 Channel
[sofia/internal/100 at 10.10.10.10] has been answered
EXECUTE sofia/internal/100 at 10.10.10.10 voicemail(default mydomainHERE
12345678901)
2010-02-14 17:31:16.888821 [DEBUG] mod_voicemail.c:728 [default] rwlock
2010-02-14 17:31:16.888821 [NOTICE] mod_python.c:118 Invoking py module: obadamy
2010-02-14 17:31:16.888821 [DEBUG] mod_python.c:188 Call python script
2010-02-14 17:31:16.888821 [DEBUG] mod_python.c:191 Finished calling
python script
2010-02-14 17:31:16.888821 [ERR] mod_python.c:200 Error calling python script
2010-02-14 17:31:16.888821 [WARNING] mod_voicemail.c:2923 Can't find
user [12345678901 at mydomainHERE]
2010-02-14 17:31:16.888821 [DEBUG] switch_ivr_play_say.c:118 No
language specified - Using [en]
2010-02-14 17:31:16.888821 [DEBUG] sofia.c:4110 Channel
sofia/internal/100 at 10.10.10.10 entering state [ready][200]
2010-02-14 17:31:16.888821 [DEBUG] switch_ivr_play_say.c:273 Handle
play-file:[voicemail/vm-goodbye.wav] (en:en)
2010-02-14 17:31:16.888821 [DEBUG] switch_ivr_play_say.c:1158 Codec
Activated L16 at 8000hz 1 channels 20ms

The same output with simplified python script.
def xml_fetch(params):

        xml = '''
<?xml version="1.0" encoding="UTF-8" standalone="no"?>
<document type="freeswitch/xml">
  <section name="directory">
    <domain name="62.29.162.90">
      <user id="48223891699">
        <params>
        </params>
      </user>
    </domain>
  </section>
</document>
'''

        return xml



As for now I have no idea how to solve this, but still digging.
Funny is that dialplan bindings work okay.

Any help pls.
Thx,
Maciej


2010/2/13 mbsip <mbsip at gazeta.pl>:
> Jo?o,
>
> Thanks for hint, because i don't know how the db fetched data could be
> used with voicemail.
> I am about to ready it carefully :P
>
> Thanks,
> Maciej
>
> 2010/2/13 Jo?o Mesquita <jmesquita at freeswitch.org>:
>> Maciej,
>>
>> Take a look at the xml_hooks we have on mod_python. Might do the trick for
>> you.
>>
>> http://svn.freeswitch.org/svn/freeswitch/trunk/src/mod/languages/mod_python/python_example.py
>>
>> JM
>>
>>
>> On Sat, Feb 13, 2010 at 6:59 PM, mbsip <mbsip at gazeta.pl> wrote:
>>>
>>> There is a lack of connection between fatched data and voicemail and I
>>> dont know how to achieve it.
>>>
>>> Thx,
>>> Maciej.
>>>
>>>
>>> 2010/2/13 mbsip <mbsip at gazeta.pl>:
>>> > Thx for prompt reply.
>>> >
>>> > The main task is to be able to use Mysql db in conjunction with VM
>>> > (but not only voicemail_msgs, voicemail_prefs).
>>> >
>>> > Lets imagine sb is calling 1000 and wants to record the message.
>>> > According to mod_voicemail settings message should be sent to some
>>> > email address.
>>> > But the information about user 1000 and his settings like email
>>> > address, passwd, quota should be fetched from Mysql db, not from
>>> > directory/default/1000.xml.
>>> > That's why I am using <action application="python" data="test"/> in my
>>> > dialplan to work with python script which in turn should do the magic.
>>> > The script should be able to gather all necessery data about user 1000
>>> > (like email address in shown example) and use them in VM.
>>> >
>>> > So the problem is how to modify the script to force voicemail app to
>>> > use data from DB.
>>> > Currently ?session.execute("voicemail", "default ${domain} " +
>>> > the_dest) is still using .xml files.
>>> >
>>> > Thx,
>>> > Maciej.
>>> >
>>> >
>>> > 2010/2/13 Michael Jerris <mike at jerris.com>:
>>> >> Can you describe what your trying to accomplish, I don't understand
>>> >> what the goal is. ?What feature are you looking for that does not already
>>> >> exist in mod_voiceamil.
>>> >>
>>> >> Mike
>>> >>
>>> >> On Feb 13, 2010, at 3:03 PM, mbsip wrote:
>>> >>
>>> >>> Hello,
>>> >>>
>>> >>> I am trying to use mod_python to fetch data from Mysql db (through
>>> >>> ODBC) and execute voicemail application.
>>> >>> Below a part of my script:
>>> >>>
>>> >>> db=MySQLdb.connect("localhost","root","","test")
>>> >>> ? ? ? Cursor=db.cursor()
>>> >>> ? ? ? sql = "SELECT email FROM VM WHERE called_num=%s" % the_dest
>>> >>> ? ? ? Cursor.execute(sql)
>>> >>> ? ? ? while (1):
>>> >>> ? ? ? ? ? ? ? Results = Cursor.fetchone()
>>> >>> ? ? ? ? ? ? ? if Results == None:
>>> >>> ? ? ? ? ? ? ? ? ? ? ? break
>>> >>> ? ? ? ? ? ? ? consoleLog("debug", "Found email " + Results[0] +"\n")
>>> >>> ? ? ? ? ? ? ? the_recipient = Results[0]
>>> >>> ? ? ? db.close()
>>> >>>
>>> >>> Now i have email address corresponding with called number. The
>>> >>> question is how to use it for voicemail application?
>>> >>> So it also means how to omit all /directory/default/....xml, where
>>> >>> there are all VM parameters set and use fetched data.
>>> >>>
>>> >>> ? ? ? session.answer()
>>> >>> ? ? ? session.execute("voicemail", "default ${domain} " + the_dest)
>>> >>>
>>> >>> Is this possible or should I start all VM app in python from the
>>> >>> scratch?
>>> >>
>>> >>
>>> >> _______________________________________________
>>> >> FreeSWITCH-users mailing list
>>> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>
>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> http://www.freeswitch.org
>>> >>
>>> >
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>


From errotan at gmail.com  Sun Feb 14 06:52:31 2010
From: errotan at gmail.com (=?iso-8859-1?q?Pusk=E1s_Zsolt?=)
Date: Sun, 14 Feb 2010 15:52:31 +0100
Subject: [Freeswitch-users] mod_skypiax (skype endpoint) new timing
In-Reply-To: <7b197bef1002131935j3fdf80c4w4050f10bea105d49@mail.gmail.com>
References: <7b197bef1002131935j3fdf80c4w4050f10bea105d49@mail.gmail.com>
Message-ID: <201002141552.31159.errotan@gmail.com>

2010. febru?r 14. 04.35.01 Giovanni Maruzzelli d?tummal ezt ?rta:
> Hello FreeSWITCHers,
> 
> I've just committed on svn16640 new timing for mod_skypiax, and I
> would like if you guys give it a test in the various use cases.
> 
> ciao for now,
> 
> -giovanni
> 

It is not really working:

2010-02-14 15:37:39.182686 [ERR] skypiax_protocol.c:196 rev 16640M[(nil)|37     
][ERRORA  196  ][gun_at_koli][-1, 0, 0] Skype got ERROR: |||ERROR 589 ALTER 
CALL: unable to alter input/output|||
2010-02-14 15:37:39.182686 [ERR] skypiax_protocol.c:198 rev 16640M[(nil)|37     
][ERRORA  198  ][gun_at_koli][-1, 0,16] skype_call now is DOWN
2010-02-14 15:37:39.182686 [DEBUG] mod_skypiax.c:1068 rev 16640M[(nil)|37     
][DEBUG_SKYPE  1068 ][gun_at_koli][-1, 1,16] skype call ended                                                                                                 
2010-02-14 15:37:39.182686 [DEBUG] mod_skypiax.c:1085 rev 16640M[(nil)|37     
][DEBUG_SKYPE  1085 ][gun_at_koli][-1, 1,16] no session


Also can't unload mod_skypiax because the channels are up:

freeswitch at internal> reload mod_skypiax
-ERR unloading module [Module in use.]

2010-02-14 15:40:13.570678 [WARNING] switch_loadable_module.c:1268 Module 
mod_skypiax is in use, cannot unload.
freeswitch at internal> show channels
uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,write_codec,write_rate,secure,hostname,presence_id,presence_data
4f302086-1976-11df-a1d8-47a9ffb132d9,inbound,2010-02-14 
15:36:16,1266158176,skypiax/gun_at_koli,CS_EXECUTE,Pusk?s 
Zsolt,errotan,,4219,echo,,XML,default,L16,16000,L16,16000,,pc,,
80007580-1976-11df-a1d9-47a9ffb132d9,inbound,2010-02-14 
15:37:38,1266158258,skypiax/gun_at_koli,CS_EXECUTE,Pusk?s 
Zsolt,errotan,,4219,echo,,XML,default,L16,16000,L16,16000,,pc,,

2 total.

Have to do a fsctl shutdown to kill those channels.

My home skype (errotan) client displays: Remote sound problem. when I try to 
call the other (gun_at_koli).


From anthony.minessale at gmail.com  Sun Feb 14 06:54:27 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sun, 14 Feb 2010 08:54:27 -0600
Subject: [Freeswitch-users] Bypass-media and REFER method
In-Reply-To: <191c3a031002140645s358688c8g68280267d5d650d3@mail.gmail.com>
References: <B816EF85C732094BA2016B2112F3DC2B0402C777@isr-brass.interwise.com>
	<B816EF85C732094BA2016B2112F3DC2B040C5CA6@isr-brass.interwise.com>
	<191c3a031002100645n5d827949xdefd24143f10759@mail.gmail.com>
	<B816EF85C732094BA2016B2112F3DC2B040C6061@isr-brass.interwise.com>
	<191c3a031002111501y7e5917b8g4faae4b8752e6fc2@mail.gmail.com>
	<B816EF85C732094BA2016B2112F3DC2B040C693B@isr-brass.interwise.com>
	<191c3a031002140645s358688c8g68280267d5d650d3@mail.gmail.com>
Message-ID: <191c3a031002140654x37b65f1alc45907b545249cb4@mail.gmail.com>

We don't support the series of 100,180,200 in the notifies that is typically
a pure sip pbx feature.
We are a b2b and protocol agnostic softswitch.
When you do refer to a sofia leg you are talking to that leg independantly.
We do not track the progress of what the channel does once you transfer it,
instead we send the channel back to the dialplan and accept the refer.  This
operation if sucessful will generate 202.
The resume media on hold and bypass after att xfer sofia options or
bypass_media_after_bridge var may be what you need if you want no media on
fs.
We have to make some sacrifices on sip madness to gain some general
flexibility and call volume. =/

On Feb 14, 2010 3:39 AM, "Victor Maruani" <vmaruani at interwise.com> wrote:

 Hi,



I would say it fails in 2 points:

First in the fact that a "NOTIFY 200 OK" (line 1070) is sent right after FS
gets the REFER.



Then in the REINVITE (line 1094) sent to A (10.10.5.19)  just after this
NOTIFY,

This REINVITE contains the SDP of the FS (10.10.5.92) causing the A side to
send media to FS. There will be no REINVITE with SDP of C (10.10.5.48)



But as you say, just afterwards, the REFER action is actually done and C is
invited by FS with the SDP of A.



Conclusion :

1)

B is notified of success just after it sent the REFER and is disconnected.

B may be notified of every step of the connection to C (100 trying? 200 OK)
when these actually happen.

What if C is down? Can't FS notify a failure? (didn't test that.)



2)

'A' gets to send media to FS Because of a REINVITE which disconnect him from
B (we are in bypass media mode) .

during the process of REFER, A should be still connected to B from a media
perspective.

The REINVITE is not done at the right time with the right params.

Here, a pseudo bridge (on way voice)  is established when C gets the INVITE
and is sending media to A.



A can hear C but C can't hear A after the REFER.

If C was down, A would be "lost" in FS?



I believe the correct behavior would be:

B sends REFER.

FS INVITE C

C replies 100, 180? 200 and FS notifies B in accordance.

Once C has sent 200 OK with its SDP. B is disconnected and A is updated
(REINVITE) with C's SDP.



Please share your thoughts, I still don't know if it's a bug or if I
configured something wrong although I don't think so.

Hasn't anyone done that before?



Thank you.







*From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony
Minessale
*Sent:* Friday, February 12, 2010 1:02 AM


To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Bypass-media and REFER me...





Where do you think it's failing?
that log shows it get refer, go back to dp, invite to 3341 and...

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100214/086a6d4c/attachment-0001.html 

From anthony.minessale at gmail.com  Sun Feb 14 07:04:09 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sun, 14 Feb 2010 09:04:09 -0600
Subject: [Freeswitch-users] Other party does not hear voice when
	connecting with X-Lite
In-Reply-To: <a858e7691002140206q2292e055t3511f98ffa6a22fb@mail.gmail.com>
References: <a858e7691002120133q5e4c806fx309517e4fdec45c5@mail.gmail.com>
	<2F81A680-6A5C-4E3F-9590-FFF6AC4EA80D@freeswitch.org>
	<a858e7691002130247s439177acg23d6a41de3a9d21@mail.gmail.com>
	<a858e7691002130346y2018746o2a71a084a3fd12ee@mail.gmail.com>
	<D160C09B-1623-4A0A-BB14-1A1CC04A9276@freeswitch.org>
	<a858e7691002140206q2292e055t3511f98ffa6a22fb@mail.gmail.com>
Message-ID: <191c3a031002140704g705bfc73rd8dd103f3d846062@mail.gmail.com>

You need to describe this again its too confusing now.
List each device, freeswitch, the phones and which ip and combo of addrs it
uses with the topology clearly stated.
Your attempt to simplify your explanation is actually making it harder to
follow.
Also consider a debug/sip trace as well.

Include
sofia status profile default.

Then capture a test call after entering these commands.

console loglevel debug.
sofa profile internal siptrace on

On Feb 14, 2010 4:12 AM, "Vitalii Colosov" <vetali100 at gmail.com> wrote:

No, it is done on the different PCs...

Sorry, when I started the topic, I have described the problem how it is
visible from PC of my friend.
Then I tried to reproduce the same on my own PC, and you are right...I was
not able to hear anything as well, not only both party wasn't.
Also, from my PC I was NOT able to hear guitar on test number "9999".

This log reflects the call from my PC.
SIP header sent by XLITE was :

       INVITE sip:9999 at sip.voipsler.com <sip%3A9999 at sip.voipsler.com>SIP/2.0

       Via: SIP/2.0/UDP 192.168.2.10:8342<<<LOCAL IP>>>
;branch=z9hG4bK-d8754z-2f03fe469803ff0f-1---d8754z-;rport

       Contact: <sip:1000 at 192.168.2.10:8342<<<LOCAL IP>>>>




*Yesterday I put STUN server* at the XLITE settings and started to hear
guitar on "9999", and SIP header has changed.

       INVITE sip:1001 at sip.voipsler.com <sip%3A1001 at sip.voipsler.com>SIP/2.0

       Via: SIP/2.0/UDP 192.168.2.10:8216<<<LOCAL
IP>>>;branch=z9hG4bK-d8754z-1758da08c07c1e37-1---d8754z-;rport


       Contact: <sip:1000@<<<*my-global-ip-adress*>>>:8216>

BUT I am still NOT able to hear anything on my PC...

All ports are open on my PC and on FS server.


I tried to use the following option, but no luck: <param
name="aggressive-nat-detection" value="true"/>

http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#NATing_.5Bapply-nat-acl.2C_aggressive-nat-detection.5D


                           "This will enable NAT mode if the network IP/port
from which the request was received differs from the IP/Port combination in
the SIP Via: header, or if the Via: header contains the received parameter
(regardless of what it contains.) "


Do you know what else can I try?
Thank you,
Vitalii



2010/2/13 Brian West <brian at freeswitch.org>


>
> That should work either way then...are you trying to do this all on the
same machine?
>
> /b
>
...


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100214/f7c82c8e/attachment.html 

From anthony.minessale at gmail.com  Sun Feb 14 07:05:55 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sun, 14 Feb 2010 09:05:55 -0600
Subject: [Freeswitch-users] Is it possible to repeat music in playback
In-Reply-To: <7d79b3931002140129k6c9655c8o9a6956966bb22b70@mail.gmail.com>
References: <13529f9d1002110233g57c8a1a5j3cdd38e22fa6a13d@mail.gmail.com>
	<2FBDB302-2B54-40CC-B694-332854219774@freeswitch.org>
	<191c3a031002110639h3cf14efdvf7715b973fe3caed@mail.gmail.com>
	<7d79b3931002120323x7a2520fcr1610af69ba4ca51@mail.gmail.com>
	<7d79b3931002140129k6c9655c8o9a6956966bb22b70@mail.gmail.com>
Message-ID: <191c3a031002140705m525a8753h67ef2609ff51fe78@mail.gmail.com>

There is no way, that's why its called endless playback.
You could put in a feature request or a bounty and if it was deemed a
sensible req, it could be added.

On Feb 14, 2010 3:37 AM, "lakshmanan ganapathy" <lakindia89 at gmail.com>
wrote:

Hi all,
Any update on this. How to stop an endless_playback???



On Fri, Feb 12, 2010 at 4:53 PM, lakshmanan ganapathy <lakindia89 at gmail.com>
wrote:
>
> Hi antony,...

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100214/00d9ccdc/attachment.html 

From errotan at gmail.com  Sun Feb 14 07:09:18 2010
From: errotan at gmail.com (=?iso-8859-1?q?Pusk=E1s_Zsolt?=)
Date: Sun, 14 Feb 2010 16:09:18 +0100
Subject: [Freeswitch-users] mod_skypiax (skype endpoint) new timing
In-Reply-To: <201002141552.31159.errotan@gmail.com>
References: <7b197bef1002131935j3fdf80c4w4050f10bea105d49@mail.gmail.com>
	<201002141552.31159.errotan@gmail.com>
Message-ID: <201002141609.18711.errotan@gmail.com>

2010. febru?r 14. 15.52.31 Pusk?s Zsolt d?tummal ezt ?rta:
> 2010. febru?r 14. 04.35.01 Giovanni Maruzzelli d?tummal ezt ?rta:
> > Hello FreeSWITCHers,
> >
> > I've just committed on svn16640 new timing for mod_skypiax, and I
> > would like if you guys give it a test in the various use cases.
> >
> > ciao for now,
> >
> > -giovanni
> 
> It is not really working:
> 
> 2010-02-14 15:37:39.182686 [ERR] skypiax_protocol.c:196 rev 16640M[(nil)|37
> ][ERRORA  196  ][gun_at_koli][-1, 0, 0] Skype got ERROR: |||ERROR 589 ALTER
> CALL: unable to alter input/output|||
> 2010-02-14 15:37:39.182686 [ERR] skypiax_protocol.c:198 rev 16640M[(nil)|37
> ][ERRORA  198  ][gun_at_koli][-1, 0,16] skype_call now is DOWN
> 2010-02-14 15:37:39.182686 [DEBUG] mod_skypiax.c:1068 rev 16640M[(nil)|37
> ][DEBUG_SKYPE  1068 ][gun_at_koli][-1, 1,16] skype call ended
> 2010-02-14 15:37:39.182686 [DEBUG] mod_skypiax.c:1085 rev 16640M[(nil)|37
> ][DEBUG_SKYPE  1085 ][gun_at_koli][-1, 1,16] no session
> 
> 
> Also can't unload mod_skypiax because the channels are up:
> 
> freeswitch at internal> reload mod_skypiax
> -ERR unloading module [Module in use.]
> 
> 2010-02-14 15:40:13.570678 [WARNING] switch_loadable_module.c:1268 Module
> mod_skypiax is in use, cannot unload.
> freeswitch at internal> show channels
> uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,de
> st,application,application_data,dialplan,context,read_codec,read_rate,write
> _codec,write_rate,secure,hostname,presence_id,presence_data
>  4f302086-1976-11df-a1d8-47a9ffb132d9,inbound,2010-02-14
> 15:36:16,1266158176,skypiax/gun_at_koli,CS_EXECUTE,Pusk?s
> Zsolt,errotan,,4219,echo,,XML,default,L16,16000,L16,16000,,pc,,
> 80007580-1976-11df-a1d9-47a9ffb132d9,inbound,2010-02-14
> 15:37:38,1266158258,skypiax/gun_at_koli,CS_EXECUTE,Pusk?s
> Zsolt,errotan,,4219,echo,,XML,default,L16,16000,L16,16000,,pc,,
> 
> 2 total.
> 
> Have to do a fsctl shutdown to kill those channels.
> 
> My home skype (errotan) client displays: Remote sound problem. when I try
>  to call the other (gun_at_koli).
> 

Ooops sorry the user running the skype client were not in the audio group 
therefore don't have access to snd_dummy.

Works now! Tested on Debian "Lenny" x86

:)


From mbsip at gazeta.pl  Sun Feb 14 08:01:47 2010
From: mbsip at gazeta.pl (mbsip)
Date: Sun, 14 Feb 2010 17:01:47 +0100
Subject: [Freeswitch-users] can vm settings go in
	conf\directory\default\default.xml?
In-Reply-To: <28f27f5d1002131444h36df1ba7g7438dd685d8f4281@mail.gmail.com>
References: <4B77292B.6080207@apcl.us>
	<28f27f5d1002131444h36df1ba7g7438dd685d8f4281@mail.gmail.com>
Message-ID: <28f27f5d1002140801r1e952d91l85b4b643350134e2@mail.gmail.com>

Paul,

Maybe you should think about providing dynamic directory information
by using DB.
>From my point of view, Its much easier to manage DB than .xml files.

Thx,
Maciej.

2010/2/13 mbsip <mbsip at gazeta.pl>:
> So now I am with You Paul. I have the same thoughs and problem :P
>
> Thanks,
> Maciej.
>
> 2010/2/13 Paul Levin <paul at apcl.us>:
>> Thank you for the reply Maciej.
>>
>> Looks like I made an error in my first email, so let me repeat the problem.
>> If I put the lines:
>>
>> <param name="vm-email-all-messages" value="true"/>
>> <param name="vm-keep-local-after-email" value="false"/>
>> <param name="vm-attach-file" value="true"/>
>>
>> into?? conf/directory/default/Bob.xml?? then emails for voice mail are sent.
>>
>> If I remove those three lines from Bob.xml and put them into
>> conf/directory/default.xml?? then emails are not sent.
>>
>> I would have thought that putting those lines in
>> conf/directory/default.xml? would remove the requirement to have them in
>> Bob.xml.? No?
>>
>> ??? Thanks,
>> ??? Paul
>>
>>
>>
>> From:
>> mbsip <mbsip at gazeta.pl>
>> Date:
>> Sat, 13 Feb 2010 22:23:02 +0100
>>
>>> Have you tried doing the same with
>>> /usr/local/freeswitch/conf/directory/default.xml ?
>>
>>> Maciej
>>
>> 2010/2/13 Paul Levin <paul at apcl.us>:
>>
>>> I'm running FS on Windows (in case that matters here).
>>>
>>> In conf\directory\default\Bob.xml I have the settings:
>>>
>>> ??? ? <param name="vm-email-all-messages" value="true"/>
>>> ??? ? <param name="vm-keep-local-after-email" value="false"/>
>>> ??? ? <param name="vm-attach-file" value="true"/>
>>>
>>> in addition to other vm- setting that are specific to Bob.? When a voice
>>> mail is left for Bob, an email is sent to the configured email address.
>>> It
>>> is working well.? When the email is sent, I can see in the console the
>>> lines:
>>>
>>> 2010-02-10 15:41:36.949484 [DEBUG]
>>> switch_utils.c:633 Emailed file [C:\WINDOWS\TEMP\mail.12658416960810]
>>> to [bob at domain.com]
>>>
>>> 2010-02-10 15:41:36.949484 [DEBUG]
>>> mod_voicemail.c:2541 Sending message to bob at domain.com
>>>
>>> I then move those 3 <param> lines into the default\default.xml file.? Now
>>> when a voice mail is left for Bob, an email is not sent and those debug
>>> lines do not appear on the console.
>>>
>>> I don't mind keeping those 3 lines in each user file, but I'm expecting to
>>> have about 10,000 users and its kinda silly to repeat those lines in each
>>> user's file.? Can't they go in the default.xml file (and have it work)?
>>>
>>> ??? Thanks,
>>> ??? Paul
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>


From bottleman at icf.org.ru  Sun Feb 14 08:09:43 2010
From: bottleman at icf.org.ru (Georgiewskiy Yuriy)
Date: Sun, 14 Feb 2010 19:09:43 +0300 (MSK)
Subject: [Freeswitch-users] FreeSWITCH Debian packages - build issues
In-Reply-To: <20100213030812.GA19108@jdc.jasonjgw.net>
References: <20100213030812.GA19108@jdc.jasonjgw.net>
Message-ID: <alpine.DEB.2.00.1002141906400.26475@srv.icf.org.ru>

On 2010-02-13 14:08 +1100, Jason White wrote Freeswitch-users:

As i see build on debian stable is broken, it's not build many modules, 
for example mod_say_xx, play, etc, actually there no error on build process, 
but resulted packages not usable becouse many modules not present in it.

JW>Has anyone successfully built the Debian packages recently from the source
JW>repository?
JW>
JW>The problem I'm experiencing is that openzap is specified to be built, but it
JW>is never actually compiled. Consequently, the packages can't be created (the
JW>process fails due to the missing mod_openzap.so file).
JW>
JW>I don't need openzap; I can easily comment it out, but I also think the
JW>supplied package files should work as is.
JW>
JW>first step: confirm whether my experience under Debian Sid is shared by others
JW>using different versions of Debian or Ubuntu.
JW>
JW>
JW>_______________________________________________
JW>FreeSWITCH-users mailing list
JW>FreeSWITCH-users at lists.freeswitch.org
JW>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
JW>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
JW>http://www.freeswitch.org
JW>

C ?????????                       With Best Regards
???????????? ????.                Georgiewskiy Yuriy
+7 4872 711666                    +7 4872 711666
???? +7 4872 711143               fax +7 4872 711143
???????? ??? "?? ?? ??????"       IT Service Ltd
http://nkoort.ru                  http://nkoort.ru
JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
YG129-RIPE                        YG129-RIPE

From gmaruzz at celliax.org  Sun Feb 14 08:40:24 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Sun, 14 Feb 2010 17:40:24 +0100
Subject: [Freeswitch-users] mod_skypiax (skype endpoint) new timing
In-Reply-To: <201002141609.18711.errotan@gmail.com>
References: <7b197bef1002131935j3fdf80c4w4050f10bea105d49@mail.gmail.com> 
	<201002141552.31159.errotan@gmail.com>
	<201002141609.18711.errotan@gmail.com>
Message-ID: <7b197bef1002140840i6dfbdaecu26c1ae7e1f1a0633@mail.gmail.com>

Thanks Pusk?s,

any problem of growing delay in sip->fs->skype or skype->fs->sip, or
conferences, or whatever?

Also, snd-dummy is now more compatible with kernels at 1000HZ (eg:
centOS), although I believe Lenny is 100HZ.

-giovanni

On Sun, Feb 14, 2010 at 4:09 PM, Pusk?s Zsolt <errotan at gmail.com> wrote:
> 2010. febru?r 14. 15.52.31 Pusk?s Zsolt d?tummal ezt ?rta:
>> 2010. febru?r 14. 04.35.01 Giovanni Maruzzelli d?tummal ezt ?rta:
>> > Hello FreeSWITCHers,
>> >
>> > I've just committed on svn16640 new timing for mod_skypiax, and I
>> > would like if you guys give it a test in the various use cases.
>> >
>> > ciao for now,
>> >
>> > -giovanni
>>
>> It is not really working:
>>
>> 2010-02-14 15:37:39.182686 [ERR] skypiax_protocol.c:196 rev 16640M[(nil)|37
>> ][ERRORA ?196 ?][gun_at_koli][-1, 0, 0] Skype got ERROR: |||ERROR 589 ALTER
>> CALL: unable to alter input/output|||
>> 2010-02-14 15:37:39.182686 [ERR] skypiax_protocol.c:198 rev 16640M[(nil)|37
>> ][ERRORA ?198 ?][gun_at_koli][-1, 0,16] skype_call now is DOWN
>> 2010-02-14 15:37:39.182686 [DEBUG] mod_skypiax.c:1068 rev 16640M[(nil)|37
>> ][DEBUG_SKYPE ?1068 ][gun_at_koli][-1, 1,16] skype call ended
>> 2010-02-14 15:37:39.182686 [DEBUG] mod_skypiax.c:1085 rev 16640M[(nil)|37
>> ][DEBUG_SKYPE ?1085 ][gun_at_koli][-1, 1,16] no session
>>
>>
>> Also can't unload mod_skypiax because the channels are up:
>>
>> freeswitch at internal> reload mod_skypiax
>> -ERR unloading module [Module in use.]
>>
>> 2010-02-14 15:40:13.570678 [WARNING] switch_loadable_module.c:1268 Module
>> mod_skypiax is in use, cannot unload.
>> freeswitch at internal> show channels
>> uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,de
>> st,application,application_data,dialplan,context,read_codec,read_rate,write
>> _codec,write_rate,secure,hostname,presence_id,presence_data
>> ?4f302086-1976-11df-a1d8-47a9ffb132d9,inbound,2010-02-14
>> 15:36:16,1266158176,skypiax/gun_at_koli,CS_EXECUTE,Pusk?s
>> Zsolt,errotan,,4219,echo,,XML,default,L16,16000,L16,16000,,pc,,
>> 80007580-1976-11df-a1d9-47a9ffb132d9,inbound,2010-02-14
>> 15:37:38,1266158258,skypiax/gun_at_koli,CS_EXECUTE,Pusk?s
>> Zsolt,errotan,,4219,echo,,XML,default,L16,16000,L16,16000,,pc,,
>>
>> 2 total.
>>
>> Have to do a fsctl shutdown to kill those channels.
>>
>> My home skype (errotan) client displays: Remote sound problem. when I try
>> ?to call the other (gun_at_koli).
>>
>
> Ooops sorry the user running the skype client were not in the audio group
> therefore don't have access to snd_dummy.
>
> Works now! Tested on Debian "Lenny" x86
>
> :)
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From gmaruzz at celliax.org  Sun Feb 14 08:42:57 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Sun, 14 Feb 2010 17:42:57 +0100
Subject: [Freeswitch-users] mod_skypiax (skype endpoint) new timing
In-Reply-To: <7b197bef1002140840i6dfbdaecu26c1ae7e1f1a0633@mail.gmail.com>
References: <7b197bef1002131935j3fdf80c4w4050f10bea105d49@mail.gmail.com> 
	<201002141552.31159.errotan@gmail.com>
	<201002141609.18711.errotan@gmail.com> 
	<7b197bef1002140840i6dfbdaecu26c1ae7e1f1a0633@mail.gmail.com>
Message-ID: <7b197bef1002140842i3be561a6sb0c7f1b3adb3ca0b@mail.gmail.com>

I mean, snd-dummy still working very well on kernels at 100HZ, but the
new one is supposed to work with 1000HZ kernels too.

But the most changes are in mod_skypiax own timing, that are supposed
to sole the various delay problems that now and then have surfaced.

Let me know, guys 'n gals.

-gm

On Sun, Feb 14, 2010 at 5:40 PM, Giovanni Maruzzelli
<gmaruzz at celliax.org> wrote:
> Thanks Pusk?s,
>
> any problem of growing delay in sip->fs->skype or skype->fs->sip, or
> conferences, or whatever?
>
> Also, snd-dummy is now more compatible with kernels at 1000HZ (eg:
> centOS), although I believe Lenny is 100HZ.
>
> -giovanni
>
> On Sun, Feb 14, 2010 at 4:09 PM, Pusk?s Zsolt <errotan at gmail.com> wrote:
>> 2010. febru?r 14. 15.52.31 Pusk?s Zsolt d?tummal ezt ?rta:
>>> 2010. febru?r 14. 04.35.01 Giovanni Maruzzelli d?tummal ezt ?rta:
>>> > Hello FreeSWITCHers,
>>> >
>>> > I've just committed on svn16640 new timing for mod_skypiax, and I
>>> > would like if you guys give it a test in the various use cases.
>>> >
>>> > ciao for now,
>>> >
>>> > -giovanni
>>>
>>> It is not really working:
>>>
>>> 2010-02-14 15:37:39.182686 [ERR] skypiax_protocol.c:196 rev 16640M[(nil)|37
>>> ][ERRORA ?196 ?][gun_at_koli][-1, 0, 0] Skype got ERROR: |||ERROR 589 ALTER
>>> CALL: unable to alter input/output|||
>>> 2010-02-14 15:37:39.182686 [ERR] skypiax_protocol.c:198 rev 16640M[(nil)|37
>>> ][ERRORA ?198 ?][gun_at_koli][-1, 0,16] skype_call now is DOWN
>>> 2010-02-14 15:37:39.182686 [DEBUG] mod_skypiax.c:1068 rev 16640M[(nil)|37
>>> ][DEBUG_SKYPE ?1068 ][gun_at_koli][-1, 1,16] skype call ended
>>> 2010-02-14 15:37:39.182686 [DEBUG] mod_skypiax.c:1085 rev 16640M[(nil)|37
>>> ][DEBUG_SKYPE ?1085 ][gun_at_koli][-1, 1,16] no session
>>>
>>>
>>> Also can't unload mod_skypiax because the channels are up:
>>>
>>> freeswitch at internal> reload mod_skypiax
>>> -ERR unloading module [Module in use.]
>>>
>>> 2010-02-14 15:40:13.570678 [WARNING] switch_loadable_module.c:1268 Module
>>> mod_skypiax is in use, cannot unload.
>>> freeswitch at internal> show channels
>>> uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,de
>>> st,application,application_data,dialplan,context,read_codec,read_rate,write
>>> _codec,write_rate,secure,hostname,presence_id,presence_data
>>> ?4f302086-1976-11df-a1d8-47a9ffb132d9,inbound,2010-02-14
>>> 15:36:16,1266158176,skypiax/gun_at_koli,CS_EXECUTE,Pusk?s
>>> Zsolt,errotan,,4219,echo,,XML,default,L16,16000,L16,16000,,pc,,
>>> 80007580-1976-11df-a1d9-47a9ffb132d9,inbound,2010-02-14
>>> 15:37:38,1266158258,skypiax/gun_at_koli,CS_EXECUTE,Pusk?s
>>> Zsolt,errotan,,4219,echo,,XML,default,L16,16000,L16,16000,,pc,,
>>>
>>> 2 total.
>>>
>>> Have to do a fsctl shutdown to kill those channels.
>>>
>>> My home skype (errotan) client displays: Remote sound problem. when I try
>>> ?to call the other (gun_at_koli).
>>>
>>
>> Ooops sorry the user running the skype client were not in the audio group
>> therefore don't have access to snd_dummy.
>>
>> Works now! Tested on Debian "Lenny" x86
>>
>> :)
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From paul at apcl.us  Sun Feb 14 09:14:50 2010
From: paul at apcl.us (Paul Levin)
Date: Sun, 14 Feb 2010 12:14:50 -0500
Subject: [Freeswitch-users] can vm settings go in
 conf\directory\default\default.xml?
Message-ID: <4B782F8A.6010402@apcl.us>

An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100214/9eff7240/attachment.html 

From vmaruani at interwise.com  Sun Feb 14 09:34:06 2010
From: vmaruani at interwise.com (Victor Maruani)
Date: Sun, 14 Feb 2010 19:34:06 +0200
Subject: [Freeswitch-users] Bypass-media and REFER method
In-Reply-To: <191c3a031002140654x37b65f1alc45907b545249cb4@mail.gmail.com>
References: <B816EF85C732094BA2016B2112F3DC2B0402C777@isr-brass.interwise.com><B816EF85C732094BA2016B2112F3DC2B040C5CA6@isr-brass.interwise.com><191c3a031002100645n5d827949xdefd24143f10759@mail.gmail.com><B816EF85C732094BA2016B2112F3DC2B040C6061@isr-brass.interwise.com><191c3a031002111501y7e5917b8g4faae4b8752e6fc2@mail.gmail.com><B816EF85C732094BA2016B2112F3DC2B040C693B@isr-brass.interwise.com><191c3a031002140645s358688c8g68280267d5d650d3@mail.gmail.com>
	<191c3a031002140654x37b65f1alc45907b545249cb4@mail.gmail.com>
Message-ID: <B816EF85C732094BA2016B2112F3DC2B04130D8C@isr-brass.interwise.com>

Hi, 

 

I understand your point and the reason I want to bypass media has a lot
to do with load/call volume.

I don't mind the notify series being implemented or not, the notify OK
is enough if it succeeded indeed.

I think the issue here is the order of action. 

 

Instead of making a long speech and lose my point allow me to put it
like this:

What I see today is:

a-      REFER is received

b-      FS immediately sends 202 accepted ----- ok

c-       FS sends NOTIFY OK to B ---- wrong (not true yet)

d-      FS sends reINVITE to A  ------ wrong (why not after FS gets C's
sdp)

e-      B is disconnected ------- wrong (same as above , and if C is
down I lose the caller)

f-       FS INVITE C  ------ ok 

 

Without getting in sip madness, I would change the order to a,b, F ,
c,d,e 

This way only do I have the sofia leg talking to the B-leg independently
during the REFER in my view.

What do you think?

 

In the meantime I'll try playing with the config but so far I don't have
it work.

Thanks,

 

Regards,

Victor.

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Anthony Minessale
Sent: Sunday, February 14, 2010 4:54 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Bypass-media and REFER method

 

We don't support the series of 100,180,200 in the notifies that is
typically a pure sip pbx feature.
We are a b2b and protocol agnostic softswitch.
When you do refer to a sofia leg you are talking to that leg
independantly.
We do not track the progress of what the channel does once you transfer
it, instead we send the channel back to the dialplan and accept the
refer.  This operation if sucessful will generate 202.
The resume media on hold and bypass after att xfer sofia options or
bypass_media_after_bridge var may be what you need if you want no media
on fs.
We have to make some sacrifices on sip madness to gain some general
flexibility and call volume. =/

	On Feb 14, 2010 3:39 AM, "Victor Maruani"
<vmaruani at interwise.com> wrote:

	Hi,

	 

	I would say it fails in 2 points:

	First in the fact that a "NOTIFY 200 OK" (line 1070) is sent
right after FS gets the REFER.

	 

	Then in the REINVITE (line 1094) sent to A (10.10.5.19)  just
after this NOTIFY,

	This REINVITE contains the SDP of the FS (10.10.5.92) causing
the A side to send media to FS. There will be no REINVITE with SDP of C
(10.10.5.48)

	 

	But as you say, just afterwards, the REFER action is actually
done and C is invited by FS with the SDP of A.

	 

	Conclusion :

	1)

	B is notified of success just after it sent the REFER and is
disconnected.

	B may be notified of every step of the connection to C (100
trying... 200 OK) when these actually happen.

	What if C is down? Can't FS notify a failure? (didn't test
that.)

	 

	2)

	'A' gets to send media to FS Because of a REINVITE which
disconnect him from B (we are in bypass media mode) . 

	during the process of REFER, A should be still connected to B
from a media perspective.

	The REINVITE is not done at the right time with the right
params.

	Here, a pseudo bridge (on way voice)  is established when C gets
the INVITE and is sending media to A.

	 

	A can hear C but C can't hear A after the REFER.

	If C was down, A would be "lost" in FS...

	 

	I believe the correct behavior would be:

	B sends REFER.

	FS INVITE C 

	C replies 100, 180... 200 and FS notifies B in accordance.

	Once C has sent 200 OK with its SDP. B is disconnected and A is
updated (REINVITE) with C's SDP.

	 

	Please share your thoughts, I still don't know if it's a bug or
if I configured something wrong although I don't think so.

	Hasn't anyone done that before?

	 

	Thank you.

	 

	 

	 

	From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Anthony Minessale
	Sent: Friday, February 12, 2010 1:02 AM

	
	To: freeswitch-users at lists.freeswitch.org
	Subject: Re: [Freeswitch-users] Bypass-media and REFER me...

	
	
	 
	
	Where do you think it's failing?
	that log shows it get refer, go back to dp, invite to 3341
and...

	
	_______________________________________________
	FreeSWITCH-users mailing list
	FreeSWITCH-users at lists.freeswitch.org
	http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
	
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
	http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100214/1ab52697/attachment-0001.html 

From anthony.minessale at gmail.com  Sun Feb 14 10:37:53 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sun, 14 Feb 2010 12:37:53 -0600
Subject: [Freeswitch-users] Bypass-media and REFER method
In-Reply-To: <191c3a031002141037n2e8823ebic0e283aafd0dc002@mail.gmail.com>
References: <B816EF85C732094BA2016B2112F3DC2B0402C777@isr-brass.interwise.com>
	<B816EF85C732094BA2016B2112F3DC2B040C5CA6@isr-brass.interwise.com>
	<191c3a031002100645n5d827949xdefd24143f10759@mail.gmail.com>
	<B816EF85C732094BA2016B2112F3DC2B040C6061@isr-brass.interwise.com>
	<191c3a031002111501y7e5917b8g4faae4b8752e6fc2@mail.gmail.com>
	<B816EF85C732094BA2016B2112F3DC2B040C693B@isr-brass.interwise.com>
	<191c3a031002140645s358688c8g68280267d5d650d3@mail.gmail.com>
	<191c3a031002140654x37b65f1alc45907b545249cb4@mail.gmail.com>
	<B816EF85C732094BA2016B2112F3DC2B04130D8C@isr-brass.interwise.com>
	<191c3a031002141037n2e8823ebic0e283aafd0dc002@mail.gmail.com>
Message-ID: <191c3a031002141037h715fd09bsdb4fb085bfcc3d09@mail.gmail.com>

Right we can't do it
Its difficult to add

On Feb 14, 2010 11:39 AM, "Victor Maruani" <vmaruani at interwise.com> wrote:

 Hi,



I understand your point and the reason I want to bypass media has a lot to
do with load/call volume.

I don't mind the notify series being implemented or not, the notify OK is
enough if it succeeded indeed.

I think the issue here is the order of action.



Instead of making a long speech and lose my point allow me to put it like
this:

What I see today is:

a-      REFER is received

b-      FS immediately sends 202 accepted ----- ok

c-       FS sends NOTIFY OK to B ---- wrong (not true yet)

d-      FS sends reINVITE to A  ------ wrong (why not after FS gets C's sdp)

e-      B is disconnected ------- wrong (same as above , and if C is down I
lose the caller)

f-       FS INVITE C  ------ ok



Without getting in sip madness, I would change the order to a,b, F , c,d,e

This way only do I have the sofia leg talking to the B-leg independently
during the REFER in my view.

What do you think?



In the meantime I'll try playing with the config but so far I don't have it
work.

Thanks,



Regards,

Victor.





*From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony
Minessale
*Sent:* Sunday, February 14, 2010 4:54 PM


To: freeswitch-users at lists.freeswitch.org
*Subject:* Re: [Freeswitch-users] Bypass-media and REFER method





We don't support the series of 100,180,200 in the notifies that is typically
a pure sip pbx fea...

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100214/ea7fe0ff/attachment.html 

From gkuri at ieee.org  Sun Feb 14 11:03:35 2010
From: gkuri at ieee.org (Gabriel Kuri)
Date: Sun, 14 Feb 2010 11:03:35 -0800
Subject: [Freeswitch-users] Broadsoft SCA w/ Cisco SPA-5xx series
Message-ID: <8b1c9cda1002141103te523643ob16db25e3bfa877f@mail.gmail.com>

I followed Brian's directions from one of the previous threads on
configuring the SPA-5xx series phones for Broadsoft SCA and set
manage-shared-appearance=true in the internal profile. SCA appears to
be working on outgoing calls between two phones, the line key starts
flashing red on the second phone when the first phone picks up the
receiver to make a call. However on incoming calls, both phones ring
(same extension), however when one of the phones picks up the line,
the second phone's line key doesn't flash red or show the first phone
on that incoming call. Any ideas? Does shared appearance only work on
outgoing phone calls?

Thanks,
Gabe


From mbsip at gazeta.pl  Sun Feb 14 12:52:54 2010
From: mbsip at gazeta.pl (mbsip)
Date: Sun, 14 Feb 2010 21:52:54 +0100
Subject: [Freeswitch-users] can vm settings go in
	conf\directory\default\default.xml?
In-Reply-To: <4B782F8A.6010402@apcl.us>
References: <4B782F8A.6010402@apcl.us>
Message-ID: <28f27f5d1002141252x2088d193k5e57fa0006957df7@mail.gmail.com>

Paul,

VM is using sqlite as a default configuration. You can use Mysql for
instance, but first you need to play around with ODBC -->
http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core

mod_lua, mod_python or mod_xml_curl can help you to use the same Mysql db to
provide dynamic directory/dialplan information.
For more information run though mentioned modules and take the one you are
good at.

Thx,
Maciej.


2010/2/14 Paul Levin <paul at apcl.us>

>  Maciej,
>     Interesting approach Maciej.  I'm really just a beginner user of FS.
> Can you point me towards documentation that specifically describes the DB
> and how to access it?  If it matters, I'm running this on Windows.
>     Thanks,
>     Paul
>
>
>    Subject:
> Re: [Freeswitch-users] can vm settings go in
> conf\directory\default\default.xml?
> From:
> mbsip <mbsip at gazeta.pl> <mbsip at gazeta.pl>
> Date:
> Sun, 14 Feb 2010 17:01:47 +0100
> To:
> freeswitch-users at lists.freeswitch.org
>
> Paul,
>
> Maybe you should think about providing dynamic directory information
> by using DB.
> >From my point of view, Its much easier to manage DB than .xml files.
>
> Thx,
> Maciej.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100214/88603c99/attachment.html 

From errotan at gmail.com  Sun Feb 14 14:12:47 2010
From: errotan at gmail.com (=?iso-8859-1?q?Pusk=E1s_Zsolt?=)
Date: Sun, 14 Feb 2010 23:12:47 +0100
Subject: [Freeswitch-users] mod_skypiax (skype endpoint) new timing
In-Reply-To: <7b197bef1002140842i3be561a6sb0c7f1b3adb3ca0b@mail.gmail.com>
References: <7b197bef1002131935j3fdf80c4w4050f10bea105d49@mail.gmail.com>
	<7b197bef1002140840i6dfbdaecu26c1ae7e1f1a0633@mail.gmail.com>
	<7b197bef1002140842i3be561a6sb0c7f1b3adb3ca0b@mail.gmail.com>
Message-ID: <201002142312.47548.errotan@gmail.com>

2010. febru?r 14. 17.42.57 Giovanni Maruzzelli d?tummal ezt ?rta:
> I mean, snd-dummy still working very well on kernels at 100HZ, but the
> new one is supposed to work with 1000HZ kernels too.
> 
> But the most changes are in mod_skypiax own timing, that are supposed
> to sole the various delay problems that now and then have surfaced.
> 
> Let me know, guys 'n gals.
> 
> -gm
> 
> On Sun, Feb 14, 2010 at 5:40 PM, Giovanni Maruzzelli
> 
> <gmaruzz at celliax.org> wrote:
> > Thanks Pusk?s,
> >
> > any problem of growing delay in sip->fs->skype or skype->fs->sip, or
> > conferences, or whatever?
> >
> > Also, snd-dummy is now more compatible with kernels at 1000HZ (eg:
> > centOS), although I believe Lenny is 100HZ.
> >
> > -giovanni
> >
> > On Sun, Feb 14, 2010 at 4:09 PM, Pusk?s Zsolt <errotan at gmail.com> wrote:
> >> 2010. febru?r 14. 15.52.31 Pusk?s Zsolt d?tummal ezt ?rta:
> >>> 2010. febru?r 14. 04.35.01 Giovanni Maruzzelli d?tummal ezt ?rta:
> >>> > Hello FreeSWITCHers,
> >>> >
> >>> > I've just committed on svn16640 new timing for mod_skypiax, and I
> >>> > would like if you guys give it a test in the various use cases.
> >>> >
> >>> > ciao for now,
> >>> >
> >>> > -giovanni
> >>>
> >>> It is not really working:
> >>>
> >>> 2010-02-14 15:37:39.182686 [ERR] skypiax_protocol.c:196 rev
> >>> 16640M[(nil)|37 ][ERRORA  196  ][gun_at_koli][-1, 0, 0] Skype got
> >>> ERROR: |||ERROR 589 ALTER CALL: unable to alter input/output|||
> >>> 2010-02-14 15:37:39.182686 [ERR] skypiax_protocol.c:198 rev
> >>> 16640M[(nil)|37 ][ERRORA  198  ][gun_at_koli][-1, 0,16] skype_call now
> >>> is DOWN 2010-02-14 15:37:39.182686 [DEBUG] mod_skypiax.c:1068 rev
> >>> 16640M[(nil)|37 ][DEBUG_SKYPE  1068 ][gun_at_koli][-1, 1,16] skype call
> >>> ended
> >>> 2010-02-14 15:37:39.182686 [DEBUG] mod_skypiax.c:1085 rev
> >>> 16640M[(nil)|37 ][DEBUG_SKYPE  1085 ][gun_at_koli][-1, 1,16] no session
> >>>
> >>>
> >>> Also can't unload mod_skypiax because the channels are up:
> >>>
> >>> freeswitch at internal> reload mod_skypiax
> >>> -ERR unloading module [Module in use.]
> >>>
> >>> 2010-02-14 15:40:13.570678 [WARNING] switch_loadable_module.c:1268
> >>> Module mod_skypiax is in use, cannot unload.
> >>> freeswitch at internal> show channels
> >>> uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_add
> >>>r,de
> >>> st,application,application_data,dialplan,context,read_codec,read_rate,w
> >>>rite _codec,write_rate,secure,hostname,presence_id,presence_data
> >>>  4f302086-1976-11df-a1d8-47a9ffb132d9,inbound,2010-02-14
> >>> 15:36:16,1266158176,skypiax/gun_at_koli,CS_EXECUTE,Pusk?s
> >>> Zsolt,errotan,,4219,echo,,XML,default,L16,16000,L16,16000,,pc,,
> >>> 80007580-1976-11df-a1d9-47a9ffb132d9,inbound,2010-02-14
> >>> 15:37:38,1266158258,skypiax/gun_at_koli,CS_EXECUTE,Pusk?s
> >>> Zsolt,errotan,,4219,echo,,XML,default,L16,16000,L16,16000,,pc,,
> >>>
> >>> 2 total.
> >>>
> >>> Have to do a fsctl shutdown to kill those channels.
> >>>
> >>> My home skype (errotan) client displays: Remote sound problem. when I
> >>> try to call the other (gun_at_koli).
> >>
> >> Ooops sorry the user running the skype client were not in the audio
> >> group therefore don't have access to snd_dummy.
> >>
> >> Works now! Tested on Debian "Lenny" x86
> >>
> >> :)
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> > --
> > Sincerely,
> >
> > Giovanni Maruzzelli
> > Cell : +39-347-2665618
> 

My test was skype -> fs (echo && delay_echo) .
The echo call was 10 min, the delayed echo was 1 hour long.

Now i tried with skype -> fs -> sip.
I done a 15 minute call no delay issues.


From anthony.minessale at gmail.com  Sun Feb 14 14:48:50 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sun, 14 Feb 2010 16:48:50 -0600
Subject: [Freeswitch-users] Bypass-media and REFER method
In-Reply-To: <191c3a031002141037h715fd09bsdb4fb085bfcc3d09@mail.gmail.com>
References: <B816EF85C732094BA2016B2112F3DC2B0402C777@isr-brass.interwise.com>
	<191c3a031002100645n5d827949xdefd24143f10759@mail.gmail.com>
	<B816EF85C732094BA2016B2112F3DC2B040C6061@isr-brass.interwise.com>
	<191c3a031002111501y7e5917b8g4faae4b8752e6fc2@mail.gmail.com>
	<B816EF85C732094BA2016B2112F3DC2B040C693B@isr-brass.interwise.com>
	<191c3a031002140645s358688c8g68280267d5d650d3@mail.gmail.com>
	<191c3a031002140654x37b65f1alc45907b545249cb4@mail.gmail.com>
	<B816EF85C732094BA2016B2112F3DC2B04130D8C@isr-brass.interwise.com>
	<191c3a031002141037n2e8823ebic0e283aafd0dc002@mail.gmail.com>
	<191c3a031002141037h715fd09bsdb4fb085bfcc3d09@mail.gmail.com>
Message-ID: <191c3a031002141448w2749ab61r94b40ab37e83103e@mail.gmail.com>

Let me try to explain better now that I am not on my cell.

My original explanation already stated that the receipt of the refer on a
blind xfer is always answered with a 202 because the call will transfer to
the desired extension with success.  That does not mean that the subsequent
call that is going to take place when that leg hits the dialplan and tries
to make another outbound call will be successful as well.  There is no way
to predict that.  It could easily be a conference or moh or some other 1
legged call.  The 2nd leg in your description cannot exist unless the leg
you are transferring goes back to the dialplan.

The call you sent the REFER to already has moved on in the FS state machine
and there would be no way to go back.

This is what I was trying to explain when I said we have to sacrifice some
seemingly easy features from one perspective to gain all the other things we
can do in FreeSWITCH.

So its not 100% impossible to code but it currently does not exist and I
would be concerned trying to do it would blur the abstraction lines in the
code.  So the short answer is, no, we don't support what you are asking
about.




On Sun, Feb 14, 2010 at 12:37 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> Right we can't do it
> Its difficult to add
>
> On Feb 14, 2010 11:39 AM, "Victor Maruani" <vmaruani at interwise.com> wrote:
>
>  Hi,
>
>
>
> I understand your point and the reason I want to bypass media has a lot to
> do with load/call volume.
>
> I don't mind the notify series being implemented or not, the notify OK is
> enough if it succeeded indeed.
>
> I think the issue here is the order of action.
>
>
>
> Instead of making a long speech and lose my point allow me to put it like
> this:
>
> What I see today is:
>
> a-      REFER is received
>
> b-      FS immediately sends 202 accepted ----- ok
>
> c-       FS sends NOTIFY OK to B ---- wrong (not true yet)
>
> d-      FS sends reINVITE to A  ------ wrong (why not after FS gets C's
> sdp)
>
> e-      B is disconnected ------- wrong (same as above , and if C is down
> I lose the caller)
>
> f-       FS INVITE C  ------ ok
>
>
>
> Without getting in sip madness, I would change the order to a,b, F , c,d,e
>
> This way only do I have the sofia leg talking to the B-leg independently
> during the REFER in my view.
>
> What do you think?
>
>
>
> In the meantime I'll try playing with the config but so far I don't have it
> work.
>
> Thanks,
>
>
>
> Regards,
>
> Victor.
>
>
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony
> Minessale
> *Sent:* Sunday, February 14, 2010 4:54 PM
>
>
> To: freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Bypass-media and REFER method
>
>
>
>
>
> We don't support the series of 100,180,200 in the notifies that is
> typically a pure sip pbx fea...
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100214/c35b9aa9/attachment-0001.html 

From anthony.minessale at gmail.com  Sun Feb 14 15:42:14 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sun, 14 Feb 2010 17:42:14 -0600
Subject: [Freeswitch-users] can vm settings go in
	conf\directory\default\default.xml?
In-Reply-To: <191c3a031002141541n2358d92kf80e35134e1a43f9@mail.gmail.com>
References: <4B782F8A.6010402@apcl.us>
	<28f27f5d1002141252x2088d193k5e57fa0006957df7@mail.gmail.com>
	<191c3a031002141541n2358d92kf80e35134e1a43f9@mail.gmail.com>
Message-ID: <191c3a031002141542w50e92c9k4fc321f74b68f0a9@mail.gmail.com>

Vm users do not inherit params from the domain there is a patch coming to
fix this soon.

On Feb 14, 2010 2:59 PM, "mbsip" <mbsip at gazeta.pl> wrote:

Paul,

VM is using sqlite as a default configuration. You can use Mysql for
instance, but first you need to play around with ODBC -->
http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core

mod_lua, mod_python or mod_xml_curl can help you to use the same Mysql db to
provide dynamic directory/dialplan information.
For more information run though mentioned modules and take the one you are
good at.

Thx,
Maciej.


2010/2/14 Paul Levin <paul at apcl.us>

> >
> > Maciej,
> >     Interesting approach Maciej.  I'm really just a beginner user of FS.
> Can you poin...
>
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users...
>


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100214/d10afec7/attachment.html 

From mcampbellsmith at gmail.com  Sun Feb 14 17:08:49 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Mon, 15 Feb 2010 12:08:49 +1100
Subject: [Freeswitch-users] Forbidden using UDP, works with TCP/TLS
In-Reply-To: <9D702362-BB5F-4C0F-9EE7-DABA3B926C36@freeswitch.org>
References: <33c87fa31002101824h21e3bf7lb13b68e8d4fd36a2@mail.gmail.com>
	<33c87fa31002110131w3cf6ad00n802669856085a879@mail.gmail.com>
	<CA21231C-108B-4445-A3B9-BF8D94FFE898@jerris.com>
	<33c87fa31002110213q1de0eebfsb950d8725ea9aba@mail.gmail.com>
	<0A905834-850F-4D1F-AEC5-D3A87CE4141B@jerris.com>
	<33c87fa31002121629k74f26807i131841aa4252461b@mail.gmail.com>
	<33c87fa31002130343k7a01e9abt9c33360399f246f8@mail.gmail.com>
	<C37AB6F3-F712-41B6-9D6D-7A8AE3C6CABB@freeswitch.org>
	<33c87fa31002131336j7b05db7aof6ba6e3439aebc41@mail.gmail.com>
	<9D702362-BB5F-4C0F-9EE7-DABA3B926C36@freeswitch.org>
Message-ID: <33c87fa31002141708s72cb7246h5372ecea40be615d@mail.gmail.com>

Hi,

The sip trace provided only contains 4 SIP messages.  Do you need the
IP's to decode the messages?

from udp/[121.xxx.xxx.xxx] is the SPA3102
from (udp/192.168.1.120:5060) is FS server

I can register the device to all my voip providers successfully using
UDP but it will not register to FS using UDP.
It can register using TLS and TCP.

Very confusing as to why that would be.


On Sun, Feb 14, 2010 at 10:11 AM, Brian West <brian at freeswitch.org> wrote:
> You know you could have obscured the first part of the IP and not the LAST... kinda removes the ability to tell WHO sent what.
>
> >From that log I guess your password is wrong.
>
> /b
>
> On Feb 13, 2010, at 3:36 PM, Mark Campbell-Smith wrote:
>
>> Thanks Brian.
>>
>> The full log is pasted here http://pastebin.freeswitch.org/12133
>>
>>
>>
>> On Sun, Feb 14, 2010 at 2:08 AM, Brian West <brian at freeswitch.org> wrote:
>>> Can't tell since you keep cutting the lines required to figure this out... FULL console log with FULL sip trace. ?You always cut out what I wanna see. ?Just get a pcap.
>>>
>>> /b
>>>
>>> On Feb 13, 2010, at 5:43 AM, Mark Campbell-Smith wrote:
>>>
>>>> More testing. The device registers successfully to my SIP provider
>>>> directly using UDP - why would FS be rejecting the registration
>>>> request?
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From jmesquita at freeswitch.org  Sun Feb 14 17:28:24 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Sun, 14 Feb 2010 22:28:24 -0300
Subject: [Freeswitch-users] play_and_get_digits + OutboundESL
Message-ID: <ebbc2d551002141728xa335781qecaa284d32544ae0@mail.gmail.com>

Gentleman,

I am trying to use TTS with play_and_get_digits with Outbound ESL with
little luck.

There seems to be some kind of parsing problem that I don't really know how
to solve.

Here is the command (my python script) that I am issuing:

http://pastebin.freeswitch.org/12147

Here is the FreeSWITCH log output:

http://pastebin.freeswitch.org/12146

And finally here is the ESL log:

http://pastebin.freeswitch.org/12145

It needs to be stated that the tts_engine and tts_voice channel vars have
been set before on the script.

If I use these commands the exact same way on the dialplan, it works. Is
this a bug or am I overlooking something really obvious?

Regards,
JM
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100214/758c3b36/attachment.html 

From infos at madovsky.org  Sun Feb 14 09:58:43 2010
From: infos at madovsky.org (Madovsky)
Date: Sun, 14 Feb 2010 12:58:43 -0500
Subject: [Freeswitch-users] inbound outbound transparent proxy question
Message-ID: <059C32DA5C3647908A1499C8027464B4@MOBILEE1705>

Hi,

I'm new to freeswitch world and apologize if my question is not relevant.
I need to install freeswitch as inbound outbound transparent proxy (media only if possible, but not sure it's the right terms to use)
example :


Caller A (he can be anonymous with his own provider) connect with a softphone we provide in my network (nat and public)
so he sets the softphone as user callerA, domain iptel.org (which is not my domain) proxy 192.168.0.1 (which is my FS proxy).
So he calls an external user (who is unknown) like callerB at bobo.dot. So callerB receive the invite (until now he can see the caller user name but
with proxy ip subsituted from the domain (but I want the original caller domain kept intact) and the call is established.
But, for the contrary (callerB wants to call callerA), it doesn't work.

Is anyone have idea or example of this kind of configuration ?

Best Regards

Franck Chionna
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100214/8f0df933/attachment.html 

From infos at madovsky.org  Sun Feb 14 11:33:19 2010
From: infos at madovsky.org (Madovsky)
Date: Sun, 14 Feb 2010 14:33:19 -0500
Subject: [Freeswitch-users] Fw: inbound outbound transparent proxy question
Message-ID: <650FD98CB36544129A1C325A0DBD95AB@MOBILEE1705>

Hi,

I'm new to freeswitch world and apologize if my question is not relevant.
I need to install freeswitch as inbound outbound transparent proxy (media only if possible, but not sure it's the right terms to use)
example :


Caller A (he can be anonymous with his own provider) connect with a softphone we provide in my network (nat and public)
so he sets the softphone as user callerA, domain iptel.org (which is not my domain) proxy 192.168.0.1 (which is my FS proxy).
So he calls an external user (who is unknown) like callerB at bobo.dot. So callerB receive the invite (until now he can see the caller user name but
with proxy ip subsituted from the domain (but I want the original caller domain kept intact) and the call is established.
But, for the contrary (callerB wants to call callerA), it doesn't work.

Is anyone have idea or example of this kind of configuration ?

Best Regards

Franck Chionna
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100214/c4c213a6/attachment.html 

From brian at freeswitch.org  Sun Feb 14 19:58:15 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 14 Feb 2010 21:58:15 -0600
Subject: [Freeswitch-users] Broadsoft SCA w/ Cisco SPA-5xx series
In-Reply-To: <8b1c9cda1002141103te523643ob16db25e3bfa877f@mail.gmail.com>
References: <8b1c9cda1002141103te523643ob16db25e3bfa877f@mail.gmail.com>
Message-ID: <0D67132C-69D8-43AD-B7E6-C7545B544322@freeswitch.org>

Works fine here... is your box slow or something?

/b

On Feb 14, 2010, at 1:03 PM, Gabriel Kuri wrote:

> I followed Brian's directions from one of the previous threads on
> configuring the SPA-5xx series phones for Broadsoft SCA and set
> manage-shared-appearance=true in the internal profile. SCA appears to
> be working on outgoing calls between two phones, the line key starts
> flashing red on the second phone when the first phone picks up the
> receiver to make a call. However on incoming calls, both phones ring
> (same extension), however when one of the phones picks up the line,
> the second phone's line key doesn't flash red or show the first phone
> on that incoming call. Any ideas? Does shared appearance only work on
> outgoing phone calls?
> 
> Thanks,
> Gabe



From brian at freeswitch.org  Sun Feb 14 20:07:31 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 14 Feb 2010 22:07:31 -0600
Subject: [Freeswitch-users] Forbidden using UDP, works with TCP/TLS
In-Reply-To: <33c87fa31002141708s72cb7246h5372ecea40be615d@mail.gmail.com>
References: <33c87fa31002101824h21e3bf7lb13b68e8d4fd36a2@mail.gmail.com>
	<33c87fa31002110131w3cf6ad00n802669856085a879@mail.gmail.com>
	<CA21231C-108B-4445-A3B9-BF8D94FFE898@jerris.com>
	<33c87fa31002110213q1de0eebfsb950d8725ea9aba@mail.gmail.com>
	<0A905834-850F-4D1F-AEC5-D3A87CE4141B@jerris.com>
	<33c87fa31002121629k74f26807i131841aa4252461b@mail.gmail.com>
	<33c87fa31002130343k7a01e9abt9c33360399f246f8@mail.gmail.com>
	<C37AB6F3-F712-41B6-9D6D-7A8AE3C6CABB@freeswitch.org>
	<33c87fa31002131336j7b05db7aof6ba6e3439aebc41@mail.gmail.com>
	<9D702362-BB5F-4C0F-9EE7-DABA3B926C36@freeswitch.org>
	<33c87fa31002141708s72cb7246h5372ecea40be615d@mail.gmail.com>
Message-ID: <1888B1CA-D0A8-407C-9C44-B86E0F1925D6@freeswitch.org>

Is the device behind nat with your FreeSWITCH?  If so disable stun on the device.  If FS is 192.168.1.120 and your device is 121.x.x.x something then I suspect its doing a hair pin thru your router.  Your network is busted which is my final answer.  

/b

On Feb 14, 2010, at 7:08 PM, Mark Campbell-Smith wrote:

> Hi,
> 
> The sip trace provided only contains 4 SIP messages.  Do you need the
> IP's to decode the messages?
> 
> from udp/[121.xxx.xxx.xxx] is the SPA3102
> from (udp/192.168.1.120:5060) is FS server
> 
> I can register the device to all my voip providers successfully using
> UDP but it will not register to FS using UDP.
> It can register using TLS and TCP.
> 
> Very confusing as to why that would be.



From mcampbellsmith at gmail.com  Sun Feb 14 20:17:15 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Mon, 15 Feb 2010 15:17:15 +1100
Subject: [Freeswitch-users] Forbidden using UDP, works with TCP/TLS
In-Reply-To: <1888B1CA-D0A8-407C-9C44-B86E0F1925D6@freeswitch.org>
References: <33c87fa31002101824h21e3bf7lb13b68e8d4fd36a2@mail.gmail.com>
	<33c87fa31002110213q1de0eebfsb950d8725ea9aba@mail.gmail.com>
	<0A905834-850F-4D1F-AEC5-D3A87CE4141B@jerris.com>
	<33c87fa31002121629k74f26807i131841aa4252461b@mail.gmail.com>
	<33c87fa31002130343k7a01e9abt9c33360399f246f8@mail.gmail.com>
	<C37AB6F3-F712-41B6-9D6D-7A8AE3C6CABB@freeswitch.org>
	<33c87fa31002131336j7b05db7aof6ba6e3439aebc41@mail.gmail.com>
	<9D702362-BB5F-4C0F-9EE7-DABA3B926C36@freeswitch.org>
	<33c87fa31002141708s72cb7246h5372ecea40be615d@mail.gmail.com>
	<1888B1CA-D0A8-407C-9C44-B86E0F1925D6@freeswitch.org>
Message-ID: <33c87fa31002142017v49d8e7c5nf572d086ec89282b@mail.gmail.com>

FS and the ATA are on different networks.  FS is nat'd (192.168.1.120,
upnp enabled on the router) and the ATA is on the internet at another
location.

Any other ideas?

On Mon, Feb 15, 2010 at 3:07 PM, Brian West <brian at freeswitch.org> wrote:
> Is the device behind nat with your FreeSWITCH? ?If so disable stun on the device. ?If FS is 192.168.1.120 and your device is 121.x.x.x something then I suspect its doing a hair pin thru your router. ?Your network is busted which is my final answer.
>
> /b
>
> On Feb 14, 2010, at 7:08 PM, Mark Campbell-Smith wrote:
>
>> Hi,
>>
>> The sip trace provided only contains 4 SIP messages. ?Do you need the
>> IP's to decode the messages?
>>
>> from udp/[121.xxx.xxx.xxx] is the SPA3102
>> from (udp/192.168.1.120:5060) is FS server
>>
>> I can register the device to all my voip providers successfully using
>> UDP but it will not register to FS using UDP.
>> It can register using TLS and TCP.
>>
>> Very confusing as to why that would be.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From mcampbellsmith at gmail.com  Sun Feb 14 21:09:35 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Mon, 15 Feb 2010 16:09:35 +1100
Subject: [Freeswitch-users] Forbidden using UDP, works with TCP/TLS
In-Reply-To: <33c87fa31002142017v49d8e7c5nf572d086ec89282b@mail.gmail.com>
References: <33c87fa31002101824h21e3bf7lb13b68e8d4fd36a2@mail.gmail.com>
	<0A905834-850F-4D1F-AEC5-D3A87CE4141B@jerris.com>
	<33c87fa31002121629k74f26807i131841aa4252461b@mail.gmail.com>
	<33c87fa31002130343k7a01e9abt9c33360399f246f8@mail.gmail.com>
	<C37AB6F3-F712-41B6-9D6D-7A8AE3C6CABB@freeswitch.org>
	<33c87fa31002131336j7b05db7aof6ba6e3439aebc41@mail.gmail.com>
	<9D702362-BB5F-4C0F-9EE7-DABA3B926C36@freeswitch.org>
	<33c87fa31002141708s72cb7246h5372ecea40be615d@mail.gmail.com>
	<1888B1CA-D0A8-407C-9C44-B86E0F1925D6@freeswitch.org>
	<33c87fa31002142017v49d8e7c5nf572d086ec89282b@mail.gmail.com>
Message-ID: <33c87fa31002142109l5fcabed3ydba21b865ed6cbbb@mail.gmail.com>

A little more testing.    I noticed that the Authorization field
differs when TCP or UDP:

UDP (fails)
Digest username=\"2010\", realm=\"mydns.dyndns.org\",
nonce=\"e5f119c6-19e9-11df-bd09-773b7a755f78\",
uri=\"sip:mydns.dyndns.org:5060\",
response=\"e37be3e49c159d4f98e8bd04b36f2bd7\", algorithm=MD5,
qop=\"1fffcc9f\"

TCP (works)
Digest username=\"2010\",realm=\"mydns.dyndns.org\",nonce=\"5d9e75c2-19ea-11df-bd0b-773b7a755f78\",uri=\"sip:mydns.dyndns.org:5060\",algorithm=MD5,response=\"45ba55d3fbafcbf2bc2aa6418656ecc2\",qop=auth,nc=00000001,cnonce=\"3a650454\"

Is qop = 1fffcc9f valid in SIP?  Does a cnonce need to be included also?

FS sends qop=auth in the Unauthorized response.

Thanks



On Mon, Feb 15, 2010 at 3:17 PM, Mark Campbell-Smith
<mcampbellsmith at gmail.com> wrote:
> FS and the ATA are on different networks. ?FS is nat'd (192.168.1.120,
> upnp enabled on the router) and the ATA is on the internet at another
> location.
>
> Any other ideas?
>
> On Mon, Feb 15, 2010 at 3:07 PM, Brian West <brian at freeswitch.org> wrote:
>> Is the device behind nat with your FreeSWITCH? ?If so disable stun on the device. ?If FS is 192.168.1.120 and your device is 121.x.x.x something then I suspect its doing a hair pin thru your router. ?Your network is busted which is my final answer.
>>
>> /b
>>
>> On Feb 14, 2010, at 7:08 PM, Mark Campbell-Smith wrote:
>>
>>> Hi,
>>>
>>> The sip trace provided only contains 4 SIP messages. ?Do you need the
>>> IP's to decode the messages?
>>>
>>> from udp/[121.xxx.xxx.xxx] is the SPA3102
>>> from (udp/192.168.1.120:5060) is FS server
>>>
>>> I can register the device to all my voip providers successfully using
>>> UDP but it will not register to FS using UDP.
>>> It can register using TLS and TCP.
>>>
>>> Very confusing as to why that would be.
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>


From brian at freeswitch.org  Sun Feb 14 21:18:48 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 14 Feb 2010 23:18:48 -0600
Subject: [Freeswitch-users] Forbidden using UDP, works with TCP/TLS
In-Reply-To: <33c87fa31002142109l5fcabed3ydba21b865ed6cbbb@mail.gmail.com>
References: <33c87fa31002101824h21e3bf7lb13b68e8d4fd36a2@mail.gmail.com>
	<0A905834-850F-4D1F-AEC5-D3A87CE4141B@jerris.com>
	<33c87fa31002121629k74f26807i131841aa4252461b@mail.gmail.com>
	<33c87fa31002130343k7a01e9abt9c33360399f246f8@mail.gmail.com>
	<C37AB6F3-F712-41B6-9D6D-7A8AE3C6CABB@freeswitch.org>
	<33c87fa31002131336j7b05db7aof6ba6e3439aebc41@mail.gmail.com>
	<9D702362-BB5F-4C0F-9EE7-DABA3B926C36@freeswitch.org>
	<33c87fa31002141708s72cb7246h5372ecea40be615d@mail.gmail.com>
	<1888B1CA-D0A8-407C-9C44-B86E0F1925D6@freeswitch.org>
	<33c87fa31002142017v49d8e7c5nf572d086ec89282b@mail.gmail.com>
	<33c87fa31002142109l5fcabed3ydba21b865ed6cbbb@mail.gmail.com>
Message-ID: <308A282B-27A0-497B-B250-ED2EC02D0BD5@freeswitch.org>

Sounds like the device is BUSTED.  :P

/b

On Feb 14, 2010, at 11:09 PM, Mark Campbell-Smith wrote:

> A little more testing.    I noticed that the Authorization field
> differs when TCP or UDP:
> 
> UDP (fails)
> Digest username=\"2010\", realm=\"mydns.dyndns.org\",
> nonce=\"e5f119c6-19e9-11df-bd09-773b7a755f78\",
> uri=\"sip:mydns.dyndns.org:5060\",
> response=\"e37be3e49c159d4f98e8bd04b36f2bd7\", algorithm=MD5,
> qop=\"1fffcc9f\"
> 
> TCP (works)
> Digest username=\"2010\",realm=\"mydns.dyndns.org\",nonce=\"5d9e75c2-19ea-11df-bd0b-773b7a755f78\",uri=\"sip:mydns.dyndns.org:5060\",algorithm=MD5,response=\"45ba55d3fbafcbf2bc2aa6418656ecc2\",qop=auth,nc=00000001,cnonce=\"3a650454\"
> 
> Is qop = 1fffcc9f valid in SIP?  Does a cnonce need to be included also?
> 
> FS sends qop=auth in the Unauthorized response.
> 
> Thanks

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100214/ee387f1a/attachment.html 

From nagalenoj at gmail.com  Sun Feb 14 21:31:54 2010
From: nagalenoj at gmail.com (Nagalenoj H.)
Date: Mon, 15 Feb 2010 11:01:54 +0530
Subject: [Freeswitch-users] Play music to A leg.
In-Reply-To: <191c3a031002120907l28fbdf2dgab5df7dd1b5a2f76@mail.gmail.com>
References: <fe67a3341002090519r2cc6e8edm7702b8d012fc2be3@mail.gmail.com>
	<87f2f3b91002090737l4efa8en29bde9457fef7ea5@mail.gmail.com>
	<fe67a3341002092157x7484f3d2id6678510bbec66cc@mail.gmail.com>
	<87f2f3b91002100702u7581589ua866d2492a008b23@mail.gmail.com>
	<fe67a3341002110211w60390e1cm2d12fadd1e802778@mail.gmail.com>
	<87f2f3b91002110754laf9d01dxa94a36bc1b8c83df@mail.gmail.com>
	<191c3a031002110804m4c18e7e0y317f44272b42c8b4@mail.gmail.com>
	<7d79b3931002112018y2512954cuca83ce21ff3406fe@mail.gmail.com>
	<191c3a031002120907l28fbdf2dgab5df7dd1b5a2f76@mail.gmail.com>
Message-ID: <fe67a3341002142131m28d7bc5ao8028c4ce8706f97c@mail.gmail.com>

Usually it works as follows,
bridge {group_confirm_key=exec,group_confirm_file=perl xx.pl
}user/1000,user/1001,user/1002

But, I want it like,
bridge [group_confirm_key=exec,group_confirm_file=perl
xx.pl]user/1000,[group_confirm_key=exec,group_confirm_file=perl
xx.pl]user/1001,user/1002

So, I don't want the script to be executed if 1002 answers the call.

Also, I need only one to answer the call. When the first person answers the
call, the other extensions have to stop ringing immediately.
But, this works as 'bridges with who completes the script first',

On Fri, Feb 12, 2010 at 10:37 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> the script executes for everyone and gives them a chance to dial multiple
> digits to test for, this is what he asked for, instead of 1 digit dial
> multiple digits.  you set the correct string as a variable on the channel
> and everybody runs the script and whoever dials the right digits wins the
> rest will be hungup on.
>
>
>
> On Thu, Feb 11, 2010 at 10:18 PM, lakshmanan ganapathy <
> lakindia89 at gmail.com> wrote:
>
>> Dear Antony,
>> In bridge if we are making parallel calls, then group_confirm_key in
>> execute mode will execute for all the extensions, and whomsoever finishes
>> the script first, will be bridged.
>>
>> But I think nagalenoj need to execute the script for the extension which
>> answers the call first, not for all the extension.!!!.
>>
>> From nanalenoj's post
>>
>>             " but when I do originate for multiple extensions, I want this
>> script to work based on the answered extension."
>>
>>
>> On Thu, Feb 11, 2010 at 9:34 PM, Anthony Minessale <
>> anthony.minessale at gmail.com> wrote:
>>
>>> group_confirm_key in execute mode can execute a lua script instead that
>>> can read as many digits as you want and parse the results.
>>>
>>>
>>>
>>> On Thu, Feb 11, 2010 at 9:54 AM, Michael Collins <msc at freeswitch.org>wrote:
>>>
>>>> Hehe, this is getting more and more complicated. You may want to
>>>> consider using the event socket and have your call control be done from a
>>>> more 3rd party-ish perspective. If you've got all these different scenarios
>>>> it might be better to let an external script do all the work.
>>>>
>>>> http://wiki.freeswitch.org/wiki/Event_Socket
>>>>
>>>> -MC
>>>>
>>>>
>>>> On Thu, Feb 11, 2010 at 2:11 AM, Nagalenoj H. <nagalenoj at gmail.com>wrote:
>>>>
>>>>> But My scenario is,
>>>>>    After I get the call from X.
>>>>>    I answer the call in some scenarios and won't answer the call. So,
>>>>> this leg can either be answered or unanswered.
>>>>>    I originate a call to another number.
>>>>>    After getting some digits from this originated leg.
>>>>>    I do uuid_bridge of these 2 legs.
>>>>>
>>>>> I want to play some file[ringback] to leg A before bridging to B.
>>>>>
>>>>> On Wed, Feb 10, 2010 at 8:32 PM, Michael Collins <msc at freeswitch.org>wrote:
>>>>>
>>>>>>
>>>>>>
>>>>>> On Tue, Feb 9, 2010 at 9:57 PM, Nagalenoj H. <nagalenoj at gmail.com>wrote:
>>>>>>
>>>>>>> Because, I want to get some digits before bridging the legs. I've
>>>>>>> tried group_confirm_key, but it accepts only one digit, I need multiple
>>>>>>> digits, so I can't use.
>>>>>>> I've also tried group_confirm_file, but when I do originate for
>>>>>>> multiple extensions, I want this script to work based on the answered
>>>>>>> extension.
>>>>>>>
>>>>>>> So, I've originated and processed the events to do my job.
>>>>>>>
>>>>>>> How do I play some music to A leg?
>>>>>>>
>>>>>>> I might be missing something, but couldn't you just park the call ("A
>>>>>> leg") until you connect to the other party ("B leg") and then uuid_bridge at
>>>>>> whatever point you want?
>>>>>> -MC
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> Regards,
>>>>> Nagalenoj H.
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>> iax:guest at conference.freeswitch.org/888
>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> pstn:+19193869900
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Regards,
Nagalenoj H.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100215/aae903bf/attachment-0001.html 

From gorand at donevtechconsulting.com  Sun Feb 14 20:47:14 2010
From: gorand at donevtechconsulting.com (Goran Donev)
Date: Sun, 14 Feb 2010 22:47:14 -0600
Subject: [Freeswitch-users] Version 1.5
In-Reply-To: <mailman.6373.1266207467.2495.freeswitch-users@lists.freeswitch.org>
References: <mailman.6373.1266207467.2495.freeswitch-users@lists.freeswitch.org>
Message-ID: <041101caadf9$f36d77e0$da4867a0$@com>

When is version 1.5 of free switch going to be released. The last update on
the website was on the week of Feb 8th. I still have not seen anything else.


Thanks



From jaybinks at gmail.com  Sun Feb 14 21:42:40 2010
From: jaybinks at gmail.com (jay binks)
Date: Mon, 15 Feb 2010 15:42:40 +1000
Subject: [Freeswitch-users] Version 1.5
In-Reply-To: <041101caadf9$f36d77e0$da4867a0$@com>
References: <mailman.6373.1266207467.2495.freeswitch-users@lists.freeswitch.org>
	<041101caadf9$f36d77e0$da4867a0$@com>
Message-ID: <ec2ce17e1002142142g1d71be24yf55132344b880514@mail.gmail.com>

did you contribute to the dinner ??
maybe thats why it hasnt been released yet ...

Jokes...

J

On Mon, Feb 15, 2010 at 2:47 PM, Goran Donev <gorand at donevtechconsulting.com
> wrote:

> When is version 1.5 of free switch going to be released. The last update on
> the website was on the week of Feb 8th. I still have not seen anything
> else.
>
>
> Thanks
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely

Jay
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100215/c6728b40/attachment.html 

From gkuri at ieee.org  Sun Feb 14 21:50:58 2010
From: gkuri at ieee.org (Gabriel Kuri)
Date: Sun, 14 Feb 2010 21:50:58 -0800
Subject: [Freeswitch-users] Broadsoft SCA w/ Cisco SPA-5xx series
In-Reply-To: <0D67132C-69D8-43AD-B7E6-C7545B544322@freeswitch.org>
References: <8b1c9cda1002141103te523643ob16db25e3bfa877f@mail.gmail.com>
	<0D67132C-69D8-43AD-B7E6-C7545B544322@freeswitch.org>
Message-ID: <8b1c9cda1002142150i48e17045yac9596cb32ee6ee2@mail.gmail.com>

It's an Atom N330. Not sure why it doesn't work on incoming calls, are
there any other settings that need to be set on the phones other than
setting the lines to shared and server type to Broadsoft?

Not sure if this matters, since everything else seems to be working,
but the phones are behind one NAT and FreeSWITCH is behind a totally
different NAT. Everything else seems to be working, I don't have any
one way audio or other funny things going on that would point to NAT,
so I'm not sure NAT is the issue.

What should be sent to the phone, to light up the light after the
other phone is answered, a NOTIFY? I'm seeing a NOTIFY sent to the
other phone with Event: call-info, but the light isn't turning on to
indicate SCA.

Thanks,
Gabe





On Sun, Feb 14, 2010 at 7:58 PM, Brian West <brian at freeswitch.org> wrote:
> Works fine here... is your box slow or something?
>
> /b
>
> On Feb 14, 2010, at 1:03 PM, Gabriel Kuri wrote:
>
>> I followed Brian's directions from one of the previous threads on
>> configuring the SPA-5xx series phones for Broadsoft SCA and set
>> manage-shared-appearance=true in the internal profile. SCA appears to
>> be working on outgoing calls between two phones, the line key starts
>> flashing red on the second phone when the first phone picks up the
>> receiver to make a call. However on incoming calls, both phones ring
>> (same extension), however when one of the phones picks up the line,
>> the second phone's line key doesn't flash red or show the first phone
>> on that incoming call. Any ideas? Does shared appearance only work on
>> outgoing phone calls?
>>
>> Thanks,
>> Gabe
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From gkuri at ieee.org  Sun Feb 14 21:59:35 2010
From: gkuri at ieee.org (Gabriel Kuri)
Date: Sun, 14 Feb 2010 21:59:35 -0800
Subject: [Freeswitch-users] Broadsoft SCA w/ Cisco SPA-5xx series
In-Reply-To: <0D67132C-69D8-43AD-B7E6-C7545B544322@freeswitch.org>
References: <8b1c9cda1002141103te523643ob16db25e3bfa877f@mail.gmail.com>
	<0D67132C-69D8-43AD-B7E6-C7545B544322@freeswitch.org>
Message-ID: <8b1c9cda1002142159h5efcc765q9a41baa1bf694c8c@mail.gmail.com>

BTW, here's a copy of the NOTIFY (event call-info) sent to the other
phone after the first phone is answered, should this have a Call-Info
line with an "appearance-state=seized" to turn on the light on the
other phone?


NOTIFY sip:2551@<removed>:54446 SIP/2.0.
Via: SIP/2.0/UDP <removed>:9430;rport;branch=z9hG4bK71pN2cXgH851K.
Max-Forwards: 70.
From: <sip:2551 at mydomain.com>;tag=XeB6ZrKDevpHp.
To: <sip:2551 at mydomain.com>;tag=c2d34993aac6ea.
Call-ID: 34c34987-8b6fa786@<removed>.
CSeq: 126950830 NOTIFY.
Contact: <sip:2551@<removed>:9430>.
Expires: 3959.
Call-Info: <sip:mydomain.com>;appearance-index=*;appearance-state=idle.
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16633M.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, PRACK, NOTIFY, PUBLISH, SUBSCRIBE.
Supported: 100rel, timer, precondition, path, replaces.
Event: call-info.
Allow-Events: talk, presence, dialog, line-seize, call-info, sla,
include-session-description, presence.winfo, message-summary, refer.
Subscription-State: active;expires=3959.
Content-Length: 0.



On Sun, Feb 14, 2010 at 7:58 PM, Brian West <brian at freeswitch.org> wrote:
> Works fine here... is your box slow or something?
>
> /b
>
> On Feb 14, 2010, at 1:03 PM, Gabriel Kuri wrote:
>
>> I followed Brian's directions from one of the previous threads on
>> configuring the SPA-5xx series phones for Broadsoft SCA and set
>> manage-shared-appearance=true in the internal profile. SCA appears to
>> be working on outgoing calls between two phones, the line key starts
>> flashing red on the second phone when the first phone picks up the
>> receiver to make a call. However on incoming calls, both phones ring
>> (same extension), however when one of the phones picks up the line,
>> the second phone's line key doesn't flash red or show the first phone
>> on that incoming call. Any ideas? Does shared appearance only work on
>> outgoing phone calls?
>>
>> Thanks,
>> Gabe
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From mike at jerris.com  Sun Feb 14 22:19:32 2010
From: mike at jerris.com (Michael Jerris)
Date: Mon, 15 Feb 2010 01:19:32 -0500
Subject: [Freeswitch-users] Version 1.5
In-Reply-To: <ec2ce17e1002142142g1d71be24yf55132344b880514@mail.gmail.com>
References: <mailman.6373.1266207467.2495.freeswitch-users@lists.freeswitch.org>
	<041101caadf9$f36d77e0$da4867a0$@com>
	<ec2ce17e1002142142g1d71be24yf55132344b880514@mail.gmail.com>
Message-ID: <976016A5-C29F-4962-8779-F4DC257952F3@jerris.com>

I think 1.5 is probably quite a long ways off, 1.0.5 should be very soon now.

Mike

On Feb 15, 2010, at 12:42 AM, jay binks wrote:

> did you contribute to the dinner ??
> maybe thats why it hasnt been released yet ...
> 
> 
> On Mon, Feb 15, 2010 at 2:47 PM, Goran Donev <gorand at donevtechconsulting.com> wrote:
> When is version 1.5 of free switch going to be released. The last update on
> the website was on the week of Feb 8th. I still have not seen anything else.

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100215/0569e1b1/attachment.html 

From jhonsonj at live.com  Sun Feb 14 22:18:52 2010
From: jhonsonj at live.com (John Jhonson)
Date: Mon, 15 Feb 2010 11:18:52 +0500
Subject: [Freeswitch-users] Looking Forward to know to create Partition in FS
Message-ID: <SNT126-W3516D0822131A1155DE0A5B84A0@phx.gbl>


Hi all,

I'm newbie in FS. I want to know how can I create/setup partitions in FS, like market well known VoIP Soft Switch products i.e. Nextone, VoIP Switch, Myra,etc can give partitions instead on buying whole product. Like I want to buy partition of 300 channels from any above mentioned products and add my carriers terminating gateways for whole sale scenario. 

Kindly advise/suggest me how can I do this setup in FS?



--

Regards,

John  


 		 	   		  
_________________________________________________________________
Hotmail: Free, trusted and rich email service.
https://signup.live.com/signup.aspx?id=60969
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100215/193390ba/attachment.html 

From mike at jerris.com  Sun Feb 14 22:38:16 2010
From: mike at jerris.com (Michael Jerris)
Date: Mon, 15 Feb 2010 01:38:16 -0500
Subject: [Freeswitch-users] FreeSWITCH Debian packages - build issues
In-Reply-To: <20100214042033.GA19822@jdc.jasonjgw.net>
References: <20100213030812.GA19108@jdc.jasonjgw.net>
	<201002130957.11633.errotan@gmail.com>
	<20100213090907.GA29452@jdc.jasonjgw.net>
	<201002131109.35877.errotan@gmail.com>
	<20100213225320.GA4990@jdc.jasonjgw.net>
	<20100214042033.GA19822@jdc.jasonjgw.net>
Message-ID: <9CF8766D-2367-4275-ACCF-0F9C6ABC7A09@jerris.com>

Did anyone bother opening a bug on jira for this or are we going to just tag 1.0.5 without deb packages?

Mie

On Feb 13, 2010, at 11:20 PM, Jason White wrote:

> Just to close this thread for now, FreeSWITCH builds correctly if I remove the
> memcache module from the Debian package files.
> 
> Maybe when memcache in FreeSWITCH is updated to libmemcache 0.37 (which is in
> Debian unstable currently) the autoconf problem, which I'm not inclined to
> track down myself at the moement as I don't use memcache, will go away.
> 



From jason at jasonjgw.net  Sun Feb 14 22:46:16 2010
From: jason at jasonjgw.net (Jason White)
Date: Mon, 15 Feb 2010 17:46:16 +1100
Subject: [Freeswitch-users] FreeSWITCH Debian packages - build issues
In-Reply-To: <9CF8766D-2367-4275-ACCF-0F9C6ABC7A09@jerris.com>
References: <20100213030812.GA19108@jdc.jasonjgw.net>
	<201002130957.11633.errotan@gmail.com>
	<20100213090907.GA29452@jdc.jasonjgw.net>
	<201002131109.35877.errotan@gmail.com>
	<20100213225320.GA4990@jdc.jasonjgw.net>
	<20100214042033.GA19822@jdc.jasonjgw.net>
	<9CF8766D-2367-4275-ACCF-0F9C6ABC7A09@jerris.com>
Message-ID: <20100215064616.GA32700@jdc.jasonjgw.net>

Michael Jerris <mike at jerris.com> wrote:
> Did anyone bother opening a bug on jira for this or are we going to just tag
> 1.0.5 without deb packages?

Has anyone tried building these on Ubuntu 9.10 or Debian 5.0? I'm not in a
position to do so at the moment.



From jason at jasonjgw.net  Sun Feb 14 22:50:57 2010
From: jason at jasonjgw.net (Jason White)
Date: Mon, 15 Feb 2010 17:50:57 +1100
Subject: [Freeswitch-users] Version 1.5
In-Reply-To: <041101caadf9$f36d77e0$da4867a0$@com>
References: <mailman.6373.1266207467.2495.freeswitch-users@lists.freeswitch.org>
	<041101caadf9$f36d77e0$da4867a0$@com>
Message-ID: <20100215065057.GB32700@jdc.jasonjgw.net>

Goran Donev <gorand at donevtechconsulting.com> wrote:
> When is version 1.5 of free switch going to be released. The last update on
> the website was on the week of Feb 8th. I still have not seen anything else.

It will happen sooner if you help to find and fix the bugs, or if you
contribute funding to the development effort.



From mike at jerris.com  Sun Feb 14 22:52:01 2010
From: mike at jerris.com (Michael Jerris)
Date: Mon, 15 Feb 2010 01:52:01 -0500
Subject: [Freeswitch-users] Looking Forward to know to create Partition
	in FS
In-Reply-To: <SNT126-W3516D0822131A1155DE0A5B84A0@phx.gbl>
References: <SNT126-W3516D0822131A1155DE0A5B84A0@phx.gbl>
Message-ID: <8B6411F5-9ED2-4FF1-BDEF-5DA9CF49DDAB@jerris.com>

You can buy as many partitions from me as you like, I take paypal.

Mike

p.s., take a look on the wiki at sofia profiles and domains and mod limit, or just run multiple instances.



On Feb 15, 2010, at 1:18 AM, John Jhonson wrote:

> Hi all,
> 
> I'm newbie in FS. I want to know how can I create/setup partitions in FS, like market well known VoIP Soft Switch products i.e. Nextone, VoIP Switch, Myra,etc can give partitions instead on buying whole product. Like I want to buy partition of 300 channels from any above mentioned products and add my carriers terminating gateways for whole sale scenario. 
> 
> Kindly advise/suggest me how can I do this setup in FS?

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100215/5a8d480c/attachment-0001.html 

From dome at tel.co.th  Sun Feb 14 23:19:42 2010
From: dome at tel.co.th (Dome Charoenyost)
Date: Mon, 15 Feb 2010 14:19:42 +0700
Subject: [Freeswitch-users] Looking Forward to know to create Partition
	in FS
In-Reply-To: <SNT126-W3516D0822131A1155DE0A5B84A0@phx.gbl>
References: <SNT126-W3516D0822131A1155DE0A5B84A0@phx.gbl>
Message-ID: <8ccbff061002142319y4b2a6929m2ee6596654f1adc2@mail.gmail.com>

http://wiki.freeswitch.org/wiki/Multi-tenant

2010/2/15 John Jhonson <jhonsonj at live.com>:
> Hi all,
>
> I'm newbie in FS. I want to know how can I create/setup partitions in FS,
> like market well known VoIP Soft Switch products i.e. Nextone, VoIP Switch,
> Myra,etc can give partitions instead on buying whole product. Like I want to
> buy partition of 300 channels from any above mentioned products and add my
> carriers terminating gateways for whole sale scenario.
>
> Kindly advise/suggest me how can I do this setup in FS?
>
>
>
> --
>
> Regards,
>
> John
>
>
>
> ________________________________
> Hotmail: Free, trusted and rich email service. Get it now.
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From steve at justfone.com  Mon Feb 15 00:50:44 2010
From: steve at justfone.com (Steven Brown)
Date: Mon, 15 Feb 2010 08:50:44 +0000
Subject: [Freeswitch-users] Other party does not hear voice when
	connecting with X-Lite
Message-ID: <3e6d7b0c1002150050g3ccd23aas2e7f05f230bff4a7@mail.gmail.com>

I had the same problem with XLite / Freeswitch  a while back that I never
fully understood,  however the problem vanished when I disabled all codecs
on Xlite except G711 uLaw, as I say, no idea what was going on but this
might be worth trying.

Steve


Message: 1
Date: Sun, 14 Feb 2010 09:04:09 -0600
From: Anthony Minessale <anthony.minessale at gmail.com>
Subject: Re: [Freeswitch-users] Other party does not hear voice when
       connecting with X-Lite
To: freeswitch-users at lists.freeswitch.org
Message-ID:
       <191c3a031002140704g705bfc73rd8dd103f3d846062 at mail.gmail.com
>
Content-Type: text/plain; charset="iso-8859-1"

You need to describe this again its too confusing now.
List each device, freeswitch, the phones and which ip and combo of addrs it
uses with the topology clearly stated.
Your attempt to simplify your explanation is actually making it harder to
follow.
Also consider a debug/sip trace as well.

Include
sofia status profile default.

Then capture a test call after entering these commands.

console loglevel debug.
sofa profile internal siptrace on

On Feb 14, 2010 4:12 AM, "Vitalii Colosov" <vetali100 at gmail.com> wrote:

No, it is done on the different PCs...

Sorry, when I started the topic, I have described the problem how it is
visible from PC of my friend.
Then I tried to reproduce the same on my own PC, and you are right...I was
not able to hear anything as well, not only both party wasn't.
Also, from my PC I was NOT able to hear guitar on test number "9999".
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100215/37c7f74d/attachment.html 

From kond at nstel.ru  Mon Feb 15 01:46:00 2010
From: kond at nstel.ru (Nikolay Kondratyev)
Date: Mon, 15 Feb 2010 12:46:00 +0300
Subject: [Freeswitch-users] h323 - sip call is not working
In-Reply-To: <65d96fc81002121246r48e867abp3c11f7f72a0ee906@mail.gmail.com>
Message-ID: <20100215094600.AED9312036@mail.nstel.ru>

Tihomir, 

I've just sent the trace to your gmail address.

Nikolay.

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Tihomir
Culjaga
Sent: Friday, February 12, 2010 11:46 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] h323 - sip call is not working

 

 

On Fri, Feb 12, 2010 at 3:29 PM, Brian West <brian at freeswitch.org> wrote:

This is a rather broad assumption.  I have seen 40ms, 60ms and even 80ms in
the wild.  It all depends on what you want to do.  It lowers overhead and
increases efficiency on the wire.

 

/b

 

On Feb 12, 2010, at 5:24 AM, Nikolay Kondratyev wrote:





By the way, I know that one can use different packetization times for the
same codec, but I've never heard, that somebody really uses 30 ms for
G711Alaw. Always 20ms.

 


everything above 60 ms is a nonsense ... and ugly :) It screws your voice
quality not even thinking VBD (voice band data) over that line :).

Anyhow, Nikolay, your problem is broken IPO h323 stack and the know avaya
"flexibility" when interoping with other vendor equipments. Here IPO is
unable to negotiate a different framing size than the default and sadly this
is the core of the problem.

Please, can you send me two tcpdump captures of calls between IPO and FS:

1.	a capture with fast start & h245tunneling=true
2.	a captire with fast start & h245tunelling=true + h245inSetup


I just want to be sure of something.

T.

 

 


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100215/5a399dac/attachment.html 

From mbsip at gazeta.pl  Mon Feb 15 02:50:30 2010
From: mbsip at gazeta.pl (mbsip)
Date: Mon, 15 Feb 2010 11:50:30 +0100
Subject: [Freeswitch-users] mod_python fetching data from mysql for VM
In-Reply-To: <28f27f5d1002140652q656ab6deh40109b223d4f2f4a@mail.gmail.com>
References: <28f27f5d1002131203j132c6230sa260358611a16247@mail.gmail.com>
	<8E4B95E9-3914-4FEE-9C8C-9A5B4C5E2AB2@jerris.com>
	<28f27f5d1002131354k23a0eaf5jc51b05106c68d719@mail.gmail.com>
	<28f27f5d1002131359h164e4077x326f646ff2367ad9@mail.gmail.com>
	<ebbc2d551002131420v58cf95f2lfc9356f2377728b6@mail.gmail.com>
	<28f27f5d1002131438l58718bbat304407f49397c965@mail.gmail.com>
	<28f27f5d1002140652q656ab6deh40109b223d4f2f4a@mail.gmail.com>
Message-ID: <28f27f5d1002150250u7add2d0fq79b3a41803587f@mail.gmail.com>

Anybody could help with this?

Thx,
Maciej.


> Hi.
>
> Please correct me if my approach is okay.
> 1. in python.conf.xml
> ? ?<param name="xml-handler-script" value="testscript"/>
> ? ?<param name="xml-handler-bindings" value="directory"/>
> 2. in dialplan
> ? ?<extension name="public_did_11digits">
> ? ? ?<condition field="destination_number" expression="^(\d{11})$">
> ? ? ? ?<action application="answer"/>
> ? ? ? ?<action application="voicemail" data="default $${domain} $1"/>
> ? ? ?</condition>
> ? ?</extension>
> 3.testscript.py (as for now only static entries)
> def xml_fetch(params):
>
> ? ? ? ?xml = '''
> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
> <document type="freeswitch/xml">
> ?<section name="directory">
> ? ?<domain name="mydomainHERE">
> ? ? ?<params>
> ? ? ? ?<param name="dial-string"
> value="{presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
> ? ? ?</params>
> ? ? ?<groups>
> ? ? ? ?<group name="default">
> ? ? ? ? <users>
> ? ? ? ? ?<user id="12345678901">
> ? ? ? ? ? ?<params>
> ? ? ? ? ? ? ? <param name="vm-mailto" value="email at aaa.com"/
> ? ? ? ? ? ?</params>
> ? ? ? ? ?</user>
> ? ? ? ?</group>
> ? ? ? </users>
> ? ? ?</groups>
> ? ?</domain>
> ?</section>
> </document>
> '''
>
> ? ? ? ?return xml
>
>
> Unfortunately aforemetnioned configuration does not work at all and
> produce following errors:
> 2010-02-14 17:31:16.878878 [DEBUG] sofia.c:4110 Channel
> sofia/internal/100 at 10.10.10.10 entering state [completed][200]
> 2010-02-14 17:31:16.878878 [DEBUG] switch_core_session.c:638 Send
> signal sofia/internal/100 at 10.10.10.10 [BREAK]
> 2010-02-14 17:31:16.878878 [NOTICE] mod_dptools.c:715 Channel
> [sofia/internal/100 at 10.10.10.10] has been answered
> EXECUTE sofia/internal/100 at 10.10.10.10 voicemail(default mydomainHERE
> 12345678901)
> 2010-02-14 17:31:16.888821 [DEBUG] mod_voicemail.c:728 [default] rwlock
> 2010-02-14 17:31:16.888821 [NOTICE] mod_python.c:118 Invoking py module: obadamy
> 2010-02-14 17:31:16.888821 [DEBUG] mod_python.c:188 Call python script
> 2010-02-14 17:31:16.888821 [DEBUG] mod_python.c:191 Finished calling
> python script
> 2010-02-14 17:31:16.888821 [ERR] mod_python.c:200 Error calling python script
> 2010-02-14 17:31:16.888821 [WARNING] mod_voicemail.c:2923 Can't find
> user [12345678901 at mydomainHERE]
> 2010-02-14 17:31:16.888821 [DEBUG] switch_ivr_play_say.c:118 No
> language specified - Using [en]
> 2010-02-14 17:31:16.888821 [DEBUG] sofia.c:4110 Channel
> sofia/internal/100 at 10.10.10.10 entering state [ready][200]
> 2010-02-14 17:31:16.888821 [DEBUG] switch_ivr_play_say.c:273 Handle
> play-file:[voicemail/vm-goodbye.wav] (en:en)
> 2010-02-14 17:31:16.888821 [DEBUG] switch_ivr_play_say.c:1158 Codec
> Activated L16 at 8000hz 1 channels 20ms
>
> The same output with simplified python script.
> def xml_fetch(params):
>
> ? ? ? ?xml = '''
> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
> <document type="freeswitch/xml">
> ?<section name="directory">
> ? ?<domain name="62.29.162.90">
> ? ? ?<user id="48223891699">
> ? ? ? ?<params>
> ? ? ? ?</params>
> ? ? ?</user>
> ? ?</domain>
> ?</section>
> </document>
> '''
>
> ? ? ? ?return xml
>
>
>
> As for now I have no idea how to solve this, but still digging.
> Funny is that dialplan bindings work okay.
>
> Any help pls.
> Thx,
> Maciej
>
>
> 2010/2/13 mbsip <mbsip at gazeta.pl>:
>> Jo?o,
>>
>> Thanks for hint, because i don't know how the db fetched data could be
>> used with voicemail.
>> I am about to ready it carefully :P
>>
>> Thanks,
>> Maciej
>>
>> 2010/2/13 Jo?o Mesquita <jmesquita at freeswitch.org>:
>>> Maciej,
>>>
>>> Take a look at the xml_hooks we have on mod_python. Might do the trick for
>>> you.
>>>
>>> http://svn.freeswitch.org/svn/freeswitch/trunk/src/mod/languages/mod_python/python_example.py
>>>
>>> JM
>>>
>>>
>>> On Sat, Feb 13, 2010 at 6:59 PM, mbsip <mbsip at gazeta.pl> wrote:
>>>>
>>>> There is a lack of connection between fatched data and voicemail and I
>>>> dont know how to achieve it.
>>>>
>>>> Thx,
>>>> Maciej.
>>>>
>>>>
>>>> 2010/2/13 mbsip <mbsip at gazeta.pl>:
>>>> > Thx for prompt reply.
>>>> >
>>>> > The main task is to be able to use Mysql db in conjunction with VM
>>>> > (but not only voicemail_msgs, voicemail_prefs).
>>>> >
>>>> > Lets imagine sb is calling 1000 and wants to record the message.
>>>> > According to mod_voicemail settings message should be sent to some
>>>> > email address.
>>>> > But the information about user 1000 and his settings like email
>>>> > address, passwd, quota should be fetched from Mysql db, not from
>>>> > directory/default/1000.xml.
>>>> > That's why I am using <action application="python" data="test"/> in my
>>>> > dialplan to work with python script which in turn should do the magic.
>>>> > The script should be able to gather all necessery data about user 1000
>>>> > (like email address in shown example) and use them in VM.
>>>> >
>>>> > So the problem is how to modify the script to force voicemail app to
>>>> > use data from DB.
>>>> > Currently ?session.execute("voicemail", "default ${domain} " +
>>>> > the_dest) is still using .xml files.
>>>> >
>>>> > Thx,
>>>> > Maciej.
>>>> >
>>>> >
>>>> > 2010/2/13 Michael Jerris <mike at jerris.com>:
>>>> >> Can you describe what your trying to accomplish, I don't understand
>>>> >> what the goal is. ?What feature are you looking for that does not already
>>>> >> exist in mod_voiceamil.
>>>> >>
>>>> >> Mike
>>>> >>
>>>> >> On Feb 13, 2010, at 3:03 PM, mbsip wrote:
>>>> >>
>>>> >>> Hello,
>>>> >>>
>>>> >>> I am trying to use mod_python to fetch data from Mysql db (through
>>>> >>> ODBC) and execute voicemail application.
>>>> >>> Below a part of my script:
>>>> >>>
>>>> >>> db=MySQLdb.connect("localhost","root","","test")
>>>> >>> ? ? ? Cursor=db.cursor()
>>>> >>> ? ? ? sql = "SELECT email FROM VM WHERE called_num=%s" % the_dest
>>>> >>> ? ? ? Cursor.execute(sql)
>>>> >>> ? ? ? while (1):
>>>> >>> ? ? ? ? ? ? ? Results = Cursor.fetchone()
>>>> >>> ? ? ? ? ? ? ? if Results == None:
>>>> >>> ? ? ? ? ? ? ? ? ? ? ? break
>>>> >>> ? ? ? ? ? ? ? consoleLog("debug", "Found email " + Results[0] +"\n")
>>>> >>> ? ? ? ? ? ? ? the_recipient = Results[0]
>>>> >>> ? ? ? db.close()
>>>> >>>
>>>> >>> Now i have email address corresponding with called number. The
>>>> >>> question is how to use it for voicemail application?
>>>> >>> So it also means how to omit all /directory/default/....xml, where
>>>> >>> there are all VM parameters set and use fetched data.
>>>> >>>
>>>> >>> ? ? ? session.answer()
>>>> >>> ? ? ? session.execute("voicemail", "default ${domain} " + the_dest)
>>>> >>>
>>>> >>> Is this possible or should I start all VM app in python from the
>>>> >>> scratch?
>>>> >>
>>>> >>
>>>> >> _______________________________________________
>>>> >> FreeSWITCH-users mailing list
>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >>
>>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> >> http://www.freeswitch.org
>>>> >>
>>>> >
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>


From mbsip at gazeta.pl  Mon Feb 15 03:31:00 2010
From: mbsip at gazeta.pl (mbsip)
Date: Mon, 15 Feb 2010 12:31:00 +0100
Subject: [Freeswitch-users] mod_python fetching data from mysql for VM
In-Reply-To: <28f27f5d1002150250u7add2d0fq79b3a41803587f@mail.gmail.com>
References: <28f27f5d1002131203j132c6230sa260358611a16247@mail.gmail.com>
	<8E4B95E9-3914-4FEE-9C8C-9A5B4C5E2AB2@jerris.com>
	<28f27f5d1002131354k23a0eaf5jc51b05106c68d719@mail.gmail.com>
	<28f27f5d1002131359h164e4077x326f646ff2367ad9@mail.gmail.com>
	<ebbc2d551002131420v58cf95f2lfc9356f2377728b6@mail.gmail.com>
	<28f27f5d1002131438l58718bbat304407f49397c965@mail.gmail.com>
	<28f27f5d1002140652q656ab6deh40109b223d4f2f4a@mail.gmail.com>
	<28f27f5d1002150250u7add2d0fq79b3a41803587f@mail.gmail.com>
Message-ID: <28f27f5d1002150331k1001ac14o2e2bbbd27a96cbc7@mail.gmail.com>

Similiar script written in lua works okay.
Maybe there is sth wrong with providing dynamic directory information
via mod_python.

Thx,
Maciej

2010/2/15 mbsip <mbsip at gazeta.pl>:
> Anybody could help with this?
>
> Thx,
> Maciej.
>
>
>> Hi.
>>
>> Please correct me if my approach is okay.
>> 1. in python.conf.xml
>> ? ?<param name="xml-handler-script" value="testscript"/>
>> ? ?<param name="xml-handler-bindings" value="directory"/>
>> 2. in dialplan
>> ? ?<extension name="public_did_11digits">
>> ? ? ?<condition field="destination_number" expression="^(\d{11})$">
>> ? ? ? ?<action application="answer"/>
>> ? ? ? ?<action application="voicemail" data="default $${domain} $1"/>
>> ? ? ?</condition>
>> ? ?</extension>
>> 3.testscript.py (as for now only static entries)
>> def xml_fetch(params):
>>
>> ? ? ? ?xml = '''
>> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
>> <document type="freeswitch/xml">
>> ?<section name="directory">
>> ? ?<domain name="mydomainHERE">
>> ? ? ?<params>
>> ? ? ? ?<param name="dial-string"
>> value="{presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
>> ? ? ?</params>
>> ? ? ?<groups>
>> ? ? ? ?<group name="default">
>> ? ? ? ? <users>
>> ? ? ? ? ?<user id="12345678901">
>> ? ? ? ? ? ?<params>
>> ? ? ? ? ? ? ? <param name="vm-mailto" value="email at aaa.com"/
>> ? ? ? ? ? ?</params>
>> ? ? ? ? ?</user>
>> ? ? ? ?</group>
>> ? ? ? </users>
>> ? ? ?</groups>
>> ? ?</domain>
>> ?</section>
>> </document>
>> '''
>>
>> ? ? ? ?return xml
>>
>>
>> Unfortunately aforemetnioned configuration does not work at all and
>> produce following errors:
>> 2010-02-14 17:31:16.878878 [DEBUG] sofia.c:4110 Channel
>> sofia/internal/100 at 10.10.10.10 entering state [completed][200]
>> 2010-02-14 17:31:16.878878 [DEBUG] switch_core_session.c:638 Send
>> signal sofia/internal/100 at 10.10.10.10 [BREAK]
>> 2010-02-14 17:31:16.878878 [NOTICE] mod_dptools.c:715 Channel
>> [sofia/internal/100 at 10.10.10.10] has been answered
>> EXECUTE sofia/internal/100 at 10.10.10.10 voicemail(default mydomainHERE
>> 12345678901)
>> 2010-02-14 17:31:16.888821 [DEBUG] mod_voicemail.c:728 [default] rwlock
>> 2010-02-14 17:31:16.888821 [NOTICE] mod_python.c:118 Invoking py module: obadamy
>> 2010-02-14 17:31:16.888821 [DEBUG] mod_python.c:188 Call python script
>> 2010-02-14 17:31:16.888821 [DEBUG] mod_python.c:191 Finished calling
>> python script
>> 2010-02-14 17:31:16.888821 [ERR] mod_python.c:200 Error calling python script
>> 2010-02-14 17:31:16.888821 [WARNING] mod_voicemail.c:2923 Can't find
>> user [12345678901 at mydomainHERE]
>> 2010-02-14 17:31:16.888821 [DEBUG] switch_ivr_play_say.c:118 No
>> language specified - Using [en]
>> 2010-02-14 17:31:16.888821 [DEBUG] sofia.c:4110 Channel
>> sofia/internal/100 at 10.10.10.10 entering state [ready][200]
>> 2010-02-14 17:31:16.888821 [DEBUG] switch_ivr_play_say.c:273 Handle
>> play-file:[voicemail/vm-goodbye.wav] (en:en)
>> 2010-02-14 17:31:16.888821 [DEBUG] switch_ivr_play_say.c:1158 Codec
>> Activated L16 at 8000hz 1 channels 20ms
>>
>> The same output with simplified python script.
>> def xml_fetch(params):
>>
>> ? ? ? ?xml = '''
>> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
>> <document type="freeswitch/xml">
>> ?<section name="directory">
>> ? ?<domain name="62.29.162.90">
>> ? ? ?<user id="48223891699">
>> ? ? ? ?<params>
>> ? ? ? ?</params>
>> ? ? ?</user>
>> ? ?</domain>
>> ?</section>
>> </document>
>> '''
>>
>> ? ? ? ?return xml
>>
>>
>>
>> As for now I have no idea how to solve this, but still digging.
>> Funny is that dialplan bindings work okay.
>>
>> Any help pls.
>> Thx,
>> Maciej
>>
>>
>> 2010/2/13 mbsip <mbsip at gazeta.pl>:
>>> Jo?o,
>>>
>>> Thanks for hint, because i don't know how the db fetched data could be
>>> used with voicemail.
>>> I am about to ready it carefully :P
>>>
>>> Thanks,
>>> Maciej
>>>
>>> 2010/2/13 Jo?o Mesquita <jmesquita at freeswitch.org>:
>>>> Maciej,
>>>>
>>>> Take a look at the xml_hooks we have on mod_python. Might do the trick for
>>>> you.
>>>>
>>>> http://svn.freeswitch.org/svn/freeswitch/trunk/src/mod/languages/mod_python/python_example.py
>>>>
>>>> JM
>>>>
>>>>
>>>> On Sat, Feb 13, 2010 at 6:59 PM, mbsip <mbsip at gazeta.pl> wrote:
>>>>>
>>>>> There is a lack of connection between fatched data and voicemail and I
>>>>> dont know how to achieve it.
>>>>>
>>>>> Thx,
>>>>> Maciej.
>>>>>
>>>>>
>>>>> 2010/2/13 mbsip <mbsip at gazeta.pl>:
>>>>> > Thx for prompt reply.
>>>>> >
>>>>> > The main task is to be able to use Mysql db in conjunction with VM
>>>>> > (but not only voicemail_msgs, voicemail_prefs).
>>>>> >
>>>>> > Lets imagine sb is calling 1000 and wants to record the message.
>>>>> > According to mod_voicemail settings message should be sent to some
>>>>> > email address.
>>>>> > But the information about user 1000 and his settings like email
>>>>> > address, passwd, quota should be fetched from Mysql db, not from
>>>>> > directory/default/1000.xml.
>>>>> > That's why I am using <action application="python" data="test"/> in my
>>>>> > dialplan to work with python script which in turn should do the magic.
>>>>> > The script should be able to gather all necessery data about user 1000
>>>>> > (like email address in shown example) and use them in VM.
>>>>> >
>>>>> > So the problem is how to modify the script to force voicemail app to
>>>>> > use data from DB.
>>>>> > Currently ?session.execute("voicemail", "default ${domain} " +
>>>>> > the_dest) is still using .xml files.
>>>>> >
>>>>> > Thx,
>>>>> > Maciej.
>>>>> >
>>>>> >
>>>>> > 2010/2/13 Michael Jerris <mike at jerris.com>:
>>>>> >> Can you describe what your trying to accomplish, I don't understand
>>>>> >> what the goal is. ?What feature are you looking for that does not already
>>>>> >> exist in mod_voiceamil.
>>>>> >>
>>>>> >> Mike
>>>>> >>
>>>>> >> On Feb 13, 2010, at 3:03 PM, mbsip wrote:
>>>>> >>
>>>>> >>> Hello,
>>>>> >>>
>>>>> >>> I am trying to use mod_python to fetch data from Mysql db (through
>>>>> >>> ODBC) and execute voicemail application.
>>>>> >>> Below a part of my script:
>>>>> >>>
>>>>> >>> db=MySQLdb.connect("localhost","root","","test")
>>>>> >>> ? ? ? Cursor=db.cursor()
>>>>> >>> ? ? ? sql = "SELECT email FROM VM WHERE called_num=%s" % the_dest
>>>>> >>> ? ? ? Cursor.execute(sql)
>>>>> >>> ? ? ? while (1):
>>>>> >>> ? ? ? ? ? ? ? Results = Cursor.fetchone()
>>>>> >>> ? ? ? ? ? ? ? if Results == None:
>>>>> >>> ? ? ? ? ? ? ? ? ? ? ? break
>>>>> >>> ? ? ? ? ? ? ? consoleLog("debug", "Found email " + Results[0] +"\n")
>>>>> >>> ? ? ? ? ? ? ? the_recipient = Results[0]
>>>>> >>> ? ? ? db.close()
>>>>> >>>
>>>>> >>> Now i have email address corresponding with called number. The
>>>>> >>> question is how to use it for voicemail application?
>>>>> >>> So it also means how to omit all /directory/default/....xml, where
>>>>> >>> there are all VM parameters set and use fetched data.
>>>>> >>>
>>>>> >>> ? ? ? session.answer()
>>>>> >>> ? ? ? session.execute("voicemail", "default ${domain} " + the_dest)
>>>>> >>>
>>>>> >>> Is this possible or should I start all VM app in python from the
>>>>> >>> scratch?
>>>>> >>
>>>>> >>
>>>>> >> _______________________________________________
>>>>> >> FreeSWITCH-users mailing list
>>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> >>
>>>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> >> http://www.freeswitch.org
>>>>> >>
>>>>> >
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>
>


From paul at apcl.us  Mon Feb 15 04:46:10 2010
From: paul at apcl.us (Paul Levin)
Date: Mon, 15 Feb 2010 07:46:10 -0500
Subject: [Freeswitch-users] can vm settings go in
 conf\directory\default\default.xml?
Message-ID: <4B794212.9010406@apcl.us>

An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100215/a0454a49/attachment.html 

From tculjaga at gmail.com  Mon Feb 15 05:49:43 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Mon, 15 Feb 2010 14:49:43 +0100
Subject: [Freeswitch-users] h323 - sip call is not working
In-Reply-To: <20100215094600.AED9312036@mail.nstel.ru>
References: <65d96fc81002121246r48e867abp3c11f7f72a0ee906@mail.gmail.com>
	<20100215094600.AED9312036@mail.nstel.ru>
Message-ID: <65d96fc81002150549j4cd129a8m8566e442e17de1f8@mail.gmail.com>

On Mon, Feb 15, 2010 at 10:46 AM, Nikolay Kondratyev <kond at nstel.ru> wrote:

>  Tihomir,
>
> I?ve just sent the trace to your gmail address?
>

didn't get anything...


> Nikolay.
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100215/fe2c30b5/attachment.html 

From brian at freeswitch.org  Mon Feb 15 06:36:20 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 15 Feb 2010 08:36:20 -0600
Subject: [Freeswitch-users] Forbidden using UDP, works with TCP/TLS
In-Reply-To: <33c87fa31002142109l5fcabed3ydba21b865ed6cbbb@mail.gmail.com>
References: <33c87fa31002101824h21e3bf7lb13b68e8d4fd36a2@mail.gmail.com>
	<0A905834-850F-4D1F-AEC5-D3A87CE4141B@jerris.com>
	<33c87fa31002121629k74f26807i131841aa4252461b@mail.gmail.com>
	<33c87fa31002130343k7a01e9abt9c33360399f246f8@mail.gmail.com>
	<C37AB6F3-F712-41B6-9D6D-7A8AE3C6CABB@freeswitch.org>
	<33c87fa31002131336j7b05db7aof6ba6e3439aebc41@mail.gmail.com>
	<9D702362-BB5F-4C0F-9EE7-DABA3B926C36@freeswitch.org>
	<33c87fa31002141708s72cb7246h5372ecea40be615d@mail.gmail.com>
	<1888B1CA-D0A8-407C-9C44-B86E0F1925D6@freeswitch.org>
	<33c87fa31002142017v49d8e7c5nf572d086ec89282b@mail.gmail.com>
	<33c87fa31002142109l5fcabed3ydba21b865ed6cbbb@mail.gmail.com>
Message-ID: <D8F079A5-3EFA-41C1-8F37-780DBF5DDD44@freeswitch.org>

Interesting... I wonder if we have to echo back the qop token?

/b

On Feb 14, 2010, at 11:09 PM, Mark Campbell-Smith wrote:

> A little more testing.    I noticed that the Authorization field
> differs when TCP or UDP:
> 
> UDP (fails)
> Digest username=\"2010\", realm=\"mydns.dyndns.org\",
> nonce=\"e5f119c6-19e9-11df-bd09-773b7a755f78\",
> uri=\"sip:mydns.dyndns.org:5060\",
> response=\"e37be3e49c159d4f98e8bd04b36f2bd7\", algorithm=MD5,
> qop=\"1fffcc9f\"
> 
> TCP (works)
> Digest username=\"2010\",realm=\"mydns.dyndns.org\",nonce=\"5d9e75c2-19ea-11df-bd0b-773b7a755f78\",uri=\"sip:mydns.dyndns.org:5060\",algorithm=MD5,response=\"45ba55d3fbafcbf2bc2aa6418656ecc2\",qop=auth,nc=00000001,cnonce=\"3a650454\"
> 
> Is qop = 1fffcc9f valid in SIP?  Does a cnonce need to be included also?
> 
> FS sends qop=auth in the Unauthorized response.
> 
> Thanks

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100215/adb9e06f/attachment.html 

From brian at freeswitch.org  Mon Feb 15 07:03:22 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 15 Feb 2010 09:03:22 -0600
Subject: [Freeswitch-users] Forbidden using UDP, works with TCP/TLS
In-Reply-To: <33c87fa31002142109l5fcabed3ydba21b865ed6cbbb@mail.gmail.com>
References: <33c87fa31002101824h21e3bf7lb13b68e8d4fd36a2@mail.gmail.com>
	<0A905834-850F-4D1F-AEC5-D3A87CE4141B@jerris.com>
	<33c87fa31002121629k74f26807i131841aa4252461b@mail.gmail.com>
	<33c87fa31002130343k7a01e9abt9c33360399f246f8@mail.gmail.com>
	<C37AB6F3-F712-41B6-9D6D-7A8AE3C6CABB@freeswitch.org>
	<33c87fa31002131336j7b05db7aof6ba6e3439aebc41@mail.gmail.com>
	<9D702362-BB5F-4C0F-9EE7-DABA3B926C36@freeswitch.org>
	<33c87fa31002141708s72cb7246h5372ecea40be615d@mail.gmail.com>
	<1888B1CA-D0A8-407C-9C44-B86E0F1925D6@freeswitch.org>
	<33c87fa31002142017v49d8e7c5nf572d086ec89282b@mail.gmail.com>
	<33c87fa31002142109l5fcabed3ydba21b865ed6cbbb@mail.gmail.com>
Message-ID: <147993F8-9C9E-4D96-9AFE-5EEE4E8F93F0@freeswitch.org>

Ok looks like the token is not used at all in digest auth.  This is the first time I have seen a device send back something other than auth or auth-int.

/b

On Feb 14, 2010, at 11:09 PM, Mark Campbell-Smith wrote:

> A little more testing.    I noticed that the Authorization field
> differs when TCP or UDP:
> 
> UDP (fails)
> Digest username=\"2010\", realm=\"mydns.dyndns.org\",
> nonce=\"e5f119c6-19e9-11df-bd09-773b7a755f78\",
> uri=\"sip:mydns.dyndns.org:5060\",
> response=\"e37be3e49c159d4f98e8bd04b36f2bd7\", algorithm=MD5,
> qop=\"1fffcc9f\"
> 
> TCP (works)
> Digest username=\"2010\",realm=\"mydns.dyndns.org\",nonce=\"5d9e75c2-19ea-11df-bd0b-773b7a755f78\",uri=\"sip:mydns.dyndns.org:5060\",algorithm=MD5,response=\"45ba55d3fbafcbf2bc2aa6418656ecc2\",qop=auth,nc=00000001,cnonce=\"3a650454\"
> 
> Is qop = 1fffcc9f valid in SIP?  Does a cnonce need to be included also?
> 
> FS sends qop=auth in the Unauthorized response.
> 
> Thanks

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100215/27cdb563/attachment-0001.html 

From kond at nstel.ru  Mon Feb 15 07:26:01 2010
From: kond at nstel.ru (Nikolay Kondratyev)
Date: Mon, 15 Feb 2010 18:26:01 +0300
Subject: [Freeswitch-users] h323 - sip call is not working
In-Reply-To: <65d96fc81002150549j4cd129a8m8566e442e17de1f8@mail.gmail.com>
Message-ID: <20100215152601.D569811FA6@mail.nstel.ru>

Sent again..

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Tihomir
Culjaga
Sent: Monday, February 15, 2010 4:50 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] h323 - sip call is not working

 

 

On Mon, Feb 15, 2010 at 10:46 AM, Nikolay Kondratyev <kond at nstel.ru> wrote:

Tihomir, 

I've just sent the trace to your gmail address.


didn't get anything...
 

Nikolay.

 

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100215/60631e74/attachment.html 

From tculjaga at gmail.com  Mon Feb 15 07:39:38 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Mon, 15 Feb 2010 16:39:38 +0100
Subject: [Freeswitch-users] h323 - sip call is not working
In-Reply-To: <20100215152601.D569811FA6@mail.nstel.ru>
References: <65d96fc81002150549j4cd129a8m8566e442e17de1f8@mail.gmail.com>
	<20100215152601.D569811FA6@mail.nstel.ru>
Message-ID: <65d96fc81002150739l242407d2u2d5cb5081f96e7eb@mail.gmail.com>

It looks like audio level issue .. can you lower the gain on IPO ?

T.

On Mon, Feb 15, 2010 at 4:26 PM, Nikolay Kondratyev <kond at nstel.ru> wrote:

>  Sent again..
>
>
>   ------------------------------
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Tihomir
> Culjaga
> *Sent:* Monday, February 15, 2010 4:50 PM
>
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] h323 - sip call is not working
>
>
>
>
>
> On Mon, Feb 15, 2010 at 10:46 AM, Nikolay Kondratyev <kond at nstel.ru>
> wrote:
>
> Tihomir,
>
> I?ve just sent the trace to your gmail address?
>
>
> didn't get anything...
>
>
>  Nikolay.
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100215/af17b8d2/attachment.html 

From vkozak at abisoft.spb.ru  Mon Feb 15 03:54:17 2010
From: vkozak at abisoft.spb.ru (Kozak Vladimir)
Date: Mon, 15 Feb 2010 14:54:17 +0300
Subject: [Freeswitch-users] CS_REPORTING state and CHANNEL_HANGUP event
Message-ID: <D0F4BBA35AD84876B35F99FCD443DAF9@abisoft.biz>

Hello all.
I have the following questions:
    1) Sometime channel whis CS_REPORTING state remain. What is channel whis CS_REPORTING state mean? for example: 2c90ac6b-7147-4aac-82fb-a23d6d5c4185,outbound,2010-02-10         11:53:02,1265820782,sofia/internal/sip:7100 at 76.74.160.163:57312,CS_REPORTING,FreeSWITCH,sipp,,7100,,,,default,,,,,,pst01.localdomain.com,, 
How can I kill this channel?  uuid_kill 2c90ac6b-7147-4aac-82fb-a23d6d5c4185 is not work.
    2) Sometime I get CHANNEL_HANGUP event but real call in eyeBeam is live for the present. When come CHANNEL_HANGUP event and response on api uuid_kill command? It's come when FS send hangup to phone or when phone confirm hangup?
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100215/924552f0/attachment.html 

From kond at nstel.ru  Mon Feb 15 08:14:08 2010
From: kond at nstel.ru (Nikolay Kondratyev)
Date: Mon, 15 Feb 2010 19:14:08 +0300
Subject: [Freeswitch-users] h323 - sip call is not working
In-Reply-To: <65d96fc81002150739l242407d2u2d5cb5081f96e7eb@mail.gmail.com>
Message-ID: <20100215161410.BF0D911F5E@mail.nstel.ru>

Mmm. I doubt that it is an audio level problem.

Look at the picture that wireshark "voip calls -> player" shows for that
wheezy rbt.

Here it is:



 

This graph shows that FS really plays "stutter" tone instead of normal rbt.
Or am I mistaken?

 

And wireshark analysis of that rtp stream is strange.it shows 22 _seconds_
for that rtp stream..

Here is the text summary of wireshark rtp stream analysis of the stream from
FS to IPO:

Max delta = 50,65 ms at packet no. 1147 

Max jitter = 22556,56 ms. Mean jitter = 238,60 ms.

Max skew = 355392,29 ms.

Total RTP packets = 1538   (expected 1538)   Lost RTP packets = 0 (0,00%)
Sequence errors = 0 

Duration 36,14 s (513134 ms clock drift, corresponding to 121603 Hz
(+1420,03%)

 

I think the problem is on FS site.

 

Thanks and regards,

Nikolay.

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Tihomir
Culjaga
Sent: Monday, February 15, 2010 6:40 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] h323 - sip call is not working

 

It looks like audio level issue .. can you lower the gain on IPO ?

T.

On Mon, Feb 15, 2010 at 4:26 PM, Nikolay Kondratyev <kond at nstel.ru> wrote:

Sent again..

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Tihomir
Culjaga
Sent: Monday, February 15, 2010 4:50 PM


To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] h323 - sip call is not working

 

 

On Mon, Feb 15, 2010 at 10:46 AM, Nikolay Kondratyev <kond at nstel.ru> wrote:

Tihomir, 

I've just sent the trace to your gmail address.


didn't get anything...
 

Nikolay.

 

 


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100215/3ef129a9/attachment-0001.html 
-------------- next part --------------
A non-text attachment was scrubbed...
Name: not available
Type: image/jpeg
Size: 75817 bytes
Desc: not available
Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100215/3ef129a9/attachment-0001.jpe 

From michal.kalinowski at interia.pl  Mon Feb 15 08:24:07 2010
From: michal.kalinowski at interia.pl (michal kalinowski)
Date: Mon, 15 Feb 2010 17:24:07 +0100
Subject: [Freeswitch-users]  ivr from mysql
Message-ID: <7c74f5761002150824i41a2ed5did008f36cbb4499a4@mail.gmail.com>

Hi,

I need build ivr script/aplication which will take dynamically
configuration from mysql db (ivr menu, prompts, etc.).
My first idea is generate xml from python script. But it's not working properly.

Anybody has some idea or have this aplication already done ?

BR,
Micha?


From anthony.minessale at gmail.com  Mon Feb 15 08:32:11 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 15 Feb 2010 10:32:11 -0600
Subject: [Freeswitch-users] Broadsoft SCA w/ Cisco SPA-5xx series
In-Reply-To: <8b1c9cda1002142159h5efcc765q9a41baa1bf694c8c@mail.gmail.com>
References: <8b1c9cda1002141103te523643ob16db25e3bfa877f@mail.gmail.com>
	<0D67132C-69D8-43AD-B7E6-C7545B544322@freeswitch.org>
	<8b1c9cda1002142159h5efcc765q9a41baa1bf694c8c@mail.gmail.com>
Message-ID: <191c3a031002150832x7faa9a78xbc160cde63f09d09@mail.gmail.com>

it should be active not seized.
seized is when you take it off hook.

We need some more debugging to be sure.
Can we work in real time on it or can you get a more detailed log?

edit sofia.conf.xml and add the param to the "settings" section.

<param name="debug-sla" value="100"/>


then restart and enable sip trace and debug level

//do this for every profile involved in the call.
sofia profile <profilename> siptrace on

//also do this
console loglevel debug


if you can let us ssh, we can do all the for you if you can make the test
calls.




On Sun, Feb 14, 2010 at 11:59 PM, Gabriel Kuri <gkuri at ieee.org> wrote:

> BTW, here's a copy of the NOTIFY (event call-info) sent to the other
> phone after the first phone is answered, should this have a Call-Info
> line with an "appearance-state=seized" to turn on the light on the
> other phone?
>
>
> NOTIFY sip:2551@<removed>:54446 SIP/2.0.
> Via: SIP/2.0/UDP <removed>:9430;rport;branch=z9hG4bK71pN2cXgH851K.
> Max-Forwards: 70.
> From: <sip:2551 at mydomain.com <sip%3A2551 at mydomain.com>>;tag=XeB6ZrKDevpHp.
> To: <sip:2551 at mydomain.com <sip%3A2551 at mydomain.com>>;tag=c2d34993aac6ea.
> Call-ID: 34c34987-8b6fa786@<removed>.
> CSeq: 126950830 NOTIFY.
> Contact: <sip:2551@<removed>:9430>.
> Expires: 3959.
> Call-Info: <sip:mydomain.com>;appearance-index=*;appearance-state=idle.
> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16633M.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, PRACK, NOTIFY, PUBLISH, SUBSCRIBE.
> Supported: 100rel, timer, precondition, path, replaces.
> Event: call-info.
> Allow-Events: talk, presence, dialog, line-seize, call-info, sla,
> include-session-description, presence.winfo, message-summary, refer.
> Subscription-State: active;expires=3959.
> Content-Length: 0.
>
>
>
> On Sun, Feb 14, 2010 at 7:58 PM, Brian West <brian at freeswitch.org> wrote:
> > Works fine here... is your box slow or something?
> >
> > /b
> >
> > On Feb 14, 2010, at 1:03 PM, Gabriel Kuri wrote:
> >
> >> I followed Brian's directions from one of the previous threads on
> >> configuring the SPA-5xx series phones for Broadsoft SCA and set
> >> manage-shared-appearance=true in the internal profile. SCA appears to
> >> be working on outgoing calls between two phones, the line key starts
> >> flashing red on the second phone when the first phone picks up the
> >> receiver to make a call. However on incoming calls, both phones ring
> >> (same extension), however when one of the phones picks up the line,
> >> the second phone's line key doesn't flash red or show the first phone
> >> on that incoming call. Any ideas? Does shared appearance only work on
> >> outgoing phone calls?
> >>
> >> Thanks,
> >> Gabe
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100215/23d10814/attachment.html 

From bottleman at icf.org.ru  Mon Feb 15 08:36:05 2010
From: bottleman at icf.org.ru (Georgiewskiy Yuriy)
Date: Mon, 15 Feb 2010 19:36:05 +0300 (MSK)
Subject: [Freeswitch-users] FreeSWITCH Debian packages - build issues
In-Reply-To: <20100214042033.GA19822@jdc.jasonjgw.net>
References: <20100213030812.GA19108@jdc.jasonjgw.net>
	<201002130957.11633.errotan@gmail.com>
	<20100213090907.GA29452@jdc.jasonjgw.net>
	<201002131109.35877.errotan@gmail.com>
	<20100213225320.GA4990@jdc.jasonjgw.net>
	<20100214042033.GA19822@jdc.jasonjgw.net>
Message-ID: <alpine.DEB.2.00.1002151931200.6101@srv.icf.org.ru>

On 2010-02-14 15:20 +1100, Jason White wrote freeswitch-users at lists.freeswi...:

JW>Just to close this thread for now, FreeSWITCH builds correctly if I remove the
JW>memcache module from the Debian package files.

after remove memcache it's not build mod_say_xx modules, and sound packages too.

JW>Maybe when memcache in FreeSWITCH is updated to libmemcache 0.37 (which is in
JW>Debian unstable currently) the autoconf problem, which I'm not inclined to
JW>track down myself at the moement as I don't use memcache, will go away.
JW>
JW>
JW>_______________________________________________
JW>FreeSWITCH-users mailing list
JW>FreeSWITCH-users at lists.freeswitch.org
JW>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
JW>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
JW>http://www.freeswitch.org
JW>

C ?????????                       With Best Regards
???????????? ????.                Georgiewskiy Yuriy
+7 4872 711666                    +7 4872 711666
???? +7 4872 711143               fax +7 4872 711143
???????? ??? "?? ?? ??????"       IT Service Ltd
http://nkoort.ru                  http://nkoort.ru
JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
YG129-RIPE                        YG129-RIPE

From msc at freeswitch.org  Mon Feb 15 09:59:41 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 15 Feb 2010 09:59:41 -0800
Subject: [Freeswitch-users] ASTPP For FreeSWITCH
Message-ID: <87f2f3b91002150959q6accece5od67b262fb38ec41e@mail.gmail.com>

Hey all,

Here's a quick story <http://www.freeswitch.org/node/239> about ASTPP and
FreeSWITCH. If you are using ASTPP with FreeSWITCH please let me know how it
works. I didn't see any information on our wiki about ASTPP. If ASTPP is
viable then we should document it as best we can.

Thanks!
-Michael
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100215/dc6c763a/attachment.html 

From msc at freeswitch.org  Mon Feb 15 10:02:12 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 15 Feb 2010 10:02:12 -0800
Subject: [Freeswitch-users] ivr from mysql
In-Reply-To: <7c74f5761002150824i41a2ed5did008f36cbb4499a4@mail.gmail.com>
References: <7c74f5761002150824i41a2ed5did008f36cbb4499a4@mail.gmail.com>
Message-ID: <87f2f3b91002151002v6bbc3e7fgb67dd6d3e72a915@mail.gmail.com>

When you say "xml from python" what exactly do you mean? Are you trying to
use mod_xml_curl? If not you might want to check it out. The other choice is
to use Lua from the dialplan, although I have a gut feeling that
mod_xml_curl might be better for you.
-MC

2010/2/15 michal kalinowski <michal.kalinowski at interia.pl>

> Hi,
>
> I need build ivr script/aplication which will take dynamically
> configuration from mysql db (ivr menu, prompts, etc.).
> My first idea is generate xml from python script. But it's not working
> properly.
>
> Anybody has some idea or have this aplication already done ?
>
> BR,
> Micha?
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100215/651f0453/attachment.html 

From gkuri at ieee.org  Mon Feb 15 10:48:13 2010
From: gkuri at ieee.org (Gabriel Kuri)
Date: Mon, 15 Feb 2010 10:48:13 -0800
Subject: [Freeswitch-users] Broadsoft SCA w/ Cisco SPA-5xx series
In-Reply-To: <191c3a031002150832x7faa9a78xbc160cde63f09d09@mail.gmail.com>
References: <8b1c9cda1002141103te523643ob16db25e3bfa877f@mail.gmail.com>
	<0D67132C-69D8-43AD-B7E6-C7545B544322@freeswitch.org>
	<8b1c9cda1002142159h5efcc765q9a41baa1bf694c8c@mail.gmail.com>
	<191c3a031002150832x7faa9a78xbc160cde63f09d09@mail.gmail.com>
Message-ID: <8b1c9cda1002151048o4b45c034qb22732a9366024ae@mail.gmail.com>

Here's a call from ext 2551 to 2552 (two phones). I noticed a bunch of
errors related to SQL UPDATE for presence ...

     http://pastebin.freeswitch.org/12152

Thanks,
Gabe


On Mon, Feb 15, 2010 at 8:32 AM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> it should be active not seized.
> seized is when you take it off hook.
>
> We need some more debugging to be sure.
> Can we work in real time on it or can you get a more detailed log?
>
> edit sofia.conf.xml and add the param to the "settings" section.
>
> <param name="debug-sla" value="100"/>
>
>
> then restart and enable sip trace and debug level
>
> //do this for every profile involved in the call.
> sofia profile <profilename> siptrace on
>
> //also do this
> console loglevel debug
>
>
> if you can let us ssh, we can do all the for you if you can make the test
> calls.
>
>
>
>
> On Sun, Feb 14, 2010 at 11:59 PM, Gabriel Kuri <gkuri at ieee.org> wrote:
>>
>> BTW, here's a copy of the NOTIFY (event call-info) sent to the other
>> phone after the first phone is answered, should this have a Call-Info
>> line with an "appearance-state=seized" to turn on the light on the
>> other phone?
>>
>>
>> NOTIFY sip:2551@<removed>:54446 SIP/2.0.
>> Via: SIP/2.0/UDP <removed>:9430;rport;branch=z9hG4bK71pN2cXgH851K.
>> Max-Forwards: 70.
>> From: <sip:2551 at mydomain.com>;tag=XeB6ZrKDevpHp.
>> To: <sip:2551 at mydomain.com>;tag=c2d34993aac6ea.
>> Call-ID: 34c34987-8b6fa786@<removed>.
>> CSeq: 126950830 NOTIFY.
>> Contact: <sip:2551@<removed>:9430>.
>> Expires: 3959.
>> Call-Info: <sip:mydomain.com>;appearance-index=*;appearance-state=idle.
>> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16633M.
>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>> REGISTER, REFER, PRACK, NOTIFY, PUBLISH, SUBSCRIBE.
>> Supported: 100rel, timer, precondition, path, replaces.
>> Event: call-info.
>> Allow-Events: talk, presence, dialog, line-seize, call-info, sla,
>> include-session-description, presence.winfo, message-summary, refer.
>> Subscription-State: active;expires=3959.
>> Content-Length: 0.
>>
>>
>>
>> On Sun, Feb 14, 2010 at 7:58 PM, Brian West <brian at freeswitch.org> wrote:
>> > Works fine here... is your box slow or something?
>> >
>> > /b
>> >
>> > On Feb 14, 2010, at 1:03 PM, Gabriel Kuri wrote:
>> >
>> >> I followed Brian's directions from one of the previous threads on
>> >> configuring the SPA-5xx series phones for Broadsoft SCA and set
>> >> manage-shared-appearance=true in the internal profile. SCA appears to
>> >> be working on outgoing calls between two phones, the line key starts
>> >> flashing red on the second phone when the first phone picks up the
>> >> receiver to make a call. However on incoming calls, both phones ring
>> >> (same extension), however when one of the phones picks up the line,
>> >> the second phone's line key doesn't flash red or show the first phone
>> >> on that incoming call. Any ideas? Does shared appearance only work on
>> >> outgoing phone calls?
>> >>
>> >> Thanks,
>> >> Gabe
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From darren at aleph-com.net  Mon Feb 15 10:53:45 2010
From: darren at aleph-com.net (Darren Wiebe)
Date: Mon, 15 Feb 2010 11:53:45 -0700
Subject: [Freeswitch-users] ASTPP For FreeSWITCH
In-Reply-To: <87f2f3b91002150959q6accece5od67b262fb38ec41e@mail.gmail.com>
References: <87f2f3b91002150959q6accece5od67b262fb38ec41e@mail.gmail.com>
Message-ID: <4B799839.2090008@aleph-com.net>

I will comment.  We've been using ASTPP for rating freeswitch cdrs for 
some time already.  It provides lcr from a database as well as sip user 
management.  It uses the mod_xml_curl and mod_xml_cdr modules for 
routing as well as realtime rating.  It also has an application that can 
listen to freeswitch and rate calls in realtime that way.  I patched a 
couple of bugs earlier this morning and I would not say that it's bug 
free but it's certainly in testing.

Darren Wiebe
darren at aleph-com.net


On 02/15/2010 10:59 AM, Michael Collins wrote:
> Hey all,
>
> Here's a quick story <http://www.freeswitch.org/node/239> about ASTPP 
> and FreeSWITCH. If you are using ASTPP with FreeSWITCH please let me 
> know how it works. I didn't see any information on our wiki about 
> ASTPP. If ASTPP is viable then we should document it as best we can.
>
> Thanks!
> -Michael
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>    

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100215/5c92b492/attachment.html 

From anthony.minessale at gmail.com  Mon Feb 15 10:55:34 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 15 Feb 2010 12:55:34 -0600
Subject: [Freeswitch-users] Broadsoft SCA w/ Cisco SPA-5xx series
In-Reply-To: <8b1c9cda1002151048o4b45c034qb22732a9366024ae@mail.gmail.com>
References: <8b1c9cda1002141103te523643ob16db25e3bfa877f@mail.gmail.com>
	<0D67132C-69D8-43AD-B7E6-C7545B544322@freeswitch.org>
	<8b1c9cda1002142159h5efcc765q9a41baa1bf694c8c@mail.gmail.com>
	<191c3a031002150832x7faa9a78xbc160cde63f09d09@mail.gmail.com>
	<8b1c9cda1002151048o4b45c034qb22732a9366024ae@mail.gmail.com>
Message-ID: <191c3a031002151055v1938ab91p89232341f4087c8e@mail.gmail.com>

we log the sql stmts on err so they are red and easier to read.



On Mon, Feb 15, 2010 at 12:48 PM, Gabriel Kuri <gkuri at ieee.org> wrote:

> Here's a call from ext 2551 to 2552 (two phones). I noticed a bunch of
> errors related to SQL UPDATE for presence ...
>
>     http://pastebin.freeswitch.org/12152
>
> Thanks,
> Gabe
>
>
> On Mon, Feb 15, 2010 at 8:32 AM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
> > it should be active not seized.
> > seized is when you take it off hook.
> >
> > We need some more debugging to be sure.
> > Can we work in real time on it or can you get a more detailed log?
> >
> > edit sofia.conf.xml and add the param to the "settings" section.
> >
> > <param name="debug-sla" value="100"/>
> >
> >
> > then restart and enable sip trace and debug level
> >
> > //do this for every profile involved in the call.
> > sofia profile <profilename> siptrace on
> >
> > //also do this
> > console loglevel debug
> >
> >
> > if you can let us ssh, we can do all the for you if you can make the test
> > calls.
> >
> >
> >
> >
> > On Sun, Feb 14, 2010 at 11:59 PM, Gabriel Kuri <gkuri at ieee.org> wrote:
> >>
> >> BTW, here's a copy of the NOTIFY (event call-info) sent to the other
> >> phone after the first phone is answered, should this have a Call-Info
> >> line with an "appearance-state=seized" to turn on the light on the
> >> other phone?
> >>
> >>
> >> NOTIFY sip:2551@<removed>:54446 SIP/2.0.
> >> Via: SIP/2.0/UDP <removed>:9430;rport;branch=z9hG4bK71pN2cXgH851K.
> >> Max-Forwards: 70.
> >> From: <sip:2551 at mydomain.com <sip%3A2551 at mydomain.com>
> >;tag=XeB6ZrKDevpHp.
> >> To: <sip:2551 at mydomain.com <sip%3A2551 at mydomain.com>
> >;tag=c2d34993aac6ea.
> >> Call-ID: 34c34987-8b6fa786@<removed>.
> >> CSeq: 126950830 NOTIFY.
> >> Contact: <sip:2551@<removed>:9430>.
> >> Expires: 3959.
> >> Call-Info: <sip:mydomain.com>;appearance-index=*;appearance-state=idle.
> >> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16633M.
> >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> >> REGISTER, REFER, PRACK, NOTIFY, PUBLISH, SUBSCRIBE.
> >> Supported: 100rel, timer, precondition, path, replaces.
> >> Event: call-info.
> >> Allow-Events: talk, presence, dialog, line-seize, call-info, sla,
> >> include-session-description, presence.winfo, message-summary, refer.
> >> Subscription-State: active;expires=3959.
> >> Content-Length: 0.
> >>
> >>
> >>
> >> On Sun, Feb 14, 2010 at 7:58 PM, Brian West <brian at freeswitch.org>
> wrote:
> >> > Works fine here... is your box slow or something?
> >> >
> >> > /b
> >> >
> >> > On Feb 14, 2010, at 1:03 PM, Gabriel Kuri wrote:
> >> >
> >> >> I followed Brian's directions from one of the previous threads on
> >> >> configuring the SPA-5xx series phones for Broadsoft SCA and set
> >> >> manage-shared-appearance=true in the internal profile. SCA appears to
> >> >> be working on outgoing calls between two phones, the line key starts
> >> >> flashing red on the second phone when the first phone picks up the
> >> >> receiver to make a call. However on incoming calls, both phones ring
> >> >> (same extension), however when one of the phones picks up the line,
> >> >> the second phone's line key doesn't flash red or show the first phone
> >> >> on that incoming call. Any ideas? Does shared appearance only work on
> >> >> outgoing phone calls?
> >> >>
> >> >> Thanks,
> >> >> Gabe
> >> >
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> > Twitter: http://twitter.com/FreeSWITCH_wire
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> > IRC: irc.freenode.net #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> > iax:guest at conference.freeswitch.org/888
> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > pstn:+19193869900
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100215/04330486/attachment.html 

From anthony.minessale at gmail.com  Mon Feb 15 11:04:24 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 15 Feb 2010 13:04:24 -0600
Subject: [Freeswitch-users] Broadsoft SCA w/ Cisco SPA-5xx series
In-Reply-To: <191c3a031002151055v1938ab91p89232341f4087c8e@mail.gmail.com>
References: <8b1c9cda1002141103te523643ob16db25e3bfa877f@mail.gmail.com>
	<0D67132C-69D8-43AD-B7E6-C7545B544322@freeswitch.org>
	<8b1c9cda1002142159h5efcc765q9a41baa1bf694c8c@mail.gmail.com>
	<191c3a031002150832x7faa9a78xbc160cde63f09d09@mail.gmail.com>
	<8b1c9cda1002151048o4b45c034qb22732a9366024ae@mail.gmail.com>
	<191c3a031002151055v1938ab91p89232341f4087c8e@mail.gmail.com>
Message-ID: <191c3a031002151104u7413b17dh84e68ab1cf703cb9@mail.gmail.com>

I don't see any notifies at all in this trace do the profiles in question
have:
manage-shared-appearance set to true?
and are you on latest trunk?


On Mon, Feb 15, 2010 at 12:55 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> we log the sql stmts on err so they are red and easier to read.
>
>
>
>
> On Mon, Feb 15, 2010 at 12:48 PM, Gabriel Kuri <gkuri at ieee.org> wrote:
>
>> Here's a call from ext 2551 to 2552 (two phones). I noticed a bunch of
>> errors related to SQL UPDATE for presence ...
>>
>>     http://pastebin.freeswitch.org/12152
>>
>> Thanks,
>> Gabe
>>
>>
>> On Mon, Feb 15, 2010 at 8:32 AM, Anthony Minessale
>> <anthony.minessale at gmail.com> wrote:
>> > it should be active not seized.
>> > seized is when you take it off hook.
>> >
>> > We need some more debugging to be sure.
>> > Can we work in real time on it or can you get a more detailed log?
>> >
>> > edit sofia.conf.xml and add the param to the "settings" section.
>> >
>> > <param name="debug-sla" value="100"/>
>> >
>> >
>> > then restart and enable sip trace and debug level
>> >
>> > //do this for every profile involved in the call.
>> > sofia profile <profilename> siptrace on
>> >
>> > //also do this
>> > console loglevel debug
>> >
>> >
>> > if you can let us ssh, we can do all the for you if you can make the
>> test
>> > calls.
>> >
>> >
>> >
>> >
>> > On Sun, Feb 14, 2010 at 11:59 PM, Gabriel Kuri <gkuri at ieee.org> wrote:
>> >>
>> >> BTW, here's a copy of the NOTIFY (event call-info) sent to the other
>> >> phone after the first phone is answered, should this have a Call-Info
>> >> line with an "appearance-state=seized" to turn on the light on the
>> >> other phone?
>> >>
>> >>
>> >> NOTIFY sip:2551@<removed>:54446 SIP/2.0.
>> >> Via: SIP/2.0/UDP <removed>:9430;rport;branch=z9hG4bK71pN2cXgH851K.
>> >> Max-Forwards: 70.
>> >> From: <sip:2551 at mydomain.com <sip%3A2551 at mydomain.com>
>> >;tag=XeB6ZrKDevpHp.
>> >> To: <sip:2551 at mydomain.com <sip%3A2551 at mydomain.com>
>> >;tag=c2d34993aac6ea.
>> >> Call-ID: 34c34987-8b6fa786@<removed>.
>> >> CSeq: 126950830 NOTIFY.
>> >> Contact: <sip:2551@<removed>:9430>.
>> >> Expires: 3959.
>> >> Call-Info: <sip:mydomain.com
>> >;appearance-index=*;appearance-state=idle.
>> >> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16633M.
>> >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>> >> REGISTER, REFER, PRACK, NOTIFY, PUBLISH, SUBSCRIBE.
>> >> Supported: 100rel, timer, precondition, path, replaces.
>> >> Event: call-info.
>> >> Allow-Events: talk, presence, dialog, line-seize, call-info, sla,
>> >> include-session-description, presence.winfo, message-summary, refer.
>> >> Subscription-State: active;expires=3959.
>> >> Content-Length: 0.
>> >>
>> >>
>> >>
>> >> On Sun, Feb 14, 2010 at 7:58 PM, Brian West <brian at freeswitch.org>
>> wrote:
>> >> > Works fine here... is your box slow or something?
>> >> >
>> >> > /b
>> >> >
>> >> > On Feb 14, 2010, at 1:03 PM, Gabriel Kuri wrote:
>> >> >
>> >> >> I followed Brian's directions from one of the previous threads on
>> >> >> configuring the SPA-5xx series phones for Broadsoft SCA and set
>> >> >> manage-shared-appearance=true in the internal profile. SCA appears
>> to
>> >> >> be working on outgoing calls between two phones, the line key starts
>> >> >> flashing red on the second phone when the first phone picks up the
>> >> >> receiver to make a call. However on incoming calls, both phones ring
>> >> >> (same extension), however when one of the phones picks up the line,
>> >> >> the second phone's line key doesn't flash red or show the first
>> phone
>> >> >> on that incoming call. Any ideas? Does shared appearance only work
>> on
>> >> >> outgoing phone calls?
>> >> >>
>> >> >> Thanks,
>> >> >> Gabe
>> >> >
>> >> >
>> >> > _______________________________________________
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> > http://www.freeswitch.org
>> >> >
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> >
>> > --
>> > Anthony Minessale II
>> >
>> > FreeSWITCH http://www.freeswitch.org/
>> > ClueCon http://www.cluecon.com/
>> > Twitter: http://twitter.com/FreeSWITCH_wire
>> >
>> > AIM: anthm
>> > MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> > IRC: irc.freenode.net #freeswitch
>> >
>> > FreeSWITCH Developer Conference
>> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> > iax:guest at conference.freeswitch.org/888
>> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> > pstn:+19193869900
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100215/1563828c/attachment-0001.html 

From gkuri at ieee.org  Mon Feb 15 11:47:49 2010
From: gkuri at ieee.org (Gabriel Kuri)
Date: Mon, 15 Feb 2010 11:47:49 -0800
Subject: [Freeswitch-users] Broadsoft SCA w/ Cisco SPA-5xx series
In-Reply-To: <191c3a031002151104u7413b17dh84e68ab1cf703cb9@mail.gmail.com>
References: <8b1c9cda1002141103te523643ob16db25e3bfa877f@mail.gmail.com>
	<0D67132C-69D8-43AD-B7E6-C7545B544322@freeswitch.org>
	<8b1c9cda1002142159h5efcc765q9a41baa1bf694c8c@mail.gmail.com>
	<191c3a031002150832x7faa9a78xbc160cde63f09d09@mail.gmail.com>
	<8b1c9cda1002151048o4b45c034qb22732a9366024ae@mail.gmail.com>
	<191c3a031002151055v1938ab91p89232341f4087c8e@mail.gmail.com>
	<191c3a031002151104u7413b17dh84e68ab1cf703cb9@mail.gmail.com>
Message-ID: <8b1c9cda1002151147l1e072d00y69c0022e3924c3b4@mail.gmail.com>

OK, I don't know what happened there, here's another one with the
NOTIFYs. I'm on trunk rev 16633 and I have
"managed-shared-appeareance=true" on the internal profile. I'm just
making calls between internal phones.

     http://pastebin.freeswitch.org/12153

Thanks,
Gabe

On Mon, Feb 15, 2010 at 11:04 AM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> I don't see any notifies at all in this trace do the profiles in question
> have:
> manage-shared-appearance set to true?
> and are you on latest trunk?
>
>
> On Mon, Feb 15, 2010 at 12:55 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>>
>> we log the sql stmts on err so they are red and easier to read.
>>
>>
>>
>> On Mon, Feb 15, 2010 at 12:48 PM, Gabriel Kuri <gkuri at ieee.org> wrote:
>>>
>>> Here's a call from ext 2551 to 2552 (two phones). I noticed a bunch of
>>> errors related to SQL UPDATE for presence ...
>>>
>>> ? ? http://pastebin.freeswitch.org/12152
>>>
>>> Thanks,
>>> Gabe
>>>
>>>
>>> On Mon, Feb 15, 2010 at 8:32 AM, Anthony Minessale
>>> <anthony.minessale at gmail.com> wrote:
>>> > it should be active not seized.
>>> > seized is when you take it off hook.
>>> >
>>> > We need some more debugging to be sure.
>>> > Can we work in real time on it or can you get a more detailed log?
>>> >
>>> > edit sofia.conf.xml and add the param to the "settings" section.
>>> >
>>> > <param name="debug-sla" value="100"/>
>>> >
>>> >
>>> > then restart and enable sip trace and debug level
>>> >
>>> > //do this for every profile involved in the call.
>>> > sofia profile <profilename> siptrace on
>>> >
>>> > //also do this
>>> > console loglevel debug
>>> >
>>> >
>>> > if you can let us ssh, we can do all the for you if you can make the
>>> > test
>>> > calls.
>>> >
>>> >
>>> >
>>> >
>>> > On Sun, Feb 14, 2010 at 11:59 PM, Gabriel Kuri <gkuri at ieee.org> wrote:
>>> >>
>>> >> BTW, here's a copy of the NOTIFY (event call-info) sent to the other
>>> >> phone after the first phone is answered, should this have a Call-Info
>>> >> line with an "appearance-state=seized" to turn on the light on the
>>> >> other phone?
>>> >>
>>> >>
>>> >> NOTIFY sip:2551@<removed>:54446 SIP/2.0.
>>> >> Via: SIP/2.0/UDP <removed>:9430;rport;branch=z9hG4bK71pN2cXgH851K.
>>> >> Max-Forwards: 70.
>>> >> From: <sip:2551 at mydomain.com>;tag=XeB6ZrKDevpHp.
>>> >> To: <sip:2551 at mydomain.com>;tag=c2d34993aac6ea.
>>> >> Call-ID: 34c34987-8b6fa786@<removed>.
>>> >> CSeq: 126950830 NOTIFY.
>>> >> Contact: <sip:2551@<removed>:9430>.
>>> >> Expires: 3959.
>>> >> Call-Info:
>>> >> <sip:mydomain.com>;appearance-index=*;appearance-state=idle.
>>> >> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16633M.
>>> >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>>> >> REGISTER, REFER, PRACK, NOTIFY, PUBLISH, SUBSCRIBE.
>>> >> Supported: 100rel, timer, precondition, path, replaces.
>>> >> Event: call-info.
>>> >> Allow-Events: talk, presence, dialog, line-seize, call-info, sla,
>>> >> include-session-description, presence.winfo, message-summary, refer.
>>> >> Subscription-State: active;expires=3959.
>>> >> Content-Length: 0.
>>> >>
>>> >>
>>> >>
>>> >> On Sun, Feb 14, 2010 at 7:58 PM, Brian West <brian at freeswitch.org>
>>> >> wrote:
>>> >> > Works fine here... is your box slow or something?
>>> >> >
>>> >> > /b
>>> >> >
>>> >> > On Feb 14, 2010, at 1:03 PM, Gabriel Kuri wrote:
>>> >> >
>>> >> >> I followed Brian's directions from one of the previous threads on
>>> >> >> configuring the SPA-5xx series phones for Broadsoft SCA and set
>>> >> >> manage-shared-appearance=true in the internal profile. SCA appears
>>> >> >> to
>>> >> >> be working on outgoing calls between two phones, the line key
>>> >> >> starts
>>> >> >> flashing red on the second phone when the first phone picks up the
>>> >> >> receiver to make a call. However on incoming calls, both phones
>>> >> >> ring
>>> >> >> (same extension), however when one of the phones picks up the line,
>>> >> >> the second phone's line key doesn't flash red or show the first
>>> >> >> phone
>>> >> >> on that incoming call. Any ideas? Does shared appearance only work
>>> >> >> on
>>> >> >> outgoing phone calls?
>>> >> >>
>>> >> >> Thanks,
>>> >> >> Gabe
>>> >> >
>>> >> >
>>> >> > _______________________________________________
>>> >> > FreeSWITCH-users mailing list
>>> >> > FreeSWITCH-users at lists.freeswitch.org
>>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> >
>>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> > http://www.freeswitch.org
>>> >> >
>>> >>
>>> >> _______________________________________________
>>> >> FreeSWITCH-users mailing list
>>> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>
>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> http://www.freeswitch.org
>>> >
>>> >
>>> >
>>> > --
>>> > Anthony Minessale II
>>> >
>>> > FreeSWITCH http://www.freeswitch.org/
>>> > ClueCon http://www.cluecon.com/
>>> > Twitter: http://twitter.com/FreeSWITCH_wire
>>> >
>>> > AIM: anthm
>>> > MSN:anthony_minessale at hotmail.com
>>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>> > IRC: irc.freenode.net #freeswitch
>>> >
>>> > FreeSWITCH Developer Conference
>>> > sip:888 at conference.freeswitch.org
>>> > iax:guest at conference.freeswitch.org/888
>>> > googletalk:conf+888 at conference.freeswitch.org
>>> > pstn:+19193869900
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >
>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>> >
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From tculjaga at gmail.com  Mon Feb 15 12:02:15 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Mon, 15 Feb 2010 21:02:15 +0100
Subject: [Freeswitch-users] h323 - sip call is not working
In-Reply-To: <20100215161410.BF0D911F5E@mail.nstel.ru>
References: <65d96fc81002150739l242407d2u2d5cb5081f96e7eb@mail.gmail.com>
	<20100215161410.BF0D911F5E@mail.nstel.ru>
Message-ID: <65d96fc81002151202r4fb18b6dma3cfb34fc98adc23@mail.gmail.com>

ok, than try this:


edit h323plus/src/h323caps.cxx, grep it for "H323AudioCapability(240, 30) //
240ms max, 30ms desired" ... it should be at line 2598....


replace 30 with 20, recompile (make && make install) make sure you use the
new compiled library and start FS.

let me know if you still have audio issues.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100215/1dc5a80e/attachment.html 

From brian at freeswitch.org  Mon Feb 15 12:28:55 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 15 Feb 2010 14:28:55 -0600
Subject: [Freeswitch-users] Broadsoft SCA w/ Cisco SPA-5xx series
In-Reply-To: <8b1c9cda1002151147l1e072d00y69c0022e3924c3b4@mail.gmail.com>
References: <8b1c9cda1002141103te523643ob16db25e3bfa877f@mail.gmail.com>
	<0D67132C-69D8-43AD-B7E6-C7545B544322@freeswitch.org>
	<8b1c9cda1002142159h5efcc765q9a41baa1bf694c8c@mail.gmail.com>
	<191c3a031002150832x7faa9a78xbc160cde63f09d09@mail.gmail.com>
	<8b1c9cda1002151048o4b45c034qb22732a9366024ae@mail.gmail.com>
	<191c3a031002151055v1938ab91p89232341f4087c8e@mail.gmail.com>
	<191c3a031002151104u7413b17dh84e68ab1cf703cb9@mail.gmail.com>
	<8b1c9cda1002151147l1e072d00y69c0022e3924c3b4@mail.gmail.com>
Message-ID: <7C6C0110-C6AE-45CA-A1B4-413E39223286@freeswitch.org>

Can you outline the topology a bit better...  I sense you have FS behind nat... devices behind nat and ext-rtp-ip and ext-sip-ip set.

Can you confirm this?

/b

On Feb 15, 2010, at 1:47 PM, Gabriel Kuri wrote:

> OK, I don't know what happened there, here's another one with the
> NOTIFYs. I'm on trunk rev 16633 and I have
> "managed-shared-appeareance=true" on the internal profile. I'm just
> making calls between internal phones.
> 
>     http://pastebin.freeswitch.org/12153
> 
> Thanks,
> Gabe

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100215/60cf05e1/attachment.html 

From peder at networkoblivion.com  Mon Feb 15 12:34:12 2010
From: peder at networkoblivion.com (Peder)
Date: Mon, 15 Feb 2010 14:34:12 -0600
Subject: [Freeswitch-users] Broadsoft SCA w/ Cisco SPA-5xx series
In-Reply-To: <8b1c9cda1002151147l1e072d00y69c0022e3924c3b4@mail.gmail.com>
References: <8b1c9cda1002141103te523643ob16db25e3bfa877f@mail.gmail.com>	<0D67132C-69D8-43AD-B7E6-C7545B544322@freeswitch.org>	<8b1c9cda1002142159h5efcc765q9a41baa1bf694c8c@mail.gmail.com>	<191c3a031002150832x7faa9a78xbc160cde63f09d09@mail.gmail.com>	<8b1c9cda1002151048o4b45c034qb22732a9366024ae@mail.gmail.com>	<191c3a031002151055v1938ab91p89232341f4087c8e@mail.gmail.com>	<191c3a031002151104u7413b17dh84e68ab1cf703cb9@mail.gmail.com>
	<8b1c9cda1002151147l1e072d00y69c0022e3924c3b4@mail.gmail.com>
Message-ID: <0dd701caae7e$3c108b70$b431a250$@com>

Is this a typo "managed-shared-appeareance=true" or is there an extra e in
appearance in your config?

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Gabriel
Kuri
Sent: Monday, February 15, 2010 1:48 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Broadsoft SCA w/ Cisco SPA-5xx series

OK, I don't know what happened there, here's another one with the
NOTIFYs. I'm on trunk rev 16633 and I have
"managed-shared-appeareance=true" on the internal profile. I'm just
making calls between internal phones.

     http://pastebin.freeswitch.org/12153

Thanks,
Gabe

On Mon, Feb 15, 2010 at 11:04 AM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> I don't see any notifies at all in this trace do the profiles in question
> have:
> manage-shared-appearance set to true?
> and are you on latest trunk?
>
>
> On Mon, Feb 15, 2010 at 12:55 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>>
>> we log the sql stmts on err so they are red and easier to read.
>>
>>
>>
>> On Mon, Feb 15, 2010 at 12:48 PM, Gabriel Kuri <gkuri at ieee.org> wrote:
>>>
>>> Here's a call from ext 2551 to 2552 (two phones). I noticed a bunch of
>>> errors related to SQL UPDATE for presence ...
>>>
>>> ? ? http://pastebin.freeswitch.org/12152
>>>
>>> Thanks,
>>> Gabe
>>>
>>>
>>> On Mon, Feb 15, 2010 at 8:32 AM, Anthony Minessale
>>> <anthony.minessale at gmail.com> wrote:
>>> > it should be active not seized.
>>> > seized is when you take it off hook.
>>> >
>>> > We need some more debugging to be sure.
>>> > Can we work in real time on it or can you get a more detailed log?
>>> >
>>> > edit sofia.conf.xml and add the param to the "settings" section.
>>> >
>>> > <param name="debug-sla" value="100"/>
>>> >
>>> >
>>> > then restart and enable sip trace and debug level
>>> >
>>> > //do this for every profile involved in the call.
>>> > sofia profile <profilename> siptrace on
>>> >
>>> > //also do this
>>> > console loglevel debug
>>> >
>>> >
>>> > if you can let us ssh, we can do all the for you if you can make the
>>> > test
>>> > calls.
>>> >
>>> >
>>> >
>>> >
>>> > On Sun, Feb 14, 2010 at 11:59 PM, Gabriel Kuri <gkuri at ieee.org> wrote:
>>> >>
>>> >> BTW, here's a copy of the NOTIFY (event call-info) sent to the other
>>> >> phone after the first phone is answered, should this have a Call-Info
>>> >> line with an "appearance-state=seized" to turn on the light on the
>>> >> other phone?
>>> >>
>>> >>
>>> >> NOTIFY sip:2551@<removed>:54446 SIP/2.0.
>>> >> Via: SIP/2.0/UDP <removed>:9430;rport;branch=z9hG4bK71pN2cXgH851K.
>>> >> Max-Forwards: 70.
>>> >> From: <sip:2551 at mydomain.com>;tag=XeB6ZrKDevpHp.
>>> >> To: <sip:2551 at mydomain.com>;tag=c2d34993aac6ea.
>>> >> Call-ID: 34c34987-8b6fa786@<removed>.
>>> >> CSeq: 126950830 NOTIFY.
>>> >> Contact: <sip:2551@<removed>:9430>.
>>> >> Expires: 3959.
>>> >> Call-Info:
>>> >> <sip:mydomain.com>;appearance-index=*;appearance-state=idle.
>>> >> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16633M.
>>> >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>>> >> REGISTER, REFER, PRACK, NOTIFY, PUBLISH, SUBSCRIBE.
>>> >> Supported: 100rel, timer, precondition, path, replaces.
>>> >> Event: call-info.
>>> >> Allow-Events: talk, presence, dialog, line-seize, call-info, sla,
>>> >> include-session-description, presence.winfo, message-summary, refer.
>>> >> Subscription-State: active;expires=3959.
>>> >> Content-Length: 0.
>>> >>
>>> >>
>>> >>
>>> >> On Sun, Feb 14, 2010 at 7:58 PM, Brian West <brian at freeswitch.org>
>>> >> wrote:
>>> >> > Works fine here... is your box slow or something?
>>> >> >
>>> >> > /b
>>> >> >
>>> >> > On Feb 14, 2010, at 1:03 PM, Gabriel Kuri wrote:
>>> >> >
>>> >> >> I followed Brian's directions from one of the previous threads on
>>> >> >> configuring the SPA-5xx series phones for Broadsoft SCA and set
>>> >> >> manage-shared-appearance=true in the internal profile. SCA appears
>>> >> >> to
>>> >> >> be working on outgoing calls between two phones, the line key
>>> >> >> starts
>>> >> >> flashing red on the second phone when the first phone picks up the
>>> >> >> receiver to make a call. However on incoming calls, both phones
>>> >> >> ring
>>> >> >> (same extension), however when one of the phones picks up the
line,
>>> >> >> the second phone's line key doesn't flash red or show the first
>>> >> >> phone
>>> >> >> on that incoming call. Any ideas? Does shared appearance only work
>>> >> >> on
>>> >> >> outgoing phone calls?
>>> >> >>
>>> >> >> Thanks,
>>> >> >> Gabe
>>> >> >
>>> >> >
>>> >> > _______________________________________________
>>> >> > FreeSWITCH-users mailing list
>>> >> > FreeSWITCH-users at lists.freeswitch.org
>>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> >
>>> >> >
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> > http://www.freeswitch.org
>>> >> >
>>> >>
>>> >> _______________________________________________
>>> >> FreeSWITCH-users mailing list
>>> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>
>>> >>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> http://www.freeswitch.org
>>> >
>>> >
>>> >
>>> > --
>>> > Anthony Minessale II
>>> >
>>> > FreeSWITCH http://www.freeswitch.org/
>>> > ClueCon http://www.cluecon.com/
>>> > Twitter: http://twitter.com/FreeSWITCH_wire
>>> >
>>> > AIM: anthm
>>> > MSN:anthony_minessale at hotmail.com
>>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>> > IRC: irc.freenode.net #freeswitch
>>> >
>>> > FreeSWITCH Developer Conference
>>> > sip:888 at conference.freeswitch.org
>>> > iax:guest at conference.freeswitch.org/888
>>> > googletalk:conf+888 at conference.freeswitch.org
>>> > pstn:+19193869900
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >
>>> >
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>> >
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



From msc at freeswitch.org  Mon Feb 15 12:45:45 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 15 Feb 2010 12:45:45 -0800
Subject: [Freeswitch-users] Broadsoft SCA w/ Cisco SPA-5xx series
In-Reply-To: <0dd701caae7e$3c108b70$b431a250$@com>
References: <8b1c9cda1002141103te523643ob16db25e3bfa877f@mail.gmail.com>
	<0D67132C-69D8-43AD-B7E6-C7545B544322@freeswitch.org>
	<8b1c9cda1002142159h5efcc765q9a41baa1bf694c8c@mail.gmail.com>
	<191c3a031002150832x7faa9a78xbc160cde63f09d09@mail.gmail.com>
	<8b1c9cda1002151048o4b45c034qb22732a9366024ae@mail.gmail.com>
	<191c3a031002151055v1938ab91p89232341f4087c8e@mail.gmail.com>
	<191c3a031002151104u7413b17dh84e68ab1cf703cb9@mail.gmail.com>
	<8b1c9cda1002151147l1e072d00y69c0022e3924c3b4@mail.gmail.com>
	<0dd701caae7e$3c108b70$b431a250$@com>
Message-ID: <87f2f3b91002151245s3987beecy41853d95d522ceb0@mail.gmail.com>

Yes, it is a typo from the original article I wrote. Also "appearance" is
misspelled. The correct line is:
<param name="manage-shared-appearance" value="true"/>

BTW, in internal.xml this line is commented out so if you need to copy &
paste just look in there...
-MC

On Mon, Feb 15, 2010 at 12:34 PM, Peder <peder at networkoblivion.com> wrote:

> Is this a typo "managed-shared-appeareance=true" or is there an extra e in
> appearance in your config?
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Gabriel
> Kuri
> Sent: Monday, February 15, 2010 1:48 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Broadsoft SCA w/ Cisco SPA-5xx series
>
> OK, I don't know what happened there, here's another one with the
> NOTIFYs. I'm on trunk rev 16633 and I have
> "managed-shared-appeareance=true" on the internal profile. I'm just
> making calls between internal phones.
>
>     http://pastebin.freeswitch.org/12153
>
> Thanks,
> Gabe
>
> On Mon, Feb 15, 2010 at 11:04 AM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
> > I don't see any notifies at all in this trace do the profiles in question
> > have:
> > manage-shared-appearance set to true?
> > and are you on latest trunk?
> >
> >
> > On Mon, Feb 15, 2010 at 12:55 PM, Anthony Minessale
> > <anthony.minessale at gmail.com> wrote:
> >>
> >> we log the sql stmts on err so they are red and easier to read.
> >>
> >>
> >>
> >> On Mon, Feb 15, 2010 at 12:48 PM, Gabriel Kuri <gkuri at ieee.org> wrote:
> >>>
> >>> Here's a call from ext 2551 to 2552 (two phones). I noticed a bunch of
> >>> errors related to SQL UPDATE for presence ...
> >>>
> >>>     http://pastebin.freeswitch.org/12152
> >>>
> >>> Thanks,
> >>> Gabe
> >>>
> >>>
> >>> On Mon, Feb 15, 2010 at 8:32 AM, Anthony Minessale
> >>> <anthony.minessale at gmail.com> wrote:
> >>> > it should be active not seized.
> >>> > seized is when you take it off hook.
> >>> >
> >>> > We need some more debugging to be sure.
> >>> > Can we work in real time on it or can you get a more detailed log?
> >>> >
> >>> > edit sofia.conf.xml and add the param to the "settings" section.
> >>> >
> >>> > <param name="debug-sla" value="100"/>
> >>> >
> >>> >
> >>> > then restart and enable sip trace and debug level
> >>> >
> >>> > //do this for every profile involved in the call.
> >>> > sofia profile <profilename> siptrace on
> >>> >
> >>> > //also do this
> >>> > console loglevel debug
> >>> >
> >>> >
> >>> > if you can let us ssh, we can do all the for you if you can make the
> >>> > test
> >>> > calls.
> >>> >
> >>> >
> >>> >
> >>> >
> >>> > On Sun, Feb 14, 2010 at 11:59 PM, Gabriel Kuri <gkuri at ieee.org>
> wrote:
> >>> >>
> >>> >> BTW, here's a copy of the NOTIFY (event call-info) sent to the other
> >>> >> phone after the first phone is answered, should this have a
> Call-Info
> >>> >> line with an "appearance-state=seized" to turn on the light on the
> >>> >> other phone?
> >>> >>
> >>> >>
> >>> >> NOTIFY sip:2551@<removed>:54446 SIP/2.0.
> >>> >> Via: SIP/2.0/UDP <removed>:9430;rport;branch=z9hG4bK71pN2cXgH851K.
> >>> >> Max-Forwards: 70.
> >>> >> From: <sip:2551 at mydomain.com <sip%3A2551 at mydomain.com>
> >;tag=XeB6ZrKDevpHp.
> >>> >> To: <sip:2551 at mydomain.com <sip%3A2551 at mydomain.com>
> >;tag=c2d34993aac6ea.
> >>> >> Call-ID: 34c34987-8b6fa786@<removed>.
> >>> >> CSeq: 126950830 NOTIFY.
> >>> >> Contact: <sip:2551@<removed>:9430>.
> >>> >> Expires: 3959.
> >>> >> Call-Info:
> >>> >> <sip:mydomain.com>;appearance-index=*;appearance-state=idle.
> >>> >> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16633M.
> >>> >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> >>> >> REGISTER, REFER, PRACK, NOTIFY, PUBLISH, SUBSCRIBE.
> >>> >> Supported: 100rel, timer, precondition, path, replaces.
> >>> >> Event: call-info.
> >>> >> Allow-Events: talk, presence, dialog, line-seize, call-info, sla,
> >>> >> include-session-description, presence.winfo, message-summary, refer.
> >>> >> Subscription-State: active;expires=3959.
> >>> >> Content-Length: 0.
> >>> >>
> >>> >>
> >>> >>
> >>> >> On Sun, Feb 14, 2010 at 7:58 PM, Brian West <brian at freeswitch.org>
> >>> >> wrote:
> >>> >> > Works fine here... is your box slow or something?
> >>> >> >
> >>> >> > /b
> >>> >> >
> >>> >> > On Feb 14, 2010, at 1:03 PM, Gabriel Kuri wrote:
> >>> >> >
> >>> >> >> I followed Brian's directions from one of the previous threads on
> >>> >> >> configuring the SPA-5xx series phones for Broadsoft SCA and set
> >>> >> >> manage-shared-appearance=true in the internal profile. SCA
> appears
> >>> >> >> to
> >>> >> >> be working on outgoing calls between two phones, the line key
> >>> >> >> starts
> >>> >> >> flashing red on the second phone when the first phone picks up
> the
> >>> >> >> receiver to make a call. However on incoming calls, both phones
> >>> >> >> ring
> >>> >> >> (same extension), however when one of the phones picks up the
> line,
> >>> >> >> the second phone's line key doesn't flash red or show the first
> >>> >> >> phone
> >>> >> >> on that incoming call. Any ideas? Does shared appearance only
> work
> >>> >> >> on
> >>> >> >> outgoing phone calls?
> >>> >> >>
> >>> >> >> Thanks,
> >>> >> >> Gabe
> >>> >> >
> >>> >> >
> >>> >> > _______________________________________________
> >>> >> > FreeSWITCH-users mailing list
> >>> >> > FreeSWITCH-users at lists.freeswitch.org
> >>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> >> >
> >>> >> >
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> >> > http://www.freeswitch.org
> >>> >> >
> >>> >>
> >>> >> _______________________________________________
> >>> >> FreeSWITCH-users mailing list
> >>> >> FreeSWITCH-users at lists.freeswitch.org
> >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> >>
> >>> >>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> >> http://www.freeswitch.org
> >>> >
> >>> >
> >>> >
> >>> > --
> >>> > Anthony Minessale II
> >>> >
> >>> > FreeSWITCH http://www.freeswitch.org/
> >>> > ClueCon http://www.cluecon.com/
> >>> > Twitter: http://twitter.com/FreeSWITCH_wire
> >>> >
> >>> > AIM: anthm
> >>> > MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
> >>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >>> > IRC: irc.freenode.net #freeswitch
> >>> >
> >>> > FreeSWITCH Developer Conference
> >>> > sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> >>> > iax:guest at conference.freeswitch.org/888
> >>> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >>> > pstn:+19193869900
> >>> >
> >>> > _______________________________________________
> >>> > FreeSWITCH-users mailing list
> >>> > FreeSWITCH-users at lists.freeswitch.org
> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> >
> >>> >
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> > http://www.freeswitch.org
> >>> >
> >>> >
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>
> >>
> >>
> >> --
> >> Anthony Minessale II
> >>
> >> FreeSWITCH http://www.freeswitch.org/
> >> ClueCon http://www.cluecon.com/
> >> Twitter: http://twitter.com/FreeSWITCH_wire
> >>
> >> AIM: anthm
> >> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >> IRC: irc.freenode.net #freeswitch
> >>
> >> FreeSWITCH Developer Conference
> >> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> >> iax:guest at conference.freeswitch.org/888
> >> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >> pstn:+19193869900
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> > Twitter: http://twitter.com/FreeSWITCH_wire
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> > IRC: irc.freenode.net #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> > iax:guest at conference.freeswitch.org/888
> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > pstn:+19193869900
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100215/391884c7/attachment-0001.html 

From jerry.richards at teotech.com  Mon Feb 15 12:49:38 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Mon, 15 Feb 2010 12:49:38 -0800
Subject: [Freeswitch-users] external_sip_address and external_rtp_address
	Question
Message-ID: <CDD4A2D8AA494ED8BF64401C5C953F72@greyhawk.tonecommander.com>

I only see one example for setting of external_sip_address and
external_rtp_address tags.  Is it true they are used to specify a SIP
provider outside of a LAN (i.e. through a router)?  If so, then can these
tags be set for each sip_profile?  So, if I have multiple external SIP
providers that are accessed through NAT, they would each have their own
external_sip_address and external_rtp_address?

Best Regards,
Jerry




From gkuri at ieee.org  Mon Feb 15 12:52:58 2010
From: gkuri at ieee.org (Gabriel Kuri)
Date: Mon, 15 Feb 2010 12:52:58 -0800
Subject: [Freeswitch-users] Broadsoft SCA w/ Cisco SPA-5xx series
In-Reply-To: <7C6C0110-C6AE-45CA-A1B4-413E39223286@freeswitch.org>
References: <8b1c9cda1002141103te523643ob16db25e3bfa877f@mail.gmail.com>
	<0D67132C-69D8-43AD-B7E6-C7545B544322@freeswitch.org>
	<8b1c9cda1002142159h5efcc765q9a41baa1bf694c8c@mail.gmail.com>
	<191c3a031002150832x7faa9a78xbc160cde63f09d09@mail.gmail.com>
	<8b1c9cda1002151048o4b45c034qb22732a9366024ae@mail.gmail.com>
	<191c3a031002151055v1938ab91p89232341f4087c8e@mail.gmail.com>
	<191c3a031002151104u7413b17dh84e68ab1cf703cb9@mail.gmail.com>
	<8b1c9cda1002151147l1e072d00y69c0022e3924c3b4@mail.gmail.com>
	<7C6C0110-C6AE-45CA-A1B4-413E39223286@freeswitch.org>
Message-ID: <8b1c9cda1002151252jbc500bax3f137a882171485f@mail.gmail.com>

Yes, that is correct. FS is behind a NAT and the phones behind another NAT.

I have ext-rtp-ip and ext-sip-ip set to the public IP address. Phones
calls and everything else seem to be working.

Thanks,
Gabe

On Mon, Feb 15, 2010 at 12:28 PM, Brian West <brian at freeswitch.org> wrote:
> Can you outline the topology a bit better... ?I sense you have FS behind
> nat... devices behind nat and ext-rtp-ip and ext-sip-ip set.
> Can you confirm this?
> /b
> On Feb 15, 2010, at 1:47 PM, Gabriel Kuri wrote:
>
> OK, I don't know what happened there, here's another one with the
> NOTIFYs. I'm on trunk rev 16633 and I have
> "managed-shared-appeareance=true" on the internal profile. I'm just
> making calls between internal phones.
>
> ????http://pastebin.freeswitch.org/12153
>
> Thanks,
> Gabe
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From gkuri at ieee.org  Mon Feb 15 12:53:40 2010
From: gkuri at ieee.org (Gabriel Kuri)
Date: Mon, 15 Feb 2010 12:53:40 -0800
Subject: [Freeswitch-users] Broadsoft SCA w/ Cisco SPA-5xx series
In-Reply-To: <0dd701caae7e$3c108b70$b431a250$@com>
References: <8b1c9cda1002141103te523643ob16db25e3bfa877f@mail.gmail.com>
	<0D67132C-69D8-43AD-B7E6-C7545B544322@freeswitch.org>
	<8b1c9cda1002142159h5efcc765q9a41baa1bf694c8c@mail.gmail.com>
	<191c3a031002150832x7faa9a78xbc160cde63f09d09@mail.gmail.com>
	<8b1c9cda1002151048o4b45c034qb22732a9366024ae@mail.gmail.com>
	<191c3a031002151055v1938ab91p89232341f4087c8e@mail.gmail.com>
	<191c3a031002151104u7413b17dh84e68ab1cf703cb9@mail.gmail.com>
	<8b1c9cda1002151147l1e072d00y69c0022e3924c3b4@mail.gmail.com>
	<0dd701caae7e$3c108b70$b431a250$@com>
Message-ID: <8b1c9cda1002151253t32c22663wf66245e0220a4e6@mail.gmail.com>

No, that was a typo. I have it correct in the config file.

Gabe

On Mon, Feb 15, 2010 at 12:34 PM, Peder <peder at networkoblivion.com> wrote:
> Is this a typo "managed-shared-appeareance=true" or is there an extra e in
> appearance in your config?
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Gabriel
> Kuri
> Sent: Monday, February 15, 2010 1:48 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Broadsoft SCA w/ Cisco SPA-5xx series
>
> OK, I don't know what happened there, here's another one with the
> NOTIFYs. I'm on trunk rev 16633 and I have
> "managed-shared-appeareance=true" on the internal profile. I'm just
> making calls between internal phones.
>
> ? ? http://pastebin.freeswitch.org/12153
>
> Thanks,
> Gabe
>
> On Mon, Feb 15, 2010 at 11:04 AM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>> I don't see any notifies at all in this trace do the profiles in question
>> have:
>> manage-shared-appearance set to true?
>> and are you on latest trunk?
>>
>>
>> On Mon, Feb 15, 2010 at 12:55 PM, Anthony Minessale
>> <anthony.minessale at gmail.com> wrote:
>>>
>>> we log the sql stmts on err so they are red and easier to read.
>>>
>>>
>>>
>>> On Mon, Feb 15, 2010 at 12:48 PM, Gabriel Kuri <gkuri at ieee.org> wrote:
>>>>
>>>> Here's a call from ext 2551 to 2552 (two phones). I noticed a bunch of
>>>> errors related to SQL UPDATE for presence ...
>>>>
>>>> ? ? http://pastebin.freeswitch.org/12152
>>>>
>>>> Thanks,
>>>> Gabe
>>>>
>>>>
>>>> On Mon, Feb 15, 2010 at 8:32 AM, Anthony Minessale
>>>> <anthony.minessale at gmail.com> wrote:
>>>> > it should be active not seized.
>>>> > seized is when you take it off hook.
>>>> >
>>>> > We need some more debugging to be sure.
>>>> > Can we work in real time on it or can you get a more detailed log?
>>>> >
>>>> > edit sofia.conf.xml and add the param to the "settings" section.
>>>> >
>>>> > <param name="debug-sla" value="100"/>
>>>> >
>>>> >
>>>> > then restart and enable sip trace and debug level
>>>> >
>>>> > //do this for every profile involved in the call.
>>>> > sofia profile <profilename> siptrace on
>>>> >
>>>> > //also do this
>>>> > console loglevel debug
>>>> >
>>>> >
>>>> > if you can let us ssh, we can do all the for you if you can make the
>>>> > test
>>>> > calls.
>>>> >
>>>> >
>>>> >
>>>> >
>>>> > On Sun, Feb 14, 2010 at 11:59 PM, Gabriel Kuri <gkuri at ieee.org> wrote:
>>>> >>
>>>> >> BTW, here's a copy of the NOTIFY (event call-info) sent to the other
>>>> >> phone after the first phone is answered, should this have a Call-Info
>>>> >> line with an "appearance-state=seized" to turn on the light on the
>>>> >> other phone?
>>>> >>
>>>> >>
>>>> >> NOTIFY sip:2551@<removed>:54446 SIP/2.0.
>>>> >> Via: SIP/2.0/UDP <removed>:9430;rport;branch=z9hG4bK71pN2cXgH851K.
>>>> >> Max-Forwards: 70.
>>>> >> From: <sip:2551 at mydomain.com>;tag=XeB6ZrKDevpHp.
>>>> >> To: <sip:2551 at mydomain.com>;tag=c2d34993aac6ea.
>>>> >> Call-ID: 34c34987-8b6fa786@<removed>.
>>>> >> CSeq: 126950830 NOTIFY.
>>>> >> Contact: <sip:2551@<removed>:9430>.
>>>> >> Expires: 3959.
>>>> >> Call-Info:
>>>> >> <sip:mydomain.com>;appearance-index=*;appearance-state=idle.
>>>> >> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16633M.
>>>> >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>>>> >> REGISTER, REFER, PRACK, NOTIFY, PUBLISH, SUBSCRIBE.
>>>> >> Supported: 100rel, timer, precondition, path, replaces.
>>>> >> Event: call-info.
>>>> >> Allow-Events: talk, presence, dialog, line-seize, call-info, sla,
>>>> >> include-session-description, presence.winfo, message-summary, refer.
>>>> >> Subscription-State: active;expires=3959.
>>>> >> Content-Length: 0.
>>>> >>
>>>> >>
>>>> >>
>>>> >> On Sun, Feb 14, 2010 at 7:58 PM, Brian West <brian at freeswitch.org>
>>>> >> wrote:
>>>> >> > Works fine here... is your box slow or something?
>>>> >> >
>>>> >> > /b
>>>> >> >
>>>> >> > On Feb 14, 2010, at 1:03 PM, Gabriel Kuri wrote:
>>>> >> >
>>>> >> >> I followed Brian's directions from one of the previous threads on
>>>> >> >> configuring the SPA-5xx series phones for Broadsoft SCA and set
>>>> >> >> manage-shared-appearance=true in the internal profile. SCA appears
>>>> >> >> to
>>>> >> >> be working on outgoing calls between two phones, the line key
>>>> >> >> starts
>>>> >> >> flashing red on the second phone when the first phone picks up the
>>>> >> >> receiver to make a call. However on incoming calls, both phones
>>>> >> >> ring
>>>> >> >> (same extension), however when one of the phones picks up the
> line,
>>>> >> >> the second phone's line key doesn't flash red or show the first
>>>> >> >> phone
>>>> >> >> on that incoming call. Any ideas? Does shared appearance only work
>>>> >> >> on
>>>> >> >> outgoing phone calls?
>>>> >> >>
>>>> >> >> Thanks,
>>>> >> >> Gabe
>>>> >> >
>>>> >> >
>>>> >> > _______________________________________________
>>>> >> > FreeSWITCH-users mailing list
>>>> >> > FreeSWITCH-users at lists.freeswitch.org
>>>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >> >
>>>> >> >
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> >> > http://www.freeswitch.org
>>>> >> >
>>>> >>
>>>> >> _______________________________________________
>>>> >> FreeSWITCH-users mailing list
>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >>
>>>> >>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> >> http://www.freeswitch.org
>>>> >
>>>> >
>>>> >
>>>> > --
>>>> > Anthony Minessale II
>>>> >
>>>> > FreeSWITCH http://www.freeswitch.org/
>>>> > ClueCon http://www.cluecon.com/
>>>> > Twitter: http://twitter.com/FreeSWITCH_wire
>>>> >
>>>> > AIM: anthm
>>>> > MSN:anthony_minessale at hotmail.com
>>>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>>> > IRC: irc.freenode.net #freeswitch
>>>> >
>>>> > FreeSWITCH Developer Conference
>>>> > sip:888 at conference.freeswitch.org
>>>> > iax:guest at conference.freeswitch.org/888
>>>> > googletalk:conf+888 at conference.freeswitch.org
>>>> > pstn:+19193869900
>>>> >
>>>> > _______________________________________________
>>>> > FreeSWITCH-users mailing list
>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >
>>>> >
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> > http://www.freeswitch.org
>>>> >
>>>> >
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org
>>> iax:guest at conference.freeswitch.org/888
>>> googletalk:conf+888 at conference.freeswitch.org
>>> pstn:+19193869900
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From anthony.minessale at gmail.com  Mon Feb 15 13:04:01 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 15 Feb 2010 15:04:01 -0600
Subject: [Freeswitch-users] Broadsoft SCA w/ Cisco SPA-5xx series
In-Reply-To: <8b1c9cda1002151253t32c22663wf66245e0220a4e6@mail.gmail.com>
References: <8b1c9cda1002141103te523643ob16db25e3bfa877f@mail.gmail.com>
	<0D67132C-69D8-43AD-B7E6-C7545B544322@freeswitch.org>
	<8b1c9cda1002142159h5efcc765q9a41baa1bf694c8c@mail.gmail.com>
	<191c3a031002150832x7faa9a78xbc160cde63f09d09@mail.gmail.com>
	<8b1c9cda1002151048o4b45c034qb22732a9366024ae@mail.gmail.com>
	<191c3a031002151055v1938ab91p89232341f4087c8e@mail.gmail.com>
	<191c3a031002151104u7413b17dh84e68ab1cf703cb9@mail.gmail.com>
	<8b1c9cda1002151147l1e072d00y69c0022e3924c3b4@mail.gmail.com>
	<0dd701caae7e$3c108b70$b431a250$@com>
	<8b1c9cda1002151253t32c22663wf66245e0220a4e6@mail.gmail.com>
Message-ID: <191c3a031002151304h54c262c5o8ffd5c195e0512d2@mail.gmail.com>

you are missing something because you have no seize events when you go on
and off hook.
is every phone in the correct mode?


On Mon, Feb 15, 2010 at 2:53 PM, Gabriel Kuri <gkuri at ieee.org> wrote:

> No, that was a typo. I have it correct in the config file.
>
> Gabe
>
> On Mon, Feb 15, 2010 at 12:34 PM, Peder <peder at networkoblivion.com> wrote:
> > Is this a typo "managed-shared-appeareance=true" or is there an extra e
> in
> > appearance in your config?
> >
> > -----Original Message-----
> > From: freeswitch-users-bounces at lists.freeswitch.org
> > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Gabriel
> > Kuri
> > Sent: Monday, February 15, 2010 1:48 PM
> > To: freeswitch-users at lists.freeswitch.org
> > Subject: Re: [Freeswitch-users] Broadsoft SCA w/ Cisco SPA-5xx series
> >
> > OK, I don't know what happened there, here's another one with the
> > NOTIFYs. I'm on trunk rev 16633 and I have
> > "managed-shared-appeareance=true" on the internal profile. I'm just
> > making calls between internal phones.
> >
> >     http://pastebin.freeswitch.org/12153
> >
> > Thanks,
> > Gabe
> >
> > On Mon, Feb 15, 2010 at 11:04 AM, Anthony Minessale
> > <anthony.minessale at gmail.com> wrote:
> >> I don't see any notifies at all in this trace do the profiles in
> question
> >> have:
> >> manage-shared-appearance set to true?
> >> and are you on latest trunk?
> >>
> >>
> >> On Mon, Feb 15, 2010 at 12:55 PM, Anthony Minessale
> >> <anthony.minessale at gmail.com> wrote:
> >>>
> >>> we log the sql stmts on err so they are red and easier to read.
> >>>
> >>>
> >>>
> >>> On Mon, Feb 15, 2010 at 12:48 PM, Gabriel Kuri <gkuri at ieee.org> wrote:
> >>>>
> >>>> Here's a call from ext 2551 to 2552 (two phones). I noticed a bunch of
> >>>> errors related to SQL UPDATE for presence ...
> >>>>
> >>>>     http://pastebin.freeswitch.org/12152
> >>>>
> >>>> Thanks,
> >>>> Gabe
> >>>>
> >>>>
> >>>> On Mon, Feb 15, 2010 at 8:32 AM, Anthony Minessale
> >>>> <anthony.minessale at gmail.com> wrote:
> >>>> > it should be active not seized.
> >>>> > seized is when you take it off hook.
> >>>> >
> >>>> > We need some more debugging to be sure.
> >>>> > Can we work in real time on it or can you get a more detailed log?
> >>>> >
> >>>> > edit sofia.conf.xml and add the param to the "settings" section.
> >>>> >
> >>>> > <param name="debug-sla" value="100"/>
> >>>> >
> >>>> >
> >>>> > then restart and enable sip trace and debug level
> >>>> >
> >>>> > //do this for every profile involved in the call.
> >>>> > sofia profile <profilename> siptrace on
> >>>> >
> >>>> > //also do this
> >>>> > console loglevel debug
> >>>> >
> >>>> >
> >>>> > if you can let us ssh, we can do all the for you if you can make the
> >>>> > test
> >>>> > calls.
> >>>> >
> >>>> >
> >>>> >
> >>>> >
> >>>> > On Sun, Feb 14, 2010 at 11:59 PM, Gabriel Kuri <gkuri at ieee.org>
> wrote:
> >>>> >>
> >>>> >> BTW, here's a copy of the NOTIFY (event call-info) sent to the
> other
> >>>> >> phone after the first phone is answered, should this have a
> Call-Info
> >>>> >> line with an "appearance-state=seized" to turn on the light on the
> >>>> >> other phone?
> >>>> >>
> >>>> >>
> >>>> >> NOTIFY sip:2551@<removed>:54446 SIP/2.0.
> >>>> >> Via: SIP/2.0/UDP <removed>:9430;rport;branch=z9hG4bK71pN2cXgH851K.
> >>>> >> Max-Forwards: 70.
> >>>> >> From: <sip:2551 at mydomain.com <sip%3A2551 at mydomain.com>
> >;tag=XeB6ZrKDevpHp.
> >>>> >> To: <sip:2551 at mydomain.com <sip%3A2551 at mydomain.com>
> >;tag=c2d34993aac6ea.
> >>>> >> Call-ID: 34c34987-8b6fa786@<removed>.
> >>>> >> CSeq: 126950830 NOTIFY.
> >>>> >> Contact: <sip:2551@<removed>:9430>.
> >>>> >> Expires: 3959.
> >>>> >> Call-Info:
> >>>> >> <sip:mydomain.com>;appearance-index=*;appearance-state=idle.
> >>>> >> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16633M.
> >>>> >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> >>>> >> REGISTER, REFER, PRACK, NOTIFY, PUBLISH, SUBSCRIBE.
> >>>> >> Supported: 100rel, timer, precondition, path, replaces.
> >>>> >> Event: call-info.
> >>>> >> Allow-Events: talk, presence, dialog, line-seize, call-info, sla,
> >>>> >> include-session-description, presence.winfo, message-summary,
> refer.
> >>>> >> Subscription-State: active;expires=3959.
> >>>> >> Content-Length: 0.
> >>>> >>
> >>>> >>
> >>>> >>
> >>>> >> On Sun, Feb 14, 2010 at 7:58 PM, Brian West <brian at freeswitch.org>
> >>>> >> wrote:
> >>>> >> > Works fine here... is your box slow or something?
> >>>> >> >
> >>>> >> > /b
> >>>> >> >
> >>>> >> > On Feb 14, 2010, at 1:03 PM, Gabriel Kuri wrote:
> >>>> >> >
> >>>> >> >> I followed Brian's directions from one of the previous threads
> on
> >>>> >> >> configuring the SPA-5xx series phones for Broadsoft SCA and set
> >>>> >> >> manage-shared-appearance=true in the internal profile. SCA
> appears
> >>>> >> >> to
> >>>> >> >> be working on outgoing calls between two phones, the line key
> >>>> >> >> starts
> >>>> >> >> flashing red on the second phone when the first phone picks up
> the
> >>>> >> >> receiver to make a call. However on incoming calls, both phones
> >>>> >> >> ring
> >>>> >> >> (same extension), however when one of the phones picks up the
> > line,
> >>>> >> >> the second phone's line key doesn't flash red or show the first
> >>>> >> >> phone
> >>>> >> >> on that incoming call. Any ideas? Does shared appearance only
> work
> >>>> >> >> on
> >>>> >> >> outgoing phone calls?
> >>>> >> >>
> >>>> >> >> Thanks,
> >>>> >> >> Gabe
> >>>> >> >
> >>>> >> >
> >>>> >> > _______________________________________________
> >>>> >> > FreeSWITCH-users mailing list
> >>>> >> > FreeSWITCH-users at lists.freeswitch.org
> >>>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>> >> >
> >>>> >> >
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>> >> > http://www.freeswitch.org
> >>>> >> >
> >>>> >>
> >>>> >> _______________________________________________
> >>>> >> FreeSWITCH-users mailing list
> >>>> >> FreeSWITCH-users at lists.freeswitch.org
> >>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>> >>
> >>>> >>
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>> >> http://www.freeswitch.org
> >>>> >
> >>>> >
> >>>> >
> >>>> > --
> >>>> > Anthony Minessale II
> >>>> >
> >>>> > FreeSWITCH http://www.freeswitch.org/
> >>>> > ClueCon http://www.cluecon.com/
> >>>> > Twitter: http://twitter.com/FreeSWITCH_wire
> >>>> >
> >>>> > AIM: anthm
> >>>> > MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
> >>>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >>>> > IRC: irc.freenode.net #freeswitch
> >>>> >
> >>>> > FreeSWITCH Developer Conference
> >>>> > sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> >>>> > iax:guest at conference.freeswitch.org/888
> >>>> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >>>> > pstn:+19193869900
> >>>> >
> >>>> > _______________________________________________
> >>>> > FreeSWITCH-users mailing list
> >>>> > FreeSWITCH-users at lists.freeswitch.org
> >>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>> >
> >>>> >
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>> > http://www.freeswitch.org
> >>>> >
> >>>> >
> >>>>
> >>>> _______________________________________________
> >>>> FreeSWITCH-users mailing list
> >>>> FreeSWITCH-users at lists.freeswitch.org
> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>> http://www.freeswitch.org
> >>>
> >>>
> >>>
> >>> --
> >>> Anthony Minessale II
> >>>
> >>> FreeSWITCH http://www.freeswitch.org/
> >>> ClueCon http://www.cluecon.com/
> >>> Twitter: http://twitter.com/FreeSWITCH_wire
> >>>
> >>> AIM: anthm
> >>> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
> >>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >>> IRC: irc.freenode.net #freeswitch
> >>>
> >>> FreeSWITCH Developer Conference
> >>> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> >>> iax:guest at conference.freeswitch.org/888
> >>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >>> pstn:+19193869900
> >>
> >>
> >>
> >> --
> >> Anthony Minessale II
> >>
> >> FreeSWITCH http://www.freeswitch.org/
> >> ClueCon http://www.cluecon.com/
> >> Twitter: http://twitter.com/FreeSWITCH_wire
> >>
> >> AIM: anthm
> >> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >> IRC: irc.freenode.net #freeswitch
> >>
> >> FreeSWITCH Developer Conference
> >> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> >> iax:guest at conference.freeswitch.org/888
> >> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >> pstn:+19193869900
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100215/f04cb603/attachment-0001.html 

From peder at networkoblivion.com  Mon Feb 15 13:17:57 2010
From: peder at networkoblivion.com (Peder)
Date: Mon, 15 Feb 2010 15:17:57 -0600
Subject: [Freeswitch-users] Broadsoft SCA w/ Cisco SPA-5xx series
In-Reply-To: <191c3a031002151304h54c262c5o8ffd5c195e0512d2@mail.gmail.com>
References: <8b1c9cda1002141103te523643ob16db25e3bfa877f@mail.gmail.com>	<0D67132C-69D8-43AD-B7E6-C7545B544322@freeswitch.org>	<8b1c9cda1002142159h5efcc765q9a41baa1bf694c8c@mail.gmail.com>	<191c3a031002150832x7faa9a78xbc160cde63f09d09@mail.gmail.com>	<8b1c9cda1002151048o4b45c034qb22732a9366024ae@mail.gmail.com>	<191c3a031002151055v1938ab91p89232341f4087c8e@mail.gmail.com>	<191c3a031002151104u7413b17dh84e68ab1cf703cb9@mail.gmail.com>	<8b1c9cda1002151147l1e072d00y69c0022e3924c3b4@mail.gmail.com>	<0dd701caae7e$3c108b70$b431a250$@com>	<8b1c9cda1002151253t32c22663wf66245e0220a4e6@mail.gmail.com>
	<191c3a031002151304h54c262c5o8ffd5c195e0512d2@mail.gmail.com>
Message-ID: <0e7301caae84$58b47ce0$0a1d76a0$@com>

On the phone itself, do you have the line set to shared and "Broadsoft SCA"
enabled?

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Monday, February 15, 2010 3:04 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Broadsoft SCA w/ Cisco SPA-5xx series

 

you are missing something because you have no seize events when you go on
and off hook.
is every phone in the correct mode?



On Mon, Feb 15, 2010 at 2:53 PM, Gabriel Kuri <gkuri at ieee.org> wrote:

No, that was a typo. I have it correct in the config file.

Gabe


On Mon, Feb 15, 2010 at 12:34 PM, Peder <peder at networkoblivion.com> wrote:

> Is this a typo "managed-shared-appeareance=true" or is there an extra e in
> appearance in your config?
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Gabriel
> Kuri
> Sent: Monday, February 15, 2010 1:48 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Broadsoft SCA w/ Cisco SPA-5xx series
>
> OK, I don't know what happened there, here's another one with the
> NOTIFYs. I'm on trunk rev 16633 and I have
> "managed-shared-appeareance=true" on the internal profile. I'm just
> making calls between internal phones.
>
>     http://pastebin.freeswitch.org/12153
>
> Thanks,
> Gabe
>
> On Mon, Feb 15, 2010 at 11:04 AM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>> I don't see any notifies at all in this trace do the profiles in question
>> have:
>> manage-shared-appearance set to true?
>> and are you on latest trunk?
>>
>>
>> On Mon, Feb 15, 2010 at 12:55 PM, Anthony Minessale
>> <anthony.minessale at gmail.com> wrote:
>>>
>>> we log the sql stmts on err so they are red and easier to read.
>>>
>>>
>>>
>>> On Mon, Feb 15, 2010 at 12:48 PM, Gabriel Kuri <gkuri at ieee.org> wrote:
>>>>
>>>> Here's a call from ext 2551 to 2552 (two phones). I noticed a bunch of
>>>> errors related to SQL UPDATE for presence ...
>>>>
>>>>     http://pastebin.freeswitch.org/12152
>>>>
>>>> Thanks,
>>>> Gabe
>>>>
>>>>
>>>> On Mon, Feb 15, 2010 at 8:32 AM, Anthony Minessale
>>>> <anthony.minessale at gmail.com> wrote:
>>>> > it should be active not seized.
>>>> > seized is when you take it off hook.
>>>> >
>>>> > We need some more debugging to be sure.
>>>> > Can we work in real time on it or can you get a more detailed log?
>>>> >
>>>> > edit sofia.conf.xml and add the param to the "settings" section.
>>>> >
>>>> > <param name="debug-sla" value="100"/>
>>>> >
>>>> >
>>>> > then restart and enable sip trace and debug level
>>>> >
>>>> > //do this for every profile involved in the call.
>>>> > sofia profile <profilename> siptrace on
>>>> >
>>>> > //also do this
>>>> > console loglevel debug
>>>> >
>>>> >
>>>> > if you can let us ssh, we can do all the for you if you can make the
>>>> > test
>>>> > calls.
>>>> >
>>>> >
>>>> >
>>>> >
>>>> > On Sun, Feb 14, 2010 at 11:59 PM, Gabriel Kuri <gkuri at ieee.org>
wrote:
>>>> >>
>>>> >> BTW, here's a copy of the NOTIFY (event call-info) sent to the other
>>>> >> phone after the first phone is answered, should this have a
Call-Info
>>>> >> line with an "appearance-state=seized" to turn on the light on the
>>>> >> other phone?
>>>> >>
>>>> >>
>>>> >> NOTIFY sip:2551@<removed>:54446 SIP/2.0.
>>>> >> Via: SIP/2.0/UDP <removed>:9430;rport;branch=z9hG4bK71pN2cXgH851K.
>>>> >> Max-Forwards: 70.
>>>> >> From: <sip:2551 at mydomain.com <mailto:sip%3A2551 at mydomain.com>
>;tag=XeB6ZrKDevpHp.
>>>> >> To: <sip:2551 at mydomain.com <mailto:sip%3A2551 at mydomain.com>
>;tag=c2d34993aac6ea.
>>>> >> Call-ID: 34c34987-8b6fa786@<removed>.
>>>> >> CSeq: 126950830 NOTIFY.
>>>> >> Contact: <sip:2551@<removed>:9430>.
>>>> >> Expires: 3959.
>>>> >> Call-Info:
>>>> >> <sip:mydomain.com>;appearance-index=*;appearance-state=idle.
>>>> >> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16633M.
>>>> >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>>>> >> REGISTER, REFER, PRACK, NOTIFY, PUBLISH, SUBSCRIBE.
>>>> >> Supported: 100rel, timer, precondition, path, replaces.
>>>> >> Event: call-info.
>>>> >> Allow-Events: talk, presence, dialog, line-seize, call-info, sla,
>>>> >> include-session-description, presence.winfo, message-summary, refer.
>>>> >> Subscription-State: active;expires=3959.
>>>> >> Content-Length: 0.
>>>> >>
>>>> >>
>>>> >>
>>>> >> On Sun, Feb 14, 2010 at 7:58 PM, Brian West <brian at freeswitch.org>
>>>> >> wrote:
>>>> >> > Works fine here... is your box slow or something?
>>>> >> >
>>>> >> > /b
>>>> >> >
>>>> >> > On Feb 14, 2010, at 1:03 PM, Gabriel Kuri wrote:
>>>> >> >
>>>> >> >> I followed Brian's directions from one of the previous threads on
>>>> >> >> configuring the SPA-5xx series phones for Broadsoft SCA and set
>>>> >> >> manage-shared-appearance=true in the internal profile. SCA
appears
>>>> >> >> to
>>>> >> >> be working on outgoing calls between two phones, the line key
>>>> >> >> starts
>>>> >> >> flashing red on the second phone when the first phone picks up
the
>>>> >> >> receiver to make a call. However on incoming calls, both phones
>>>> >> >> ring
>>>> >> >> (same extension), however when one of the phones picks up the
> line,
>>>> >> >> the second phone's line key doesn't flash red or show the first
>>>> >> >> phone
>>>> >> >> on that incoming call. Any ideas? Does shared appearance only
work
>>>> >> >> on
>>>> >> >> outgoing phone calls?
>>>> >> >>
>>>> >> >> Thanks,
>>>> >> >> Gabe
>>>> >> >
>>>> >> >
>>>> >> > _______________________________________________
>>>> >> > FreeSWITCH-users mailing list
>>>> >> > FreeSWITCH-users at lists.freeswitch.org
>>>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >> >
>>>> >> >
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> >> > http://www.freeswitch.org
>>>> >> >
>>>> >>
>>>> >> _______________________________________________
>>>> >> FreeSWITCH-users mailing list
>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >>
>>>> >>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> >> http://www.freeswitch.org
>>>> >
>>>> >
>>>> >
>>>> > --
>>>> > Anthony Minessale II
>>>> >
>>>> > FreeSWITCH http://www.freeswitch.org/
>>>> > ClueCon http://www.cluecon.com/
>>>> > Twitter: http://twitter.com/FreeSWITCH_wire
>>>> >
>>>> > AIM: anthm
>>>> > MSN:anthony_minessale at hotmail.com
<mailto:MSN%3Aanthony_minessale at hotmail.com> 
>>>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
<mailto:PAYPAL%3Aanthony.minessale at gmail.com> 
>>>> > IRC: irc.freenode.net #freeswitch
>>>> >
>>>> > FreeSWITCH Developer Conference
>>>> > sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
>>>> > iax:guest at conference.freeswitch.org/888
>>>> > googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
>>>> > pstn:+19193869900
>>>> >
>>>> > _______________________________________________
>>>> > FreeSWITCH-users mailing list
>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >
>>>> >
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> > http://www.freeswitch.org
>>>> >
>>>> >
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com
<mailto:MSN%3Aanthony_minessale at hotmail.com> 
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
<mailto:PAYPAL%3Aanthony.minessale at gmail.com> 
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
>>> iax:guest at conference.freeswitch.org/888
>>> googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
>>> pstn:+19193869900
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
<mailto:MSN%3Aanthony_minessale at hotmail.com> 
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
<mailto:PAYPAL%3Aanthony.minessale at gmail.com> 
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
<mailto:MSN%3Aanthony_minessale at hotmail.com> 
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
<mailto:PAYPAL%3Aanthony.minessale at gmail.com> 
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:+19193869900

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100215/2d11e79a/attachment-0001.html 

From rupa at rupa.com  Mon Feb 15 13:39:12 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Mon, 15 Feb 2010 15:39:12 -0600
Subject: [Freeswitch-users] CS_REPORTING state and CHANNEL_HANGUP event
In-Reply-To: <D0F4BBA35AD84876B35F99FCD443DAF9@abisoft.biz>
References: <D0F4BBA35AD84876B35F99FCD443DAF9@abisoft.biz>
Message-ID: <b39273531002151339k28aadcffyfd1e04df48d01da7@mail.gmail.com>

CS_REPORTING is where CDRs are generated.   How are you doing CDRs or are
you trying to bill at call hangup time?

2010/2/15 Kozak Vladimir <vkozak at abisoft.spb.ru>

>  Hello all.
> I have the following questions:
>     1) Sometime channel whis CS_REPORTING state remain. What is channel
> whis CS_REPORTING state mean? for example: 2c90ac6b-7147-4aac-82fb-a23d6d5c4185,outbound,2010-02-10
> *        11:53:02,1265820782,sofia/internal/sip:7100 at 76.74.160.163:57312
> ,CS_REPORTING,FreeSWITCH,sipp,,7100,,,,default,,,,,,pst01.localdomain.com,
> *,
> How can I kill this channel?  uuid_kill
> 2c90ac6b-7147-4aac-82fb-a23d6d5c4185 is not work.
>     2) Sometime I get CHANNEL_HANGUP event but real call in eyeBeam is
> live for the present. When come CHANNEL_HANGUP event and response on api
> uuid_kill command? It's come when FS send hangup to phone or when phone
> confirm hangup?
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100215/25cff3e1/attachment.html 

From errotan at gmail.com  Mon Feb 15 13:57:28 2010
From: errotan at gmail.com (=?iso-8859-1?q?Pusk=E1s_Zsolt?=)
Date: Mon, 15 Feb 2010 22:57:28 +0100
Subject: [Freeswitch-users] FreeSWITCH Debian packages - build issues
In-Reply-To: <20100215064616.GA32700@jdc.jasonjgw.net>
References: <20100213030812.GA19108@jdc.jasonjgw.net>
	<9CF8766D-2367-4275-ACCF-0F9C6ABC7A09@jerris.com>
	<20100215064616.GA32700@jdc.jasonjgw.net>
Message-ID: <201002152257.28593.errotan@gmail.com>

2010. febru?r 15. 07.46.16 Jason White d?tummal ezt ?rta:
> Michael Jerris <mike at jerris.com> wrote:
> > Did anyone bother opening a bug on jira for this or are we going to just
> > tag 1.0.5 without deb packages?
> 
> Has anyone tried building these on Ubuntu 9.10 or Debian 5.0? I'm not in a
> position to do so at the moment.
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 

Just done with dpkg-buildpackage on Debian 5.0 "lenny" on x86.
It built without errors and everything looks ok and all modules exists, but 
haven't tried to install or run it.


From michal.kalinowski at interia.pl  Mon Feb 15 14:25:58 2010
From: michal.kalinowski at interia.pl (michal kalinowski)
Date: Mon, 15 Feb 2010 23:25:58 +0100
Subject: [Freeswitch-users] ivr from mysql
In-Reply-To: <87f2f3b91002151002v6bbc3e7fgb67dd6d3e72a915@mail.gmail.com>
References: <7c74f5761002150824i41a2ed5did008f36cbb4499a4@mail.gmail.com>
	<87f2f3b91002151002v6bbc3e7fgb67dd6d3e72a915@mail.gmail.com>
Message-ID: <7c74f5761002151425g512fa685n13c58d8f4b894740@mail.gmail.com>

I'm trying use in python something like this:

from freeswitch import *

def xml_fetch( param1, param2 ):
        xml = '''
<?xml version="1.0" encoding="UTF-8" standalone="no"?>
<document type="freeswitch/xml">
  <section name="dialplan" description="RE Dial Plan For FreeSwitch">
    <context name="default">
      <extension name="generated">
        <condition field="destination_number" expression="^9992$">>
         <action application="answer"/>
         <action application="playback" data="${hold_music}"/>
        </condition>
      </extension>
    </context>
  </section>
</document>
'''
        return xml

Of course XML context is with ivr parameters.

So I will try mod_xml_curl in my configuration.

BR,
Micha?


W dniu 15 lutego 2010 19:02 u?ytkownik Michael Collins
<msc at freeswitch.org> napisa?:
> When you say "xml from python" what exactly do you mean? Are you trying to
> use mod_xml_curl? If not you might want to check it out. The other choice is
> to use Lua from the dialplan, although I have a gut feeling that
> mod_xml_curl might be better for you.
> -MC
>
> 2010/2/15 michal kalinowski <michal.kalinowski at interia.pl>
>>
>> Hi,
>>
>> I need build ivr script/aplication which will take dynamically
>> configuration from mysql db (ivr menu, prompts, etc.).
>> My first idea is generate xml from python script. But it's not working
>> properly.
>>
>> Anybody has some idea or have this aplication already done ?
>>
>> BR,
>> Micha?
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From mrene_lists at avgs.ca  Mon Feb 15 14:32:48 2010
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Mon, 15 Feb 2010 17:32:48 -0500
Subject: [Freeswitch-users] ivr from mysql
In-Reply-To: <7c74f5761002151425g512fa685n13c58d8f4b894740@mail.gmail.com>
References: <7c74f5761002150824i41a2ed5did008f36cbb4499a4@mail.gmail.com>
	<87f2f3b91002151002v6bbc3e7fgb67dd6d3e72a915@mail.gmail.com>
	<7c74f5761002151425g512fa685n13c58d8f4b894740@mail.gmail.com>
Message-ID: <0FD30AAA-C2E2-4228-A09B-DB09F4E26811@avgs.ca>

You need to examine the parameters passed into the xml_fetch callback.  
The request will be in the configuration section for "ivr.conf" and  
its parameters will tell you which ivr should be loaded.

Also note that you have to return all the submenus at the same time  
(this is also valid for xml_curl) as freeswitch loads all of them at  
the same time to optimize the amount of times it does queries.

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 15-Feb-10, at 5:25 PM, michal kalinowski wrote:

> I'm trying use in python something like this:
>
> from freeswitch import *
>
> def xml_fetch( param1, param2 ):
>        xml = '''
> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
> <document type="freeswitch/xml">
>  <section name="dialplan" description="RE Dial Plan For FreeSwitch">
>    <context name="default">
>      <extension name="generated">
>        <condition field="destination_number" expression="^9992$">>
>         <action application="answer"/>
>         <action application="playback" data="${hold_music}"/>
>        </condition>
>      </extension>
>    </context>
>  </section>
> </document>
> '''
>        return xml
>
> Of course XML context is with ivr parameters.
>
> So I will try mod_xml_curl in my configuration.
>
> BR,
> Micha?
>
>
> W dniu 15 lutego 2010 19:02 u?ytkownik Michael Collins
> <msc at freeswitch.org> napisa?:
>> When you say "xml from python" what exactly do you mean? Are you  
>> trying to
>> use mod_xml_curl? If not you might want to check it out. The other  
>> choice is
>> to use Lua from the dialplan, although I have a gut feeling that
>> mod_xml_curl might be better for you.
>> -MC
>>
>> 2010/2/15 michal kalinowski <michal.kalinowski at interia.pl>
>>>
>>> Hi,
>>>
>>> I need build ivr script/aplication which will take dynamically
>>> configuration from mysql db (ivr menu, prompts, etc.).
>>> My first idea is generate xml from python script. But it's not  
>>> working
>>> properly.
>>>
>>> Anybody has some idea or have this aplication already done ?
>>>
>>> BR,
>>> Micha?
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From anthony.minessale at gmail.com  Mon Feb 15 14:49:25 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 15 Feb 2010 16:49:25 -0600
Subject: [Freeswitch-users] CS_REPORTING state and CHANNEL_HANGUP event
In-Reply-To: <b39273531002151339k28aadcffyfd1e04df48d01da7@mail.gmail.com>
References: <D0F4BBA35AD84876B35F99FCD443DAF9@abisoft.biz>
	<b39273531002151339k28aadcffyfd1e04df48d01da7@mail.gmail.com>
Message-ID: <191c3a031002151449i3f575a89me9ce23d4fc50dc6b@mail.gmail.com>

make sure you have this problem on latest trunk

On Mon, Feb 15, 2010 at 3:39 PM, Rupa Schomaker <rupa at rupa.com> wrote:

> CS_REPORTING is where CDRs are generated.   How are you doing CDRs or are
> you trying to bill at call hangup time?
>
> 2010/2/15 Kozak Vladimir <vkozak at abisoft.spb.ru>
>
>>  Hello all.
>> I have the following questions:
>>     1) Sometime channel whis CS_REPORTING state remain. What is channel
>> whis CS_REPORTING state mean? for example: 2c90ac6b-7147-4aac-82fb-a23d6d5c4185,outbound,2010-02-10
>> *        11:53:02,1265820782,sofia/internal/sip:7100 at 76.74.160.163:57312
>> ,CS_REPORTING,FreeSWITCH,sipp,,7100,,,,default,,,,,,pst01.localdomain.com
>> ,*,
>> How can I kill this channel?  uuid_kill
>> 2c90ac6b-7147-4aac-82fb-a23d6d5c4185 is not work.
>>     2) Sometime I get CHANNEL_HANGUP event but real call in eyeBeam is
>> live for the present. When come CHANNEL_HANGUP event and response on api
>> uuid_kill command? It's come when FS send hangup to phone or when phone
>> confirm hangup?
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100215/3adbe950/attachment.html 

From errotan at gmail.com  Mon Feb 15 14:53:43 2010
From: errotan at gmail.com (=?iso-8859-1?q?Pusk=E1s_Zsolt?=)
Date: Mon, 15 Feb 2010 23:53:43 +0100
Subject: [Freeswitch-users] FreeSWITCH.Managed.dll deletes on make distclean
Message-ID: <201002152353.43666.errotan@gmail.com>

I usually do svn-clean than svn up when i compile a new version of fs.
I noticed that if i do make distclean the file @ 
src/mod/languages/mod_managed/managed/FreeSWITCH.Managed.dll got deleted.
When i do svn up it gets 'restored':

Restored 'src/mod/languages/mod_managed/managed/FreeSWITCH.Managed.dll'

Is this normal ?

I don't use mod_managed btw.


From gkuri at ieee.org  Mon Feb 15 14:54:12 2010
From: gkuri at ieee.org (Gabriel Kuri)
Date: Mon, 15 Feb 2010 14:54:12 -0800
Subject: [Freeswitch-users] Broadsoft SCA w/ Cisco SPA-5xx series
In-Reply-To: <191c3a031002151304h54c262c5o8ffd5c195e0512d2@mail.gmail.com>
References: <8b1c9cda1002141103te523643ob16db25e3bfa877f@mail.gmail.com>
	<8b1c9cda1002142159h5efcc765q9a41baa1bf694c8c@mail.gmail.com>
	<191c3a031002150832x7faa9a78xbc160cde63f09d09@mail.gmail.com>
	<8b1c9cda1002151048o4b45c034qb22732a9366024ae@mail.gmail.com>
	<191c3a031002151055v1938ab91p89232341f4087c8e@mail.gmail.com>
	<191c3a031002151104u7413b17dh84e68ab1cf703cb9@mail.gmail.com>
	<8b1c9cda1002151147l1e072d00y69c0022e3924c3b4@mail.gmail.com>
	<0dd701caae7e$3c108b70$b431a250$@com>
	<8b1c9cda1002151253t32c22663wf66245e0220a4e6@mail.gmail.com>
	<191c3a031002151304h54c262c5o8ffd5c195e0512d2@mail.gmail.com>
Message-ID: <8b1c9cda1002151454m45afd322o9fd3bc9f0a6fa58f@mail.gmail.com>

The phone I am dialing from (ext 2552) is an SPA-942 and I do not have
it configured for SCA. I only have the two phones than I am calling
(ext 2551) configured for SCA, so that's probably why you're not
seeing the seize events from the SPA-942.

Thanks,
Gabe



On Mon, Feb 15, 2010 at 1:04 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> you are missing something because you have no seize events when you go on
> and off hook.
> is every phone in the correct mode?
>
>
> On Mon, Feb 15, 2010 at 2:53 PM, Gabriel Kuri <gkuri at ieee.org> wrote:
>>
>> No, that was a typo. I have it correct in the config file.
>>
>> Gabe
>>
>> On Mon, Feb 15, 2010 at 12:34 PM, Peder <peder at networkoblivion.com> wrote:
>> > Is this a typo "managed-shared-appeareance=true" or is there an extra e
>> > in
>> > appearance in your config?
>> >
>> > -----Original Message-----
>> > From: freeswitch-users-bounces at lists.freeswitch.org
>> > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
>> > Gabriel
>> > Kuri
>> > Sent: Monday, February 15, 2010 1:48 PM
>> > To: freeswitch-users at lists.freeswitch.org
>> > Subject: Re: [Freeswitch-users] Broadsoft SCA w/ Cisco SPA-5xx series
>> >
>> > OK, I don't know what happened there, here's another one with the
>> > NOTIFYs. I'm on trunk rev 16633 and I have
>> > "managed-shared-appeareance=true" on the internal profile. I'm just
>> > making calls between internal phones.
>> >
>> > ? ? http://pastebin.freeswitch.org/12153
>> >
>> > Thanks,
>> > Gabe
>> >
>> > On Mon, Feb 15, 2010 at 11:04 AM, Anthony Minessale
>> > <anthony.minessale at gmail.com> wrote:
>> >> I don't see any notifies at all in this trace do the profiles in
>> >> question
>> >> have:
>> >> manage-shared-appearance set to true?
>> >> and are you on latest trunk?
>> >>
>> >>
>> >> On Mon, Feb 15, 2010 at 12:55 PM, Anthony Minessale
>> >> <anthony.minessale at gmail.com> wrote:
>> >>>
>> >>> we log the sql stmts on err so they are red and easier to read.
>> >>>
>> >>>
>> >>>
>> >>> On Mon, Feb 15, 2010 at 12:48 PM, Gabriel Kuri <gkuri at ieee.org> wrote:
>> >>>>
>> >>>> Here's a call from ext 2551 to 2552 (two phones). I noticed a bunch
>> >>>> of
>> >>>> errors related to SQL UPDATE for presence ...
>> >>>>
>> >>>> ? ? http://pastebin.freeswitch.org/12152
>> >>>>
>> >>>> Thanks,
>> >>>> Gabe
>> >>>>
>> >>>>
>> >>>> On Mon, Feb 15, 2010 at 8:32 AM, Anthony Minessale
>> >>>> <anthony.minessale at gmail.com> wrote:
>> >>>> > it should be active not seized.
>> >>>> > seized is when you take it off hook.
>> >>>> >
>> >>>> > We need some more debugging to be sure.
>> >>>> > Can we work in real time on it or can you get a more detailed log?
>> >>>> >
>> >>>> > edit sofia.conf.xml and add the param to the "settings" section.
>> >>>> >
>> >>>> > <param name="debug-sla" value="100"/>
>> >>>> >
>> >>>> >
>> >>>> > then restart and enable sip trace and debug level
>> >>>> >
>> >>>> > //do this for every profile involved in the call.
>> >>>> > sofia profile <profilename> siptrace on
>> >>>> >
>> >>>> > //also do this
>> >>>> > console loglevel debug
>> >>>> >
>> >>>> >
>> >>>> > if you can let us ssh, we can do all the for you if you can make
>> >>>> > the
>> >>>> > test
>> >>>> > calls.
>> >>>> >
>> >>>> >
>> >>>> >
>> >>>> >
>> >>>> > On Sun, Feb 14, 2010 at 11:59 PM, Gabriel Kuri <gkuri at ieee.org>
>> >>>> > wrote:
>> >>>> >>
>> >>>> >> BTW, here's a copy of the NOTIFY (event call-info) sent to the
>> >>>> >> other
>> >>>> >> phone after the first phone is answered, should this have a
>> >>>> >> Call-Info
>> >>>> >> line with an "appearance-state=seized" to turn on the light on the
>> >>>> >> other phone?
>> >>>> >>
>> >>>> >>
>> >>>> >> NOTIFY sip:2551@<removed>:54446 SIP/2.0.
>> >>>> >> Via: SIP/2.0/UDP <removed>:9430;rport;branch=z9hG4bK71pN2cXgH851K.
>> >>>> >> Max-Forwards: 70.
>> >>>> >> From: <sip:2551 at mydomain.com>;tag=XeB6ZrKDevpHp.
>> >>>> >> To: <sip:2551 at mydomain.com>;tag=c2d34993aac6ea.
>> >>>> >> Call-ID: 34c34987-8b6fa786@<removed>.
>> >>>> >> CSeq: 126950830 NOTIFY.
>> >>>> >> Contact: <sip:2551@<removed>:9430>.
>> >>>> >> Expires: 3959.
>> >>>> >> Call-Info:
>> >>>> >> <sip:mydomain.com>;appearance-index=*;appearance-state=idle.
>> >>>> >> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16633M.
>> >>>> >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>> >>>> >> REGISTER, REFER, PRACK, NOTIFY, PUBLISH, SUBSCRIBE.
>> >>>> >> Supported: 100rel, timer, precondition, path, replaces.
>> >>>> >> Event: call-info.
>> >>>> >> Allow-Events: talk, presence, dialog, line-seize, call-info, sla,
>> >>>> >> include-session-description, presence.winfo, message-summary,
>> >>>> >> refer.
>> >>>> >> Subscription-State: active;expires=3959.
>> >>>> >> Content-Length: 0.
>> >>>> >>
>> >>>> >>
>> >>>> >>
>> >>>> >> On Sun, Feb 14, 2010 at 7:58 PM, Brian West <brian at freeswitch.org>
>> >>>> >> wrote:
>> >>>> >> > Works fine here... is your box slow or something?
>> >>>> >> >
>> >>>> >> > /b
>> >>>> >> >
>> >>>> >> > On Feb 14, 2010, at 1:03 PM, Gabriel Kuri wrote:
>> >>>> >> >
>> >>>> >> >> I followed Brian's directions from one of the previous threads
>> >>>> >> >> on
>> >>>> >> >> configuring the SPA-5xx series phones for Broadsoft SCA and set
>> >>>> >> >> manage-shared-appearance=true in the internal profile. SCA
>> >>>> >> >> appears
>> >>>> >> >> to
>> >>>> >> >> be working on outgoing calls between two phones, the line key
>> >>>> >> >> starts
>> >>>> >> >> flashing red on the second phone when the first phone picks up
>> >>>> >> >> the
>> >>>> >> >> receiver to make a call. However on incoming calls, both phones
>> >>>> >> >> ring
>> >>>> >> >> (same extension), however when one of the phones picks up the
>> > line,
>> >>>> >> >> the second phone's line key doesn't flash red or show the first
>> >>>> >> >> phone
>> >>>> >> >> on that incoming call. Any ideas? Does shared appearance only
>> >>>> >> >> work
>> >>>> >> >> on
>> >>>> >> >> outgoing phone calls?
>> >>>> >> >>
>> >>>> >> >> Thanks,
>> >>>> >> >> Gabe
>> >>>> >> >
>> >>>> >> >
>> >>>> >> > _______________________________________________
>> >>>> >> > FreeSWITCH-users mailing list
>> >>>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >>>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>> >> >
>> >>>> >> >
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>>> >> > http://www.freeswitch.org
>> >>>> >> >
>> >>>> >>
>> >>>> >> _______________________________________________
>> >>>> >> FreeSWITCH-users mailing list
>> >>>> >> FreeSWITCH-users at lists.freeswitch.org
>> >>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>> >>
>> >>>> >>
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>>> >> http://www.freeswitch.org
>> >>>> >
>> >>>> >
>> >>>> >
>> >>>> > --
>> >>>> > Anthony Minessale II
>> >>>> >
>> >>>> > FreeSWITCH http://www.freeswitch.org/
>> >>>> > ClueCon http://www.cluecon.com/
>> >>>> > Twitter: http://twitter.com/FreeSWITCH_wire
>> >>>> >
>> >>>> > AIM: anthm
>> >>>> > MSN:anthony_minessale at hotmail.com
>> >>>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> >>>> > IRC: irc.freenode.net #freeswitch
>> >>>> >
>> >>>> > FreeSWITCH Developer Conference
>> >>>> > sip:888 at conference.freeswitch.org
>> >>>> > iax:guest at conference.freeswitch.org/888
>> >>>> > googletalk:conf+888 at conference.freeswitch.org
>> >>>> > pstn:+19193869900
>> >>>> >
>> >>>> > _______________________________________________
>> >>>> > FreeSWITCH-users mailing list
>> >>>> > FreeSWITCH-users at lists.freeswitch.org
>> >>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>> >
>> >>>> >
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>>> > http://www.freeswitch.org
>> >>>> >
>> >>>> >
>> >>>>
>> >>>> _______________________________________________
>> >>>> FreeSWITCH-users mailing list
>> >>>> FreeSWITCH-users at lists.freeswitch.org
>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>>
>> >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>>> http://www.freeswitch.org
>> >>>
>> >>>
>> >>>
>> >>> --
>> >>> Anthony Minessale II
>> >>>
>> >>> FreeSWITCH http://www.freeswitch.org/
>> >>> ClueCon http://www.cluecon.com/
>> >>> Twitter: http://twitter.com/FreeSWITCH_wire
>> >>>
>> >>> AIM: anthm
>> >>> MSN:anthony_minessale at hotmail.com
>> >>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> >>> IRC: irc.freenode.net #freeswitch
>> >>>
>> >>> FreeSWITCH Developer Conference
>> >>> sip:888 at conference.freeswitch.org
>> >>> iax:guest at conference.freeswitch.org/888
>> >>> googletalk:conf+888 at conference.freeswitch.org
>> >>> pstn:+19193869900
>> >>
>> >>
>> >>
>> >> --
>> >> Anthony Minessale II
>> >>
>> >> FreeSWITCH http://www.freeswitch.org/
>> >> ClueCon http://www.cluecon.com/
>> >> Twitter: http://twitter.com/FreeSWITCH_wire
>> >>
>> >> AIM: anthm
>> >> MSN:anthony_minessale at hotmail.com
>> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> >> IRC: irc.freenode.net #freeswitch
>> >>
>> >> FreeSWITCH Developer Conference
>> >> sip:888 at conference.freeswitch.org
>> >> iax:guest at conference.freeswitch.org/888
>> >> googletalk:conf+888 at conference.freeswitch.org
>> >> pstn:+19193869900
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >>
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From gkuri at ieee.org  Mon Feb 15 14:56:07 2010
From: gkuri at ieee.org (Gabriel Kuri)
Date: Mon, 15 Feb 2010 14:56:07 -0800
Subject: [Freeswitch-users] Broadsoft SCA w/ Cisco SPA-5xx series
In-Reply-To: <0e7301caae84$58b47ce0$0a1d76a0$@com>
References: <8b1c9cda1002141103te523643ob16db25e3bfa877f@mail.gmail.com>
	<191c3a031002150832x7faa9a78xbc160cde63f09d09@mail.gmail.com>
	<8b1c9cda1002151048o4b45c034qb22732a9366024ae@mail.gmail.com>
	<191c3a031002151055v1938ab91p89232341f4087c8e@mail.gmail.com>
	<191c3a031002151104u7413b17dh84e68ab1cf703cb9@mail.gmail.com>
	<8b1c9cda1002151147l1e072d00y69c0022e3924c3b4@mail.gmail.com>
	<0dd701caae7e$3c108b70$b431a250$@com>
	<8b1c9cda1002151253t32c22663wf66245e0220a4e6@mail.gmail.com>
	<191c3a031002151304h54c262c5o8ffd5c195e0512d2@mail.gmail.com>
	<0e7301caae84$58b47ce0$0a1d76a0$@com>
Message-ID: <8b1c9cda1002151456s5dec6dbbtaa2063e0f70fe5dd@mail.gmail.com>

Yes, the two phones being called (SPA-509Gs) have SCA enabled and it
works when making outgoing calls from those phones. But incoming calls
to those two phones don't seem to have the line key light up on the
other phone when one of the phones is answered (same extension).

Thanks,
Gabe

On Mon, Feb 15, 2010 at 1:17 PM, Peder <peder at networkoblivion.com> wrote:
> On the phone itself, do you have the line set to shared and ?Broadsoft SCA?
> enabled?
>
>
>
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
> Minessale
> Sent: Monday, February 15, 2010 3:04 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Broadsoft SCA w/ Cisco SPA-5xx series
>
>
>
> you are missing something because you have no seize events when you go on
> and off hook.
> is every phone in the correct mode?
>
> On Mon, Feb 15, 2010 at 2:53 PM, Gabriel Kuri <gkuri at ieee.org> wrote:
>
> No, that was a typo. I have it correct in the config file.
>
> Gabe
>
> On Mon, Feb 15, 2010 at 12:34 PM, Peder <peder at networkoblivion.com> wrote:
>
>> Is this a typo "managed-shared-appeareance=true" or is there an extra e in
>> appearance in your config?
>>
>> -----Original Message-----
>> From: freeswitch-users-bounces at lists.freeswitch.org
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
>> Gabriel
>> Kuri
>> Sent: Monday, February 15, 2010 1:48 PM
>> To: freeswitch-users at lists.freeswitch.org
>> Subject: Re: [Freeswitch-users] Broadsoft SCA w/ Cisco SPA-5xx series
>>
>> OK, I don't know what happened there, here's another one with the
>> NOTIFYs. I'm on trunk rev 16633 and I have
>> "managed-shared-appeareance=true" on the internal profile. I'm just
>> making calls between internal phones.
>>
>> ? ? http://pastebin.freeswitch.org/12153
>>
>> Thanks,
>> Gabe
>>
>> On Mon, Feb 15, 2010 at 11:04 AM, Anthony Minessale
>> <anthony.minessale at gmail.com> wrote:
>>> I don't see any notifies at all in this trace do the profiles in question
>>> have:
>>> manage-shared-appearance set to true?
>>> and are you on latest trunk?
>>>
>>>
>>> On Mon, Feb 15, 2010 at 12:55 PM, Anthony Minessale
>>> <anthony.minessale at gmail.com> wrote:
>>>>
>>>> we log the sql stmts on err so they are red and easier to read.
>>>>
>>>>
>>>>
>>>> On Mon, Feb 15, 2010 at 12:48 PM, Gabriel Kuri <gkuri at ieee.org> wrote:
>>>>>
>>>>> Here's a call from ext 2551 to 2552 (two phones). I noticed a bunch of
>>>>> errors related to SQL UPDATE for presence ...
>>>>>
>>>>> ? ? http://pastebin.freeswitch.org/12152
>>>>>
>>>>> Thanks,
>>>>> Gabe
>>>>>
>>>>>
>>>>> On Mon, Feb 15, 2010 at 8:32 AM, Anthony Minessale
>>>>> <anthony.minessale at gmail.com> wrote:
>>>>> > it should be active not seized.
>>>>> > seized is when you take it off hook.
>>>>> >
>>>>> > We need some more debugging to be sure.
>>>>> > Can we work in real time on it or can you get a more detailed log?
>>>>> >
>>>>> > edit sofia.conf.xml and add the param to the "settings" section.
>>>>> >
>>>>> > <param name="debug-sla" value="100"/>
>>>>> >
>>>>> >
>>>>> > then restart and enable sip trace and debug level
>>>>> >
>>>>> > //do this for every profile involved in the call.
>>>>> > sofia profile <profilename> siptrace on
>>>>> >
>>>>> > //also do this
>>>>> > console loglevel debug
>>>>> >
>>>>> >
>>>>> > if you can let us ssh, we can do all the for you if you can make the
>>>>> > test
>>>>> > calls.
>>>>> >
>>>>> >
>>>>> >
>>>>> >
>>>>> > On Sun, Feb 14, 2010 at 11:59 PM, Gabriel Kuri <gkuri at ieee.org>
>>>>> > wrote:
>>>>> >>
>>>>> >> BTW, here's a copy of the NOTIFY (event call-info) sent to the other
>>>>> >> phone after the first phone is answered, should this have a
>>>>> >> Call-Info
>>>>> >> line with an "appearance-state=seized" to turn on the light on the
>>>>> >> other phone?
>>>>> >>
>>>>> >>
>>>>> >> NOTIFY sip:2551@<removed>:54446 SIP/2.0.
>>>>> >> Via: SIP/2.0/UDP <removed>:9430;rport;branch=z9hG4bK71pN2cXgH851K.
>>>>> >> Max-Forwards: 70.
>>>>> >> From: <sip:2551 at mydomain.com>;tag=XeB6ZrKDevpHp.
>>>>> >> To: <sip:2551 at mydomain.com>;tag=c2d34993aac6ea.
>>>>> >> Call-ID: 34c34987-8b6fa786@<removed>.
>>>>> >> CSeq: 126950830 NOTIFY.
>>>>> >> Contact: <sip:2551@<removed>:9430>.
>>>>> >> Expires: 3959.
>>>>> >> Call-Info:
>>>>> >> <sip:mydomain.com>;appearance-index=*;appearance-state=idle.
>>>>> >> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16633M.
>>>>> >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>>>>> >> REGISTER, REFER, PRACK, NOTIFY, PUBLISH, SUBSCRIBE.
>>>>> >> Supported: 100rel, timer, precondition, path, replaces.
>>>>> >> Event: call-info.
>>>>> >> Allow-Events: talk, presence, dialog, line-seize, call-info, sla,
>>>>> >> include-session-description, presence.winfo, message-summary, refer.
>>>>> >> Subscription-State: active;expires=3959.
>>>>> >> Content-Length: 0.
>>>>> >>
>>>>> >>
>>>>> >>
>>>>> >> On Sun, Feb 14, 2010 at 7:58 PM, Brian West <brian at freeswitch.org>
>>>>> >> wrote:
>>>>> >> > Works fine here... is your box slow or something?
>>>>> >> >
>>>>> >> > /b
>>>>> >> >
>>>>> >> > On Feb 14, 2010, at 1:03 PM, Gabriel Kuri wrote:
>>>>> >> >
>>>>> >> >> I followed Brian's directions from one of the previous threads on
>>>>> >> >> configuring the SPA-5xx series phones for Broadsoft SCA and set
>>>>> >> >> manage-shared-appearance=true in the internal profile. SCA
>>>>> >> >> appears
>>>>> >> >> to
>>>>> >> >> be working on outgoing calls between two phones, the line key
>>>>> >> >> starts
>>>>> >> >> flashing red on the second phone when the first phone picks up
>>>>> >> >> the
>>>>> >> >> receiver to make a call. However on incoming calls, both phones
>>>>> >> >> ring
>>>>> >> >> (same extension), however when one of the phones picks up the
>> line,
>>>>> >> >> the second phone's line key doesn't flash red or show the first
>>>>> >> >> phone
>>>>> >> >> on that incoming call. Any ideas? Does shared appearance only
>>>>> >> >> work
>>>>> >> >> on
>>>>> >> >> outgoing phone calls?
>>>>> >> >>
>>>>> >> >> Thanks,
>>>>> >> >> Gabe
>>>>> >> >
>>>>> >> >
>>>>> >> > _______________________________________________
>>>>> >> > FreeSWITCH-users mailing list
>>>>> >> > FreeSWITCH-users at lists.freeswitch.org
>>>>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> >> >
>>>>> >> >
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> >> > http://www.freeswitch.org
>>>>> >> >
>>>>> >>
>>>>> >> _______________________________________________
>>>>> >> FreeSWITCH-users mailing list
>>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> >>
>>>>> >>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> >> http://www.freeswitch.org
>>>>> >
>>>>> >
>>>>> >
>>>>> > --
>>>>> > Anthony Minessale II
>>>>> >
>>>>> > FreeSWITCH http://www.freeswitch.org/
>>>>> > ClueCon http://www.cluecon.com/
>>>>> > Twitter: http://twitter.com/FreeSWITCH_wire
>>>>> >
>>>>> > AIM: anthm
>>>>> > MSN:anthony_minessale at hotmail.com
>>>>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>>>> > IRC: irc.freenode.net #freeswitch
>>>>> >
>>>>> > FreeSWITCH Developer Conference
>>>>> > sip:888 at conference.freeswitch.org
>>>>> > iax:guest at conference.freeswitch.org/888
>>>>> > googletalk:conf+888 at conference.freeswitch.org
>>>>> > pstn:+19193869900
>>>>> >
>>>>> > _______________________________________________
>>>>> > FreeSWITCH-users mailing list
>>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> >
>>>>> >
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> > http://www.freeswitch.org
>>>>> >
>>>>> >
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>> --
>>>> Anthony Minessale II
>>>>
>>>> FreeSWITCH http://www.freeswitch.org/
>>>> ClueCon http://www.cluecon.com/
>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>
>>>> AIM: anthm
>>>> MSN:anthony_minessale at hotmail.com
>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>>> IRC: irc.freenode.net #freeswitch
>>>>
>>>> FreeSWITCH Developer Conference
>>>> sip:888 at conference.freeswitch.org
>>>> iax:guest at conference.freeswitch.org/888
>>>> googletalk:conf+888 at conference.freeswitch.org
>>>> pstn:+19193869900
>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org
>>> iax:guest at conference.freeswitch.org/888
>>> googletalk:conf+888 at conference.freeswitch.org
>>> pstn:+19193869900
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From brian at freeswitch.org  Mon Feb 15 14:56:42 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 15 Feb 2010 16:56:42 -0600
Subject: [Freeswitch-users] FreeSWITCH.Managed.dll deletes on make
	distclean
In-Reply-To: <201002152353.43666.errotan@gmail.com>
References: <201002152353.43666.errotan@gmail.com>
Message-ID: <D5940D2A-0A92-46E4-B83C-14FB8AF69D9D@freeswitch.org>

YES.

/b

On Feb 15, 2010, at 4:53 PM, Pusk?s Zsolt wrote:

> Is this normal ?



From brian at freeswitch.org  Mon Feb 15 14:57:15 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 15 Feb 2010 16:57:15 -0600
Subject: [Freeswitch-users] Broadsoft SCA w/ Cisco SPA-5xx series
In-Reply-To: <8b1c9cda1002151454m45afd322o9fd3bc9f0a6fa58f@mail.gmail.com>
References: <8b1c9cda1002141103te523643ob16db25e3bfa877f@mail.gmail.com>
	<8b1c9cda1002142159h5efcc765q9a41baa1bf694c8c@mail.gmail.com>
	<191c3a031002150832x7faa9a78xbc160cde63f09d09@mail.gmail.com>
	<8b1c9cda1002151048o4b45c034qb22732a9366024ae@mail.gmail.com>
	<191c3a031002151055v1938ab91p89232341f4087c8e@mail.gmail.com>
	<191c3a031002151104u7413b17dh84e68ab1cf703cb9@mail.gmail.com>
	<8b1c9cda1002151147l1e072d00y69c0022e3924c3b4@mail.gmail.com>
	<0dd701caae7e$3c108b70$b431a250$@com>
	<8b1c9cda1002151253t32c22663wf66245e0220a4e6@mail.gmail.com>
	<191c3a031002151304h54c262c5o8ffd5c195e0512d2@mail.gmail.com>
	<8b1c9cda1002151454m45afd322o9fd3bc9f0a6fa58f@mail.gmail.com>
Message-ID: <AC324FC3-2298-4765-887C-451F6F17E951@freeswitch.org>

You'll see it when you life the handset... thats the going to happen no matter what.. are you lifting the handset at all?

/b

On Feb 15, 2010, at 4:54 PM, Gabriel Kuri wrote:

> The phone I am dialing from (ext 2552) is an SPA-942 and I do not have
> it configured for SCA. I only have the two phones than I am calling
> (ext 2551) configured for SCA, so that's probably why you're not
> seeing the seize events from the SPA-942.
> 
> Thanks,
> Gabe



From gorand at donevtechconsulting.com  Mon Feb 15 15:12:41 2010
From: gorand at donevtechconsulting.com (Goran Donev)
Date: Mon, 15 Feb 2010 17:12:41 -0600
Subject: [Freeswitch-users] Version 1.05 release
In-Reply-To: <mailman.6410.1266268685.2495.freeswitch-users@lists.freeswitch.org>
References: <mailman.6410.1266268685.2495.freeswitch-users@lists.freeswitch.org>
Message-ID: <054c01caae94$6131b1c0$23951540$@com>

I really didn't get a definitive answer on when 1.05 is slated to be
released. We are running into some issues that I hope that 1.05 fixes. Do we
have an eta?

Thx



From brian at freeswitch.org  Mon Feb 15 15:17:26 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 15 Feb 2010 17:17:26 -0600
Subject: [Freeswitch-users] Version 1.05 release
In-Reply-To: <054c01caae94$6131b1c0$23951540$@com>
References: <mailman.6410.1266268685.2495.freeswitch-users@lists.freeswitch.org>
	<054c01caae94$6131b1c0$23951540$@com>
Message-ID: <DA13E7EA-D100-4A19-8477-25D5B57A1ECB@freeswitch.org>

Well if you're running into issues you should have already updated to SVN trunk.  1.0.5 is marching to release.  You're better off being on SVN trunk right now anyway. 

/b

On Feb 15, 2010, at 5:12 PM, Goran Donev wrote:

> I really didn't get a definitive answer on when 1.05 is slated to be
> released. We are running into some issues that I hope that 1.05 fixes. Do we
> have an eta?
> 
> Thx
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From brian at freeswitch.org  Mon Feb 15 15:22:04 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 15 Feb 2010 17:22:04 -0600
Subject: [Freeswitch-users] Broadsoft SCA w/ Cisco SPA-5xx series
In-Reply-To: <8b1c9cda1002151252jbc500bax3f137a882171485f@mail.gmail.com>
References: <8b1c9cda1002141103te523643ob16db25e3bfa877f@mail.gmail.com>
	<0D67132C-69D8-43AD-B7E6-C7545B544322@freeswitch.org>
	<8b1c9cda1002142159h5efcc765q9a41baa1bf694c8c@mail.gmail.com>
	<191c3a031002150832x7faa9a78xbc160cde63f09d09@mail.gmail.com>
	<8b1c9cda1002151048o4b45c034qb22732a9366024ae@mail.gmail.com>
	<191c3a031002151055v1938ab91p89232341f4087c8e@mail.gmail.com>
	<191c3a031002151104u7413b17dh84e68ab1cf703cb9@mail.gmail.com>
	<8b1c9cda1002151147l1e072d00y69c0022e3924c3b4@mail.gmail.com>
	<7C6C0110-C6AE-45CA-A1B4-413E39223286@freeswitch.org>
	<8b1c9cda1002151252jbc500bax3f137a882171485f@mail.gmail.com>
Message-ID: <D62B55B2-5DE9-4585-A533-0F77593FA125@freeswitch.org>

Can you try something for me?

Connect to your sofia db and dump its contents and put it on pastebin please.

/b
On Feb 15, 2010, at 2:52 PM, Gabriel Kuri wrote:

> Yes, that is correct. FS is behind a NAT and the phones behind another NAT.
> 
> I have ext-rtp-ip and ext-sip-ip set to the public IP address. Phones
> calls and everything else seem to be working.
> 
> Thanks,
> Gabe



From michal.kalinowski at interia.pl  Mon Feb 15 15:25:32 2010
From: michal.kalinowski at interia.pl (michal kalinowski)
Date: Tue, 16 Feb 2010 00:25:32 +0100
Subject: [Freeswitch-users] ivr from mysql
In-Reply-To: <0FD30AAA-C2E2-4228-A09B-DB09F4E26811@avgs.ca>
References: <7c74f5761002150824i41a2ed5did008f36cbb4499a4@mail.gmail.com>
	<87f2f3b91002151002v6bbc3e7fgb67dd6d3e72a915@mail.gmail.com>
	<7c74f5761002151425g512fa685n13c58d8f4b894740@mail.gmail.com>
	<0FD30AAA-C2E2-4228-A09B-DB09F4E26811@avgs.ca>
Message-ID: <7c74f5761002151525p7cc690ces632d46c6ab5e236c@mail.gmail.com>

Mathieu,

Could you insert several examples here?

BR,
Micha?

W dniu 15 lutego 2010 23:32 u?ytkownik Mathieu Rene
<mrene_lists at avgs.ca> napisa?:
> You need to examine the parameters passed into the xml_fetch callback.
> The request will be in the configuration section for "ivr.conf" and
> its parameters will tell you which ivr should be loaded.
>
> Also note that you have to return all the submenus at the same time
> (this is also valid for xml_curl) as freeswitch loads all of them at
> the same time to optimize the amount of times it does queries.
>
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>
>
>
>
> On 15-Feb-10, at 5:25 PM, michal kalinowski wrote:
>
>> I'm trying use in python something like this:
>>
>> from freeswitch import *
>>
>> def xml_fetch( param1, param2 ):
>> ? ? ? ?xml = '''
>> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
>> <document type="freeswitch/xml">
>> ?<section name="dialplan" description="RE Dial Plan For FreeSwitch">
>> ? ?<context name="default">
>> ? ? ?<extension name="generated">
>> ? ? ? ?<condition field="destination_number" expression="^9992$">>
>> ? ? ? ? <action application="answer"/>
>> ? ? ? ? <action application="playback" data="${hold_music}"/>
>> ? ? ? ?</condition>
>> ? ? ?</extension>
>> ? ?</context>
>> ?</section>
>> </document>
>> '''
>> ? ? ? ?return xml
>>
>> Of course XML context is with ivr parameters.
>>
>> So I will try mod_xml_curl in my configuration.
>>
>> BR,
>> Micha?
>>
>>
>> W dniu 15 lutego 2010 19:02 u?ytkownik Michael Collins
>> <msc at freeswitch.org> napisa?:
>>> When you say "xml from python" what exactly do you mean? Are you
>>> trying to
>>> use mod_xml_curl? If not you might want to check it out. The other
>>> choice is
>>> to use Lua from the dialplan, although I have a gut feeling that
>>> mod_xml_curl might be better for you.
>>> -MC
>>>
>>> 2010/2/15 michal kalinowski <michal.kalinowski at interia.pl>
>>>>
>>>> Hi,
>>>>
>>>> I need build ivr script/aplication which will take dynamically
>>>> configuration from mysql db (ivr menu, prompts, etc.).
>>>> My first idea is generate xml from python script. But it's not
>>>> working
>>>> properly.
>>>>
>>>> Anybody has some idea or have this aplication already done ?
>>>>
>>>> BR,
>>>> Micha?
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From brian at freeswitch.org  Mon Feb 15 15:29:27 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 15 Feb 2010 17:29:27 -0600
Subject: [Freeswitch-users] ivr from mysql
In-Reply-To: <7c74f5761002151525p7cc690ces632d46c6ab5e236c@mail.gmail.com>
References: <7c74f5761002150824i41a2ed5did008f36cbb4499a4@mail.gmail.com>
	<87f2f3b91002151002v6bbc3e7fgb67dd6d3e72a915@mail.gmail.com>
	<7c74f5761002151425g512fa685n13c58d8f4b894740@mail.gmail.com>
	<0FD30AAA-C2E2-4228-A09B-DB09F4E26811@avgs.ca>
	<7c74f5761002151525p7cc690ces632d46c6ab5e236c@mail.gmail.com>
Message-ID: <F2D49DD4-79DC-4F52-A1C1-4C48C79B4637@freeswitch.org>

Examples?  like we need more of those :)  where is the fun in that?

/b
PS: yes we need more docs... and code samples... anyone willing to help out?
On Feb 15, 2010, at 5:25 PM, michal kalinowski wrote:

> Mathieu,
> 
> Could you insert several examples here?
> 
> BR,
> Micha?



From anthony.minessale at gmail.com  Mon Feb 15 15:31:04 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 15 Feb 2010 17:31:04 -0600
Subject: [Freeswitch-users] Broadsoft SCA w/ Cisco SPA-5xx series
In-Reply-To: <8b1c9cda1002151456s5dec6dbbtaa2063e0f70fe5dd@mail.gmail.com>
References: <8b1c9cda1002141103te523643ob16db25e3bfa877f@mail.gmail.com>
	<8b1c9cda1002151048o4b45c034qb22732a9366024ae@mail.gmail.com>
	<191c3a031002151055v1938ab91p89232341f4087c8e@mail.gmail.com>
	<191c3a031002151104u7413b17dh84e68ab1cf703cb9@mail.gmail.com>
	<8b1c9cda1002151147l1e072d00y69c0022e3924c3b4@mail.gmail.com>
	<0dd701caae7e$3c108b70$b431a250$@com>
	<8b1c9cda1002151253t32c22663wf66245e0220a4e6@mail.gmail.com>
	<191c3a031002151304h54c262c5o8ffd5c195e0512d2@mail.gmail.com>
	<0e7301caae84$58b47ce0$0a1d76a0$@com>
	<8b1c9cda1002151456s5dec6dbbtaa2063e0f70fe5dd@mail.gmail.com>
Message-ID: <191c3a031002151531k148d8504w15e6f4c6c1e3ccd@mail.gmail.com>

Do the domain names match on what the remote phones are using?

When the call is active, can you attach to sqlite with the sqlite3 app and
select * from sip_dialogs

sqlite3 /usr/local/freeswitch/db/sofia_reg_internal.db
> select * from sip_dialogs;

remember to do it while the call is up.


I am going to bet the domain name in that table is not the same as your
actual domain.


I tried to make this easier by asking to ssh to your box and work with you
to fix it but now 9 hours later its starting to resemble diffusing a bomb
over a telegraph wire.




On Mon, Feb 15, 2010 at 4:56 PM, Gabriel Kuri <gkuri at ieee.org> wrote:

> Yes, the two phones being called (SPA-509Gs) have SCA enabled and it
> works when making outgoing calls from those phones. But incoming calls
> to those two phones don't seem to have the line key light up on the
> other phone when one of the phones is answered (same extension).
>
> Thanks,
> Gabe
>
> On Mon, Feb 15, 2010 at 1:17 PM, Peder <peder at networkoblivion.com> wrote:
> > On the phone itself, do you have the line set to shared and ?Broadsoft
> SCA?
> > enabled?
> >
> >
> >
> > From: freeswitch-users-bounces at lists.freeswitch.org
> > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Anthony
> > Minessale
> > Sent: Monday, February 15, 2010 3:04 PM
> > To: freeswitch-users at lists.freeswitch.org
> > Subject: Re: [Freeswitch-users] Broadsoft SCA w/ Cisco SPA-5xx series
> >
> >
> >
> > you are missing something because you have no seize events when you go on
> > and off hook.
> > is every phone in the correct mode?
> >
> > On Mon, Feb 15, 2010 at 2:53 PM, Gabriel Kuri <gkuri at ieee.org> wrote:
> >
> > No, that was a typo. I have it correct in the config file.
> >
> > Gabe
> >
> > On Mon, Feb 15, 2010 at 12:34 PM, Peder <peder at networkoblivion.com>
> wrote:
> >
> >> Is this a typo "managed-shared-appeareance=true" or is there an extra e
> in
> >> appearance in your config?
> >>
> >> -----Original Message-----
> >> From: freeswitch-users-bounces at lists.freeswitch.org
> >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> >> Gabriel
> >> Kuri
> >> Sent: Monday, February 15, 2010 1:48 PM
> >> To: freeswitch-users at lists.freeswitch.org
> >> Subject: Re: [Freeswitch-users] Broadsoft SCA w/ Cisco SPA-5xx series
> >>
> >> OK, I don't know what happened there, here's another one with the
> >> NOTIFYs. I'm on trunk rev 16633 and I have
> >> "managed-shared-appeareance=true" on the internal profile. I'm just
> >> making calls between internal phones.
> >>
> >>     http://pastebin.freeswitch.org/12153
> >>
> >> Thanks,
> >> Gabe
> >>
> >> On Mon, Feb 15, 2010 at 11:04 AM, Anthony Minessale
> >> <anthony.minessale at gmail.com> wrote:
> >>> I don't see any notifies at all in this trace do the profiles in
> question
> >>> have:
> >>> manage-shared-appearance set to true?
> >>> and are you on latest trunk?
> >>>
> >>>
> >>> On Mon, Feb 15, 2010 at 12:55 PM, Anthony Minessale
> >>> <anthony.minessale at gmail.com> wrote:
> >>>>
> >>>> we log the sql stmts on err so they are red and easier to read.
> >>>>
> >>>>
> >>>>
> >>>> On Mon, Feb 15, 2010 at 12:48 PM, Gabriel Kuri <gkuri at ieee.org>
> wrote:
> >>>>>
> >>>>> Here's a call from ext 2551 to 2552 (two phones). I noticed a bunch
> of
> >>>>> errors related to SQL UPDATE for presence ...
> >>>>>
> >>>>>     http://pastebin.freeswitch.org/12152
> >>>>>
> >>>>> Thanks,
> >>>>> Gabe
> >>>>>
> >>>>>
> >>>>> On Mon, Feb 15, 2010 at 8:32 AM, Anthony Minessale
> >>>>> <anthony.minessale at gmail.com> wrote:
> >>>>> > it should be active not seized.
> >>>>> > seized is when you take it off hook.
> >>>>> >
> >>>>> > We need some more debugging to be sure.
> >>>>> > Can we work in real time on it or can you get a more detailed log?
> >>>>> >
> >>>>> > edit sofia.conf.xml and add the param to the "settings" section.
> >>>>> >
> >>>>> > <param name="debug-sla" value="100"/>
> >>>>> >
> >>>>> >
> >>>>> > then restart and enable sip trace and debug level
> >>>>> >
> >>>>> > //do this for every profile involved in the call.
> >>>>> > sofia profile <profilename> siptrace on
> >>>>> >
> >>>>> > //also do this
> >>>>> > console loglevel debug
> >>>>> >
> >>>>> >
> >>>>> > if you can let us ssh, we can do all the for you if you can make
> the
> >>>>> > test
> >>>>> > calls.
> >>>>> >
> >>>>> >
> >>>>> >
> >>>>> >
> >>>>> > On Sun, Feb 14, 2010 at 11:59 PM, Gabriel Kuri <gkuri at ieee.org>
> >>>>> > wrote:
> >>>>> >>
> >>>>> >> BTW, here's a copy of the NOTIFY (event call-info) sent to the
> other
> >>>>> >> phone after the first phone is answered, should this have a
> >>>>> >> Call-Info
> >>>>> >> line with an "appearance-state=seized" to turn on the light on the
> >>>>> >> other phone?
> >>>>> >>
> >>>>> >>
> >>>>> >> NOTIFY sip:2551@<removed>:54446 SIP/2.0.
> >>>>> >> Via: SIP/2.0/UDP <removed>:9430;rport;branch=z9hG4bK71pN2cXgH851K.
> >>>>> >> Max-Forwards: 70.
> >>>>> >> From: <sip:2551 at mydomain.com <sip%3A2551 at mydomain.com>
> >;tag=XeB6ZrKDevpHp.
> >>>>> >> To: <sip:2551 at mydomain.com <sip%3A2551 at mydomain.com>
> >;tag=c2d34993aac6ea.
> >>>>> >> Call-ID: 34c34987-8b6fa786@<removed>.
> >>>>> >> CSeq: 126950830 NOTIFY.
> >>>>> >> Contact: <sip:2551@<removed>:9430>.
> >>>>> >> Expires: 3959.
> >>>>> >> Call-Info:
> >>>>> >> <sip:mydomain.com>;appearance-index=*;appearance-state=idle.
> >>>>> >> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16633M.
> >>>>> >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> >>>>> >> REGISTER, REFER, PRACK, NOTIFY, PUBLISH, SUBSCRIBE.
> >>>>> >> Supported: 100rel, timer, precondition, path, replaces.
> >>>>> >> Event: call-info.
> >>>>> >> Allow-Events: talk, presence, dialog, line-seize, call-info, sla,
> >>>>> >> include-session-description, presence.winfo, message-summary,
> refer.
> >>>>> >> Subscription-State: active;expires=3959.
> >>>>> >> Content-Length: 0.
> >>>>> >>
> >>>>> >>
> >>>>> >>
> >>>>> >> On Sun, Feb 14, 2010 at 7:58 PM, Brian West <brian at freeswitch.org
> >
> >>>>> >> wrote:
> >>>>> >> > Works fine here... is your box slow or something?
> >>>>> >> >
> >>>>> >> > /b
> >>>>> >> >
> >>>>> >> > On Feb 14, 2010, at 1:03 PM, Gabriel Kuri wrote:
> >>>>> >> >
> >>>>> >> >> I followed Brian's directions from one of the previous threads
> on
> >>>>> >> >> configuring the SPA-5xx series phones for Broadsoft SCA and set
> >>>>> >> >> manage-shared-appearance=true in the internal profile. SCA
> >>>>> >> >> appears
> >>>>> >> >> to
> >>>>> >> >> be working on outgoing calls between two phones, the line key
> >>>>> >> >> starts
> >>>>> >> >> flashing red on the second phone when the first phone picks up
> >>>>> >> >> the
> >>>>> >> >> receiver to make a call. However on incoming calls, both phones
> >>>>> >> >> ring
> >>>>> >> >> (same extension), however when one of the phones picks up the
> >> line,
> >>>>> >> >> the second phone's line key doesn't flash red or show the first
> >>>>> >> >> phone
> >>>>> >> >> on that incoming call. Any ideas? Does shared appearance only
> >>>>> >> >> work
> >>>>> >> >> on
> >>>>> >> >> outgoing phone calls?
> >>>>> >> >>
> >>>>> >> >> Thanks,
> >>>>> >> >> Gabe
> >>>>> >> >
> >>>>> >> >
> >>>>> >> > _______________________________________________
> >>>>> >> > FreeSWITCH-users mailing list
> >>>>> >> > FreeSWITCH-users at lists.freeswitch.org
> >>>>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>> >> >
> >>>>> >> >
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>> >> > http://www.freeswitch.org
> >>>>> >> >
> >>>>> >>
> >>>>> >> _______________________________________________
> >>>>> >> FreeSWITCH-users mailing list
> >>>>> >> FreeSWITCH-users at lists.freeswitch.org
> >>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>> >>
> >>>>> >>
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>> >> http://www.freeswitch.org
> >>>>> >
> >>>>> >
> >>>>> >
> >>>>> > --
> >>>>> > Anthony Minessale II
> >>>>> >
> >>>>> > FreeSWITCH http://www.freeswitch.org/
> >>>>> > ClueCon http://www.cluecon.com/
> >>>>> > Twitter: http://twitter.com/FreeSWITCH_wire
> >>>>> >
> >>>>> > AIM: anthm
> >>>>> > MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
> >>>>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >>>>> > IRC: irc.freenode.net #freeswitch
> >>>>> >
> >>>>> > FreeSWITCH Developer Conference
> >>>>> > sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> >>>>> > iax:guest at conference.freeswitch.org/888
> >>>>> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >>>>> > pstn:+19193869900
> >>>>> >
> >>>>> > _______________________________________________
> >>>>> > FreeSWITCH-users mailing list
> >>>>> > FreeSWITCH-users at lists.freeswitch.org
> >>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>> >
> >>>>> >
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>> > http://www.freeswitch.org
> >>>>> >
> >>>>> >
> >>>>>
> >>>>> _______________________________________________
> >>>>> FreeSWITCH-users mailing list
> >>>>> FreeSWITCH-users at lists.freeswitch.org
> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>>
> >>>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>> http://www.freeswitch.org
> >>>>
> >>>>
> >>>>
> >>>> --
> >>>> Anthony Minessale II
> >>>>
> >>>> FreeSWITCH http://www.freeswitch.org/
> >>>> ClueCon http://www.cluecon.com/
> >>>> Twitter: http://twitter.com/FreeSWITCH_wire
> >>>>
> >>>> AIM: anthm
> >>>> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
> >>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >>>> IRC: irc.freenode.net #freeswitch
> >>>>
> >>>> FreeSWITCH Developer Conference
> >>>> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> >>>> iax:guest at conference.freeswitch.org/888
> >>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >>>> pstn:+19193869900
> >>>
> >>>
> >>>
> >>> --
> >>> Anthony Minessale II
> >>>
> >>> FreeSWITCH http://www.freeswitch.org/
> >>> ClueCon http://www.cluecon.com/
> >>> Twitter: http://twitter.com/FreeSWITCH_wire
> >>>
> >>> AIM: anthm
> >>> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
> >>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >>> IRC: irc.freenode.net #freeswitch
> >>>
> >>> FreeSWITCH Developer Conference
> >>> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> >>> iax:guest at conference.freeswitch.org/888
> >>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >>> pstn:+19193869900
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>>
> >>>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> > Twitter: http://twitter.com/FreeSWITCH_wire
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> > IRC: irc.freenode.net #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> > iax:guest at conference.freeswitch.org/888
> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > pstn:+19193869900
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100215/122fa7df/attachment-0001.html 

From anthony.minessale at gmail.com  Mon Feb 15 15:43:03 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 15 Feb 2010 17:43:03 -0600
Subject: [Freeswitch-users] Version 1.05 release
In-Reply-To: <054c01caae94$6131b1c0$23951540$@com>
References: <mailman.6410.1266268685.2495.freeswitch-users@lists.freeswitch.org>
	<054c01caae94$6131b1c0$23951540$@com>
Message-ID: <191c3a031002151543q6a00a936vf7a5e9d4b69d0ebc@mail.gmail.com>

Since you are advertising services on your website most likely all provided
by our free software, I hope you can learn some patience while you wait for
us to provide you with your next release to sell to your customers.

The release will be as soon as we make sure all the bugs are fixed, unless
you prefer it to be buggy.

BTW,

If your issues are still present in 1.0.5 and it's because you never
reported them, then we will, of course, be very unhappy.

If this is unacceptable we do offer a triple-your-money-back guarantee that
we will upgrade to quadruple if you act now.


On Mon, Feb 15, 2010 at 5:12 PM, Goran Donev <gorand at donevtechconsulting.com
> wrote:

> I really didn't get a definitive answer on when 1.05 is slated to be
> released. We are running into some issues that I hope that 1.05 fixes. Do
> we
> have an eta?
>
> Thx
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100215/3da9decd/attachment.html 

From rupa at rupa.com  Mon Feb 15 16:00:51 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Mon, 15 Feb 2010 18:00:51 -0600
Subject: [Freeswitch-users] FreeSWITCH Debian packages - build issues
In-Reply-To: <201002152257.28593.errotan@gmail.com>
References: <20100213030812.GA19108@jdc.jasonjgw.net>
	<9CF8766D-2367-4275-ACCF-0F9C6ABC7A09@jerris.com>
	<20100215064616.GA32700@jdc.jasonjgw.net>
	<201002152257.28593.errotan@gmail.com>
Message-ID: <b39273531002151600l5c5620eapd2eb745d7bb3bbc0@mail.gmail.com>

I made a minor change to mod_memcache's Makefile -- can you try re-enabling
mod_memcache in your debian build and see if it builds for you?

On Mon, Feb 15, 2010 at 3:57 PM, Pusk?s Zsolt <errotan at gmail.com> wrote:

> 2010. febru?r 15. 07.46.16 Jason White d?tummal ezt ?rta:
> > Michael Jerris <mike at jerris.com> wrote:
> > > Did anyone bother opening a bug on jira for this or are we going to
> just
> > > tag 1.0.5 without deb packages?
> >
> > Has anyone tried building these on Ubuntu 9.10 or Debian 5.0? I'm not in
> a
> > position to do so at the moment.
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> Just done with dpkg-buildpackage on Debian 5.0 "lenny" on x86.
> It built without errors and everything looks ok and all modules exists, but
> haven't tried to install or run it.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
-Rupa
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100215/63b31383/attachment.html 

From jason at jasonjgw.net  Mon Feb 15 18:09:34 2010
From: jason at jasonjgw.net (Jason White)
Date: Tue, 16 Feb 2010 13:09:34 +1100
Subject: [Freeswitch-users] FreeSWITCH Debian packages - build issues
In-Reply-To: <b39273531002151600l5c5620eapd2eb745d7bb3bbc0@mail.gmail.com>
References: <20100213030812.GA19108@jdc.jasonjgw.net>
	<9CF8766D-2367-4275-ACCF-0F9C6ABC7A09@jerris.com>
	<20100215064616.GA32700@jdc.jasonjgw.net>
	<201002152257.28593.errotan@gmail.com>
	<b39273531002151600l5c5620eapd2eb745d7bb3bbc0@mail.gmail.com>
Message-ID: <20100216020934.GA10717@jdc.jasonjgw.net>

Rupa Schomaker <rupa at rupa.com> wrote:
> I made a minor change to mod_memcache's Makefile -- can you try re-enabling
> mod_memcache in your debian build and see if it builds for you?

It built this time:

-rw-r----- 1 freeswitch daemon 73552 Feb 16 02:02 mod_memcache.so



From gkuri at ieee.org  Mon Feb 15 22:20:59 2010
From: gkuri at ieee.org (Gabriel Kuri)
Date: Mon, 15 Feb 2010 22:20:59 -0800
Subject: [Freeswitch-users] Broadsoft SCA w/ Cisco SPA-5xx series
In-Reply-To: <D62B55B2-5DE9-4585-A533-0F77593FA125@freeswitch.org>
References: <8b1c9cda1002141103te523643ob16db25e3bfa877f@mail.gmail.com>
	<8b1c9cda1002142159h5efcc765q9a41baa1bf694c8c@mail.gmail.com>
	<191c3a031002150832x7faa9a78xbc160cde63f09d09@mail.gmail.com>
	<8b1c9cda1002151048o4b45c034qb22732a9366024ae@mail.gmail.com>
	<191c3a031002151055v1938ab91p89232341f4087c8e@mail.gmail.com>
	<191c3a031002151104u7413b17dh84e68ab1cf703cb9@mail.gmail.com>
	<8b1c9cda1002151147l1e072d00y69c0022e3924c3b4@mail.gmail.com>
	<7C6C0110-C6AE-45CA-A1B4-413E39223286@freeswitch.org>
	<8b1c9cda1002151252jbc500bax3f137a882171485f@mail.gmail.com>
	<D62B55B2-5DE9-4585-A533-0F77593FA125@freeswitch.org>
Message-ID: <8b1c9cda1002152220i710d7174n9ad7cdaa22c6452@mail.gmail.com>

Here's the database dump ...

     http://pastebin.freeswitch.org/12158

Thanks,
Gabe


On Mon, Feb 15, 2010 at 3:22 PM, Brian West <brian at freeswitch.org> wrote:
> Can you try something for me?
>
> Connect to your sofia db and dump its contents and put it on pastebin please.
>
> /b
> On Feb 15, 2010, at 2:52 PM, Gabriel Kuri wrote:
>
>> Yes, that is correct. FS is behind a NAT and the phones behind another NAT.
>>
>> I have ext-rtp-ip and ext-sip-ip set to the public IP address. Phones
>> calls and everything else seem to be working.
>>
>> Thanks,
>> Gabe
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From gkuri at ieee.org  Mon Feb 15 22:25:56 2010
From: gkuri at ieee.org (Gabriel Kuri)
Date: Mon, 15 Feb 2010 22:25:56 -0800
Subject: [Freeswitch-users] Broadsoft SCA w/ Cisco SPA-5xx series
In-Reply-To: <191c3a031002151531k148d8504w15e6f4c6c1e3ccd@mail.gmail.com>
References: <8b1c9cda1002141103te523643ob16db25e3bfa877f@mail.gmail.com>
	<191c3a031002151055v1938ab91p89232341f4087c8e@mail.gmail.com>
	<191c3a031002151104u7413b17dh84e68ab1cf703cb9@mail.gmail.com>
	<8b1c9cda1002151147l1e072d00y69c0022e3924c3b4@mail.gmail.com>
	<0dd701caae7e$3c108b70$b431a250$@com>
	<8b1c9cda1002151253t32c22663wf66245e0220a4e6@mail.gmail.com>
	<191c3a031002151304h54c262c5o8ffd5c195e0512d2@mail.gmail.com>
	<0e7301caae84$58b47ce0$0a1d76a0$@com>
	<8b1c9cda1002151456s5dec6dbbtaa2063e0f70fe5dd@mail.gmail.com>
	<191c3a031002151531k148d8504w15e6f4c6c1e3ccd@mail.gmail.com>
Message-ID: <8b1c9cda1002152225i7700bdfcgd2c112bf8180e36@mail.gmail.com>

Yeah, the domain name matches on the internal profile.

Thanks for all your help, I can arrange ssh access tomorrow, today
just wasn't one of those good days to do so, I've been running in and
out too much to coordinate it.

Here's the pastebin for the sip_dialogs table while the call is up ...

     http://pastebin.freeswitch.org/12159

Thanks,
Gabe


On Mon, Feb 15, 2010 at 3:31 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> Do the domain names match on what the remote phones are using?
>
> When the call is active, can you attach to sqlite with the sqlite3 app and
> select * from sip_dialogs
>
> sqlite3 /usr/local/freeswitch/db/sofia_reg_internal.db
>> select * from sip_dialogs;
>
> remember to do it while the call is up.
>
>
> I am going to bet the domain name in that table is not the same as your
> actual domain.
>
>
> I tried to make this easier by asking to ssh to your box and work with you
> to fix it but now 9 hours later its starting to resemble diffusing a bomb
> over a telegraph wire.
>
>
>
>
> On Mon, Feb 15, 2010 at 4:56 PM, Gabriel Kuri <gkuri at ieee.org> wrote:
>>
>> Yes, the two phones being called (SPA-509Gs) have SCA enabled and it
>> works when making outgoing calls from those phones. But incoming calls
>> to those two phones don't seem to have the line key light up on the
>> other phone when one of the phones is answered (same extension).
>>
>> Thanks,
>> Gabe
>>
>> On Mon, Feb 15, 2010 at 1:17 PM, Peder <peder at networkoblivion.com> wrote:
>> > On the phone itself, do you have the line set to shared and ?Broadsoft
>> > SCA?
>> > enabled?
>> >
>> >
>> >
>> > From: freeswitch-users-bounces at lists.freeswitch.org
>> > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
>> > Anthony
>> > Minessale
>> > Sent: Monday, February 15, 2010 3:04 PM
>> > To: freeswitch-users at lists.freeswitch.org
>> > Subject: Re: [Freeswitch-users] Broadsoft SCA w/ Cisco SPA-5xx series
>> >
>> >
>> >
>> > you are missing something because you have no seize events when you go
>> > on
>> > and off hook.
>> > is every phone in the correct mode?
>> >
>> > On Mon, Feb 15, 2010 at 2:53 PM, Gabriel Kuri <gkuri at ieee.org> wrote:
>> >
>> > No, that was a typo. I have it correct in the config file.
>> >
>> > Gabe
>> >
>> > On Mon, Feb 15, 2010 at 12:34 PM, Peder <peder at networkoblivion.com>
>> > wrote:
>> >
>> >> Is this a typo "managed-shared-appeareance=true" or is there an extra e
>> >> in
>> >> appearance in your config?
>> >>
>> >> -----Original Message-----
>> >> From: freeswitch-users-bounces at lists.freeswitch.org
>> >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
>> >> Gabriel
>> >> Kuri
>> >> Sent: Monday, February 15, 2010 1:48 PM
>> >> To: freeswitch-users at lists.freeswitch.org
>> >> Subject: Re: [Freeswitch-users] Broadsoft SCA w/ Cisco SPA-5xx series
>> >>
>> >> OK, I don't know what happened there, here's another one with the
>> >> NOTIFYs. I'm on trunk rev 16633 and I have
>> >> "managed-shared-appeareance=true" on the internal profile. I'm just
>> >> making calls between internal phones.
>> >>
>> >> ? ? http://pastebin.freeswitch.org/12153
>> >>
>> >> Thanks,
>> >> Gabe
>> >>
>> >> On Mon, Feb 15, 2010 at 11:04 AM, Anthony Minessale
>> >> <anthony.minessale at gmail.com> wrote:
>> >>> I don't see any notifies at all in this trace do the profiles in
>> >>> question
>> >>> have:
>> >>> manage-shared-appearance set to true?
>> >>> and are you on latest trunk?
>> >>>
>> >>>
>> >>> On Mon, Feb 15, 2010 at 12:55 PM, Anthony Minessale
>> >>> <anthony.minessale at gmail.com> wrote:
>> >>>>
>> >>>> we log the sql stmts on err so they are red and easier to read.
>> >>>>
>> >>>>
>> >>>>
>> >>>> On Mon, Feb 15, 2010 at 12:48 PM, Gabriel Kuri <gkuri at ieee.org>
>> >>>> wrote:
>> >>>>>
>> >>>>> Here's a call from ext 2551 to 2552 (two phones). I noticed a bunch
>> >>>>> of
>> >>>>> errors related to SQL UPDATE for presence ...
>> >>>>>
>> >>>>> ? ? http://pastebin.freeswitch.org/12152
>> >>>>>
>> >>>>> Thanks,
>> >>>>> Gabe
>> >>>>>
>> >>>>>
>> >>>>> On Mon, Feb 15, 2010 at 8:32 AM, Anthony Minessale
>> >>>>> <anthony.minessale at gmail.com> wrote:
>> >>>>> > it should be active not seized.
>> >>>>> > seized is when you take it off hook.
>> >>>>> >
>> >>>>> > We need some more debugging to be sure.
>> >>>>> > Can we work in real time on it or can you get a more detailed log?
>> >>>>> >
>> >>>>> > edit sofia.conf.xml and add the param to the "settings" section.
>> >>>>> >
>> >>>>> > <param name="debug-sla" value="100"/>
>> >>>>> >
>> >>>>> >
>> >>>>> > then restart and enable sip trace and debug level
>> >>>>> >
>> >>>>> > //do this for every profile involved in the call.
>> >>>>> > sofia profile <profilename> siptrace on
>> >>>>> >
>> >>>>> > //also do this
>> >>>>> > console loglevel debug
>> >>>>> >
>> >>>>> >
>> >>>>> > if you can let us ssh, we can do all the for you if you can make
>> >>>>> > the
>> >>>>> > test
>> >>>>> > calls.
>> >>>>> >
>> >>>>> >
>> >>>>> >
>> >>>>> >
>> >>>>> > On Sun, Feb 14, 2010 at 11:59 PM, Gabriel Kuri <gkuri at ieee.org>
>> >>>>> > wrote:
>> >>>>> >>
>> >>>>> >> BTW, here's a copy of the NOTIFY (event call-info) sent to the
>> >>>>> >> other
>> >>>>> >> phone after the first phone is answered, should this have a
>> >>>>> >> Call-Info
>> >>>>> >> line with an "appearance-state=seized" to turn on the light on
>> >>>>> >> the
>> >>>>> >> other phone?
>> >>>>> >>
>> >>>>> >>
>> >>>>> >> NOTIFY sip:2551@<removed>:54446 SIP/2.0.
>> >>>>> >> Via: SIP/2.0/UDP
>> >>>>> >> <removed>:9430;rport;branch=z9hG4bK71pN2cXgH851K.
>> >>>>> >> Max-Forwards: 70.
>> >>>>> >> From: <sip:2551 at mydomain.com>;tag=XeB6ZrKDevpHp.
>> >>>>> >> To: <sip:2551 at mydomain.com>;tag=c2d34993aac6ea.
>> >>>>> >> Call-ID: 34c34987-8b6fa786@<removed>.
>> >>>>> >> CSeq: 126950830 NOTIFY.
>> >>>>> >> Contact: <sip:2551@<removed>:9430>.
>> >>>>> >> Expires: 3959.
>> >>>>> >> Call-Info:
>> >>>>> >> <sip:mydomain.com>;appearance-index=*;appearance-state=idle.
>> >>>>> >> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16633M.
>> >>>>> >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>> >>>>> >> REGISTER, REFER, PRACK, NOTIFY, PUBLISH, SUBSCRIBE.
>> >>>>> >> Supported: 100rel, timer, precondition, path, replaces.
>> >>>>> >> Event: call-info.
>> >>>>> >> Allow-Events: talk, presence, dialog, line-seize, call-info, sla,
>> >>>>> >> include-session-description, presence.winfo, message-summary,
>> >>>>> >> refer.
>> >>>>> >> Subscription-State: active;expires=3959.
>> >>>>> >> Content-Length: 0.
>> >>>>> >>
>> >>>>> >>
>> >>>>> >>
>> >>>>> >> On Sun, Feb 14, 2010 at 7:58 PM, Brian West
>> >>>>> >> <brian at freeswitch.org>
>> >>>>> >> wrote:
>> >>>>> >> > Works fine here... is your box slow or something?
>> >>>>> >> >
>> >>>>> >> > /b
>> >>>>> >> >
>> >>>>> >> > On Feb 14, 2010, at 1:03 PM, Gabriel Kuri wrote:
>> >>>>> >> >
>> >>>>> >> >> I followed Brian's directions from one of the previous threads
>> >>>>> >> >> on
>> >>>>> >> >> configuring the SPA-5xx series phones for Broadsoft SCA and
>> >>>>> >> >> set
>> >>>>> >> >> manage-shared-appearance=true in the internal profile. SCA
>> >>>>> >> >> appears
>> >>>>> >> >> to
>> >>>>> >> >> be working on outgoing calls between two phones, the line key
>> >>>>> >> >> starts
>> >>>>> >> >> flashing red on the second phone when the first phone picks up
>> >>>>> >> >> the
>> >>>>> >> >> receiver to make a call. However on incoming calls, both
>> >>>>> >> >> phones
>> >>>>> >> >> ring
>> >>>>> >> >> (same extension), however when one of the phones picks up the
>> >> line,
>> >>>>> >> >> the second phone's line key doesn't flash red or show the
>> >>>>> >> >> first
>> >>>>> >> >> phone
>> >>>>> >> >> on that incoming call. Any ideas? Does shared appearance only
>> >>>>> >> >> work
>> >>>>> >> >> on
>> >>>>> >> >> outgoing phone calls?
>> >>>>> >> >>
>> >>>>> >> >> Thanks,
>> >>>>> >> >> Gabe
>> >>>>> >> >
>> >>>>> >> >
>> >>>>> >> > _______________________________________________
>> >>>>> >> > FreeSWITCH-users mailing list
>> >>>>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >>>>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>>> >> >
>> >>>>> >> >
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>>>> >> > http://www.freeswitch.org
>> >>>>> >> >
>> >>>>> >>
>> >>>>> >> _______________________________________________
>> >>>>> >> FreeSWITCH-users mailing list
>> >>>>> >> FreeSWITCH-users at lists.freeswitch.org
>> >>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>>> >>
>> >>>>> >>
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>>>> >> http://www.freeswitch.org
>> >>>>> >
>> >>>>> >
>> >>>>> >
>> >>>>> > --
>> >>>>> > Anthony Minessale II
>> >>>>> >
>> >>>>> > FreeSWITCH http://www.freeswitch.org/
>> >>>>> > ClueCon http://www.cluecon.com/
>> >>>>> > Twitter: http://twitter.com/FreeSWITCH_wire
>> >>>>> >
>> >>>>> > AIM: anthm
>> >>>>> > MSN:anthony_minessale at hotmail.com
>> >>>>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> >>>>> > IRC: irc.freenode.net #freeswitch
>> >>>>> >
>> >>>>> > FreeSWITCH Developer Conference
>> >>>>> > sip:888 at conference.freeswitch.org
>> >>>>> > iax:guest at conference.freeswitch.org/888
>> >>>>> > googletalk:conf+888 at conference.freeswitch.org
>> >>>>> > pstn:+19193869900
>> >>>>> >
>> >>>>> > _______________________________________________
>> >>>>> > FreeSWITCH-users mailing list
>> >>>>> > FreeSWITCH-users at lists.freeswitch.org
>> >>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>>> >
>> >>>>> >
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>>>> > http://www.freeswitch.org
>> >>>>> >
>> >>>>> >
>> >>>>>
>> >>>>> _______________________________________________
>> >>>>> FreeSWITCH-users mailing list
>> >>>>> FreeSWITCH-users at lists.freeswitch.org
>> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>>>
>> >>>>>
>> >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>>>> http://www.freeswitch.org
>> >>>>
>> >>>>
>> >>>>
>> >>>> --
>> >>>> Anthony Minessale II
>> >>>>
>> >>>> FreeSWITCH http://www.freeswitch.org/
>> >>>> ClueCon http://www.cluecon.com/
>> >>>> Twitter: http://twitter.com/FreeSWITCH_wire
>> >>>>
>> >>>> AIM: anthm
>> >>>> MSN:anthony_minessale at hotmail.com
>> >>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> >>>> IRC: irc.freenode.net #freeswitch
>> >>>>
>> >>>> FreeSWITCH Developer Conference
>> >>>> sip:888 at conference.freeswitch.org
>> >>>> iax:guest at conference.freeswitch.org/888
>> >>>> googletalk:conf+888 at conference.freeswitch.org
>> >>>> pstn:+19193869900
>> >>>
>> >>>
>> >>>
>> >>> --
>> >>> Anthony Minessale II
>> >>>
>> >>> FreeSWITCH http://www.freeswitch.org/
>> >>> ClueCon http://www.cluecon.com/
>> >>> Twitter: http://twitter.com/FreeSWITCH_wire
>> >>>
>> >>> AIM: anthm
>> >>> MSN:anthony_minessale at hotmail.com
>> >>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> >>> IRC: irc.freenode.net #freeswitch
>> >>>
>> >>> FreeSWITCH Developer Conference
>> >>> sip:888 at conference.freeswitch.org
>> >>> iax:guest at conference.freeswitch.org/888
>> >>> googletalk:conf+888 at conference.freeswitch.org
>> >>> pstn:+19193869900
>> >>>
>> >>> _______________________________________________
>> >>> FreeSWITCH-users mailing list
>> >>> FreeSWITCH-users at lists.freeswitch.org
>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>
>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> http://www.freeswitch.org
>> >>>
>> >>>
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>> > --
>> > Anthony Minessale II
>> >
>> > FreeSWITCH http://www.freeswitch.org/
>> > ClueCon http://www.cluecon.com/
>> > Twitter: http://twitter.com/FreeSWITCH_wire
>> >
>> > AIM: anthm
>> > MSN:anthony_minessale at hotmail.com
>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> > IRC: irc.freenode.net #freeswitch
>> >
>> > FreeSWITCH Developer Conference
>> > sip:888 at conference.freeswitch.org
>> > iax:guest at conference.freeswitch.org/888
>> > googletalk:conf+888 at conference.freeswitch.org
>> > pstn:+19193869900
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From tculjaga at gmail.com  Tue Feb 16 00:10:30 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 16 Feb 2010 09:10:30 +0100
Subject: [Freeswitch-users] DTMF problem
In-Reply-To: <1265922753047-4557446.post@n2.nabble.com>
References: <89c9bbf81002111209v6e2f36b3x86a628c5e1390599@mail.gmail.com>
	<F960CC45-C9D2-4150-A108-C0E8B230284D@freeswitch.org>
	<1265922753047-4557446.post@n2.nabble.com>
Message-ID: <65d96fc81002160010j798353b4y5d38894e774b85e0@mail.gmail.com>

what DTMF method are you using ?

InBand as pure voice or by rfc2833?


if it is as pure voice than you will need this "<action
application="start_dtmf" />" in FS dialplan receiving the call.

If you are using 2833, than you should check your FS config for DTMF method.

T.


On Thu, Feb 11, 2010 at 10:12 PM, maxim.tsvetov <maxim.tsvetov at gmail.com>wrote:

>
> I already added "dtmf-relay rtp-nte" and this doesn't work.
>
> Also I don't have "dtmf-interworking rtp-nte" command in Cisco.
> --
> View this message in context:
> http://n2.nabble.com/DTMF-problem-tp4557122p4557446.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100216/7595082c/attachment.html 

From kond at nstel.ru  Tue Feb 16 01:08:39 2010
From: kond at nstel.ru (Nikolay Kondratyev)
Date: Tue, 16 Feb 2010 12:08:39 +0300
Subject: [Freeswitch-users] h323 - sip call is not working
In-Reply-To: <65d96fc81002151202r4fb18b6dma3cfb34fc98adc23@mail.gmail.com>
Message-ID: <20100216090836.C841011FDD@mail.nstel.ru>

Tihomir,

Thanks a lot, I recompiled h323plus libs as you told me and rbt and voice is
ok now.

Thanks again,

Nikolay.

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Tihomir
Culjaga
Sent: Monday, February 15, 2010 11:02 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] h323 - sip call is not working

 

ok, than try this:


edit h323plus/src/h323caps.cxx, grep it for "H323AudioCapability(240, 30) //
240ms max, 30ms desired" ... it should be at line 2598.... 


replace 30 with 20, recompile (make && make install) make sure you use the
new compiled library and start FS.

let me know if you still have audio issues.




-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100216/648dbf2d/attachment.html 

From yehavi.bourvine at gmail.com  Tue Feb 16 01:09:03 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Tue, 16 Feb 2010 11:09:03 +0200
Subject: [Freeswitch-users] DTMF problem
In-Reply-To: <65d96fc81002160010j798353b4y5d38894e774b85e0@mail.gmail.com>
References: <89c9bbf81002111209v6e2f36b3x86a628c5e1390599@mail.gmail.com>
	<F960CC45-C9D2-4150-A108-C0E8B230284D@freeswitch.org>
	<1265922753047-4557446.post@n2.nabble.com>
	<65d96fc81002160010j798353b4y5d38894e774b85e0@mail.gmail.com>
Message-ID: <d361ca7b1002160109r539f98dfra29669a8d7c28e60@mail.gmail.com>

I have a similar problem. In debug mode I see that the Cisco decodes the
DTMFs but does *not* send them to the PSTN.

The only way I managed to make it working is setting the DTMF mode to be
INFO on Freeswitch. This is problematic when you have both phones and the
gateway on the same profile, and the phones use RFC-2833. The solution is to
have a separate profile for the gateway.

                      Regards, __Yehavi:

 2010/2/16 Tihomir Culjaga <tculjaga at gmail.com>

> what DTMF method are you using ?
>
> InBand as pure voice or by rfc2833?
>
>
> if it is as pure voice than you will need this "<action
> application="start_dtmf" />" in FS dialplan receiving the call.
>
> If you are using 2833, than you should check your FS config for DTMF
> method.
>
> T.
>
>
>
> On Thu, Feb 11, 2010 at 10:12 PM, maxim.tsvetov <maxim.tsvetov at gmail.com>wrote:
>
>>
>> I already added "dtmf-relay rtp-nte" and this doesn't work.
>>
>> Also I don't have "dtmf-interworking rtp-nte" command in Cisco.
>> --
>> View this message in context:
>> http://n2.nabble.com/DTMF-problem-tp4557122p4557446.html
>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>
>> _______________________________________________
>>  FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100216/21ffa450/attachment.html 

From devel at thom.fr.eu.org  Tue Feb 16 01:33:44 2010
From: devel at thom.fr.eu.org (=?UTF-8?Q?Fran=C3=A7ois_Legal?=)
Date: Tue, 16 Feb 2010 10:33:44 +0100
Subject: [Freeswitch-users] Sending message notifications with openzap
Message-ID: <e749d5a76cc31b4a3bb2459aaf8b78d9@thom.fr.eu.org>



Hello, 

I was wondering whether or not a facility was available to send
message waiting indicator (from mpd_voicemail) using openzap. 

I know this
feature is not available in openzap (I'm in the process of coding it for
analog channels) but as message indication can be sent via mod_sofia, I
wonder if a frame is available in FS core. 

Anybody can shed me some light
? 

Fran?ois
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100216/754ff240/attachment.html 

From tculjaga at gmail.com  Tue Feb 16 01:39:08 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 16 Feb 2010 10:39:08 +0100
Subject: [Freeswitch-users] h323 - sip call is not working
In-Reply-To: <20100216090836.C841011FDD@mail.nstel.ru>
References: <65d96fc81002151202r4fb18b6dma3cfb34fc98adc23@mail.gmail.com>
	<20100216090836.C841011FDD@mail.nstel.ru>
Message-ID: <65d96fc81002160139x7d002c5al6292e78539b5094c@mail.gmail.com>

On Tue, Feb 16, 2010 at 10:08 AM, Nikolay Kondratyev <kond at nstel.ru> wrote:

>  Tihomir,
>
> Thanks a lot, I recompiled h323plus libs as you told me and rbt and voice
> is ok now.
>

I was afraid so, we will need to work on mod_h323 to allow async codec
framing.


> Thanks again,
>
> Nikolay.
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100216/54b4e03e/attachment.html 

From maxim.tsvetov at gmail.com  Tue Feb 16 02:27:03 2010
From: maxim.tsvetov at gmail.com (maxim.tsvetov)
Date: Tue, 16 Feb 2010 02:27:03 -0800 (PST)
Subject: [Freeswitch-users] DTMF problem
In-Reply-To: <65d96fc81002160010j798353b4y5d38894e774b85e0@mail.gmail.com>
References: <89c9bbf81002111209v6e2f36b3x86a628c5e1390599@mail.gmail.com>
	<F960CC45-C9D2-4150-A108-C0E8B230284D@freeswitch.org>
	<1265922753047-4557446.post@n2.nabble.com>
	<65d96fc81002160010j798353b4y5d38894e774b85e0@mail.gmail.com>
Message-ID: <89c9bbf81002160226u61df2a95j695fcd960f9b2449@mail.gmail.com>


I tried both inband with "start_dtmf" and rfc2833. They are not working.

Maybe there is a method how can I check whether DTMF tones come to FS server
or not ?
(like logs or sniffer)

On Tue, Feb 16, 2010 at 11:18 AM, Tihomir Culjaga [via freeswitch-users] <
ml-node+4579097-1272436358 at n2.nabble.com<ml-node%2B4579097-1272436358 at n2.nabble.com>
> wrote:

> what DTMF method are you using ?
>
> InBand as pure voice or by rfc2833?
>
>
> if it is as pure voice than you will need this "<action
> application="start_dtmf" />" in FS dialplan receiving the call.
>
> If you are using 2833, than you should check your FS config for DTMF
> method.
>
> T.
>
>
> On Thu, Feb 11, 2010 at 10:12 PM, maxim.tsvetov <[hidden email]<http://n2.nabble.com/user/SendEmail.jtp?type=node&node=4579097&i=0>
> > wrote:
>
>>
>> I already added "dtmf-relay rtp-nte" and this doesn't work.
>>
>> Also I don't have "dtmf-interworking rtp-nte" command in Cisco.
>> --
>> View this message in context:
>> http://n2.nabble.com/DTMF-problem-tp4557122p4557446.html
>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> [hidden email]<http://n2.nabble.com/user/SendEmail.jtp?type=node&node=4579097&i=1>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> [hidden email]<http://n2.nabble.com/user/SendEmail.jtp?type=node&node=4579097&i=2>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> ------------------------------
>  View message @ http://n2.nabble.com/DTMF-problem-tp4557122p4579097.html
> To unsubscribe from Re: DTMF problem, click here< (link removed) >.
>
>
>

-- 
View this message in context: http://n2.nabble.com/DTMF-problem-tp4557122p4579539.html
Sent from the freeswitch-users mailing list archive at Nabble.com.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100216/01a50173/attachment-0001.html 

From maxim.tsvetov at gmail.com  Tue Feb 16 02:32:26 2010
From: maxim.tsvetov at gmail.com (maxim.tsvetov)
Date: Tue, 16 Feb 2010 02:32:26 -0800 (PST)
Subject: [Freeswitch-users] DTMF problem
In-Reply-To: <d361ca7b1002160109r539f98dfra29669a8d7c28e60@mail.gmail.com>
References: <89c9bbf81002111209v6e2f36b3x86a628c5e1390599@mail.gmail.com>
	<F960CC45-C9D2-4150-A108-C0E8B230284D@freeswitch.org>
	<1265922753047-4557446.post@n2.nabble.com>
	<65d96fc81002160010j798353b4y5d38894e774b85e0@mail.gmail.com>
	<d361ca7b1002160109r539f98dfra29669a8d7c28e60@mail.gmail.com>
Message-ID: <1266316346819-4579554.post@n2.nabble.com>


Could you please send me example of your FS and Cisco config?

Regards,
Maxim Tsvetov
-- 
View this message in context: http://n2.nabble.com/DTMF-problem-tp4557122p4579554.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From tculjaga at gmail.com  Tue Feb 16 05:22:43 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 16 Feb 2010 14:22:43 +0100
Subject: [Freeswitch-users] DTMF problem
In-Reply-To: <1266316346819-4579554.post@n2.nabble.com>
References: <89c9bbf81002111209v6e2f36b3x86a628c5e1390599@mail.gmail.com>
	<F960CC45-C9D2-4150-A108-C0E8B230284D@freeswitch.org>
	<1265922753047-4557446.post@n2.nabble.com>
	<65d96fc81002160010j798353b4y5d38894e774b85e0@mail.gmail.com>
	<d361ca7b1002160109r539f98dfra29669a8d7c28e60@mail.gmail.com>
	<1266316346819-4579554.post@n2.nabble.com>
Message-ID: <65d96fc81002160522y14ae88eex810928cadf496fe7@mail.gmail.com>

Please send me a wireshark sniff taken on FS for the call establishment and
DTMF transmition.

not filtered please!

T.

On Tue, Feb 16, 2010 at 11:32 AM, maxim.tsvetov <maxim.tsvetov at gmail.com>wrote:

>
> Could you please send me example of your FS and Cisco config?
>
> Regards,
> Maxim Tsvetov
> --
> View this message in context:
> http://n2.nabble.com/DTMF-problem-tp4557122p4579554.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100216/2862a3f5/attachment.html 

From maxim.tsvetov at gmail.com  Tue Feb 16 05:54:22 2010
From: maxim.tsvetov at gmail.com (maxim.tsvetov)
Date: Tue, 16 Feb 2010 05:54:22 -0800 (PST)
Subject: [Freeswitch-users] DTMF problem
In-Reply-To: <1266316346819-4579554.post@n2.nabble.com>
References: <89c9bbf81002111209v6e2f36b3x86a628c5e1390599@mail.gmail.com>
	<F960CC45-C9D2-4150-A108-C0E8B230284D@freeswitch.org>
	<1265922753047-4557446.post@n2.nabble.com>
	<65d96fc81002160010j798353b4y5d38894e774b85e0@mail.gmail.com>
	<d361ca7b1002160109r539f98dfra29669a8d7c28e60@mail.gmail.com>
	<1266316346819-4579554.post@n2.nabble.com>
Message-ID: <1266328462314-4580309.post@n2.nabble.com>


http://n2.nabble.com/file/n4580309/fs.pcap fs.pcap 
-- 
View this message in context: http://n2.nabble.com/DTMF-problem-tp4557122p4580309.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From maxim.tsvetov at gmail.com  Tue Feb 16 05:55:06 2010
From: maxim.tsvetov at gmail.com (maxim.tsvetov)
Date: Tue, 16 Feb 2010 05:55:06 -0800 (PST)
Subject: [Freeswitch-users] DTMF problem
In-Reply-To: <65d96fc81002160522y14ae88eex810928cadf496fe7@mail.gmail.com>
References: <89c9bbf81002111209v6e2f36b3x86a628c5e1390599@mail.gmail.com>
	<F960CC45-C9D2-4150-A108-C0E8B230284D@freeswitch.org>
	<1265922753047-4557446.post@n2.nabble.com>
	<65d96fc81002160010j798353b4y5d38894e774b85e0@mail.gmail.com>
	<d361ca7b1002160109r539f98dfra29669a8d7c28e60@mail.gmail.com>
	<1266316346819-4579554.post@n2.nabble.com>
	<65d96fc81002160522y14ae88eex810928cadf496fe7@mail.gmail.com>
Message-ID: <1266328506892-4580313.post@n2.nabble.com>


http://n2.nabble.com/file/n4580313/fs.pcap fs.pcap 
-- 
View this message in context: http://n2.nabble.com/DTMF-problem-tp4557122p4580313.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From kond at nstel.ru  Tue Feb 16 06:16:34 2010
From: kond at nstel.ru (Nikolay Kondratyev)
Date: Tue, 16 Feb 2010 17:16:34 +0300
Subject: [Freeswitch-users] How to tie context to a gateway?
Message-ID: <20100216141634.75B3511FC6@mail.nstel.ru>

Hi all,

I have several gateways in the external profile.

Let's say GW1 and GW2. I'd like to process calls from the GW1 in the context
C1 and calls from GW2 in the context C2.

Parameter "context", as far as I understand works for the whole profile, not
for individual gateways in the profile.

How do send calls from GW1 into context C1?

What will be a good practice to do that?

Thanks in advance,

Nikolay.

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100216/41af1172/attachment.html 

From tculjaga at gmail.com  Tue Feb 16 06:27:17 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 16 Feb 2010 15:27:17 +0100
Subject: [Freeswitch-users] DTMF problem
In-Reply-To: <1266328506892-4580313.post@n2.nabble.com>
References: <89c9bbf81002111209v6e2f36b3x86a628c5e1390599@mail.gmail.com>
	<F960CC45-C9D2-4150-A108-C0E8B230284D@freeswitch.org>
	<1265922753047-4557446.post@n2.nabble.com>
	<65d96fc81002160010j798353b4y5d38894e774b85e0@mail.gmail.com>
	<d361ca7b1002160109r539f98dfra29669a8d7c28e60@mail.gmail.com>
	<1266316346819-4579554.post@n2.nabble.com>
	<65d96fc81002160522y14ae88eex810928cadf496fe7@mail.gmail.com>
	<1266328506892-4580313.post@n2.nabble.com>
Message-ID: <65d96fc81002160627o79869d84mee3773b34d993d4d@mail.gmail.com>

On Tue, Feb 16, 2010 at 2:55 PM, maxim.tsvetov <maxim.tsvetov at gmail.com>wrote:

>
> http://n2.nabble.com/file/n4580313/fs.pcap fs.pcap
>

the call is not even established so no DTMF can be exchanged :)
please establish a call and send me the sniff.

T.



> --
> View this message in context:
> http://n2.nabble.com/DTMF-problem-tp4557122p4580313.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100216/31299b1e/attachment.html 

From anthony.minessale at gmail.com  Tue Feb 16 06:41:14 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 16 Feb 2010 08:41:14 -0600
Subject: [Freeswitch-users] Broadsoft SCA w/ Cisco SPA-5xx series
In-Reply-To: <8b1c9cda1002152225i7700bdfcgd2c112bf8180e36@mail.gmail.com>
References: <8b1c9cda1002141103te523643ob16db25e3bfa877f@mail.gmail.com>
	<191c3a031002151104u7413b17dh84e68ab1cf703cb9@mail.gmail.com>
	<8b1c9cda1002151147l1e072d00y69c0022e3924c3b4@mail.gmail.com>
	<0dd701caae7e$3c108b70$b431a250$@com>
	<8b1c9cda1002151253t32c22663wf66245e0220a4e6@mail.gmail.com>
	<191c3a031002151304h54c262c5o8ffd5c195e0512d2@mail.gmail.com>
	<0e7301caae84$58b47ce0$0a1d76a0$@com>
	<8b1c9cda1002151456s5dec6dbbtaa2063e0f70fe5dd@mail.gmail.com>
	<191c3a031002151531k148d8504w15e6f4c6c1e3ccd@mail.gmail.com>
	<8b1c9cda1002152225i7700bdfcgd2c112bf8180e36@mail.gmail.com>
Message-ID: <191c3a031002160641q61cb1c53r38add08fc32f3a5a@mail.gmail.com>

as I expected, you have IP addrs in the table which do not match your domain
name.
the phones behind nat should have your domain name in them same as the local
phones.
And the proxy addr should be set to the ip.

If the IP and the DOMAIN do not match you will get mismatches.
Most people make the false assumption that this is like dns where the ip and
hostname are interchangeable.

We can look at making a patch to force the hostname to always be the right
value in the db like we do for reg possibly.



On Tue, Feb 16, 2010 at 12:25 AM, Gabriel Kuri <gkuri at ieee.org> wrote:

> Yeah, the domain name matches on the internal profile.
>
> Thanks for all your help, I can arrange ssh access tomorrow, today
> just wasn't one of those good days to do so, I've been running in and
> out too much to coordinate it.
>
> Here's the pastebin for the sip_dialogs table while the call is up ...
>
>     http://pastebin.freeswitch.org/12159
>
> Thanks,
> Gabe
>
>
> On Mon, Feb 15, 2010 at 3:31 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
> > Do the domain names match on what the remote phones are using?
> >
> > When the call is active, can you attach to sqlite with the sqlite3 app
> and
> > select * from sip_dialogs
> >
> > sqlite3 /usr/local/freeswitch/db/sofia_reg_internal.db
> >> select * from sip_dialogs;
> >
> > remember to do it while the call is up.
> >
> >
> > I am going to bet the domain name in that table is not the same as your
> > actual domain.
> >
> >
> > I tried to make this easier by asking to ssh to your box and work with
> you
> > to fix it but now 9 hours later its starting to resemble diffusing a bomb
> > over a telegraph wire.
> >
> >
> >
> >
> > On Mon, Feb 15, 2010 at 4:56 PM, Gabriel Kuri <gkuri at ieee.org> wrote:
> >>
> >> Yes, the two phones being called (SPA-509Gs) have SCA enabled and it
> >> works when making outgoing calls from those phones. But incoming calls
> >> to those two phones don't seem to have the line key light up on the
> >> other phone when one of the phones is answered (same extension).
> >>
> >> Thanks,
> >> Gabe
> >>
> >> On Mon, Feb 15, 2010 at 1:17 PM, Peder <peder at networkoblivion.com>
> wrote:
> >> > On the phone itself, do you have the line set to shared and ?Broadsoft
> >> > SCA?
> >> > enabled?
> >> >
> >> >
> >> >
> >> > From: freeswitch-users-bounces at lists.freeswitch.org
> >> > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> >> > Anthony
> >> > Minessale
> >> > Sent: Monday, February 15, 2010 3:04 PM
> >> > To: freeswitch-users at lists.freeswitch.org
> >> > Subject: Re: [Freeswitch-users] Broadsoft SCA w/ Cisco SPA-5xx series
> >> >
> >> >
> >> >
> >> > you are missing something because you have no seize events when you go
> >> > on
> >> > and off hook.
> >> > is every phone in the correct mode?
> >> >
> >> > On Mon, Feb 15, 2010 at 2:53 PM, Gabriel Kuri <gkuri at ieee.org> wrote:
> >> >
> >> > No, that was a typo. I have it correct in the config file.
> >> >
> >> > Gabe
> >> >
> >> > On Mon, Feb 15, 2010 at 12:34 PM, Peder <peder at networkoblivion.com>
> >> > wrote:
> >> >
> >> >> Is this a typo "managed-shared-appeareance=true" or is there an extra
> e
> >> >> in
> >> >> appearance in your config?
> >> >>
> >> >> -----Original Message-----
> >> >> From: freeswitch-users-bounces at lists.freeswitch.org
> >> >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> >> >> Gabriel
> >> >> Kuri
> >> >> Sent: Monday, February 15, 2010 1:48 PM
> >> >> To: freeswitch-users at lists.freeswitch.org
> >> >> Subject: Re: [Freeswitch-users] Broadsoft SCA w/ Cisco SPA-5xx series
> >> >>
> >> >> OK, I don't know what happened there, here's another one with the
> >> >> NOTIFYs. I'm on trunk rev 16633 and I have
> >> >> "managed-shared-appeareance=true" on the internal profile. I'm just
> >> >> making calls between internal phones.
> >> >>
> >> >>     http://pastebin.freeswitch.org/12153
> >> >>
> >> >> Thanks,
> >> >> Gabe
> >> >>
> >> >> On Mon, Feb 15, 2010 at 11:04 AM, Anthony Minessale
> >> >> <anthony.minessale at gmail.com> wrote:
> >> >>> I don't see any notifies at all in this trace do the profiles in
> >> >>> question
> >> >>> have:
> >> >>> manage-shared-appearance set to true?
> >> >>> and are you on latest trunk?
> >> >>>
> >> >>>
> >> >>> On Mon, Feb 15, 2010 at 12:55 PM, Anthony Minessale
> >> >>> <anthony.minessale at gmail.com> wrote:
> >> >>>>
> >> >>>> we log the sql stmts on err so they are red and easier to read.
> >> >>>>
> >> >>>>
> >> >>>>
> >> >>>> On Mon, Feb 15, 2010 at 12:48 PM, Gabriel Kuri <gkuri at ieee.org>
> >> >>>> wrote:
> >> >>>>>
> >> >>>>> Here's a call from ext 2551 to 2552 (two phones). I noticed a
> bunch
> >> >>>>> of
> >> >>>>> errors related to SQL UPDATE for presence ...
> >> >>>>>
> >> >>>>>     http://pastebin.freeswitch.org/12152
> >> >>>>>
> >> >>>>> Thanks,
> >> >>>>> Gabe
> >> >>>>>
> >> >>>>>
> >> >>>>> On Mon, Feb 15, 2010 at 8:32 AM, Anthony Minessale
> >> >>>>> <anthony.minessale at gmail.com> wrote:
> >> >>>>> > it should be active not seized.
> >> >>>>> > seized is when you take it off hook.
> >> >>>>> >
> >> >>>>> > We need some more debugging to be sure.
> >> >>>>> > Can we work in real time on it or can you get a more detailed
> log?
> >> >>>>> >
> >> >>>>> > edit sofia.conf.xml and add the param to the "settings" section.
> >> >>>>> >
> >> >>>>> > <param name="debug-sla" value="100"/>
> >> >>>>> >
> >> >>>>> >
> >> >>>>> > then restart and enable sip trace and debug level
> >> >>>>> >
> >> >>>>> > //do this for every profile involved in the call.
> >> >>>>> > sofia profile <profilename> siptrace on
> >> >>>>> >
> >> >>>>> > //also do this
> >> >>>>> > console loglevel debug
> >> >>>>> >
> >> >>>>> >
> >> >>>>> > if you can let us ssh, we can do all the for you if you can make
> >> >>>>> > the
> >> >>>>> > test
> >> >>>>> > calls.
> >> >>>>> >
> >> >>>>> >
> >> >>>>> >
> >> >>>>> >
> >> >>>>> > On Sun, Feb 14, 2010 at 11:59 PM, Gabriel Kuri <gkuri at ieee.org>
> >> >>>>> > wrote:
> >> >>>>> >>
> >> >>>>> >> BTW, here's a copy of the NOTIFY (event call-info) sent to the
> >> >>>>> >> other
> >> >>>>> >> phone after the first phone is answered, should this have a
> >> >>>>> >> Call-Info
> >> >>>>> >> line with an "appearance-state=seized" to turn on the light on
> >> >>>>> >> the
> >> >>>>> >> other phone?
> >> >>>>> >>
> >> >>>>> >>
> >> >>>>> >> NOTIFY sip:2551@<removed>:54446 SIP/2.0.
> >> >>>>> >> Via: SIP/2.0/UDP
> >> >>>>> >> <removed>:9430;rport;branch=z9hG4bK71pN2cXgH851K.
> >> >>>>> >> Max-Forwards: 70.
> >> >>>>> >> From: <sip:2551 at mydomain.com <sip%3A2551 at mydomain.com>
> >;tag=XeB6ZrKDevpHp.
> >> >>>>> >> To: <sip:2551 at mydomain.com <sip%3A2551 at mydomain.com>
> >;tag=c2d34993aac6ea.
> >> >>>>> >> Call-ID: 34c34987-8b6fa786@<removed>.
> >> >>>>> >> CSeq: 126950830 NOTIFY.
> >> >>>>> >> Contact: <sip:2551@<removed>:9430>.
> >> >>>>> >> Expires: 3959.
> >> >>>>> >> Call-Info:
> >> >>>>> >> <sip:mydomain.com>;appearance-index=*;appearance-state=idle.
> >> >>>>> >> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16633M.
> >> >>>>> >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE,
> INFO,
> >> >>>>> >> REGISTER, REFER, PRACK, NOTIFY, PUBLISH, SUBSCRIBE.
> >> >>>>> >> Supported: 100rel, timer, precondition, path, replaces.
> >> >>>>> >> Event: call-info.
> >> >>>>> >> Allow-Events: talk, presence, dialog, line-seize, call-info,
> sla,
> >> >>>>> >> include-session-description, presence.winfo, message-summary,
> >> >>>>> >> refer.
> >> >>>>> >> Subscription-State: active;expires=3959.
> >> >>>>> >> Content-Length: 0.
> >> >>>>> >>
> >> >>>>> >>
> >> >>>>> >>
> >> >>>>> >> On Sun, Feb 14, 2010 at 7:58 PM, Brian West
> >> >>>>> >> <brian at freeswitch.org>
> >> >>>>> >> wrote:
> >> >>>>> >> > Works fine here... is your box slow or something?
> >> >>>>> >> >
> >> >>>>> >> > /b
> >> >>>>> >> >
> >> >>>>> >> > On Feb 14, 2010, at 1:03 PM, Gabriel Kuri wrote:
> >> >>>>> >> >
> >> >>>>> >> >> I followed Brian's directions from one of the previous
> threads
> >> >>>>> >> >> on
> >> >>>>> >> >> configuring the SPA-5xx series phones for Broadsoft SCA and
> >> >>>>> >> >> set
> >> >>>>> >> >> manage-shared-appearance=true in the internal profile. SCA
> >> >>>>> >> >> appears
> >> >>>>> >> >> to
> >> >>>>> >> >> be working on outgoing calls between two phones, the line
> key
> >> >>>>> >> >> starts
> >> >>>>> >> >> flashing red on the second phone when the first phone picks
> up
> >> >>>>> >> >> the
> >> >>>>> >> >> receiver to make a call. However on incoming calls, both
> >> >>>>> >> >> phones
> >> >>>>> >> >> ring
> >> >>>>> >> >> (same extension), however when one of the phones picks up
> the
> >> >> line,
> >> >>>>> >> >> the second phone's line key doesn't flash red or show the
> >> >>>>> >> >> first
> >> >>>>> >> >> phone
> >> >>>>> >> >> on that incoming call. Any ideas? Does shared appearance
> only
> >> >>>>> >> >> work
> >> >>>>> >> >> on
> >> >>>>> >> >> outgoing phone calls?
> >> >>>>> >> >>
> >> >>>>> >> >> Thanks,
> >> >>>>> >> >> Gabe
> >> >>>>> >> >
> >> >>>>> >> >
> >> >>>>> >> > _______________________________________________
> >> >>>>> >> > FreeSWITCH-users mailing list
> >> >>>>> >> > FreeSWITCH-users at lists.freeswitch.org
> >> >>>>> >> >
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>>>> >> >
> >> >>>>> >> >
> >> >>
> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >>>>> >> > http://www.freeswitch.org
> >> >>>>> >> >
> >> >>>>> >>
> >> >>>>> >> _______________________________________________
> >> >>>>> >> FreeSWITCH-users mailing list
> >> >>>>> >> FreeSWITCH-users at lists.freeswitch.org
> >> >>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>>>> >>
> >> >>>>> >>
> >> >>
> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >>>>> >> http://www.freeswitch.org
> >> >>>>> >
> >> >>>>> >
> >> >>>>> >
> >> >>>>> > --
> >> >>>>> > Anthony Minessale II
> >> >>>>> >
> >> >>>>> > FreeSWITCH http://www.freeswitch.org/
> >> >>>>> > ClueCon http://www.cluecon.com/
> >> >>>>> > Twitter: http://twitter.com/FreeSWITCH_wire
> >> >>>>> >
> >> >>>>> > AIM: anthm
> >> >>>>> > MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
> >> >>>>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >> >>>>> > IRC: irc.freenode.net #freeswitch
> >> >>>>> >
> >> >>>>> > FreeSWITCH Developer Conference
> >> >>>>> > sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> >> >>>>> > iax:guest at conference.freeswitch.org/888
> >> >>>>> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >> >>>>> > pstn:+19193869900
> >> >>>>> >
> >> >>>>> > _______________________________________________
> >> >>>>> > FreeSWITCH-users mailing list
> >> >>>>> > FreeSWITCH-users at lists.freeswitch.org
> >> >>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>>>> >
> >> >>>>> >
> >> >>
> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >>>>> > http://www.freeswitch.org
> >> >>>>> >
> >> >>>>> >
> >> >>>>>
> >> >>>>> _______________________________________________
> >> >>>>> FreeSWITCH-users mailing list
> >> >>>>> FreeSWITCH-users at lists.freeswitch.org
> >> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>>>>
> >> >>>>>
> >> >>>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >>>>> http://www.freeswitch.org
> >> >>>>
> >> >>>>
> >> >>>>
> >> >>>> --
> >> >>>> Anthony Minessale II
> >> >>>>
> >> >>>> FreeSWITCH http://www.freeswitch.org/
> >> >>>> ClueCon http://www.cluecon.com/
> >> >>>> Twitter: http://twitter.com/FreeSWITCH_wire
> >> >>>>
> >> >>>> AIM: anthm
> >> >>>> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
> >> >>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >> >>>> IRC: irc.freenode.net #freeswitch
> >> >>>>
> >> >>>> FreeSWITCH Developer Conference
> >> >>>> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> >> >>>> iax:guest at conference.freeswitch.org/888
> >> >>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >> >>>> pstn:+19193869900
> >> >>>
> >> >>>
> >> >>>
> >> >>> --
> >> >>> Anthony Minessale II
> >> >>>
> >> >>> FreeSWITCH http://www.freeswitch.org/
> >> >>> ClueCon http://www.cluecon.com/
> >> >>> Twitter: http://twitter.com/FreeSWITCH_wire
> >> >>>
> >> >>> AIM: anthm
> >> >>> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
> >> >>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >> >>> IRC: irc.freenode.net #freeswitch
> >> >>>
> >> >>> FreeSWITCH Developer Conference
> >> >>> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> >> >>> iax:guest at conference.freeswitch.org/888
> >> >>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >> >>> pstn:+19193869900
> >> >>>
> >> >>> _______________________________________________
> >> >>> FreeSWITCH-users mailing list
> >> >>> FreeSWITCH-users at lists.freeswitch.org
> >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>>
> >> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >>> http://www.freeswitch.org
> >> >>>
> >> >>>
> >> >>
> >> >> _______________________________________________
> >> >> FreeSWITCH-users mailing list
> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>
> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> http://www.freeswitch.org
> >> >>
> >> >>
> >> >> _______________________________________________
> >> >> FreeSWITCH-users mailing list
> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>
> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> http://www.freeswitch.org
> >> >>
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >> > --
> >> > Anthony Minessale II
> >> >
> >> > FreeSWITCH http://www.freeswitch.org/
> >> > ClueCon http://www.cluecon.com/
> >> > Twitter: http://twitter.com/FreeSWITCH_wire
> >> >
> >> > AIM: anthm
> >> > MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
> >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >> > IRC: irc.freenode.net #freeswitch
> >> >
> >> > FreeSWITCH Developer Conference
> >> > sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> >> > iax:guest at conference.freeswitch.org/888
> >> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >> > pstn:+19193869900
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> > Twitter: http://twitter.com/FreeSWITCH_wire
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> > IRC: irc.freenode.net #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> > iax:guest at conference.freeswitch.org/888
> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > pstn:+19193869900
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100216/0d5c4dcb/attachment-0001.html 

From yehavi.bourvine at gmail.com  Tue Feb 16 06:56:22 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Tue, 16 Feb 2010 16:56:22 +0200
Subject: [Freeswitch-users] SQL ERR: library routine called out of sequence
Message-ID: <d361ca7b1002160656w7d616f6dn5e80087ebde94dbe@mail.gmail.com>

Hello,

  After upgrading from 1.0.5pre10 to 1.0.5-20100216-0400 we started getting
the above errors (I append bellow two samples). It seems Freeswitch fails to
read a database using Sqlite.
Anyone have seen this?

Other details: Fedora 10, SQlite 3.5.9.
We also do SQLite quesries during call setup via LUA from CoreDB. Is it an
SQLite problem?

                         Thanks! __Yehavi:

The samples:
2010-02-16 16:22:13.762679 [ERR] switch_core_sqldb.c:404 SQL ERR [library
routin
e called out of sequence]
delete from sip_dialogs where call_id='
8656841832142-120172129116107 at 10.64.1.2'

2010-02-16 16:42:00.802442 [ERR] switch_core_sqldb.c:722 SQL ERR: [select
call_i
d,sip_user,sip_host,contact,status,rpid,expires,user_agent,server_user,server_ho
st,profile_name,hostname,network_ip,network_port,sip_username,sip_realm,mwi_user
,mwi_host from sip_registrations where profile_name='phones' and contact
like '%
80635%'] library routine called out of sequence
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100216/11d23da8/attachment.html 

From anthony.minessale at gmail.com  Tue Feb 16 07:05:12 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 16 Feb 2010 09:05:12 -0600
Subject: [Freeswitch-users] SQL ERR: library routine called out of
	sequence
In-Reply-To: <d361ca7b1002160656w7d616f6dn5e80087ebde94dbe@mail.gmail.com>
References: <d361ca7b1002160656w7d616f6dn5e80087ebde94dbe@mail.gmail.com>
Message-ID: <191c3a031002160705w4050c3cdk2bf2922e9aff5ba3@mail.gmail.com>

try removing all the .db files from /usr/local/freeswitch/db


On Tue, Feb 16, 2010 at 8:56 AM, Yehavi Bourvine
<yehavi.bourvine at gmail.com>wrote:

> Hello,
>
>   After upgrading from 1.0.5pre10 to 1.0.5-20100216-0400 we started getting
> the above errors (I append bellow two samples). It seems Freeswitch fails to
> read a database using Sqlite.
> Anyone have seen this?
>
> Other details: Fedora 10, SQlite 3.5.9.
> We also do SQLite quesries during call setup via LUA from CoreDB. Is it an
> SQLite problem?
>
>                          Thanks! __Yehavi:
>
> The samples:
> 2010-02-16 16:22:13.762679 [ERR] switch_core_sqldb.c:404 SQL ERR [library
> routin
> e called out of sequence]
> delete from sip_dialogs where call_id='
> 8656841832142-120172129116107 at 10.64.1.2'
>
> 2010-02-16 16:42:00.802442 [ERR] switch_core_sqldb.c:722 SQL ERR: [select
> call_i
>
> d,sip_user,sip_host,contact,status,rpid,expires,user_agent,server_user,server_ho
>
> st,profile_name,hostname,network_ip,network_port,sip_username,sip_realm,mwi_user
> ,mwi_host from sip_registrations where profile_name='phones' and contact
> like '%
> 80635%'] library routine called out of sequence
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100216/5112a388/attachment.html 

From yehavi.bourvine at gmail.com  Tue Feb 16 07:27:50 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Tue, 16 Feb 2010 17:27:50 +0200
Subject: [Freeswitch-users] SQL ERR: library routine called out of
	sequence
In-Reply-To: <191c3a031002160705w4050c3cdk2bf2922e9aff5ba3@mail.gmail.com>
References: <d361ca7b1002160656w7d616f6dn5e80087ebde94dbe@mail.gmail.com>
	<191c3a031002160705w4050c3cdk2bf2922e9aff5ba3@mail.gmail.com>
Message-ID: <d361ca7b1002160727ucb6e0c8id62e803757333213@mail.gmail.com>

Tried this, but it didn't help. I delete these DB files before any upgrade
just to be sure.

                  Thanks! __Yehavi:

 2010/2/16 Anthony Minessale <anthony.minessale at gmail.com>

> try removing all the .db files from /usr/local/freeswitch/db
>
>
>   On Tue, Feb 16, 2010 at 8:56 AM, Yehavi Bourvine <
> yehavi.bourvine at gmail.com> wrote:
>
>>   Hello,
>>
>>   After upgrading from 1.0.5pre10 to 1.0.5-20100216-0400 we started
>> getting the above errors (I append bellow two samples). It seems Freeswitch
>> fails to read a database using Sqlite.
>> Anyone have seen this?
>>
>> Other details: Fedora 10, SQlite 3.5.9.
>> We also do SQLite quesries during call setup via LUA from CoreDB. Is it an
>> SQLite problem?
>>
>>                          Thanks! __Yehavi:
>>
>> The samples:
>> 2010-02-16 16:22:13.762679 [ERR] switch_core_sqldb.c:404 SQL ERR [library
>> routin
>> e called out of sequence]
>> delete from sip_dialogs where call_id='
>> 8656841832142-120172129116107 at 10.64.1.2'
>>
>> 2010-02-16 16:42:00.802442 [ERR] switch_core_sqldb.c:722 SQL ERR: [select
>> call_i
>>
>> d,sip_user,sip_host,contact,status,rpid,expires,user_agent,server_user,server_ho
>>
>> st,profile_name,hostname,network_ip,network_port,sip_username,sip_realm,mwi_user
>> ,mwi_host from sip_registrations where profile_name='phones' and contact
>> like '%
>> 80635%'] library routine called out of sequence
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100216/3ef8c588/attachment.html 

From gkuri at ieee.org  Tue Feb 16 07:38:27 2010
From: gkuri at ieee.org (Gabriel Kuri)
Date: Tue, 16 Feb 2010 07:38:27 -0800
Subject: [Freeswitch-users] Broadsoft SCA w/ Cisco SPA-5xx series
In-Reply-To: <191c3a031002160641q61cb1c53r38add08fc32f3a5a@mail.gmail.com>
References: <8b1c9cda1002141103te523643ob16db25e3bfa877f@mail.gmail.com>
	<8b1c9cda1002151147l1e072d00y69c0022e3924c3b4@mail.gmail.com>
	<0dd701caae7e$3c108b70$b431a250$@com>
	<8b1c9cda1002151253t32c22663wf66245e0220a4e6@mail.gmail.com>
	<191c3a031002151304h54c262c5o8ffd5c195e0512d2@mail.gmail.com>
	<0e7301caae84$58b47ce0$0a1d76a0$@com>
	<8b1c9cda1002151456s5dec6dbbtaa2063e0f70fe5dd@mail.gmail.com>
	<191c3a031002151531k148d8504w15e6f4c6c1e3ccd@mail.gmail.com>
	<8b1c9cda1002152225i7700bdfcgd2c112bf8180e36@mail.gmail.com>
	<191c3a031002160641q61cb1c53r38add08fc32f3a5a@mail.gmail.com>
Message-ID: <8b1c9cda1002160738r619b2a3cs3bf5dd7d1322121e@mail.gmail.com>

The phones are currently setup with the domain in their "Proxy" field
and set to use SRV to lookup the IP. The "Outbound Proxy" field is
left empty. How should the phones be setup? The Proxy field with the
domain and Outbound Proxy set to the IP?

Thanks,
Gabe

On Tue, Feb 16, 2010 at 6:41 AM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> as I expected, you have IP addrs in the table which do not match your domain
> name.
> the phones behind nat should have your domain name in them same as the local
> phones.
> And the proxy addr should be set to the ip.
>
> If the IP and the DOMAIN do not match you will get mismatches.
> Most people make the false assumption that this is like dns where the ip and
> hostname are interchangeable.
>
> We can look at making a patch to force the hostname to always be the right
> value in the db like we do for reg possibly.
>
>
>
> On Tue, Feb 16, 2010 at 12:25 AM, Gabriel Kuri <gkuri at ieee.org> wrote:
>>
>> Yeah, the domain name matches on the internal profile.
>>
>> Thanks for all your help, I can arrange ssh access tomorrow, today
>> just wasn't one of those good days to do so, I've been running in and
>> out too much to coordinate it.
>>
>> Here's the pastebin for the sip_dialogs table while the call is up ...
>>
>> ? ? http://pastebin.freeswitch.org/12159
>>
>> Thanks,
>> Gabe
>>
>>
>> On Mon, Feb 15, 2010 at 3:31 PM, Anthony Minessale
>> <anthony.minessale at gmail.com> wrote:
>> > Do the domain names match on what the remote phones are using?
>> >
>> > When the call is active, can you attach to sqlite with the sqlite3 app
>> > and
>> > select * from sip_dialogs
>> >
>> > sqlite3 /usr/local/freeswitch/db/sofia_reg_internal.db
>> >> select * from sip_dialogs;
>> >
>> > remember to do it while the call is up.
>> >
>> >
>> > I am going to bet the domain name in that table is not the same as your
>> > actual domain.
>> >
>> >
>> > I tried to make this easier by asking to ssh to your box and work with
>> > you
>> > to fix it but now 9 hours later its starting to resemble diffusing a
>> > bomb
>> > over a telegraph wire.
>> >
>> >
>> >
>> >
>> > On Mon, Feb 15, 2010 at 4:56 PM, Gabriel Kuri <gkuri at ieee.org> wrote:
>> >>
>> >> Yes, the two phones being called (SPA-509Gs) have SCA enabled and it
>> >> works when making outgoing calls from those phones. But incoming calls
>> >> to those two phones don't seem to have the line key light up on the
>> >> other phone when one of the phones is answered (same extension).
>> >>
>> >> Thanks,
>> >> Gabe
>> >>
>> >> On Mon, Feb 15, 2010 at 1:17 PM, Peder <peder at networkoblivion.com>
>> >> wrote:
>> >> > On the phone itself, do you have the line set to shared and
>> >> > ?Broadsoft
>> >> > SCA?
>> >> > enabled?
>> >> >
>> >> >
>> >> >
>> >> > From: freeswitch-users-bounces at lists.freeswitch.org
>> >> > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
>> >> > Anthony
>> >> > Minessale
>> >> > Sent: Monday, February 15, 2010 3:04 PM
>> >> > To: freeswitch-users at lists.freeswitch.org
>> >> > Subject: Re: [Freeswitch-users] Broadsoft SCA w/ Cisco SPA-5xx series
>> >> >
>> >> >
>> >> >
>> >> > you are missing something because you have no seize events when you
>> >> > go
>> >> > on
>> >> > and off hook.
>> >> > is every phone in the correct mode?
>> >> >
>> >> > On Mon, Feb 15, 2010 at 2:53 PM, Gabriel Kuri <gkuri at ieee.org> wrote:
>> >> >
>> >> > No, that was a typo. I have it correct in the config file.
>> >> >
>> >> > Gabe
>> >> >
>> >> > On Mon, Feb 15, 2010 at 12:34 PM, Peder <peder at networkoblivion.com>
>> >> > wrote:
>> >> >
>> >> >> Is this a typo "managed-shared-appeareance=true" or is there an
>> >> >> extra e
>> >> >> in
>> >> >> appearance in your config?
>> >> >>
>> >> >> -----Original Message-----
>> >> >> From: freeswitch-users-bounces at lists.freeswitch.org
>> >> >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
>> >> >> Gabriel
>> >> >> Kuri
>> >> >> Sent: Monday, February 15, 2010 1:48 PM
>> >> >> To: freeswitch-users at lists.freeswitch.org
>> >> >> Subject: Re: [Freeswitch-users] Broadsoft SCA w/ Cisco SPA-5xx
>> >> >> series
>> >> >>
>> >> >> OK, I don't know what happened there, here's another one with the
>> >> >> NOTIFYs. I'm on trunk rev 16633 and I have
>> >> >> "managed-shared-appeareance=true" on the internal profile. I'm just
>> >> >> making calls between internal phones.
>> >> >>
>> >> >> ? ? http://pastebin.freeswitch.org/12153
>> >> >>
>> >> >> Thanks,
>> >> >> Gabe
>> >> >>
>> >> >> On Mon, Feb 15, 2010 at 11:04 AM, Anthony Minessale
>> >> >> <anthony.minessale at gmail.com> wrote:
>> >> >>> I don't see any notifies at all in this trace do the profiles in
>> >> >>> question
>> >> >>> have:
>> >> >>> manage-shared-appearance set to true?
>> >> >>> and are you on latest trunk?
>> >> >>>
>> >> >>>
>> >> >>> On Mon, Feb 15, 2010 at 12:55 PM, Anthony Minessale
>> >> >>> <anthony.minessale at gmail.com> wrote:
>> >> >>>>
>> >> >>>> we log the sql stmts on err so they are red and easier to read.
>> >> >>>>
>> >> >>>>
>> >> >>>>
>> >> >>>> On Mon, Feb 15, 2010 at 12:48 PM, Gabriel Kuri <gkuri at ieee.org>
>> >> >>>> wrote:
>> >> >>>>>
>> >> >>>>> Here's a call from ext 2551 to 2552 (two phones). I noticed a
>> >> >>>>> bunch
>> >> >>>>> of
>> >> >>>>> errors related to SQL UPDATE for presence ...
>> >> >>>>>
>> >> >>>>> ? ? http://pastebin.freeswitch.org/12152
>> >> >>>>>
>> >> >>>>> Thanks,
>> >> >>>>> Gabe
>> >> >>>>>
>> >> >>>>>
>> >> >>>>> On Mon, Feb 15, 2010 at 8:32 AM, Anthony Minessale
>> >> >>>>> <anthony.minessale at gmail.com> wrote:
>> >> >>>>> > it should be active not seized.
>> >> >>>>> > seized is when you take it off hook.
>> >> >>>>> >
>> >> >>>>> > We need some more debugging to be sure.
>> >> >>>>> > Can we work in real time on it or can you get a more detailed
>> >> >>>>> > log?
>> >> >>>>> >
>> >> >>>>> > edit sofia.conf.xml and add the param to the "settings"
>> >> >>>>> > section.
>> >> >>>>> >
>> >> >>>>> > <param name="debug-sla" value="100"/>
>> >> >>>>> >
>> >> >>>>> >
>> >> >>>>> > then restart and enable sip trace and debug level
>> >> >>>>> >
>> >> >>>>> > //do this for every profile involved in the call.
>> >> >>>>> > sofia profile <profilename> siptrace on
>> >> >>>>> >
>> >> >>>>> > //also do this
>> >> >>>>> > console loglevel debug
>> >> >>>>> >
>> >> >>>>> >
>> >> >>>>> > if you can let us ssh, we can do all the for you if you can
>> >> >>>>> > make
>> >> >>>>> > the
>> >> >>>>> > test
>> >> >>>>> > calls.
>> >> >>>>> >
>> >> >>>>> >
>> >> >>>>> >
>> >> >>>>> >
>> >> >>>>> > On Sun, Feb 14, 2010 at 11:59 PM, Gabriel Kuri <gkuri at ieee.org>
>> >> >>>>> > wrote:
>> >> >>>>> >>
>> >> >>>>> >> BTW, here's a copy of the NOTIFY (event call-info) sent to the
>> >> >>>>> >> other
>> >> >>>>> >> phone after the first phone is answered, should this have a
>> >> >>>>> >> Call-Info
>> >> >>>>> >> line with an "appearance-state=seized" to turn on the light on
>> >> >>>>> >> the
>> >> >>>>> >> other phone?
>> >> >>>>> >>
>> >> >>>>> >>
>> >> >>>>> >> NOTIFY sip:2551@<removed>:54446 SIP/2.0.
>> >> >>>>> >> Via: SIP/2.0/UDP
>> >> >>>>> >> <removed>:9430;rport;branch=z9hG4bK71pN2cXgH851K.
>> >> >>>>> >> Max-Forwards: 70.
>> >> >>>>> >> From: <sip:2551 at mydomain.com>;tag=XeB6ZrKDevpHp.
>> >> >>>>> >> To: <sip:2551 at mydomain.com>;tag=c2d34993aac6ea.
>> >> >>>>> >> Call-ID: 34c34987-8b6fa786@<removed>.
>> >> >>>>> >> CSeq: 126950830 NOTIFY.
>> >> >>>>> >> Contact: <sip:2551@<removed>:9430>.
>> >> >>>>> >> Expires: 3959.
>> >> >>>>> >> Call-Info:
>> >> >>>>> >> <sip:mydomain.com>;appearance-index=*;appearance-state=idle.
>> >> >>>>> >> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16633M.
>> >> >>>>> >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE,
>> >> >>>>> >> INFO,
>> >> >>>>> >> REGISTER, REFER, PRACK, NOTIFY, PUBLISH, SUBSCRIBE.
>> >> >>>>> >> Supported: 100rel, timer, precondition, path, replaces.
>> >> >>>>> >> Event: call-info.
>> >> >>>>> >> Allow-Events: talk, presence, dialog, line-seize, call-info,
>> >> >>>>> >> sla,
>> >> >>>>> >> include-session-description, presence.winfo, message-summary,
>> >> >>>>> >> refer.
>> >> >>>>> >> Subscription-State: active;expires=3959.
>> >> >>>>> >> Content-Length: 0.
>> >> >>>>> >>
>> >> >>>>> >>
>> >> >>>>> >>
>> >> >>>>> >> On Sun, Feb 14, 2010 at 7:58 PM, Brian West
>> >> >>>>> >> <brian at freeswitch.org>
>> >> >>>>> >> wrote:
>> >> >>>>> >> > Works fine here... is your box slow or something?
>> >> >>>>> >> >
>> >> >>>>> >> > /b
>> >> >>>>> >> >
>> >> >>>>> >> > On Feb 14, 2010, at 1:03 PM, Gabriel Kuri wrote:
>> >> >>>>> >> >
>> >> >>>>> >> >> I followed Brian's directions from one of the previous
>> >> >>>>> >> >> threads
>> >> >>>>> >> >> on
>> >> >>>>> >> >> configuring the SPA-5xx series phones for Broadsoft SCA and
>> >> >>>>> >> >> set
>> >> >>>>> >> >> manage-shared-appearance=true in the internal profile. SCA
>> >> >>>>> >> >> appears
>> >> >>>>> >> >> to
>> >> >>>>> >> >> be working on outgoing calls between two phones, the line
>> >> >>>>> >> >> key
>> >> >>>>> >> >> starts
>> >> >>>>> >> >> flashing red on the second phone when the first phone picks
>> >> >>>>> >> >> up
>> >> >>>>> >> >> the
>> >> >>>>> >> >> receiver to make a call. However on incoming calls, both
>> >> >>>>> >> >> phones
>> >> >>>>> >> >> ring
>> >> >>>>> >> >> (same extension), however when one of the phones picks up
>> >> >>>>> >> >> the
>> >> >> line,
>> >> >>>>> >> >> the second phone's line key doesn't flash red or show the
>> >> >>>>> >> >> first
>> >> >>>>> >> >> phone
>> >> >>>>> >> >> on that incoming call. Any ideas? Does shared appearance
>> >> >>>>> >> >> only
>> >> >>>>> >> >> work
>> >> >>>>> >> >> on
>> >> >>>>> >> >> outgoing phone calls?
>> >> >>>>> >> >>
>> >> >>>>> >> >> Thanks,
>> >> >>>>> >> >> Gabe
>> >> >>>>> >> >
>> >> >>>>> >> >
>> >> >>>>> >> > _______________________________________________
>> >> >>>>> >> > FreeSWITCH-users mailing list
>> >> >>>>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> >>>>> >> >
>> >> >>>>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>>>> >> >
>> >> >>>>> >> >
>> >> >>
>> >> >>
>> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >>>>> >> > http://www.freeswitch.org
>> >> >>>>> >> >
>> >> >>>>> >>
>> >> >>>>> >> _______________________________________________
>> >> >>>>> >> FreeSWITCH-users mailing list
>> >> >>>>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> >>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>>>> >>
>> >> >>>>> >>
>> >> >>
>> >> >>
>> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >>>>> >> http://www.freeswitch.org
>> >> >>>>> >
>> >> >>>>> >
>> >> >>>>> >
>> >> >>>>> > --
>> >> >>>>> > Anthony Minessale II
>> >> >>>>> >
>> >> >>>>> > FreeSWITCH http://www.freeswitch.org/
>> >> >>>>> > ClueCon http://www.cluecon.com/
>> >> >>>>> > Twitter: http://twitter.com/FreeSWITCH_wire
>> >> >>>>> >
>> >> >>>>> > AIM: anthm
>> >> >>>>> > MSN:anthony_minessale at hotmail.com
>> >> >>>>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> >> >>>>> > IRC: irc.freenode.net #freeswitch
>> >> >>>>> >
>> >> >>>>> > FreeSWITCH Developer Conference
>> >> >>>>> > sip:888 at conference.freeswitch.org
>> >> >>>>> > iax:guest at conference.freeswitch.org/888
>> >> >>>>> > googletalk:conf+888 at conference.freeswitch.org
>> >> >>>>> > pstn:+19193869900
>> >> >>>>> >
>> >> >>>>> > _______________________________________________
>> >> >>>>> > FreeSWITCH-users mailing list
>> >> >>>>> > FreeSWITCH-users at lists.freeswitch.org
>> >> >>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>>>> >
>> >> >>>>> >
>> >> >>
>> >> >>
>> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >>>>> > http://www.freeswitch.org
>> >> >>>>> >
>> >> >>>>> >
>> >> >>>>>
>> >> >>>>> _______________________________________________
>> >> >>>>> FreeSWITCH-users mailing list
>> >> >>>>> FreeSWITCH-users at lists.freeswitch.org
>> >> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>>>>
>> >> >>>>>
>> >> >>>>>
>> >> >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >>>>> http://www.freeswitch.org
>> >> >>>>
>> >> >>>>
>> >> >>>>
>> >> >>>> --
>> >> >>>> Anthony Minessale II
>> >> >>>>
>> >> >>>> FreeSWITCH http://www.freeswitch.org/
>> >> >>>> ClueCon http://www.cluecon.com/
>> >> >>>> Twitter: http://twitter.com/FreeSWITCH_wire
>> >> >>>>
>> >> >>>> AIM: anthm
>> >> >>>> MSN:anthony_minessale at hotmail.com
>> >> >>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> >> >>>> IRC: irc.freenode.net #freeswitch
>> >> >>>>
>> >> >>>> FreeSWITCH Developer Conference
>> >> >>>> sip:888 at conference.freeswitch.org
>> >> >>>> iax:guest at conference.freeswitch.org/888
>> >> >>>> googletalk:conf+888 at conference.freeswitch.org
>> >> >>>> pstn:+19193869900
>> >> >>>
>> >> >>>
>> >> >>>
>> >> >>> --
>> >> >>> Anthony Minessale II
>> >> >>>
>> >> >>> FreeSWITCH http://www.freeswitch.org/
>> >> >>> ClueCon http://www.cluecon.com/
>> >> >>> Twitter: http://twitter.com/FreeSWITCH_wire
>> >> >>>
>> >> >>> AIM: anthm
>> >> >>> MSN:anthony_minessale at hotmail.com
>> >> >>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> >> >>> IRC: irc.freenode.net #freeswitch
>> >> >>>
>> >> >>> FreeSWITCH Developer Conference
>> >> >>> sip:888 at conference.freeswitch.org
>> >> >>> iax:guest at conference.freeswitch.org/888
>> >> >>> googletalk:conf+888 at conference.freeswitch.org
>> >> >>> pstn:+19193869900
>> >> >>>
>> >> >>> _______________________________________________
>> >> >>> FreeSWITCH-users mailing list
>> >> >>> FreeSWITCH-users at lists.freeswitch.org
>> >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>>
>> >> >>>
>> >> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >>> http://www.freeswitch.org
>> >> >>>
>> >> >>>
>> >> >>
>> >> >> _______________________________________________
>> >> >> FreeSWITCH-users mailing list
>> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>
>> >> >>
>> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> http://www.freeswitch.org
>> >> >>
>> >> >>
>> >> >> _______________________________________________
>> >> >> FreeSWITCH-users mailing list
>> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>
>> >> >>
>> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> http://www.freeswitch.org
>> >> >>
>> >> >
>> >> > _______________________________________________
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >
>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> > http://www.freeswitch.org
>> >> >
>> >> >
>> >> > --
>> >> > Anthony Minessale II
>> >> >
>> >> > FreeSWITCH http://www.freeswitch.org/
>> >> > ClueCon http://www.cluecon.com/
>> >> > Twitter: http://twitter.com/FreeSWITCH_wire
>> >> >
>> >> > AIM: anthm
>> >> > MSN:anthony_minessale at hotmail.com
>> >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> >> > IRC: irc.freenode.net #freeswitch
>> >> >
>> >> > FreeSWITCH Developer Conference
>> >> > sip:888 at conference.freeswitch.org
>> >> > iax:guest at conference.freeswitch.org/888
>> >> > googletalk:conf+888 at conference.freeswitch.org
>> >> > pstn:+19193869900
>> >> >
>> >> > _______________________________________________
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >
>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> > http://www.freeswitch.org
>> >> >
>> >> >
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> >
>> > --
>> > Anthony Minessale II
>> >
>> > FreeSWITCH http://www.freeswitch.org/
>> > ClueCon http://www.cluecon.com/
>> > Twitter: http://twitter.com/FreeSWITCH_wire
>> >
>> > AIM: anthm
>> > MSN:anthony_minessale at hotmail.com
>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> > IRC: irc.freenode.net #freeswitch
>> >
>> > FreeSWITCH Developer Conference
>> > sip:888 at conference.freeswitch.org
>> > iax:guest at conference.freeswitch.org/888
>> > googletalk:conf+888 at conference.freeswitch.org
>> > pstn:+19193869900
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From anthony.minessale at gmail.com  Tue Feb 16 07:55:02 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 16 Feb 2010 09:55:02 -0600
Subject: [Freeswitch-users] SQL ERR: library routine called out of
	sequence
In-Reply-To: <d361ca7b1002160727ucb6e0c8id62e803757333213@mail.gmail.com>
References: <d361ca7b1002160656w7d616f6dn5e80087ebde94dbe@mail.gmail.com>
	<191c3a031002160705w4050c3cdk2bf2922e9aff5ba3@mail.gmail.com>
	<d361ca7b1002160727ucb6e0c8id62e803757333213@mail.gmail.com>
Message-ID: <191c3a031002160755q569fe2ednfefb01b6d9cf7f0e@mail.gmail.com>

you may want to do a clean wipe of all files related to FS then.
you clearly have some problem with legacy something or other because we
don't see that on dozens of dev boxes.

What os is it?


On Tue, Feb 16, 2010 at 9:27 AM, Yehavi Bourvine
<yehavi.bourvine at gmail.com>wrote:

> Tried this, but it didn't help. I delete these DB files before any upgrade
> just to be sure.
>
>                   Thanks! __Yehavi:
>
>  2010/2/16 Anthony Minessale <anthony.minessale at gmail.com>
>
>> try removing all the .db files from /usr/local/freeswitch/db
>>
>>
>>   On Tue, Feb 16, 2010 at 8:56 AM, Yehavi Bourvine <
>> yehavi.bourvine at gmail.com> wrote:
>>
>>>   Hello,
>>>
>>>   After upgrading from 1.0.5pre10 to 1.0.5-20100216-0400 we started
>>> getting the above errors (I append bellow two samples). It seems Freeswitch
>>> fails to read a database using Sqlite.
>>> Anyone have seen this?
>>>
>>> Other details: Fedora 10, SQlite 3.5.9.
>>> We also do SQLite quesries during call setup via LUA from CoreDB. Is it
>>> an SQLite problem?
>>>
>>>                          Thanks! __Yehavi:
>>>
>>> The samples:
>>> 2010-02-16 16:22:13.762679 [ERR] switch_core_sqldb.c:404 SQL ERR [library
>>> routin
>>> e called out of sequence]
>>> delete from sip_dialogs where call_id='
>>> 8656841832142-120172129116107 at 10.64.1.2'
>>>
>>> 2010-02-16 16:42:00.802442 [ERR] switch_core_sqldb.c:722 SQL ERR: [select
>>> call_i
>>>
>>> d,sip_user,sip_host,contact,status,rpid,expires,user_agent,server_user,server_ho
>>>
>>> st,profile_name,hostname,network_ip,network_port,sip_username,sip_realm,mwi_user
>>> ,mwi_host from sip_registrations where profile_name='phones' and contact
>>> like '%
>>> 80635%'] library routine called out of sequence
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100216/b1e80285/attachment.html 

From freeswitchlistreceiver at gmail.com  Mon Feb 15 23:41:09 2010
From: freeswitchlistreceiver at gmail.com (Thomas Switch)
Date: Tue, 16 Feb 2010 08:41:09 +0100
Subject: [Freeswitch-users] Call Monitoring and Recording with FS?
Message-ID: <106852de1002152341v5ec56da9s2fc30b580a989a18@mail.gmail.com>

Hello FreeSWITCH folks,

I was asked to join a project in the VoIP field. Being a newbie to VoIP, I
read a couple of books, many web pages and came across FreeSWITCH.

Hope you don't mind answering two questions of mine:


a) Similar to a call centre application, I'd need to record *all*conversation.
Like "For quality assurance, all conversations will be recorded..."... Could
I do that with FS or do I need an additional piece of software or hardware?
If I cannot record all conversation easily, I might be able to water the
requirement down to an "Operator monitoring a call, can record the call on
demand" (see also the second question)? What about that?

b) As an operator, I need to be able to monitor any call.
I understand, that one can get the active connection from FS. Is there a
possibility to get into these calls?
Or do I need to hack a "standard" call silently into a conference call with
the operator? If yes, is it possible to do that without the participants in
the call noticing it?

Thanks a lot for your time and patience.

Daniel
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100216/f4ed72f7/attachment.html 

From Russell.Mosemann at cune.org  Tue Feb 16 08:13:09 2010
From: Russell.Mosemann at cune.org (Russell.Mosemann at cune.org)
Date: Tue, 16 Feb 2010 16:13:09 -0000
Subject: [Freeswitch-users] Call Monitoring and Recording with FS?
In-Reply-To: <106852de1002152341v5ec56da9s2fc30b580a989a18@mail.gmail.com>
Message-ID: <20100216161309.5D0D6156285@cuneorg-email.cune.pri>

Thomas Switch <freeswitchlistreceiver at gmail.com> said:

> I was asked to join a project in the VoIP field. Being a newbie to VoIP, I
> read a couple of books, many web pages and came across FreeSWITCH.

Well, then, you are primed and ready for this.

http://wiki.freeswitch.org/

-- 
Russell Mosemann



________________________________________________________
Concordia University, Nebraska
See http://www.cune.edu/ for the latest news and events!



From rupa at rupa.com  Tue Feb 16 08:22:06 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Tue, 16 Feb 2010 10:22:06 -0600
Subject: [Freeswitch-users] Call Monitoring and Recording with FS?
In-Reply-To: <106852de1002152341v5ec56da9s2fc30b580a989a18@mail.gmail.com>
References: <106852de1002152341v5ec56da9s2fc30b580a989a18@mail.gmail.com>
Message-ID: <b39273531002160822s52e2b2e1u86c734c47f5e904e@mail.gmail.com>

a) http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session
<http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session>(I'd
suggest recording to ${uuid}.wav, you can then use the uuid from the CDR to
find the wav file)

b) http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_eavesdrop

<http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_eavesdrop>The second
(at least) is in the default configs

On Tue, Feb 16, 2010 at 1:41 AM, Thomas Switch <
freeswitchlistreceiver at gmail.com> wrote:

> Hello FreeSWITCH folks,
>
> I was asked to join a project in the VoIP field. Being a newbie to VoIP, I
> read a couple of books, many web pages and came across FreeSWITCH.
>
> Hope you don't mind answering two questions of mine:
>
>
> a) Similar to a call centre application, I'd need to record *all*conversation.
> Like "For quality assurance, all conversations will be recorded..."...
> Could I do that with FS or do I need an additional piece of software or
> hardware?
> If I cannot record all conversation easily, I might be able to water the
> requirement down to an "Operator monitoring a call, can record the call on
> demand" (see also the second question)? What about that?
>
> b) As an operator, I need to be able to monitor any call.
> I understand, that one can get the active connection from FS. Is there a
> possibility to get into these calls?
> Or do I need to hack a "standard" call silently into a conference call with
> the operator? If yes, is it possible to do that without the participants in
> the call noticing it?
>
> Thanks a lot for your time and patience.
>
> Daniel
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100216/041a8ac2/attachment.html 

From rob4manhere at gmail.com  Tue Feb 16 08:22:52 2010
From: rob4manhere at gmail.com (Rob Forman)
Date: Tue, 16 Feb 2010 10:22:52 -0600
Subject: [Freeswitch-users] Call Monitoring and Recording with FS?
In-Reply-To: <106852de1002152341v5ec56da9s2fc30b580a989a18@mail.gmail.com>
References: <106852de1002152341v5ec56da9s2fc30b580a989a18@mail.gmail.com>
Message-ID: <AB27AF06-74B2-49FE-B943-CD5F007A3F13@gmail.com>

Hi Daniel,

FreeSWITCH can definitely handle those requirements, and more, without  
hacking.  I would encourage you to search around the wiki.

For recording:
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session

You can enable it on any and all calls via your dialplan.

For call monitoring:
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_eavesdrop


Good luck!
Rob

On Feb 16, 2010, at 1:41 AM, Thomas Switch wrote:

> Hello FreeSWITCH folks,
>
> I was asked to join a project in the VoIP field. Being a newbie to  
> VoIP, I read a couple of books, many web pages and came across  
> FreeSWITCH.
>
> Hope you don't mind answering two questions of mine:
>
>
> a) Similar to a call centre application, I'd need to record all  
> conversation.
> Like "For quality assurance, all conversations will be  
> recorded..."... Could I do that with FS or do I need an additional  
> piece of software or hardware?
> If I cannot record all conversation easily, I might be able to water  
> the requirement down to an "Operator monitoring a call, can record  
> the call on demand" (see also the second question)? What about that?
>
> b) As an operator, I need to be able to monitor any call.
> I understand, that one can get the active connection from FS. Is  
> there a possibility to get into these calls?
> Or do I need to hack a "standard" call silently into a conference  
> call with the operator? If yes, is it possible to do that without  
> the participants in the call noticing it?
>
> Thanks a lot for your time and patience.
>
> Daniel
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100216/74955172/attachment-0001.html 

From brian at freeswitch.org  Tue Feb 16 08:23:40 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 16 Feb 2010 10:23:40 -0600
Subject: [Freeswitch-users] Call Monitoring and Recording with FS?
In-Reply-To: <106852de1002152341v5ec56da9s2fc30b580a989a18@mail.gmail.com>
References: <106852de1002152341v5ec56da9s2fc30b580a989a18@mail.gmail.com>
Message-ID: <23F6840C-4278-4A86-8615-E8D69A3147C8@freeswitch.org>

Word of advice... our community is a unique one.  Just don't take anything personally and roll with the punches.  You'll fit right in if you take that advice.

/b
PS:  We have you now... you'll be hooked before long. muahahahah

On Feb 16, 2010, at 1:41 AM, Thomas Switch wrote:

> Hello FreeSWITCH folks,
> 
> I was asked to join a project in the VoIP field. Being a newbie to VoIP, I read a couple of books, many web pages and came across FreeSWITCH. 
> 
> Hope you don't mind answering two questions of mine:
> 
> 
> a) Similar to a call centre application, I'd need to record all conversation. 
> Like "For quality assurance, all conversations will be recorded..."... Could I do that with FS or do I need an additional piece of software or hardware? 
> If I cannot record all conversation easily, I might be able to water the requirement down to an "Operator monitoring a call, can record the call on demand" (see also the second question)? What about that?
> 
> b) As an operator, I need to be able to monitor any call. 
> I understand, that one can get the active connection from FS. Is there a possibility to get into these calls?
> Or do I need to hack a "standard" call silently into a conference call with the operator? If yes, is it possible to do that without the participants in the call noticing it?
> 
> Thanks a lot for your time and patience.
> 
> Daniel

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100216/f3e15fa0/attachment.html 

From moizchinoy at gmail.com  Tue Feb 16 08:35:17 2010
From: moizchinoy at gmail.com (Moiz Chinoy)
Date: Tue, 16 Feb 2010 20:35:17 +0400
Subject: [Freeswitch-users] Dingaling - external rtp supported?
Message-ID: <29b888f81002160835y6e290319ub0d04c088caa1b79@mail.gmail.com>

Hi All,

In mod_dingaling <param name="ext-rtp-ip"
value="$${external_rtp_ip}"/> supported? Whenever I uncomment this
line in client.xml (jingle profile) FS crashes as soon a call lands
(sip call) and dialplan bridges the call to a gtalk user.

I am running FS on windows and build is 16642.

-- 
Regards,
Moiz Chinoy.


From brian at freeswitch.org  Tue Feb 16 08:41:04 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 16 Feb 2010 10:41:04 -0600
Subject: [Freeswitch-users] Dingaling - external rtp supported?
In-Reply-To: <29b888f81002160835y6e290319ub0d04c088caa1b79@mail.gmail.com>
References: <29b888f81002160835y6e290319ub0d04c088caa1b79@mail.gmail.com>
Message-ID: <DEA39AD1-0EE9-473A-B78C-CDDF3F100578@freeswitch.org>

can you please update, try again and post a jira?

/b

On Feb 16, 2010, at 10:35 AM, Moiz Chinoy wrote:

> Hi All,
> 
> In mod_dingaling <param name="ext-rtp-ip"
> value="$${external_rtp_ip}"/> supported? Whenever I uncomment this
> line in client.xml (jingle profile) FS crashes as soon a call lands
> (sip call) and dialplan bridges the call to a gtalk user.
> 
> I am running FS on windows and build is 16642.
> 
> -- 
> Regards,
> Moiz Chinoy.



From jerry.richards at teotech.com  Tue Feb 16 09:11:31 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Tue, 16 Feb 2010 09:11:31 -0800
Subject: [Freeswitch-users] Presence PUBLISH Not Updating AfterSoftphone
	OffLine Then Available
In-Reply-To: <191c3a031002091557jfc9b486lb832ea538883bce9@mail.gmail.com>
References: <31FEE9072F104AE18B26812978FCA341@greyhawk.tonecommander.com><45963EB5-DFE5-43FA-AFCC-857E1E719E2D@jerris.com>
	<191c3a031002091557jfc9b486lb832ea538883bce9@mail.gmail.com>
Message-ID: <68A4C011A72B4ABA91A185F32480EB99@greyhawk.tonecommander.com>

I got version freeswitch-1.0.5-20100215-0400, built it, and ran it, and I am
seeing the same issue.  That is, once I set the Bria softphone status to
'Appear Offline', FS does not forward presence states until resubscription
time (i.e. tens of minutes later).  I posted a trace at
http://pastebin.freeswitch.org/12164.  At line 359 of the trace, FS is
logging an ERR at sofia_presence.c:662.  Here is the scenario:
 
1) Set Bria softphone presence state to 'Appear Offline'
2) Subscibing softphones reflect offline status
3) Set Bria softphone presence state to 'Available'
4) *** Subscibing softphones do not get status update ***
 
Thanks And Best Regards,
Jerry
 


  _____  

From: Anthony Minessale [mailto:anthony.minessale at gmail.com] 
Sent: Tuesday, February 09, 2010 3:58 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Presence PUBLISH Not Updating AfterSoftphone
OffLine Then Available


he means update to trunk first then try it again obviously.



On Tue, Feb 9, 2010 at 3:10 PM, Michael Jerris <mike at jerris.com> wrote:


Try this again, I think I saw changes go in for this issue.

Mike


On Feb 5, 2010, at 2:38 PM, Jerry Richards wrote:

> I found a scenario where presence status is not distributed to
subscribers.
> This is using the latest changes (as of Feb 03, 2010).  The scenario
> follows:
>
> 1) Register two Bria softphones (A and B), which each have the other as a
> contact.
> 2) Set softphone B's presence status to 'Appear Offline'.
> 3) Softphone A correctly reflects contact B is offline.
> 4) Set softphone B's presence status to 'Available'.
> 5) ******* There is no change to contact B's status at softphone A *******
>
> I posted a log at http://pastebin.freeswitch.org/12054.  At line 773,
there
> is an error when FS is processing the PUBLISH from softphone B:
>
> 773.2010-02-05 10:29:21.254221 [ERR] sofia_presence.c:674 DUMP PRESENCE
SQL:
>
> I did notice that after about 30 minutes, softphone B's status gets
> reflected at softphone A.



_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org





-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
<mailto:MSN%3Aanthony_minessale at hotmail.com> 
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
<mailto:PAYPAL%3Aanthony.minessale at gmail.com> 
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:+19193869900


-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100216/b3237ee4/attachment.html 

From msc at freeswitch.org  Tue Feb 16 09:31:14 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 16 Feb 2010 09:31:14 -0800
Subject: [Freeswitch-users] Call Monitoring and Recording with FS?
In-Reply-To: <106852de1002152341v5ec56da9s2fc30b580a989a18@mail.gmail.com>
References: <106852de1002152341v5ec56da9s2fc30b580a989a18@mail.gmail.com>
Message-ID: <87f2f3b91002160931g7fac3f52m4ecd95b617eece87@mail.gmail.com>

Daniel,

Welcome to the wonderful and crazy world of VoIP! FreeSWITCH is totally
awesome and can do all sorts of things. It is like a cross between a Hemi
and a box of Lego bricks: it is extremely powerful and you can build all
sorts of things with it. Here's the caveat: there probably isn't a
pre-rolled solution for your setup. That being said, if you have any
programming skills, or if you have access to some IT resources at your firm,
then you can probably build something for your enterprise. Alternatively,
you can email consulting at freeswitch.org and seek professional (i.e. paid)
help.

My recommendation is to learn more about FreeSWITCH from these resources:
wiki.freeswitch.org
lists.freeswitch.org (freeswitch-users is the main list)
#freeswitch channel on irc.freenode.net
Community conf call on Fridays:
http://wiki.freeswitch.org/wiki/Weekly_Conference_Call

A very gentle intro to FreeSWITCH can be found here: http://bit.ly/EpVrv
After that then it's off to the wiki. There is a FreeSWITCH book in writing,
probably due out by summer time. In the meantime the community will be happy
to answer questions, especially if you roll up your sleeves and try things
before you ask.

Last pieces of advice: Use Linux distro CentOS 5.3, and use x86_64 if you
have 64bit hardware. We've seen crazy things happen with using 32bit Linux
on 64bit hardware. Also, use latest SVN trunk or download from
latest.freeswitch.org. FreeSWITCH is a unique project where the latest SVN
trunk is almost always more stable than the "latest stable" release. Hint:
if you want to update your FreeSWITCH installation to "the latest" then just
go to your source directory and type "make current" and it will do the rest.


Have fun!
-Michael

On Mon, Feb 15, 2010 at 11:41 PM, Thomas Switch <
freeswitchlistreceiver at gmail.com> wrote:

> Hello FreeSWITCH folks,
>
> I was asked to join a project in the VoIP field. Being a newbie to VoIP, I
> read a couple of books, many web pages and came across FreeSWITCH.
>
> Hope you don't mind answering two questions of mine:
>
>
> a) Similar to a call centre application, I'd need to record *all*conversation.
> Like "For quality assurance, all conversations will be recorded..."...
> Could I do that with FS or do I need an additional piece of software or
> hardware?
> If I cannot record all conversation easily, I might be able to water the
> requirement down to an "Operator monitoring a call, can record the call on
> demand" (see also the second question)? What about that?
>
> b) As an operator, I need to be able to monitor any call.
> I understand, that one can get the active connection from FS. Is there a
> possibility to get into these calls?
> Or do I need to hack a "standard" call silently into a conference call with
> the operator? If yes, is it possible to do that without the participants in
> the call noticing it?
>
> Thanks a lot for your time and patience.
>
> Daniel
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100216/67c9317f/attachment-0001.html 

From ivan at myrvold.org  Tue Feb 16 09:41:15 2010
From: ivan at myrvold.org (Ivan C Myrvold)
Date: Tue, 16 Feb 2010 18:41:15 +0100
Subject: [Freeswitch-users] mod_zeroconf
Message-ID: <8FFE625C-FFBE-4414-A95B-C54C1D21BFBF@myrvold.org>

How can I build FreeSWITCH with mod_zeroconf? I can't find any mod_zeroconf to uncomment in modules.conf

Ivan

From mrene_lists at avgs.ca  Tue Feb 16 09:44:55 2010
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Tue, 16 Feb 2010 12:44:55 -0500
Subject: [Freeswitch-users] mod_zeroconf
In-Reply-To: <8FFE625C-FFBE-4414-A95B-C54C1D21BFBF@myrvold.org>
References: <8FFE625C-FFBE-4414-A95B-C54C1D21BFBF@myrvold.org>
Message-ID: <45AE1465-DD7C-4101-A71A-B148A7DBCD74@avgs.ca>

mod_zeroconf has been moved to unsupported. you can still get it, cd  
to src/mod/applications/, type in svn co http://svn.freeswitch.org/svn/unsupported/mod_zeroconf/ 
  and then add a line in modules.conf it should then build  
auttomatically when you do make.

You can also speed that up and type make mod_zeroconf / make  
mod_zeroconf-install once the line in modules.conf has been added

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 16-Feb-10, at 12:41 PM, Ivan C Myrvold wrote:

> How can I build FreeSWITCH with mod_zeroconf? I can't find any  
> mod_zeroconf to uncomment in modules.conf
>
> Ivan
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From lists at redbonez.net  Tue Feb 16 10:03:52 2010
From: lists at redbonez.net (Adam Ford)
Date: Tue, 16 Feb 2010 11:03:52 -0700
Subject: [Freeswitch-users] Version 1.05 release
In-Reply-To: <191c3a031002151543q6a00a936vf7a5e9d4b69d0ebc@mail.gmail.com>
References: <mailman.6410.1266268685.2495.freeswitch-users@lists.freeswitch.org>	<054c01caae94$6131b1c0$23951540$@com>
	<191c3a031002151543q6a00a936vf7a5e9d4b69d0ebc@mail.gmail.com>
Message-ID: <005101caaf32$679da250$36d8e6f0$@net>

I'm sure you guys get this all the time, but I just wanted to throw in and
say I appreciate what all you FreeSWITCH devs have done in creating this
software.   Getting to learn and implement a FreeSWITCH system for our
office has been the most fun project I have had in years as a sys/network
admin.

 

-Adam

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Monday, February 15, 2010 4:43 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Version 1.05 release

 


Since you are advertising services on your website most likely all provided
by our free software, I hope you can learn some patience while you wait for
us to provide you with your next release to sell to your customers.

The release will be as soon as we make sure all the bugs are fixed, unless
you prefer it to be buggy.

BTW,

If your issues are still present in 1.0.5 and it's because you never
reported them, then we will, of course, be very unhappy.

If this is unacceptable we do offer a triple-your-money-back guarantee that
we will upgrade to quadruple if you act now.



On Mon, Feb 15, 2010 at 5:12 PM, Goran Donev
<gorand at donevtechconsulting.com> wrote:

I really didn't get a definitive answer on when 1.05 is slated to be
released. We are running into some issues that I hope that 1.05 fixes. Do we
have an eta?

Thx


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
<mailto:MSN%3Aanthony_minessale at hotmail.com> 
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
<mailto:PAYPAL%3Aanthony.minessale at gmail.com> 
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:+19193869900

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100216/594f00d7/attachment.html 

From brian at freeswitch.org  Tue Feb 16 10:12:20 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 16 Feb 2010 12:12:20 -0600
Subject: [Freeswitch-users] Version 1.05 release
In-Reply-To: <005101caaf32$679da250$36d8e6f0$@net>
References: <mailman.6410.1266268685.2495.freeswitch-users@lists.freeswitch.org>	<054c01caae94$6131b1c0$23951540$@com>
	<191c3a031002151543q6a00a936vf7a5e9d4b69d0ebc@mail.gmail.com>
	<005101caaf32$679da250$36d8e6f0$@net>
Message-ID: <1C2E630C-E859-42ED-B5CE-BFF89212B71B@freeswitch.org>

Thank you... hope to see you at ClueCon this year too... Registration is open but I'm working out the last few minor details with the Trump hotel in Chicago. 

http://www.cluecon.com  (website is being worked on and updated as we move forward) 

Thanks,

/b

On Feb 16, 2010, at 12:03 PM, Adam Ford wrote:

> I?m sure you guys get this all the time, but I just wanted to throw in and say I appreciate what all you FreeSWITCH devs have done in creating this software.   Getting to learn and implement a FreeSWITCH system for our office has been the most fun project I have had in years as a sys/network admin.
>  
> -Adam
>  

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100216/a791c50f/attachment.html 

From msc at freeswitch.org  Tue Feb 16 10:18:01 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 16 Feb 2010 10:18:01 -0800
Subject: [Freeswitch-users] Version 1.05 release
In-Reply-To: <005101caaf32$679da250$36d8e6f0$@net>
References: <mailman.6410.1266268685.2495.freeswitch-users@lists.freeswitch.org>
	<054c01caae94$6131b1c0$23951540$@com>
	<191c3a031002151543q6a00a936vf7a5e9d4b69d0ebc@mail.gmail.com>
	<005101caaf32$679da250$36d8e6f0$@net>
Message-ID: <87f2f3b91002161018m5b0e0517xda3a6e95467cd496@mail.gmail.com>

On Tue, Feb 16, 2010 at 10:03 AM, Adam Ford <lists at redbonez.net> wrote:

>  I?m sure you guys get this all the time, but I just wanted to throw in
> and say I appreciate what all you FreeSWITCH devs have done in creating this
> software.   Getting to learn and implement a FreeSWITCH system for our
> office has been the most fun project I have had in years as a sys/network
> admin.
>
> The devs don't get this kind of email often enough, so many thanks for your
recognition. The guys work extremely hard on FreeSWITCH and related
projects, not the least of which is ClueCon. Thanks for chiming in.

BTW, the dev dinner was great! Thanks to all who give active support to the
community.
-MC

>
>
> -Adam
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony
> Minessale
> *Sent:* Monday, February 15, 2010 4:43 PM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Version 1.05 release
>
>
>
>
> Since you are advertising services on your website most likely all provided
> by our free software, I hope you can learn some patience while you wait for
> us to provide you with your next release to sell to your customers.
>
> The release will be as soon as we make sure all the bugs are fixed, unless
> you prefer it to be buggy.
>
> BTW,
>
> If your issues are still present in 1.0.5 and it's because you never
> reported them, then we will, of course, be very unhappy.
>
> If this is unacceptable we do offer a triple-your-money-back guarantee that
> we will upgrade to quadruple if you act now.
>
>  On Mon, Feb 15, 2010 at 5:12 PM, Goran Donev <
> gorand at donevtechconsulting.com> wrote:
>
> I really didn't get a definitive answer on when 1.05 is slated to be
> released. We are running into some issues that I hope that 1.05 fixes. Do
> we
> have an eta?
>
> Thx
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100216/f76a00fd/attachment-0001.html 

From freeswitch at peely.com  Tue Feb 16 10:55:50 2010
From: freeswitch at peely.com (peely)
Date: Tue, 16 Feb 2010 10:55:50 -0800 (PST)
Subject: [Freeswitch-users] event-socket outbound: Dialplan failover on
 socket error?
Message-ID: <27613245.post@talk.nabble.com>


Hi,

I'd like to implement a dialplan entry to use a secondary event-socket
application if the first esl server is down or could not be connected to.

Could somebody please tell me what continue_on_fail options I could set so
that I will continue ONLY if the event-socket outbound connection was
unsuccessful i.e. an "[ERR] mod_event_socket.c:414 Socket Error!"? By
default it seems to continue if the outbound bridge is unsuccessful however
I don;t want to do that.

Alternatively, is there something I can set from the event-socket session to
stop the continue_on_fail from failing over?


Thanks,



Neil.
-- 
View this message in context: http://old.nabble.com/event-socket-outbound%3A-Dialplan-failover-on-socket-error--tp27613245p27613245.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From yehavi.bourvine at gmail.com  Tue Feb 16 10:59:26 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Tue, 16 Feb 2010 20:59:26 +0200
Subject: [Freeswitch-users] SQL ERR: library routine called out of
	sequence
In-Reply-To: <191c3a031002160755q569fe2ednfefb01b6d9cf7f0e@mail.gmail.com>
References: <d361ca7b1002160656w7d616f6dn5e80087ebde94dbe@mail.gmail.com>
	<191c3a031002160705w4050c3cdk2bf2922e9aff5ba3@mail.gmail.com>
	<d361ca7b1002160727ucb6e0c8id62e803757333213@mail.gmail.com>
	<191c3a031002160755q569fe2ednfefb01b6d9cf7f0e@mail.gmail.com>
Message-ID: <d361ca7b1002161059w1ef9484dr87e69a33aa2b62fc@mail.gmail.com>

The OS is Fedora-10 (soon to be upgraded to 12).

What I do when I want to test a new version:

   - Download the latest one into a fresh directory
   - bootstrap.sh, configure and make
   - stop Freeswitch, delete everything in lib, mod, bin ,db
   - make install and run it.


Is there additional place to clean?

                        Thanks! __Yehavi:

 2010/2/16 Anthony Minessale <anthony.minessale at gmail.com>

> you may want to do a clean wipe of all files related to FS then.
> you clearly have some problem with legacy something or other because we
> don't see that on dozens of dev boxes.
>
> What os is it?
>
>
>
> On Tue, Feb 16, 2010 at 9:27 AM, Yehavi Bourvine <
> yehavi.bourvine at gmail.com> wrote:
>
>>  Tried this, but it didn't help. I delete these DB files before any
>> upgrade just to be sure.
>>
>>                   Thanks! __Yehavi:
>>
>>  2010/2/16 Anthony Minessale <anthony.minessale at gmail.com>
>>
>>> try removing all the .db files from /usr/local/freeswitch/db
>>>
>>>
>>>   On Tue, Feb 16, 2010 at 8:56 AM, Yehavi Bourvine <
>>> yehavi.bourvine at gmail.com> wrote:
>>>
>>>>   Hello,
>>>>
>>>>   After upgrading from 1.0.5pre10 to 1.0.5-20100216-0400 we started
>>>> getting the above errors (I append bellow two samples). It seems Freeswitch
>>>> fails to read a database using Sqlite.
>>>> Anyone have seen this?
>>>>
>>>> Other details: Fedora 10, SQlite 3.5.9.
>>>> We also do SQLite quesries during call setup via LUA from CoreDB. Is it
>>>> an SQLite problem?
>>>>
>>>>                          Thanks! __Yehavi:
>>>>
>>>> The samples:
>>>> 2010-02-16 16:22:13.762679 [ERR] switch_core_sqldb.c:404 SQL ERR
>>>> [library routin
>>>> e called out of sequence]
>>>> delete from sip_dialogs where call_id='
>>>> 8656841832142-120172129116107 at 10.64.1.2'
>>>>
>>>> 2010-02-16 16:42:00.802442 [ERR] switch_core_sqldb.c:722 SQL ERR:
>>>> [select call_i
>>>>
>>>> d,sip_user,sip_host,contact,status,rpid,expires,user_agent,server_user,server_ho
>>>>
>>>> st,profile_name,hostname,network_ip,network_port,sip_username,sip_realm,mwi_user
>>>> ,mwi_host from sip_registrations where profile_name='phones' and contact
>>>> like '%
>>>> 80635%'] library routine called out of sequence
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>> iax:guest at conference.freeswitch.org/888
>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> pstn:+19193869900
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100216/c8a4c162/attachment.html 

From anthony.minessale at gmail.com  Tue Feb 16 11:08:54 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 16 Feb 2010 13:08:54 -0600
Subject: [Freeswitch-users] event-socket outbound: Dialplan failover on
	socket error?
In-Reply-To: <27613245.post@talk.nabble.com>
References: <27613245.post@talk.nabble.com>
Message-ID: <191c3a031002161108j1f77925fpd58b4497b9b4d54a@mail.gmail.com>

continue_on_fail only applies to origination from the bridge app
the socket app should always continue to the next entry in the dp when it
fails.


On Tue, Feb 16, 2010 at 12:55 PM, peely <freeswitch at peely.com> wrote:

>
> Hi,
>
> I'd like to implement a dialplan entry to use a secondary event-socket
> application if the first esl server is down or could not be connected to.
>
> Could somebody please tell me what continue_on_fail options I could set so
> that I will continue ONLY if the event-socket outbound connection was
> unsuccessful i.e. an "[ERR] mod_event_socket.c:414 Socket Error!"? By
> default it seems to continue if the outbound bridge is unsuccessful however
> I don;t want to do that.
>
> Alternatively, is there something I can set from the event-socket session
> to
> stop the continue_on_fail from failing over?
>
>
> Thanks,
>
>
>
> Neil.
> --
> View this message in context:
> http://old.nabble.com/event-socket-outbound%3A-Dialplan-failover-on-socket-error--tp27613245p27613245.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100216/64686c89/attachment.html 

From anthony.minessale at gmail.com  Tue Feb 16 11:10:12 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 16 Feb 2010 13:10:12 -0600
Subject: [Freeswitch-users] SQL ERR: library routine called out of
	sequence
In-Reply-To: <d361ca7b1002161059w1ef9484dr87e69a33aa2b62fc@mail.gmail.com>
References: <d361ca7b1002160656w7d616f6dn5e80087ebde94dbe@mail.gmail.com>
	<191c3a031002160705w4050c3cdk2bf2922e9aff5ba3@mail.gmail.com>
	<d361ca7b1002160727ucb6e0c8id62e803757333213@mail.gmail.com>
	<191c3a031002160755q569fe2ednfefb01b6d9cf7f0e@mail.gmail.com>
	<d361ca7b1002161059w1ef9484dr87e69a33aa2b62fc@mail.gmail.com>
Message-ID: <191c3a031002161110s637a48a2r717ae94557fe9e9a@mail.gmail.com>

That sounds about right.

That error usually has something to do with using db calls on a closed file
or something along those lines.
Maybe you have a permission problem on the directory where the db files are?


On Tue, Feb 16, 2010 at 12:59 PM, Yehavi Bourvine <yehavi.bourvine at gmail.com
> wrote:

> The OS is Fedora-10 (soon to be upgraded to 12).
>
> What I do when I want to test a new version:
>
>    - Download the latest one into a fresh directory
>    - bootstrap.sh, configure and make
>    - stop Freeswitch, delete everything in lib, mod, bin ,db
>    - make install and run it.
>
>
> Is there additional place to clean?
>
>                         Thanks! __Yehavi:
>
>  2010/2/16 Anthony Minessale <anthony.minessale at gmail.com>
>
>> you may want to do a clean wipe of all files related to FS then.
>> you clearly have some problem with legacy something or other because we
>> don't see that on dozens of dev boxes.
>>
>> What os is it?
>>
>>
>>
>> On Tue, Feb 16, 2010 at 9:27 AM, Yehavi Bourvine <
>> yehavi.bourvine at gmail.com> wrote:
>>
>>>  Tried this, but it didn't help. I delete these DB files before any
>>> upgrade just to be sure.
>>>
>>>                   Thanks! __Yehavi:
>>>
>>>  2010/2/16 Anthony Minessale <anthony.minessale at gmail.com>
>>>
>>>> try removing all the .db files from /usr/local/freeswitch/db
>>>>
>>>>
>>>>   On Tue, Feb 16, 2010 at 8:56 AM, Yehavi Bourvine <
>>>> yehavi.bourvine at gmail.com> wrote:
>>>>
>>>>>   Hello,
>>>>>
>>>>>   After upgrading from 1.0.5pre10 to 1.0.5-20100216-0400 we started
>>>>> getting the above errors (I append bellow two samples). It seems Freeswitch
>>>>> fails to read a database using Sqlite.
>>>>> Anyone have seen this?
>>>>>
>>>>> Other details: Fedora 10, SQlite 3.5.9.
>>>>> We also do SQLite quesries during call setup via LUA from CoreDB. Is it
>>>>> an SQLite problem?
>>>>>
>>>>>                          Thanks! __Yehavi:
>>>>>
>>>>> The samples:
>>>>> 2010-02-16 16:22:13.762679 [ERR] switch_core_sqldb.c:404 SQL ERR
>>>>> [library routin
>>>>> e called out of sequence]
>>>>> delete from sip_dialogs where call_id='
>>>>> 8656841832142-120172129116107 at 10.64.1.2'
>>>>>
>>>>> 2010-02-16 16:42:00.802442 [ERR] switch_core_sqldb.c:722 SQL ERR:
>>>>> [select call_i
>>>>>
>>>>> d,sip_user,sip_host,contact,status,rpid,expires,user_agent,server_user,server_ho
>>>>>
>>>>> st,profile_name,hostname,network_ip,network_port,sip_username,sip_realm,mwi_user
>>>>> ,mwi_host from sip_registrations where profile_name='phones' and
>>>>> contact like '%
>>>>> 80635%'] library routine called out of sequence
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>>
>>>> --
>>>> Anthony Minessale II
>>>>
>>>> FreeSWITCH http://www.freeswitch.org/
>>>> ClueCon http://www.cluecon.com/
>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>
>>>> AIM: anthm
>>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>> IRC: irc.freenode.net #freeswitch
>>>>
>>>> FreeSWITCH Developer Conference
>>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>>> iax:guest at conference.freeswitch.org/888
>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>> pstn:+19193869900
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100216/ae8618dc/attachment-0001.html 

From yehavi.bourvine at gmail.com  Tue Feb 16 11:30:33 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Tue, 16 Feb 2010 21:30:33 +0200
Subject: [Freeswitch-users] SQL ERR: library routine called out of
	sequence
In-Reply-To: <191c3a031002161110s637a48a2r717ae94557fe9e9a@mail.gmail.com>
References: <d361ca7b1002160656w7d616f6dn5e80087ebde94dbe@mail.gmail.com>
	<191c3a031002160705w4050c3cdk2bf2922e9aff5ba3@mail.gmail.com>
	<d361ca7b1002160727ucb6e0c8id62e803757333213@mail.gmail.com>
	<191c3a031002160755q569fe2ednfefb01b6d9cf7f0e@mail.gmail.com>
	<d361ca7b1002161059w1ef9484dr87e69a33aa2b62fc@mail.gmail.com>
	<191c3a031002161110s637a48a2r717ae94557fe9e9a@mail.gmail.com>
Message-ID: <d361ca7b1002161130k3eb2d88by3100b30a9d29bd51@mail.gmail.com>

Most of the queries are ok, only some fail, thus it doesn't look like
permission problem. Furthermore, under 1.0.5pre10 it works for months.

Might it be thread unsafe function calls? I've found the following while
searching the WEB:

*According to the MSDN docs, System.Timers.Timer operates in a thread pool.
If that's the case, your code is breaking the "connections cannot be shared
across threads" rule for SQLit*

Although it quotes MSDN, it might be related to Linux as well.

                    Thanks, __Yehavi:

 2010/2/16 Anthony Minessale <anthony.minessale at gmail.com>

> That sounds about right.
>
> That error usually has something to do with using db calls on a closed file
> or something along those lines.
> Maybe you have a permission problem on the directory where the db files
> are?
>
>
>
> On Tue, Feb 16, 2010 at 12:59 PM, Yehavi Bourvine <
> yehavi.bourvine at gmail.com> wrote:
>
>>  The OS is Fedora-10 (soon to be upgraded to 12).
>>
>> What I do when I want to test a new version:
>>
>>    - Download the latest one into a fresh directory
>>    - bootstrap.sh, configure and make
>>    - stop Freeswitch, delete everything in lib, mod, bin ,db
>>    - make install and run it.
>>
>>
>> Is there additional place to clean?
>>
>>                         Thanks! __Yehavi:
>>
>>  2010/2/16 Anthony Minessale <anthony.minessale at gmail.com>
>>
>>> you may want to do a clean wipe of all files related to FS then.
>>> you clearly have some problem with legacy something or other because we
>>> don't see that on dozens of dev boxes.
>>>
>>> What os is it?
>>>
>>>
>>>
>>> On Tue, Feb 16, 2010 at 9:27 AM, Yehavi Bourvine <
>>> yehavi.bourvine at gmail.com> wrote:
>>>
>>>>  Tried this, but it didn't help. I delete these DB files before any
>>>> upgrade just to be sure.
>>>>
>>>>                   Thanks! __Yehavi:
>>>>
>>>>  2010/2/16 Anthony Minessale <anthony.minessale at gmail.com>
>>>>
>>>>> try removing all the .db files from /usr/local/freeswitch/db
>>>>>
>>>>>
>>>>>   On Tue, Feb 16, 2010 at 8:56 AM, Yehavi Bourvine <
>>>>> yehavi.bourvine at gmail.com> wrote:
>>>>>
>>>>>>   Hello,
>>>>>>
>>>>>>   After upgrading from 1.0.5pre10 to 1.0.5-20100216-0400 we started
>>>>>> getting the above errors (I append bellow two samples). It seems Freeswitch
>>>>>> fails to read a database using Sqlite.
>>>>>> Anyone have seen this?
>>>>>>
>>>>>> Other details: Fedora 10, SQlite 3.5.9.
>>>>>> We also do SQLite quesries during call setup via LUA from CoreDB. Is
>>>>>> it an SQLite problem?
>>>>>>
>>>>>>                          Thanks! __Yehavi:
>>>>>>
>>>>>> The samples:
>>>>>> 2010-02-16 16:22:13.762679 [ERR] switch_core_sqldb.c:404 SQL ERR
>>>>>> [library routin
>>>>>> e called out of sequence]
>>>>>> delete from sip_dialogs where call_id='
>>>>>> 8656841832142-120172129116107 at 10.64.1.2'
>>>>>>
>>>>>> 2010-02-16 16:42:00.802442 [ERR] switch_core_sqldb.c:722 SQL ERR:
>>>>>> [select call_i
>>>>>>
>>>>>> d,sip_user,sip_host,contact,status,rpid,expires,user_agent,server_user,server_ho
>>>>>>
>>>>>> st,profile_name,hostname,network_ip,network_port,sip_username,sip_realm,mwi_user
>>>>>> ,mwi_host from sip_registrations where profile_name='phones' and
>>>>>> contact like '%
>>>>>> 80635%'] library routine called out of sequence
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> Anthony Minessale II
>>>>>
>>>>> FreeSWITCH http://www.freeswitch.org/
>>>>> ClueCon http://www.cluecon.com/
>>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>>
>>>>> AIM: anthm
>>>>> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>>> IRC: irc.freenode.net #freeswitch
>>>>>
>>>>> FreeSWITCH Developer Conference
>>>>> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>>>>> iax:guest at conference.freeswitch.org/888
>>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>>> pstn:+19193869900
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>> iax:guest at conference.freeswitch.org/888
>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> pstn:+19193869900
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100216/d3a65942/attachment.html 

From anthony.minessale at gmail.com  Tue Feb 16 12:30:41 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 16 Feb 2010 14:30:41 -0600
Subject: [Freeswitch-users] SQL ERR: library routine called out of
	sequence
In-Reply-To: <d361ca7b1002161130k3eb2d88by3100b30a9d29bd51@mail.gmail.com>
References: <d361ca7b1002160656w7d616f6dn5e80087ebde94dbe@mail.gmail.com>
	<191c3a031002160705w4050c3cdk2bf2922e9aff5ba3@mail.gmail.com>
	<d361ca7b1002160727ucb6e0c8id62e803757333213@mail.gmail.com>
	<191c3a031002160755q569fe2ednfefb01b6d9cf7f0e@mail.gmail.com>
	<d361ca7b1002161059w1ef9484dr87e69a33aa2b62fc@mail.gmail.com>
	<191c3a031002161110s637a48a2r717ae94557fe9e9a@mail.gmail.com>
	<d361ca7b1002161130k3eb2d88by3100b30a9d29bd51@mail.gmail.com>
Message-ID: <191c3a031002161230h2b6e6f9wf3f65ddb71f08297@mail.gmail.com>

Strange, even on abusive testing we have not seen this problem.

please update to latest trunk.
There was only one change I can think of that may cause your issue and I
added a patch for it.
If it persists try setting the sql-in-transactions profile param to false.




On Tue, Feb 16, 2010 at 1:30 PM, Yehavi Bourvine
<yehavi.bourvine at gmail.com>wrote:

> Most of the queries are ok, only some fail, thus it doesn't look like
> permission problem. Furthermore, under 1.0.5pre10 it works for months.
>
> Might it be thread unsafe function calls? I've found the following while
> searching the WEB:
>
> *According to the MSDN docs, System.Timers.Timer operates in a thread
> pool.  If that's the case, your code is breaking the "connections cannot be
> shared across threads" rule for SQLit*
>
> Although it quotes MSDN, it might be related to Linux as well.
>
>                     Thanks, __Yehavi:
>
>  2010/2/16 Anthony Minessale <anthony.minessale at gmail.com>
>
>> That sounds about right.
>>
>> That error usually has something to do with using db calls on a closed
>> file or something along those lines.
>> Maybe you have a permission problem on the directory where the db files
>> are?
>>
>>
>>
>> On Tue, Feb 16, 2010 at 12:59 PM, Yehavi Bourvine <
>> yehavi.bourvine at gmail.com> wrote:
>>
>>>  The OS is Fedora-10 (soon to be upgraded to 12).
>>>
>>> What I do when I want to test a new version:
>>>
>>>    - Download the latest one into a fresh directory
>>>    - bootstrap.sh, configure and make
>>>    - stop Freeswitch, delete everything in lib, mod, bin ,db
>>>    - make install and run it.
>>>
>>>
>>> Is there additional place to clean?
>>>
>>>                         Thanks! __Yehavi:
>>>
>>>  2010/2/16 Anthony Minessale <anthony.minessale at gmail.com>
>>>
>>>> you may want to do a clean wipe of all files related to FS then.
>>>> you clearly have some problem with legacy something or other because we
>>>> don't see that on dozens of dev boxes.
>>>>
>>>> What os is it?
>>>>
>>>>
>>>>
>>>> On Tue, Feb 16, 2010 at 9:27 AM, Yehavi Bourvine <
>>>> yehavi.bourvine at gmail.com> wrote:
>>>>
>>>>>  Tried this, but it didn't help. I delete these DB files before any
>>>>> upgrade just to be sure.
>>>>>
>>>>>                   Thanks! __Yehavi:
>>>>>
>>>>>  2010/2/16 Anthony Minessale <anthony.minessale at gmail.com>
>>>>>
>>>>>> try removing all the .db files from /usr/local/freeswitch/db
>>>>>>
>>>>>>
>>>>>>   On Tue, Feb 16, 2010 at 8:56 AM, Yehavi Bourvine <
>>>>>> yehavi.bourvine at gmail.com> wrote:
>>>>>>
>>>>>>>   Hello,
>>>>>>>
>>>>>>>   After upgrading from 1.0.5pre10 to 1.0.5-20100216-0400 we started
>>>>>>> getting the above errors (I append bellow two samples). It seems Freeswitch
>>>>>>> fails to read a database using Sqlite.
>>>>>>> Anyone have seen this?
>>>>>>>
>>>>>>> Other details: Fedora 10, SQlite 3.5.9.
>>>>>>> We also do SQLite quesries during call setup via LUA from CoreDB. Is
>>>>>>> it an SQLite problem?
>>>>>>>
>>>>>>>                          Thanks! __Yehavi:
>>>>>>>
>>>>>>> The samples:
>>>>>>> 2010-02-16 16:22:13.762679 [ERR] switch_core_sqldb.c:404 SQL ERR
>>>>>>> [library routin
>>>>>>> e called out of sequence]
>>>>>>> delete from sip_dialogs where call_id='
>>>>>>> 8656841832142-120172129116107 at 10.64.1.2'
>>>>>>>
>>>>>>> 2010-02-16 16:42:00.802442 [ERR] switch_core_sqldb.c:722 SQL ERR:
>>>>>>> [select call_i
>>>>>>>
>>>>>>> d,sip_user,sip_host,contact,status,rpid,expires,user_agent,server_user,server_ho
>>>>>>>
>>>>>>> st,profile_name,hostname,network_ip,network_port,sip_username,sip_realm,mwi_user
>>>>>>> ,mwi_host from sip_registrations where profile_name='phones' and
>>>>>>> contact like '%
>>>>>>> 80635%'] library routine called out of sequence
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>> Anthony Minessale II
>>>>>>
>>>>>> FreeSWITCH http://www.freeswitch.org/
>>>>>> ClueCon http://www.cluecon.com/
>>>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>>>
>>>>>> AIM: anthm
>>>>>> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
>>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>>>> IRC: irc.freenode.net #freeswitch
>>>>>>
>>>>>> FreeSWITCH Developer Conference
>>>>>> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>>>>>> iax:guest at conference.freeswitch.org/888
>>>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>>>> pstn:+19193869900
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>>
>>>> --
>>>> Anthony Minessale II
>>>>
>>>> FreeSWITCH http://www.freeswitch.org/
>>>> ClueCon http://www.cluecon.com/
>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>
>>>> AIM: anthm
>>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>> IRC: irc.freenode.net #freeswitch
>>>>
>>>> FreeSWITCH Developer Conference
>>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>>> iax:guest at conference.freeswitch.org/888
>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>> pstn:+19193869900
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100216/2fcd1a2f/attachment-0001.html 

From brian at freeswitch.org  Tue Feb 16 12:36:45 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 16 Feb 2010 14:36:45 -0600
Subject: [Freeswitch-users] SQL ERR: library routine called out of
	sequence
In-Reply-To: <191c3a031002161230h2b6e6f9wf3f65ddb71f08297@mail.gmail.com>
References: <d361ca7b1002160656w7d616f6dn5e80087ebde94dbe@mail.gmail.com>
	<191c3a031002160705w4050c3cdk2bf2922e9aff5ba3@mail.gmail.com>
	<d361ca7b1002160727ucb6e0c8id62e803757333213@mail.gmail.com>
	<191c3a031002160755q569fe2ednfefb01b6d9cf7f0e@mail.gmail.com>
	<d361ca7b1002161059w1ef9484dr87e69a33aa2b62fc@mail.gmail.com>
	<191c3a031002161110s637a48a2r717ae94557fe9e9a@mail.gmail.com>
	<d361ca7b1002161130k3eb2d88by3100b30a9d29bd51@mail.gmail.com>
	<191c3a031002161230h2b6e6f9wf3f65ddb71f08297@mail.gmail.com>
Message-ID: <CBE5A464-0321-4A55-9195-7CAB0E3405AE@freeswitch.org>

What distro are you on and kernel version?

cat /proc/cpuinfo
uname -a

and such

/b

On Feb 16, 2010, at 2:30 PM, Anthony Minessale wrote:

> Strange, even on abusive testing we have not seen this problem.
> 
> please update to latest trunk.
> There was only one change I can think of that may cause your issue and I added a patch for it.
> If it persists try setting the sql-in-transactions profile param to false.
> 
> 
> 
> 
> On Tue, Feb 16, 2010 at 1:30 PM, Yehavi Bourvine <yehavi.bourvine at gmail.com> wrote:
> Most of the queries are ok, only some fail, thus it doesn't look like permission problem. Furthermore, under 1.0.5pre10 it works for months.
>  
> Might it be thread unsafe function calls? I've found the following while searching the WEB:
>  
> According to the MSDN docs, System.Timers.Timer operates in a thread pool.  If that's the case, your code is breaking the "connections cannot be shared across threads" rule for SQLit
>  

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100216/6bbf2331/attachment.html 

From lists at redbonez.net  Tue Feb 16 13:39:35 2010
From: lists at redbonez.net (Adam Ford)
Date: Tue, 16 Feb 2010 14:39:35 -0700
Subject: [Freeswitch-users] Multiple versions of FreeSWITCH on one box
Message-ID: <00cb01caaf50$8a1dbe50$9e593af0$@net>

According to the wiki it is possible to run multiple FreeSWITCH instances on
one box.  Are there any known issues with having those instances be
different versions? 

 

Unfortunately I don't have a separate box to test the migration from 1.0.4
to 1.0.5, so I am looking to build a separate instance of 1.0.5 on the same
box.   

 

-Adam

 

 

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100216/db6524c3/attachment.html 

From brian at freeswitch.org  Tue Feb 16 13:45:10 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 16 Feb 2010 15:45:10 -0600
Subject: [Freeswitch-users] Multiple versions of FreeSWITCH on one box
In-Reply-To: <00cb01caaf50$8a1dbe50$9e593af0$@net>
References: <00cb01caaf50$8a1dbe50$9e593af0$@net>
Message-ID: <E7A31EA6-D007-41D1-BACE-C113064E3860@freeswitch.org>

./configure --prefix=/usr/local/freeswitch-1.0.5

then edit for different ip's and ports 

/b

On Feb 16, 2010, at 3:39 PM, Adam Ford wrote:

> According to the wiki it is possible to run multiple FreeSWITCH instances on one box.  Are there any known issues with having those instances be different versions?
>  
> Unfortunately I don?t have a separate box to test the migration from 1.0.4 to 1.0.5, so I am looking to build a separate instance of 1.0.5 on the same box.   
>  
> -Adam

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100216/3bb746f0/attachment.html 

From leo.zibi at gmail.com  Tue Feb 16 14:50:25 2010
From: leo.zibi at gmail.com (leo.zibi at gmail.com)
Date: Tue, 16 Feb 2010 23:50:25 +0100
Subject: [Freeswitch-users] Multiple versions of FreeSWITCH on one box
In-Reply-To: <00cb01caaf50$8a1dbe50$9e593af0$@net>
References: <00cb01caaf50$8a1dbe50$9e593af0$@net>
Message-ID: <4B7B2131.4010003@gmail.com>

An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100216/9f1997be/attachment.html 

From lists at redbonez.net  Tue Feb 16 14:57:39 2010
From: lists at redbonez.net (Adam Ford)
Date: Tue, 16 Feb 2010 15:57:39 -0700
Subject: [Freeswitch-users] Multiple versions of FreeSWITCH on one box
In-Reply-To: <4B7B2131.4010003@gmail.com>
References: <00cb01caaf50$8a1dbe50$9e593af0$@net> <4B7B2131.4010003@gmail.com>
Message-ID: <00e401caaf5b$71cdfc10$5569f430$@net>

Yeah, that is what I was referencing when I said 'According to the wiki it
is possible..' My question was if it matter if they were different versions
of FreeSWITCH (1.0.4 and 1.0.5).  I will take this as a no.

 

Thanks you guys.

-Adam

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
leo.zibi at gmail.com
Sent: Tuesday, February 16, 2010 3:50 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Multiple versions of FreeSWITCH on one box

 

http://wiki.freeswitch.org/wiki/Deployment_Setup

Adam Ford wrote: 

According to the wiki it is possible to run multiple FreeSWITCH instances on
one box.  Are there any known issues with having those instances be
different versions? 

 

Unfortunately I don't have a separate box to test the migration from 1.0.4
to 1.0.5, so I am looking to build a separate instance of 1.0.5 on the same
box.   

 

-Adam

 

 

 

 



  _____  



 
_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
  

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100216/422c1747/attachment-0001.html 

From mike at jerris.com  Tue Feb 16 15:56:37 2010
From: mike at jerris.com (Michael Jerris)
Date: Tue, 16 Feb 2010 18:56:37 -0500
Subject: [Freeswitch-users] external_sip_address and
	external_rtp_address Question
In-Reply-To: <CDD4A2D8AA494ED8BF64401C5C953F72@greyhawk.tonecommander.com>
References: <CDD4A2D8AA494ED8BF64401C5C953F72@greyhawk.tonecommander.com>
Message-ID: <6D6A08D3-68FD-4B19-90A5-A19A9BC9100E@jerris.com>

not sure what those vars are, in the sip profiles we have ext-sip-ip and ext-rtp-ip.  They should be documented in the default configs and on the wiki.  These are per-profile settings, not per provider or gateway (unless of course you have a profile for each).

Mike

On Feb 15, 2010, at 3:49 PM, Jerry Richards wrote:

> I only see one example for setting of external_sip_address and
> external_rtp_address tags.  Is it true they are used to specify a SIP
> provider outside of a LAN (i.e. through a router)?  If so, then can these
> tags be set for each sip_profile?  So, if I have multiple external SIP
> providers that are accessed through NAT, they would each have their own
> external_sip_address and external_rtp_address?



From robert.hadley at teotech.com  Tue Feb 16 17:01:45 2010
From: robert.hadley at teotech.com (Robert Hadley)
Date: Tue, 16 Feb 2010 17:01:45 -0800
Subject: [Freeswitch-users] mod_fax receives fax to file but logs error msg
Message-ID: <02F74983D688435EA3F6954B48A14D6D@greyhawk.tonecommander.com>

I have been playing around with mod_fax and can successfully receive a fax
to file.  However, while doing so mod_fax is logging an error message.  Does
anybody know what this error means?

 

Dialplan: OpenZAP/2:1/1011 parsing [default->REH_test_fax_receive]
continue=false

Dialplan: OpenZAP/2:1/1011 Regex (PASS) [REH_test_fax_receive]
destination_number(1011) =~ /^1011$/ break=on-false

Dialplan: OpenZAP/2:1/1011 Action disable_ec()

Dialplan: OpenZAP/2:1/1011 Action answer()

Dialplan: OpenZAP/2:1/1011 Action playback(silence_stream://2000)

Dialplan: OpenZAP/2:1/1011 Action rxfax(/tmp/rxfax.tif)

Dialplan: OpenZAP/2:1/1011 Action hangup()

2010-02-16 16:27:08.542746 [DEBUG] switch_core_state_machine.c:122
(OpenZAP/2:1/1011) State Change CS_ROUTING -> CS_EXECUTE

2010-02-16 16:27:08.542746 [DEBUG] switch_core_session.c:1019 Send signal
OpenZAP/2:1/1011 [BREAK]

2010-02-16 16:27:08.542746 [DEBUG] switch_core_state_machine.c:341
(OpenZAP/2:1/1011) State ROUTING going to sleep

2010-02-16 16:27:08.542746 [DEBUG] switch_core_state_machine.c:314
(OpenZAP/2:1/1011) Running State Change CS_EXECUTE

2010-02-16 16:27:08.542746 [DEBUG] switch_core_state_machine.c:348
(OpenZAP/2:1/1011) State EXECUTE

2010-02-16 16:27:08.542746 [DEBUG] mod_openzap.c:434 OpenZAP/2:1/1011
CHANNEL EXECUTE

2010-02-16 16:27:08.542746 [DEBUG] switch_core_state_machine.c:159
OpenZAP/2:1/1011 Standard EXECUTE

2010-02-16 16:27:08.542746 [DEBUG] switch_core_session.c:1521 Application
disable_ec Requires media! pre_answering channel OpenZAP/2:1/1011

2010-02-16 16:27:08.542746 [DEBUG] switch_core_session.c:1523
OpenZAP/2:1/1011 receive message [PROGRESS]

2010-02-16 16:27:08.542746 [DEBUG] mod_openzap.c:960 Changing state on 2:1
from IDLE to UP

2010-02-16 16:27:08.542746 [NOTICE] mod_openzap.c:961 Channel
[OpenZAP/2:1/1011] has been answered

2010-02-16 16:27:08.542746 [DEBUG] switch_channel.c:182 OpenZAP/2:1/1011
receive message [AUDIO_SYNC]

2010-02-16 16:27:08.542746 [DEBUG] switch_core_session.c:634 Send signal
OpenZAP/2:1/1011 [BREAK]

EXECUTE OpenZAP/2:1/1011 disable_ec()

2010-02-16 16:27:08.542746 [INFO] mod_openzap.c:2951 Echo Canceller Disabled

EXECUTE OpenZAP/2:1/1011 answer()

EXECUTE OpenZAP/2:1/1011 playback(silence_stream://2000)

2010-02-16 16:27:08.542746 [DEBUG] switch_ivr_play_say.c:1162 Codec
Activated L16 at 8000hz 1 channels 20ms

2010-02-16 16:27:08.542746 [DEBUG] switch_core_io.c:652 OpenZAP/2:1/1011
receive message [TRANSCODING_NECESSARY]

2010-02-16 16:27:08.562747 [DEBUG] ozmod_analog.c:450 Executing state
handler on 2:1 for UP

2010-02-16 16:27:08.562747 [DEBUG] mod_openzap.c:1463 got FXS sig [UP]

2010-02-16 16:27:10.522277 [DEBUG] switch_ivr_play_say.c:1454 done playing
file

EXECUTE OpenZAP/2:1/1011 rxfax(/tmp/rxfax.tif)

2010-02-16 16:27:10.522277 [DEBUG] mod_fax.c:591 Raw read codec activation
Success L16 20000

2010-02-16 16:27:10.522277 [DEBUG] switch_core_codec.c:112 OpenZAP/2:1/1011
Push codec L16:10

2010-02-16 16:27:10.522277 [DEBUG] mod_fax.c:607 Raw write codec activation
Success L16

2010-02-16 16:27:10.522277 [DEBUG] switch_channel.c:182 OpenZAP/2:1/1011
receive message [AUDIO_SYNC]

2010-02-16 16:27:10.542275 [DEBUG] switch_core_io.c:234 OpenZAP/2:1/1011
receive message [TRANSCODING_NECESSARY]

2010-02-16 16:28:14.982013 [DEBUG] ozmod_analog.c:788 EVENT [ONHOOK][2:1]
STATE [UP]

2010-02-16 16:28:14.982013 [DEBUG] ozmod_analog.c:824 Changing state on 2:1
from UP to DOWN

2010-02-16 16:28:14.992014 [DEBUG] ozmod_analog.c:450 Executing state
handler on 2:1 for DOWN

2010-02-16 16:28:14.992014 [DEBUG] mod_openzap.c:1463 got FXS sig [STOP]

2010-02-16 16:28:14.992014 [NOTICE] mod_openzap.c:1554 Hangup
OpenZAP/2:1/1011 [CS_EXECUTE] [NORMAL_CLEARING]

2010-02-16 16:28:14.992014 [DEBUG] switch_channel.c:1976 Send signal
OpenZAP/2:1/1011 [KILL]

2010-02-16 16:28:15.002011 [ERR] mod_fax.c:666 Cannot write frame [datalen:
320, samples: 160]

2010-02-16 16:28:15.002011 [DEBUG] mod_fax.c:167
============================================================================
==

2010-02-16 16:28:15.002011 [DEBUG] mod_fax.c:174 Fax successfully received.

2010-02-16 16:28:15.002011 [DEBUG] mod_fax.c:185 Remote station id: 206 742
3831

2010-02-16 16:28:15.002011 [DEBUG] mod_fax.c:186 Local station id:  SpanDSP
Fax Ident

2010-02-16 16:28:15.002011 [DEBUG] mod_fax.c:187 Pages transferred: 1

2010-02-16 16:28:15.002011 [DEBUG] mod_fax.c:189 Total fax pages:   1

2010-02-16 16:28:15.002011 [DEBUG] mod_fax.c:190 Image resolution:
8031x3850

2010-02-16 16:28:15.002011 [DEBUG] mod_fax.c:191 Transfer Rate:     9600

2010-02-16 16:28:15.002011 [DEBUG] mod_fax.c:193 ECM status         off

2010-02-16 16:28:15.002011 [DEBUG] mod_fax.c:194 remote country:

2010-02-16 16:28:15.002011 [DEBUG] mod_fax.c:195 remote vendor:

2010-02-16 16:28:15.002011 [DEBUG] mod_fax.c:196 remote model:

2010-02-16 16:28:15.002011 [DEBUG] mod_fax.c:198
============================================================================
==

 

I traced the error message as coming from switch_core_session_write_frame in
src/switch_core_io.c which returns SWITCH_STATUS_FALSE.

 

Thanks for any information,

Robert

 

 

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100216/5d51c234/attachment.html 

From brian at freeswitch.org  Tue Feb 16 17:07:29 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 16 Feb 2010 19:07:29 -0600
Subject: [Freeswitch-users] mod_fax receives fax to file but logs error
	msg
In-Reply-To: <02F74983D688435EA3F6954B48A14D6D@greyhawk.tonecommander.com>
References: <02F74983D688435EA3F6954B48A14D6D@greyhawk.tonecommander.com>
Message-ID: <80AC3154-4DC5-4432-BEF7-442C4DB47553@freeswitch.org>

Usually means what it says... I think thats harmless if the fax worked.

/b

On Feb 16, 2010, at 7:01 PM, Robert Hadley wrote:

> I have been playing around with mod_fax and can successfully receive a fax to file.  However, while doing so mod_fax is logging an error message.  Does anybody know what this error means?

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100216/a4d051e5/attachment-0001.html 

From anthony.minessale at gmail.com  Tue Feb 16 17:51:15 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 16 Feb 2010 19:51:15 -0600
Subject: [Freeswitch-users] Multiple versions of FreeSWITCH on one box
In-Reply-To: <00e401caaf5b$71cdfc10$5569f430$@net>
References: <00cb01caaf50$8a1dbe50$9e593af0$@net> <4B7B2131.4010003@gmail.com>
	<00e401caaf5b$71cdfc10$5569f430$@net>
Message-ID: <191c3a031002161751p719a177ah8acac55864817c37@mail.gmail.com>

it makes no difference at all what version is was as long as it has
alternate config paths and uses a different ip/port


On Tue, Feb 16, 2010 at 4:57 PM, Adam Ford <lists at redbonez.net> wrote:

>  Yeah, that is what I was referencing when I said ?According to the wiki
> it is possible?.? My question was if it matter if they were different
> versions of FreeSWITCH (1.0.4 and 1.0.5).  I will take this as a no.
>
>
>
> Thanks you guys.
>
> -Adam
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *
> leo.zibi at gmail.com
> *Sent:* Tuesday, February 16, 2010 3:50 PM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Multiple versions of FreeSWITCH on one
> box
>
>
>
> http://wiki.freeswitch.org/wiki/Deployment_Setup
>
> Adam Ford wrote:
>
> According to the wiki it is possible to run multiple FreeSWITCH instances
> on one box.  Are there any known issues with having those instances be
> different versions?
>
>
>
> Unfortunately I don?t have a separate box to test the migration from 1.0.4
> to 1.0.5, so I am looking to build a separate instance of 1.0.5 on the same
> box.
>
>
>
> -Adam
>
>
>
>
>
>
>
>
>
> ------------------------------
>
>
>
> _______________________________________________
>
> FreeSWITCH-users mailing list
>
> FreeSWITCH-users at lists.freeswitch.org
>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
> http://www.freeswitch.org
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100216/1160aa68/attachment.html 

From anthony.minessale at gmail.com  Tue Feb 16 17:53:30 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 16 Feb 2010 19:53:30 -0600
Subject: [Freeswitch-users] Broadsoft SCA w/ Cisco SPA-5xx series
In-Reply-To: <8b1c9cda1002160738r619b2a3cs3bf5dd7d1322121e@mail.gmail.com>
References: <8b1c9cda1002141103te523643ob16db25e3bfa877f@mail.gmail.com>
	<0dd701caae7e$3c108b70$b431a250$@com>
	<8b1c9cda1002151253t32c22663wf66245e0220a4e6@mail.gmail.com>
	<191c3a031002151304h54c262c5o8ffd5c195e0512d2@mail.gmail.com>
	<0e7301caae84$58b47ce0$0a1d76a0$@com>
	<8b1c9cda1002151456s5dec6dbbtaa2063e0f70fe5dd@mail.gmail.com>
	<191c3a031002151531k148d8504w15e6f4c6c1e3ccd@mail.gmail.com>
	<8b1c9cda1002152225i7700bdfcgd2c112bf8180e36@mail.gmail.com>
	<191c3a031002160641q61cb1c53r38add08fc32f3a5a@mail.gmail.com>
	<8b1c9cda1002160738r619b2a3cs3bf5dd7d1322121e@mail.gmail.com>
Message-ID: <191c3a031002161753g27878c85nb20ebb86d9646dc1@mail.gmail.com>

ideally they should have the host name in the packets to match what your
configured domain is so when it does invites the domain is in the to from
etc.  your external phones are putting the ip in the packets which do not
match the domain name.

Another solution is to set the domain to be that IP addr in your config.



On Tue, Feb 16, 2010 at 9:38 AM, Gabriel Kuri <gkuri at ieee.org> wrote:

> The phones are currently setup with the domain in their "Proxy" field
> and set to use SRV to lookup the IP. The "Outbound Proxy" field is
> left empty. How should the phones be setup? The Proxy field with the
> domain and Outbound Proxy set to the IP?
>
> Thanks,
> Gabe
>
> On Tue, Feb 16, 2010 at 6:41 AM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
> > as I expected, you have IP addrs in the table which do not match your
> domain
> > name.
> > the phones behind nat should have your domain name in them same as the
> local
> > phones.
> > And the proxy addr should be set to the ip.
> >
> > If the IP and the DOMAIN do not match you will get mismatches.
> > Most people make the false assumption that this is like dns where the ip
> and
> > hostname are interchangeable.
> >
> > We can look at making a patch to force the hostname to always be the
> right
> > value in the db like we do for reg possibly.
> >
> >
> >
> > On Tue, Feb 16, 2010 at 12:25 AM, Gabriel Kuri <gkuri at ieee.org> wrote:
> >>
> >> Yeah, the domain name matches on the internal profile.
> >>
> >> Thanks for all your help, I can arrange ssh access tomorrow, today
> >> just wasn't one of those good days to do so, I've been running in and
> >> out too much to coordinate it.
> >>
> >> Here's the pastebin for the sip_dialogs table while the call is up ...
> >>
> >>     http://pastebin.freeswitch.org/12159
> >>
> >> Thanks,
> >> Gabe
> >>
> >>
> >> On Mon, Feb 15, 2010 at 3:31 PM, Anthony Minessale
> >> <anthony.minessale at gmail.com> wrote:
> >> > Do the domain names match on what the remote phones are using?
> >> >
> >> > When the call is active, can you attach to sqlite with the sqlite3 app
> >> > and
> >> > select * from sip_dialogs
> >> >
> >> > sqlite3 /usr/local/freeswitch/db/sofia_reg_internal.db
> >> >> select * from sip_dialogs;
> >> >
> >> > remember to do it while the call is up.
> >> >
> >> >
> >> > I am going to bet the domain name in that table is not the same as
> your
> >> > actual domain.
> >> >
> >> >
> >> > I tried to make this easier by asking to ssh to your box and work with
> >> > you
> >> > to fix it but now 9 hours later its starting to resemble diffusing a
> >> > bomb
> >> > over a telegraph wire.
> >> >
> >> >
> >> >
> >> >
> >> > On Mon, Feb 15, 2010 at 4:56 PM, Gabriel Kuri <gkuri at ieee.org> wrote:
> >> >>
> >> >> Yes, the two phones being called (SPA-509Gs) have SCA enabled and it
> >> >> works when making outgoing calls from those phones. But incoming
> calls
> >> >> to those two phones don't seem to have the line key light up on the
> >> >> other phone when one of the phones is answered (same extension).
> >> >>
> >> >> Thanks,
> >> >> Gabe
> >> >>
> >> >> On Mon, Feb 15, 2010 at 1:17 PM, Peder <peder at networkoblivion.com>
> >> >> wrote:
> >> >> > On the phone itself, do you have the line set to shared and
> >> >> > ?Broadsoft
> >> >> > SCA?
> >> >> > enabled?
> >> >> >
> >> >> >
> >> >> >
> >> >> > From: freeswitch-users-bounces at lists.freeswitch.org
> >> >> > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf
> Of
> >> >> > Anthony
> >> >> > Minessale
> >> >> > Sent: Monday, February 15, 2010 3:04 PM
> >> >> > To: freeswitch-users at lists.freeswitch.org
> >> >> > Subject: Re: [Freeswitch-users] Broadsoft SCA w/ Cisco SPA-5xx
> series
> >> >> >
> >> >> >
> >> >> >
> >> >> > you are missing something because you have no seize events when you
> >> >> > go
> >> >> > on
> >> >> > and off hook.
> >> >> > is every phone in the correct mode?
> >> >> >
> >> >> > On Mon, Feb 15, 2010 at 2:53 PM, Gabriel Kuri <gkuri at ieee.org>
> wrote:
> >> >> >
> >> >> > No, that was a typo. I have it correct in the config file.
> >> >> >
> >> >> > Gabe
> >> >> >
> >> >> > On Mon, Feb 15, 2010 at 12:34 PM, Peder <peder at networkoblivion.com
> >
> >> >> > wrote:
> >> >> >
> >> >> >> Is this a typo "managed-shared-appeareance=true" or is there an
> >> >> >> extra e
> >> >> >> in
> >> >> >> appearance in your config?
> >> >> >>
> >> >> >> -----Original Message-----
> >> >> >> From: freeswitch-users-bounces at lists.freeswitch.org
> >> >> >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf
> Of
> >> >> >> Gabriel
> >> >> >> Kuri
> >> >> >> Sent: Monday, February 15, 2010 1:48 PM
> >> >> >> To: freeswitch-users at lists.freeswitch.org
> >> >> >> Subject: Re: [Freeswitch-users] Broadsoft SCA w/ Cisco SPA-5xx
> >> >> >> series
> >> >> >>
> >> >> >> OK, I don't know what happened there, here's another one with the
> >> >> >> NOTIFYs. I'm on trunk rev 16633 and I have
> >> >> >> "managed-shared-appeareance=true" on the internal profile. I'm
> just
> >> >> >> making calls between internal phones.
> >> >> >>
> >> >> >>     http://pastebin.freeswitch.org/12153
> >> >> >>
> >> >> >> Thanks,
> >> >> >> Gabe
> >> >> >>
> >> >> >> On Mon, Feb 15, 2010 at 11:04 AM, Anthony Minessale
> >> >> >> <anthony.minessale at gmail.com> wrote:
> >> >> >>> I don't see any notifies at all in this trace do the profiles in
> >> >> >>> question
> >> >> >>> have:
> >> >> >>> manage-shared-appearance set to true?
> >> >> >>> and are you on latest trunk?
> >> >> >>>
> >> >> >>>
> >> >> >>> On Mon, Feb 15, 2010 at 12:55 PM, Anthony Minessale
> >> >> >>> <anthony.minessale at gmail.com> wrote:
> >> >> >>>>
> >> >> >>>> we log the sql stmts on err so they are red and easier to read.
> >> >> >>>>
> >> >> >>>>
> >> >> >>>>
> >> >> >>>> On Mon, Feb 15, 2010 at 12:48 PM, Gabriel Kuri <gkuri at ieee.org>
> >> >> >>>> wrote:
> >> >> >>>>>
> >> >> >>>>> Here's a call from ext 2551 to 2552 (two phones). I noticed a
> >> >> >>>>> bunch
> >> >> >>>>> of
> >> >> >>>>> errors related to SQL UPDATE for presence ...
> >> >> >>>>>
> >> >> >>>>>     http://pastebin.freeswitch.org/12152
> >> >> >>>>>
> >> >> >>>>> Thanks,
> >> >> >>>>> Gabe
> >> >> >>>>>
> >> >> >>>>>
> >> >> >>>>> On Mon, Feb 15, 2010 at 8:32 AM, Anthony Minessale
> >> >> >>>>> <anthony.minessale at gmail.com> wrote:
> >> >> >>>>> > it should be active not seized.
> >> >> >>>>> > seized is when you take it off hook.
> >> >> >>>>> >
> >> >> >>>>> > We need some more debugging to be sure.
> >> >> >>>>> > Can we work in real time on it or can you get a more detailed
> >> >> >>>>> > log?
> >> >> >>>>> >
> >> >> >>>>> > edit sofia.conf.xml and add the param to the "settings"
> >> >> >>>>> > section.
> >> >> >>>>> >
> >> >> >>>>> > <param name="debug-sla" value="100"/>
> >> >> >>>>> >
> >> >> >>>>> >
> >> >> >>>>> > then restart and enable sip trace and debug level
> >> >> >>>>> >
> >> >> >>>>> > //do this for every profile involved in the call.
> >> >> >>>>> > sofia profile <profilename> siptrace on
> >> >> >>>>> >
> >> >> >>>>> > //also do this
> >> >> >>>>> > console loglevel debug
> >> >> >>>>> >
> >> >> >>>>> >
> >> >> >>>>> > if you can let us ssh, we can do all the for you if you can
> >> >> >>>>> > make
> >> >> >>>>> > the
> >> >> >>>>> > test
> >> >> >>>>> > calls.
> >> >> >>>>> >
> >> >> >>>>> >
> >> >> >>>>> >
> >> >> >>>>> >
> >> >> >>>>> > On Sun, Feb 14, 2010 at 11:59 PM, Gabriel Kuri <
> gkuri at ieee.org>
> >> >> >>>>> > wrote:
> >> >> >>>>> >>
> >> >> >>>>> >> BTW, here's a copy of the NOTIFY (event call-info) sent to
> the
> >> >> >>>>> >> other
> >> >> >>>>> >> phone after the first phone is answered, should this have a
> >> >> >>>>> >> Call-Info
> >> >> >>>>> >> line with an "appearance-state=seized" to turn on the light
> on
> >> >> >>>>> >> the
> >> >> >>>>> >> other phone?
> >> >> >>>>> >>
> >> >> >>>>> >>
> >> >> >>>>> >> NOTIFY sip:2551@<removed>:54446 SIP/2.0.
> >> >> >>>>> >> Via: SIP/2.0/UDP
> >> >> >>>>> >> <removed>:9430;rport;branch=z9hG4bK71pN2cXgH851K.
> >> >> >>>>> >> Max-Forwards: 70.
> >> >> >>>>> >> From: <sip:2551 at mydomain.com <sip%3A2551 at mydomain.com>
> >;tag=XeB6ZrKDevpHp.
> >> >> >>>>> >> To: <sip:2551 at mydomain.com <sip%3A2551 at mydomain.com>
> >;tag=c2d34993aac6ea.
> >> >> >>>>> >> Call-ID: 34c34987-8b6fa786@<removed>.
> >> >> >>>>> >> CSeq: 126950830 NOTIFY.
> >> >> >>>>> >> Contact: <sip:2551@<removed>:9430>.
> >> >> >>>>> >> Expires: 3959.
> >> >> >>>>> >> Call-Info:
> >> >> >>>>> >> <sip:mydomain.com
> >;appearance-index=*;appearance-state=idle.
> >> >> >>>>> >> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16633M.
> >> >> >>>>> >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE,
> >> >> >>>>> >> INFO,
> >> >> >>>>> >> REGISTER, REFER, PRACK, NOTIFY, PUBLISH, SUBSCRIBE.
> >> >> >>>>> >> Supported: 100rel, timer, precondition, path, replaces.
> >> >> >>>>> >> Event: call-info.
> >> >> >>>>> >> Allow-Events: talk, presence, dialog, line-seize, call-info,
> >> >> >>>>> >> sla,
> >> >> >>>>> >> include-session-description, presence.winfo,
> message-summary,
> >> >> >>>>> >> refer.
> >> >> >>>>> >> Subscription-State: active;expires=3959.
> >> >> >>>>> >> Content-Length: 0.
> >> >> >>>>> >>
> >> >> >>>>> >>
> >> >> >>>>> >>
> >> >> >>>>> >> On Sun, Feb 14, 2010 at 7:58 PM, Brian West
> >> >> >>>>> >> <brian at freeswitch.org>
> >> >> >>>>> >> wrote:
> >> >> >>>>> >> > Works fine here... is your box slow or something?
> >> >> >>>>> >> >
> >> >> >>>>> >> > /b
> >> >> >>>>> >> >
> >> >> >>>>> >> > On Feb 14, 2010, at 1:03 PM, Gabriel Kuri wrote:
> >> >> >>>>> >> >
> >> >> >>>>> >> >> I followed Brian's directions from one of the previous
> >> >> >>>>> >> >> threads
> >> >> >>>>> >> >> on
> >> >> >>>>> >> >> configuring the SPA-5xx series phones for Broadsoft SCA
> and
> >> >> >>>>> >> >> set
> >> >> >>>>> >> >> manage-shared-appearance=true in the internal profile.
> SCA
> >> >> >>>>> >> >> appears
> >> >> >>>>> >> >> to
> >> >> >>>>> >> >> be working on outgoing calls between two phones, the line
> >> >> >>>>> >> >> key
> >> >> >>>>> >> >> starts
> >> >> >>>>> >> >> flashing red on the second phone when the first phone
> picks
> >> >> >>>>> >> >> up
> >> >> >>>>> >> >> the
> >> >> >>>>> >> >> receiver to make a call. However on incoming calls, both
> >> >> >>>>> >> >> phones
> >> >> >>>>> >> >> ring
> >> >> >>>>> >> >> (same extension), however when one of the phones picks up
> >> >> >>>>> >> >> the
> >> >> >> line,
> >> >> >>>>> >> >> the second phone's line key doesn't flash red or show the
> >> >> >>>>> >> >> first
> >> >> >>>>> >> >> phone
> >> >> >>>>> >> >> on that incoming call. Any ideas? Does shared appearance
> >> >> >>>>> >> >> only
> >> >> >>>>> >> >> work
> >> >> >>>>> >> >> on
> >> >> >>>>> >> >> outgoing phone calls?
> >> >> >>>>> >> >>
> >> >> >>>>> >> >> Thanks,
> >> >> >>>>> >> >> Gabe
> >> >> >>>>> >> >
> >> >> >>>>> >> >
> >> >> >>>>> >> > _______________________________________________
> >> >> >>>>> >> > FreeSWITCH-users mailing list
> >> >> >>>>> >> > FreeSWITCH-users at lists.freeswitch.org
> >> >> >>>>> >> >
> >> >> >>>>> >> >
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >>>>> >> >
> >> >> >>>>> >> >
> >> >> >>
> >> >> >>
> >> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >>>>> >> > http://www.freeswitch.org
> >> >> >>>>> >> >
> >> >> >>>>> >>
> >> >> >>>>> >> _______________________________________________
> >> >> >>>>> >> FreeSWITCH-users mailing list
> >> >> >>>>> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> >>>>> >>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >>>>> >>
> >> >> >>>>> >>
> >> >> >>
> >> >> >>
> >> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >>>>> >> http://www.freeswitch.org
> >> >> >>>>> >
> >> >> >>>>> >
> >> >> >>>>> >
> >> >> >>>>> > --
> >> >> >>>>> > Anthony Minessale II
> >> >> >>>>> >
> >> >> >>>>> > FreeSWITCH http://www.freeswitch.org/
> >> >> >>>>> > ClueCon http://www.cluecon.com/
> >> >> >>>>> > Twitter: http://twitter.com/FreeSWITCH_wire
> >> >> >>>>> >
> >> >> >>>>> > AIM: anthm
> >> >> >>>>> > MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
> >> >> >>>>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >> >> >>>>> > IRC: irc.freenode.net #freeswitch
> >> >> >>>>> >
> >> >> >>>>> > FreeSWITCH Developer Conference
> >> >> >>>>> > sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> >> >> >>>>> > iax:guest at conference.freeswitch.org/888
> >> >> >>>>> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >> >> >>>>> > pstn:+19193869900
> >> >> >>>>> >
> >> >> >>>>> > _______________________________________________
> >> >> >>>>> > FreeSWITCH-users mailing list
> >> >> >>>>> > FreeSWITCH-users at lists.freeswitch.org
> >> >> >>>>> >
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >>>>> >
> >> >> >>>>> >
> >> >> >>
> >> >> >>
> >> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >>>>> > http://www.freeswitch.org
> >> >> >>>>> >
> >> >> >>>>> >
> >> >> >>>>>
> >> >> >>>>> _______________________________________________
> >> >> >>>>> FreeSWITCH-users mailing list
> >> >> >>>>> FreeSWITCH-users at lists.freeswitch.org
> >> >> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >>>>>
> >> >> >>>>>
> >> >> >>>>>
> >> >> >>>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >>>>> http://www.freeswitch.org
> >> >> >>>>
> >> >> >>>>
> >> >> >>>>
> >> >> >>>> --
> >> >> >>>> Anthony Minessale II
> >> >> >>>>
> >> >> >>>> FreeSWITCH http://www.freeswitch.org/
> >> >> >>>> ClueCon http://www.cluecon.com/
> >> >> >>>> Twitter: http://twitter.com/FreeSWITCH_wire
> >> >> >>>>
> >> >> >>>> AIM: anthm
> >> >> >>>> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
> >> >> >>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >> >> >>>> IRC: irc.freenode.net #freeswitch
> >> >> >>>>
> >> >> >>>> FreeSWITCH Developer Conference
> >> >> >>>> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> >> >> >>>> iax:guest at conference.freeswitch.org/888
> >> >> >>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >> >> >>>> pstn:+19193869900
> >> >> >>>
> >> >> >>>
> >> >> >>>
> >> >> >>> --
> >> >> >>> Anthony Minessale II
> >> >> >>>
> >> >> >>> FreeSWITCH http://www.freeswitch.org/
> >> >> >>> ClueCon http://www.cluecon.com/
> >> >> >>> Twitter: http://twitter.com/FreeSWITCH_wire
> >> >> >>>
> >> >> >>> AIM: anthm
> >> >> >>> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
> >> >> >>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >> >> >>> IRC: irc.freenode.net #freeswitch
> >> >> >>>
> >> >> >>> FreeSWITCH Developer Conference
> >> >> >>> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> >> >> >>> iax:guest at conference.freeswitch.org/888
> >> >> >>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >> >> >>> pstn:+19193869900
> >> >> >>>
> >> >> >>> _______________________________________________
> >> >> >>> FreeSWITCH-users mailing list
> >> >> >>> FreeSWITCH-users at lists.freeswitch.org
> >> >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >>>
> >> >> >>>
> >> >> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >>> http://www.freeswitch.org
> >> >> >>>
> >> >> >>>
> >> >> >>
> >> >> >> _______________________________________________
> >> >> >> FreeSWITCH-users mailing list
> >> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >>
> >> >> >>
> >> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >> http://www.freeswitch.org
> >> >> >>
> >> >> >>
> >> >> >> _______________________________________________
> >> >> >> FreeSWITCH-users mailing list
> >> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >>
> >> >> >>
> >> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >> http://www.freeswitch.org
> >> >> >>
> >> >> >
> >> >> > _______________________________________________
> >> >> > FreeSWITCH-users mailing list
> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >
> >> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> > http://www.freeswitch.org
> >> >> >
> >> >> >
> >> >> > --
> >> >> > Anthony Minessale II
> >> >> >
> >> >> > FreeSWITCH http://www.freeswitch.org/
> >> >> > ClueCon http://www.cluecon.com/
> >> >> > Twitter: http://twitter.com/FreeSWITCH_wire
> >> >> >
> >> >> > AIM: anthm
> >> >> > MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
> >> >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >> >> > IRC: irc.freenode.net #freeswitch
> >> >> >
> >> >> > FreeSWITCH Developer Conference
> >> >> > sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> >> >> > iax:guest at conference.freeswitch.org/888
> >> >> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >> >> > pstn:+19193869900
> >> >> >
> >> >> > _______________________________________________
> >> >> > FreeSWITCH-users mailing list
> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >
> >> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> > http://www.freeswitch.org
> >> >> >
> >> >> >
> >> >>
> >> >> _______________________________________________
> >> >> FreeSWITCH-users mailing list
> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>
> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> http://www.freeswitch.org
> >> >
> >> >
> >> >
> >> > --
> >> > Anthony Minessale II
> >> >
> >> > FreeSWITCH http://www.freeswitch.org/
> >> > ClueCon http://www.cluecon.com/
> >> > Twitter: http://twitter.com/FreeSWITCH_wire
> >> >
> >> > AIM: anthm
> >> > MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
> >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >> > IRC: irc.freenode.net #freeswitch
> >> >
> >> > FreeSWITCH Developer Conference
> >> > sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> >> > iax:guest at conference.freeswitch.org/888
> >> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >> > pstn:+19193869900
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> > Twitter: http://twitter.com/FreeSWITCH_wire
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> > IRC: irc.freenode.net #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> > iax:guest at conference.freeswitch.org/888
> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > pstn:+19193869900
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100216/b3ccf694/attachment-0001.html 

From anthony.minessale at gmail.com  Tue Feb 16 18:08:38 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 16 Feb 2010 20:08:38 -0600
Subject: [Freeswitch-users] mod_fax receives fax to file but logs error
	msg
In-Reply-To: <80AC3154-4DC5-4432-BEF7-442C4DB47553@freeswitch.org>
References: <02F74983D688435EA3F6954B48A14D6D@greyhawk.tonecommander.com>
	<80AC3154-4DC5-4432-BEF7-442C4DB47553@freeswitch.org>
Message-ID: <191c3a031002161808u32282f7esbee131305a607786@mail.gmail.com>

when a channel is hungup the read and write will fail to stop the media,
this is typical.


On Tue, Feb 16, 2010 at 7:07 PM, Brian West <brian at freeswitch.org> wrote:

> Usually means what it says... I think thats harmless if the fax worked.
>
> /b
>
> On Feb 16, 2010, at 7:01 PM, Robert Hadley wrote:
>
> I have been playing around with mod_fax and can successfully receive a fax
> to file.  However, while doing so mod_fax is logging an error message.  Does
> anybody know what this error means?
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100216/1c9a5e9b/attachment.html 

From infos at madovsky.org  Tue Feb 16 20:02:42 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 16 Feb 2010 23:02:42 -0500
Subject: [Freeswitch-users] freeswitch as proxy
Message-ID: <82FAF11BF2BB4A0787E86706EABF755E@MOBILEE1705>

Hi,

First I'd like to felicitate the huge work of the freeswitch founders, their patience and humbleness
(Anthony Minessale (is there 1 and 2 ?), Michael Jerris, Brian West and Others ) and people need to
know and understand how hard is to maintain open source for years...

well, maybe I didn't look for very well in archives and google so I hope my request won't be an old one....
I set freeswitch with proxy_media and blind reg and auth on true since I want everybody who uses my network
use a softphone to register their own sip account (whatever domain outside my network).

user at hisdomain -> myproxy IP -> hisdomain registrar -> confirm register and wait call.
user at hisdomain calls other at otherdomain everywhere in the world.
other at otherdomain can also call user at hisdomain

is it possible ? if yes is there any link example ? I guess maybe it's a dialplan rule....

Sorry I'm novice yet

Regards

Franck
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100216/49356588/attachment.html 

From yehavi.bourvine at gmail.com  Tue Feb 16 20:49:30 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Wed, 17 Feb 2010 06:49:30 +0200
Subject: [Freeswitch-users] SQL ERR: library routine called out of
	sequence
In-Reply-To: <CBE5A464-0321-4A55-9195-7CAB0E3405AE@freeswitch.org>
References: <d361ca7b1002160656w7d616f6dn5e80087ebde94dbe@mail.gmail.com>
	<191c3a031002160705w4050c3cdk2bf2922e9aff5ba3@mail.gmail.com>
	<d361ca7b1002160727ucb6e0c8id62e803757333213@mail.gmail.com>
	<191c3a031002160755q569fe2ednfefb01b6d9cf7f0e@mail.gmail.com>
	<d361ca7b1002161059w1ef9484dr87e69a33aa2b62fc@mail.gmail.com>
	<191c3a031002161110s637a48a2r717ae94557fe9e9a@mail.gmail.com>
	<d361ca7b1002161130k3eb2d88by3100b30a9d29bd51@mail.gmail.com>
	<191c3a031002161230h2b6e6f9wf3f65ddb71f08297@mail.gmail.com>
	<CBE5A464-0321-4A55-9195-7CAB0E3405AE@freeswitch.org>
Message-ID: <d361ca7b1002162049y6810073cj55765ce9fdfc2f48@mail.gmail.com>

Hello,

  I'll try the latest snapshot during the weekend as this is a production
system.

I am using FedoraCore 10 with a kernel from kernel.org (as I recall there
was some issue with Freeswitch and Fedora's kernel). Here is the output of
uname and cpuinfo:



Linux control.huji.ac.il 2.6.32.5 #1 SMP Sat Jan 23 11:17:10 IST 2010 i686
i686 i386 GNU/Linux



processor   : 0

vendor_id   : GenuineIntel

cpu family  : 15

model       : 4

model name  : Intel(R) Xeon(TM) CPU 3.00GHz

stepping    : 1

cpu MHz           : 3000.000

cache size  : 1024 KB

physical id : 0

siblings    : 2

core id           : 0

cpu cores   : 1

apicid            : 0

initial apicid    : 0

fdiv_bug    : no

hlt_bug           : no

f00f_bug    : no

coma_bug    : no

fpu         : yes

fpu_exception     : yes

cpuid level : 5

wp          : yes

flags       : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca cmov
pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe lm constant_tsc
pebs bts pni dtes64 monitor ds_cpl cid cx16 xtpr

bogomips    : 6000.32

clflush size      : 64

cache_alignment   : 128

address sizes     : 36 bits physical, 48 bits virtual

power management:



processor   : 1

vendor_id   : GenuineIntel

cpu family  : 15

model       : 4

model name  : Intel(R) Xeon(TM) CPU 3.00GHz

stepping    : 1

cpu MHz           : 3000.000

cache size  : 1024 KB

physical id : 0

siblings    : 2

core id           : 0

cpu cores   : 1

apicid            : 1

initial apicid    : 1

fdiv_bug    : no

hlt_bug           : no

f00f_bug    : no

coma_bug    : no

fpu         : yes

fpu_exception     : yes

cpuid level : 5

wp          : yes

flags       : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca cmov
pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe lm constant_tsc
pebs bts pni dtes64 monitor ds_cpl cid cx16 xtpr

bogomips    : 5999.17

clflush size      : 64

cache_alignment   : 128

address sizes     : 36 bits physical, 48 bits virtual

power management:



                Thanks! __Yehavi:


 2010/2/16 Brian West <brian at freeswitch.org>

>  What distro are you on and kernel version?
>
> cat /proc/cpuinfo
> uname -a
>
> and such
>
> /b
>
>  On Feb 16, 2010, at 2:30 PM, Anthony Minessale wrote:
>
> Strange, even on abusive testing we have not seen this problem.
>
> please update to latest trunk.
> There was only one change I can think of that may cause your issue and I
> added a patch for it.
> If it persists try setting the sql-in-transactions profile param to false.
>
>
>
>
> On Tue, Feb 16, 2010 at 1:30 PM, Yehavi Bourvine <
> yehavi.bourvine at gmail.com> wrote:
>
>>  Most of the queries are ok, only some fail, thus it doesn't look like
>> permission problem. Furthermore, under 1.0.5pre10 it works for months.
>>
>> Might it be thread unsafe function calls? I've found the following while
>> searching the WEB:
>>
>> *According to the MSDN docs, System.Timers.Timer operates in a thread
>> pool.  If that's the case, your code is breaking the "connections cannot be
>> shared across threads" rule for SQLit*
>>
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100217/d02228a0/attachment-0001.html 

From anthony.minessale at gmail.com  Tue Feb 16 21:39:53 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 16 Feb 2010 23:39:53 -0600
Subject: [Freeswitch-users] Presence PUBLISH Not Updating AfterSoftphone
	OffLine Then Available
In-Reply-To: <68A4C011A72B4ABA91A185F32480EB99@greyhawk.tonecommander.com>
References: <31FEE9072F104AE18B26812978FCA341@greyhawk.tonecommander.com>
	<45963EB5-DFE5-43FA-AFCC-857E1E719E2D@jerris.com>
	<191c3a031002091557jfc9b486lb832ea538883bce9@mail.gmail.com>
	<68A4C011A72B4ABA91A185F32480EB99@greyhawk.tonecommander.com>
Message-ID: <191c3a031002162139p57c94046y660fb8d01fd76f46@mail.gmail.com>

You see one case at the top where it sends a notify and more where it doesnt
.  You have the sql stmts right there (they are not errs just logging in red
so they are obvious) run them manually and figure out why there are no
matches.  No subscriptions maybe?

Its beginning to sound like a broken record with so many bria isssues, its a
new software afterall and not free like we are, why must we support it so
much?

Also if you are actually concerned with this issue, maybe you can come back
sooner than once every week or 2 weeks.  We quickly lose track of threads
like this that linger for a month, that's what jira is for....

Maybe you can stop by irc or keep an eye on your email client so we can
confirm what you are doing wrong or if we have an interop with bria, a pay
softphone none of us have a copy of........

On Feb 16, 2010 11:18 AM, "Jerry Richards" <jerry.richards at teotech.com>
wrote:

 I got version freeswitch-1.0.5-20100215-0400, built it, and ran it, and I
am seeing the same issue.  That is, once I set the Bria softphone status to
'Appear Offline', FS does not forward presence states until resubscription
time (i.e. tens of minutes later).  I posted a trace at
http://pastebin.freeswitch.org/12164.  At line 359 of the trace, FS is
logging an ERR at sofia_presence.c:662.  Here is the scenario:

1) Set Bria softphone presence state to 'Appear Offline'
2) Subscibing softphones reflect offline status
 3) Set Bria softphone presence state to 'Available'
4) *** Subscibing softphones do not get status update ***

Thanks And Best Regards,
Jerry


 ------------------------------
*From:* Anthony Minessale [mailto:anthony.minessale at gmail.com]
*Sent:* Tuesday, February 09, 2010 3:58 PM
*To:* freeswitch-users at lists.freeswitch.org
*Subject:* Re: [Freeswitch-users] Presence PUBLISH Not Updating
AfterSoftphone OffLine Then Available

> he means update to trunk first then try it again obviously.
>
>
> On Tue, Feb 9, 2010 at 3:10 PM, ...


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100216/4841dd82/attachment.html 

From ledoktre at meanie.us  Tue Feb 16 18:53:58 2010
From: ledoktre at meanie.us (Doc)
Date: Tue, 16 Feb 2010 20:53:58 -0600
Subject: [Freeswitch-users] Greetings and a couple of questions
Message-ID: <4B7B5A46.3090304@meanie.us>

First, to all, greetings.

I am just beginning a quest in trying to setup a simple FS box to route 
my incoming skype account to a SIP ATA (SPA-1001).

I have this installed on Ubuntu 8.04 (since I read that it came with a 
stock tickless kernel with 100HZ tick).  I have (as far as I can tell) 
compiled the alsa 1.0.20 drivers and included the mod_skypiax dummy 
file.  I followed install instructions in the FS wiki for Hardy & FS & 
skypiax.

1) I am able to see a call come in, and it gets routed to the sample IVR 
to start.  The first thing off is that when I dial from PSTN -> 
Skype-In, it does not let me push any buttons.  If I launch a second 
skype client, and dial the skype user on FS directly, it works fine.  
Any ideas?

2) No matter how Skype works, when I hang up, it throws 3 or 4 errors 
(2010-02-16 20:41:45.384633 [ERR] mod_skypiax.c:826 rev 
16619M[(nil)|37     ][ERRORA  826  ][interface1][-1, 5,21] EXIT? 
sent=-1), and when I try subsequent dials, I get pages and pages of this 
error, and no audio.  Eventually it crashes FS.  Any ideas?

3) When dialing extension to extension, or even testing out the IVR, it 
all works fine - no errors.  The only thing it does make me do, and I 
haven't tracked it down yet is it waits like 30 seconds before 
responding (unless I press the # after I type the extension number).  
Any ideas?

4) When I run startskype.sh, I see these errors (and are they worth 
concerning?) :

expected keysym, got XF86KbdLightOnOff: line 70 of pc
expected keysym, got XF86KbdBrightnessDown: line 71 of pc
expected keysym, got XF86KbdBrightnessUp: line 72 of pc
Could not init font path element /usr/share/fonts/X11/cyrillic, removing 
from list!
Could not init font path element 
/var/lib/defoma/x-ttcidfont-conf.d/dirs/TrueType, removing from list!
ALSA lib pcm.c:2211:(snd_pcm_open_noupdate) Unknown PCM null
ALSA lib pcm.c:2211:(snd_pcm_open_noupdate) Unknown PCM null

Thanks for the patience in letting me email to the group.  I used 
Asterisk in the past, and even though I am finding a bit of a learning 
curve using FS, I am enjoying it.  Hopefully someone will have some 
insight at least into where or how I can settle the above items down, 
and I can start building a proper dialplan.

Thanks,

Doc


From ledoktre at meanie.us  Tue Feb 16 19:13:45 2010
From: ledoktre at meanie.us (Doc)
Date: Tue, 16 Feb 2010 21:13:45 -0600
Subject: [Freeswitch-users] One more thing..
Message-ID: <4B7B5EE9.4020705@meanie.us>

Greetings one more time,

I just remembered.  I also ran into one other issue in my testing.

When I route the incoming skype call to an extension (Sipura SPA 1001), 
it plays hold music on the callers side (good....), and when I pick up 
the extension, calling party drops, internal phone is left with a fast 
busy signal :

2010-02-16 21:09:31.687204 [NOTICE] sofia.c:4690 Channel 
[sofia/internal/sip:1001 at 10.24.72.12:5060] has been answered
2010-02-16 21:09:31.987714 [WARNING] mod_sofia.c:918 We were told to use 
ptime 3 but what they meant to say was 20
This issue has so far been identified to happen on the following broken 
platforms/devices:
Linksys/Sipura aka Cisco
ShoreTel
Sonus/L3
We will try to fix it but some of the devices on this list are so broken 
who knows what will happen..
2010-02-16 21:09:32.027243 [ERR] switch_core_io.c:121 
sofia/internal/sip:1001 at 10.24.72.12:5060 has no read codec.

First thing I notice is this unusual error about the ptime?  and the one 
that hangs me up I think is the "sofia... has no read codec".  I have 
been poking around my configuration - any suggestions?

Thanks,

Doc



From moizchinoy at gmail.com  Wed Feb 17 03:18:54 2010
From: moizchinoy at gmail.com (Moiz Chinoy)
Date: Wed, 17 Feb 2010 15:18:54 +0400
Subject: [Freeswitch-users] Dingaling - external rtp supported?
In-Reply-To: <DEA39AD1-0EE9-473A-B78C-CDDF3F100578@freeswitch.org>
References: <29b888f81002160835y6e290319ub0d04c088caa1b79@mail.gmail.com>
	<DEA39AD1-0EE9-473A-B78C-CDDF3F100578@freeswitch.org>
Message-ID: <29b888f81002170318p63622f3fhc8d26ad0e85602d2@mail.gmail.com>

Hi,

FS rev: 16673
Platform: Windows

More details:

FS is behind NAT and machine is running a VPN connection.

FS and GTalk client on the same machine:
--------------------------------------------------------------------------------------------------
jingle profile client.xml has following line:
<param name="rtp-ip" value="$${bind_server_ip}"/>

External SIP call is successfully bridged to GTalk client.


FS and GTalk client on the different machine:
--------------------------------------------------------------------------------------------------
jingle profile client.xml has following lines:
<param name="rtp-ip" value="$${bind_server_ip}"/>
<param name="ext-rtp-ip" value="$${external_rtp_ip}"/>

As soon as external SIP call land and I try to bridge the call to
GTalk client, FS crashes.


NAT Details:
---------------------------
I think my NAT does not support UpNP or PMP. The reason I say it
because when FS starts following message is displayed:

2010-02-17 09:15:42.809674 [INFO] switch_nat.c:409 Scanning for NAT
2010-02-17 09:15:42.809674 [ERR] switch_nat.c:197 Error checking for
PMP [init failed]
2010-02-17 09:15:42.809674 [DEBUG] switch_nat.c:414 Checking for UPnP
2010-02-17 09:15:56.825388 [DEBUG] switch_nat.c:114 No
InternetGatewayDevice, using first entry as default
(http://192.168.16.17:50144/).
2010-02-17 09:15:56.887889 [INFO] switch_nat.c:429 No PMP or UPnP NAT
devices detected!



On Tue, Feb 16, 2010 at 8:41 PM, Brian West <brian at freeswitch.org> wrote:
> can you please update, try again and post a jira?
>
> /b
>
> On Feb 16, 2010, at 10:35 AM, Moiz Chinoy wrote:
>
>> Hi All,
>>
>> In mod_dingaling <param name="ext-rtp-ip"
>> value="$${external_rtp_ip}"/> supported? Whenever I uncomment this
>> line in client.xml (jingle profile) FS crashes as soon a call lands
>> (sip call) and dialplan bridges the call to a gtalk user.
>>
>> I am running FS on windows and build is 16642.
>>
>> --
>> Regards,
>> Moiz Chinoy.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Regards,
Moiz Chinoy.


From scott.torr.fs at letterboxes.org  Wed Feb 17 04:08:23 2010
From: scott.torr.fs at letterboxes.org (Scott Torr)
Date: Wed, 17 Feb 2010 23:08:23 +1100
Subject: [Freeswitch-users] Greetings and a couple of questions
In-Reply-To: <4B7B5A46.3090304@meanie.us>
References: <4B7B5A46.3090304@meanie.us>
Message-ID: <1266408503.10430.1360421639@webmail.messagingengine.com>

On Tue, 16 Feb 2010 20:53 -0600, "Doc" <ledoktre at meanie.us> wrote:
> 1) I am able to see a call come in, and it gets routed to the sample IVR 
> to start.  The first thing off is that when I dial from PSTN -> 
> Skype-In, it does not let me push any buttons.  If I launch a second 
> skype client, and dial the skype user on FS directly, it works fine.  
> Any ideas?

Hi Doc,

When you dial in from the PSTN the 'push button' events are present as
"in band" audio tones.
By default the sample IVR only works on "out of band" DTMF events.

This is why when you call directly from another skype client the 'push
button' events are detected because they are passed as "out of band"
signaling.


Now,
In the dial plan you can tell FS to listen for "in band" audio DTMF
tones using <action application="start_dtmf" />

However,
This currently does not work during a skype call for some reason?
http://jira.freeswitch.org/browse/MODSKYPIAX-66


A work around,
is to sign up for the "Skype SIP Beta" product where the 'push button'
events are sent to FS "out of band".

This conversion is done at the PSTN --> Skype gateway by dedicated DTMF
tone detection hardware.


Skype has either made a business decision, or a technical over sight to
pass DTMF events 'out of band' for only addition fee products.


It has also been reported in New Zealand that even the 'in band' tones
where present one day and actually filtered out the next.
This seems extreme, but either through deliberate action or a technology
change this is what was reported on one blog.

It is unclear to me if this was a technical limitation or a blunt
business decisions, but a audio sample showed the audio tones missing? 


In any case you would not want to rely on 'In band' DTMF' tones when
passed through 'lossy' codecs anyway.

Best to stick with 'out of band' signaling for reliability.


regards,
Scott Torr








From anthony.minessale at gmail.com  Wed Feb 17 05:36:05 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 17 Feb 2010 07:36:05 -0600
Subject: [Freeswitch-users] One more thing..
In-Reply-To: <191c3a031002170535i12c82731r5829417387dd3228@mail.gmail.com>
References: <4B7B5EE9.4020705@meanie.us>
	<191c3a031002170535i12c82731r5829417387dd3228@mail.gmail.com>
Message-ID: <191c3a031002170536g297a816ch2fa94eb2dcfbd16d@mail.gmail.com>

Are you doing this with the latest revision?
You would have to supply more info like a console trace on debug level with
siptrace enabled.

On Feb 17, 2010 1:51 AM, "Doc" <ledoktre at meanie.us> wrote:

Greetings one more time,

I just remembered.  I also ran into one other issue in my testing.

When I route the incoming skype call to an extension (Sipura SPA 1001),
it plays hold music on the callers side (good....), and when I pick up
the extension, calling party drops, internal phone is left with a fast
busy signal :

2010-02-16 21:09:31.687204 [NOTICE] sofia.c:4690 Channel
[sofia/internal/sip:1001 at 10.24.72.12:5060] has been answered
2010-02-16 21:09:31.987714 [WARNING] mod_sofia.c:918 We were told to use
ptime 3 but what they meant to say was 20
This issue has so far been identified to happen on the following broken
platforms/devices:
Linksys/Sipura aka Cisco
ShoreTel
Sonus/L3
We will try to fix it but some of the devices on this list are so broken
who knows what will happen..
2010-02-16 21:09:32.027243 [ERR] switch_core_io.c:121
sofia/internal/sip:1001 at 10.24.72.12:5060 has no read codec.

First thing I notice is this unusual error about the ptime?  and the one
that hangs me up I think is the "sofia... has no read codec".  I have
been poking around my configuration - any suggestions?

Thanks,

Doc


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100217/a192135e/attachment.html 

From anthony.minessale at gmail.com  Wed Feb 17 05:38:50 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 17 Feb 2010 07:38:50 -0600
Subject: [Freeswitch-users] Dingaling - external rtp supported?
In-Reply-To: <29b888f81002170318p63622f3fhc8d26ad0e85602d2@mail.gmail.com>
References: <29b888f81002160835y6e290319ub0d04c088caa1b79@mail.gmail.com>
	<DEA39AD1-0EE9-473A-B78C-CDDF3F100578@freeswitch.org>
	<29b888f81002170318p63622f3fhc8d26ad0e85602d2@mail.gmail.com>
Message-ID: <191c3a031002170538r3b029515i8a000812a70ad558@mail.gmail.com>

Obtain a stack trace from the crash.

On Feb 17, 2010 5:26 AM, "Moiz Chinoy" <moizchinoy at gmail.com> wrote:

Hi,

FS rev: 16673
Platform: Windows

More details:

FS is behind NAT and machine is running a VPN connection.

FS and GTalk client on the same machine:
--------------------------------------------------------------------------------------------------
jingle profile client.xml has following line:
<param name="rtp-ip" value="$${bind_server_ip}"/>

External SIP call is successfully bridged to GTalk client.


FS and GTalk client on the different machine:
--------------------------------------------------------------------------------------------------
jingle profile client.xml has following lines:
<param name="rtp-ip" value="$${bind_server_ip}"/>

<param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
As soon as external SIP call land and I try to bridge the call to
GTalk client, FS crashes.


NAT Details:
---------------------------
I think my NAT does not support UpNP or PMP. The reason I say it
because when FS starts following message is displayed:

2010-02-17 09:15:42.809674 [INFO] switch_nat.c:409 Scanning for NAT
2010-02-17 09:15:42.809674 [ERR] switch_nat.c:197 Error checking for
PMP [init failed]
2010-02-17 09:15:42.809674 [DEBUG] switch_nat.c:414 Checking for UPnP
2010-02-17 09:15:56.825388 [DEBUG] switch_nat.c:114 No
InternetGatewayDevice, using first entry as default
(http://192.168.16.17:50144/).
2010-02-17 09:15:56.887889 [INFO] switch_nat.c:429 No PMP or UPnP NAT
devices detected!




On Tue, Feb 16, 2010 at 8:41 PM, Brian West <brian at freeswitch.org> wrote:
> can you please update...
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100217/f54b8d97/attachment-0001.html 

From yehavi.bourvine at gmail.com  Wed Feb 17 05:49:29 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Wed, 17 Feb 2010 15:49:29 +0200
Subject: [Freeswitch-users] Announcement: FreeSWITCH Add Broadsoft SCA
	Support
In-Reply-To: <9D328485-D4C3-4B14-808A-CB63BBF06DC1@freeswitch.org>
References: <87f2f3b91001111742k691966cfs82515db77c13d4d4@mail.gmail.com>
	<d361ca7b1002082110h4e3c60d0p88412f753444ec3d@mail.gmail.com>
	<016d01caa98e$f6df25f0$e49d71d0$@com>
	<d361ca7b1002090817h4aa3c13dka4d4db8b4adc7606@mail.gmail.com>
	<1C0A1BEE-3480-4A33-916F-52067AF5B9C9@freeswitch.org>
	<d361ca7b1002090838j6c42dcd2of8affb5f281900fd@mail.gmail.com>
	<767B677E-0D19-43A9-89DE-D96EC4A715EC@freeswitch.org>
	<8D062B6E-4691-4BA6-B038-0EC6E8642DDA@jerris.com>
	<9D328485-D4C3-4B14-808A-CB63BBF06DC1@freeswitch.org>
Message-ID: <d361ca7b1002170549g608d6e16vb2eb3f087fff9752@mail.gmail.com>

I am trying to migrate from 1.0.5pre10 to the latest (FreeSWITCH Version
1.0.5-20100216-0400 (16659M)), and shared apearance stopped working...

We have Polycom phones and I use the same configuration for both versions.
With the old version I see that after the phone registers with FreeSwitch,
the server subscribes to the phone for the extension; with the latest
version this does not happen. Furthermore, the table
sip_shared_appearance_dialogs is empty.

I don't find anything I can change on the phone config (the line is already
set to shared).

Here is the relevant config from the sip profile:

    <param name="manage-presence" value="passive"/> (set to TRUE only on one
profile).
    <param name="manage-shared-appearance" value="true"/>
    <param name="dbname" value="share_presence"/>
    <param name="presence-hosts" value="pbx-gr-backup.ip-phones.huji.ac.il
"/>
Any idea where to look?
                           Thanks, __Yehavi:
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100217/f7b628b6/attachment.html 

From anthony.minessale at gmail.com  Wed Feb 17 05:57:13 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 17 Feb 2010 07:57:13 -0600
Subject: [Freeswitch-users] Announcement: FreeSWITCH Add Broadsoft SCA
	Support
In-Reply-To: <191c3a031002170556p5dfd4c55le3c279f194808d92@mail.gmail.com>
References: <87f2f3b91001111742k691966cfs82515db77c13d4d4@mail.gmail.com>
	<016d01caa98e$f6df25f0$e49d71d0$@com>
	<d361ca7b1002090817h4aa3c13dka4d4db8b4adc7606@mail.gmail.com>
	<1C0A1BEE-3480-4A33-916F-52067AF5B9C9@freeswitch.org>
	<d361ca7b1002090838j6c42dcd2of8affb5f281900fd@mail.gmail.com>
	<767B677E-0D19-43A9-89DE-D96EC4A715EC@freeswitch.org>
	<8D062B6E-4691-4BA6-B038-0EC6E8642DDA@jerris.com>
	<9D328485-D4C3-4B14-808A-CB63BBF06DC1@freeswitch.org>
	<d361ca7b1002170549g608d6e16vb2eb3f087fff9752@mail.gmail.com>
	<191c3a031002170556p5dfd4c55le3c279f194808d92@mail.gmail.com>
Message-ID: <191c3a031002170557g5d8c9df9xf40d1b74b9a6f19d@mail.gmail.com>

You have to set the param to sylantro to get that mode.
Or configure your phone to use broadsoft.

On Feb 17, 2010 7:54 AM, "Yehavi Bourvine" <yehavi.bourvine at gmail.com>
wrote:

I am trying to migrate from 1.0.5pre10 to the latest (FreeSWITCH Version
1.0.5-20100216-0400 (16659M)), and shared apearance stopped working...

We have Polycom phones and I use the same configuration for both versions.
With the old version I see that after the phone registers with FreeSwitch,
the server subscribes to the phone for the extension; with the latest
version this does not happen. Furthermore, the table
sip_shared_appearance_dialogs is empty.

I don't find anything I can change on the phone config (the line is already
set to shared).

Here is the relevant config from the sip profile:

    <param name="manage-presence" value="passive"/> (set to TRUE only on one
profile).
    <param name="manage-shared-appearance" value="true"/>


    <param name="dbname" value="share_presence"/>
    <param name="presence-hosts" value="pbx-gr-backup.ip-phones.huji.ac.il
"/>
Any idea where to look?
                           Thanks, __Yehavi:


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100217/35d7982e/attachment.html 

From brian at freeswitch.org  Wed Feb 17 06:26:32 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 17 Feb 2010 08:26:32 -0600
Subject: [Freeswitch-users] Dingaling - external rtp supported?
In-Reply-To: <29b888f81002170318p63622f3fhc8d26ad0e85602d2@mail.gmail.com>
References: <29b888f81002160835y6e290319ub0d04c088caa1b79@mail.gmail.com>
	<DEA39AD1-0EE9-473A-B78C-CDDF3F100578@freeswitch.org>
	<29b888f81002170318p63622f3fhc8d26ad0e85602d2@mail.gmail.com>
Message-ID: <2F1D6EF8-CA94-44D6-B947-6E4B75D14A62@freeswitch.org>

http://wiki.freeswitch.org/wiki/Reporting_Bugs

/b

On Feb 17, 2010, at 5:18 AM, Moiz Chinoy wrote:

> Hi,
> 
> FS rev: 16673
> Platform: Windows
> 
> More details:
> 
> FS is behind NAT and machine is running a VPN connection.
> 
> FS and GTalk client on the same machine:
> --------------------------------------------------------------------------------------------------
> jingle profile client.xml has following line:
> <param name="rtp-ip" value="$${bind_server_ip}"/>
> 
> External SIP call is successfully bridged to GTalk client.
> 
> 
> FS and GTalk client on the different machine:
> --------------------------------------------------------------------------------------------------
> jingle profile client.xml has following lines:
> <param name="rtp-ip" value="$${bind_server_ip}"/>
> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
> 
> As soon as external SIP call land and I try to bridge the call to
> GTalk client, FS crashes.
> 
> 
> NAT Details:
> ---------------------------
> I think my NAT does not support UpNP or PMP. The reason I say it
> because when FS starts following message is displayed:
> 
> 2010-02-17 09:15:42.809674 [INFO] switch_nat.c:409 Scanning for NAT
> 2010-02-17 09:15:42.809674 [ERR] switch_nat.c:197 Error checking for
> PMP [init failed]
> 2010-02-17 09:15:42.809674 [DEBUG] switch_nat.c:414 Checking for UPnP
> 2010-02-17 09:15:56.825388 [DEBUG] switch_nat.c:114 No
> InternetGatewayDevice, using first entry as default
> (http://192.168.16.17:50144/).
> 2010-02-17 09:15:56.887889 [INFO] switch_nat.c:429 No PMP or UPnP NAT
> devices detected!
> 
> 
> 
> On Tue, Feb 16, 2010 at 8:41 PM, Brian West <brian at freeswitch.org> wrote:
>> can you please update, try again and post a jira?
>> 
>> /b
>> 
>> On Feb 16, 2010, at 10:35 AM, Moiz Chinoy wrote:
>> 
>>> Hi All,
>>> 
>>> In mod_dingaling <param name="ext-rtp-ip"
>>> value="$${external_rtp_ip}"/> supported? Whenever I uncomment this
>>> line in client.xml (jingle profile) FS crashes as soon a call lands
>>> (sip call) and dialplan bridges the call to a gtalk user.
>>> 
>>> I am running FS on windows and build is 16642.
>>> 
>>> --
>>> Regards,
>>> Moiz Chinoy.
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
> 
> 
> 
> -- 
> Regards,
> Moiz Chinoy.
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From chris.chen2004 at gmail.com  Wed Feb 17 06:40:36 2010
From: chris.chen2004 at gmail.com (Chris Chen)
Date: Wed, 17 Feb 2010 09:40:36 -0500
Subject: [Freeswitch-users] Announcement: FreeSWITCH Add Broadsoft SCA
	Support
In-Reply-To: <191c3a031002170557g5d8c9df9xf40d1b74b9a6f19d@mail.gmail.com>
References: <87f2f3b91001111742k691966cfs82515db77c13d4d4@mail.gmail.com>
	<d361ca7b1002090817h4aa3c13dka4d4db8b4adc7606@mail.gmail.com>
	<1C0A1BEE-3480-4A33-916F-52067AF5B9C9@freeswitch.org>
	<d361ca7b1002090838j6c42dcd2of8affb5f281900fd@mail.gmail.com>
	<767B677E-0D19-43A9-89DE-D96EC4A715EC@freeswitch.org>
	<8D062B6E-4691-4BA6-B038-0EC6E8642DDA@jerris.com>
	<9D328485-D4C3-4B14-808A-CB63BBF06DC1@freeswitch.org>
	<d361ca7b1002170549g608d6e16vb2eb3f087fff9752@mail.gmail.com>
	<191c3a031002170556p5dfd4c55le3c279f194808d92@mail.gmail.com>
	<191c3a031002170557g5d8c9df9xf40d1b74b9a6f19d@mail.gmail.com>
Message-ID: <507898381002170640u5586698cg310f9cd228da5936@mail.gmail.com>

Hi Tony, do you mean in the internal.xml under
/usr/local/freeswitch/local/sip_profiles
we should set <param name="manage-shared-
appearance" value="sylantro"/>

to get the SCA mode working properly?
Please confirm
Thanks,

Chris


On Wed, Feb 17, 2010 at 8:57 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> You have to set the param to sylantro to get that mode.
> Or configure your phone to use broadsoft.
>
> On Feb 17, 2010 7:54 AM, "Yehavi Bourvine" <yehavi.bourvine at gmail.com>
> wrote:
>
> I am trying to migrate from 1.0.5pre10 to the latest (FreeSWITCH Version
> 1.0.5-20100216-0400 (16659M)), and shared apearance stopped working...
>
> We have Polycom phones and I use the same configuration for both versions.
> With the old version I see that after the phone registers with FreeSwitch,
> the server subscribes to the phone for the extension; with the latest
> version this does not happen. Furthermore, the table
> sip_shared_appearance_dialogs is empty.
>
> I don't find anything I can change on the phone config (the line is already
> set to shared).
>
> Here is the relevant config from the sip profile:
>
>     <param name="manage-presence" value="passive"/> (set to TRUE only on
> one profile).
>     <param name="manage-shared-appearance" value="true"/>
>
>
>     <param name="dbname" value="share_presence"/>
>     <param name="presence-hosts" value="pbx-gr-backup.ip-phones.huji.ac.il
> "/>
> Any idea where to look?
>                            Thanks, __Yehavi:
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100217/a6bcf3b9/attachment.html 

From anthony.minessale at gmail.com  Wed Feb 17 06:51:57 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 17 Feb 2010 08:51:57 -0600
Subject: [Freeswitch-users] Announcement: FreeSWITCH Add Broadsoft SCA
	Support
In-Reply-To: <191c3a031002170649i2e7c0cb9t7b806367b781b06a@mail.gmail.com>
References: <87f2f3b91001111742k691966cfs82515db77c13d4d4@mail.gmail.com>
	<d361ca7b1002090838j6c42dcd2of8affb5f281900fd@mail.gmail.com>
	<767B677E-0D19-43A9-89DE-D96EC4A715EC@freeswitch.org>
	<8D062B6E-4691-4BA6-B038-0EC6E8642DDA@jerris.com>
	<9D328485-D4C3-4B14-808A-CB63BBF06DC1@freeswitch.org>
	<d361ca7b1002170549g608d6e16vb2eb3f087fff9752@mail.gmail.com>
	<191c3a031002170556p5dfd4c55le3c279f194808d92@mail.gmail.com>
	<191c3a031002170557g5d8c9df9xf40d1b74b9a6f19d@mail.gmail.com>
	<507898381002170640u5586698cg310f9cd228da5936@mail.gmail.com>
	<191c3a031002170649i2e7c0cb9t7b806367b781b06a@mail.gmail.com>
Message-ID: <191c3a031002170651k3f0c421cj87f3e2bd796b7711@mail.gmail.com>

If by properly you mean the previously supported way then yes.

We added support for the broadsoft method now as the default because its
supported on more phones.

You may want to try setting your phones to that mode to compare but setting
the profile param to sylantro should restore the previous default behaviour.

On Feb 17, 2010 8:46 AM, "Chris Chen" <chris.chen2004 at gmail.com> wrote:

Hi Tony, do you mean in the internal.xml under
/usr/local/freeswitch/local/sip_profiles
we should set <param name="manage-shared-
appearance" value="sylantro"/>

to get the SCA mode working properly?
Please confirm
Thanks,

Chris




On Wed, Feb 17, 2010 at 8:57 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:
>
> You h...

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100217/ff44348f/attachment.html 

From yehavi.bourvine at gmail.com  Wed Feb 17 07:00:17 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Wed, 17 Feb 2010 17:00:17 +0200
Subject: [Freeswitch-users] Announcement: FreeSWITCH Add Broadsoft SCA
	Support
In-Reply-To: <191c3a031002170557g5d8c9df9xf40d1b74b9a6f19d@mail.gmail.com>
References: <87f2f3b91001111742k691966cfs82515db77c13d4d4@mail.gmail.com>
	<d361ca7b1002090817h4aa3c13dka4d4db8b4adc7606@mail.gmail.com>
	<1C0A1BEE-3480-4A33-916F-52067AF5B9C9@freeswitch.org>
	<d361ca7b1002090838j6c42dcd2of8affb5f281900fd@mail.gmail.com>
	<767B677E-0D19-43A9-89DE-D96EC4A715EC@freeswitch.org>
	<8D062B6E-4691-4BA6-B038-0EC6E8642DDA@jerris.com>
	<9D328485-D4C3-4B14-808A-CB63BBF06DC1@freeswitch.org>
	<d361ca7b1002170549g608d6e16vb2eb3f087fff9752@mail.gmail.com>
	<191c3a031002170556p5dfd4c55le3c279f194808d92@mail.gmail.com>
	<191c3a031002170557g5d8c9df9xf40d1b74b9a6f19d@mail.gmail.com>
Message-ID: <d361ca7b1002170700x15630ab2tabc6f3d91ebc3303@mail.gmail.com>

Hello Anthony,

  Since Polycom has no place to define the server type I've set
manage-shared-appearance="sylantro" and have some progress.

Now I see that both the server and the phone subscribe to each other(the
server subscribes twice), but the last NOTIFY from the phone to the server
gets rejected with "Call does not exist".

On the console I get the following error message: [ERR]
sofia_presence.c:2135 Gateway information missing.
                  Thanks, __Yehavi:

 2010/2/17 Anthony Minessale <anthony.minessale at gmail.com>

>  You have to set the param to sylantro to get that mode.
> Or configure your phone to use broadsoft.
>
>   On Feb 17, 2010 7:54 AM, "Yehavi Bourvine" <yehavi.bourvine at gmail.com>
> wrote:
>
>  I am trying to migrate from 1.0.5pre10 to the latest (FreeSWITCH Version
> 1.0.5-20100216-0400 (16659M)), and shared apearance stopped working...
>
> We have Polycom phones and I use the same configuration for both versions.
> With the old version I see that after the phone registers with FreeSwitch,
> the server subscribes to the phone for the extension; with the latest
> version this does not happen. Furthermore, the table
> sip_shared_appearance_dialogs is empty.
>
> I don't find anything I can change on the phone config (the line is already
> set to shared).
>
> Here is the relevant config from the sip profile:
>
>     <param name="manage-presence" value="passive"/> (set to TRUE only on
> one profile).
>     <param name="manage-shared-appearance" value="true"/>
>
>
>     <param name="dbname" value="share_presence"/>
>     <param name="presence-hosts" value="pbx-gr-backup.ip-phones.huji.ac.il
> "/>
> Any idea where to look?
>                            Thanks, __Yehavi:
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100217/317d66f4/attachment-0001.html 

From moizchinoy at gmail.com  Wed Feb 17 07:41:51 2010
From: moizchinoy at gmail.com (Moiz Chinoy)
Date: Wed, 17 Feb 2010 19:41:51 +0400
Subject: [Freeswitch-users] Dingaling - external rtp supported?
In-Reply-To: <191c3a031002170538r3b029515i8a000812a70ad558@mail.gmail.com>
References: <29b888f81002160835y6e290319ub0d04c088caa1b79@mail.gmail.com>
	<DEA39AD1-0EE9-473A-B78C-CDDF3F100578@freeswitch.org>
	<29b888f81002170318p63622f3fhc8d26ad0e85602d2@mail.gmail.com>
	<191c3a031002170538r3b029515i8a000812a70ad558@mail.gmail.com>
Message-ID: <29b888f81002170741y3d13cb57ycb4436a372c4eded@mail.gmail.com>

Guys I am unable to produce the crash but now both parties cannot hear
each other!

Vars.xml has following lines:
  <X-PRE-PROCESS cmd="set" data="external_rtp_ip=stun:stun.freeswitch.org"/>
  <X-PRE-PROCESS cmd="set" data="external_sip_ip=stun:stun.freeswitch.org"/>

Jingle Client.xml has following lines:
    <param name="rtp-ip" value="$${bind_server_ip}"/>
    <!-- <param name="ext-rtp-ip" value="auto-nat"/> -->
    <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
    <param name="disable-rtp-auto-adjust" value="true"/>
    <param name="local-network-acl" value="localnet.auto"/>

On Wed, Feb 17, 2010 at 5:38 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> Obtain a stack trace from the crash.
>
> On Feb 17, 2010 5:26 AM, "Moiz Chinoy" <moizchinoy at gmail.com> wrote:
>
> Hi,
>
> FS rev: 16673
> Platform: Windows
>
> More details:
>
> FS is behind NAT and machine is running a VPN connection.
>
> FS and GTalk client on the same machine:
> --------------------------------------------------------------------------------------------------
> jingle profile client.xml has following line:
> <param name="rtp-ip" value="$${bind_server_ip}"/>
>
> External SIP call is successfully bridged to GTalk client.
>
>
> FS and GTalk client on the different machine:
> --------------------------------------------------------------------------------------------------
> jingle profile client.xml has following lines:
> <param name="rtp-ip" value="$${bind_server_ip}"/>
>
> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
>
> As soon as external SIP call land and I try to bridge the call to
> GTalk client, FS crashes.
>
>
> NAT Details:
> ---------------------------
> I think my NAT does not support UpNP or PMP. The reason I say it
> because when FS starts following message is displayed:
>
> 2010-02-17 09:15:42.809674 [INFO] switch_nat.c:409 Scanning for NAT
> 2010-02-17 09:15:42.809674 [ERR] switch_nat.c:197 Error checking for
> PMP [init failed]
> 2010-02-17 09:15:42.809674 [DEBUG] switch_nat.c:414 Checking for UPnP
> 2010-02-17 09:15:56.825388 [DEBUG] switch_nat.c:114 No
> InternetGatewayDevice, using first entry as default
> (http://192.168.16.17:50144/).
> 2010-02-17 09:15:56.887889 [INFO] switch_nat.c:429 No PMP or UPnP NAT
> devices detected!
>
>
>
> On Tue, Feb 16, 2010 at 8:41 PM, Brian West <brian at freeswitch.org> wrote:
>> can you please update...
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Regards,
Moiz Chinoy.


From msc at freeswitch.org  Wed Feb 17 08:19:22 2010
From: msc at freeswitch.org (Michael S Collins)
Date: Wed, 17 Feb 2010 08:19:22 -0800
Subject: [Freeswitch-users] Dingaling - external rtp supported?
In-Reply-To: <29b888f81002170741y3d13cb57ycb4436a372c4eded@mail.gmail.com>
References: <29b888f81002160835y6e290319ub0d04c088caa1b79@mail.gmail.com>
	<DEA39AD1-0EE9-473A-B78C-CDDF3F100578@freeswitch.org>
	<29b888f81002170318p63622f3fhc8d26ad0e85602d2@mail.gmail.com>
	<191c3a031002170538r3b029515i8a000812a70ad558@mail.gmail.com>
	<29b888f81002170741y3d13cb57ycb4436a372c4eded@mail.gmail.com>
Message-ID: <055E2932-0D54-44E8-85F7-503D2B6CD592@freeswitch.org>

Get a console log and SIP trace of the call. The wiki page Brian sent  
has all the details on how to collect this information.
-MC

Sent from my iPhone

On Feb 17, 2010, at 7:41 AM, Moiz Chinoy <moizchinoy at gmail.com> wrote:

> Guys I am unable to produce the crash but now both parties cannot hear
> each other!
>
> Vars.xml has following lines:
>  <X-PRE-PROCESS cmd="set"  
> data="external_rtp_ip=stun:stun.freeswitch.org"/>
>  <X-PRE-PROCESS cmd="set"  
> data="external_sip_ip=stun:stun.freeswitch.org"/>
>
> Jingle Client.xml has following lines:
>    <param name="rtp-ip" value="$${bind_server_ip}"/>
>    <!-- <param name="ext-rtp-ip" value="auto-nat"/> -->
>    <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
>    <param name="disable-rtp-auto-adjust" value="true"/>
>    <param name="local-network-acl" value="localnet.auto"/>
>
> On Wed, Feb 17, 2010 at 5:38 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>> Obtain a stack trace from the crash.
>>
>> On Feb 17, 2010 5:26 AM, "Moiz Chinoy" <moizchinoy at gmail.com> wrote:
>>
>> Hi,
>>
>> FS rev: 16673
>> Platform: Windows
>>
>> More details:
>>
>> FS is behind NAT and machine is running a VPN connection.
>>
>> FS and GTalk client on the same machine:
>> --- 
>> --- 
>> --- 
>> --- 
>> --- 
>> --- 
>> --- 
>> --- 
>> --- 
>> --- 
>> --------------------------------------------------------------------
>> jingle profile client.xml has following line:
>> <param name="rtp-ip" value="$${bind_server_ip}"/>
>>
>> External SIP call is successfully bridged to GTalk client.
>>
>>
>> FS and GTalk client on the different machine:
>> --- 
>> --- 
>> --- 
>> --- 
>> --- 
>> --- 
>> --- 
>> --- 
>> --- 
>> --- 
>> --------------------------------------------------------------------
>> jingle profile client.xml has following lines:
>> <param name="rtp-ip" value="$${bind_server_ip}"/>
>>
>> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
>>
>> As soon as external SIP call land and I try to bridge the call to
>> GTalk client, FS crashes.
>>
>>
>> NAT Details:
>> ---------------------------
>> I think my NAT does not support UpNP or PMP. The reason I say it
>> because when FS starts following message is displayed:
>>
>> 2010-02-17 09:15:42.809674 [INFO] switch_nat.c:409 Scanning for NAT
>> 2010-02-17 09:15:42.809674 [ERR] switch_nat.c:197 Error checking for
>> PMP [init failed]
>> 2010-02-17 09:15:42.809674 [DEBUG] switch_nat.c:414 Checking for UPnP
>> 2010-02-17 09:15:56.825388 [DEBUG] switch_nat.c:114 No
>> InternetGatewayDevice, using first entry as default
>> (http://192.168.16.17:50144/).
>> 2010-02-17 09:15:56.887889 [INFO] switch_nat.c:429 No PMP or UPnP NAT
>> devices detected!
>>
>>
>>
>> On Tue, Feb 16, 2010 at 8:41 PM, Brian West <brian at freeswitch.org>  
>> wrote:
>>> can you please update...
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>> users
>> http://www.freeswitch.org
>>
>>
>
>
>
> -- 
> Regards,
> Moiz Chinoy.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org


From brian at freeswitch.org  Wed Feb 17 08:24:47 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 17 Feb 2010 10:24:47 -0600
Subject: [Freeswitch-users] Dingaling - external rtp supported?
In-Reply-To: <055E2932-0D54-44E8-85F7-503D2B6CD592@freeswitch.org>
References: <29b888f81002160835y6e290319ub0d04c088caa1b79@mail.gmail.com>
	<DEA39AD1-0EE9-473A-B78C-CDDF3F100578@freeswitch.org>
	<29b888f81002170318p63622f3fhc8d26ad0e85602d2@mail.gmail.com>
	<191c3a031002170538r3b029515i8a000812a70ad558@mail.gmail.com>
	<29b888f81002170741y3d13cb57ycb4436a372c4eded@mail.gmail.com>
	<055E2932-0D54-44E8-85F7-503D2B6CD592@freeswitch.org>
Message-ID: <D4D98039-293E-4DF4-9DB2-E695EA84F3C9@freeswitch.org>

I have tested this with empty values... and it works fine.

/b

On Feb 17, 2010, at 10:19 AM, Michael S Collins wrote:

> Get a console log and SIP trace of the call. The wiki page Brian sent  
> has all the details on how to collect this information.
> -MC
> 
> Sent from my iPhone



From e.brolman at telecats.nl  Wed Feb 17 05:56:32 2010
From: e.brolman at telecats.nl (=?iso-8859-1?Q?Eelco_Br=F6lman?=)
Date: Wed, 17 Feb 2010 14:56:32 +0100
Subject: [Freeswitch-users] Calls being parked on DTMF (2)
Message-ID: <0A1FDB5DAA23564F8758BA05D26DCD747F9A09@exchange.telecats.nl>

Hi all,

I have the same problems as described in the thread "Calls being parked on DTMF".

We have a lab-test setup, which consist of the following:

1) a box with 4 E1 trunks with a call generator application (let's call it CG) 
2) connected with a FreeSwitch box (Sangoma A104D card) (let's call that FS)

The CG dials a number on the FS box (number 921000), which is bridged out back to the CG server (number 1000). The initiating application sends some DTMF, which is replied (echo-ed) by the receiving application (both on the CG server).

In roughly about 80% of the calls, FS just disconnects the call:

2010-02-17 14:30:14.677125 [DEBUG] switch_ivr_originate.c:3105 Originate Resulted in Success: [OpenZAP/1:1/1000]
2010-02-17 14:30:14.677125 [DEBUG] switch_core_session.c:638 Send signal OpenZAP/1:1/1000 [BREAK]
2010-02-17 14:30:14.677125 [DEBUG] switch_core_session.c:638 Send signal OpenZAP/1:60/921000 [BREAK]
2010-02-17 14:30:14.677125 [DEBUG] switch_ivr_bridge.c:1175 (OpenZAP/1:1/1000) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA
2010-02-17 14:30:14.677125 [DEBUG] switch_core_session.c:1018 Send signal OpenZAP/1:1/1000 [BREAK]
2010-02-17 14:30:14.677125 [DEBUG] switch_core_state_machine.c:314 (OpenZAP/1:1/1000) Running State Change CS_EXCHANGE_MEDIA
2010-02-17 14:30:14.677125 [DEBUG] switch_core_state_machine.c:351 (OpenZAP/1:1/1000) State EXCHANGE_MEDIA
2010-02-17 14:30:14.677125 [DEBUG] mod_openzap.c:558 CHANNEL EXCHANGE_MEDIA
2010-02-17 14:30:14.687135 [DEBUG] switch_core_session.c:699 Send signal OpenZAP/1:1/1000 [BREAK]
2010-02-17 14:30:14.687135 [DEBUG] switch_core_session.c:699 Send signal OpenZAP/1:60/921000 [BREAK]
2010-02-17 14:30:15.986882 [DEBUG] mod_openzap.c:684 queue DTMF [7]
2010-02-17 14:30:16.026733 [DEBUG] zap_io.c:1870 1:1 GENERATE DTMF [7]
2010-02-17 14:30:16.106418 [DEBUG] switch_ivr_bridge.c:469 OpenZAP/1:1/1000 ending bridge by request from write function
2010-02-17 14:30:16.106418 [DEBUG] switch_core_session.c:638 Send signal OpenZAP/1:60/921000 [BREAK]
2010-02-17 14:30:16.126335 [DEBUG] switch_ivr_bridge.c:475 OpenZAP/1:1/1000 ending bridge by request from read function
2010-02-17 14:30:16.126335 [DEBUG] switch_core_session.c:638 Send signal OpenZAP/1:1/1000 [BREAK]
2010-02-17 14:30:16.126335 [DEBUG] switch_ivr_bridge.c:562 BRIDGE THREAD DONE [OpenZAP/1:60/921000]
2010-02-17 14:30:16.126335 [DEBUG] switch_ivr_bridge.c:582 Send signal OpenZAP/1:1/1000 [BREAK]
2010-02-17 14:30:16.126335 [DEBUG] switch_ivr_bridge.c:562 BRIDGE THREAD DONE [OpenZAP/1:1/1000]
2010-02-17 14:30:16.126335 [DEBUG] switch_ivr_bridge.c:582 Send signal OpenZAP/1:60/921000 [BREAK]
2010-02-17 14:30:16.126335 [NOTICE] switch_ivr_bridge.c:634 Hangup OpenZAP/1:1/1000 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
2010-02-17 14:30:16.126335 [DEBUG] switch_channel.c:2063 Send signal OpenZAP/1:1/1000 [KILL]


Where the line "ending bridge by request from write function" is curious of course.

I'm running:
- Freeswitch trunk (r16674) (updated, make clean, reconfigured and make install this morning)
- Wanpipe 3.5.10.smg-2 (see http://wiki.sangoma.com/wanpipe-SmgPriInstallation)
- Using the ozmod_sangoma_boost SCTP


If anyone needs more information, tests or config files, I'll be happy to provide them.



Kind regards, 

Eelco Br?lman



From mike at jerris.com  Wed Feb 17 08:52:20 2010
From: mike at jerris.com (Michael Jerris)
Date: Wed, 17 Feb 2010 11:52:20 -0500
Subject: [Freeswitch-users] FreeSWITCH.Managed.dll deletes on make
	distclean
In-Reply-To: <201002152353.43666.errotan@gmail.com>
References: <201002152353.43666.errotan@gmail.com>
Message-ID: <097535D1-2601-4BBE-88D8-A113685A9435@jerris.com>

we do not currently have a working distclean target.  using it will likely make your tree in an unusable state.

Mike

On Feb 15, 2010, at 5:53 PM, Pusk?s Zsolt wrote:

> I usually do svn-clean than svn up when i compile a new version of fs.
> I noticed that if i do make distclean the file @ 
> src/mod/languages/mod_managed/managed/FreeSWITCH.Managed.dll got deleted.
> When i do svn up it gets 'restored':
> 
> Restored 'src/mod/languages/mod_managed/managed/FreeSWITCH.Managed.dll'



From brian at freeswitch.org  Wed Feb 17 09:04:38 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 17 Feb 2010 11:04:38 -0600
Subject: [Freeswitch-users] One more thing..
In-Reply-To: <191c3a031002170536g297a816ch2fa94eb2dcfbd16d@mail.gmail.com>
References: <4B7B5EE9.4020705@meanie.us>
	<191c3a031002170535i12c82731r5829417387dd3228@mail.gmail.com>
	<191c3a031002170536g297a816ch2fa94eb2dcfbd16d@mail.gmail.com>
Message-ID: <2CEBE2C8-3982-49B6-A06F-9406C1D1F123@freeswitch.org>

Its the same bug the linksys SPA has... where the RTP time is set to 0.030 and an inbound call to the device doesn't 200ok with the right ptime.

/b

On Feb 17, 2010, at 7:36 AM, Anthony Minessale wrote:

> Are you doing this with the latest revision?
> You would have to supply more info like a console trace on debug level with siptrace enabled.



From ledoktre at meanie.us  Wed Feb 17 09:19:06 2010
From: ledoktre at meanie.us (Doc)
Date: Wed, 17 Feb 2010 11:19:06 -0600
Subject: [Freeswitch-users] One more thing..
In-Reply-To: <191c3a031002170536g297a816ch2fa94eb2dcfbd16d@mail.gmail.com>
References: <4B7B5EE9.4020705@meanie.us>	<191c3a031002170535i12c82731r5829417387dd3228@mail.gmail.com>
	<191c3a031002170536g297a816ch2fa94eb2dcfbd16d@mail.gmail.com>
Message-ID: <4B7C250A.3050801@meanie.us>

I need to be a little quicker on my replies :-)  FS tells me "FreeSWITCH 
Version 1.0.trunk (16619M)".  I installed it this past Saturday from SVN 
trunk as I recall.  I can supply anything you need, but I am pretty new 
to FS and definately wet behind the ears.  Thanks-

Anthony Minessale wrote:
>
> Are you doing this with the latest revision?
> You would have to supply more info like a console trace on debug level 
> with siptrace enabled.
>
>> On Feb 17, 2010 1:51 AM, "Doc" <ledoktre at meanie.us 
>> <mailto:ledoktre at meanie.us>> wrote:
>>
>> Greetings one more time,
>>
>> I just remembered.  I also ran into one other issue in my testing.
>>
>> When I route the incoming skype call to an extension (Sipura SPA 1001),
>> it plays hold music on the callers side (good....), and when I pick up
>> the extension, calling party drops, internal phone is left with a fast
>> busy signal :
>>
>> 2010-02-16 21:09:31.687204 [NOTICE] sofia.c:4690 Channel
>> [sofia/internal/sip:1001 at 10.24.72.12:5060 
>> <http://sip:1001 at 10.24.72.12:5060>] has been answered
>> 2010-02-16 21:09:31.987714 [WARNING] mod_sofia.c:918 We were told to use
>> ptime 3 but what they meant to say was 20
>> This issue has so far been identified to happen on the following broken
>> platforms/devices:
>> Linksys/Sipura aka Cisco
>> ShoreTel
>> Sonus/L3
>> We will try to fix it but some of the devices on this list are so broken
>> who knows what will happen..
>> 2010-02-16 21:09:32.027243 [ERR] switch_core_io.c:121
>> sofia/internal/sip:1001 at 10.24.72.12:5060 
>> <http://sip:1001 at 10.24.72.12:5060> has no read codec.
>>
>> First thing I notice is this unusual error about the ptime?  and the one
>> that hangs me up I think is the "sofia... has no read codec".  I have
>> been poking around my configuration - any suggestions?
>>
>> Thanks,
>>
>> Doc
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org 
>> <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> ------------------------------------------------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   


From ledoktre at meanie.us  Wed Feb 17 09:19:59 2010
From: ledoktre at meanie.us (Doc)
Date: Wed, 17 Feb 2010 11:19:59 -0600
Subject: [Freeswitch-users] One more thing..
In-Reply-To: <2CEBE2C8-3982-49B6-A06F-9406C1D1F123@freeswitch.org>
References: <4B7B5EE9.4020705@meanie.us>	<191c3a031002170535i12c82731r5829417387dd3228@mail.gmail.com>	<191c3a031002170536g297a816ch2fa94eb2dcfbd16d@mail.gmail.com>
	<2CEBE2C8-3982-49B6-A06F-9406C1D1F123@freeswitch.org>
Message-ID: <4B7C253F.9090905@meanie.us>

Is there any configuration to change in the Sipura, any changes to FS 
configuration, or does it boil down to using different SIP hardware? -Thanks

Brian West wrote:
> Its the same bug the linksys SPA has... where the RTP time is set to 0.030 and an inbound call to the device doesn't 200ok with the right ptime.
>
> /b
>
> On Feb 17, 2010, at 7:36 AM, Anthony Minessale wrote:
>
>   
>> Are you doing this with the latest revision?
>> You would have to supply more info like a console trace on debug level with siptrace enabled.
>>     
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   


From ledoktre at meanie.us  Wed Feb 17 09:23:40 2010
From: ledoktre at meanie.us (Doc)
Date: Wed, 17 Feb 2010 11:23:40 -0600
Subject: [Freeswitch-users] Greetings and a couple of questions
In-Reply-To: <1266408503.10430.1360421639@webmail.messagingengine.com>
References: <4B7B5A46.3090304@meanie.us>
	<1266408503.10430.1360421639@webmail.messagingengine.com>
Message-ID: <4B7C261C.1040802@meanie.us>

Once you mentioned the in-band versus out-of-band DTMF, it made sense.  
I've been reading on it since your post, and I'm going to try a few 
things.  Wonder if there is any app for voice controlled IVR (so many 
systems seem to support it these days, it would be a nice way to 
circumvent the DTMF issue... :-) ).

Any ideas on the rest of my points?   -Thanks,

Scott Torr wrote:
> On Tue, 16 Feb 2010 20:53 -0600, "Doc" <ledoktre at meanie.us> wrote:
>   
>> 1) I am able to see a call come in, and it gets routed to the sample IVR 
>> to start.  The first thing off is that when I dial from PSTN -> 
>> Skype-In, it does not let me push any buttons.  If I launch a second 
>> skype client, and dial the skype user on FS directly, it works fine.  
>> Any ideas?
>>     
>
> Hi Doc,
>
> When you dial in from the PSTN the 'push button' events are present as
> "in band" audio tones.
> By default the sample IVR only works on "out of band" DTMF events.
>
> This is why when you call directly from another skype client the 'push
> button' events are detected because they are passed as "out of band"
> signaling.
>
>
> Now,
> In the dial plan you can tell FS to listen for "in band" audio DTMF
> tones using <action application="start_dtmf" />
>
> However,
> This currently does not work during a skype call for some reason?
> http://jira.freeswitch.org/browse/MODSKYPIAX-66
>
>
> A work around,
> is to sign up for the "Skype SIP Beta" product where the 'push button'
> events are sent to FS "out of band".
>
> This conversion is done at the PSTN --> Skype gateway by dedicated DTMF
> tone detection hardware.
>
>
> Skype has either made a business decision, or a technical over sight to
> pass DTMF events 'out of band' for only addition fee products.
>
>
> It has also been reported in New Zealand that even the 'in band' tones
> where present one day and actually filtered out the next.
> This seems extreme, but either through deliberate action or a technology
> change this is what was reported on one blog.
>
> It is unclear to me if this was a technical limitation or a blunt
> business decisions, but a audio sample showed the audio tones missing? 
>
>
> In any case you would not want to rely on 'In band' DTMF' tones when
> passed through 'lossy' codecs anyway.
>
> Best to stick with 'out of band' signaling for reliability.
>
>
> regards,
> Scott Torr
>
>
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   


From brian at freeswitch.org  Wed Feb 17 09:26:44 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 17 Feb 2010 11:26:44 -0600
Subject: [Freeswitch-users] One more thing..
In-Reply-To: <4B7C253F.9090905@meanie.us>
References: <4B7B5EE9.4020705@meanie.us>	<191c3a031002170535i12c82731r5829417387dd3228@mail.gmail.com>	<191c3a031002170536g297a816ch2fa94eb2dcfbd16d@mail.gmail.com>
	<2CEBE2C8-3982-49B6-A06F-9406C1D1F123@freeswitch.org>
	<4B7C253F.9090905@meanie.us>
Message-ID: <5A00F97E-41C9-43AD-8CFA-1C8B87E5EECC@freeswitch.org>

change the rtp time to 0.020 from 0.030

/b

On Feb 17, 2010, at 11:19 AM, Doc wrote:

> Is there any configuration to change in the Sipura, any changes to FS 
> configuration, or does it boil down to using different SIP hardware? -Thanks



From ledoktre at meanie.us  Wed Feb 17 09:56:19 2010
From: ledoktre at meanie.us (Doc)
Date: Wed, 17 Feb 2010 11:56:19 -0600
Subject: [Freeswitch-users] One more thing..
In-Reply-To: <5A00F97E-41C9-43AD-8CFA-1C8B87E5EECC@freeswitch.org>
References: <4B7B5EE9.4020705@meanie.us>	<191c3a031002170535i12c82731r5829417387dd3228@mail.gmail.com>	<191c3a031002170536g297a816ch2fa94eb2dcfbd16d@mail.gmail.com>	<2CEBE2C8-3982-49B6-A06F-9406C1D1F123@freeswitch.org>	<4B7C253F.9090905@meanie.us>
	<5A00F97E-41C9-43AD-8CFA-1C8B87E5EECC@freeswitch.org>
Message-ID: <4B7C2DC3.20502@meanie.us>

I did try changing the rtp time in my Sipura (advanced, sip, RTP Packet 
Size) from 0.030 to 0.020, and the error went away.  I was able to then 
test (and succeed) dialing from secondary skype user to skypiax, and 
have the call bridged automatically to one of my Sipura SPA-1001's.  I 
could speak into the phone, and hear it come through skype (on my 
laptop) no problem.  When I would talk on the laptop, I would get garble 
back on the phone, but that could be a sound issue on my laptop (my 
skype sometimes does this with Pulse audio on my laptop, Ubuntu Karmic).

The interesting thing it did do, however, was eventually (within a 
minute?) It threw a couple of errors :

2010-02-17 11:47:34.007909 [ERR] mod_skypiax.c:826 rev 
16619M[(nil)|37     ][ERRORA  826  ][interface1][-1, 5,21] EXIT? sent=-1
2010-02-17 11:47:34.027776 [ERR] mod_skypiax.c:826 rev 
16619M[(nil)|37     ][ERRORA  826  ][interface1][-1, 5,21] EXIT? sent=-1
2010-02-17 11:47:34.047643 [ERR] mod_skypiax.c:826 rev 
16619M[(nil)|37     ][ERRORA  826  ][interface1][-1, 5,21] EXIT? sent=-1

And on a subsequent test, I received no audio (the above error rolling 
on the console), and within a matter of seconds, FS crashed with this :

Segmentation fault (core dumped)

I have another request open on the group actually for the above error, 
so I think it'd make sense to discuss further there (rather than 
duplicating material).

Thanks!  At any rate, the error for the missing codec, etc, seemed to be 
gone once I updated the RTP time in my ATA.  Hope this helps someone!!

Brian West wrote:
> change the rtp time to 0.020 from 0.030
>
> /b


From brian at freeswitch.org  Wed Feb 17 11:23:03 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 17 Feb 2010 13:23:03 -0600
Subject: [Freeswitch-users] Announcement: FreeSWITCH Add Broadsoft SCA
	Support
In-Reply-To: <d361ca7b1002170700x15630ab2tabc6f3d91ebc3303@mail.gmail.com>
References: <87f2f3b91001111742k691966cfs82515db77c13d4d4@mail.gmail.com>
	<d361ca7b1002090817h4aa3c13dka4d4db8b4adc7606@mail.gmail.com>
	<1C0A1BEE-3480-4A33-916F-52067AF5B9C9@freeswitch.org>
	<d361ca7b1002090838j6c42dcd2of8affb5f281900fd@mail.gmail.com>
	<767B677E-0D19-43A9-89DE-D96EC4A715EC@freeswitch.org>
	<8D062B6E-4691-4BA6-B038-0EC6E8642DDA@jerris.com>
	<9D328485-D4C3-4B14-808A-CB63BBF06DC1@freeswitch.org>
	<d361ca7b1002170549g608d6e16vb2eb3f087fff9752@mail.gmail.com>
	<191c3a031002170556p5dfd4c55le3c279f194808d92@mail.gmail.com>
	<191c3a031002170557g5d8c9df9xf40d1b74b9a6f19d@mail.gmail.com>
	<d361ca7b1002170700x15630ab2tabc6f3d91ebc3303@mail.gmail.com>
Message-ID: <ADC8B813-4161-481C-877E-93FC012858B4@freeswitch.org>

Step 1. Enable manage-shared-appearance=true

Step 2. Now in the phone's config Configure the phone as usually, set the line shared and DO NOT set the third party name.

Step 3. Reboot

It should work.

I wish someone that has this working would write some wiki docs these threads about it not working are getting rather old when I know for a fact they work fine.

The gateway info missing is a gateway you have configured getting a notify.  It has nothing to do with SCA.

/b

On Feb 17, 2010, at 9:00 AM, Yehavi Bourvine wrote:

> .



From anthony.minessale at gmail.com  Wed Feb 17 11:23:06 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 17 Feb 2010 13:23:06 -0600
Subject: [Freeswitch-users] One more thing..
In-Reply-To: <4B7C2DC3.20502@meanie.us>
References: <4B7B5EE9.4020705@meanie.us>
	<191c3a031002170535i12c82731r5829417387dd3228@mail.gmail.com>
	<191c3a031002170536g297a816ch2fa94eb2dcfbd16d@mail.gmail.com>
	<2CEBE2C8-3982-49B6-A06F-9406C1D1F123@freeswitch.org>
	<4B7C253F.9090905@meanie.us>
	<5A00F97E-41C9-43AD-8CFA-1C8B87E5EECC@freeswitch.org>
	<4B7C2DC3.20502@meanie.us>
Message-ID: <191c3a031002171123r416f6413wbecddcea498b5616@mail.gmail.com>

keep updating, the maintainer of mod_skypeiax is adding new patches every
few hours.
You can also join irc on irc.freenode.net #freeswitch and #freeswitch-dev to
interact with him live.


One hint around here when we ask if you updated we work in 1 minute
increments.



On Wed, Feb 17, 2010 at 11:56 AM, Doc <ledoktre at meanie.us> wrote:

> I did try changing the rtp time in my Sipura (advanced, sip, RTP Packet
> Size) from 0.030 to 0.020, and the error went away.  I was able to then
> test (and succeed) dialing from secondary skype user to skypiax, and
> have the call bridged automatically to one of my Sipura SPA-1001's.  I
> could speak into the phone, and hear it come through skype (on my
> laptop) no problem.  When I would talk on the laptop, I would get garble
> back on the phone, but that could be a sound issue on my laptop (my
> skype sometimes does this with Pulse audio on my laptop, Ubuntu Karmic).
>
> The interesting thing it did do, however, was eventually (within a
> minute?) It threw a couple of errors :
>
> 2010-02-17 11:47:34.007909 [ERR] mod_skypiax.c:826 rev
> 16619M[(nil)|37     ][ERRORA  826  ][interface1][-1, 5,21] EXIT? sent=-1
> 2010-02-17 11:47:34.027776 [ERR] mod_skypiax.c:826 rev
> 16619M[(nil)|37     ][ERRORA  826  ][interface1][-1, 5,21] EXIT? sent=-1
> 2010-02-17 11:47:34.047643 [ERR] mod_skypiax.c:826 rev
> 16619M[(nil)|37     ][ERRORA  826  ][interface1][-1, 5,21] EXIT? sent=-1
>
> And on a subsequent test, I received no audio (the above error rolling
> on the console), and within a matter of seconds, FS crashed with this :
>
> Segmentation fault (core dumped)
>
> I have another request open on the group actually for the above error,
> so I think it'd make sense to discuss further there (rather than
> duplicating material).
>
> Thanks!  At any rate, the error for the missing codec, etc, seemed to be
> gone once I updated the RTP time in my ATA.  Hope this helps someone!!
>
> Brian West wrote:
> > change the rtp time to 0.020 from 0.030
> >
> > /b
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100217/9657cfda/attachment.html 

From msc at freeswitch.org  Wed Feb 17 11:29:40 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 17 Feb 2010 11:29:40 -0800
Subject: [Freeswitch-users] One more thing..
In-Reply-To: <4B7C2DC3.20502@meanie.us>
References: <4B7B5EE9.4020705@meanie.us>
	<191c3a031002170535i12c82731r5829417387dd3228@mail.gmail.com>
	<191c3a031002170536g297a816ch2fa94eb2dcfbd16d@mail.gmail.com>
	<2CEBE2C8-3982-49B6-A06F-9406C1D1F123@freeswitch.org>
	<4B7C253F.9090905@meanie.us>
	<5A00F97E-41C9-43AD-8CFA-1C8B87E5EECC@freeswitch.org>
	<4B7C2DC3.20502@meanie.us>
Message-ID: <87f2f3b91002171129v59518b2bg9673212cff957a6c@mail.gmail.com>

Are you able to update your FS to the latest trunk? ("make current" in the
FS src dir) Also, I notice that you are on 16619M - the "M" indicates a
modification of some kind. Did you make any changes to the FS source?

-MC

On Wed, Feb 17, 2010 at 9:56 AM, Doc <ledoktre at meanie.us> wrote:

> I did try changing the rtp time in my Sipura (advanced, sip, RTP Packet
> Size) from 0.030 to 0.020, and the error went away.  I was able to then
> test (and succeed) dialing from secondary skype user to skypiax, and
> have the call bridged automatically to one of my Sipura SPA-1001's.  I
> could speak into the phone, and hear it come through skype (on my
> laptop) no problem.  When I would talk on the laptop, I would get garble
> back on the phone, but that could be a sound issue on my laptop (my
> skype sometimes does this with Pulse audio on my laptop, Ubuntu Karmic).
>
> The interesting thing it did do, however, was eventually (within a
> minute?) It threw a couple of errors :
>
> 2010-02-17 11:47:34.007909 [ERR] mod_skypiax.c:826 rev
> 16619M[(nil)|37     ][ERRORA  826  ][interface1][-1, 5,21] EXIT? sent=-1
> 2010-02-17 11:47:34.027776 [ERR] mod_skypiax.c:826 rev
> 16619M[(nil)|37     ][ERRORA  826  ][interface1][-1, 5,21] EXIT? sent=-1
> 2010-02-17 11:47:34.047643 [ERR] mod_skypiax.c:826 rev
> 16619M[(nil)|37     ][ERRORA  826  ][interface1][-1, 5,21] EXIT? sent=-1
>
> And on a subsequent test, I received no audio (the above error rolling
> on the console), and within a matter of seconds, FS crashed with this :
>
> Segmentation fault (core dumped)
>
> I have another request open on the group actually for the above error,
> so I think it'd make sense to discuss further there (rather than
> duplicating material).
>
> Thanks!  At any rate, the error for the missing codec, etc, seemed to be
> gone once I updated the RTP time in my ATA.  Hope this helps someone!!
>
> Brian West wrote:
> > change the rtp time to 0.020 from 0.030
> >
> > /b
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100217/842ca09c/attachment-0001.html 

From ledoktre at meanie.us  Wed Feb 17 11:35:59 2010
From: ledoktre at meanie.us (Doc)
Date: Wed, 17 Feb 2010 13:35:59 -0600
Subject: [Freeswitch-users] One more thing..
In-Reply-To: <191c3a031002171123r416f6413wbecddcea498b5616@mail.gmail.com>
References: <4B7B5EE9.4020705@meanie.us>	<191c3a031002170535i12c82731r5829417387dd3228@mail.gmail.com>	<191c3a031002170536g297a816ch2fa94eb2dcfbd16d@mail.gmail.com>	<2CEBE2C8-3982-49B6-A06F-9406C1D1F123@freeswitch.org>	<4B7C253F.9090905@meanie.us>	<5A00F97E-41C9-43AD-8CFA-1C8B87E5EECC@freeswitch.org>	<4B7C2DC3.20502@meanie.us>
	<191c3a031002171123r416f6413wbecddcea498b5616@mail.gmail.com>
Message-ID: <4B7C451F.1050104@meanie.us>

I will update the software and report back.  It it good to know that you 
think in "1 minute increments" - I figured mine was not too outdated 
;-)   -Thanks,

Anthony Minessale wrote:
> keep updating, the maintainer of mod_skypeiax is adding new patches 
> every few hours.
> You can also join irc on irc.freenode.net <http://irc.freenode.net> 
> #freeswitch and #freeswitch-dev to interact with him live.
>
>
> One hint around here when we ask if you updated we work in 1 minute 
> increments.
>
>
>
> On Wed, Feb 17, 2010 at 11:56 AM, Doc <ledoktre at meanie.us 
> <mailto:ledoktre at meanie.us>> wrote:
>
>     I did try changing the rtp time in my Sipura (advanced, sip, RTP
>     Packet
>     Size) from 0.030 to 0.020, and the error went away.  I was able to
>     then
>     test (and succeed) dialing from secondary skype user to skypiax, and
>     have the call bridged automatically to one of my Sipura SPA-1001's.  I
>     could speak into the phone, and hear it come through skype (on my
>     laptop) no problem.  When I would talk on the laptop, I would get
>     garble
>     back on the phone, but that could be a sound issue on my laptop (my
>     skype sometimes does this with Pulse audio on my laptop, Ubuntu
>     Karmic).
>
>     The interesting thing it did do, however, was eventually (within a
>     minute?) It threw a couple of errors :
>
>     2010-02-17 11:47:34.007909 [ERR] mod_skypiax.c:826 rev
>     16619M[(nil)|37     ][ERRORA  826  ][interface1][-1, 5,21] EXIT?
>     sent=-1
>     2010-02-17 11:47:34.027776 [ERR] mod_skypiax.c:826 rev
>     16619M[(nil)|37     ][ERRORA  826  ][interface1][-1, 5,21] EXIT?
>     sent=-1
>     2010-02-17 11:47:34.047643 [ERR] mod_skypiax.c:826 rev
>     16619M[(nil)|37     ][ERRORA  826  ][interface1][-1, 5,21] EXIT?
>     sent=-1
>
>     And on a subsequent test, I received no audio (the above error rolling
>     on the console), and within a matter of seconds, FS crashed with
>     this :
>
>     Segmentation fault (core dumped)
>
>     I have another request open on the group actually for the above error,
>     so I think it'd make sense to discuss further there (rather than
>     duplicating material).
>
>     Thanks!  At any rate, the error for the missing codec, etc, seemed
>     to be
>     gone once I updated the RTP time in my ATA.  Hope this helps someone!!
>
>     Brian West wrote:
>     > change the rtp time to 0.020 from 0.030
>     >
>     > /b
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com 
> <mailto:MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com 
> <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org 
> <mailto:sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888 
> <http://iax:guest at conference.freeswitch.org/888>
> googletalk:conf+888 at conference.freeswitch.org 
> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
> ------------------------------------------------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   


From anthony.minessale at gmail.com  Wed Feb 17 11:36:49 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 17 Feb 2010 13:36:49 -0600
Subject: [Freeswitch-users] Dingaling - external rtp supported?
In-Reply-To: <29b888f81002170741y3d13cb57ycb4436a372c4eded@mail.gmail.com>
References: <29b888f81002160835y6e290319ub0d04c088caa1b79@mail.gmail.com>
	<DEA39AD1-0EE9-473A-B78C-CDDF3F100578@freeswitch.org>
	<29b888f81002170318p63622f3fhc8d26ad0e85602d2@mail.gmail.com>
	<191c3a031002170538r3b029515i8a000812a70ad558@mail.gmail.com>
	<29b888f81002170741y3d13cb57ycb4436a372c4eded@mail.gmail.com>
Message-ID: <191c3a031002171136q7424e9bai34379abb542daf93@mail.gmail.com>

you cant combine stun and gtalk and boxes in the same lan very easily if you
do need to do that you will need to mess with

<param name="candidate-acl" value="wan.auto"/>
<param name="local-network-acl" value="localnet.auto"/>





On Wed, Feb 17, 2010 at 9:41 AM, Moiz Chinoy <moizchinoy at gmail.com> wrote:

> Guys I am unable to produce the crash but now both parties cannot hear
> each other!
>
> Vars.xml has following lines:
>  <X-PRE-PROCESS cmd="set" data="external_rtp_ip=stun:stun.freeswitch.org
> "/>
>  <X-PRE-PROCESS cmd="set" data="external_sip_ip=stun:stun.freeswitch.org
> "/>
>
> Jingle Client.xml has following lines:
>     <param name="rtp-ip" value="$${bind_server_ip}"/>
>     <!-- <param name="ext-rtp-ip" value="auto-nat"/> -->
>     <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
>     <param name="disable-rtp-auto-adjust" value="true"/>
>    <param name="local-network-acl" value="localnet.auto"/>
>
> On Wed, Feb 17, 2010 at 5:38 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
> > Obtain a stack trace from the crash.
> >
> > On Feb 17, 2010 5:26 AM, "Moiz Chinoy" <moizchinoy at gmail.com> wrote:
> >
> > Hi,
> >
> > FS rev: 16673
> > Platform: Windows
> >
> > More details:
> >
> > FS is behind NAT and machine is running a VPN connection.
> >
> > FS and GTalk client on the same machine:
> >
> --------------------------------------------------------------------------------------------------
> > jingle profile client.xml has following line:
> > <param name="rtp-ip" value="$${bind_server_ip}"/>
> >
> > External SIP call is successfully bridged to GTalk client.
> >
> >
> > FS and GTalk client on the different machine:
> >
> --------------------------------------------------------------------------------------------------
> > jingle profile client.xml has following lines:
> > <param name="rtp-ip" value="$${bind_server_ip}"/>
> >
> > <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
> >
> > As soon as external SIP call land and I try to bridge the call to
> > GTalk client, FS crashes.
> >
> >
> > NAT Details:
> > ---------------------------
> > I think my NAT does not support UpNP or PMP. The reason I say it
> > because when FS starts following message is displayed:
> >
> > 2010-02-17 09:15:42.809674 [INFO] switch_nat.c:409 Scanning for NAT
> > 2010-02-17 09:15:42.809674 [ERR] switch_nat.c:197 Error checking for
> > PMP [init failed]
> > 2010-02-17 09:15:42.809674 [DEBUG] switch_nat.c:414 Checking for UPnP
> > 2010-02-17 09:15:56.825388 [DEBUG] switch_nat.c:114 No
> > InternetGatewayDevice, using first entry as default
> > (http://192.168.16.17:50144/).
> > 2010-02-17 09:15:56.887889 [INFO] switch_nat.c:429 No PMP or UPnP NAT
> > devices detected!
> >
> >
> >
> > On Tue, Feb 16, 2010 at 8:41 PM, Brian West <brian at freeswitch.org>
> wrote:
> >> can you please update...
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Regards,
> Moiz Chinoy.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100217/bb5f5244/attachment.html 

From yehavi.bourvine at gmail.com  Wed Feb 17 11:37:52 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Wed, 17 Feb 2010 21:37:52 +0200
Subject: [Freeswitch-users] Announcement: FreeSWITCH Add Broadsoft SCA
	Support
In-Reply-To: <ADC8B813-4161-481C-877E-93FC012858B4@freeswitch.org>
References: <87f2f3b91001111742k691966cfs82515db77c13d4d4@mail.gmail.com>
	<d361ca7b1002090838j6c42dcd2of8affb5f281900fd@mail.gmail.com>
	<767B677E-0D19-43A9-89DE-D96EC4A715EC@freeswitch.org>
	<8D062B6E-4691-4BA6-B038-0EC6E8642DDA@jerris.com>
	<9D328485-D4C3-4B14-808A-CB63BBF06DC1@freeswitch.org>
	<d361ca7b1002170549g608d6e16vb2eb3f087fff9752@mail.gmail.com>
	<191c3a031002170556p5dfd4c55le3c279f194808d92@mail.gmail.com>
	<191c3a031002170557g5d8c9df9xf40d1b74b9a6f19d@mail.gmail.com>
	<d361ca7b1002170700x15630ab2tabc6f3d91ebc3303@mail.gmail.com>
	<ADC8B813-4161-481C-877E-93FC012858B4@freeswitch.org>
Message-ID: <d361ca7b1002171137u7ad51bfaia34176a3b8181f0f@mail.gmail.com>

Hello Brian,

  I'll try what you suggest on Friday, and if it works I will document it in
the wiki under the Polycom page I already wrote.

                                 Thanks, __Yehavi:

2010/2/17 Brian West <brian at freeswitch.org>

> Step 1. Enable manage-shared-appearance=true
>
> Step 2. Now in the phone's config Configure the phone as usually, set the
> line shared and DO NOT set the third party name.
>
> Step 3. Reboot
>
> It should work.
>
> I wish someone that has this working would write some wiki docs these
> threads about it not working are getting rather old when I know for a fact
> they work fine.
>
> The gateway info missing is a gateway you have configured getting a notify.
>  It has nothing to do with SCA.
>
> /b
>
> On Feb 17, 2010, at 9:00 AM, Yehavi Bourvine wrote:
>
> > .
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100217/e397f872/attachment.html 

From anthony.minessale at gmail.com  Wed Feb 17 11:41:21 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 17 Feb 2010 13:41:21 -0600
Subject: [Freeswitch-users] Calls being parked on DTMF (2)
In-Reply-To: <0A1FDB5DAA23564F8758BA05D26DCD747F9A09@exchange.telecats.nl>
References: <0A1FDB5DAA23564F8758BA05D26DCD747F9A09@exchange.telecats.nl>
Message-ID: <191c3a031002171141ne49886en5552177104f10bf3@mail.gmail.com>

For the 10 gazillionth time:

Please report issues to jira, as soon as I read an email with a problem in
it, it's marked read and gets lost in a sea of 1000 other unread emails.
Jira is designed to track issues so we may help you.

Please indicate if you are using a board with an echo canceler on it or not?



On Wed, Feb 17, 2010 at 7:56 AM, Eelco Br?lman <e.brolman at telecats.nl>wrote:

> Hi all,
>
> I have the same problems as described in the thread "Calls being parked on
> DTMF".
>
> We have a lab-test setup, which consist of the following:
>
> 1) a box with 4 E1 trunks with a call generator application (let's call it
> CG)
> 2) connected with a FreeSwitch box (Sangoma A104D card) (let's call that
> FS)
>
> The CG dials a number on the FS box (number 921000), which is bridged out
> back to the CG server (number 1000). The initiating application sends some
> DTMF, which is replied (echo-ed) by the receiving application (both on the
> CG server).
>
> In roughly about 80% of the calls, FS just disconnects the call:
>
> 2010-02-17 14:30:14.677125 [DEBUG] switch_ivr_originate.c:3105 Originate
> Resulted in Success: [OpenZAP/1:1/1000]
> 2010-02-17 14:30:14.677125 [DEBUG] switch_core_session.c:638 Send signal
> OpenZAP/1:1/1000 [BREAK]
> 2010-02-17 14:30:14.677125 [DEBUG] switch_core_session.c:638 Send signal
> OpenZAP/1:60/921000 [BREAK]
> 2010-02-17 14:30:14.677125 [DEBUG] switch_ivr_bridge.c:1175
> (OpenZAP/1:1/1000) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA
> 2010-02-17 14:30:14.677125 [DEBUG] switch_core_session.c:1018 Send signal
> OpenZAP/1:1/1000 [BREAK]
> 2010-02-17 14:30:14.677125 [DEBUG] switch_core_state_machine.c:314
> (OpenZAP/1:1/1000) Running State Change CS_EXCHANGE_MEDIA
> 2010-02-17 14:30:14.677125 [DEBUG] switch_core_state_machine.c:351
> (OpenZAP/1:1/1000) State EXCHANGE_MEDIA
> 2010-02-17 14:30:14.677125 [DEBUG] mod_openzap.c:558 CHANNEL EXCHANGE_MEDIA
> 2010-02-17 14:30:14.687135 [DEBUG] switch_core_session.c:699 Send signal
> OpenZAP/1:1/1000 [BREAK]
> 2010-02-17 14:30:14.687135 [DEBUG] switch_core_session.c:699 Send signal
> OpenZAP/1:60/921000 [BREAK]
> 2010-02-17 14:30:15.986882 [DEBUG] mod_openzap.c:684 queue DTMF [7]
> 2010-02-17 14:30:16.026733 [DEBUG] zap_io.c:1870 1:1 GENERATE DTMF [7]
> 2010-02-17 14:30:16.106418 [DEBUG] switch_ivr_bridge.c:469 OpenZAP/1:1/1000
> ending bridge by request from write function
> 2010-02-17 14:30:16.106418 [DEBUG] switch_core_session.c:638 Send signal
> OpenZAP/1:60/921000 [BREAK]
> 2010-02-17 14:30:16.126335 [DEBUG] switch_ivr_bridge.c:475 OpenZAP/1:1/1000
> ending bridge by request from read function
> 2010-02-17 14:30:16.126335 [DEBUG] switch_core_session.c:638 Send signal
> OpenZAP/1:1/1000 [BREAK]
> 2010-02-17 14:30:16.126335 [DEBUG] switch_ivr_bridge.c:562 BRIDGE THREAD
> DONE [OpenZAP/1:60/921000]
> 2010-02-17 14:30:16.126335 [DEBUG] switch_ivr_bridge.c:582 Send signal
> OpenZAP/1:1/1000 [BREAK]
> 2010-02-17 14:30:16.126335 [DEBUG] switch_ivr_bridge.c:562 BRIDGE THREAD
> DONE [OpenZAP/1:1/1000]
> 2010-02-17 14:30:16.126335 [DEBUG] switch_ivr_bridge.c:582 Send signal
> OpenZAP/1:60/921000 [BREAK]
> 2010-02-17 14:30:16.126335 [NOTICE] switch_ivr_bridge.c:634 Hangup
> OpenZAP/1:1/1000 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
> 2010-02-17 14:30:16.126335 [DEBUG] switch_channel.c:2063 Send signal
> OpenZAP/1:1/1000 [KILL]
>
>
> Where the line "ending bridge by request from write function" is curious of
> course.
>
> I'm running:
> - Freeswitch trunk (r16674) (updated, make clean, reconfigured and make
> install this morning)
> - Wanpipe 3.5.10.smg-2 (see
> http://wiki.sangoma.com/wanpipe-SmgPriInstallation)
> - Using the ozmod_sangoma_boost SCTP
>
>
> If anyone needs more information, tests or config files, I'll be happy to
> provide them.
>
>
>
> Kind regards,
>
> Eelco Br?lman
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100217/62761552/attachment-0001.html 

From ledoktre at meanie.us  Wed Feb 17 11:43:06 2010
From: ledoktre at meanie.us (Doc)
Date: Wed, 17 Feb 2010 13:43:06 -0600
Subject: [Freeswitch-users] One more thing..
In-Reply-To: <87f2f3b91002171129v59518b2bg9673212cff957a6c@mail.gmail.com>
References: <4B7B5EE9.4020705@meanie.us>	<191c3a031002170535i12c82731r5829417387dd3228@mail.gmail.com>	<191c3a031002170536g297a816ch2fa94eb2dcfbd16d@mail.gmail.com>	<2CEBE2C8-3982-49B6-A06F-9406C1D1F123@freeswitch.org>	<4B7C253F.9090905@meanie.us>	<5A00F97E-41C9-43AD-8CFA-1C8B87E5EECC@freeswitch.org>	<4B7C2DC3.20502@meanie.us>
	<87f2f3b91002171129v59518b2bg9673212cff957a6c@mail.gmail.com>
Message-ID: <4B7C46CA.6060708@meanie.us>

I should be able to update to the latest trunk.  Thanks for including 
the instructions.  The only "modification" I did to the source was to 
edit the modules file and enable mod_skypiax.  I will repost with 
results once I obtain them.  -Thanks,

Michael Collins wrote:
> Are you able to update your FS to the latest trunk? ("make current" in 
> the FS src dir) Also, I notice that you are on 16619M - the "M" 
> indicates a modification of some kind. Did you make any changes to the 
> FS source?
>
> -MC


From ledoktre at meanie.us  Wed Feb 17 16:13:07 2010
From: ledoktre at meanie.us (Doc)
Date: Wed, 17 Feb 2010 18:13:07 -0600
Subject: [Freeswitch-users] One more thing..
In-Reply-To: <87f2f3b91002171129v59518b2bg9673212cff957a6c@mail.gmail.com>
References: <4B7B5EE9.4020705@meanie.us>	<191c3a031002170535i12c82731r5829417387dd3228@mail.gmail.com>	<191c3a031002170536g297a816ch2fa94eb2dcfbd16d@mail.gmail.com>	<2CEBE2C8-3982-49B6-A06F-9406C1D1F123@freeswitch.org>	<4B7C253F.9090905@meanie.us>	<5A00F97E-41C9-43AD-8CFA-1C8B87E5EECC@freeswitch.org>	<4B7C2DC3.20502@meanie.us>
	<87f2f3b91002171129v59518b2bg9673212cff957a6c@mail.gmail.com>
Message-ID: <4B7C8613.606@meanie.us>

I have updated to "FreeSWITCH Version 1.0.trunk (16679)" per your 
instructions.  My issues with this request all seem to be handled now.  
I had one or two things yet in my other posting, but I will respond 
there.  THANK YOU !!

Michael Collins wrote:
> Are you able to update your FS to the latest trunk? ("make current" in 
> the FS src dir) Also, I notice that you are on 16619M - the "M" 
> indicates a modification of some kind. Did you make any changes to the 
> FS source?
>
> -MC
>
> On Wed, Feb 17, 2010 at 9:56 AM, Doc <ledoktre at meanie.us 
> <mailto:ledoktre at meanie.us>> wrote:
>
>     I did try changing the rtp time in my Sipura (advanced, sip, RTP
>     Packet
>     Size) from 0.030 to 0.020, and the error went away.  I was able to
>     then
>     test (and succeed) dialing from secondary skype user to skypiax, and
>     have the call bridged automatically to one of my Sipura SPA-1001's.  I
>     could speak into the phone, and hear it come through skype (on my
>     laptop) no problem.  When I would talk on the laptop, I would get
>     garble
>     back on the phone, but that could be a sound issue on my laptop (my
>     skype sometimes does this with Pulse audio on my laptop, Ubuntu
>     Karmic).
>
>     The interesting thing it did do, however, was eventually (within a
>     minute?) It threw a couple of errors :
>
>     2010-02-17 11:47:34.007909 [ERR] mod_skypiax.c:826 rev
>     16619M[(nil)|37     ][ERRORA  826  ][interface1][-1, 5,21] EXIT?
>     sent=-1
>     2010-02-17 11:47:34.027776 [ERR] mod_skypiax.c:826 rev
>     16619M[(nil)|37     ][ERRORA  826  ][interface1][-1, 5,21] EXIT?
>     sent=-1
>     2010-02-17 11:47:34.047643 [ERR] mod_skypiax.c:826 rev
>     16619M[(nil)|37     ][ERRORA  826  ][interface1][-1, 5,21] EXIT?
>     sent=-1
>
>     And on a subsequent test, I received no audio (the above error rolling
>     on the console), and within a matter of seconds, FS crashed with
>     this :
>
>     Segmentation fault (core dumped)
>
>     I have another request open on the group actually for the above error,
>     so I think it'd make sense to discuss further there (rather than
>     duplicating material).
>
>     Thanks!  At any rate, the error for the missing codec, etc, seemed
>     to be
>     gone once I updated the RTP time in my ATA.  Hope this helps someone!!
>
>     Brian West wrote:
>     > change the rtp time to 0.020 from 0.030
>     >
>     > /b
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   


From ledoktre at meanie.us  Wed Feb 17 16:19:07 2010
From: ledoktre at meanie.us (Doc)
Date: Wed, 17 Feb 2010 18:19:07 -0600
Subject: [Freeswitch-users] One more thing..
In-Reply-To: <191c3a031002171123r416f6413wbecddcea498b5616@mail.gmail.com>
References: <4B7B5EE9.4020705@meanie.us>	<191c3a031002170535i12c82731r5829417387dd3228@mail.gmail.com>	<191c3a031002170536g297a816ch2fa94eb2dcfbd16d@mail.gmail.com>	<2CEBE2C8-3982-49B6-A06F-9406C1D1F123@freeswitch.org>	<4B7C253F.9090905@meanie.us>	<5A00F97E-41C9-43AD-8CFA-1C8B87E5EECC@freeswitch.org>	<4B7C2DC3.20502@meanie.us>
	<191c3a031002171123r416f6413wbecddcea498b5616@mail.gmail.com>
Message-ID: <4B7C877B.1080303@meanie.us>

I have updated to "FreeSWITCH Version 1.0.trunk (16679)" which seems to 
have cleared up part # 2 of my question (issues with the errors being 
thrown on the screen during a call, and then subsequent calls failing).

About the DTMF issues, for fun, I got onto Skype chat.  I was told by 
the technician there that Skype does not support DTMF on SkypeIn -OR- 
Skype for SIP.  He said sorry, but you won't get anywhere.  I was also 
told they are not doing any blocking, that the reason its not working is 
probably due to all the transcoding that is taking place has degraded 
the signal too much.  I guess I might try and see if there is a module 
available for a voice driven IVR? (any suggestions?)

And on the rest of my first posting, any thoughts on why I must push a # 
sign when dialing an extension, etc, in FS?  If I don't, I have to wait 
like 30 seconds.  Also, the errors when I run startskype,sh (I am 
thinking must be no big deal) listed, are they any problem :

expected keysym, got XF86KbdLightOnOff: line 70 of pc
expected keysym, got XF86KbdBrightnessDown: line 71 of pc
expected keysym, got XF86KbdBrightnessUp: line 72 of pc
Could not init font path element /usr/share/fonts/X11/cyrillic, removing 
from list!
Could not init font path element 
/var/lib/defoma/x-ttcidfont-conf.d/dirs/TrueType, removing from list!
ALSA lib pcm.c:2211:(snd_pcm_open_noupdate) Unknown PCM null
ALSA lib pcm.c:2211:(snd_pcm_open_noupdate) Unknown PCM null

Thanks again you guys for all your help!

Doc

Anthony Minessale wrote:
> keep updating, the maintainer of mod_skypeiax is adding new patches 
> every few hours.
> You can also join irc on irc.freenode.net <http://irc.freenode.net> 
> #freeswitch and #freeswitch-dev to interact with him live.
>
>
> One hint around here when we ask if you updated we work in 1 minute 
> increments.


From anthony.minessale at gmail.com  Wed Feb 17 16:37:17 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 17 Feb 2010 18:37:17 -0600
Subject: [Freeswitch-users] One more thing..
In-Reply-To: <191c3a031002171636h7232cc81kd54c22532ba04b28@mail.gmail.com>
References: <4B7B5EE9.4020705@meanie.us>
	<191c3a031002170536g297a816ch2fa94eb2dcfbd16d@mail.gmail.com>
	<2CEBE2C8-3982-49B6-A06F-9406C1D1F123@freeswitch.org>
	<4B7C253F.9090905@meanie.us>
	<5A00F97E-41C9-43AD-8CFA-1C8B87E5EECC@freeswitch.org>
	<4B7C2DC3.20502@meanie.us>
	<191c3a031002171123r416f6413wbecddcea498b5616@mail.gmail.com>
	<4B7C877B.1080303@meanie.us>
	<191c3a031002171634j33f70843j40ce8a3d60044d0b@mail.gmail.com>
	<191c3a031002171636h7232cc81kd54c22532ba04b28@mail.gmail.com>
Message-ID: <191c3a031002171637t5c1a2339jadacf8cd89b6fe52@mail.gmail.com>

One hint is to add keywords to the subject like "problems with skypeiax" so
the guy will see your issue he doesn't read every mail looking for skype
issues.  Even better, open a new issue on jira.freeswitch.org under skypeiax
related category......

On Feb 17, 2010 6:23 PM, "Doc" <ledoktre at meanie.us> wrote:

I have updated to "FreeSWITCH Version 1.0.trunk (16679)" which seems to
have cleared up part # 2 of my question (issues with the errors being
thrown on the screen during a call, and then subsequent calls failing).

About the DTMF issues, for fun, I got onto Skype chat.  I was told by
the technician there that Skype does not support DTMF on SkypeIn -OR-
Skype for SIP.  He said sorry, but you won't get anywhere.  I was also
told they are not doing any blocking, that the reason its not working is
probably due to all the transcoding that is taking place has degraded
the signal too much.  I guess I might try and see if there is a module
available for a voice driven IVR? (any suggestions?)

And on the rest of my first posting, any thoughts on why I must push a #
sign when dialing an extension, etc, in FS?  If I don't, I have to wait
like 30 seconds.  Also, the errors when I run startskype,sh (I am
thinking must be no big deal) listed, are they any problem :

expected keysym, got XF86KbdLightOnOff: line 70 of pc
expected keysym, got XF86KbdBrightnessDown: line 71 of pc
expected keysym, got XF86KbdBrightnessUp: line 72 of pc
Could not init font path element /usr/share/fonts/X11/cyrillic, removing
from list!
Could not init font path element
/var/lib/defoma/x-ttcidfont-conf.d/dirs/TrueType, removing from list!
ALSA lib pcm.c:2211:(snd_pcm_open_noupdate) Unknown PCM null
ALSA lib pcm.c:2211:(snd_pcm_open_noupdate) Unknown PCM null

Thanks again you guys for all your help!

Doc


Anthony Minessale wrote:
> keep updating, the maintainer of mod_skypeiax is adding new patches
> e...
> You can also join irc on irc.freenode.net <http://irc.freenode.net>

> #freeswitch and #freeswitch-dev to interact with him live.
>
>
> One hint around here when we ask ...

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists...
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100217/8591a34b/attachment.html 

From troy at tlainvestments.com  Wed Feb 17 16:44:36 2010
From: troy at tlainvestments.com (Troy Anderson)
Date: Wed, 17 Feb 2010 17:44:36 -0700
Subject: [Freeswitch-users] tone_detect timeout
Message-ID: <EC840323-2A76-426E-A386-8DA111195768@tlainvestments.com>

In the wiki about tone_detect, the docs state that the timeout is in seconds (e.g. +2 for 2 seconds from now), but all the examples have +5000, suggesting that it may really be milliseconds.  I'd be happy to update the wiki if someone could say if it's seconds or millisconds or otherwise.

I tried tracing it back in the code, but got lost looking for the definition of switch_media_bug_t!

http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_tone_detect

Thanks!
Troy

From wangdq.no1 at gmail.com  Wed Feb 17 19:17:26 2010
From: wangdq.no1 at gmail.com (daqiang wang)
Date: Thu, 18 Feb 2010 11:17:26 +0800
Subject: [Freeswitch-users] how to use mod_erlang ?
Message-ID: <ecd05ce31002171917w2713f8detcee3f7e36039e9f2@mail.gmail.com>

hello !
I test mod_erlang from : http://wiki.freeswitch.org/wiki/Mod_erlang_event
but when I input :
 >

{foo, freeswitch at localhost } ! {api, status, ""}.
I received
  (test at wangdq-laptop)2>
=ERROR REPORT==== 18-Feb-2010::11:10:13 ===
Error in process <0.41.0> on node 'test at wangdq-laptop' with exit
value: {badarg,[{erlang,list_to_existing_atom,["freeswitch at wangdq-laptop"]},{dist_util,recv_challenge,1},{dist_util,handshake_we_started,1}]}

and at freeswitch console :
   [NOTICE] mod_erlang_event.c:1720 Ignorable error in ei_accept -
probable bad client version, bad cookie or bad nodename


and I think the mod_erlang config file is :


<configuration name="erlang_event.conf" description="Erlang Socket Client">
  <settings>
    <param name="listen-ip" value="0.0.0.0"/>
    <param name="listen-port" value="8031"/>
    <param name="cookie" value="ClueCon"/>
    <param name="shortname" value="true"/>
    <!-- in additon to cookie, optionally restrict by ACL -->
    <param name="apply-inbound-acl" value="lan"/>
    <!-- alternative is "binary" -->
    <param name="encoding" value="string"/>
    <!-- provide compatability with previous OTP release (use with care) -->
    <!--<param name="compat-rel" value="12"/> -->
  </settings>
</configuration>
~


 Thanks !
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100218/055ca0c0/attachment.html 

From mike at jerris.com  Wed Feb 17 21:10:54 2010
From: mike at jerris.com (Michael Jerris)
Date: Thu, 18 Feb 2010 00:10:54 -0500
Subject: [Freeswitch-users] ivr from mysql
In-Reply-To: <7c74f5761002151525p7cc690ces632d46c6ab5e236c@mail.gmail.com>
References: <7c74f5761002150824i41a2ed5did008f36cbb4499a4@mail.gmail.com>
	<87f2f3b91002151002v6bbc3e7fgb67dd6d3e72a915@mail.gmail.com>
	<7c74f5761002151425g512fa685n13c58d8f4b894740@mail.gmail.com>
	<0FD30AAA-C2E2-4228-A09B-DB09F4E26811@avgs.ca>
	<7c74f5761002151525p7cc690ces632d46c6ab5e236c@mail.gmail.com>
Message-ID: <E9EBF591-5DEC-4FF3-B2D8-20DDA54B4FF5@jerris.com>

an example is available here :   http://svn.freeswitch.org/svn/freeswitch/trunk/conf/ivr_menus/demo_ivr.xml

Mike

On Feb 15, 2010, at 6:25 PM, michal kalinowski wrote:
> Could you insert several examples here?



From mike at jerris.com  Wed Feb 17 21:13:08 2010
From: mike at jerris.com (Michael Jerris)
Date: Thu, 18 Feb 2010 00:13:08 -0500
Subject: [Freeswitch-users] Sending message notifications with openzap
In-Reply-To: <e749d5a76cc31b4a3bb2459aaf8b78d9@thom.fr.eu.org>
References: <e749d5a76cc31b4a3bb2459aaf8b78d9@thom.fr.eu.org>
Message-ID: <1B676793-9D83-4C8D-A9C4-78891F8C70B5@jerris.com>

This all uses the event system.  Take a look at how we handle the events in mod_sofia, it should be pretty easy to do the same in mod_openzap.

Mike

On Feb 16, 2010, at 4:33 AM, Fran?ois Legal wrote:
> I was wondering whether or not a facility was available to send message waiting indicator (from mpd_voicemail) using openzap.
> 
> I know this feature is not available in openzap (I'm in the process of coding it for analog channels) but as message indication can be sent via mod_sofia, I wonder if a frame is available in FS core.
> 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100218/0fd81b01/attachment-0001.html 

From scottferri09 at gmail.com  Thu Feb 18 00:12:04 2010
From: scottferri09 at gmail.com (Scott Fernandez)
Date: Thu, 18 Feb 2010 13:42:04 +0530
Subject: [Freeswitch-users] Establishing a Call from .Net based
	application
In-Reply-To: <6E8D2069C08AA84A83D336E996AE4C67032C9D5346@mse17be1.mse17.exchange.ms>
References: <874941.17255.qm@web33502.mail.mud.yahoo.com>
	<df97b9c51001310545n54195cadr9faa3db46917ca9e@mail.gmail.com>
	<367751821002010737o4ca83476j62c96cb60c8a456a@mail.gmail.com>
	<df97b9c51002130752h48aae119ja80797ab8a3f5b2d@mail.gmail.com>
	<6E8D2069C08AA84A83D336E996AE4C67032C9D5346@mse17be1.mse17.exchange.ms>
Message-ID: <df97b9c51002180012wc4d9421ye6e81ae6199321c8@mail.gmail.com>

Hi Diego & Michael,

Thanks for your reply and support.

However, I have some clarifications required from both of you.

1. Here is the question for Diego,

Simple Example:

using FreeSWITCH;
using FreeSWITCH.Native;

namespace BITS.Ivr.Bp.Server
{
 public class CIV_BPFSProcess : FreeSWITCH.IAppPlugin
{
 public void Run(AppContext context)
 {
  //answer call
  context.Session.Answer();
  //sleep 2 seconds
  context.Session.sleep(2000, 1);
  //hangup call
  context.Session.Hangup("
NORMAL_CLEARING");
 }
 }
}

I understand that the concept of your example code. However, would like to
know as to how would my .NET C# know the IP address of Freeswitch to talk to
it as there is no indication for that?. If not here, where would we need to
reference the IP address of FS in .NET code?

I guess the IP address of FS needs to be mentioned in the Target section of
the below web.config file in .NET. If I am right, how to specify the IP
address over here. If I am wrong, please let me know where do we need to
mention the IP address of FS.

    <configuration>
            <dllmap dll="mod_managed.dll" target="mod_managed.so"/>
    </configuration>


2. Here is the question for Michael,

You mentioned that "mod_managed.so will be in your freeswitch mod directory".
This is very clear and what is mod_managed.dll in my .NET application and
the purpose of it?

Thanks for all your help.

Regards,
Scott.



On Sun, Feb 14, 2010 at 1:15 AM, Michael Giagnocavo <mgg at giagnocavo.net>wrote:

>
> 2. There is a configuration settings required to Map the "DLL" to ".so"
> object in CentOS.
> Now, the question is which DLL and .so file to be made available and where?
>
> ?
>
> If you are experiencing NullReferenceExceptions with any plugin run through
> the dialplan, make sure you have included the appropriate entry in your
> dllmap <http://mono-project.com/Config_DllMap> configuration:
>
>  <dllmap dll="mod_managed.dll" target="mod_managed.so" os="!windows"/>
>
> ?
>
> mod_managed.so will be in your freeswitch mod directory.
>
>
> All I need is to initiate a call from .NET application and then it should
> talk to mod_managed module and establish a call. Secondly, I need to know
> the status of the call such as Ringing, Active, Hangup etc.
>
> To initiate a call, try ManagedSession.Originate.
>
> -Michael
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100218/3fadf0d3/attachment.html 

From mcampbellsmith at gmail.com  Thu Feb 18 00:22:54 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Thu, 18 Feb 2010 19:22:54 +1100
Subject: [Freeswitch-users] Forbidden using UDP, works with TCP/TLS
In-Reply-To: <147993F8-9C9E-4D96-9AFE-5EEE4E8F93F0@freeswitch.org>
References: <33c87fa31002101824h21e3bf7lb13b68e8d4fd36a2@mail.gmail.com>
	<33c87fa31002130343k7a01e9abt9c33360399f246f8@mail.gmail.com>
	<C37AB6F3-F712-41B6-9D6D-7A8AE3C6CABB@freeswitch.org>
	<33c87fa31002131336j7b05db7aof6ba6e3439aebc41@mail.gmail.com>
	<9D702362-BB5F-4C0F-9EE7-DABA3B926C36@freeswitch.org>
	<33c87fa31002141708s72cb7246h5372ecea40be615d@mail.gmail.com>
	<1888B1CA-D0A8-407C-9C44-B86E0F1925D6@freeswitch.org>
	<33c87fa31002142017v49d8e7c5nf572d086ec89282b@mail.gmail.com>
	<33c87fa31002142109l5fcabed3ydba21b865ed6cbbb@mail.gmail.com>
	<147993F8-9C9E-4D96-9AFE-5EEE4E8F93F0@freeswitch.org>
Message-ID: <33c87fa31002180022g2e91c8efs65f52bb4275cb59@mail.gmail.com>

Thanks for looking at that Brian.  If the token is not used, I assume
this is not the reason for FS rejecting the Registration attempt?

Also, when is stale=true set in the WWW-Authentication?  I notice that
for this device, I do not see stale=true, but for all my other
devices, I see stale=true (at least from the logs I've taken today).


On Tue, Feb 16, 2010 at 2:03 AM, Brian West <brian at freeswitch.org> wrote:
> Ok looks like the token is not used at all in digest auth. ?This is the
> first time I have seen a device send back something other than auth or
> auth-int.
> /b
> On Feb 14, 2010, at 11:09 PM, Mark Campbell-Smith wrote:
>
> A little more testing. ???I noticed that the Authorization field
> differs when TCP or UDP:
>
> UDP (fails)
> Digest username=\"2010\", realm=\"mydns.dyndns.org\",
> nonce=\"e5f119c6-19e9-11df-bd09-773b7a755f78\",
> uri=\"sip:mydns.dyndns.org:5060\",
> response=\"e37be3e49c159d4f98e8bd04b36f2bd7\", algorithm=MD5,
> qop=\"1fffcc9f\"
>
> TCP (works)
> Digest
> username=\"2010\",realm=\"mydns.dyndns.org\",nonce=\"5d9e75c2-19ea-11df-bd0b-773b7a755f78\",uri=\"sip:mydns.dyndns.org:5060\",algorithm=MD5,response=\"45ba55d3fbafcbf2bc2aa6418656ecc2\",qop=auth,nc=00000001,cnonce=\"3a650454\"
>
> Is qop = 1fffcc9f valid in SIP? ?Does a cnonce need to be included also?
>
> FS sends qop=auth in the Unauthorized response.
>
> Thanks
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From brian at microcomaustralia.com.au  Thu Feb 18 00:46:38 2010
From: brian at microcomaustralia.com.au (Brian May)
Date: Thu, 18 Feb 2010 19:46:38 +1100
Subject: [Freeswitch-users] building for Lenny
Message-ID: <3c5cf5261002180046vb0015e7t787599c9903aedc@mail.gmail.com>

Hello,

What is the most recent instructions?

According to <http://wiki.freeswitch.org/wiki/Installation_Guide>, I
should install the freeswitch-sounds-music-8000 package, but no sounds
packages are created in the build process.

Thanks
-- 
Brian May <brian at microcomaustralia.com.au>


From mcampbellsmith at gmail.com  Thu Feb 18 00:56:47 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Thu, 18 Feb 2010 19:56:47 +1100
Subject: [Freeswitch-users] building for Lenny
In-Reply-To: <3c5cf5261002180046vb0015e7t787599c9903aedc@mail.gmail.com>
References: <3c5cf5261002180046vb0015e7t787599c9903aedc@mail.gmail.com>
Message-ID: <33c87fa31002180056g51ef8496qa9a35746d0fe75ec@mail.gmail.com>

I think you need to download the gzip file from http://files.freeswitch.org/

latest.freeswitch.org does not contain sound files as far as I'm aware ..

On Thu, Feb 18, 2010 at 7:46 PM, Brian May
<brian at microcomaustralia.com.au> wrote:
> Hello,
>
> What is the most recent instructions?
>
> According to <http://wiki.freeswitch.org/wiki/Installation_Guide>, I
> should install the freeswitch-sounds-music-8000 package, but no sounds
> packages are created in the build process.
>
> Thanks
> --
> Brian May <brian at microcomaustralia.com.au>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From brian at microcomaustralia.com.au  Thu Feb 18 01:10:04 2010
From: brian at microcomaustralia.com.au (Brian May)
Date: Thu, 18 Feb 2010 20:10:04 +1100
Subject: [Freeswitch-users] building for Lenny
In-Reply-To: <33c87fa31002180056g51ef8496qa9a35746d0fe75ec@mail.gmail.com>
References: <3c5cf5261002180046vb0015e7t787599c9903aedc@mail.gmail.com>
	<33c87fa31002180056g51ef8496qa9a35746d0fe75ec@mail.gmail.com>
Message-ID: <3c5cf5261002180110he0ae815uc713f77f289ac625@mail.gmail.com>

On 18 February 2010 19:56, Mark Campbell-Smith <mcampbellsmith at gmail.com> wrote:
> I think you need to download the gzip file from http://files.freeswitch.org/
>
> latest.freeswitch.org does not contain sound files as far as I'm aware ..

Does this mean I shouldn't be trying to use the debian packages? *.deb
files would certainly make it easier to compile it on a fast computer
and then transfer to my net5501.

Unfortunately, the website seems have a lot of old information,
including references to obsolete Ubuntu Hardy packages.

I see a thread from late last year that suggests there should be
prebuilt packages, everything I can find so far seems very old
however.
-- 
Brian May <brian at microcomaustralia.com.au>


From devel at thom.fr.eu.org  Thu Feb 18 01:50:30 2010
From: devel at thom.fr.eu.org (=?UTF-8?Q?Fran=C3=A7ois_Legal?=)
Date: Thu, 18 Feb 2010 10:50:30 +0100
Subject: [Freeswitch-users] Sending message notifications with openzap
In-Reply-To: <1B676793-9D83-4C8D-A9C4-78891F8C70B5@jerris.com>
References: <e749d5a76cc31b4a3bb2459aaf8b78d9@thom.fr.eu.org>
	<1B676793-9D83-4C8D-A9C4-78891F8C70B5@jerris.com>
Message-ID: <0ae2b96926e5340a50e659891efb26cc@thom.fr.eu.org>



Thanks, 

yes I could see that this was handled with events. 

Could you
please give more details (that does not seem obvious to me while looking at
sofia_presence.c) 

When and why is the event triggered ? 

What
information do I get with the event ? 

Then I have a design question, is
it mandatory to (as this is done in mod_sofia but I guess for a lot of
reasons) process the event in a separate thread ? 

Thanks for the
information (and sorry if my questions are not relevant, I don't know the
event system at all). 

Fran?ois 

On Thu, 18 Feb 2010 00:13:08 -0500,
Michael Jerris  wrote: This all uses the event system. Take a look at how
we handle the events in mod_sofia, it should be pretty easy to do the same
in mod_openzap.  Mike 
  On Feb 16, 2010, at 4:33 AM, Fran?ois Legal wrote:


I was wondering whether or not a facility was available to send message
waiting indicator (from mpd_voicemail) using openzap. 

I know this feature
is not available in openzap (I'm in the process of coding it for analog
channels) but as message indication can be sent via mod_sofia, I wonder if
a frame is available in FS core.     

 
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100218/a4daf1bd/attachment.html 

From mgg at giagnocavo.net  Thu Feb 18 03:12:29 2010
From: mgg at giagnocavo.net (Michael Giagnocavo)
Date: Thu, 18 Feb 2010 06:12:29 -0500
Subject: [Freeswitch-users] Establishing a Call from .Net
	based	application
In-Reply-To: <df97b9c51002180012wc4d9421ye6e81ae6199321c8@mail.gmail.com>
References: <874941.17255.qm@web33502.mail.mud.yahoo.com>
	<df97b9c51001310545n54195cadr9faa3db46917ca9e@mail.gmail.com>
	<367751821002010737o4ca83476j62c96cb60c8a456a@mail.gmail.com>
	<df97b9c51002130752h48aae119ja80797ab8a3f5b2d@mail.gmail.com>
	<6E8D2069C08AA84A83D336E996AE4C67032C9D5346@mse17be1.mse17.exchange.ms>
	<df97b9c51002180012wc4d9421ye6e81ae6199321c8@mail.gmail.com>
Message-ID: <6E8D2069C08AA84A83D336E996AE4C67032C9D5BAD@mse17be1.mse17.exchange.ms>

I'm not sure what the FreeSWITCH APIs are to figure out what IP Sofia SIP has bound to. Whatever it is, you'd call the same thing in C#. What do you want to do with the API?

mod_managed.dll or .so is the FreeSWITCH native code module that loads the CLR or Mono into the FreeSWITCH process and loads FreeSWITCH.Managed.dll. The managed DLL contains the bulk of the managed-unmanaged interop code (.NET definitions of all the FS C functions).

-Michael

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Scott Fernandez
Sent: Thursday, February 18, 2010 1:12 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Establishing a Call from .Net based application

Hi Diego & Michael,

Thanks for your reply and support.

However, I have some clarifications required from both of you.

1. Here is the question for Diego,

Simple Example:

using FreeSWITCH;
using FreeSWITCH.Native;

namespace BITS.Ivr.Bp.Server
{
 public class CIV_BPFSProcess : FreeSWITCH.IAppPlugin
{
 public void Run(AppContext context)
 {
  //answer call
  context.Session.Answer();
  //sleep 2 seconds
  context.Session.sleep(2000, 1);
  //hangup call
  context.Session.Hangup("
NORMAL_CLEARING");
 }
 }
}

I understand that the concept of your example code. However, would like to know as to how would my .NET C# know the IP address of Freeswitch to talk to it as there is no indication for that?. If not here, where would we need to reference the IP address of FS in .NET code?

I guess the IP address of FS needs to be mentioned in the Target section of the below web.config file in .NET. If I am right, how to specify the IP address over here. If I am wrong, please let me know where do we need to mention the IP address of FS.

    <configuration>
            <dllmap dll="mod_managed.dll" target="mod_managed.so"/>
    </configuration>


2. Here is the question for Michael,

You mentioned that "mod_managed.so will be in your freeswitch mod directory". This is very clear and what is mod_managed.dll in my .NET application and the purpose of it?

Thanks for all your help.

Regards,
Scott.


On Sun, Feb 14, 2010 at 1:15 AM, Michael Giagnocavo <mgg at giagnocavo.net<mailto:mgg at giagnocavo.net>> wrote:

2. There is a configuration settings required to Map the "DLL" to ".so" object in CentOS.
Now, the question is which DLL and .so file to be made available and where?
"
If you are experiencing NullReferenceExceptions with any plugin run through the dialplan, make sure you have included the appropriate entry in your dllmap<http://mono-project.com/Config_DllMap> configuration:
 <dllmap dll="mod_managed.dll" target="mod_managed.so" os="!windows"/>
"
mod_managed.so will be in your freeswitch mod directory.

All I need is to initiate a call from .NET application and then it should talk to mod_managed module and establish a call. Secondly, I need to know the status of the call such as Ringing, Active, Hangup etc.
To initiate a call, try ManagedSession.Originate.
-Michael

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100218/adeb933c/attachment-0001.html 

From matt at webcontracts.co.uk  Thu Feb 18 04:02:50 2010
From: matt at webcontracts.co.uk (Matthew Law)
Date: Thu, 18 Feb 2010 12:02:50 -0000
Subject: [Freeswitch-users] building for Lenny
In-Reply-To: <3c5cf5261002180110he0ae815uc713f77f289ac625@mail.gmail.com>
References: <3c5cf5261002180046vb0015e7t787599c9903aedc@mail.gmail.com>
	<33c87fa31002180056g51ef8496qa9a35746d0fe75ec@mail.gmail.com>
	<3c5cf5261002180110he0ae815uc713f77f289ac625@mail.gmail.com>
Message-ID: <a9f4977a70b3fcf1b30413ea562ca962.squirrel@www.webcontracts.co.uk>


On Thu, February 18, 2010 9:10 am, Brian May wrote:
> On 18 February 2010 19:56, Mark Campbell-Smith <mcampbellsmith at gmail.com>
> wrote:
>> I think you need to download the gzip file from
>> http://files.freeswitch.org/
>>
>> latest.freeswitch.org does not contain sound files as far as I'm aware
>> ..
>
> Does this mean I shouldn't be trying to use the debian packages? *.deb
> files would certainly make it easier to compile it on a fast computer
> and then transfer to my net5501.
>
> Unfortunately, the website seems have a lot of old information,
> including references to obsolete Ubuntu Hardy packages.
>
> I see a thread from late last year that suggests there should be
> prebuilt packages, everything I can find so far seems very old
> however.

Brian,

I had similar problems recently and decided to compile it from svn trunk. 
The dependencies for the build are all available in apt, so I installed
those, checked out trunk and configured it with a prefix of
/usr/local/freeswitch.

I got the impression it is self contained, so you should be able to tar up
the entire /usr/local/freeswitch dir and scp it over - I stand to be
corrected on that, though.

Good luck with it,

Matt.




From scott.torr.fs at letterboxes.org  Thu Feb 18 04:35:52 2010
From: scott.torr.fs at letterboxes.org (Scott Torr)
Date: Thu, 18 Feb 2010 23:35:52 +1100
Subject: [Freeswitch-users] One more thing..
In-Reply-To: <4B7C877B.1080303@meanie.us>
References: <4B7B5EE9.4020705@meanie.us>
	<191c3a031002170535i12c82731r5829417387dd3228@mail.gmail.com>
	<191c3a031002170536g297a816ch2fa94eb2dcfbd16d@mail.gmail.com>
	<2CEBE2C8-3982-49B6-A06F-9406C1D1F123@freeswitch.org>
	<4B7C253F.9090905@meanie.us>
	<5A00F97E-41C9-43AD-8CFA-1C8B87E5EECC@freeswitch.org>
	<4B7C2DC3.20502@meanie.us><191c3a031002171123r416f6413wbecddcea498b5616@mail.gmail.com>
	<4B7C877B.1080303@meanie.us>
Message-ID: <1266496552.11533.1360626205@webmail.messagingengine.com>

On Wed, 17 Feb 2010 18:19 -0600, "Doc" <ledoktre at meanie.us> wrote:
> About the DTMF issues, for fun, I got onto Skype chat.  I was told by 
> the technician there that Skype does not support DTMF on SkypeIn -OR- 
> Skype for SIP.  He said sorry, but you won't get anywhere. 



Hi Doc,

I can confirm that out-of-band DTMF signaling works for the Skype SIP
product using a SkypeIn PSTN number.



>From the Skype support site.
https://support.skype.com/en/faq/FA10292/Do-you-support-DTMF-with-Skype-for-SIP?frompage=search&q=dtmf

<quote>
Do you support DTMF with Skype for SIP?

Yes. Skype supports out-of-band DTMF signaling in accordance with the
RFC 2833 standard. Details of RFC 2833 are located on the IETF website.

We do *not* support in-band DTMF signaling. 
</quote>


Regards,
Scott Torr


regards


From vetali100 at gmail.com  Thu Feb 18 04:52:46 2010
From: vetali100 at gmail.com (Vitalii Colosov)
Date: Thu, 18 Feb 2010 14:52:46 +0200
Subject: [Freeswitch-users] Other party does not hear voice when
	connecting with X-Lite
In-Reply-To: <3e6d7b0c1002150050g3ccd23aas2e7f05f230bff4a7@mail.gmail.com>
References: <3e6d7b0c1002150050g3ccd23aas2e7f05f230bff4a7@mail.gmail.com>
Message-ID: <a858e7691002180452n6ef91ed7k3d9b2f7246d773f1@mail.gmail.com>

Thanks a lot for your advices and sorry for confusion, my description was
not consistent, agree...

We are testing X-Lite client from 4 different machines and using 2 different
FS servers and getting different results - sometimes everything works,
sometimes only signalling works - no voice, and sometimes even signalling
does not work.
YATE client always works without any such issues...

When we will come to some sort of understanding, I will try to share the
results.
If we will stuck, I will try to get some debug information and continue
bothering the professionals. :)

 BTW, I tried to change the codecs in X-Lite, same result.
It is for sure related to our network configuration + X-Lite's way of
sending SIP data, looks like...

Regards,
Vitalii





2010/2/15 Steven Brown <steve at justfone.com>

> I had the same problem with XLite / Freeswitch  a while back that I never
> fully understood,  however the problem vanished when I disabled all codecs
> on Xlite except G711 uLaw, as I say, no idea what was going on but this
> might be worth trying.
>
> Steve
>
>
> Message: 1
> Date: Sun, 14 Feb 2010 09:04:09 -0600
> From: Anthony Minessale <anthony.minessale at gmail.com>
> Subject: Re: [Freeswitch-users] Other party does not hear voice when
>        connecting with X-Lite
> To: freeswitch-users at lists.freeswitch.org
> Message-ID:
>        <191c3a031002140704g705bfc73rd8dd103f3d846062 at mail.gmail.com
> >
> Content-Type: text/plain; charset="iso-8859-1"
>
>
> You need to describe this again its too confusing now.
> List each device, freeswitch, the phones and which ip and combo of addrs it
> uses with the topology clearly stated.
> Your attempt to simplify your explanation is actually making it harder to
> follow.
> Also consider a debug/sip trace as well.
>
> Include
> sofia status profile default.
>
> Then capture a test call after entering these commands.
>
> console loglevel debug.
> sofa profile internal siptrace on
>
> On Feb 14, 2010 4:12 AM, "Vitalii Colosov" <vetali100 at gmail.com> wrote:
>
> No, it is done on the different PCs...
>
> Sorry, when I started the topic, I have described the problem how it is
> visible from PC of my friend.
> Then I tried to reproduce the same on my own PC, and you are right...I was
> not able to hear anything as well, not only both party wasn't.
> Also, from my PC I was NOT able to hear guitar on test number "9999".
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100218/9d8e2527/attachment.html 

From brian at freeswitch.org  Thu Feb 18 05:04:13 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 18 Feb 2010 07:04:13 -0600
Subject: [Freeswitch-users] Forbidden using UDP, works with TCP/TLS
In-Reply-To: <33c87fa31002180022g2e91c8efs65f52bb4275cb59@mail.gmail.com>
References: <33c87fa31002101824h21e3bf7lb13b68e8d4fd36a2@mail.gmail.com>
	<33c87fa31002130343k7a01e9abt9c33360399f246f8@mail.gmail.com>
	<C37AB6F3-F712-41B6-9D6D-7A8AE3C6CABB@freeswitch.org>
	<33c87fa31002131336j7b05db7aof6ba6e3439aebc41@mail.gmail.com>
	<9D702362-BB5F-4C0F-9EE7-DABA3B926C36@freeswitch.org>
	<33c87fa31002141708s72cb7246h5372ecea40be615d@mail.gmail.com>
	<1888B1CA-D0A8-407C-9C44-B86E0F1925D6@freeswitch.org>
	<33c87fa31002142017v49d8e7c5nf572d086ec89282b@mail.gmail.com>
	<33c87fa31002142109l5fcabed3ydba21b865ed6cbbb@mail.gmail.com>
	<147993F8-9C9E-4D96-9AFE-5EEE4E8F93F0@freeswitch.org>
	<33c87fa31002180022g2e91c8efs65f52bb4275cb59@mail.gmail.com>
Message-ID: <7EA7C626-D317-4011-BF6E-A7411C765EC1@freeswitch.org>

I think your device is broken.

/b

On Feb 18, 2010, at 2:22 AM, Mark Campbell-Smith wrote:

> Thanks for looking at that Brian.  If the token is not used, I assume
> this is not the reason for FS rejecting the Registration attempt?
> 
> Also, when is stale=true set in the WWW-Authentication?  I notice that
> for this device, I do not see stale=true, but for all my other
> devices, I see stale=true (at least from the logs I've taken today).



From testeador01 at gmail.com  Thu Feb 18 05:27:15 2010
From: testeador01 at gmail.com (Milena)
Date: Thu, 18 Feb 2010 08:27:15 -0500
Subject: [Freeswitch-users] Greetings and a couple of questions
In-Reply-To: <4B7C261C.1040802@meanie.us>
References: <4B7B5A46.3090304@meanie.us>
	<1266408503.10430.1360421639@webmail.messagingengine.com>
	<4B7C261C.1040802@meanie.us>
Message-ID: <c29b930b1002180527l5f8c3f40icd905072f302f651@mail.gmail.com>

Hello!

For ASR menus check this out:
http://wiki.freeswitch.org/wiki/Mod_pocketsphinx
<http://wiki.freeswitch.org/wiki/Mod_pocketsphinx>I wouldn't say it is
perfect but it is nice, on the other hand there is this dialplan tool that
makes fs detect inband dtmf:
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf

<http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf>hope it
helps.

-Milena

2010/2/17 Doc <ledoktre at meanie.us>

> Once you mentioned the in-band versus out-of-band DTMF, it made sense.
> I've been reading on it since your post, and I'm going to try a few
> things.  Wonder if there is any app for voice controlled IVR (so many
> systems seem to support it these days, it would be a nice way to
> circumvent the DTMF issue... :-) ).
>
> Any ideas on the rest of my points?   -Thanks,
>
> Scott Torr wrote:
> > On Tue, 16 Feb 2010 20:53 -0600, "Doc" <ledoktre at meanie.us> wrote:
> >
> >> 1) I am able to see a call come in, and it gets routed to the sample IVR
> >> to start.  The first thing off is that when I dial from PSTN ->
> >> Skype-In, it does not let me push any buttons.  If I launch a second
> >> skype client, and dial the skype user on FS directly, it works fine.
> >> Any ideas?
> >>
> >
> > Hi Doc,
> >
> > When you dial in from the PSTN the 'push button' events are present as
> > "in band" audio tones.
> > By default the sample IVR only works on "out of band" DTMF events.
> >
> > This is why when you call directly from another skype client the 'push
> > button' events are detected because they are passed as "out of band"
> > signaling.
> >
> >
> > Now,
> > In the dial plan you can tell FS to listen for "in band" audio DTMF
> > tones using <action application="start_dtmf" />
> >
> > However,
> > This currently does not work during a skype call for some reason?
> > http://jira.freeswitch.org/browse/MODSKYPIAX-66
> >
> >
> > A work around,
> > is to sign up for the "Skype SIP Beta" product where the 'push button'
> > events are sent to FS "out of band".
> >
> > This conversion is done at the PSTN --> Skype gateway by dedicated DTMF
> > tone detection hardware.
> >
> >
> > Skype has either made a business decision, or a technical over sight to
> > pass DTMF events 'out of band' for only addition fee products.
> >
> >
> > It has also been reported in New Zealand that even the 'in band' tones
> > where present one day and actually filtered out the next.
> > This seems extreme, but either through deliberate action or a technology
> > change this is what was reported on one blog.
> >
> > It is unclear to me if this was a technical limitation or a blunt
> > business decisions, but a audio sample showed the audio tones missing?
> >
> >
> > In any case you would not want to rely on 'In band' DTMF' tones when
> > passed through 'lossy' codecs anyway.
> >
> > Best to stick with 'out of band' signaling for reliability.
> >
> >
> > regards,
> > Scott Torr
> >
> >
> >
> >
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100218/ee00b5f2/attachment.html 

From mike at jerris.com  Thu Feb 18 08:02:33 2010
From: mike at jerris.com (Michael Jerris)
Date: Thu, 18 Feb 2010 11:02:33 -0500
Subject: [Freeswitch-users] Sending message notifications with openzap
In-Reply-To: <0ae2b96926e5340a50e659891efb26cc@thom.fr.eu.org>
References: <e749d5a76cc31b4a3bb2459aaf8b78d9@thom.fr.eu.org>
	<1B676793-9D83-4C8D-A9C4-78891F8C70B5@jerris.com>
	<0ae2b96926e5340a50e659891efb26cc@thom.fr.eu.org>
Message-ID: <738F6BDF-8A79-4918-A125-C6CE9445C57B@jerris.com>

its 2 way, an event is sent out to request mwi, and another in hte other direction (that mod_voicemail sends) that triggers the notify in sip.  Looking in mod_sofia where we handle the registration request, you will see there where we send the request, and look in mod_voicemail, or the event handler in sofia for the other direction.

Mike
On Feb 18, 2010, at 4:50 AM, Fran?ois Legal wrote:

> Thanks,
> 
>  
> yes I could see that this was handled with events.
> 
>  
> Could you please give more details (that does not seem obvious to me while looking at sofia_presence.c)
> 
> When and why is the event triggered ?
> 
> What information do I get with the event ?
> 
>  
> Then I have a design question, is it mandatory to (as this is done in mod_sofia but I guess for a lot of reasons) process the event in a separate thread ?
> 
>  
> Thanks for the information (and sorry if my questions are not relevant, I don't know the event system at all).
> 
>  
> Fran?ois
> 
>  
> On Thu, 18 Feb 2010 00:13:08 -0500, Michael Jerris <mike at jerris.com> wrote:
> 
> This all uses the event system.  Take a look at how we handle the events in mod_sofia, it should be pretty easy to do the same in mod_openzap.
> Mike
> 
> On Feb 16, 2010, at 4:33 AM, Fran?ois Legal wrote:
> I was wondering whether or not a facility was available to send message waiting indicator (from mpd_voicemail) using openzap.
> 
> I know this feature is not available in openzap (I'm in the process of coding it for analog channels) but as message indication can be sent via mod_sofia, I wonder if a frame is available in FS core.
> 
>  
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100218/71ead5f9/attachment-0001.html 

From anthony.minessale at gmail.com  Thu Feb 18 08:03:14 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 18 Feb 2010 10:03:14 -0600
Subject: [Freeswitch-users] Forbidden using UDP, works with TCP/TLS
In-Reply-To: <7EA7C626-D317-4011-BF6E-A7411C765EC1@freeswitch.org>
References: <33c87fa31002101824h21e3bf7lb13b68e8d4fd36a2@mail.gmail.com>
	<33c87fa31002131336j7b05db7aof6ba6e3439aebc41@mail.gmail.com>
	<9D702362-BB5F-4C0F-9EE7-DABA3B926C36@freeswitch.org>
	<33c87fa31002141708s72cb7246h5372ecea40be615d@mail.gmail.com>
	<1888B1CA-D0A8-407C-9C44-B86E0F1925D6@freeswitch.org>
	<33c87fa31002142017v49d8e7c5nf572d086ec89282b@mail.gmail.com>
	<33c87fa31002142109l5fcabed3ydba21b865ed6cbbb@mail.gmail.com>
	<147993F8-9C9E-4D96-9AFE-5EEE4E8F93F0@freeswitch.org>
	<33c87fa31002180022g2e91c8efs65f52bb4275cb59@mail.gmail.com>
	<7EA7C626-D317-4011-BF6E-A7411C765EC1@freeswitch.org>
Message-ID: <191c3a031002180803u5d862c9blf181d90988c30415@mail.gmail.com>

everyone chip in and send him a new phone so this thread can go away!

On Thu, Feb 18, 2010 at 7:04 AM, Brian West <brian at freeswitch.org> wrote:

> I think your device is broken.
>
> /b
>
> On Feb 18, 2010, at 2:22 AM, Mark Campbell-Smith wrote:
>
> > Thanks for looking at that Brian.  If the token is not used, I assume
> > this is not the reason for FS rejecting the Registration attempt?
> >
> > Also, when is stale=true set in the WWW-Authentication?  I notice that
> > for this device, I do not see stale=true, but for all my other
> > devices, I see stale=true (at least from the logs I've taken today).
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100218/3869949c/attachment.html 

From anthony.minessale at gmail.com  Thu Feb 18 08:05:18 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 18 Feb 2010 10:05:18 -0600
Subject: [Freeswitch-users] Other party does not hear voice when
	connecting with X-Lite
In-Reply-To: <a858e7691002180452n6ef91ed7k3d9b2f7246d773f1@mail.gmail.com>
References: <3e6d7b0c1002150050g3ccd23aas2e7f05f230bff4a7@mail.gmail.com>
	<a858e7691002180452n6ef91ed7k3d9b2f7246d773f1@mail.gmail.com>
Message-ID: <191c3a031002180805m512655a7s3543d3a2744453bc@mail.gmail.com>

maybe you need x-heavy aka eyebeam that has more options.
Counterpath likes to hold required features in SIP hostage for money for
some reason.


On Thu, Feb 18, 2010 at 6:52 AM, Vitalii Colosov <vetali100 at gmail.com>wrote:

> Thanks a lot for your advices and sorry for confusion, my description was
> not consistent, agree...
>
> We are testing X-Lite client from 4 different machines and using 2
> different FS servers and getting different results - sometimes everything
> works, sometimes only signalling works - no voice, and sometimes even
> signalling does not work.
> YATE client always works without any such issues...
>
> When we will come to some sort of understanding, I will try to share the
> results.
> If we will stuck, I will try to get some debug information and continue
> bothering the professionals. :)
>
>  BTW, I tried to change the codecs in X-Lite, same result.
> It is for sure related to our network configuration + X-Lite's way of
> sending SIP data, looks like...
>
> Regards,
> Vitalii
>
>
>
>
>
> 2010/2/15 Steven Brown <steve at justfone.com>
>
>> I had the same problem with XLite / Freeswitch  a while back that I never
>> fully understood,  however the problem vanished when I disabled all codecs
>> on Xlite except G711 uLaw, as I say, no idea what was going on but this
>> might be worth trying.
>>
>> Steve
>>
>>
>> Message: 1
>> Date: Sun, 14 Feb 2010 09:04:09 -0600
>> From: Anthony Minessale <anthony.minessale at gmail.com>
>> Subject: Re: [Freeswitch-users] Other party does not hear voice when
>>        connecting with X-Lite
>> To: freeswitch-users at lists.freeswitch.org
>> Message-ID:
>>        <191c3a031002140704g705bfc73rd8dd103f3d846062 at mail.gmail.com
>> >
>> Content-Type: text/plain; charset="iso-8859-1"
>>
>>
>> You need to describe this again its too confusing now.
>> List each device, freeswitch, the phones and which ip and combo of addrs
>> it
>> uses with the topology clearly stated.
>> Your attempt to simplify your explanation is actually making it harder to
>> follow.
>> Also consider a debug/sip trace as well.
>>
>> Include
>> sofia status profile default.
>>
>> Then capture a test call after entering these commands.
>>
>> console loglevel debug.
>> sofa profile internal siptrace on
>>
>> On Feb 14, 2010 4:12 AM, "Vitalii Colosov" <vetali100 at gmail.com> wrote:
>>
>> No, it is done on the different PCs...
>>
>> Sorry, when I started the topic, I have described the problem how it is
>> visible from PC of my friend.
>> Then I tried to reproduce the same on my own PC, and you are right...I was
>> not able to hear anything as well, not only both party wasn't.
>> Also, from my PC I was NOT able to hear guitar on test number "9999".
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100218/884c3eec/attachment.html 

From frank at carmickle.com  Thu Feb 18 08:17:41 2010
From: frank at carmickle.com (Frank Carmickle)
Date: Thu, 18 Feb 2010 11:17:41 -0500
Subject: [Freeswitch-users] building for Lenny
In-Reply-To: <a9f4977a70b3fcf1b30413ea562ca962.squirrel@www.webcontracts.co.uk>
References: <3c5cf5261002180046vb0015e7t787599c9903aedc@mail.gmail.com>
	<33c87fa31002180056g51ef8496qa9a35746d0fe75ec@mail.gmail.com>
	<3c5cf5261002180110he0ae815uc713f77f289ac625@mail.gmail.com>
	<a9f4977a70b3fcf1b30413ea562ca962.squirrel@www.webcontracts.co.uk>
Message-ID: <20100218161741.GC4236@base.carmickle.com>

On Thu, Feb 18, Matthew Law wrote:
> 
> On Thu, February 18, 2010 9:10 am, Brian May wrote:
> > On 18 February 2010 19:56, Mark Campbell-Smith <mcampbellsmith at gmail.com>
> > wrote:
> >> I think you need to download the gzip file from
> >> http://files.freeswitch.org/
> >>
> >> latest.freeswitch.org does not contain sound files as far as I'm aware
> >> ..
> >
> > Does this mean I shouldn't be trying to use the debian packages? *.deb
> > files would certainly make it easier to compile it on a fast computer
> > and then transfer to my net5501.
> >
> > Unfortunately, the website seems have a lot of old information,
> > including references to obsolete Ubuntu Hardy packages.

Yes, the wiki is quite out of date.  There are a few of us working on getting an apt repository for freeswitch packages.  Sorry it's taking so long.  You can build the debs from what's in tree now.  The only bit is that sounds are not included.  You will have to get them from files.freeswitch.org.  Once we have an apt repo all of this will become much less painless..

> >
> > I see a thread from late last year that suggests there should be
> > prebuilt packages, everything I can find so far seems very old
> > however.
> 
> Brian,
> 
> I had similar problems recently and decided to compile it from svn trunk. 
> The dependencies for the build are all available in apt, so I installed
> those, checked out trunk and configured it with a prefix of
> /usr/local/freeswitch.

Like I said you can and should build debs from svn.  As far as I see it there is no reason to not build debs.

> 
> I got the impression it is self contained, so you should be able to tar up
> the entire /usr/local/freeswitch dir and scp it over - I stand to be
> corrected on that, though.

That's true but then you lose the convenience of doing upgrades with the package management.

--FC


From infos at madovsky.org  Thu Feb 18 08:36:48 2010
From: infos at madovsky.org (Madovsky)
Date: Thu, 18 Feb 2010 11:36:48 -0500
Subject: [Freeswitch-users] register sipphone to an outside network
	registrar within freeswitch
References: <33c87fa31002101824h21e3bf7lb13b68e8d4fd36a2@mail.gmail.com><33c87fa31002130343k7a01e9abt9c33360399f246f8@mail.gmail.com><C37AB6F3-F712-41B6-9D6D-7A8AE3C6CABB@freeswitch.org><33c87fa31002131336j7b05db7aof6ba6e3439aebc41@mail.gmail.com><9D702362-BB5F-4C0F-9EE7-DABA3B926C36@freeswitch.org><33c87fa31002141708s72cb7246h5372ecea40be615d@mail.gmail.com><1888B1CA-D0A8-407C-9C44-B86E0F1925D6@freeswitch.org><33c87fa31002142017v49d8e7c5nf572d086ec89282b@mail.gmail.com><33c87fa31002142109l5fcabed3ydba21b865ed6cbbb@mail.gmail.com><147993F8-9C9E-4D96-9AFE-5EEE4E8F93F0@freeswitch.org><33c87fa31002180022g2e91c8efs65f52bb4275cb59@mail.gmail.com>
	<7EA7C626-D317-4011-BF6E-A7411C765EC1@freeswitch.org>
Message-ID: <55649042222E4C6481C086FC4D9B2AF0@MOBILEE1705>

Hi,

I googled hours without real success.
Is it possible to use a softphone in a local network and login
into a registrar outside it within freeswitch as proxy and call
and receive calls as normal ?
I'd like to use this config because I need to check codec transcoding.

Thank you

Franck


From brian at freeswitch.org  Thu Feb 18 08:44:16 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 18 Feb 2010 10:44:16 -0600
Subject: [Freeswitch-users] register sipphone to an outside network
	registrar within freeswitch
In-Reply-To: <55649042222E4C6481C086FC4D9B2AF0@MOBILEE1705>
References: <33c87fa31002101824h21e3bf7lb13b68e8d4fd36a2@mail.gmail.com><33c87fa31002130343k7a01e9abt9c33360399f246f8@mail.gmail.com><C37AB6F3-F712-41B6-9D6D-7A8AE3C6CABB@freeswitch.org><33c87fa31002131336j7b05db7aof6ba6e3439aebc41@mail.gmail.com><9D702362-BB5F-4C0F-9EE7-DABA3B926C36@freeswitch.org><33c87fa31002141708s72cb7246h5372ecea40be615d@mail.gmail.com><1888B1CA-D0A8-407C-9C44-B86E0F1925D6@freeswitch.org><33c87fa31002142017v49d8e7c5nf572d086ec89282b@mail.gmail.com><33c87fa31002142109l5fcabed3ydba21b865ed6cbbb@mail.gmail.com><147993F8-9C9E-4D96-9AFE-5EEE4E8F93F0@freeswitch.org><33c87fa31002180022g2e91c8efs65f52bb4275cb59@mail.gmail.com>
	<7EA7C626-D317-4011-BF6E-A7411C765EC1@freeswitch.org>
	<55649042222E4C6481C086FC4D9B2AF0@MOBILEE1705>
Message-ID: <7D69E8BC-3875-48AA-8CB5-70B7C207D845@freeswitch.org>

I have to say this first.... Please DO NOT hijack threads.  Click "new message", type the address and then your message and then press send.  What you did is click reply... change the subject and then replace the body. That is how you hijack a thread.

(the archive also lists them hijacked too.)

Yes its possible.  In fact the default config works like this.

/b

On Feb 18, 2010, at 10:36 AM, Madovsky wrote:

> Hi,
> 
> I googled hours without real success.
> Is it possible to use a softphone in a local network and login
> into a registrar outside it within freeswitch as proxy and call
> and receive calls as normal ?
> I'd like to use this config because I need to check codec transcoding.
> 
> Thank you
> 
> Franck



From infos at madovsky.org  Thu Feb 18 08:55:30 2010
From: infos at madovsky.org (Madovsky)
Date: Thu, 18 Feb 2010 11:55:30 -0500
Subject: [Freeswitch-users] register sipphone to an outside
	networkregistrar within freeswitch
References: <33c87fa31002101824h21e3bf7lb13b68e8d4fd36a2@mail.gmail.com><33c87fa31002130343k7a01e9abt9c33360399f246f8@mail.gmail.com><C37AB6F3-F712-41B6-9D6D-7A8AE3C6CABB@freeswitch.org><33c87fa31002131336j7b05db7aof6ba6e3439aebc41@mail.gmail.com><9D702362-BB5F-4C0F-9EE7-DABA3B926C36@freeswitch.org><33c87fa31002141708s72cb7246h5372ecea40be615d@mail.gmail.com><1888B1CA-D0A8-407C-9C44-B86E0F1925D6@freeswitch.org><33c87fa31002142017v49d8e7c5nf572d086ec89282b@mail.gmail.com><33c87fa31002142109l5fcabed3ydba21b865ed6cbbb@mail.gmail.com><147993F8-9C9E-4D96-9AFE-5EEE4E8F93F0@freeswitch.org><33c87fa31002180022g2e91c8efs65f52bb4275cb59@mail.gmail.com><7EA7C626-D317-4011-BF6E-A7411C765EC1@freeswitch.org><55649042222E4C6481C086FC4D9B2AF0@MOBILEE1705>
	<7D69E8BC-3875-48AA-8CB5-70B7C207D845@freeswitch.org>
Message-ID: <74FC05481EED41E2BB1737C8E2D931EC@MOBILEE1705>


----- Original Message ----- 
From: "Brian West" <brian at freeswitch.org>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Thursday, February 18, 2010 11:44 AM
Subject: Re: [Freeswitch-users] register sipphone to an outside 
networkregistrar within freeswitch


>I have to say this first.... Please DO NOT hijack threads.  Click "new 
>message", type the address and then your message and then press send.  What 
>you did is click reply... change the subject and then replace the body. 
>That is how you hijack a thread.
>
> (the archive also lists them hijacked too.)
>
> Yes its possible.  In fact the default config works like this.
>
> /b
>
> On Feb 18, 2010, at 10:36 AM, Madovsky wrote:
>
>> Hi,
>>
>> I googled hours without real success.
>> Is it possible to use a softphone in a local network and login
>> into a registrar outside it within freeswitch as proxy and call
>> and receive calls as normal ?
>> I'd like to use this config because I need to check codec transcoding.
>>
>> Thank you
>>
>> Franck
>
>

I know sorry I forgot to remove RE:




From brian at freeswitch.org  Thu Feb 18 09:01:02 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 18 Feb 2010 11:01:02 -0600
Subject: [Freeswitch-users] register sipphone to an outside
	networkregistrar within freeswitch
In-Reply-To: <74FC05481EED41E2BB1737C8E2D931EC@MOBILEE1705>
References: <33c87fa31002101824h21e3bf7lb13b68e8d4fd36a2@mail.gmail.com><33c87fa31002130343k7a01e9abt9c33360399f246f8@mail.gmail.com><C37AB6F3-F712-41B6-9D6D-7A8AE3C6CABB@freeswitch.org><33c87fa31002131336j7b05db7aof6ba6e3439aebc41@mail.gmail.com><9D702362-BB5F-4C0F-9EE7-DABA3B926C36@freeswitch.org><33c87fa31002141708s72cb7246h5372ecea40be615d@mail.gmail.com><1888B1CA-D0A8-407C-9C44-B86E0F1925D6@freeswitch.org><33c87fa31002142017v49d8e7c5nf572d086ec89282b@mail.gmail.com><33c87fa31002142109l5fcabed3ydba21b865ed6cbbb@mail.gmail.com><147993F8-9C9E-4D96-9AFE-5EEE4E8F93F0@freeswitch.org><33c87fa31002180022g2e91c8efs65f52bb4275cb59@mail.gmail.com><7EA7C626-D317-4011-BF6E-A7411C765EC1@freeswitch.org><55649042222E4C6481C086FC4D9B2AF0@MOBILEE1705>
	<7D69E8BC-3875-48AA-8CB5-70B7C207D845@freeswitch.org>
	<74FC05481EED41E2BB1737C8E2D931EC@MOBILEE1705>
Message-ID: <D4F504FD-DCBA-402C-8A25-72595CB4459D@freeswitch.org>

No that still hijacks the thread :P  YOU MUST click new message.  Because their are headers that are reflected back to the list server.

/b

On Feb 18, 2010, at 10:55 AM, Madovsky wrote:

> I know sorry I forgot to remove RE:



From brian at freeswitch.org  Thu Feb 18 09:23:42 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 18 Feb 2010 11:23:42 -0600
Subject: [Freeswitch-users] Testers
Message-ID: <8BDBC6EB-00D4-48B7-9234-1067E01DEE3D@freeswitch.org>

Here is what I need:

Testers for our G729 installer for linux 32bit and 64bit installs.

I have tested CentOS 5x, Debian (excluding 4.0 unsupported)

Please contact me off list so I can issue you a temp. license and a make sure it all works.

Thanks,
Brian



From infos at madovsky.org  Thu Feb 18 09:34:19 2010
From: infos at madovsky.org (Madovsky)
Date: Thu, 18 Feb 2010 12:34:19 -0500
Subject: [Freeswitch-users] freeswitch config expert wanted
Message-ID: <3E855EBF921248F28F231F1CDCF9B8E0@MOBILEE1705>

Hi again,

as I'm a researcher and work on a (personal) project since 5 years now,
I decided to add a "light" VOIP functionality to my project, but
after 3 weeks of learning, I decided to stop protocols headdick and waste of my time
(as I never wanted to be in the VOIP dev world), it's definetly not my cup of tea.
So, if anybody in this emailist wants to help me (I pay of course) to configure freeeswitch
as I'm expecting for (I'm sure my config request is really easy but I have not the right informations and example until now)
it would be great, and maybe I will take my retirement more early... ;)

Thanks

Franck Chionna
infos

at
madovsky
dot


org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100218/ffd485b4/attachment.html 

From devel at thom.fr.eu.org  Thu Feb 18 09:36:31 2010
From: devel at thom.fr.eu.org (=?UTF-8?Q?Fran=C3=A7ois_Legal?=)
Date: Thu, 18 Feb 2010 18:36:31 +0100
Subject: [Freeswitch-users] Sending message notifications with openzap
In-Reply-To: <738F6BDF-8A79-4918-A125-C6CE9445C57B@jerris.com>
References: <e749d5a76cc31b4a3bb2459aaf8b78d9@thom.fr.eu.org>
	<1B676793-9D83-4C8D-A9C4-78891F8C70B5@jerris.com>
	<0ae2b96926e5340a50e659891efb26cc@thom.fr.eu.org>
	<738F6BDF-8A79-4918-A125-C6CE9445C57B@jerris.com>
Message-ID: <68be1f4017938f41a48f7a619a3e667b@thom.fr.eu.org>


I not sure to understand this process, so please correct me if I'm
wrong.


A module, to receive the event notification, does not only have to
bind
to the event (with switch_event_bind or switch_event_bind_removable)
but
also have to send the event (using switch_event_create
and
switch_event_fire) for registration ? 

What is the
variable
mod_sofia_globals.mwi_node used for ? Is it used to queue the
events until
they are processed ? If yes, is it mandatory to queue the
events in openzap
? 

As a side question, I was wondering how to associate
a mailbox (and/or
a user) with an openzap channel. I guess I need that kind
of association to
bring MWI to openzap. I thought I could use the
MWI-Account (something like
MWI-Account = openzap/x/y). Would it be a
correct way to go ? 

Thanks

Fran?ois 

On Thu, 18 Feb 2010 11:02:33
-0500, Michael Jerris  wrote: its 2 way, an event is sent out to request
mwi, and another in hte other direction (that mod_voicemail sends) that
triggers the notify in sip. Looking in mod_sofia where we handle the
registration request, you will see there where we send the request, and
look in mod_voicemail, or the event handler in sofia for the other
direction.  Mike
  On Feb 18, 2010, at 4:50 AM, Fran?ois Legal wrote:


Thanks,  

yes I could see that this was handled with events.  

Could
you please give more details (that does not seem obvious to me while
looking at sofia_presence.c) 

When and why is the event triggered ? 

What
information do I get with the event ?  

Then I have a design question, is
it mandatory to (as this is done in mod_sofia but I guess for a lot of
reasons) process the event in a separate thread ?  

Thanks for the
information (and sorry if my questions are not relevant, I don't know the
event system at all).  

Fran?ois  

On Thu, 18 Feb 2010 00:13:08 -0500,
Michael Jerris  wrote: This all uses the event system. Take a look at how
we handle the events in mod_sofia, it should be pretty easy to do the same
in mod_openzap. Mike 
  On Feb 16, 2010, at 4:33 AM, Fran?ois Legal wrote: 


I was wondering whether or not a facility was available to send message
waiting indicator (from mpd_voicemail) using openzap. 

I know this feature
is not available in openzap (I'm in the process of coding it for analog
channels) but as message indication can be sent via mod_sofia, I wonder if
a frame is available in FS core.      

_______________________________________________
FreeSWITCH-users mailing
list
FreeSWITCH-users at lists.freeswitch.org
[2]
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




Links:
------
[1] mailto:mike at jerris.com
[2]
mailto:FreeSWITCH-users at lists.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100218/651a60fd/attachment.html 

From msc at freeswitch.org  Thu Feb 18 09:43:37 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 18 Feb 2010 09:43:37 -0800
Subject: [Freeswitch-users] tone_detect timeout
In-Reply-To: <EC840323-2A76-426E-A386-8DA111195768@tlainvestments.com>
References: <EC840323-2A76-426E-A386-8DA111195768@tlainvestments.com>
Message-ID: <87f2f3b91002180943u5ef06108rcec1f8d4665783fe@mail.gmail.com>

On Wed, Feb 17, 2010 at 4:44 PM, Troy Anderson <troy at tlainvestments.com>wrote:

> In the wiki about tone_detect, the docs state that the timeout is in
> seconds (e.g. +2 for 2 seconds from now), but all the examples have +5000,
> suggesting that it may really be milliseconds.  I'd be happy to update the
> wiki if someone could say if it's seconds or millisconds or otherwise.
>
> I tried tracing it back in the code, but got lost looking for the
> definition of switch_media_bug_t!
>
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_tone_detect
>
> Thanks!
> Troy
>

Troy,

Good catch. It is definitely milliseconds. Please update the wiki. You get a
gold star for taking the initiative. BTW, it's okay to make the change if
you're reasonably certain it's correct and then update us here. It's easy to
undo a wiki edit.

-MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100218/b27b5cfa/attachment.html 

From frank at carmickle.com  Thu Feb 18 09:49:05 2010
From: frank at carmickle.com (Frank Carmickle)
Date: Thu, 18 Feb 2010 12:49:05 -0500
Subject: [Freeswitch-users] freeswitch config expert wanted
In-Reply-To: <3E855EBF921248F28F231F1CDCF9B8E0@MOBILEE1705>
References: <3E855EBF921248F28F231F1CDCF9B8E0@MOBILEE1705>
Message-ID: <20100218174905.GD4236@base.carmickle.com>

On Thu, Feb 18, Madovsky wrote:
> Hi again,
> 
> as I'm a researcher and work on a (personal) project since 5 years now,
> I decided to add a "light" VOIP functionality to my project, but
> after 3 weeks of learning, I decided to stop protocols headdick and waste of my time
> (as I never wanted to be in the VOIP dev world), it's definetly not my cup of tea.
> So, if anybody in this emailist wants to help me (I pay of course) to configure freeeswitch
> as I'm expecting for (I'm sure my config request is really easy but I have not the right informations and example until now)
> it would be great, and maybe I will take my retirement more early... ;)

What is it that you require?  If it would help you to speak on the phone my number is +1 (315) 703-1608.  Feel free to call at any time.  If I do not answer please leave a message.

Regards
--Frank


From msc at freeswitch.org  Thu Feb 18 09:58:15 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 18 Feb 2010 09:58:15 -0800
Subject: [Freeswitch-users] freeswitch config expert wanted
In-Reply-To: <20100218174905.GD4236@base.carmickle.com>
References: <3E855EBF921248F28F231F1CDCF9B8E0@MOBILEE1705>
	<20100218174905.GD4236@base.carmickle.com>
Message-ID: <87f2f3b91002180958w4a3c71ddt789052f45ad95af2@mail.gmail.com>

On Thu, Feb 18, 2010 at 9:49 AM, Frank Carmickle <frank at carmickle.com>wrote:

> On Thu, Feb 18, Madovsky wrote:
> > Hi again,
> >
> > as I'm a researcher and work on a (personal) project since 5 years now,
> > I decided to add a "light" VOIP functionality to my project, but
> > after 3 weeks of learning, I decided to stop protocols headdick and waste
> of my time
> > (as I never wanted to be in the VOIP dev world), it's definetly not my
> cup of tea.
> > So, if anybody in this emailist wants to help me (I pay of course) to
> configure freeeswitch
> > as I'm expecting for (I'm sure my config request is really easy but I
> have not the right informations and example until now)
> > it would be great, and maybe I will take my retirement more early... ;)
>
> What is it that you require?  If it would help you to speak on the phone my
> number is +1 (315) 703-1608.  Feel free to call at any time.  If I do not
> answer please leave a message.
>
> Regards
> --Frank
>

Additionally, we have a community conference call tomorrow, so if you would
like to join us and talk about your project that would be okay. We have an
official agenda that we discuss:

http://wiki.freeswitch.org/wiki/FS_weekly_2010_02_19

And after the agenda people take turns asking questions and talking about
subjects of interest.
-MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100218/1809e2f7/attachment-0001.html 

From troy at tlainvestments.com  Thu Feb 18 10:07:11 2010
From: troy at tlainvestments.com (Troy Anderson)
Date: Thu, 18 Feb 2010 11:07:11 -0700
Subject: [Freeswitch-users] tone_detect timeout
In-Reply-To: <87f2f3b91002180943u5ef06108rcec1f8d4665783fe@mail.gmail.com>
References: <EC840323-2A76-426E-A386-8DA111195768@tlainvestments.com>
	<87f2f3b91002180943u5ef06108rcec1f8d4665783fe@mail.gmail.com>
Message-ID: <758D6E07-90B3-4117-97AD-3CA899B4CAE5@tlainvestments.com>

Changed to indicate milliseconds for relative and seconds for the absolute option as I assume the absolute option is for a unix timestamp which is, indeed, seconds.

-Troy


On Feb 18, 2010, at 10:43 AM, Michael Collins wrote:

> 
> 
> On Wed, Feb 17, 2010 at 4:44 PM, Troy Anderson <troy at tlainvestments.com> wrote:
> In the wiki about tone_detect, the docs state that the timeout is in seconds (e.g. +2 for 2 seconds from now), but all the examples have +5000, suggesting that it may really be milliseconds.  I'd be happy to update the wiki if someone could say if it's seconds or millisconds or otherwise.
> 
> I tried tracing it back in the code, but got lost looking for the definition of switch_media_bug_t!
> 
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_tone_detect
> 
> Thanks!
> Troy
> 
> Troy,
> 
> Good catch. It is definitely milliseconds. Please update the wiki. You get a gold star for taking the initiative. BTW, it's okay to make the change if you're reasonably certain it's correct and then update us here. It's easy to undo a wiki edit.
> 
> -MC
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100218/60395ef5/attachment.html 

From msc at freeswitch.org  Thu Feb 18 10:13:00 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 18 Feb 2010 10:13:00 -0800
Subject: [Freeswitch-users] tone_detect timeout
In-Reply-To: <758D6E07-90B3-4117-97AD-3CA899B4CAE5@tlainvestments.com>
References: <EC840323-2A76-426E-A386-8DA111195768@tlainvestments.com>
	<87f2f3b91002180943u5ef06108rcec1f8d4665783fe@mail.gmail.com>
	<758D6E07-90B3-4117-97AD-3CA899B4CAE5@tlainvestments.com>
Message-ID: <87f2f3b91002181013w7bec493cm37946405ec46ee5f@mail.gmail.com>

On Thu, Feb 18, 2010 at 10:07 AM, Troy Anderson <troy at tlainvestments.com>wrote:

> Changed to indicate milliseconds for relative and seconds for the absolute
> option as I assume the absolute option is for a unix timestamp which is,
> indeed, seconds.
>
> -Troy
>
> Another gold star 4 U! Thanks for pitching in. We really appreciate it when
folks lend a hand documenting the stuff they know. Keep up the good work.
-MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100218/fc015664/attachment.html 

From max.bridgewater at gmail.com  Thu Feb 18 10:26:37 2010
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Thu, 18 Feb 2010 10:26:37 -0800
Subject: [Freeswitch-users] Skypiax snd-dummy, One way audio
Message-ID: <c55d46951002181026n49b1dbcl8cfa868f902a8a0d@mail.gmail.com>

Hi Skypiax Lovers,

I'm trying to get Skypiax running on CentOS5.3 but I keep having
problems with snd-dummy. My problem is that I have sound only in one
direction (inbound). I suspect the sound devices are not set properly.
In particular, I can't find "hw:dummy"  in the Skype options (Static
Build 2.1).

Here are the sound devices I see in Skype options:
a) Dummy, Dummy, PCM Default Audio 0default:CARD=Dummy)
b) HDA Intel, ALC662 Analog (hw:0,0)
c) Dummy, Dummy PCM (hw:1,0)
d) hdmi (unlnown)

Here is what i notice:

1) modprobe snd-dummy returns nothing; implying to me that the module
was loaded properly.
2) lsmod | grep "snd" returns:

snd_dummy              15553  0
snd_hda_intel         343537  0
snd_hwdep              12869  1 snd_hda_intel
snd_seq_dummy           7877  0
snd_seq_oss            32577  0
snd_seq_midi_event     11073  1 snd_seq_oss
snd_seq                49585  5 snd_seq_dummy,snd_seq_oss,snd_seq_midi_event
snd_seq_device         11725  3 snd_seq_dummy,snd_seq_oss,snd_seq
snd_pcm_oss            42817  0
snd_mixer_oss          19009  1 snd_pcm_oss
snd_pcm                72133  3 snd_dummy,snd_hda_intel,snd_pcm_oss
snd_timer              24517  2 snd_seq,snd_pcm
snd                    55237  10
snd_dummy,snd_hda_intel,snd_hwdep,snd_seq_oss,snd_seq,snd_seq_device,snd_pcm_oss,snd_mixer_oss,snd_pcm,snd_timer
soundcore              11553  1 snd
snd_page_alloc         14281  2 snd_hda_intel,snd_pcm

Any idea?
Thanks,
Max.


From gmaruzz at celliax.org  Thu Feb 18 10:52:02 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Thu, 18 Feb 2010 19:52:02 +0100
Subject: [Freeswitch-users] Skypiax snd-dummy, One way audio
In-Reply-To: <c55d46951002181026n49b1dbcl8cfa868f902a8a0d@mail.gmail.com>
References: <c55d46951002181026n49b1dbcl8cfa868f902a8a0d@mail.gmail.com>
Message-ID: <7b197bef1002181052x37a01b75q17624efb6fc5fca4@mail.gmail.com>

Hi Max,

on centos 5.3 is working well, but it will consume much cpu time. If
you are planning to use few skype channels, then ok.
If you want to "scale", you must use a kernel at 100HZ, there is a
.config file in the mod_skypiax sources in the subdirectory kernel,
and instructions in the wiki page.

You can keep your same centos running, just with the new compiled kernel.

That said, for few channels will be ok.

You have to use the last static build of beta skype (.2.1.0.81).

In your case you will use the:
c) Dummy, Dummy PCM (hw:1,0)
device.

check with dmesg if the module is running properly, it will add a line
were it tells on how many HZ is running. If there is not that line,
the custom snd-dummy is not loaded.

Also, you can test with
aplay -l
if it show you a soundcard with 127 subdevices, you're ok.

Please svn update mod_skypiax directory 'cause last change was few
minutes ago ;).

Let know if you still encounter problems, you can find me in IRC
(irc.freenode.net #freeswitch) as gmaruzz, if you need,

ciao for now,

-giovanni


On Thu, Feb 18, 2010 at 7:26 PM, Max Bridgewater
<max.bridgewater at gmail.com> wrote:
> Hi Skypiax Lovers,
>
> I'm trying to get Skypiax running on CentOS5.3 but I keep having
> problems with snd-dummy. My problem is that I have sound only in one
> direction (inbound). I suspect the sound devices are not set properly.
> In particular, I can't find "hw:dummy" ?in the Skype options (Static
> Build 2.1).
>
> Here are the sound devices I see in Skype options:
> a) Dummy, Dummy, PCM Default Audio 0default:CARD=Dummy)
> b) HDA Intel, ALC662 Analog (hw:0,0)
> c) Dummy, Dummy PCM (hw:1,0)
> d) hdmi (unlnown)
>
> Here is what i notice:
>
> 1) modprobe snd-dummy returns nothing; implying to me that the module
> was loaded properly.
> 2) lsmod | grep "snd" returns:
>
> snd_dummy ? ? ? ? ? ? ?15553 ?0
> snd_hda_intel ? ? ? ? 343537 ?0
> snd_hwdep ? ? ? ? ? ? ?12869 ?1 snd_hda_intel
> snd_seq_dummy ? ? ? ? ? 7877 ?0
> snd_seq_oss ? ? ? ? ? ?32577 ?0
> snd_seq_midi_event ? ? 11073 ?1 snd_seq_oss
> snd_seq ? ? ? ? ? ? ? ?49585 ?5 snd_seq_dummy,snd_seq_oss,snd_seq_midi_event
> snd_seq_device ? ? ? ? 11725 ?3 snd_seq_dummy,snd_seq_oss,snd_seq
> snd_pcm_oss ? ? ? ? ? ?42817 ?0
> snd_mixer_oss ? ? ? ? ?19009 ?1 snd_pcm_oss
> snd_pcm ? ? ? ? ? ? ? ?72133 ?3 snd_dummy,snd_hda_intel,snd_pcm_oss
> snd_timer ? ? ? ? ? ? ?24517 ?2 snd_seq,snd_pcm
> snd ? ? ? ? ? ? ? ? ? ?55237 ?10
> snd_dummy,snd_hda_intel,snd_hwdep,snd_seq_oss,snd_seq,snd_seq_device,snd_pcm_oss,snd_mixer_oss,snd_pcm,snd_timer
> soundcore ? ? ? ? ? ? ?11553 ?1 snd
> snd_page_alloc ? ? ? ? 14281 ?2 snd_hda_intel,snd_pcm
>
> Any idea?
> Thanks,
> Max.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From andrew at hijacked.us  Thu Feb 18 11:01:29 2010
From: andrew at hijacked.us (Andrew Thompson)
Date: Thu, 18 Feb 2010 14:01:29 -0500
Subject: [Freeswitch-users] how to use mod_erlang ?
In-Reply-To: <ecd05ce31002171917w2713f8detcee3f7e36039e9f2@mail.gmail.com>
References: <ecd05ce31002171917w2713f8detcee3f7e36039e9f2@mail.gmail.com>
Message-ID: <20100218190129.GD8518@hijacked.us>

On Thu, Feb 18, 2010 at 11:17:26AM +0800, daqiang wang wrote:
> hello !
> I test mod_erlang from : http://wiki.freeswitch.org/wiki/Mod_erlang_event
> but when I input :
>  >
> 
> {foo, freeswitch at localhost } ! {api, status, ""}.
> I received
>   (test at wangdq-laptop)2>
> =ERROR REPORT==== 18-Feb-2010::11:10:13 ===
> Error in process <0.41.0> on node 'test at wangdq-laptop' with exit
> value: {badarg,[{erlang,list_to_existing_atom,["freeswitch at wangdq-laptop"]},{dist_util,recv_challenge,1},{dist_util,handshake_we_started,1}]}
>

Try {foo, 'freeswitch at wangdq-laptop' } ! {api, status, ""}. instead.

Andrew


From Prometheus001 at gmx.net  Thu Feb 18 11:27:16 2010
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Thu, 18 Feb 2010 20:27:16 +0100
Subject: [Freeswitch-users] more than 1 profile for sofia/user ?
Message-ID: <4B7D9494.8050208@gmx.net>

Hello,

in the standard setup - if a phone is registering to port 5060 - it is
bound to the "internal" profile. And I can dial it via sofia/user/xxxx then.

However due to NAT issues I would like to have to 2 seperate profiles
for SIP phones. For example I have a "local" profile for all devices
inside the LAN (e.g. Pattons und in future: local phones) and another
"internal" profile which allows also external phones via
external-xxx-ip. That way I would like to ensure that local phones have
nothing to do with natted adresses and that external phones can register
via external IPs.

Question How do I manage that I can register a phone to the "local"
profile and being able to dial that phone via sofia/user/xxxxx?

Or do I think too complicated and there is simply nothing special to do?

Best regards
Peter


From anthony.minessale at gmail.com  Thu Feb 18 11:40:28 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 18 Feb 2010 13:40:28 -0600
Subject: [Freeswitch-users] more than 1 profile for sofia/user ?
In-Reply-To: <4B7D9494.8050208@gmx.net>
References: <4B7D9494.8050208@gmx.net>
Message-ID: <191c3a031002181140y44135106yb34e6a6f73a04773@mail.gmail.com>

edit the dial-string for that user in the directory xml to try the extension
on both profile at once

On Thu, Feb 18, 2010 at 1:27 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:

> Hello,
>
> in the standard setup - if a phone is registering to port 5060 - it is
> bound to the "internal" profile. And I can dial it via sofia/user/xxxx
> then.
>
> However due to NAT issues I would like to have to 2 seperate profiles
> for SIP phones. For example I have a "local" profile for all devices
> inside the LAN (e.g. Pattons und in future: local phones) and another
> "internal" profile which allows also external phones via
> external-xxx-ip. That way I would like to ensure that local phones have
> nothing to do with natted adresses and that external phones can register
> via external IPs.
>
> Question How do I manage that I can register a phone to the "local"
> profile and being able to dial that phone via sofia/user/xxxxx?
>
> Or do I think too complicated and there is simply nothing special to do?
>
> Best regards
> Peter
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100218/a7b1f7d9/attachment.html 

From jerry.richards at teotech.com  Thu Feb 18 12:26:28 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Thu, 18 Feb 2010 12:26:28 -0800
Subject: [Freeswitch-users] Presence PUBLISH Not Updating
	AfterSoftphoneOffLine Then Available
In-Reply-To: <191c3a031002162139p57c94046y660fb8d01fd76f46@mail.gmail.com>
References: <31FEE9072F104AE18B26812978FCA341@greyhawk.tonecommander.com><45963EB5-DFE5-43FA-AFCC-857E1E719E2D@jerris.com><191c3a031002091557jfc9b486lb832ea538883bce9@mail.gmail.com><68A4C011A72B4ABA91A185F32480EB99@greyhawk.tonecommander.com>
	<191c3a031002162139p57c94046y660fb8d01fd76f46@mail.gmail.com>
Message-ID: <E69C19EDE6F541198E27BE0883CB773D@greyhawk.tonecommander.com>

Okay, you have made some good suggestions.  I will look into this further on
my end (I think I might know the cause).  If I find it to be an FS bug, I
will open a Jira Issue.  Yes, these are Bria (CounterPath) phones, but these
are phones that I'm using and they are popular, and as far as I know,
faithful to the SIP RFCs, so I think it will make FS more robust.  I haven't
use the IRC much in the past, but I can try to login there when I have
issues in the future.
 
Sorry for my slow response.  I have a lot to do at the moment and sometimes
I must do some SIP research.  By the way, I think Freeswitch is a great
design and you all are doing a great job with this project.
 
Thanks And Best Regards,
Jerry
 


  _____  

From: Anthony Minessale [mailto:anthony.minessale at gmail.com] 
Sent: Tuesday, February 16, 2010 9:40 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Presence PUBLISH Not Updating
AfterSoftphoneOffLine Then Available



You see one case at the top where it sends a notify and more where it doesnt
.  You have the sql stmts right there (they are not errs just logging in red
so they are obvious) run them manually and figure out why there are no
matches.  No subscriptions maybe?

Its beginning to sound like a broken record with so many bria isssues, its a
new software afterall and not free like we are, why must we support it so
much?

Also if you are actually concerned with this issue, maybe you can come back
sooner than once every week or 2 weeks.  We quickly lose track of threads
like this that linger for a month, that's what jira is for....

Maybe you can stop by irc or keep an eye on your email client so we can
confirm what you are doing wrong or if we have an interop with bria, a pay
softphone none of us have a copy of........




On Feb 16, 2010 11:18 AM, "Jerry Richards" <jerry.richards at teotech.com>
wrote:


I got version freeswitch-1.0.5-20100215-0400, built it, and ran it, and I am
seeing the same issue.  That is, once I set the Bria softphone status to
'Appear Offline', FS does not forward presence states until resubscription
time (i.e. tens of minutes later).  I posted a trace at
http://pastebin.freeswitch.org/12164.  At line 359 of the trace, FS is
logging an ERR at sofia_presence.c:662.  Here is the scenario:
 
1) Set Bria softphone presence state to 'Appear Offline'
2) Subscibing softphones reflect offline status

3) Set Bria softphone presence state to 'Available'
4) *** Subscibing softphones do not get status update ***
 

Thanks And Best Regards,
Jerry
 


  _____  

From: Anthony Minessale [mailto:anthony.minessale at gmail.com] 
Sent: Tuesday, February 09, 2010 3:58 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Presence PUBLISH Not Updating AfterSoftphone
OffLine Then Available



> he means update to trunk first then try it again obviously.
>
>
> On Tue, Feb 9, 2010 at 3:10 PM, ...


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org





-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100218/ee19c628/attachment-0001.html 

From mrene_lists at avgs.ca  Thu Feb 18 13:10:53 2010
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Thu, 18 Feb 2010 16:10:53 -0500
Subject: [Freeswitch-users] Establishing a Call from .Net based
	application
In-Reply-To: <6E8D2069C08AA84A83D336E996AE4C67032C9D5BAD@mse17be1.mse17.exchange.ms>
References: <874941.17255.qm@web33502.mail.mud.yahoo.com>
	<df97b9c51001310545n54195cadr9faa3db46917ca9e@mail.gmail.com>
	<367751821002010737o4ca83476j62c96cb60c8a456a@mail.gmail.com>
	<df97b9c51002130752h48aae119ja80797ab8a3f5b2d@mail.gmail.com>
	<6E8D2069C08AA84A83D336E996AE4C67032C9D5346@mse17be1.mse17.exchange.ms>
	<df97b9c51002180012wc4d9421ye6e81ae6199321c8@mail.gmail.com>
	<6E8D2069C08AA84A83D336E996AE4C67032C9D5BAD@mse17be1.mse17.exchange.ms>
Message-ID: <9ABD2E73-F7EB-4B2E-9306-8BBC6718F215@avgs.ca>

parse "sofia xmlstatus"

<profile>
<name>external</name>
<type>profile</type>
<data>sip:mod_sofia at 192.168.0.9:5080</data>
<state>RUNNING (0)</state>
</profile>


Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 18-Feb-10, at 6:12 AM, Michael Giagnocavo wrote:

> I?m not sure what the FreeSWITCH APIs are to figure out what IP  
> Sofia SIP has bound to. Whatever it is, you?d call the same thing in  
> C#. What do you want to do with the API?
>
> mod_managed.dll or .so is the FreeSWITCH native code module that  
> loads the CLR or Mono into the FreeSWITCH process and loads  
> FreeSWITCH.Managed.dll. The managed DLL contains the bulk of the  
> managed-unmanaged interop code (.NET definitions of all the FS C  
> functions).
>
> -Michael
>
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org 
> ] On Behalf Of Scott Fernandez
> Sent: Thursday, February 18, 2010 1:12 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Establishing a Call from .Net based  
> application
>
> Hi Diego & Michael,
>
> Thanks for your reply and support.
>
> However, I have some clarifications required from both of you.
>
> 1. Here is the question for Diego,
>
> Simple Example:
>
> using FreeSWITCH;
> using FreeSWITCH.Native;
>
> namespace BITS.Ivr.Bp.Server
> {
>  public class CIV_BPFSProcess : FreeSWITCH.IAppPlugin
> {
>  public void Run(AppContext context)
>  {
>   //answer call
>   context.Session.Answer();
>   //sleep 2 seconds
>   context.Session.sleep(2000, 1);
>   //hangup call
>   context.Session.Hangup("
> NORMAL_CLEARING");
>  }
>  }
> }
>
> I understand that the concept of your example code. However, would  
> like to know as to how would my .NET C# know the IP address of  
> Freeswitch to talk to it as there is no indication for that?. If not  
> here, where would we need to reference the IP address of FS in .NET  
> code?
>
> I guess the IP address of FS needs to be mentioned in the Target  
> section of the below web.config file in .NET. If I am right, how to  
> specify the IP address over here. If I am wrong, please let me know  
> where do we need to mention the IP address of FS.
>
>     <configuration>
>             <dllmap dll="mod_managed.dll" target="mod_managed.so"/>
>     </configuration>
>
>
> 2. Here is the question for Michael,
>
> You mentioned that "mod_managed.so will be in your freeswitch mod  
> directory". This is very clear and what is mod_managed.dll in  
> my .NET application and the purpose of it?
>
> Thanks for all your help.
>
> Regards,
> Scott.
>
>
>
> On Sun, Feb 14, 2010 at 1:15 AM, Michael Giagnocavo <mgg at giagnocavo.net 
> > wrote:
>
> 2. There is a configuration settings required to Map the "DLL" to  
> ".so" object in CentOS.
> Now, the question is which DLL and .so file to be made available and  
> where?
>
> ?
> If you are experiencing NullReferenceExceptions with any plugin run  
> through the dialplan, make sure you have included the appropriate  
> entry in your dllmap configuration:
>  <dllmap dll="mod_managed.dll" target="mod_managed.so" os="!windows"/>
> ?
>
> mod_managed.so will be in your freeswitch mod directory.
>
>
> All I need is to initiate a call from .NET application and then it  
> should talk to mod_managed module and establish a call. Secondly, I  
> need to know the status of the call such as Ringing, Active, Hangup  
> etc.
>
> To initiate a call, try ManagedSession.Originate.
> -Michael
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100218/43223e9c/attachment.html 

From Prometheus001 at gmx.net  Thu Feb 18 13:14:48 2010
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Thu, 18 Feb 2010 22:14:48 +0100
Subject: [Freeswitch-users] more than 1 profile for sofia/user ?
In-Reply-To: <191c3a031002181140y44135106yb34e6a6f73a04773@mail.gmail.com>
References: <4B7D9494.8050208@gmx.net>
	<191c3a031002181140y44135106yb34e6a6f73a04773@mail.gmail.com>
Message-ID: <4B7DADC8.1060405@gmx.net>

Any idea how to do this?

currently I have
{presence_id=${dialed_user}@${dialed_domain},transfer_fallback_extension=${dialed_user}}${sofia_contact(${dialed_user}@${dialed_domain})}


Best regards
Peter

Anthony Minessale schrieb:
> edit the dial-string for that user in the directory xml to try the
> extension on both profile at once
>
> On Thu, Feb 18, 2010 at 1:27 PM, Peter P GMX <Prometheus001 at gmx.net
> <mailto:Prometheus001 at gmx.net>> wrote:
>
>     Hello,
>
>     in the standard setup - if a phone is registering to port 5060 - it is
>     bound to the "internal" profile. And I can dial it via
>     sofia/user/xxxx then.
>
>     However due to NAT issues I would like to have to 2 seperate profiles
>     for SIP phones. For example I have a "local" profile for all devices
>     inside the LAN (e.g. Pattons und in future: local phones) and another
>     "internal" profile which allows also external phones via
>     external-xxx-ip. That way I would like to ensure that local phones
>     have
>     nothing to do with natted adresses and that external phones can
>     register
>     via external IPs.
>
>     Question How do I manage that I can register a phone to the "local"
>     profile and being able to dial that phone via sofia/user/xxxxx?
>
>     Or do I think too complicated and there is simply nothing special
>     to do?
>
>     Best regards
>     Peter
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> <mailto:MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> <mailto:sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> <http://iax:guest at conference.freeswitch.org/888>
> googletalk:conf+888 at conference.freeswitch.org
> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
> ------------------------------------------------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   


From anthony.minessale at gmail.com  Thu Feb 18 13:53:28 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 18 Feb 2010 15:53:28 -0600
Subject: [Freeswitch-users] more than 1 profile for sofia/user ?
In-Reply-To: <4B7DADC8.1060405@gmx.net>
References: <4B7D9494.8050208@gmx.net>
	<191c3a031002181140y44135106yb34e6a6f73a04773@mail.gmail.com>
	<4B7DADC8.1060405@gmx.net>
Message-ID: <191c3a031002181353r2dbf15c6h206509f9b1399148@mail.gmail.com>

add on a , then another dial string to reflect the other profile too

On Thu, Feb 18, 2010 at 3:14 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:

> Any idea how to do this?
>
> currently I have
> {presence_id=${dialed_user}@
> ${dialed_domain},transfer_fallback_extension=${dialed_user}}${sofia_contact(${dialed_user}@
> ${dialed_domain})}
>
>
> Best regards
> Peter
>
> Anthony Minessale schrieb:
> > edit the dial-string for that user in the directory xml to try the
> > extension on both profile at once
> >
> > On Thu, Feb 18, 2010 at 1:27 PM, Peter P GMX <Prometheus001 at gmx.net
> > <mailto:Prometheus001 at gmx.net>> wrote:
> >
> >     Hello,
> >
> >     in the standard setup - if a phone is registering to port 5060 - it
> is
> >     bound to the "internal" profile. And I can dial it via
> >     sofia/user/xxxx then.
> >
> >     However due to NAT issues I would like to have to 2 seperate profiles
> >     for SIP phones. For example I have a "local" profile for all devices
> >     inside the LAN (e.g. Pattons und in future: local phones) and another
> >     "internal" profile which allows also external phones via
> >     external-xxx-ip. That way I would like to ensure that local phones
> >     have
> >     nothing to do with natted adresses and that external phones can
> >     register
> >     via external IPs.
> >
> >     Question How do I manage that I can register a phone to the "local"
> >     profile and being able to dial that phone via sofia/user/xxxxx?
> >
> >     Or do I think too complicated and there is simply nothing special
> >     to do?
> >
> >     Best regards
> >     Peter
> >
> >     _______________________________________________
> >     FreeSWITCH-users mailing list
> >     FreeSWITCH-users at lists.freeswitch.org
> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>
> >     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >     UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >     http://www.freeswitch.org
> >
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> > Twitter: http://twitter.com/FreeSWITCH_wire
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> > <mailto:MSN%3Aanthony_minessale at hotmail.com<MSN%253Aanthony_minessale at hotmail.com>
> >
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> > <mailto:PAYPAL%3Aanthony.minessale at gmail.com<PAYPAL%253Aanthony.minessale at gmail.com>
> >
> > IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> > <mailto:sip%3A888 at conference.freeswitch.org<sip%253A888 at conference.freeswitch.org>
> >
> > iax:guest at conference.freeswitch.org/888
> > <http://iax:guest at conference.freeswitch.org/888>
> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org<googletalk%253Aconf%252B888 at conference.freeswitch.org>
> >
> > pstn:+19193869900
> > ------------------------------------------------------------------------
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100218/4545016c/attachment-0001.html 

From lloyd.aloysius at gmail.com  Thu Feb 18 14:10:24 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Thu, 18 Feb 2010 17:10:24 -0500
Subject: [Freeswitch-users] IVR greeting - first two words missing
Message-ID: <8a19bf2e1002181410x44b40216j374a30a5675fd162@mail.gmail.com>

Hi All,

I setup a simple IVR. Here is the script.

<menu name="test_ivr"
          greet-long="test/test-ivr.wav"
          greet-short="tset/test-ivr.wav"
          invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"
          exit-sound="voicemail/vm-goodbye.wav"
          timeout ="10000"
                  inter-digit-timeout="2000"
                  max-failures="3">
 <entry action="menu-exec-app" digits="/^(20[0-9])$/" param="transfer $1 XML
default"/>
 <entry action="menu-exec-app" digits="9999" param="transfer 9999 XML
default"/>
</menu>
  </menus>
</configuration>

*Dial Plan*

<action application="answer"/>
<action application="ivr" data="test_ivr"/>

Every time when I reach the IVR . I am getting first one or two words
missing( or may be not clear). How can I fix this issue.

Thanks,
Lloyd
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100218/72232ec7/attachment.html 

From Prometheus001 at gmx.net  Thu Feb 18 14:41:06 2010
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Thu, 18 Feb 2010 23:41:06 +0100
Subject: [Freeswitch-users] more than 1 profile for sofia/user ?
In-Reply-To: <191c3a031002181353r2dbf15c6h206509f9b1399148@mail.gmail.com>
References: <4B7D9494.8050208@gmx.net>	<191c3a031002181140y44135106yb34e6a6f73a04773@mail.gmail.com>	<4B7DADC8.1060405@gmx.net>
	<191c3a031002181353r2dbf15c6h206509f9b1399148@mail.gmail.com>
Message-ID: <4B7DC202.7090409@gmx.net>

Hello Anthony,

>add on a , then another dial string to reflect the other profile too
I really tried to understand this, but
can you give me an example?

Best regards
Peter

Anthony Minessale schrieb:
> add on a , then another dial string to reflect the other profile too
>
> On Thu, Feb 18, 2010 at 3:14 PM, Peter P GMX <Prometheus001 at gmx.net
> <mailto:Prometheus001 at gmx.net>> wrote:
>
>     Any idea how to do this?
>
>     currently I have
>     {presence_id=${dialed_user}@${dialed_domain},transfer_fallback_extension=${dialed_user}}${sofia_contact(${dialed_user}@${dialed_domain})}
>
>
>     Best regards
>     Peter
>
>     Anthony Minessale schrieb:
>     > edit the dial-string for that user in the directory xml to try the
>     > extension on both profile at once
>     >
>     > On Thu, Feb 18, 2010 at 1:27 PM, Peter P GMX
>     <Prometheus001 at gmx.net <mailto:Prometheus001 at gmx.net>
>     > <mailto:Prometheus001 at gmx.net <mailto:Prometheus001 at gmx.net>>>
>     wrote:
>     >
>     >     Hello,
>     >
>     >     in the standard setup - if a phone is registering to port
>     5060 - it is
>     >     bound to the "internal" profile. And I can dial it via
>     >     sofia/user/xxxx then.
>     >
>     >     However due to NAT issues I would like to have to 2 seperate
>     profiles
>     >     for SIP phones. For example I have a "local" profile for all
>     devices
>     >     inside the LAN (e.g. Pattons und in future: local phones)
>     and another
>     >     "internal" profile which allows also external phones via
>     >     external-xxx-ip. That way I would like to ensure that local
>     phones
>     >     have
>     >     nothing to do with natted adresses and that external phones can
>     >     register
>     >     via external IPs.
>     >
>     >     Question How do I manage that I can register a phone to the
>     "local"
>     >     profile and being able to dial that phone via sofia/user/xxxxx?
>     >
>     >     Or do I think too complicated and there is simply nothing
>     special
>     >     to do?
>     >
>     >     Best regards
>     >     Peter
>     >
>     >     _______________________________________________
>     >     FreeSWITCH-users mailing list
>     >     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     >     <mailto:FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>>
>     >     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     >    
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     >     http://www.freeswitch.org
>     >
>     >
>     >
>     >
>     > --
>     > Anthony Minessale II
>     >
>     > FreeSWITCH http://www.freeswitch.org/
>     > ClueCon http://www.cluecon.com/
>     > Twitter: http://twitter.com/FreeSWITCH_wire
>     >
>     > AIM: anthm
>     > MSN:anthony_minessale at hotmail.com
>     <mailto:MSN%3Aanthony_minessale at hotmail.com>
>     > <mailto:MSN%3Aanthony_minessale at hotmail.com
>     <mailto:MSN%253Aanthony_minessale at hotmail.com>>
>     > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>     <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
>     > <mailto:PAYPAL%3Aanthony.minessale at gmail.com
>     <mailto:PAYPAL%253Aanthony.minessale at gmail.com>>
>     > IRC: irc.freenode.net <http://irc.freenode.net>
>     <http://irc.freenode.net> #freeswitch
>     >
>     > FreeSWITCH Developer Conference
>     > sip:888 at conference.freeswitch.org
>     <mailto:sip%3A888 at conference.freeswitch.org>
>     > <mailto:sip%3A888 at conference.freeswitch.org
>     <mailto:sip%253A888 at conference.freeswitch.org>>
>     > iax:guest at conference.freeswitch.org/888
>     <http://iax:guest at conference.freeswitch.org/888>
>     > <http://iax:guest at conference.freeswitch.org/888>
>     > googletalk:conf+888 at conference.freeswitch.org
>     <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
>     > <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org
>     <mailto:googletalk%253Aconf%252B888 at conference.freeswitch.org>>
>     > pstn:+19193869900
>     >
>     ------------------------------------------------------------------------
>     >
>     > _______________________________________________
>     > FreeSWITCH-users mailing list
>     > FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     >
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     > http://www.freeswitch.org
>     >
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> <mailto:MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> <mailto:sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> <http://iax:guest at conference.freeswitch.org/888>
> googletalk:conf+888 at conference.freeswitch.org
> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
> ------------------------------------------------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   


From valentin.doroga at pronexus.com  Thu Feb 18 14:48:06 2010
From: valentin.doroga at pronexus.com (Valentin Doroga)
Date: Thu, 18 Feb 2010 17:48:06 -0500
Subject: [Freeswitch-users] IVR greeting - first two words missing
In-Reply-To: <8a19bf2e1002181410x44b40216j374a30a5675fd162@mail.gmail.com>
Message-ID: <mailman.3.1266533289.2492.freeswitch-users@lists.freeswitch.org>

          greet-long="test/test-ivr.wav"
          greet-short="tset/test-ivr.wav"
 
test or tset?

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Aloysius
Lloyd
Sent: Thursday, February 18, 2010 5:10 PM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] IVR greeting - first two words missing


Hi All, 

I setup a simple IVR. Here is the script. 

<menu name="test_ivr"
          greet-long="test/test-ivr.wav"
          greet-short="tset/test-ivr.wav"
          invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"
          exit-sound="voicemail/vm-goodbye.wav"
          timeout ="10000"
                  inter-digit-timeout="2000"
                  max-failures="3">
 <entry action="menu-exec-app" digits="/^(20[0-9])$/" param="transfer $1 XML
default"/>
 <entry action="menu-exec-app" digits="9999" param="transfer 9999 XML
default"/>
</menu>
  </menus>
</configuration>

Dial Plan

<action application="answer"/>
<action application="ivr" data="test_ivr"/>

Every time when I reach the IVR . I am getting first one or two words
missing( or may be not clear). How can I fix this issue.


Thanks,
Lloyd 
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100218/ab5803d0/attachment.html 

From iamcanadian at myfastmail.com  Thu Feb 18 14:50:09 2010
From: iamcanadian at myfastmail.com (Edward Stevenson)
Date: Thu, 18 Feb 2010 14:50:09 -0800 (PST)
Subject: [Freeswitch-users]  Voicemail quality
Message-ID: <27251411.post@talk.nabble.com>


I have V1.0.4 running on a production server.  It's working quite well,
except for voicemail retrieval over a satellite internet connection.  Voice
calls over satellite sound fine, other than the 600ms delay.  I'm using the
Howler G729 module for G729 transcoding.

I've noticed that in voice calls, the bandwidth over the satellite is a
steady 24 kbps in both directions.  When accessing voicemail, the bandwidth
fluctuates with voice in the call.  What I mean by that is, pauses in audio
in the voicemail, or in the IVR, cause the bandwidth of the call to drop off
momentarily.  It's almost like it's using a variable bit rate.  The audio
sounds like there's packet loss.  Pops and garbled speach.  This is not
noticable over a landline.  I'm using the built in voicemail.

Anyone else had any similar issues?
-- 
View this message in context: http://old.nabble.com/Voicemail-quality-tp27251411p27251411.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From msc at freeswitch.org  Thu Feb 18 15:22:02 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 18 Feb 2010 15:22:02 -0800
Subject: [Freeswitch-users] IVR greeting - first two words missing
In-Reply-To: <8a19bf2e1002181410x44b40216j374a30a5675fd162@mail.gmail.com>
References: <8a19bf2e1002181410x44b40216j374a30a5675fd162@mail.gmail.com>
Message-ID: <87f2f3b91002181522te296581u12527a2a9cdf1d44@mail.gmail.com>

On Thu, Feb 18, 2010 at 2:10 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:

> Hi All,
>
> I setup a simple IVR. Here is the script.
>
> <menu name="test_ivr"
>           greet-long="test/test-ivr.wav"
>           greet-short="tset/test-ivr.wav"
>           invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"
>           exit-sound="voicemail/vm-goodbye.wav"
>           timeout ="10000"
>                   inter-digit-timeout="2000"
>                   max-failures="3">
>  <entry action="menu-exec-app" digits="/^(20[0-9])$/" param="transfer $1
> XML default"/>
>  <entry action="menu-exec-app" digits="9999" param="transfer 9999 XML
> default"/>
> </menu>
>   </menus>
> </configuration>
>
> *Dial Plan*
>
> <action application="answer"/>
> <action application="ivr" data="test_ivr"/>
>
> Every time when I reach the IVR . I am getting first one or two words
> missing( or may be not clear). How can I fix this issue.
>
> Thanks,
> Lloyd
>

put a sleep after the answer:
<action application="sleep" data="1000"/>

You may have to tinker with the exact time, like maybe 1500 or 2000.
-MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100218/f3925d67/attachment.html 

From dftoro at yahoo.com  Thu Feb 18 15:34:44 2010
From: dftoro at yahoo.com (Diego Toro)
Date: Thu, 18 Feb 2010 15:34:44 -0800 (PST)
Subject: [Freeswitch-users] Establishing a Call from .Net based
	application
In-Reply-To: <6E8D2069C08AA84A83D336E996AE4C67032C9D5BAD@mse17be1.mse17.exchange.ms>
Message-ID: <922191.64755.qm@web33507.mail.mud.yahoo.com>

The managed module is loaded as a module during the startup of FreeSWITCH if set in modules.conf.xml or through the command "load mod_managed" must keep in mind that there is a directory "mod/managed. As mod_managed is loaded into FreeSWITCH process to take control of the call must be running FreeSWITCH. So to "talk" with FreeSWITCH is not necessary to know the IP, the IP depends on the profile you've defined in the configuration of the module sofia. If you need the local address of the box running FreeSWITCH try expand variable $${local_ip_v4} which is assigned automatically by FreeSWITCH.

Being more clear, when you use mod_managed including <action application="managed" data="yourclassname"/>
 in a dialplan already have way to run your C# code.
 
Now, if you need is to have control of the call to answer, originate, etc, without the application run inside FreeSWITCH process, you can use managed ESL (see examples in libs/esl/managed) this library allows your code using events "talk" with FreeSWITCH.

Diego Toro
http://lacarretade.blogspot.com/


--- On Thu, 2/18/10, Michael Giagnocavo <mgg at giagnocavo.net> wrote:

> From: Michael Giagnocavo <mgg at giagnocavo.net>
> Subject: Re: [Freeswitch-users] Establishing a Call from .Net based application
> To: "freeswitch-users at lists.freeswitch.org" <freeswitch-users at lists.freeswitch.org>
> Date: Thursday, February 18, 2010, 6:12 AM
> I?m not sure what the
> FreeSWITCH APIs are to figure out what IP Sofia SIP has
> bound to. Whatever it is, you?d call the same thing in
> C#. What do you want to do with the API? ?mod_managed.dll or .so is the
> FreeSWITCH native code module that loads the CLR or Mono
> into the FreeSWITCH process and loads
> FreeSWITCH.Managed.dll. The managed DLL contains the bulk of
> the managed-unmanaged interop code (.NET definitions of all
> the FS C functions).  ?-Michael ?From:
> freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On
> Behalf Of Scott Fernandez
> Sent: Thursday, February 18, 2010 1:12 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Establishing a Call
> from .Net based application
> ?Hi Diego & Michael,
> 
> Thanks for your reply and support.
> 
> However, I have some clarifications required from both of
> you.
> 
> 1. Here is the question for Diego,
> 
> Simple Example:
> 
> using FreeSWITCH;
> using FreeSWITCH.Native;
> 
> namespace BITS.Ivr.Bp.Server
> {
> ?public class CIV_BPFSProcess : FreeSWITCH.IAppPlugin
> {
> ?public void Run(AppContext context)
> ?{
> ? //answer call
> ? context.Session.Answer();
> ? //sleep 2 seconds
> ? context.Session.sleep(2000, 1);
> ? //hangup call
> ? context.Session.Hangup("NORMAL_CLEARING");
> ?}
> ?}
> }
> I understand that the concept of your example code.
> However, would like to know as to how would my .NET C# know the
> IP address of Freeswitch to talk to it as there is no
> indication for that?. If not here, where would we need to
> reference the IP address of FS in .NET code?
> 
> I guess the IP address of FS needs to be mentioned in the
> Target section of the below web.config file in .NET. If I am
> right, how to specify the IP address over here. If I am
> wrong, please let me know where do we need to mention the IP
> address of FS.
> 
> ??? <configuration>
> ???????????
> <dllmap dll="mod_managed.dll"
> target="mod_managed.so"/>
> ??? </configuration>
> 
> 
> 2. Here is the question for Michael,
> 
> You mentioned that "mod_managed.so will
> be in your freeswitch mod directory". This is
> very clear and what is mod_managed.dll in my .NET
> application and the purpose of it?
> 
> Thanks for all your help.
> 
> Regards,
> Scott.
> 
> 
> On Sun, Feb 14, 2010 at 1:15
> AM, Michael Giagnocavo <mgg at giagnocavo.net>
> wrote:
> 2. There is a configuration settings required to Map the
> "DLL" to ".so" object in CentOS.
> Now, the question is which DLL and .so file to be made
> available and where??If you are
> experiencing NullReferenceExceptions with any plugin run
> through the dialplan, make sure you have included the
> appropriate entry in your dllmap
> configuration: ?<dllmap dll="mod_managed.dll"
> target="mod_managed.so"
> os="!windows"/>?mod_managed.so will
> be in your freeswitch mod directory.
> All I need is to initiate a call from .NET application and
> then it should talk to mod_managed module and establish a
> call. Secondly, I need to know the status of the call such
> as Ringing, Active, Hangup etc. To initiate a
> call, try ManagedSession.Originate.-Michael
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org ?
> -----Inline Attachment Follows-----
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 


      


From brian at microcomaustralia.com.au  Thu Feb 18 15:42:47 2010
From: brian at microcomaustralia.com.au (Brian May)
Date: Fri, 19 Feb 2010 10:42:47 +1100
Subject: [Freeswitch-users] building for Lenny
In-Reply-To: <20100218161741.GC4236@base.carmickle.com>
References: <3c5cf5261002180046vb0015e7t787599c9903aedc@mail.gmail.com>
	<33c87fa31002180056g51ef8496qa9a35746d0fe75ec@mail.gmail.com>
	<3c5cf5261002180110he0ae815uc713f77f289ac625@mail.gmail.com>
	<a9f4977a70b3fcf1b30413ea562ca962.squirrel@www.webcontracts.co.uk>
	<20100218161741.GC4236@base.carmickle.com>
Message-ID: <3c5cf5261002181542y43adb7a7ofe2a15c38baf0327@mail.gmail.com>

On 19 February 2010 03:17, Frank Carmickle <frank at carmickle.com> wrote:
> Like I said you can and should build debs from svn. ?As far as
> I see it there is no reason to not build debs.

Unfortunately, that didn't create any of the packages for the sound
files, and I can't see where to get a deb package for the sound files
that really does contain the sound files.



Also I get errors when trying to start it up, not sure how many of
these I can ignore are warnings and how many are because I am doing it
wrong:

voyage:~# /opt/freeswitch/bin/freeswitch
Error: stacksize 4194303 is too large: run ulimit -s 240 or run
/opt/freeswitch/bin/freeswitch -waste.
auto-adjusting stack size for optimal performance...
2010-02-19 10:59:41.203000 [INFO] switch_event.c:580 Activate Eventing Engine.
2010-02-19 10:59:41.208000 [DEBUG] switch_event.c:568 Create event
dispatch thread 0
2010-02-19 10:59:41.428000 [ERR] switch_xml.c:1297 Couldnt open
/opt/freeswitch/conf/autoload_configs/../ivr_menus/*.xml (No such file
or directory)
Error including
/opt/freeswitch/conf/autoload_configs/../ivr_menus/*.xml (No such file
or directory)
2010-02-19 10:59:42.376000 [ERR] switch_xml.c:1297 Couldnt open
/opt/freeswitch/conf/lang/de/*.xml (No such file or directory)
Error including /opt/freeswitch/conf/lang/de/*.xml (No such file or directory)
2010-02-19 10:59:42.526000 [ERR] switch_xml.c:1297 Couldnt open
/opt/freeswitch/conf/lang/en/dir/sounds.xml (No such file or
directory)
Error including /opt/freeswitch/conf/lang/en/dir/sounds.xml (No such
file or directory)
2010-02-19 10:59:42.528000 [ERR] switch_xml.c:1297 Couldnt open
/opt/freeswitch/conf/lang/fr/*.xml (No such file or directory)
Error including /opt/freeswitch/conf/lang/fr/*.xml (No such file or directory)
2010-02-19 10:59:42.529000 [ERR] switch_xml.c:1297 Couldnt open
/opt/freeswitch/conf/lang/ru/*.xml (No such file or directory)
Error including /opt/freeswitch/conf/lang/ru/*.xml (No such file or directory)
2010-02-19 10:59:42.609000 [INFO] switch_nat.c:409 Scanning for NAT
2010-02-19 10:59:42.611000 [DEBUG] switch_nat.c:166 Checking for PMP 1/5
2010-02-19 10:59:42.614000 [ERR] switch_nat.c:197 Error checking for
PMP [general error]
2010-02-19 10:59:42.614000 [DEBUG] switch_nat.c:414 Checking for UPnP
2010-02-19 10:59:54.619000 [INFO] switch_nat.c:429 No PMP or UPnP NAT
devices detected!
2010-02-19 10:59:54.623000 [INFO] switch_core_sqldb.c:1248 Opening DB
2010-02-19 10:59:54.627000 [ERR] switch_core_sqldb.c:404 SQL ERR [no
such table: channels]
drop table channels
2010-02-19 10:59:54.628000 [ERR] switch_core_sqldb.c:404 SQL ERR [no
such table: calls]
drop table calls
2010-02-19 10:59:54.630000 [ERR] switch_core_sqldb.c:404 SQL ERR [no
such table: interfaces]
drop table interfaces
2010-02-19 10:59:54.631000 [ERR] switch_core_sqldb.c:404 SQL ERR [no
such table: tasks]
drop table tasks
2010-02-19 10:59:54.634000 [DEBUG] switch_core_sqldb.c:765 SQL ERR [no
such table: aliases]
[select hostname from aliases]
Auto Generating Table!
2010-02-19 10:59:54.636000 [DEBUG] switch_core_sqldb.c:772 SQL ERR [no
such table: aliases]
[CREATE TABLE aliases (
   sticky  INTEGER,
   alias  VARCHAR(128),
   command  VARCHAR(4096),
   hostname VARCHAR(256)
);
]
2010-02-19 10:59:54.644000 [DEBUG] switch_core_sqldb.c:765 SQL ERR [no
such table: nat]
[select hostname from nat]
Auto Generating Table!
2010-02-19 10:59:54.646000 [DEBUG] switch_core_sqldb.c:772 SQL ERR [no
such table: nat]
[CREATE TABLE nat (
   sticky  INTEGER,
	port	INTEGER,
	proto	INTEGER,
   hostname VARCHAR(256)
);
]

Am I expected to setup a SQL database to get this working? Or did it
just setup one automatically?

2010-02-19 10:59:59.684000 [CRIT] switch_loadable_module.c:882 Error
Loading module /opt/freeswitch/mod/mod_voipcodecs.so
**libjpeg.so.62: cannot open shared object file: No such file or directory**
2010-02-19 10:59:59.686000 [CRIT] switch_loadable_module.c:882 Error
Loading module /opt/freeswitch/mod/mod_g723_1.so
**/opt/freeswitch/mod/mod_g723_1.so: cannot open shared object file:
No such file or directory**
2010-02-19 10:59:59.687000 [CRIT] switch_loadable_module.c:882 Error
Loading module /opt/freeswitch/mod/mod_g729.so
**/opt/freeswitch/mod/mod_g729.so: cannot open shared object file: No
such file or directory**
2010-02-19 10:59:59.688000 [CRIT] switch_loadable_module.c:882 Error
Loading module /opt/freeswitch/mod/mod_amr.so
**/opt/freeswitch/mod/mod_amr.so: cannot open shared object file: No
such file or directory**
2010-02-19 10:59:59.835000 [CRIT] switch_loadable_module.c:882 Error
Loading module /opt/freeswitch/mod/mod_file_string.so
**/opt/freeswitch/mod/mod_file_string.so: cannot open shared object
file: No such file or directory**
2010-02-19 11:00:00.143000 [CRIT] switch_loadable_module.c:882 Error
Loading module /opt/freeswitch/mod/mod_say_ru.so
**/opt/freeswitch/mod/mod_say_ru.so: cannot open shared object file:
No such file or directory**

suspect I don't really need to worry about some of these. I assume
there is a config file somewhere where I can disable these options.


Ok, as a really pathetic question, now I have started it, how do I stop it?

freeswitch at voyage> halt
Unknown Command: halt
freeswitch at voyage> quit
Unknown Command: quit
freeswitch at voyage> exit
Unknown Command: exit
freeswitch at voyage> bye
Unknown Command: bye

-- 
Brian May <brian at microcomaustralia.com.au>


From iamcanadian at myfastmail.com  Thu Feb 18 17:21:58 2010
From: iamcanadian at myfastmail.com (Edward Stevenson)
Date: Thu, 18 Feb 2010 17:21:58 -0800 (PST)
Subject: [Freeswitch-users]  voivemail quality
Message-ID: <27648642.post@talk.nabble.com>


I have V1.0.4 running on a test/production server.  It's working quite well,
except for voicemail retrieval over a satellite internet connection.  Voice
calls over satellite sound fine, other than the 600ms delay.  I'm using the
Howler Tech G729 module for G729 transcoding.

I've noticed that in voice calls, the bandwidth over the satellite is a
steady 24 kbps in both directions.  When accessing voicemail, the bandwidth
fluctuates with voice in the call.  What I mean by that is, pauses in audio
in the voicemail, or in the IVR, cause the bandwidth of the call to drop off
momentarily.  It's almost like it's using a variable bit rate.  The audio
sounds like there's packet loss.  Pops and garbled speach.  This is not
noticable over a land internet connection.  I'm using the built in
voicemail.

If I change the phone's codec to G711, the call is perfectly clear.  Perhaps
the Howler module doesn't like transcoding from the L16 codec that
Freeswitch seems to use to play the wav file?

Anyone else had any similar issues?
-- 
View this message in context: http://old.nabble.com/voivemail-quality-tp27648642p27648642.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From iamcanadian at myfastmail.com  Thu Feb 18 17:22:48 2010
From: iamcanadian at myfastmail.com (Edward Stevenson)
Date: Thu, 18 Feb 2010 17:22:48 -0800 (PST)
Subject: [Freeswitch-users]  voivemail quality
Message-ID: <27648642.post@talk.nabble.com>


I have V1.0.4 running on a test/production server.  It's working quite well,
except for voicemail retrieval over a satellite internet connection.  Voice
calls over satellite sound fine, other than the 600ms delay.  I'm using the
Howler Tech G729 module for G729 transcoding.

I've noticed that in voice calls, the bandwidth over the satellite is a
steady 24 kbps in both directions.  When accessing voicemail, the bandwidth
fluctuates with voice in the call.  What I mean by that is, pauses in audio
in the voicemail, or in the IVR, cause the bandwidth of the call to drop off
momentarily.  It's almost like it's using a variable bit rate.  The audio
sounds like there's packet loss.  Pops and garbled speach.  This is not
noticable over a land internet connection.  I'm using the built in
voicemail.

If I change the phone's codec to G711, the call is perfectly clear.  Perhaps
the Howler module doesn't like transcoding from the L16 codec that
Freeswitch seems to use to play the wav file?

Anyone else had any similar issues?
-- 
View this message in context: http://old.nabble.com/voivemail-quality-tp27648642p27648642.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From lon at kickasspixels.com  Thu Feb 18 17:41:02 2010
From: lon at kickasspixels.com (Lon Baker)
Date: Thu, 18 Feb 2010 17:41:02 -0800
Subject: [Freeswitch-users] 1.0.4 vs. trunk vs. bugs
Message-ID: <5d3e0dc61002181741r57b91f86h4810de63c1a3d46d@mail.gmail.com>

Guys,

With all due respect I want to suggest a policy change. A while ago it
was announced that bug reports against 1.0.4 would not be accepted and
the solution was to work off the trunk or the latest nightly build.

It seems more reasonable to have a release/production branch that can
be depended on for production use. This branch would only accept
reports and fixes for critical bugs.

The development branch is where feature requests and non-critical bugs
reports would be filed for the next production release.

The current process leaves a gap between production ready and
development code that may become greater over time.

Just a thought.

Lon


From jason at jasonjgw.net  Thu Feb 18 17:51:18 2010
From: jason at jasonjgw.net (Jason White)
Date: Fri, 19 Feb 2010 12:51:18 +1100
Subject: [Freeswitch-users] 1.0.4 vs. trunk vs. bugs
In-Reply-To: <5d3e0dc61002181741r57b91f86h4810de63c1a3d46d@mail.gmail.com>
References: <5d3e0dc61002181741r57b91f86h4810de63c1a3d46d@mail.gmail.com>
Message-ID: <20100219015118.GA12983@jdc.jasonjgw.net>

Lon Baker <lon at kickasspixels.com> wrote:
 
> The development branch is where feature requests and non-critical bugs
> reports would be filed for the next production release.
> 
> The current process leaves a gap between production ready and
> development code that may become greater over time.

Did you read the statements by FreeSWITCH developers indicating that the svn
trunk is usually more stable than "released" versions, and that this is at
least partly due to a lack of testers/testing prior to release?

A change of policy isn't going to address those underlying problems.

For the record, I don't favour the proposed change.



From wangdq.no1 at gmail.com  Thu Feb 18 17:54:12 2010
From: wangdq.no1 at gmail.com (daqiang wang)
Date: Fri, 19 Feb 2010 09:54:12 +0800
Subject: [Freeswitch-users] how to use mod_erlang ?
In-Reply-To: <20100218190129.GD8518@hijacked.us>
References: <ecd05ce31002171917w2713f8detcee3f7e36039e9f2@mail.gmail.com>
	<20100218190129.GD8518@hijacked.us>
Message-ID: <ecd05ce31002181754x338f6b4bt5e3609cd74b28f3b@mail.gmail.com>

ok, thank you very much!


2010/2/19 Andrew Thompson <andrew at hijacked.us>

> On Thu, Feb 18, 2010 at 11:17:26AM +0800, daqiang wang wrote:
> > hello !
> > I test mod_erlang from :
> http://wiki.freeswitch.org/wiki/Mod_erlang_event
> > but when I input :
> >  >
> >
> > {foo, freeswitch at localhost } ! {api, status, ""}.
> > I received
> >   (test at wangdq-laptop)2>
> > =ERROR REPORT==== 18-Feb-2010::11:10:13 ===
> > Error in process <0.41.0> on node 'test at wangdq-laptop' with exit
> > value: {badarg,[{erlang,list_to_existing_atom,["freeswitch at wangdq-laptop
> "]},{dist_util,recv_challenge,1},{dist_util,handshake_we_started,1}]}
> >
>
> Try {foo, 'freeswitch at wangdq-laptop' } ! {api, status, ""}. instead.
>
> Andrew
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100219/f5d92703/attachment.html 

From technical at ttnc.co.uk  Thu Feb 18 17:57:55 2010
From: technical at ttnc.co.uk (TTNC - Technical)
Date: Fri, 19 Feb 2010 01:57:55 +0000
Subject: [Freeswitch-users] mod_fax undefined symbol
Message-ID: <6FCE44FF-70C1-412D-B722-4C8D359381BA@ttnc.co.uk>

Hi Guys

I'm having trouble getting mod_fax to load. Running on Debian testing (squeeze). Everything is installed as per - http://wiki.freeswitch.org/wiki/Installation_Guide. (dpkg-buildpackage etc)

When trying to load the fax module I get:

2010-02-19 01:48:17.554935 [CRIT] switch_loadable_module.c:882 Error Loading module /opt/freeswitch/mod/mod_fax.so
**/opt/freeswitch/mod/mod_fax.so: undefined symbol: TIFFDefaultStripSize**

And when a fax is sent, I'm getting:

 2010-02-19 01:04:57.355330 [ERR] switch_core_session.c:1490 Invalid Application rxfax 

I guess because mod_fax isn't loaded.

I've got libtiff4 and libtiff4-dev installed:

ii  libtiff4                             3.9.2-2                  Tag Image File Format (TIFF) library
ii  libtiff4-dev                         3.9.2-2                  Tag Image File Format library (TIFF), development files
ii  libtiffxx0c2                         3.9.2-2                  Tag Image File Format (TIFF) library -- C++ interface

Just tried updating to the latest svn trunk (16700M) and it hasn't made any difference.

From googling, it suggests that it could be because the module is complied against a different one currently running on the system, however I'm not sure how this can be the case, there is only the one version installed. 

Any suggestions as to what I can try?

Any help appreciated

Russ

From rupa at rupa.com  Thu Feb 18 18:01:17 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Thu, 18 Feb 2010 20:01:17 -0600
Subject: [Freeswitch-users] voivemail quality
In-Reply-To: <27648642.post@talk.nabble.com>
References: <27648642.post@talk.nabble.com>
Message-ID: <b39273531002181801kd81a6b2x3327be4887563eae@mail.gmail.com>

Have you asked Howler about this?  This is not a support channel for
commercial software that doesn't participate or contribute in the community.

On Thu, Feb 18, 2010 at 7:21 PM, Edward Stevenson <
iamcanadian at myfastmail.com> wrote:

>
> I have V1.0.4 running on a test/production server.  It's working quite
> well,
> except for voicemail retrieval over a satellite internet connection.  Voice
> calls over satellite sound fine, other than the 600ms delay.  I'm using the
> Howler Tech G729 module for G729 transcoding.
>
> I've noticed that in voice calls, the bandwidth over the satellite is a
> steady 24 kbps in both directions.  When accessing voicemail, the bandwidth
> fluctuates with voice in the call.  What I mean by that is, pauses in audio
> in the voicemail, or in the IVR, cause the bandwidth of the call to drop
> off
> momentarily.  It's almost like it's using a variable bit rate.  The audio
> sounds like there's packet loss.  Pops and garbled speach.  This is not
> noticable over a land internet connection.  I'm using the built in
> voicemail.
>
> If I change the phone's codec to G711, the call is perfectly clear.
>  Perhaps
> the Howler module doesn't like transcoding from the L16 codec that
> Freeswitch seems to use to play the wav file?
>
> Anyone else had any similar issues?
> --
> View this message in context:
> http://old.nabble.com/voivemail-quality-tp27648642p27648642.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
-Rupa
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100218/664ab29f/attachment-0001.html 

From jaybinks at gmail.com  Thu Feb 18 18:14:00 2010
From: jaybinks at gmail.com (jay binks)
Date: Fri, 19 Feb 2010 12:14:00 +1000
Subject: [Freeswitch-users] voivemail quality
In-Reply-To: <b39273531002181801kd81a6b2x3327be4887563eae@mail.gmail.com>
References: <27648642.post@talk.nabble.com>
	<b39273531002181801kd81a6b2x3327be4887563eae@mail.gmail.com>
Message-ID: <ec2ce17e1002181814l26038ddat19923218eb014ff0@mail.gmail.com>

I might suggest you try with G711 , ilbc, speex ..
see if it works with these ( and not which do / dont work )

then post back, I think Rupa's point is that it may be a codec issue.
be helpful ( either way ) to prove that it is / isnt this 3rd party codec.

Jay

On Fri, Feb 19, 2010 at 12:01 PM, Rupa Schomaker <rupa at rupa.com> wrote:

> Have you asked Howler about this?  This is not a support channel for
> commercial software that doesn't participate or contribute in the community.
>
>
> On Thu, Feb 18, 2010 at 7:21 PM, Edward Stevenson <
> iamcanadian at myfastmail.com> wrote:
>
>>
>> I have V1.0.4 running on a test/production server.  It's working quite
>> well,
>> except for voicemail retrieval over a satellite internet connection.
>>  Voice
>> calls over satellite sound fine, other than the 600ms delay.  I'm using
>> the
>> Howler Tech G729 module for G729 transcoding.
>>
>> I've noticed that in voice calls, the bandwidth over the satellite is a
>> steady 24 kbps in both directions.  When accessing voicemail, the
>> bandwidth
>> fluctuates with voice in the call.  What I mean by that is, pauses in
>> audio
>> in the voicemail, or in the IVR, cause the bandwidth of the call to drop
>> off
>> momentarily.  It's almost like it's using a variable bit rate.  The audio
>> sounds like there's packet loss.  Pops and garbled speach.  This is not
>> noticable over a land internet connection.  I'm using the built in
>> voicemail.
>>
>> If I change the phone's codec to G711, the call is perfectly clear.
>>  Perhaps
>> the Howler module doesn't like transcoding from the L16 codec that
>> Freeswitch seems to use to play the wav file?
>>
>> Anyone else had any similar issues?
>> --
>> View this message in context:
>> http://old.nabble.com/voivemail-quality-tp27648642p27648642.html
>> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Sincerely

Jay
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100219/4e62cc29/attachment.html 

From lon at kickasspixels.com  Thu Feb 18 18:28:30 2010
From: lon at kickasspixels.com (Lon Baker)
Date: Thu, 18 Feb 2010 18:28:30 -0800
Subject: [Freeswitch-users] 1.0.4 vs. trunk vs. bugs
In-Reply-To: <20100219015118.GA12983@jdc.jasonjgw.net>
References: <5d3e0dc61002181741r57b91f86h4810de63c1a3d46d@mail.gmail.com>
	<20100219015118.GA12983@jdc.jasonjgw.net>
Message-ID: <DE4923F9-EEC2-4009-9A0D-D62C1CF3BD0F@kickasspixels.com>

Jason,

Yes. I saw that. But, quite a few times I have found the trunk unstable or broken. Its why I'm proposing the changes.

To be clear, I'm not in anyway impugning the quality or expertise of the dev team. They are doing amazing work! 

I'm looking down the road. As the project grows and FS becomes a critical piece of any company's infrastructure a clear distinction between "stable" and "development" branches are pretty standard in project the scope of FS.

Lon

On Feb 18, 2010, at 5:51 PM, Jason White wrote:

> Lon Baker <lon at kickasspixels.com> wrote:
> 
>> The development branch is where feature requests and non-critical bugs
>> reports would be filed for the next production release.
>> 
>> The current process leaves a gap between production ready and
>> development code that may become greater over time.
> 
> Did you read the statements by FreeSWITCH developers indicating that the svn
> trunk is usually more stable than "released" versions, and that this is at
> least partly due to a lack of testers/testing prior to release?
> 
> A change of policy isn't going to address those underlying problems.
> 
> For the record, I don't favour the proposed change.
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From brian at freeswitch.org  Thu Feb 18 18:35:06 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 18 Feb 2010 20:35:06 -0600
Subject: [Freeswitch-users] 1.0.4 vs. trunk vs. bugs
In-Reply-To: <DE4923F9-EEC2-4009-9A0D-D62C1CF3BD0F@kickasspixels.com>
References: <5d3e0dc61002181741r57b91f86h4810de63c1a3d46d@mail.gmail.com>
	<20100219015118.GA12983@jdc.jasonjgw.net>
	<DE4923F9-EEC2-4009-9A0D-D62C1CF3BD0F@kickasspixels.com>
Message-ID: <865D1CDE-736D-4ED6-B7D6-94384143BCDA@freeswitch.org>

In the grand scheme trunk isn't unstable or broken for long... this is just downright false.  If it was why aren't you opening bugs?

We have had a few snafu's but again they don't stay long.

/b


On Feb 18, 2010, at 8:28 PM, Lon Baker wrote:

> Yes. I saw that. But, quite a few times I have found the trunk unstable or broken. Its why I'm proposing the changes.



From jason at jasonjgw.net  Thu Feb 18 18:53:11 2010
From: jason at jasonjgw.net (Jason White)
Date: Fri, 19 Feb 2010 13:53:11 +1100
Subject: [Freeswitch-users] 1.0.4 vs. trunk vs. bugs
In-Reply-To: <DE4923F9-EEC2-4009-9A0D-D62C1CF3BD0F@kickasspixels.com>
References: <5d3e0dc61002181741r57b91f86h4810de63c1a3d46d@mail.gmail.com>
	<20100219015118.GA12983@jdc.jasonjgw.net>
	<DE4923F9-EEC2-4009-9A0D-D62C1CF3BD0F@kickasspixels.com>
Message-ID: <20100219025311.GA13893@jdc.jasonjgw.net>

Lon Baker <lon at kickasspixels.com> wrote:
 
> I'm looking down the road. As the project grows and FS becomes a critical
> piece of any company's infrastructure a clear distinction between "stable"
> and "development" branches are pretty standard in project the scope of FS.

I would rather that the developers spend time fixing bugs and implementing new
features instead of backporting changes to a "stable" branch that may be very
outdated.

One way of managing this is the Linux kernel's model, with short
development/release cycles, where there is only one branch. (There are
short-lived "stable" branches as in 2.6.32.1, 2.6.32.2 etc., but my
understanding is that bug fixes destined for those branches must already have
been applied to the development branch destined for the next release; this
minimizes back-porting of fixes.)

I'm sure there are other models. My essential point is that, so far, the
FreeSWITCH developers have not chosen to maintain long-lived "stable"
branches, that there are alternatives to this approach, and that it has its
downside, especially regarding the extra time/effort/work associated with
maintaining it.



From lon at kickasspixels.com  Thu Feb 18 18:53:16 2010
From: lon at kickasspixels.com (Lon Baker)
Date: Thu, 18 Feb 2010 18:53:16 -0800
Subject: [Freeswitch-users] 1.0.4 vs. trunk vs. bugs
In-Reply-To: <865D1CDE-736D-4ED6-B7D6-94384143BCDA@freeswitch.org>
References: <5d3e0dc61002181741r57b91f86h4810de63c1a3d46d@mail.gmail.com> 
	<20100219015118.GA12983@jdc.jasonjgw.net>
	<DE4923F9-EEC2-4009-9A0D-D62C1CF3BD0F@kickasspixels.com> 
	<865D1CDE-736D-4ED6-B7D6-94384143BCDA@freeswitch.org>
Message-ID: <5d3e0dc61002181853u2b5d7022ga79d41adaeb7ae68@mail.gmail.com>

I'll drop it. You're right they don't last long. I didn't open bugs
because I assumed the trunk is being worked in, so breakage is
something I expect and anywhere from minutes to a few hours later they
are gone.

Lon

On Thu, Feb 18, 2010 at 6:35 PM, Brian West <brian at freeswitch.org> wrote:
> In the grand scheme trunk isn't unstable or broken for long... this is just downright false. ?If it was why aren't you opening bugs?
>
> We have had a few snafu's but again they don't stay long.
>
> /b
>
>
> On Feb 18, 2010, at 8:28 PM, Lon Baker wrote:
>
>> Yes. I saw that. But, quite a few times I have found the trunk unstable or broken. Its why I'm proposing the changes.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From lon at kickasspixels.com  Thu Feb 18 19:08:09 2010
From: lon at kickasspixels.com (Lon Baker)
Date: Thu, 18 Feb 2010 19:08:09 -0800
Subject: [Freeswitch-users] 1.0.4 vs. trunk vs. bugs
In-Reply-To: <20100219025311.GA13893@jdc.jasonjgw.net>
References: <5d3e0dc61002181741r57b91f86h4810de63c1a3d46d@mail.gmail.com> 
	<20100219015118.GA12983@jdc.jasonjgw.net>
	<DE4923F9-EEC2-4009-9A0D-D62C1CF3BD0F@kickasspixels.com> 
	<20100219025311.GA13893@jdc.jasonjgw.net>
Message-ID: <5d3e0dc61002181908p5534c03fw5a258f165c9d3119@mail.gmail.com>

Jason,

I agree with you. Guess its just the anticipation for 1.0.5 to be
released and clients that only allow "released" versions to be
deployed.

Lon


From jmesquita at freeswitch.org  Thu Feb 18 19:29:58 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Fri, 19 Feb 2010 00:29:58 -0300
Subject: [Freeswitch-users] more than 1 profile for sofia/user ?
In-Reply-To: <4B7DC202.7090409@gmx.net>
References: <4B7D9494.8050208@gmx.net>
	<191c3a031002181140y44135106yb34e6a6f73a04773@mail.gmail.com>
	<4B7DADC8.1060405@gmx.net>
	<191c3a031002181353r2dbf15c6h206509f9b1399148@mail.gmail.com>
	<4B7DC202.7090409@gmx.net>
Message-ID: <ebbc2d551002181929m72a17b7bj1fa1cff45b02607b@mail.gmail.com>

I would:

{presence_id=${dialed_user}@
${dialed_domain},transfer_fallback_extension=${dialed_user}}${sofia_contact(${dialed_user}@
${dialed_domain}),sofia/other_profile/${dialed_user}}

You could toy with that a bit. The dialstring is really just an origination
string that is generated by the user/<extension> ...

Hope that clears it up a bit.

Regards,
Jo?o Mesquita



On Thu, Feb 18, 2010 at 7:41 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:

> Hello Anthony,
>
> >add on a , then another dial string to reflect the other profile too
> I really tried to understand this, but
> can you give me an example?
>
> Best regards
> Peter
>
> Anthony Minessale schrieb:
> > add on a , then another dial string to reflect the other profile too
> >
> > On Thu, Feb 18, 2010 at 3:14 PM, Peter P GMX <Prometheus001 at gmx.net
> > <mailto:Prometheus001 at gmx.net>> wrote:
> >
> >     Any idea how to do this?
> >
> >     currently I have
> >     {presence_id=${dialed_user}@
> ${dialed_domain},transfer_fallback_extension=${dialed_user}}${sofia_contact(${dialed_user}@
> ${dialed_domain})}
> >
> >
> >     Best regards
> >     Peter
> >
> >     Anthony Minessale schrieb:
> >     > edit the dial-string for that user in the directory xml to try the
> >     > extension on both profile at once
> >     >
> >     > On Thu, Feb 18, 2010 at 1:27 PM, Peter P GMX
> >     <Prometheus001 at gmx.net <mailto:Prometheus001 at gmx.net>
> >     > <mailto:Prometheus001 at gmx.net <mailto:Prometheus001 at gmx.net>>>
> >     wrote:
> >     >
> >     >     Hello,
> >     >
> >     >     in the standard setup - if a phone is registering to port
> >     5060 - it is
> >     >     bound to the "internal" profile. And I can dial it via
> >     >     sofia/user/xxxx then.
> >     >
> >     >     However due to NAT issues I would like to have to 2 seperate
> >     profiles
> >     >     for SIP phones. For example I have a "local" profile for all
> >     devices
> >     >     inside the LAN (e.g. Pattons und in future: local phones)
> >     and another
> >     >     "internal" profile which allows also external phones via
> >     >     external-xxx-ip. That way I would like to ensure that local
> >     phones
> >     >     have
> >     >     nothing to do with natted adresses and that external phones can
> >     >     register
> >     >     via external IPs.
> >     >
> >     >     Question How do I manage that I can register a phone to the
> >     "local"
> >     >     profile and being able to dial that phone via sofia/user/xxxxx?
> >     >
> >     >     Or do I think too complicated and there is simply nothing
> >     special
> >     >     to do?
> >     >
> >     >     Best regards
> >     >     Peter
> >     >
> >     >     _______________________________________________
> >     >     FreeSWITCH-users mailing list
> >     >     FreeSWITCH-users at lists.freeswitch.org
> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>
> >     >     <mailto:FreeSWITCH-users at lists.freeswitch.org
> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>>
> >     >     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >     >
> >     UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >     >     http://www.freeswitch.org
> >     >
> >     >
> >     >
> >     >
> >     > --
> >     > Anthony Minessale II
> >     >
> >     > FreeSWITCH http://www.freeswitch.org/
> >     > ClueCon http://www.cluecon.com/
> >     > Twitter: http://twitter.com/FreeSWITCH_wire
> >     >
> >     > AIM: anthm
> >     > MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
> >     <mailto:MSN%3Aanthony_minessale at hotmail.com<MSN%253Aanthony_minessale at hotmail.com>
> >
> >     > <mailto:MSN%3Aanthony_minessale at hotmail.com<MSN%253Aanthony_minessale at hotmail.com>
> >     <mailto:MSN%253Aanthony_minessale at hotmail.com<MSN%25253Aanthony_minessale at hotmail.com>
> >>
> >     > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >     <mailto:PAYPAL%3Aanthony.minessale at gmail.com<PAYPAL%253Aanthony.minessale at gmail.com>
> >
> >     > <mailto:PAYPAL%3Aanthony.minessale at gmail.com<PAYPAL%253Aanthony.minessale at gmail.com>
> >     <mailto:PAYPAL%253Aanthony.minessale at gmail.com<PAYPAL%25253Aanthony.minessale at gmail.com>
> >>
> >     > IRC: irc.freenode.net <http://irc.freenode.net>
> >     <http://irc.freenode.net> #freeswitch
> >     >
> >     > FreeSWITCH Developer Conference
> >     > sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> >     <mailto:sip%3A888 at conference.freeswitch.org<sip%253A888 at conference.freeswitch.org>
> >
> >     > <mailto:sip%3A888 at conference.freeswitch.org<sip%253A888 at conference.freeswitch.org>
> >     <mailto:sip%253A888 at conference.freeswitch.org<sip%25253A888 at conference.freeswitch.org>
> >>
> >     > iax:guest at conference.freeswitch.org/888
> >     <http://iax:guest at conference.freeswitch.org/888>
> >     > <http://iax:guest at conference.freeswitch.org/888>
> >     > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >     <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org<googletalk%253Aconf%252B888 at conference.freeswitch.org>
> >
> >     > <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org<googletalk%253Aconf%252B888 at conference.freeswitch.org>
> >     <mailto:googletalk%253Aconf%252B888 at conference.freeswitch.org<googletalk%25253Aconf%25252B888 at conference.freeswitch.org>
> >>
> >     > pstn:+19193869900
> >     >
> >
> ------------------------------------------------------------------------
> >     >
> >     > _______________________________________________
> >     > FreeSWITCH-users mailing list
> >     > FreeSWITCH-users at lists.freeswitch.org
> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>
> >     > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >     >
> >     UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >     > http://www.freeswitch.org
> >     >
> >
> >     _______________________________________________
> >     FreeSWITCH-users mailing list
> >     FreeSWITCH-users at lists.freeswitch.org
> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>
> >     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >     UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >     http://www.freeswitch.org
> >
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> > Twitter: http://twitter.com/FreeSWITCH_wire
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> > <mailto:MSN%3Aanthony_minessale at hotmail.com<MSN%253Aanthony_minessale at hotmail.com>
> >
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> > <mailto:PAYPAL%3Aanthony.minessale at gmail.com<PAYPAL%253Aanthony.minessale at gmail.com>
> >
> > IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> > <mailto:sip%3A888 at conference.freeswitch.org<sip%253A888 at conference.freeswitch.org>
> >
> > iax:guest at conference.freeswitch.org/888
> > <http://iax:guest at conference.freeswitch.org/888>
> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org<googletalk%253Aconf%252B888 at conference.freeswitch.org>
> >
> > pstn:+19193869900
> > ------------------------------------------------------------------------
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100219/690877ed/attachment-0001.html 

From brian at freeswitch.org  Thu Feb 18 19:34:07 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 18 Feb 2010 21:34:07 -0600
Subject: [Freeswitch-users] 1.0.4 vs. trunk vs. bugs
In-Reply-To: <20100219025311.GA13893@jdc.jasonjgw.net>
References: <5d3e0dc61002181741r57b91f86h4810de63c1a3d46d@mail.gmail.com>
	<20100219015118.GA12983@jdc.jasonjgw.net>
	<DE4923F9-EEC2-4009-9A0D-D62C1CF3BD0F@kickasspixels.com>
	<20100219025311.GA13893@jdc.jasonjgw.net>
Message-ID: <98C2329F-D453-493E-9CC0-AA17DC023B7A@freeswitch.org>

The one thing I'm going to stop doing is helping anyone unless they write docs.  I can't count the times I have helped someone with the promise they would write docs.  They get what they want and move on... its selfish and rude.  The same goes for a stable branch... it seems its wanted/demanded but again nobody steps up to help with it.  I have been at this for over 14 hours today and yet still have a stack of stuff to do that keeps growing.

I thank each and everyone that helps out... I truly enjoy having you all involved.

Thanks,
Brian


On Feb 18, 2010, at 8:53 PM, Jason White wrote:

> I'm sure there are other models. My essential point is that, so far, the
> FreeSWITCH developers have not chosen to maintain long-lived "stable"
> branches, that there are alternatives to this approach, and that it has its
> downside, especially regarding the extra time/effort/work associated with
> maintaining it.



From brian at freeswitch.org  Thu Feb 18 19:36:57 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 18 Feb 2010 21:36:57 -0600
Subject: [Freeswitch-users] 1.0.4 vs. trunk vs. bugs
In-Reply-To: <5d3e0dc61002181853u2b5d7022ga79d41adaeb7ae68@mail.gmail.com>
References: <5d3e0dc61002181741r57b91f86h4810de63c1a3d46d@mail.gmail.com>
	<20100219015118.GA12983@jdc.jasonjgw.net>
	<DE4923F9-EEC2-4009-9A0D-D62C1CF3BD0F@kickasspixels.com>
	<865D1CDE-736D-4ED6-B7D6-94384143BCDA@freeswitch.org>
	<5d3e0dc61002181853u2b5d7022ga79d41adaeb7ae68@mail.gmail.com>
Message-ID: <32F19ACC-1D6B-4C94-BB2E-CB53FD143DC2@freeswitch.org>

We usually don't go to bed if its not fixed... I recall many times I was up at 2am dialing digits and testing while Anthony was coding the fix bugs... its part of what we do and shows just how dedicated we are to this project and how picky we are about the code.

/b

On Feb 18, 2010, at 8:53 PM, Lon Baker wrote:

> I'll drop it. You're right they don't last long. I didn't open bugs
> because I assumed the trunk is being worked in, so breakage is
> something I expect and anywhere from minutes to a few hours later they
> are gone.
> 
> Lon



From lon at kickasspixels.com  Thu Feb 18 20:00:09 2010
From: lon at kickasspixels.com (Kickass Pixels)
Date: Thu, 18 Feb 2010 20:00:09 -0800
Subject: [Freeswitch-users] 1.0.4 vs. trunk vs. bugs
In-Reply-To: <32F19ACC-1D6B-4C94-BB2E-CB53FD143DC2@freeswitch.org>
References: <5d3e0dc61002181741r57b91f86h4810de63c1a3d46d@mail.gmail.com>
	<20100219015118.GA12983@jdc.jasonjgw.net>
	<DE4923F9-EEC2-4009-9A0D-D62C1CF3BD0F@kickasspixels.com>
	<865D1CDE-736D-4ED6-B7D6-94384143BCDA@freeswitch.org>
	<5d3e0dc61002181853u2b5d7022ga79d41adaeb7ae68@mail.gmail.com>
	<32F19ACC-1D6B-4C94-BB2E-CB53FD143DC2@freeswitch.org>
Message-ID: <A10ABF8D-6900-49DE-9020-A7753C6A98BF@kickasspixels.com>

Brian,

Didn't intend to step on any toes. I do open bug reports, have  
submitted patches and have someone on my team who does contribute to  
the docs/wiki.

If the team wants to maintain a stable trunk I will try to find a team  
member to help manage it. If there is a philosophical reason for not  
doing it, I'm fine the current process. Whatever helps the dev team.

You guys do a great job and its a pleasure to work with freeswitch.

Lon

On Feb 18, 2010, at 7:36 PM, Brian West <brian at freeswitch.org> wrote:

> We usually don't go to bed if its not fixed... I recall many times I  
> was up at 2am dialing digits and testing while Anthony was coding  
> the fix bugs... its part of what we do and shows just how dedicated  
> we are to this project and how picky we are about the code.
>
> /b
>
> On Feb 18, 2010, at 8:53 PM, Lon Baker wrote:
>
>> I'll drop it. You're right they don't last long. I didn't open bugs
>> because I assumed the trunk is being worked in, so breakage is
>> something I expect and anywhere from minutes to a few hours later  
>> they
>> are gone.
>>
>> Lon
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org


From infos at madovsky.org  Thu Feb 18 20:08:30 2010
From: infos at madovsky.org (Madovsky)
Date: Thu, 18 Feb 2010 23:08:30 -0500
Subject: [Freeswitch-users] 1.0.4 vs. trunk vs. bugs
References: <5d3e0dc61002181741r57b91f86h4810de63c1a3d46d@mail.gmail.com><20100219015118.GA12983@jdc.jasonjgw.net><DE4923F9-EEC2-4009-9A0D-D62C1CF3BD0F@kickasspixels.com><20100219025311.GA13893@jdc.jasonjgw.net>
	<98C2329F-D453-493E-9CC0-AA17DC023B7A@freeswitch.org>
Message-ID: <FA6FB3864EE44C9581F0FD2A28738084@MOBILEE1705>


----- Original Message ----- 
From: "Brian West" <brian at freeswitch.org>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Thursday, February 18, 2010 10:34 PM
Subject: Re: [Freeswitch-users] 1.0.4 vs. trunk vs. bugs


> The one thing I'm going to stop doing is helping anyone unless they write 
> docs.  I can't count the times I have helped someone with the promise they 
> would write docs.  They get what they want and move on... its selfish and 
> rude.  The same goes for a stable branch... it seems its wanted/demanded 
> but again nobody steps up to help with it.  I have been at this for over 
> 14 hours today and yet still have a stack of stuff to do that keeps 
> growing.
>
> I thank each and everyone that helps out... I truly enjoy having you all 
> involved.
>
> Thanks,
> Brian
>
>
> On Feb 18, 2010, at 8:53 PM, Jason White wrote:
>
>> I'm sure there are other models. My essential point is that, so far, the
>> FreeSWITCH developers have not chosen to maintain long-lived "stable"
>> branches, that there are alternatives to this approach, and that it has 
>> its
>> downside, especially regarding the extra time/effort/work associated with
>> maintaining it.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


This kind of guy is called "leechers", since the internet exists there are 
leechers...
Don't be scared Brian, you are not alone in the same case,  i work also in 
programming and research 15 hours / days
since 8 years.... but I can say that Freeswitch has a very interesting wiki 
doc rather than other app ... 



From dave at 3c.co.uk  Thu Feb 18 20:10:05 2010
From: dave at 3c.co.uk (David Knell)
Date: Thu, 18 Feb 2010 21:10:05 -0700
Subject: [Freeswitch-users] 1.0.4 vs. trunk vs. bugs
In-Reply-To: <20100219015118.GA12983@jdc.jasonjgw.net>
References: <5d3e0dc61002181741r57b91f86h4810de63c1a3d46d@mail.gmail.com>
	<20100219015118.GA12983@jdc.jasonjgw.net>
Message-ID: <1266552605.7684.11.camel@local.freepabx.com>


> Lon Baker <lon at kickasspixels.com> wrote:
>  
> > The development branch is where feature requests and non-critical bugs
> > reports would be filed for the next production release.
> > 
> > The current process leaves a gap between production ready and
> > development code that may become greater over time.

Going against the grain here, I agree with you.  The current way of
doing things is, in my opinion, not well thought through - there's no
reason to tag and release versions if the answer to any issue is 'make
current', and support is not available unless that's been done.  Far
better to either have meaningful releases with stable and devel
branches, or not to have releases at all.

--Dave




From brian at freeswitch.org  Thu Feb 18 20:09:54 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 18 Feb 2010 22:09:54 -0600
Subject: [Freeswitch-users] 1.0.4 vs. trunk vs. bugs
In-Reply-To: <A10ABF8D-6900-49DE-9020-A7753C6A98BF@kickasspixels.com>
References: <5d3e0dc61002181741r57b91f86h4810de63c1a3d46d@mail.gmail.com>
	<20100219015118.GA12983@jdc.jasonjgw.net>
	<DE4923F9-EEC2-4009-9A0D-D62C1CF3BD0F@kickasspixels.com>
	<865D1CDE-736D-4ED6-B7D6-94384143BCDA@freeswitch.org>
	<5d3e0dc61002181853u2b5d7022ga79d41adaeb7ae68@mail.gmail.com>
	<32F19ACC-1D6B-4C94-BB2E-CB53FD143DC2@freeswitch.org>
	<A10ABF8D-6900-49DE-9020-A7753C6A98BF@kickasspixels.com>
Message-ID: <B4813D53-115C-4261-99EE-67CF06D2B945@freeswitch.org>


On Feb 18, 2010, at 10:00 PM, Kickass Pixels wrote:

> Brian,
> 
> Didn't intend to step on any toes. I do open bug reports, have  
> submitted patches and have someone on my team who does contribute to  
> the docs/wiki.

I thank them... I have to keep a close eye on the wiki because those viagra spammers hit it every now and again.  :P  So I know when and who does work pretty much real time... thanks to handy RSS feed of recent changes ;)

> If the team wants to maintain a stable trunk I will try to find a team  
> member to help manage it. If there is a philosophical reason for not  
> doing it, I'm fine the current process. Whatever helps the dev team.

If you want to assemble a team to help manage a "stable" branch then we'll allow you to do so.  I even welcome it if you want to help with this.  Its the only reason we aren't able to do it we don't have enough bodies to keep everything going.  ;)


> You guys do a great job and its a pleasure to work with freeswitch.

You too... keep up the good work... its these kinds of exchanges that bring about great ideas and change.

> 
> Lon



From dave at 3c.co.uk  Thu Feb 18 20:14:40 2010
From: dave at 3c.co.uk (David Knell)
Date: Thu, 18 Feb 2010 21:14:40 -0700
Subject: [Freeswitch-users] voivemail quality
In-Reply-To: <b39273531002181801kd81a6b2x3327be4887563eae@mail.gmail.com>
References: <27648642.post@talk.nabble.com>
	<b39273531002181801kd81a6b2x3327be4887563eae@mail.gmail.com>
Message-ID: <1266552880.7684.17.camel@local.freepabx.com>

Rupa -

Howler filled a big hole that we've been promised is going to be filled
for the last two years, but said filling is still not available.  That's
a significant risk and contribution on their part.

And the request is as to whether others have experienced similar
problems, which strikes me as completely reasonable on a users mailing
list.  If you can't help Edward, then I'd suggest you keep quiet.

--Dave

Rupa wrote:
> Have you asked Howler about this?  This is not a support channel for
> commercial software that doesn't participate or contribute in the
> community.
> 
> On Thu, Feb 18, 2010 at 7:21 PM, Edward Stevenson
> <iamcanadian at myfastmail.com> wrote:
>         
>         
>         I have V1.0.4 running on a test/production server.  It's
>         working quite well,
>         except for voicemail retrieval over a satellite internet
>         connection.  Voice
>         calls over satellite sound fine, other than the 600ms delay.
>          I'm using the
>         Howler Tech G729 module for G729 transcoding.
>         
>         I've noticed that in voice calls, the bandwidth over the
>         satellite is a
>         steady 24 kbps in both directions.  When accessing voicemail,
>         the bandwidth
>         fluctuates with voice in the call.  What I mean by that is,
>         pauses in audio
>         in the voicemail, or in the IVR, cause the bandwidth of the
>         call to drop off
>         momentarily.  It's almost like it's using a variable bit
>         rate.  The audio
>         sounds like there's packet loss.  Pops and garbled speach.
>          This is not
>         noticable over a land internet connection.  I'm using the
>         built in
>         voicemail.
>         
>         If I change the phone's codec to G711, the call is perfectly
>         clear.  Perhaps
>         the Howler module doesn't like transcoding from the L16 codec
>         that
>         Freeswitch seems to use to play the wav file?
>         
>         Anyone else had any similar issues?
>         --
>         View this message in context:
>         http://old.nabble.com/voivemail-quality-tp27648642p27648642.html
>         Sent from the Freeswitch-users mailing list archive at
>         Nabble.com.
>         
>         
>         _______________________________________________
>         FreeSWITCH-users mailing list
>         FreeSWITCH-users at lists.freeswitch.org
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         http://www.freeswitch.org
>         
> 
> 
> 
> -- 
> -Rupa
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From brian at freeswitch.org  Thu Feb 18 20:16:40 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 18 Feb 2010 22:16:40 -0600
Subject: [Freeswitch-users] 1.0.4 vs. trunk vs. bugs
In-Reply-To: <1266552605.7684.11.camel@local.freepabx.com>
References: <5d3e0dc61002181741r57b91f86h4810de63c1a3d46d@mail.gmail.com>
	<20100219015118.GA12983@jdc.jasonjgw.net>
	<1266552605.7684.11.camel@local.freepabx.com>
Message-ID: <616F57B5-182A-49A9-9825-2E5B72F5DB05@freeswitch.org>

OK so I can sign you up for the stable team?  ;)  As per my previous email i'm 100% sure we would do a stable release if we had people tending to issues.  The only problem is you would have to be on IRC tending to issues because if tony sees someone asking about a problem he'll be diving in to fix it before they can say "I have this one".  This also means working in a similar manner we do already.  Our process is very chaotic at times but it has served us well so far.  

The goal is to leave Anthony alone so he can move forward and let the stable team manage the jira's and issues on the list related to stable.

/b

On Feb 18, 2010, at 10:10 PM, David Knell wrote:

>> 
>> Lon Baker <lon at kickasspixels.com> wrote:
>> 
>>> The development branch is where feature requests and non-critical bugs
>>> reports would be filed for the next production release.
>>> 
>>> The current process leaves a gap between production ready and
>>> development code that may become greater over time.
> 
> Going against the grain here, I agree with you.  The current way of
> doing things is, in my opinion, not well thought through - there's no
> reason to tag and release versions if the answer to any issue is 'make
> current', and support is not available unless that's been done.  Far
> better to either have meaningful releases with stable and devel
> branches, or not to have releases at all.
> 
> --Dave

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100218/70f07be6/attachment-0001.html 

From brian at freeswitch.org  Thu Feb 18 20:21:11 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 18 Feb 2010 22:21:11 -0600
Subject: [Freeswitch-users] voivemail quality
In-Reply-To: <1266552880.7684.17.camel@local.freepabx.com>
References: <27648642.post@talk.nabble.com>
	<b39273531002181801kd81a6b2x3327be4887563eae@mail.gmail.com>
	<1266552880.7684.17.camel@local.freepabx.com>
Message-ID: <05F7797B-510C-431B-B78B-D1AB58A81B37@freeswitch.org>

We have g729 ready and tested... We just have to flip the switch and write a check.  I have personally been working on this.  The installer and testing of the license server and various registration details isn't a trivial task.  

I have been asking for testers over the past few weeks to flush out issues so we can lower our support overhead as our resources are already taxed to their limits.  I thank everyone that has been involved in testing.. our first g729 release should be next week for the G729  codec at $10.00 per channel direct from our website.

Thanks,
Brian

On Feb 18, 2010, at 10:14 PM, David Knell wrote:

> Rupa -
> 
> Howler filled a big hole that we've been promised is going to be filled
> for the last two years, but said filling is still not available.  That's
> a significant risk and contribution on their part.
> 
> And the request is as to whether others have experienced similar
> problems, which strikes me as completely reasonable on a users mailing
> list.  If you can't help Edward, then I'd suggest you keep quiet.
> 
> --Dave



From jaybinks at gmail.com  Thu Feb 18 20:26:24 2010
From: jaybinks at gmail.com (jay binks)
Date: Fri, 19 Feb 2010 14:26:24 +1000
Subject: [Freeswitch-users] voivemail quality
In-Reply-To: <1266552880.7684.17.camel@local.freepabx.com>
References: <27648642.post@talk.nabble.com>
	<b39273531002181801kd81a6b2x3327be4887563eae@mail.gmail.com> 
	<1266552880.7684.17.camel@local.freepabx.com>
Message-ID: <ec2ce17e1002182026m5986ab85t71006d608af73f88@mail.gmail.com>

I think suggesting the user test with other codecs is very helpful..

your attacking rupa  for making suggestions about where the issue is
and what to do in order to resolve it how does that help ??

pot, kettle, black ???

J

On Fri, Feb 19, 2010 at 2:14 PM, David Knell <dave at 3c.co.uk> wrote:

> Rupa -
>
> Howler filled a big hole that we've been promised is going to be filled
> for the last two years, but said filling is still not available.  That's
> a significant risk and contribution on their part.
>
> And the request is as to whether others have experienced similar
> problems, which strikes me as completely reasonable on a users mailing
> list.  If you can't help Edward, then I'd suggest you keep quiet.
>
> --Dave
>
> Rupa wrote:
> > Have you asked Howler about this?  This is not a support channel for
> > commercial software that doesn't participate or contribute in the
> > community.
> >
> > On Thu, Feb 18, 2010 at 7:21 PM, Edward Stevenson
> > <iamcanadian at myfastmail.com> wrote:
> >
> >
> >         I have V1.0.4 running on a test/production server.  It's
> >         working quite well,
> >         except for voicemail retrieval over a satellite internet
> >         connection.  Voice
> >         calls over satellite sound fine, other than the 600ms delay.
> >          I'm using the
> >         Howler Tech G729 module for G729 transcoding.
> >
> >         I've noticed that in voice calls, the bandwidth over the
> >         satellite is a
> >         steady 24 kbps in both directions.  When accessing voicemail,
> >         the bandwidth
> >         fluctuates with voice in the call.  What I mean by that is,
> >         pauses in audio
> >         in the voicemail, or in the IVR, cause the bandwidth of the
> >         call to drop off
> >         momentarily.  It's almost like it's using a variable bit
> >         rate.  The audio
> >         sounds like there's packet loss.  Pops and garbled speach.
> >          This is not
> >         noticable over a land internet connection.  I'm using the
> >         built in
> >         voicemail.
> >
> >         If I change the phone's codec to G711, the call is perfectly
> >         clear.  Perhaps
> >         the Howler module doesn't like transcoding from the L16 codec
> >         that
> >         Freeswitch seems to use to play the wav file?
> >
> >         Anyone else had any similar issues?
> >         --
> >         View this message in context:
> >         http://old.nabble.com/voivemail-quality-tp27648642p27648642.html
> >         Sent from the Freeswitch-users mailing list archive at
> >         Nabble.com.
> >
> >
> >         _______________________________________________
> >         FreeSWITCH-users mailing list
> >         FreeSWITCH-users at lists.freeswitch.org
> >         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >         UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >         http://www.freeswitch.org
> >
> >
> >
> >
> > --
> > -Rupa
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely

Jay
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100219/260cddfc/attachment.html 

From brian at freeswitch.org  Thu Feb 18 20:32:13 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 18 Feb 2010 22:32:13 -0600
Subject: [Freeswitch-users] voivemail quality
In-Reply-To: <ec2ce17e1002182026m5986ab85t71006d608af73f88@mail.gmail.com>
References: <27648642.post@talk.nabble.com>
	<b39273531002181801kd81a6b2x3327be4887563eae@mail.gmail.com>
	<1266552880.7684.17.camel@local.freepabx.com>
	<ec2ce17e1002182026m5986ab85t71006d608af73f88@mail.gmail.com>
Message-ID: <C8ECE002-AA2B-4E3B-AEEA-0FE92222F48C@freeswitch.org>

NOW NOW everyone just needs to get their panties out of a bunch.  All was helpful... We are all here to accomplish the same thing and by working together more we can accomplish these tasks.

After this friday Meeting our weekly meetings will be moved to wednesdays.  We have a few things to go over and some of them are all related to what happens when 1.0.5 is tagged.

/b

On Feb 18, 2010, at 10:26 PM, jay binks wrote:

> I think suggesting the user test with other codecs is very helpful..
> 
> your attacking rupa  for making suggestions about where the issue is
> and what to do in order to resolve it how does that help ??
> 
> pot, kettle, black ???
> 
> J



From dave at 3c.co.uk  Thu Feb 18 20:53:29 2010
From: dave at 3c.co.uk (David Knell)
Date: Thu, 18 Feb 2010 21:53:29 -0700
Subject: [Freeswitch-users] voivemail quality
In-Reply-To: <ec2ce17e1002182026m5986ab85t71006d608af73f88@mail.gmail.com>
References: <27648642.post@talk.nabble.com>
	<b39273531002181801kd81a6b2x3327be4887563eae@mail.gmail.com>
	<1266552880.7684.17.camel@local.freepabx.com>
	<ec2ce17e1002182026m5986ab85t71006d608af73f88@mail.gmail.com>
Message-ID: <1266555209.7684.23.camel@local.freepabx.com>


On Fri, 2010-02-19 at 14:26 +1000, jay binks wrote:
> I think suggesting the user test with other codecs is very helpful..

Edward already said that things worked with G.711 - that ought to be
enough testing with other codecs.

> your attacking rupa  for making suggestions about where the issue is
> and what to do in order to resolve it how does that help ??

It doesn't, directly.  And, had Rupa simply asked if the OP had raised
the issue with Howler - which is a completely valid thing to do,
particularly with the evidence in front of him - that'd have been one
thing.  But he didn't.

> pot, kettle, black ???

How so, exactly?

--Dave

> 
> 
> J
> 
> On Fri, Feb 19, 2010 at 2:14 PM, David Knell <dave at 3c.co.uk> wrote:
>         Rupa -
>         
>         Howler filled a big hole that we've been promised is going to
>         be filled
>         for the last two years, but said filling is still not
>         available.  That's
>         a significant risk and contribution on their part.
>         
>         And the request is as to whether others have experienced
>         similar
>         problems, which strikes me as completely reasonable on a users
>         mailing
>         list.  If you can't help Edward, then I'd suggest you keep
>         quiet.
>         
>         --Dave
>         
>         
>         Rupa wrote:
>         > Have you asked Howler about this?  This is not a support
>         channel for
>         > commercial software that doesn't participate or contribute
>         in the
>         > community.
>         >
>         > On Thu, Feb 18, 2010 at 7:21 PM, Edward Stevenson
>         > <iamcanadian at myfastmail.com> wrote:
>         >
>         >
>         >         I have V1.0.4 running on a test/production server.
>          It's
>         >         working quite well,
>         >         except for voicemail retrieval over a satellite
>         internet
>         >         connection.  Voice
>         >         calls over satellite sound fine, other than the
>         600ms delay.
>         >          I'm using the
>         >         Howler Tech G729 module for G729 transcoding.
>         >
>         >         I've noticed that in voice calls, the bandwidth over
>         the
>         >         satellite is a
>         >         steady 24 kbps in both directions.  When accessing
>         voicemail,
>         >         the bandwidth
>         >         fluctuates with voice in the call.  What I mean by
>         that is,
>         >         pauses in audio
>         >         in the voicemail, or in the IVR, cause the bandwidth
>         of the
>         >         call to drop off
>         >         momentarily.  It's almost like it's using a variable
>         bit
>         >         rate.  The audio
>         >         sounds like there's packet loss.  Pops and garbled
>         speach.
>         >          This is not
>         >         noticable over a land internet connection.  I'm
>         using the
>         >         built in
>         >         voicemail.
>         >
>         >         If I change the phone's codec to G711, the call is
>         perfectly
>         >         clear.  Perhaps
>         >         the Howler module doesn't like transcoding from the
>         L16 codec
>         >         that
>         >         Freeswitch seems to use to play the wav file?
>         >
>         >         Anyone else had any similar issues?
>         >         --
>         >         View this message in context:
>         >
>         http://old.nabble.com/voivemail-quality-tp27648642p27648642.html
>         >         Sent from the Freeswitch-users mailing list archive
>         at
>         >         Nabble.com.
>         >
>         >
>         >         _______________________________________________
>         >         FreeSWITCH-users mailing list
>         >         FreeSWITCH-users at lists.freeswitch.org
>         >
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         >
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         >         http://www.freeswitch.org
>         >
>         >
>         >
>         >
>         > --
>         > -Rupa
>         > _______________________________________________
>         > FreeSWITCH-users mailing list
>         > FreeSWITCH-users at lists.freeswitch.org
>         >
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         >
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         > http://www.freeswitch.org
>         
>         
>         _______________________________________________
>         FreeSWITCH-users mailing list
>         FreeSWITCH-users at lists.freeswitch.org
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         http://www.freeswitch.org
>         
> 
> 
> 
> -- 
> Sincerely
> 
> Jay
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From yehavi.bourvine at gmail.com  Thu Feb 18 21:01:27 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Fri, 19 Feb 2010 07:01:27 +0200
Subject: [Freeswitch-users] Announcement: FreeSWITCH Add Broadsoft SCA
	Support
In-Reply-To: <ADC8B813-4161-481C-877E-93FC012858B4@freeswitch.org>
References: <87f2f3b91001111742k691966cfs82515db77c13d4d4@mail.gmail.com>
	<d361ca7b1002090838j6c42dcd2of8affb5f281900fd@mail.gmail.com>
	<767B677E-0D19-43A9-89DE-D96EC4A715EC@freeswitch.org>
	<8D062B6E-4691-4BA6-B038-0EC6E8642DDA@jerris.com>
	<9D328485-D4C3-4B14-808A-CB63BBF06DC1@freeswitch.org>
	<d361ca7b1002170549g608d6e16vb2eb3f087fff9752@mail.gmail.com>
	<191c3a031002170556p5dfd4c55le3c279f194808d92@mail.gmail.com>
	<191c3a031002170557g5d8c9df9xf40d1b74b9a6f19d@mail.gmail.com>
	<d361ca7b1002170700x15630ab2tabc6f3d91ebc3303@mail.gmail.com>
	<ADC8B813-4161-481C-877E-93FC012858B4@freeswitch.org>
Message-ID: <d361ca7b1002182101p60b4e8f8hf4dc95e1d2251e30@mail.gmail.com>

Thanks Brian. It now works better, but not fully (using 16659M).

What happens is:

   - When one of the Polycoms seize the line it is ok - the other phone gets
   notification and the extension status is "in use".
   - When one of the Polycom phones initiates a call - all is ok:
      - The other side sees that the extension is in use.
      - When it is put to hold all phones who share this extension see it
      and can pick the call.
   -  When a call arrives, both ring; the one that did not answer gets only
   a cancel mesage *without *any further notification that the extension is
   in use by the other phone.

                         Thanks! __Yehavi:

2010/2/17 Brian West <brian at freeswitch.org>

> Step 1. Enable manage-shared-appearance=true
>
> Step 2. Now in the phone's config Configure the phone as usually, set the
> line shared and DO NOT set the third party name.
>
> Step 3. Reboot
>
> It should work.
>
> I wish someone that has this working would write some wiki docs these
> threads about it not working are getting rather old when I know for a fact
> they work fine.
>
> The gateway info missing is a gateway you have configured getting a notify.
>  It has nothing to do with SCA.
>
> /b
>
> On Feb 17, 2010, at 9:00 AM, Yehavi Bourvine wrote:
>
> > .
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100219/cd5b84dc/attachment-0001.html 

From brian at freeswitch.org  Thu Feb 18 21:09:35 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 18 Feb 2010 23:09:35 -0600
Subject: [Freeswitch-users] Announcement: FreeSWITCH Add Broadsoft SCA
	Support
In-Reply-To: <d361ca7b1002182101p60b4e8f8hf4dc95e1d2251e30@mail.gmail.com>
References: <87f2f3b91001111742k691966cfs82515db77c13d4d4@mail.gmail.com>
	<d361ca7b1002090838j6c42dcd2of8affb5f281900fd@mail.gmail.com>
	<767B677E-0D19-43A9-89DE-D96EC4A715EC@freeswitch.org>
	<8D062B6E-4691-4BA6-B038-0EC6E8642DDA@jerris.com>
	<9D328485-D4C3-4B14-808A-CB63BBF06DC1@freeswitch.org>
	<d361ca7b1002170549g608d6e16vb2eb3f087fff9752@mail.gmail.com>
	<191c3a031002170556p5dfd4c55le3c279f194808d92@mail.gmail.com>
	<191c3a031002170557g5d8c9df9xf40d1b74b9a6f19d@mail.gmail.com>
	<d361ca7b1002170700x15630ab2tabc6f3d91ebc3303@mail.gmail.com>
	<ADC8B813-4161-481C-877E-93FC012858B4@freeswitch.org>
	<d361ca7b1002182101p60b4e8f8hf4dc95e1d2251e30@mail.gmail.com>
Message-ID: <C577F89F-4621-4DE9-A62B-BD40D1235538@freeswitch.org>

That last bit is wrong... I need sip traces.

/b

On Feb 18, 2010, at 11:01 PM, Yehavi Bourvine wrote:

> Thanks Brian. It now works better, but not fully (using 16659M).
>  
> What happens is:
> When one of the Polycoms seize the line it is ok - the other phone gets notification and the extension status is "in use".
> When one of the Polycom phones initiates a call - all is ok:
> The other side sees that the extension is in use.
> When it is put to hold all phones who share this extension see it and can pick the call.
>  When a call arrives, both ring; the one that did not answer gets only a cancel mesage without any further notification that the extension is in use by the other phone.
>                          Thanks! __Yehavi:

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100218/b2902689/attachment.html 

From yehavi.bourvine at gmail.com  Thu Feb 18 21:41:43 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Fri, 19 Feb 2010 07:41:43 +0200
Subject: [Freeswitch-users] SQL ERR: library routine called out of
	sequence
In-Reply-To: <191c3a031002161230h2b6e6f9wf3f65ddb71f08297@mail.gmail.com>
References: <d361ca7b1002160656w7d616f6dn5e80087ebde94dbe@mail.gmail.com>
	<191c3a031002160705w4050c3cdk2bf2922e9aff5ba3@mail.gmail.com>
	<d361ca7b1002160727ucb6e0c8id62e803757333213@mail.gmail.com>
	<191c3a031002160755q569fe2ednfefb01b6d9cf7f0e@mail.gmail.com>
	<d361ca7b1002161059w1ef9484dr87e69a33aa2b62fc@mail.gmail.com>
	<191c3a031002161110s637a48a2r717ae94557fe9e9a@mail.gmail.com>
	<d361ca7b1002161130k3eb2d88by3100b30a9d29bd51@mail.gmail.com>
	<191c3a031002161230h2b6e6f9wf3f65ddb71f08297@mail.gmail.com>
Message-ID: <d361ca7b1002182141p53e3884fs2a9993d4739a1f91@mail.gmail.com>

Unfortunately it did not help. I still get these error messages.

Just for the record, I have two systems:

   - Production one which is running Fedora-10 and exibits this problem.
   - Test system which is running Fedora-12 and does not exibit this
   phenomenon; however, there is almost zero traffic on this system.


I plan in upgrading the production system to Fedora-12, but it is not that
trivial...

                   Thanks, __Yehavi:

 2010/2/16 Anthony Minessale <anthony.minessale at gmail.com>

> Strange, even on abusive testing we have not seen this problem.
>
> please update to latest trunk.
> There was only one change I can think of that may cause your issue and I
> added a patch for it.
> If it persists try setting the sql-in-transactions profile param to false.
>
>
>
>
>
> On Tue, Feb 16, 2010 at 1:30 PM, Yehavi Bourvine <
> yehavi.bourvine at gmail.com> wrote:
>
>>  Most of the queries are ok, only some fail, thus it doesn't look like
>> permission problem. Furthermore, under 1.0.5pre10 it works for months.
>>
>> Might it be thread unsafe function calls? I've found the following while
>> searching the WEB:
>>
>> *According to the MSDN docs, System.Timers.Timer operates in a thread
>> pool.  If that's the case, your code is breaking the "connections cannot be
>> shared across threads" rule for SQLit*
>>
>> Although it quotes MSDN, it might be related to Linux as well.
>>
>>                     Thanks, __Yehavi:
>>
>>  2010/2/16 Anthony Minessale <anthony.minessale at gmail.com>
>>
>>> That sounds about right.
>>>
>>> That error usually has something to do with using db calls on a closed
>>> file or something along those lines.
>>> Maybe you have a permission problem on the directory where the db files
>>> are?
>>>
>>>
>>>
>>> On Tue, Feb 16, 2010 at 12:59 PM, Yehavi Bourvine <
>>> yehavi.bourvine at gmail.com> wrote:
>>>
>>>>  The OS is Fedora-10 (soon to be upgraded to 12).
>>>>
>>>> What I do when I want to test a new version:
>>>>
>>>>    - Download the latest one into a fresh directory
>>>>    - bootstrap.sh, configure and make
>>>>    - stop Freeswitch, delete everything in lib, mod, bin ,db
>>>>    - make install and run it.
>>>>
>>>>
>>>> Is there additional place to clean?
>>>>
>>>>                         Thanks! __Yehavi:
>>>>
>>>>  2010/2/16 Anthony Minessale <anthony.minessale at gmail.com>
>>>>
>>>>> you may want to do a clean wipe of all files related to FS then.
>>>>> you clearly have some problem with legacy something or other because we
>>>>> don't see that on dozens of dev boxes.
>>>>>
>>>>> What os is it?
>>>>>
>>>>>
>>>>>
>>>>> On Tue, Feb 16, 2010 at 9:27 AM, Yehavi Bourvine <
>>>>> yehavi.bourvine at gmail.com> wrote:
>>>>>
>>>>>>  Tried this, but it didn't help. I delete these DB files before any
>>>>>> upgrade just to be sure.
>>>>>>
>>>>>>                   Thanks! __Yehavi:
>>>>>>
>>>>>>  2010/2/16 Anthony Minessale <anthony.minessale at gmail.com>
>>>>>>
>>>>>>> try removing all the .db files from /usr/local/freeswitch/db
>>>>>>>
>>>>>>>
>>>>>>>   On Tue, Feb 16, 2010 at 8:56 AM, Yehavi Bourvine <
>>>>>>> yehavi.bourvine at gmail.com> wrote:
>>>>>>>
>>>>>>>>   Hello,
>>>>>>>>
>>>>>>>>   After upgrading from 1.0.5pre10 to 1.0.5-20100216-0400 we started
>>>>>>>> getting the above errors (I append bellow two samples). It seems Freeswitch
>>>>>>>> fails to read a database using Sqlite.
>>>>>>>> Anyone have seen this?
>>>>>>>>
>>>>>>>> Other details: Fedora 10, SQlite 3.5.9.
>>>>>>>> We also do SQLite quesries during call setup via LUA from CoreDB. Is
>>>>>>>> it an SQLite problem?
>>>>>>>>
>>>>>>>>                          Thanks! __Yehavi:
>>>>>>>>
>>>>>>>> The samples:
>>>>>>>> 2010-02-16 16:22:13.762679 [ERR] switch_core_sqldb.c:404 SQL ERR
>>>>>>>> [library routin
>>>>>>>> e called out of sequence]
>>>>>>>> delete from sip_dialogs where call_id='
>>>>>>>> 8656841832142-120172129116107 at 10.64.1.2'
>>>>>>>>
>>>>>>>> 2010-02-16 16:42:00.802442 [ERR] switch_core_sqldb.c:722 SQL ERR:
>>>>>>>> [select call_i
>>>>>>>>
>>>>>>>> d,sip_user,sip_host,contact,status,rpid,expires,user_agent,server_user,server_ho
>>>>>>>>
>>>>>>>> st,profile_name,hostname,network_ip,network_port,sip_username,sip_realm,mwi_user
>>>>>>>> ,mwi_host from sip_registrations where profile_name='phones' and
>>>>>>>> contact like '%
>>>>>>>> 80635%'] library routine called out of sequence
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> --
>>>>>>> Anthony Minessale II
>>>>>>>
>>>>>>> FreeSWITCH http://www.freeswitch.org/
>>>>>>> ClueCon http://www.cluecon.com/
>>>>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>>>>
>>>>>>> AIM: anthm
>>>>>>> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
>>>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>>>>> IRC: irc.freenode.net #freeswitch
>>>>>>>
>>>>>>> FreeSWITCH Developer Conference
>>>>>>> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>>>>>>> iax:guest at conference.freeswitch.org/888
>>>>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>>>>> pstn:+19193869900
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> Anthony Minessale II
>>>>>
>>>>> FreeSWITCH http://www.freeswitch.org/
>>>>> ClueCon http://www.cluecon.com/
>>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>>
>>>>> AIM: anthm
>>>>> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>>> IRC: irc.freenode.net #freeswitch
>>>>>
>>>>> FreeSWITCH Developer Conference
>>>>> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>>>>> iax:guest at conference.freeswitch.org/888
>>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>>> pstn:+19193869900
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>> iax:guest at conference.freeswitch.org/888
>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> pstn:+19193869900
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100219/3124d299/attachment-0001.html 

From gkuri at ieee.org  Thu Feb 18 21:44:31 2010
From: gkuri at ieee.org (Gabriel Kuri)
Date: Thu, 18 Feb 2010 21:44:31 -0800
Subject: [Freeswitch-users] Announcement: FreeSWITCH Add Broadsoft SCA
	Support
In-Reply-To: <d361ca7b1002182101p60b4e8f8hf4dc95e1d2251e30@mail.gmail.com>
References: <87f2f3b91001111742k691966cfs82515db77c13d4d4@mail.gmail.com>
	<767B677E-0D19-43A9-89DE-D96EC4A715EC@freeswitch.org>
	<8D062B6E-4691-4BA6-B038-0EC6E8642DDA@jerris.com>
	<9D328485-D4C3-4B14-808A-CB63BBF06DC1@freeswitch.org>
	<d361ca7b1002170549g608d6e16vb2eb3f087fff9752@mail.gmail.com>
	<191c3a031002170556p5dfd4c55le3c279f194808d92@mail.gmail.com>
	<191c3a031002170557g5d8c9df9xf40d1b74b9a6f19d@mail.gmail.com>
	<d361ca7b1002170700x15630ab2tabc6f3d91ebc3303@mail.gmail.com>
	<ADC8B813-4161-481C-877E-93FC012858B4@freeswitch.org>
	<d361ca7b1002182101p60b4e8f8hf4dc95e1d2251e30@mail.gmail.com>
Message-ID: <8b1c9cda1002182144m65ccf979w71384a1e7a55e2ea@mail.gmail.com>

>  When a call arrives, both ring; the one that did not answer gets only a
> cancel mesage without any further notification that the extension is in use
> by the other phone.

These are the same exact symptoms I posted about earlier this week,
with the Cisco SPA-5xx series phones. I still have yet to figure out
why this is happening, if you find out what's going on, please post
back the solution, I'd like to know the resolution.

Thanks,
Gabe



On Thu, Feb 18, 2010 at 9:01 PM, Yehavi Bourvine
<yehavi.bourvine at gmail.com> wrote:
> Thanks Brian. It now works better, but not fully (using 16659M).
>
> What happens is:
>
> When one of the Polycoms seize the line it is ok?- the other phone gets
> notification and the extension status is "in use".
> When?one of the Polycom phones initiates a call - all is ok:
>
> The other side sees that the extension is in use.
> When it is put to hold all phones?who share this extension see it and can
> pick the call.
>

>
> ???????????????????????? Thanks! __Yehavi:
>
> 2010/2/17 Brian West <brian at freeswitch.org>
>>
>> Step 1. Enable manage-shared-appearance=true
>>
>> Step 2. Now in the phone's config Configure the phone as usually, set the
>> line shared and DO NOT set the third party name.
>>
>> Step 3. Reboot
>>
>> It should work.
>>
>> I wish someone that has this working would write some wiki docs these
>> threads about it not working are getting rather old when I know for a fact
>> they work fine.
>>
>> The gateway info missing is a gateway you have configured getting a
>> notify. ?It has nothing to do with SCA.
>>
>> /b
>>
>> On Feb 17, 2010, at 9:00 AM, Yehavi Bourvine wrote:
>>
>> > .
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From yehavi.bourvine at gmail.com  Thu Feb 18 21:54:12 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Fri, 19 Feb 2010 07:54:12 +0200
Subject: [Freeswitch-users] Announcement: FreeSWITCH Add Broadsoft SCA
	Support
In-Reply-To: <8b1c9cda1002182144m65ccf979w71384a1e7a55e2ea@mail.gmail.com>
References: <87f2f3b91001111742k691966cfs82515db77c13d4d4@mail.gmail.com>
	<8D062B6E-4691-4BA6-B038-0EC6E8642DDA@jerris.com>
	<9D328485-D4C3-4B14-808A-CB63BBF06DC1@freeswitch.org>
	<d361ca7b1002170549g608d6e16vb2eb3f087fff9752@mail.gmail.com>
	<191c3a031002170556p5dfd4c55le3c279f194808d92@mail.gmail.com>
	<191c3a031002170557g5d8c9df9xf40d1b74b9a6f19d@mail.gmail.com>
	<d361ca7b1002170700x15630ab2tabc6f3d91ebc3303@mail.gmail.com>
	<ADC8B813-4161-481C-877E-93FC012858B4@freeswitch.org>
	<d361ca7b1002182101p60b4e8f8hf4dc95e1d2251e30@mail.gmail.com>
	<8b1c9cda1002182144m65ccf979w71384a1e7a55e2ea@mail.gmail.com>
Message-ID: <d361ca7b1002182154o52352c62h8c0ec02fa66e8444@mail.gmail.com>

Hello Gabe,

  As you can see - Brian is actively investigating it, so you can expect for
some fix soon...

                       Regards, __Yehavi:

2010/2/19 Gabriel Kuri <gkuri at ieee.org>

> >  When a call arrives, both ring; the one that did not answer gets only a
> > cancel mesage without any further notification that the extension is in
> use
> > by the other phone.
>
> These are the same exact symptoms I posted about earlier this week,
> with the Cisco SPA-5xx series phones. I still have yet to figure out
> why this is happening, if you find out what's going on, please post
> back the solution, I'd like to know the resolution.
>
> Thanks,
> Gabe
>
>
>
> On Thu, Feb 18, 2010 at 9:01 PM, Yehavi Bourvine
> <yehavi.bourvine at gmail.com> wrote:
> > Thanks Brian. It now works better, but not fully (using 16659M).
> >
> > What happens is:
> >
> > When one of the Polycoms seize the line it is ok - the other phone gets
> > notification and the extension status is "in use".
> > When one of the Polycom phones initiates a call - all is ok:
> >
> > The other side sees that the extension is in use.
> > When it is put to hold all phones who share this extension see it and can
> > pick the call.
> >
>
> >
>  >                          Thanks! __Yehavi:
> >
> > 2010/2/17 Brian West <brian at freeswitch.org>
> >>
> >> Step 1. Enable manage-shared-appearance=true
> >>
> >> Step 2. Now in the phone's config Configure the phone as usually, set
> the
> >> line shared and DO NOT set the third party name.
> >>
> >> Step 3. Reboot
> >>
> >> It should work.
> >>
> >> I wish someone that has this working would write some wiki docs these
> >> threads about it not working are getting rather old when I know for a
> fact
> >> they work fine.
> >>
> >> The gateway info missing is a gateway you have configured getting a
> >> notify.  It has nothing to do with SCA.
> >>
> >> /b
> >>
> >> On Feb 17, 2010, at 9:00 AM, Yehavi Bourvine wrote:
> >>
> >> > .
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100219/bec80318/attachment.html 

From pmhshz at gmail.com  Thu Feb 18 22:09:11 2010
From: pmhshz at gmail.com (MohammedShehzad)
Date: Fri, 19 Feb 2010 11:39:11 +0530
Subject: [Freeswitch-users] stream a file multicast with mod_esf
In-Reply-To: <85C74329-0D7C-48B2-B7A8-D4AA67C6E3F2@freeswitch.org>
References: <B98B6921358D0C49BD21CD023F1F52C302623B4B@hbghs04a.ww300.siemens.net>
	<B98B6921358D0C49BD21CD023F1F52C302623BE0@hbghs04a.ww300.siemens.net>
	<7C996136-963A-42D2-B0FE-D2729F968E52@freeswitch.org>
	<27071973.post@talk.nabble.com>
	<378CDEC3-5A41-4EF9-927A-311F2531E6AB@freeswitch.org>
	<a792304b1001080725y19ff28camd37425927550f311@mail.gmail.com>
	<191c3a031001081848w9036207yf6a4b473ec233243@mail.gmail.com>
	<85C74329-0D7C-48B2-B7A8-D4AA67C6E3F2@freeswitch.org>
Message-ID: <a792304b1002182209m43ada13cp8ed3d91ad509f0a2@mail.gmail.com>

Hello,
I am now looking into the code from few days to create the RTP Multicast
Listener first.
Is this something similar coding as mod_esf need to do here to listen on the
Multicast IP & Port? Or I need to use combination some rtp related function
of (switch_rtp.c) here?

I think this not proper place to discuss here, let me know where should I
continue further discussion.

-MohammedShehzad
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100219/c679aed0/attachment.html 

From anthony.minessale at gmail.com  Thu Feb 18 22:15:30 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 19 Feb 2010 00:15:30 -0600
Subject: [Freeswitch-users] SQL ERR: library routine called out of
	sequence
In-Reply-To: <191c3a031002182214h614c8c4aladed66943939a3bf@mail.gmail.com>
References: <d361ca7b1002160656w7d616f6dn5e80087ebde94dbe@mail.gmail.com>
	<191c3a031002160705w4050c3cdk2bf2922e9aff5ba3@mail.gmail.com>
	<d361ca7b1002160727ucb6e0c8id62e803757333213@mail.gmail.com>
	<191c3a031002160755q569fe2ednfefb01b6d9cf7f0e@mail.gmail.com>
	<d361ca7b1002161059w1ef9484dr87e69a33aa2b62fc@mail.gmail.com>
	<191c3a031002161110s637a48a2r717ae94557fe9e9a@mail.gmail.com>
	<d361ca7b1002161130k3eb2d88by3100b30a9d29bd51@mail.gmail.com>
	<191c3a031002161230h2b6e6f9wf3f65ddb71f08297@mail.gmail.com>
	<d361ca7b1002182141p53e3884fs2a9993d4739a1f91@mail.gmail.com>
	<191c3a031002182214h614c8c4aladed66943939a3bf@mail.gmail.com>
Message-ID: <191c3a031002182215t7e253924rca5de90e013ec49b@mail.gmail.com>

There is no evidence of this on any box I have used.
Do you have selinux on maybe.
The best I can do now is declare we do not support your current
configuration.

On Feb 18, 2010 11:48 PM, "Yehavi Bourvine" <yehavi.bourvine at gmail.com>
wrote:

Unfortunately it did not help. I still get these error messages.

Just for the record, I have two systems:

   - Production one which is running Fedora-10 and exibits this problem.
   - Test system which is running Fedora-12 and does not exibit this
   phenomenon; however, there is almost zero traffic on this system.


I plan in upgrading the production system to Fedora-12, but it is not that
trivial...



                   Thanks, __Yehavi:

2010/2/16 Anthony Minessale <anthony.minessale at gmail.com>...

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100219/e7179a62/attachment.html 

From lloyd.aloysius at gmail.com  Thu Feb 18 22:19:00 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Fri, 19 Feb 2010 01:19:00 -0500
Subject: [Freeswitch-users] IVR greeting - first two words missing
In-Reply-To: <87f2f3b91002181522te296581u12527a2a9cdf1d44@mail.gmail.com>
References: <8a19bf2e1002181410x44b40216j374a30a5675fd162@mail.gmail.com>
	<87f2f3b91002181522te296581u12527a2a9cdf1d44@mail.gmail.com>
Message-ID: <8a19bf2e1002182219i189664f2vfac02744331d5f7@mail.gmail.com>

Thank you Michael. It is now working perfect.

Lloyd


On Thu, Feb 18, 2010 at 6:22 PM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Thu, Feb 18, 2010 at 2:10 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:
>
>> Hi All,
>>
>> I setup a simple IVR. Here is the script.
>>
>> <menu name="test_ivr"
>>           greet-long="test/test-ivr.wav"
>>           greet-short="tset/test-ivr.wav"
>>           invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"
>>           exit-sound="voicemail/vm-goodbye.wav"
>>           timeout ="10000"
>>                   inter-digit-timeout="2000"
>>                   max-failures="3">
>>  <entry action="menu-exec-app" digits="/^(20[0-9])$/" param="transfer $1
>> XML default"/>
>>  <entry action="menu-exec-app" digits="9999" param="transfer 9999 XML
>> default"/>
>> </menu>
>>   </menus>
>> </configuration>
>>
>> *Dial Plan*
>>
>> <action application="answer"/>
>> <action application="ivr" data="test_ivr"/>
>>
>> Every time when I reach the IVR . I am getting first one or two words
>> missing( or may be not clear). How can I fix this issue.
>>
>> Thanks,
>> Lloyd
>>
>
> put a sleep after the answer:
> <action application="sleep" data="1000"/>
>
> You may have to tinker with the exact time, like maybe 1500 or 2000.
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100219/7932c2a5/attachment-0001.html 

From yehavi.bourvine at gmail.com  Thu Feb 18 22:19:23 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Fri, 19 Feb 2010 08:19:23 +0200
Subject: [Freeswitch-users] SQL ERR: library routine called out of
	sequence
In-Reply-To: <d361ca7b1002182141p53e3884fs2a9993d4739a1f91@mail.gmail.com>
References: <d361ca7b1002160656w7d616f6dn5e80087ebde94dbe@mail.gmail.com>
	<191c3a031002160705w4050c3cdk2bf2922e9aff5ba3@mail.gmail.com>
	<d361ca7b1002160727ucb6e0c8id62e803757333213@mail.gmail.com>
	<191c3a031002160755q569fe2ednfefb01b6d9cf7f0e@mail.gmail.com>
	<d361ca7b1002161059w1ef9484dr87e69a33aa2b62fc@mail.gmail.com>
	<191c3a031002161110s637a48a2r717ae94557fe9e9a@mail.gmail.com>
	<d361ca7b1002161130k3eb2d88by3100b30a9d29bd51@mail.gmail.com>
	<191c3a031002161230h2b6e6f9wf3f65ddb71f08297@mail.gmail.com>
	<d361ca7b1002182141p53e3884fs2a9993d4739a1f91@mail.gmail.com>
Message-ID: <d361ca7b1002182219k3a26ad6dx40c02538ccb2da1@mail.gmail.com>

Correction: It happens also on my test system (Fedora 12, Kernel 2.6.32.8,
Pentium-III, 1GHz).

                     Thanks, __Yehavi:

2010/2/19 Yehavi Bourvine <yehavi.bourvine at gmail.com>

>  Unfortunately it did not help. I still get these error messages.
>
> Just for the record, I have two systems:
>
>    - Production one which is running Fedora-10 and exibits this problem.
>    - Test system which is running Fedora-12 and does not exibit this
>    phenomenon; however, there is almost zero traffic on this system.
>
>
> I plan in upgrading the production system to Fedora-12, but it is not that
> trivial...
>
>                    Thanks, __Yehavi:
>
>  2010/2/16 Anthony Minessale <anthony.minessale at gmail.com>
>
>> Strange, even on abusive testing we have not seen this problem.
>>
>> please update to latest trunk.
>> There was only one change I can think of that may cause your issue and I
>> added a patch for it.
>> If it persists try setting the sql-in-transactions profile param to false.
>>
>>
>>
>>
>>
>>
>> On Tue, Feb 16, 2010 at 1:30 PM, Yehavi Bourvine <
>> yehavi.bourvine at gmail.com> wrote:
>>
>>>  Most of the queries are ok, only some fail, thus it doesn't look like
>>> permission problem. Furthermore, under 1.0.5pre10 it works for months.
>>>
>>> Might it be thread unsafe function calls? I've found the following while
>>> searching the WEB:
>>>
>>> *According to the MSDN docs, System.Timers.Timer operates in a thread
>>> pool.  If that's the case, your code is breaking the "connections cannot be
>>> shared across threads" rule for SQLit*
>>>
>>> Although it quotes MSDN, it might be related to Linux as well.
>>>
>>>                     Thanks, __Yehavi:
>>>
>>>  2010/2/16 Anthony Minessale <anthony.minessale at gmail.com>
>>>
>>>> That sounds about right.
>>>>
>>>> That error usually has something to do with using db calls on a closed
>>>> file or something along those lines.
>>>> Maybe you have a permission problem on the directory where the db files
>>>> are?
>>>>
>>>>
>>>>
>>>> On Tue, Feb 16, 2010 at 12:59 PM, Yehavi Bourvine <
>>>> yehavi.bourvine at gmail.com> wrote:
>>>>
>>>>>  The OS is Fedora-10 (soon to be upgraded to 12).
>>>>>
>>>>> What I do when I want to test a new version:
>>>>>
>>>>>    - Download the latest one into a fresh directory
>>>>>    - bootstrap.sh, configure and make
>>>>>    - stop Freeswitch, delete everything in lib, mod, bin ,db
>>>>>    - make install and run it.
>>>>>
>>>>>
>>>>> Is there additional place to clean?
>>>>>
>>>>>                         Thanks! __Yehavi:
>>>>>
>>>>>  2010/2/16 Anthony Minessale <anthony.minessale at gmail.com>
>>>>>
>>>>>> you may want to do a clean wipe of all files related to FS then.
>>>>>> you clearly have some problem with legacy something or other because
>>>>>> we don't see that on dozens of dev boxes.
>>>>>>
>>>>>> What os is it?
>>>>>>
>>>>>>
>>>>>>
>>>>>> On Tue, Feb 16, 2010 at 9:27 AM, Yehavi Bourvine <
>>>>>> yehavi.bourvine at gmail.com> wrote:
>>>>>>
>>>>>>>  Tried this, but it didn't help. I delete these DB files before any
>>>>>>> upgrade just to be sure.
>>>>>>>
>>>>>>>                   Thanks! __Yehavi:
>>>>>>>
>>>>>>>  2010/2/16 Anthony Minessale <anthony.minessale at gmail.com>
>>>>>>>
>>>>>>>> try removing all the .db files from /usr/local/freeswitch/db
>>>>>>>>
>>>>>>>>
>>>>>>>>   On Tue, Feb 16, 2010 at 8:56 AM, Yehavi Bourvine <
>>>>>>>> yehavi.bourvine at gmail.com> wrote:
>>>>>>>>
>>>>>>>>>   Hello,
>>>>>>>>>
>>>>>>>>>   After upgrading from 1.0.5pre10 to 1.0.5-20100216-0400 we started
>>>>>>>>> getting the above errors (I append bellow two samples). It seems Freeswitch
>>>>>>>>> fails to read a database using Sqlite.
>>>>>>>>> Anyone have seen this?
>>>>>>>>>
>>>>>>>>> Other details: Fedora 10, SQlite 3.5.9.
>>>>>>>>> We also do SQLite quesries during call setup via LUA from CoreDB.
>>>>>>>>> Is it an SQLite problem?
>>>>>>>>>
>>>>>>>>>                          Thanks! __Yehavi:
>>>>>>>>>
>>>>>>>>> The samples:
>>>>>>>>> 2010-02-16 16:22:13.762679 [ERR] switch_core_sqldb.c:404 SQL ERR
>>>>>>>>> [library routin
>>>>>>>>> e called out of sequence]
>>>>>>>>> delete from sip_dialogs where call_id='
>>>>>>>>> 8656841832142-120172129116107 at 10.64.1.2'
>>>>>>>>>
>>>>>>>>> 2010-02-16 16:42:00.802442 [ERR] switch_core_sqldb.c:722 SQL ERR:
>>>>>>>>> [select call_i
>>>>>>>>>
>>>>>>>>> d,sip_user,sip_host,contact,status,rpid,expires,user_agent,server_user,server_ho
>>>>>>>>>
>>>>>>>>> st,profile_name,hostname,network_ip,network_port,sip_username,sip_realm,mwi_user
>>>>>>>>> ,mwi_host from sip_registrations where profile_name='phones' and
>>>>>>>>> contact like '%
>>>>>>>>> 80635%'] library routine called out of sequence
>>>>>>>>>
>>>>>>>>> _______________________________________________
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>
>>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> --
>>>>>>>> Anthony Minessale II
>>>>>>>>
>>>>>>>> FreeSWITCH http://www.freeswitch.org/
>>>>>>>> ClueCon http://www.cluecon.com/
>>>>>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>>>>>
>>>>>>>> AIM: anthm
>>>>>>>> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
>>>>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>>>>>> IRC: irc.freenode.net #freeswitch
>>>>>>>>
>>>>>>>> FreeSWITCH Developer Conference
>>>>>>>> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>>>>>>>> iax:guest at conference.freeswitch.org/888
>>>>>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>>>>>> pstn:+19193869900
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>> Anthony Minessale II
>>>>>>
>>>>>> FreeSWITCH http://www.freeswitch.org/
>>>>>> ClueCon http://www.cluecon.com/
>>>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>>>
>>>>>> AIM: anthm
>>>>>> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
>>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>>>> IRC: irc.freenode.net #freeswitch
>>>>>>
>>>>>> FreeSWITCH Developer Conference
>>>>>> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>>>>>> iax:guest at conference.freeswitch.org/888
>>>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>>>> pstn:+19193869900
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>>
>>>> --
>>>> Anthony Minessale II
>>>>
>>>> FreeSWITCH http://www.freeswitch.org/
>>>> ClueCon http://www.cluecon.com/
>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>
>>>> AIM: anthm
>>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>> IRC: irc.freenode.net #freeswitch
>>>>
>>>> FreeSWITCH Developer Conference
>>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>>> iax:guest at conference.freeswitch.org/888
>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>> pstn:+19193869900
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100219/e6a6f60c/attachment-0001.html 

From anthony.minessale at gmail.com  Thu Feb 18 22:21:43 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 19 Feb 2010 00:21:43 -0600
Subject: [Freeswitch-users] voivemail quality
In-Reply-To: <191c3a031002182219l1dc9d55dj6bfb065b9edc0263@mail.gmail.com>
References: <27648642.post@talk.nabble.com>
	<b39273531002181801kd81a6b2x3327be4887563eae@mail.gmail.com>
	<1266552880.7684.17.camel@local.freepabx.com>
	<ec2ce17e1002182026m5986ab85t71006d608af73f88@mail.gmail.com>
	<1266555209.7684.23.camel@local.freepabx.com>
	<191c3a031002182219l1dc9d55dj6bfb065b9edc0263@mail.gmail.com>
Message-ID: <191c3a031002182221rf5fac8v76f611294b4881ad@mail.gmail.com>

Our g729 would have been done a lot sooner if we did not spend all our time
making sure there is no audio hiccups on a 600ms latent connection.  Yes we
made it possible and now we have g729 too nonetheless.  The only thing
howler filled is their wallet.  Let's see them as a gold sponsor of cluecon
if they are doing us such a big favor........

On Feb 18, 2010 10:58 PM, "David Knell" <dave at 3c.co.uk> wrote:


On Fri, 2010-02-19 at 14:26 +1000, jay binks wrote:
> I think suggesting the user test with other c...
Edward already said that things worked with G.711 - that ought to be
enough testing with other codecs.


> your attacking rupa for making suggestions about where the issue is
> and what to do in order to...
It doesn't, directly.  And, had Rupa simply asked if the OP had raised
the issue with Howler - which is a completely valid thing to do,
particularly with the evidence in front of him - that'd have been one
thing.  But he didn't.

> pot, kettle, black ???

How so, exactly?

--Dave


>
>
> J
>
> On Fri, Feb 19, 2010 at 2:14 PM, David Knell <dave at 3c.co.uk> wrote:
> Rupa -...
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100219/5316e654/attachment.html 

From anthony.minessale at gmail.com  Thu Feb 18 22:23:20 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 19 Feb 2010 00:23:20 -0600
Subject: [Freeswitch-users] 1.0.4 vs. trunk vs. bugs
In-Reply-To: <616F57B5-182A-49A9-9825-2E5B72F5DB05@freeswitch.org>
References: <5d3e0dc61002181741r57b91f86h4810de63c1a3d46d@mail.gmail.com>
	<20100219015118.GA12983@jdc.jasonjgw.net>
	<1266552605.7684.11.camel@local.freepabx.com>
	<616F57B5-182A-49A9-9825-2E5B72F5DB05@freeswitch.org>
Message-ID: <191c3a031002182223o66a16cdcjc5e43b637a5b91be@mail.gmail.com>

How many more ungrateful complaints can we get in one night.... sigh

On Feb 18, 2010 10:21 PM, "Brian West" <brian at freeswitch.org> wrote:

OK so I can sign you up for the stable team?  ;)  As per my previous email
i'm 100% sure we would do a stable release if we had people tending to
issues.  The only problem is you would have to be on IRC tending to issues
because if tony sees someone asking about a problem he'll be diving in to
fix it before they can say "I have this one".  This also means working in a
similar manner we do already.  Our process is very chaotic at times but it
has served us well so far.

The goal is to leave Anthony alone so he can move forward and let the stable
team manage the jira's and issues on the list related to stable.

/b



On Feb 18, 2010, at 10:10 PM, David Knell wrote:

>>
>> Lon Baker <lon at kickasspixels.com> wrote:
>...

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100219/632ca546/attachment.html 

From mike at jerris.com  Thu Feb 18 23:01:39 2010
From: mike at jerris.com (Michael Jerris)
Date: Fri, 19 Feb 2010 02:01:39 -0500
Subject: [Freeswitch-users] Presence PUBLISH Not Updating
	AfterSoftphoneOffLine Then Available
In-Reply-To: <E69C19EDE6F541198E27BE0883CB773D@greyhawk.tonecommander.com>
References: <31FEE9072F104AE18B26812978FCA341@greyhawk.tonecommander.com><45963EB5-DFE5-43FA-AFCC-857E1E719E2D@jerris.com><191c3a031002091557jfc9b486lb832ea538883bce9@mail.gmail.com><68A4C011A72B4ABA91A185F32480EB99@greyhawk.tonecommander.com>
	<191c3a031002162139p57c94046y660fb8d01fd76f46@mail.gmail.com>
	<E69C19EDE6F541198E27BE0883CB773D@greyhawk.tonecommander.com>
Message-ID: <8E59FC36-2CDB-4B77-8736-2319D883889F@jerris.com>



On Feb 18, 2010, at 3:26 PM, "Jerry Richards" <jerry.richards at teotech.com 
 > wrote:

> Yes, these are Bria (CounterPath) phones, but these are phones that  
> I'm using and they are popular, and as far as I know, faithful to  
> the SIP RFCs,

Hahahha

/me falls over laughing 
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100219/0196b4a8/attachment.html 

From mike at jerris.com  Thu Feb 18 23:10:39 2010
From: mike at jerris.com (Michael Jerris)
Date: Fri, 19 Feb 2010 02:10:39 -0500
Subject: [Freeswitch-users] building for Lenny
In-Reply-To: <3c5cf5261002181542y43adb7a7ofe2a15c38baf0327@mail.gmail.com>
References: <3c5cf5261002180046vb0015e7t787599c9903aedc@mail.gmail.com>
	<33c87fa31002180056g51ef8496qa9a35746d0fe75ec@mail.gmail.com>
	<3c5cf5261002180110he0ae815uc713f77f289ac625@mail.gmail.com>
	<a9f4977a70b3fcf1b30413ea562ca962.squirrel@www.webcontracts.co.uk>
	<20100218161741.GC4236@base.carmickle.com>
	<3c5cf5261002181542y43adb7a7ofe2a15c38baf0327@mail.gmail.com>
Message-ID: <4C13BCBF-1DA0-47AE-A749-738AD6D8FE48@jerris.com>

Please create me a bug on http://jira.freeswitch.org for this issue.

On Feb 18, 2010, at 6:42 PM, Brian May  
<brian at microcomaustralia.com.au> wrote:

> On 19 February 2010 03:17, Frank Carmickle <frank at carmickle.com>  
> wrote:
>> Like I said you can and should build debs from svn.  As far as
>> I see it there is no reason to not build debs.
>
> Unfortunately, that didn't create any of the packages for the sound
> files, and I can't see where to get a deb package for the sound files
> that really does contain the sound files.
>
>
>
> Also I get errors when trying to start it up, not sure how many of
> these I can ignore are warnings and how many are because I am doing it
> wrong:
>
> voyage:~# /opt/freeswitch/bin/freeswitch
> Error: stacksize 4194303 is too large: run ulimit -s 240 or run
> /opt/freeswitch/bin/freeswitch -waste.
> auto-adjusting stack size for optimal performance...
> 2010-02-19 10:59:41.203000 [INFO] switch_event.c:580 Activate  
> Eventing Engine.
> 2010-02-19 10:59:41.208000 [DEBUG] switch_event.c:568 Create event
> dispatch thread 0
> 2010-02-19 10:59:41.428000 [ERR] switch_xml.c:1297 Couldnt open
> /opt/freeswitch/conf/autoload_configs/../ivr_menus/*.xml (No such file
> or directory)
> Error including
> /opt/freeswitch/conf/autoload_configs/../ivr_menus/*.xml (No such file
> or directory)
> 2010-02-19 10:59:42.376000 [ERR] switch_xml.c:1297 Couldnt open
> /opt/freeswitch/conf/lang/de/*.xml (No such file or directory)
> Error including /opt/freeswitch/conf/lang/de/*.xml (No such file or  
> directory)
> 2010-02-19 10:59:42.526000 [ERR] switch_xml.c:1297 Couldnt open
> /opt/freeswitch/conf/lang/en/dir/sounds.xml (No such file or
> directory)
> Error including /opt/freeswitch/conf/lang/en/dir/sounds.xml (No such
> file or directory)
> 2010-02-19 10:59:42.528000 [ERR] switch_xml.c:1297 Couldnt open
> /opt/freeswitch/conf/lang/fr/*.xml (No such file or directory)
> Error including /opt/freeswitch/conf/lang/fr/*.xml (No such file or  
> directory)
> 2010-02-19 10:59:42.529000 [ERR] switch_xml.c:1297 Couldnt open
> /opt/freeswitch/conf/lang/ru/*.xml (No such file or directory)
> Error including /opt/freeswitch/conf/lang/ru/*.xml (No such file or  
> directory)
> 2010-02-19 10:59:42.609000 [INFO] switch_nat.c:409 Scanning for NAT
> 2010-02-19 10:59:42.611000 [DEBUG] switch_nat.c:166 Checking for PMP  
> 1/5
> 2010-02-19 10:59:42.614000 [ERR] switch_nat.c:197 Error checking for
> PMP [general error]
> 2010-02-19 10:59:42.614000 [DEBUG] switch_nat.c:414 Checking for UPnP
> 2010-02-19 10:59:54.619000 [INFO] switch_nat.c:429 No PMP or UPnP NAT
> devices detected!
> 2010-02-19 10:59:54.623000 [INFO] switch_core_sqldb.c:1248 Opening DB
> 2010-02-19 10:59:54.627000 [ERR] switch_core_sqldb.c:404 SQL ERR [no
> such table: channels]
> drop table channels
> 2010-02-19 10:59:54.628000 [ERR] switch_core_sqldb.c:404 SQL ERR [no
> such table: calls]
> drop table calls
> 2010-02-19 10:59:54.630000 [ERR] switch_core_sqldb.c:404 SQL ERR [no
> such table: interfaces]
> drop table interfaces
> 2010-02-19 10:59:54.631000 [ERR] switch_core_sqldb.c:404 SQL ERR [no
> such table: tasks]
> drop table tasks
> 2010-02-19 10:59:54.634000 [DEBUG] switch_core_sqldb.c:765 SQL ERR [no
> such table: aliases]
> [select hostname from aliases]
> Auto Generating Table!
> 2010-02-19 10:59:54.636000 [DEBUG] switch_core_sqldb.c:772 SQL ERR [no
> such table: aliases]
> [CREATE TABLE aliases (
>   sticky  INTEGER,
>   alias  VARCHAR(128),
>   command  VARCHAR(4096),
>   hostname VARCHAR(256)
> );
> ]
> 2010-02-19 10:59:54.644000 [DEBUG] switch_core_sqldb.c:765 SQL ERR [no
> such table: nat]
> [select hostname from nat]
> Auto Generating Table!
> 2010-02-19 10:59:54.646000 [DEBUG] switch_core_sqldb.c:772 SQL ERR [no
> such table: nat]
> [CREATE TABLE nat (
>   sticky  INTEGER,
>    port    INTEGER,
>    proto    INTEGER,
>   hostname VARCHAR(256)
> );
> ]
>
> Am I expected to setup a SQL database to get this working? Or did it
> just setup one automatically?
>
> 2010-02-19 10:59:59.684000 [CRIT] switch_loadable_module.c:882 Error
> Loading module /opt/freeswitch/mod/mod_voipcodecs.so
> **libjpeg.so.62: cannot open shared object file: No such file or  
> directory**
> 2010-02-19 10:59:59.686000 [CRIT] switch_loadable_module.c:882 Error
> Loading module /opt/freeswitch/mod/mod_g723_1.so
> **/opt/freeswitch/mod/mod_g723_1.so: cannot open shared object file:
> No such file or directory**
> 2010-02-19 10:59:59.687000 [CRIT] switch_loadable_module.c:882 Error
> Loading module /opt/freeswitch/mod/mod_g729.so
> **/opt/freeswitch/mod/mod_g729.so: cannot open shared object file: No
> such file or directory**
> 2010-02-19 10:59:59.688000 [CRIT] switch_loadable_module.c:882 Error
> Loading module /opt/freeswitch/mod/mod_amr.so
> **/opt/freeswitch/mod/mod_amr.so: cannot open shared object file: No
> such file or directory**
> 2010-02-19 10:59:59.835000 [CRIT] switch_loadable_module.c:882 Error
> Loading module /opt/freeswitch/mod/mod_file_string.so
> **/opt/freeswitch/mod/mod_file_string.so: cannot open shared object
> file: No such file or directory**
> 2010-02-19 11:00:00.143000 [CRIT] switch_loadable_module.c:882 Error
> Loading module /opt/freeswitch/mod/mod_say_ru.so
> **/opt/freeswitch/mod/mod_say_ru.so: cannot open shared object file:
> No such file or directory**
>
> suspect I don't really need to worry about some of these. I assume
> there is a config file somewhere where I can disable these options.
>
>
> Ok, as a really pathetic question, now I have started it, how do I  
> stop it?
>
> freeswitch at voyage> halt
> Unknown Command: halt
> freeswitch at voyage> quit
> Unknown Command: quit
> freeswitch at voyage> exit
> Unknown Command: exit
> freeswitch at voyage> bye
> Unknown Command: bye
>
> -- 
> Brian May <brian at microcomaustralia.com.au>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org


From mrene_lists at avgs.ca  Thu Feb 18 23:14:11 2010
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Fri, 19 Feb 2010 02:14:11 -0500
Subject: [Freeswitch-users] building for Lenny
In-Reply-To: <4C13BCBF-1DA0-47AE-A749-738AD6D8FE48@jerris.com>
References: <3c5cf5261002180046vb0015e7t787599c9903aedc@mail.gmail.com>
	<33c87fa31002180056g51ef8496qa9a35746d0fe75ec@mail.gmail.com>
	<3c5cf5261002180110he0ae815uc713f77f289ac625@mail.gmail.com>
	<a9f4977a70b3fcf1b30413ea562ca962.squirrel@www.webcontracts.co.uk>
	<20100218161741.GC4236@base.carmickle.com>
	<3c5cf5261002181542y43adb7a7ofe2a15c38baf0327@mail.gmail.com>
	<4C13BCBF-1DA0-47AE-A749-738AD6D8FE48@jerris.com>
Message-ID: <155182E9-FD25-4BE2-A5EE-389E3D64B9DC@avgs.ca>

You can stop it with "..." or "fsctl shutdown"

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 19-Feb-10, at 2:10 AM, Michael Jerris wrote:

> Please create me a bug on http://jira.freeswitch.org for this issue.
>
> On Feb 18, 2010, at 6:42 PM, Brian May
> <brian at microcomaustralia.com.au> wrote:
>
>> On 19 February 2010 03:17, Frank Carmickle <frank at carmickle.com>
>> wrote:
>>> Like I said you can and should build debs from svn.  As far as
>>> I see it there is no reason to not build debs.
>>
>> Unfortunately, that didn't create any of the packages for the sound
>> files, and I can't see where to get a deb package for the sound files
>> that really does contain the sound files.
>>
>>
>>
>> Also I get errors when trying to start it up, not sure how many of
>> these I can ignore are warnings and how many are because I am doing  
>> it
>> wrong:
>>
>> voyage:~# /opt/freeswitch/bin/freeswitch
>> Error: stacksize 4194303 is too large: run ulimit -s 240 or run
>> /opt/freeswitch/bin/freeswitch -waste.
>> auto-adjusting stack size for optimal performance...
>> 2010-02-19 10:59:41.203000 [INFO] switch_event.c:580 Activate
>> Eventing Engine.
>> 2010-02-19 10:59:41.208000 [DEBUG] switch_event.c:568 Create event
>> dispatch thread 0
>> 2010-02-19 10:59:41.428000 [ERR] switch_xml.c:1297 Couldnt open
>> /opt/freeswitch/conf/autoload_configs/../ivr_menus/*.xml (No such  
>> file
>> or directory)
>> Error including
>> /opt/freeswitch/conf/autoload_configs/../ivr_menus/*.xml (No such  
>> file
>> or directory)
>> 2010-02-19 10:59:42.376000 [ERR] switch_xml.c:1297 Couldnt open
>> /opt/freeswitch/conf/lang/de/*.xml (No such file or directory)
>> Error including /opt/freeswitch/conf/lang/de/*.xml (No such file or
>> directory)
>> 2010-02-19 10:59:42.526000 [ERR] switch_xml.c:1297 Couldnt open
>> /opt/freeswitch/conf/lang/en/dir/sounds.xml (No such file or
>> directory)
>> Error including /opt/freeswitch/conf/lang/en/dir/sounds.xml (No such
>> file or directory)
>> 2010-02-19 10:59:42.528000 [ERR] switch_xml.c:1297 Couldnt open
>> /opt/freeswitch/conf/lang/fr/*.xml (No such file or directory)
>> Error including /opt/freeswitch/conf/lang/fr/*.xml (No such file or
>> directory)
>> 2010-02-19 10:59:42.529000 [ERR] switch_xml.c:1297 Couldnt open
>> /opt/freeswitch/conf/lang/ru/*.xml (No such file or directory)
>> Error including /opt/freeswitch/conf/lang/ru/*.xml (No such file or
>> directory)
>> 2010-02-19 10:59:42.609000 [INFO] switch_nat.c:409 Scanning for NAT
>> 2010-02-19 10:59:42.611000 [DEBUG] switch_nat.c:166 Checking for PMP
>> 1/5
>> 2010-02-19 10:59:42.614000 [ERR] switch_nat.c:197 Error checking for
>> PMP [general error]
>> 2010-02-19 10:59:42.614000 [DEBUG] switch_nat.c:414 Checking for UPnP
>> 2010-02-19 10:59:54.619000 [INFO] switch_nat.c:429 No PMP or UPnP NAT
>> devices detected!
>> 2010-02-19 10:59:54.623000 [INFO] switch_core_sqldb.c:1248 Opening DB
>> 2010-02-19 10:59:54.627000 [ERR] switch_core_sqldb.c:404 SQL ERR [no
>> such table: channels]
>> drop table channels
>> 2010-02-19 10:59:54.628000 [ERR] switch_core_sqldb.c:404 SQL ERR [no
>> such table: calls]
>> drop table calls
>> 2010-02-19 10:59:54.630000 [ERR] switch_core_sqldb.c:404 SQL ERR [no
>> such table: interfaces]
>> drop table interfaces
>> 2010-02-19 10:59:54.631000 [ERR] switch_core_sqldb.c:404 SQL ERR [no
>> such table: tasks]
>> drop table tasks
>> 2010-02-19 10:59:54.634000 [DEBUG] switch_core_sqldb.c:765 SQL ERR  
>> [no
>> such table: aliases]
>> [select hostname from aliases]
>> Auto Generating Table!
>> 2010-02-19 10:59:54.636000 [DEBUG] switch_core_sqldb.c:772 SQL ERR  
>> [no
>> such table: aliases]
>> [CREATE TABLE aliases (
>>  sticky  INTEGER,
>>  alias  VARCHAR(128),
>>  command  VARCHAR(4096),
>>  hostname VARCHAR(256)
>> );
>> ]
>> 2010-02-19 10:59:54.644000 [DEBUG] switch_core_sqldb.c:765 SQL ERR  
>> [no
>> such table: nat]
>> [select hostname from nat]
>> Auto Generating Table!
>> 2010-02-19 10:59:54.646000 [DEBUG] switch_core_sqldb.c:772 SQL ERR  
>> [no
>> such table: nat]
>> [CREATE TABLE nat (
>>  sticky  INTEGER,
>>   port    INTEGER,
>>   proto    INTEGER,
>>  hostname VARCHAR(256)
>> );
>> ]
>>
>> Am I expected to setup a SQL database to get this working? Or did it
>> just setup one automatically?
>>
>> 2010-02-19 10:59:59.684000 [CRIT] switch_loadable_module.c:882 Error
>> Loading module /opt/freeswitch/mod/mod_voipcodecs.so
>> **libjpeg.so.62: cannot open shared object file: No such file or
>> directory**
>> 2010-02-19 10:59:59.686000 [CRIT] switch_loadable_module.c:882 Error
>> Loading module /opt/freeswitch/mod/mod_g723_1.so
>> **/opt/freeswitch/mod/mod_g723_1.so: cannot open shared object file:
>> No such file or directory**
>> 2010-02-19 10:59:59.687000 [CRIT] switch_loadable_module.c:882 Error
>> Loading module /opt/freeswitch/mod/mod_g729.so
>> **/opt/freeswitch/mod/mod_g729.so: cannot open shared object file: No
>> such file or directory**
>> 2010-02-19 10:59:59.688000 [CRIT] switch_loadable_module.c:882 Error
>> Loading module /opt/freeswitch/mod/mod_amr.so
>> **/opt/freeswitch/mod/mod_amr.so: cannot open shared object file: No
>> such file or directory**
>> 2010-02-19 10:59:59.835000 [CRIT] switch_loadable_module.c:882 Error
>> Loading module /opt/freeswitch/mod/mod_file_string.so
>> **/opt/freeswitch/mod/mod_file_string.so: cannot open shared object
>> file: No such file or directory**
>> 2010-02-19 11:00:00.143000 [CRIT] switch_loadable_module.c:882 Error
>> Loading module /opt/freeswitch/mod/mod_say_ru.so
>> **/opt/freeswitch/mod/mod_say_ru.so: cannot open shared object file:
>> No such file or directory**
>>
>> suspect I don't really need to worry about some of these. I assume
>> there is a config file somewhere where I can disable these options.
>>
>>
>> Ok, as a really pathetic question, now I have started it, how do I
>> stop it?
>>
>> freeswitch at voyage> halt
>> Unknown Command: halt
>> freeswitch at voyage> quit
>> Unknown Command: quit
>> freeswitch at voyage> exit
>> Unknown Command: exit
>> freeswitch at voyage> bye
>> Unknown Command: bye
>>
>> -- 
>> Brian May <brian at microcomaustralia.com.au>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>> users
>> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From mike at jerris.com  Thu Feb 18 23:17:52 2010
From: mike at jerris.com (Michael Jerris)
Date: Fri, 19 Feb 2010 02:17:52 -0500
Subject: [Freeswitch-users] 1.0.4 vs. trunk vs. bugs
In-Reply-To: <616F57B5-182A-49A9-9825-2E5B72F5DB05@freeswitch.org>
References: <5d3e0dc61002181741r57b91f86h4810de63c1a3d46d@mail.gmail.com>
	<20100219015118.GA12983@jdc.jasonjgw.net>
	<1266552605.7684.11.camel@local.freepabx.com>
	<616F57B5-182A-49A9-9825-2E5B72F5DB05@freeswitch.org>
Message-ID: <7716C919-BCD1-44FA-9ED2-5268F4C42100@jerris.com>

This seems a good time to note that we are still looking for  
volunteers to assist in maintaining a stable branch.  I can not do  
this without additional volunteer resources.  We have asked several  
times recently to fairly silent response.  If anyone is interested in  
assisting with this effort, please contact  me offlist and we can  
discuss further.

Mike

On Feb 18, 2010, at 11:16 PM, Brian West <brian at freeswitch.org> wrote:

> OK so I can sign you up for the stable team?  ;)  As per my previous  
> email i'm 100% sure we would do a stable release if we had people  
> tending to issues.  The only problem is you would have to be on IRC  
> tending to issues because if tony sees someone asking about a  
> problem he'll be diving in to fix it before they can say "I have  
> this one".  This also means working in a similar manner we do  
> already.  Our process is very chaotic at times but it has served us  
> well so far.
>
> The goal is to leave Anthony alone so he can move forward and let  
> the stable team manage the jira's and issues on the list related to  
> stable.
>
> /b
>
> On Feb 18, 2010, at 10:10 PM, David Knell wrote:
>
>>>
>>> Lon Baker <lon at kickasspixels.com> wrote:
>>>
>>>> The development branch is where feature requests and non-critical  
>>>> bugs
>>>> reports would be filed for the next production release.
>>>>
>>>> The current process leaves a gap between production ready and
>>>> development code that may become greater over time.
>>
>> Going against the grain here, I agree with you.  The current way of
>> doing things is, in my opinion, not well thought through - there's no
>> reason to tag and release versions if the answer to any issue is  
>> 'make
>> current', and support is not available unless that's been done.  Far
>> better to either have meaningful releases with stable and devel
>> branches, or not to have releases at all.
>>
>> --Dave
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100219/c53d7751/attachment.html 

From mike at jerris.com  Thu Feb 18 23:22:26 2010
From: mike at jerris.com (Michael Jerris)
Date: Fri, 19 Feb 2010 02:22:26 -0500
Subject: [Freeswitch-users] mod_fax undefined symbol
In-Reply-To: <6FCE44FF-70C1-412D-B722-4C8D359381BA@ttnc.co.uk>
References: <6FCE44FF-70C1-412D-B722-4C8D359381BA@ttnc.co.uk>
Message-ID: <0F7476C7-FA7E-441F-9E17-EB2D23CD6055@jerris.com>

Please open a bug on http://jira.freeswitch.org for this issue.  As a  
note, we use our own copy of libtiff, statically linked to the  
module.  Is this recent trunk or something older?

On Feb 18, 2010, at 8:57 PM, TTNC - Technical <technical at ttnc.co.uk>  
wrote:

> Hi Guys
>
> I'm having trouble getting mod_fax to load. Running on Debian  
> testing (squeeze). Everything is installed as per - http://wiki.freeswitch.org/wiki/Installation_Guide 
> . (dpkg-buildpackage etc)
>
> When trying to load the fax module I get:
>
> 2010-02-19 01:48:17.554935 [CRIT] switch_loadable_module.c:882 Error  
> Loading module /opt/freeswitch/mod/mod_fax.so
> **/opt/freeswitch/mod/mod_fax.so: undefined symbol:  
> TIFFDefaultStripSize**
>
> And when a fax is sent, I'm getting:
>
> 2010-02-19 01:04:57.355330 [ERR] switch_core_session.c:1490 Invalid  
> Application rxfax
>
> I guess because mod_fax isn't loaded.
>
> I've got libtiff4 and libtiff4-dev installed:
>
> ii  libtiff4                             3.9.2-2                   
> Tag Image File Format (TIFF) library
> ii  libtiff4-dev                         3.9.2-2                   
> Tag Image File Format library (TIFF), development files
> ii  libtiffxx0c2                         3.9.2-2                   
> Tag Image File Format (TIFF) library -- C++ interface
>
> Just tried updating to the latest svn trunk (16700M) and it hasn't  
> made any difference.
>
>> From googling, it suggests that it could be because the module is  
>> complied against a different one currently running on the system,  
>> however I'm not sure how this can be the case, there is only the  
>> one version installed.
>
> Any suggestions as to what I can try?
>
> Any help appreciated
>
> Russ
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org


From mike at jerris.com  Thu Feb 18 23:24:37 2010
From: mike at jerris.com (Michael Jerris)
Date: Fri, 19 Feb 2010 02:24:37 -0500
Subject: [Freeswitch-users] Announcement: FreeSWITCH Add Broadsoft SCA
	Support
In-Reply-To: <d361ca7b1002182154o52352c62h8c0ec02fa66e8444@mail.gmail.com>
References: <87f2f3b91001111742k691966cfs82515db77c13d4d4@mail.gmail.com>
	<8D062B6E-4691-4BA6-B038-0EC6E8642DDA@jerris.com>
	<9D328485-D4C3-4B14-808A-CB63BBF06DC1@freeswitch.org>
	<d361ca7b1002170549g608d6e16vb2eb3f087fff9752@mail.gmail.com>
	<191c3a031002170556p5dfd4c55le3c279f194808d92@mail.gmail.com>
	<191c3a031002170557g5d8c9df9xf40d1b74b9a6f19d@mail.gmail.com>
	<d361ca7b1002170700x15630ab2tabc6f3d91ebc3303@mail.gmail.com>
	<ADC8B813-4161-481C-877E-93FC012858B4@freeswitch.org>
	<d361ca7b1002182101p60b4e8f8hf4dc95e1d2251e30@mail.gmail.com>
	<8b1c9cda1002182144m65ccf979w71384a1e7a55e2ea@mail.gmail.com>
	<d361ca7b1002182154o52352c62h8c0ec02fa66e8444@mail.gmail.com>
Message-ID: <2F87B95C-EA23-4072-B845-3F7AE0617BE3@jerris.com>

If this issue is not already on jira could you please make sure it  
gets added?

Mike

On Feb 19, 2010, at 12:54 AM, Yehavi Bourvine  
<yehavi.bourvine at gmail.com> wrote:

> Hello Gabe,
>
>   As you can see - Brian is actively investigating it, so you can  
> expect for some fix soon...
>
>                        Regards, __Yehavi:
>
> 2010/2/19 Gabriel Kuri <gkuri at ieee.org>
> >  When a call arrives, both ring; the one that did not answer gets  
> only a
> > cancel mesage without any further notification that the extension  
> is in use
> > by the other phone.
>
> These are the same exact symptoms I posted about earlier this week,
> with the Cisco SPA-5xx series phones. I still have yet to figure out
> why this is happening, if you find out what's going on, please post
> back the solution, I'd like to know the resolution.
>
> Thanks,
> Gabe
>
>
>
> On Thu, Feb 18, 2010 at 9:01 PM, Yehavi Bourvine
> <yehavi.bourvine at gmail.com> wrote:
> > Thanks Brian. It now works better, but not fully (using 16659M).
> >
> > What happens is:
> >
> > When one of the Polycoms seize the line it is ok - the other phone  
> gets
> > notification and the extension status is "in use".
> > When one of the Polycom phones initiates a call - all is ok:
> >
> > The other side sees that the extension is in use.
> > When it is put to hold all phones who share this extension see it  
> and can
> > pick the call.
> >
>
> >
> >                          Thanks! __Yehavi:
> >
> > 2010/2/17 Brian West <brian at freeswitch.org>
> >>
> >> Step 1. Enable manage-shared-appearance=true
> >>
> >> Step 2. Now in the phone's config Configure the phone as usually,  
> set the
> >> line shared and DO NOT set the third party name.
> >>
> >> Step 3. Reboot
> >>
> >> It should work.
> >>
> >> I wish someone that has this working would write some wiki docs  
> these
> >> threads about it not working are getting rather old when I know  
> for a fact
> >> they work fine.
> >>
> >> The gateway info missing is a gateway you have configured getting a
> >> notify.  It has nothing to do with SCA.
> >>
> >> /b
> >>
> >> On Feb 17, 2010, at 9:00 AM, Yehavi Bourvine wrote:
> >>
> >> > .
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/ 
> freeswitch-users
> >> http://www.freeswitch.org
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100219/4c863452/attachment.html 

From mike at jerris.com  Thu Feb 18 23:29:18 2010
From: mike at jerris.com (Michael Jerris)
Date: Fri, 19 Feb 2010 02:29:18 -0500
Subject: [Freeswitch-users] stream a file multicast with mod_esf
In-Reply-To: <a792304b1002182209m43ada13cp8ed3d91ad509f0a2@mail.gmail.com>
References: <B98B6921358D0C49BD21CD023F1F52C302623B4B@hbghs04a.ww300.siemens.net>
	<B98B6921358D0C49BD21CD023F1F52C302623BE0@hbghs04a.ww300.siemens.net>
	<7C996136-963A-42D2-B0FE-D2729F968E52@freeswitch.org>
	<27071973.post@talk.nabble.com>
	<378CDEC3-5A41-4EF9-927A-311F2531E6AB@freeswitch.org>
	<a792304b1001080725y19ff28camd37425927550f311@mail.gmail.com>
	<191c3a031001081848w9036207yf6a4b473ec233243@mail.gmail.com>
	<85C74329-0D7C-48B2-B7A8-D4AA67C6E3F2@freeswitch.org>
	<a792304b1002182209m43ada13cp8ed3d91ad509f0a2@mail.gmail.com>
Message-ID: <DBA4E291-9377-489A-841A-1A3D83880DF2@jerris.com>

Listening on multicast is noting special for multicast, it is just  
like reading any other udp socket

Mike

On Feb 19, 2010, at 1:09 AM, MohammedShehzad <pmhshz at gmail.com> wrote:

> Hello,
> I am now looking into the code from few days to create the RTP  
> Multicast Listener first.
> Is this something similar coding as mod_esf need to do here to  
> listen on the Multicast IP & Port? Or I need to use combination some  
> rtp related function of (switch_rtp.c) here?
>
> I think this not proper place to discuss here, let me know where  
> should I continue further discussion.
>
> -MohammedShehzad
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org


From mike at jerris.com  Thu Feb 18 23:30:56 2010
From: mike at jerris.com (Michael Jerris)
Date: Fri, 19 Feb 2010 02:30:56 -0500
Subject: [Freeswitch-users] building for Lenny
In-Reply-To: <155182E9-FD25-4BE2-A5EE-389E3D64B9DC@avgs.ca>
References: <3c5cf5261002180046vb0015e7t787599c9903aedc@mail.gmail.com>
	<33c87fa31002180056g51ef8496qa9a35746d0fe75ec@mail.gmail.com>
	<3c5cf5261002180110he0ae815uc713f77f289ac625@mail.gmail.com>
	<a9f4977a70b3fcf1b30413ea562ca962.squirrel@www.webcontracts.co.uk>
	<20100218161741.GC4236@base.carmickle.com>
	<3c5cf5261002181542y43adb7a7ofe2a15c38baf0327@mail.gmail.com>
	<4C13BCBF-1DA0-47AE-A749-738AD6D8FE48@jerris.com>
	<155182E9-FD25-4BE2-A5EE-389E3D64B9DC@avgs.ca>
Message-ID: <D35A6E61-BBBF-49F5-9EF1-FAFCECCF679A@jerris.com>

There is also a "help" command

Mike

On Feb 19, 2010, at 2:14 AM, Mathieu Rene <mrene_lists at avgs.ca> wrote:

> You can stop it with "..." or "fsctl shutdown"
>
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>
>
>
>
> On 19-Feb-10, at 2:10 AM, Michael Jerris wrote:
>
>> Please create me a bug on http://jira.freeswitch.org for this issue.
>>
>> On Feb 18, 2010, at 6:42 PM, Brian May
>> <brian at microcomaustralia.com.au> wrote:
>>
>>> On 19 February 2010 03:17, Frank Carmickle <frank at carmickle.com>
>>> wrote:
>>>> Like I said you can and should build debs from svn.  As far as
>>>> I see it there is no reason to not build debs.
>>>
>>> Unfortunately, that didn't create any of the packages for the sound
>>> files, and I can't see where to get a deb package for the sound  
>>> files
>>> that really does contain the sound files.
>>>
>>>
>>>
>>> Also I get errors when trying to start it up, not sure how many of
>>> these I can ignore are warnings and how many are because I am doing
>>> it
>>> wrong:
>>>
>>> voyage:~# /opt/freeswitch/bin/freeswitch
>>> Error: stacksize 4194303 is too large: run ulimit -s 240 or run
>>> /opt/freeswitch/bin/freeswitch -waste.
>>> auto-adjusting stack size for optimal performance...
>>> 2010-02-19 10:59:41.203000 [INFO] switch_event.c:580 Activate
>>> Eventing Engine.
>>> 2010-02-19 10:59:41.208000 [DEBUG] switch_event.c:568 Create event
>>> dispatch thread 0
>>> 2010-02-19 10:59:41.428000 [ERR] switch_xml.c:1297 Couldnt open
>>> /opt/freeswitch/conf/autoload_configs/../ivr_menus/*.xml (No such
>>> file
>>> or directory)
>>> Error including
>>> /opt/freeswitch/conf/autoload_configs/../ivr_menus/*.xml (No such
>>> file
>>> or directory)
>>> 2010-02-19 10:59:42.376000 [ERR] switch_xml.c:1297 Couldnt open
>>> /opt/freeswitch/conf/lang/de/*.xml (No such file or directory)
>>> Error including /opt/freeswitch/conf/lang/de/*.xml (No such file or
>>> directory)
>>> 2010-02-19 10:59:42.526000 [ERR] switch_xml.c:1297 Couldnt open
>>> /opt/freeswitch/conf/lang/en/dir/sounds.xml (No such file or
>>> directory)
>>> Error including /opt/freeswitch/conf/lang/en/dir/sounds.xml (No such
>>> file or directory)
>>> 2010-02-19 10:59:42.528000 [ERR] switch_xml.c:1297 Couldnt open
>>> /opt/freeswitch/conf/lang/fr/*.xml (No such file or directory)
>>> Error including /opt/freeswitch/conf/lang/fr/*.xml (No such file or
>>> directory)
>>> 2010-02-19 10:59:42.529000 [ERR] switch_xml.c:1297 Couldnt open
>>> /opt/freeswitch/conf/lang/ru/*.xml (No such file or directory)
>>> Error including /opt/freeswitch/conf/lang/ru/*.xml (No such file or
>>> directory)
>>> 2010-02-19 10:59:42.609000 [INFO] switch_nat.c:409 Scanning for NAT
>>> 2010-02-19 10:59:42.611000 [DEBUG] switch_nat.c:166 Checking for PMP
>>> 1/5
>>> 2010-02-19 10:59:42.614000 [ERR] switch_nat.c:197 Error checking for
>>> PMP [general error]
>>> 2010-02-19 10:59:42.614000 [DEBUG] switch_nat.c:414 Checking for  
>>> UPnP
>>> 2010-02-19 10:59:54.619000 [INFO] switch_nat.c:429 No PMP or UPnP  
>>> NAT
>>> devices detected!
>>> 2010-02-19 10:59:54.623000 [INFO] switch_core_sqldb.c:1248 Opening  
>>> DB
>>> 2010-02-19 10:59:54.627000 [ERR] switch_core_sqldb.c:404 SQL ERR [no
>>> such table: channels]
>>> drop table channels
>>> 2010-02-19 10:59:54.628000 [ERR] switch_core_sqldb.c:404 SQL ERR [no
>>> such table: calls]
>>> drop table calls
>>> 2010-02-19 10:59:54.630000 [ERR] switch_core_sqldb.c:404 SQL ERR [no
>>> such table: interfaces]
>>> drop table interfaces
>>> 2010-02-19 10:59:54.631000 [ERR] switch_core_sqldb.c:404 SQL ERR [no
>>> such table: tasks]
>>> drop table tasks
>>> 2010-02-19 10:59:54.634000 [DEBUG] switch_core_sqldb.c:765 SQL ERR
>>> [no
>>> such table: aliases]
>>> [select hostname from aliases]
>>> Auto Generating Table!
>>> 2010-02-19 10:59:54.636000 [DEBUG] switch_core_sqldb.c:772 SQL ERR
>>> [no
>>> such table: aliases]
>>> [CREATE TABLE aliases (
>>> sticky  INTEGER,
>>> alias  VARCHAR(128),
>>> command  VARCHAR(4096),
>>> hostname VARCHAR(256)
>>> );
>>> ]
>>> 2010-02-19 10:59:54.644000 [DEBUG] switch_core_sqldb.c:765 SQL ERR
>>> [no
>>> such table: nat]
>>> [select hostname from nat]
>>> Auto Generating Table!
>>> 2010-02-19 10:59:54.646000 [DEBUG] switch_core_sqldb.c:772 SQL ERR
>>> [no
>>> such table: nat]
>>> [CREATE TABLE nat (
>>> sticky  INTEGER,
>>>  port    INTEGER,
>>>  proto    INTEGER,
>>> hostname VARCHAR(256)
>>> );
>>> ]
>>>
>>> Am I expected to setup a SQL database to get this working? Or did it
>>> just setup one automatically?
>>>
>>> 2010-02-19 10:59:59.684000 [CRIT] switch_loadable_module.c:882 Error
>>> Loading module /opt/freeswitch/mod/mod_voipcodecs.so
>>> **libjpeg.so.62: cannot open shared object file: No such file or
>>> directory**
>>> 2010-02-19 10:59:59.686000 [CRIT] switch_loadable_module.c:882 Error
>>> Loading module /opt/freeswitch/mod/mod_g723_1.so
>>> **/opt/freeswitch/mod/mod_g723_1.so: cannot open shared object file:
>>> No such file or directory**
>>> 2010-02-19 10:59:59.687000 [CRIT] switch_loadable_module.c:882 Error
>>> Loading module /opt/freeswitch/mod/mod_g729.so
>>> **/opt/freeswitch/mod/mod_g729.so: cannot open shared object file:  
>>> No
>>> such file or directory**
>>> 2010-02-19 10:59:59.688000 [CRIT] switch_loadable_module.c:882 Error
>>> Loading module /opt/freeswitch/mod/mod_amr.so
>>> **/opt/freeswitch/mod/mod_amr.so: cannot open shared object file: No
>>> such file or directory**
>>> 2010-02-19 10:59:59.835000 [CRIT] switch_loadable_module.c:882 Error
>>> Loading module /opt/freeswitch/mod/mod_file_string.so
>>> **/opt/freeswitch/mod/mod_file_string.so: cannot open shared object
>>> file: No such file or directory**
>>> 2010-02-19 11:00:00.143000 [CRIT] switch_loadable_module.c:882 Error
>>> Loading module /opt/freeswitch/mod/mod_say_ru.so
>>> **/opt/freeswitch/mod/mod_say_ru.so: cannot open shared object file:
>>> No such file or directory**
>>>
>>> suspect I don't really need to worry about some of these. I assume
>>> there is a config file somewhere where I can disable these options.
>>>
>>>
>>> Ok, as a really pathetic question, now I have started it, how do I
>>> stop it?
>>>
>>> freeswitch at voyage> halt
>>> Unknown Command: halt
>>> freeswitch at voyage> quit
>>> Unknown Command: quit
>>> freeswitch at voyage> exit
>>> Unknown Command: exit
>>> freeswitch at voyage> bye
>>> Unknown Command: bye
>>>
>>> -- 
>>> Brian May <brian at microcomaustralia.com.au>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>>> users
>>> http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>> users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org


From yehavi.bourvine at gmail.com  Thu Feb 18 23:46:04 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Fri, 19 Feb 2010 09:46:04 +0200
Subject: [Freeswitch-users] Announcement: FreeSWITCH Add Broadsoft SCA
	Support
In-Reply-To: <2F87B95C-EA23-4072-B845-3F7AE0617BE3@jerris.com>
References: <87f2f3b91001111742k691966cfs82515db77c13d4d4@mail.gmail.com>
	<d361ca7b1002170549g608d6e16vb2eb3f087fff9752@mail.gmail.com>
	<191c3a031002170556p5dfd4c55le3c279f194808d92@mail.gmail.com>
	<191c3a031002170557g5d8c9df9xf40d1b74b9a6f19d@mail.gmail.com>
	<d361ca7b1002170700x15630ab2tabc6f3d91ebc3303@mail.gmail.com>
	<ADC8B813-4161-481C-877E-93FC012858B4@freeswitch.org>
	<d361ca7b1002182101p60b4e8f8hf4dc95e1d2251e30@mail.gmail.com>
	<8b1c9cda1002182144m65ccf979w71384a1e7a55e2ea@mail.gmail.com>
	<d361ca7b1002182154o52352c62h8c0ec02fa66e8444@mail.gmail.com>
	<2F87B95C-EA23-4072-B845-3F7AE0617BE3@jerris.com>
Message-ID: <d361ca7b1002182346k71fd602fh6e6725834a6de528@mail.gmail.com>

A jira issue has been created:
*MODSOFIA-61*<http://jira.freeswitch.org/browse/MODSOFIA-61>
.

                  Thanks, __Yehavi:

 2010/2/19 Michael Jerris <mike at jerris.com>

>  If this issue is not already on jira could you please make sure it gets
> added?
>
> Mike
>
>
> On Feb 19, 2010, at 12:54 AM, Yehavi Bourvine <yehavi.bourvine at gmail.com>
> wrote:
>
>     Hello Gabe,
>
>   As you can see - Brian is actively investigating it, so you can expect
> for some fix soon...
>
>                        Regards, __Yehavi:
>
> 2010/2/19 Gabriel Kuri <gkuri at ieee.org>
>
>> >  When a call arrives, both ring; the one that did not answer gets only a
>> > cancel mesage without any further notification that the extension is in
>> use
>> > by the other phone.
>>
>> These are the same exact symptoms I posted about earlier this week,
>> with the Cisco SPA-5xx series phones. I still have yet to figure out
>> why this is happening, if you find out what's going on, please post
>> back the solution, I'd like to know the resolution.
>>
>> Thanks,
>> Gabe
>>
>>
>>
>> On Thu, Feb 18, 2010 at 9:01 PM, Yehavi Bourvine
>> <yehavi.bourvine at gmail.com> wrote:
>> > Thanks Brian. It now works better, but not fully (using 16659M).
>> >
>> > What happens is:
>> >
>> > When one of the Polycoms seize the line it is ok - the other phone gets
>> > notification and the extension status is "in use".
>> > When one of the Polycom phones initiates a call - all is ok:
>> >
>> > The other side sees that the extension is in use.
>> > When it is put to hold all phones who share this extension see it and
>> can
>> > pick the call.
>> >
>>
>> >
>>  >                          Thanks! __Yehavi:
>> >
>> > 2010/2/17 Brian West <brian at freeswitch.org>
>> >>
>> >> Step 1. Enable manage-shared-appearance=true
>> >>
>> >> Step 2. Now in the phone's config Configure the phone as usually, set
>> the
>> >> line shared and DO NOT set the third party name.
>> >>
>> >> Step 3. Reboot
>> >>
>> >> It should work.
>> >>
>> >> I wish someone that has this working would write some wiki docs these
>> >> threads about it not working are getting rather old when I know for a
>> fact
>> >> they work fine.
>> >>
>> >> The gateway info missing is a gateway you have configured getting a
>> >> notify.  It has nothing to do with SCA.
>> >>
>> >> /b
>> >>
>> >> On Feb 17, 2010, at 9:00 AM, Yehavi Bourvine wrote:
>> >>
>> >> > .
>> >>
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>  _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100219/07c5c86b/attachment.html 

From pmhshz at gmail.com  Fri Feb 19 00:02:43 2010
From: pmhshz at gmail.com (MohammedShehzad)
Date: Fri, 19 Feb 2010 13:32:43 +0530
Subject: [Freeswitch-users] stream a file multicast with mod_esf
In-Reply-To: <DBA4E291-9377-489A-841A-1A3D83880DF2@jerris.com>
References: <B98B6921358D0C49BD21CD023F1F52C302623B4B@hbghs04a.ww300.siemens.net>
	<B98B6921358D0C49BD21CD023F1F52C302623BE0@hbghs04a.ww300.siemens.net>
	<7C996136-963A-42D2-B0FE-D2729F968E52@freeswitch.org>
	<27071973.post@talk.nabble.com>
	<378CDEC3-5A41-4EF9-927A-311F2531E6AB@freeswitch.org>
	<a792304b1001080725y19ff28camd37425927550f311@mail.gmail.com>
	<191c3a031001081848w9036207yf6a4b473ec233243@mail.gmail.com>
	<85C74329-0D7C-48B2-B7A8-D4AA67C6E3F2@freeswitch.org>
	<a792304b1002182209m43ada13cp8ed3d91ad509f0a2@mail.gmail.com>
	<DBA4E291-9377-489A-841A-1A3D83880DF2@jerris.com>
Message-ID: <a792304b1002190002u50ec6912g1905ffc63a119174@mail.gmail.com>

On Fri, Feb 19, 2010 at 12:59 PM, Michael Jerris <mike at jerris.com> wrote:

> Listening on multicast is noting special for multicast, it is just
> like reading any other udp socket
>
> Mike
>
> Correct, but I have to play those audio stream back to caller taking care
of the audio codec  and other things, do anybody have any idea in that part?
Please let me know that.
-- 

-MohammedShehzad
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100219/6e6998d6/attachment.html 

From technical at ttnc.co.uk  Fri Feb 19 01:26:25 2010
From: technical at ttnc.co.uk (TTNC - Technical)
Date: Fri, 19 Feb 2010 09:26:25 +0000
Subject: [Freeswitch-users] mod_fax undefined symbol
In-Reply-To: <0F7476C7-FA7E-441F-9E17-EB2D23CD6055@jerris.com>
References: <6FCE44FF-70C1-412D-B722-4C8D359381BA@ttnc.co.uk>
	<0F7476C7-FA7E-441F-9E17-EB2D23CD6055@jerris.com>
Message-ID: <EBF5F462-9A4F-4699-95FB-6947A9D2D55C@ttnc.co.uk>

Hi Michel

I'm running the latest trunk I believe, r16700.

I've opened a bug report: http://jira.freeswitch.org/browse/FSMOD-37

I had libtiff4-dev installed as per the Debian setup instructions on the wiki, wasn't sure whether it was important as I could see tiff-3.8.2 in the libs folder in FreeSWITCH.

If you could come back to me or take a look at the bug report at your earliest convenience, as it's affecting service for us, it'd be appreciated.

Thanks

Russ


On 19 Feb 2010, at 07:22, Michael Jerris wrote:

> Please open a bug on http://jira.freeswitch.org for this issue.  As a  
> note, we use our own copy of libtiff, statically linked to the  
> module.  Is this recent trunk or something older?
> 
> On Feb 18, 2010, at 8:57 PM, TTNC - Technical <technical at ttnc.co.uk>  
> wrote:
> 
>> Hi Guys
>> 
>> I'm having trouble getting mod_fax to load. Running on Debian  
>> testing (squeeze). Everything is installed as per - http://wiki.freeswitch.org/wiki/Installation_Guide 
>> . (dpkg-buildpackage etc)
>> 
>> When trying to load the fax module I get:
>> 
>> 2010-02-19 01:48:17.554935 [CRIT] switch_loadable_module.c:882 Error  
>> Loading module /opt/freeswitch/mod/mod_fax.so
>> **/opt/freeswitch/mod/mod_fax.so: undefined symbol:  
>> TIFFDefaultStripSize**
>> 
>> And when a fax is sent, I'm getting:
>> 
>> 2010-02-19 01:04:57.355330 [ERR] switch_core_session.c:1490 Invalid  
>> Application rxfax
>> 
>> I guess because mod_fax isn't loaded.
>> 
>> I've got libtiff4 and libtiff4-dev installed:
>> 
>> ii  libtiff4                             3.9.2-2                   
>> Tag Image File Format (TIFF) library
>> ii  libtiff4-dev                         3.9.2-2                   
>> Tag Image File Format library (TIFF), development files
>> ii  libtiffxx0c2                         3.9.2-2                   
>> Tag Image File Format (TIFF) library -- C++ interface
>> 
>> Just tried updating to the latest svn trunk (16700M) and it hasn't  
>> made any difference.
>> 
>>> From googling, it suggests that it could be because the module is  
>>> complied against a different one currently running on the system,  
>>> however I'm not sure how this can be the case, there is only the  
>>> one version installed.
>> 
>> Any suggestions as to what I can try?
>> 
>> Any help appreciated
>> 
>> Russ
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>> users
>> http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From gmaruzz at celliax.org  Fri Feb 19 01:46:57 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Fri, 19 Feb 2010 10:46:57 +0100
Subject: [Freeswitch-users] mod_fax undefined symbol
In-Reply-To: <EBF5F462-9A4F-4699-95FB-6947A9D2D55C@ttnc.co.uk>
References: <6FCE44FF-70C1-412D-B722-4C8D359381BA@ttnc.co.uk> 
	<0F7476C7-FA7E-441F-9E17-EB2D23CD6055@jerris.com>
	<EBF5F462-9A4F-4699-95FB-6947A9D2D55C@ttnc.co.uk>
Message-ID: <7b197bef1002190146x53492b85va88e4025c59e37f3@mail.gmail.com>

On Fri, Feb 19, 2010 at 10:26 AM, TTNC - Technical <technical at ttnc.co.uk> wrote:
> I'm running the latest trunk I believe, r16700.
>
> I've opened a bug report: http://jira.freeswitch.org/browse/FSMOD-37
>
> I had libtiff4-dev installed as per the Debian setup instructions on the wiki, wasn't sure whether it was important as I could see tiff-3.8.2 in the libs folder in FreeSWITCH.

FS mod_fax uses FS provided libtiff.

If you have not modified the Makefile, mod_fax is built like that, to
use the FS provided libtiff.

Have you modified the Makefile?

-giovanni


>
> If you could come back to me or take a look at the bug report at your earliest convenience, as it's affecting service for us, it'd be appreciated.
>
> Thanks
>
> Russ
>
>
> On 19 Feb 2010, at 07:22, Michael Jerris wrote:
>
>> Please open a bug on http://jira.freeswitch.org for this issue. ?As a
>> note, we use our own copy of libtiff, statically linked to the
>> module. ?Is this recent trunk or something older?
>>
>> On Feb 18, 2010, at 8:57 PM, TTNC - Technical <technical at ttnc.co.uk>
>> wrote:
>>
>>> Hi Guys
>>>
>>> I'm having trouble getting mod_fax to load. Running on Debian
>>> testing (squeeze). Everything is installed as per - http://wiki.freeswitch.org/wiki/Installation_Guide
>>> . (dpkg-buildpackage etc)
>>>
>>> When trying to load the fax module I get:
>>>
>>> 2010-02-19 01:48:17.554935 [CRIT] switch_loadable_module.c:882 Error
>>> Loading module /opt/freeswitch/mod/mod_fax.so
>>> **/opt/freeswitch/mod/mod_fax.so: undefined symbol:
>>> TIFFDefaultStripSize**
>>>
>>> And when a fax is sent, I'm getting:
>>>
>>> 2010-02-19 01:04:57.355330 [ERR] switch_core_session.c:1490 Invalid
>>> Application rxfax
>>>
>>> I guess because mod_fax isn't loaded.
>>>
>>> I've got libtiff4 and libtiff4-dev installed:
>>>
>>> ii ?libtiff4 ? ? ? ? ? ? ? ? ? ? ? ? ? ? 3.9.2-2
>>> Tag Image File Format (TIFF) library
>>> ii ?libtiff4-dev ? ? ? ? ? ? ? ? ? ? ? ? 3.9.2-2
>>> Tag Image File Format library (TIFF), development files
>>> ii ?libtiffxx0c2 ? ? ? ? ? ? ? ? ? ? ? ? 3.9.2-2
>>> Tag Image File Format (TIFF) library -- C++ interface
>>>
>>> Just tried updating to the latest svn trunk (16700M) and it hasn't
>>> made any difference.
>>>
>>>> From googling, it suggests that it could be because the module is
>>>> complied against a different one currently running on the system,
>>>> however I'm not sure how this can be the case, there is only the
>>>> one version installed.
>>>
>>> Any suggestions as to what I can try?
>>>
>>> Any help appreciated
>>>
>>> Russ
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>>> users
>>> http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From technical at ttnc.co.uk  Fri Feb 19 02:12:32 2010
From: technical at ttnc.co.uk (TTNC - Technical)
Date: Fri, 19 Feb 2010 10:12:32 +0000
Subject: [Freeswitch-users] mod_fax undefined symbol
In-Reply-To: <7b197bef1002190146x53492b85va88e4025c59e37f3@mail.gmail.com>
References: <6FCE44FF-70C1-412D-B722-4C8D359381BA@ttnc.co.uk>
	<0F7476C7-FA7E-441F-9E17-EB2D23CD6055@jerris.com>
	<EBF5F462-9A4F-4699-95FB-6947A9D2D55C@ttnc.co.uk>
	<7b197bef1002190146x53492b85va88e4025c59e37f3@mail.gmail.com>
Message-ID: <496A4C3D-D185-4A86-BCA7-E4F5C05DA929@ttnc.co.uk>


On 19 Feb 2010, at 09:46, Giovanni Maruzzelli wrote:

> On Fri, Feb 19, 2010 at 10:26 AM, TTNC - Technical <technical at ttnc.co.uk> wrote:
>> I'm running the latest trunk I believe, r16700.
>> 
>> I've opened a bug report: http://jira.freeswitch.org/browse/FSMOD-37
>> 
>> I had libtiff4-dev installed as per the Debian setup instructions on the wiki, wasn't sure whether it was important as I could see tiff-3.8.2 in the libs folder in FreeSWITCH.
> 
> FS mod_fax uses FS provided libtiff.
> 
> If you have not modified the Makefile, mod_fax is built like that, to
> use the FS provided libtiff.
> 
> Have you modified the Makefile?
> 
> -giovanni

I haven't touched the make file, I followed the Debian install guide: http://wiki.freeswitch.org/wiki/Installation_Guide#Debian_Linux

Could it be that the Debian version of libtiff4-dev is conflicting with the FreeSWITCH static one? Doesn't seem likely...

From technical at ttnc.co.uk  Fri Feb 19 02:26:02 2010
From: technical at ttnc.co.uk (TTNC - Technical)
Date: Fri, 19 Feb 2010 10:26:02 +0000
Subject: [Freeswitch-users] mod_fax undefined symbol
In-Reply-To: <496A4C3D-D185-4A86-BCA7-E4F5C05DA929@ttnc.co.uk>
References: <6FCE44FF-70C1-412D-B722-4C8D359381BA@ttnc.co.uk>
	<0F7476C7-FA7E-441F-9E17-EB2D23CD6055@jerris.com>
	<EBF5F462-9A4F-4699-95FB-6947A9D2D55C@ttnc.co.uk>
	<7b197bef1002190146x53492b85va88e4025c59e37f3@mail.gmail.com>
	<496A4C3D-D185-4A86-BCA7-E4F5C05DA929@ttnc.co.uk>
Message-ID: <8D506E69-CB30-40BD-8AB9-9524C18E4503@ttnc.co.uk>

On 19 Feb 2010, at 10:12, TTNC - Technical wrote:

> On 19 Feb 2010, at 09:46, Giovanni Maruzzelli wrote:
> 
>> FS mod_fax uses FS provided libtiff.
>> 
>> If you have not modified the Makefile, mod_fax is built like that, to
>> use the FS provided libtiff.
>> 
>> Have you modified the Makefile?
>> 
>> -giovanni
> 
> I haven't touched the make file, I followed the Debian install guide: http://wiki.freeswitch.org/wiki/Installation_Guide#Debian_Linux
> 
> Could it be that the Debian version of libtiff4-dev is conflicting with the FreeSWITCH static one? Doesn't seem likely...


Just as another note - I've tried an install the 'freeswitch' way rather than the 'Debian' one.

dpkg-buildpackage in FreeSWITCH root requires the Debian package libtiff4-dev to be installed before it'll work. I've uninstall all Debian libtiff* packages and gone into the FreeSWITCH root and done a 'make && make install'.

Same problem persists:

2010-02-19 10:23:01.277590 [CRIT] switch_loadable_module.c:882 Error Loading module /opt/freeswitch/mod/mod_fax.so
**/opt/freeswitch/mod/mod_fax.so: undefined symbol: TIFFDefaultStripSize**

I guess that sort of shows it's an internal problem with the FreeSWITCH libtiff or mod_fax rather than the Debian package?

I'll update the bug report with this.

Russ
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100219/ebba2d99/attachment.html 

From jason at jasonjgw.net  Fri Feb 19 02:31:28 2010
From: jason at jasonjgw.net (Jason White)
Date: Fri, 19 Feb 2010 21:31:28 +1100
Subject: [Freeswitch-users] mod_fax undefined symbol
In-Reply-To: <496A4C3D-D185-4A86-BCA7-E4F5C05DA929@ttnc.co.uk>
References: <6FCE44FF-70C1-412D-B722-4C8D359381BA@ttnc.co.uk>
	<0F7476C7-FA7E-441F-9E17-EB2D23CD6055@jerris.com>
	<EBF5F462-9A4F-4699-95FB-6947A9D2D55C@ttnc.co.uk>
	<7b197bef1002190146x53492b85va88e4025c59e37f3@mail.gmail.com>
	<496A4C3D-D185-4A86-BCA7-E4F5C05DA929@ttnc.co.uk>
Message-ID: <20100219103128.GA30809@jdc.jasonjgw.net>

TTNC - Technical <technical at ttnc.co.uk> wrote:
 
> I haven't touched the make file, I followed the Debian install guide:
> http://wiki.freeswitch.org/wiki/Installation_Guide#Debian_Linux
> 
> Could it be that the Debian version of libtiff4-dev is conflicting with the
> FreeSWITCH static one? Doesn't seem likely...

It isn't likely. You could always remove that package and recompile.

I can confirm this bug in my Debian build of R16654.

(I hadn't noticed it before, since I don't need and therefore don't load
mod_fax).



From gmaruzz at celliax.org  Fri Feb 19 02:34:19 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Fri, 19 Feb 2010 11:34:19 +0100
Subject: [Freeswitch-users] mod_fax undefined symbol
In-Reply-To: <8D506E69-CB30-40BD-8AB9-9524C18E4503@ttnc.co.uk>
References: <6FCE44FF-70C1-412D-B722-4C8D359381BA@ttnc.co.uk> 
	<0F7476C7-FA7E-441F-9E17-EB2D23CD6055@jerris.com>
	<EBF5F462-9A4F-4699-95FB-6947A9D2D55C@ttnc.co.uk> 
	<7b197bef1002190146x53492b85va88e4025c59e37f3@mail.gmail.com> 
	<496A4C3D-D185-4A86-BCA7-E4F5C05DA929@ttnc.co.uk>
	<8D506E69-CB30-40BD-8AB9-9524C18E4503@ttnc.co.uk>
Message-ID: <7b197bef1002190234m58c7e8cbyd945af9ee0f5a3a@mail.gmail.com>

On Fri, Feb 19, 2010 at 11:26 AM, TTNC - Technical <technical at ttnc.co.uk> wrote:
> Just as another note - I've tried an install the 'freeswitch' way rather
> than the 'Debian' one.
> dpkg-buildpackage in FreeSWITCH root requires the Debian package
> libtiff4-dev to be installed before it'll work. I've uninstall all Debian
> libtiff* packages and gone into the FreeSWITCH root and done a 'make && make
> install'.
> Same problem persists:
> 2010-02-19 10:23:01.277590 [CRIT] switch_loadable_module.c:882 Error Loading
> module /opt/freeswitch/mod/mod_fax.so
> **/opt/freeswitch/mod/mod_fax.so: undefined symbol: TIFFDefaultStripSize**
> I guess that sort of shows it's an internal problem with the FreeSWITCH
> libtiff or mod_fax rather than the Debian package?
> I'll update the bug report with this.

try first with a fresh checkout from svn (not the one you are using
now, checkout in another dir), then ./bootstrap.sh and ./configure,
then make && make install

So you'll be sure of the results for the report.

-gm

> Russ
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From gmaruzz at celliax.org  Fri Feb 19 02:41:23 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Fri, 19 Feb 2010 11:41:23 +0100
Subject: [Freeswitch-users] mod_fax undefined symbol
In-Reply-To: <20100219103128.GA30809@jdc.jasonjgw.net>
References: <6FCE44FF-70C1-412D-B722-4C8D359381BA@ttnc.co.uk> 
	<0F7476C7-FA7E-441F-9E17-EB2D23CD6055@jerris.com>
	<EBF5F462-9A4F-4699-95FB-6947A9D2D55C@ttnc.co.uk> 
	<7b197bef1002190146x53492b85va88e4025c59e37f3@mail.gmail.com> 
	<496A4C3D-D185-4A86-BCA7-E4F5C05DA929@ttnc.co.uk>
	<20100219103128.GA30809@jdc.jasonjgw.net>
Message-ID: <7b197bef1002190241k7e061323nb663b14800e78232@mail.gmail.com>

On Fri, Feb 19, 2010 at 11:31 AM, Jason White <jason at jasonjgw.net> wrote:
>
> I can confirm this bug in my Debian build of R16654.
>
> (I hadn't noticed it before, since I don't need and therefore don't load
> mod_fax).

oooops, I can confirm it too, in a non debian (as in "normal") build.

>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From ivdreg at gmail.com  Fri Feb 19 02:42:27 2010
From: ivdreg at gmail.com (ivdreg ivdreg)
Date: Fri, 19 Feb 2010 12:42:27 +0200
Subject: [Freeswitch-users] SDP With T.38 in INVITE Problem
Message-ID: <b437a0e1002190242y2aa64696j91e8bc76f7ae2a92@mail.gmail.com>

Hi all,

Dose someone have a problem that if there T.38 in coming from gateway
FreeSwitch drops the call because of media error ? As I see from log only
T.38 port is zero and SDP has also media port. Is it possible to configure
FS to do not break a call but if media is OK.

2010-02-19 12:26:14.305244 [NOTICE] switch_channel.c:666 New Channel
sofia/backend/XXXXXXXXXX at 10.10.1.110:7065[6cd9f634-411d-df11-99ca-003048bb99cc]
2010-02-19 12:26:14.305244 [DEBUG] mod_sofia.c:3369 (sofia/backend/
XXXXXXXXXX at 10.10.1.110:7065) State Change CS_NEW -> CS_INIT
2010-02-19 12:26:14.305244 [DEBUG] switch_core_session.c:1018 Send signal
sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [BREAK]
2010-02-19 12:26:14.305244 [DEBUG] switch_core_state_machine.c:314
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) Running State Change CS_INIT
2010-02-19 12:26:14.305244 [DEBUG] switch_core_state_machine.c:338
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State INIT
2010-02-19 12:26:14.305244 [DEBUG] mod_sofia.c:83 sofia/backend/
XXXXXXXXXX at 10.10.1.110:7065 SOFIA INIT
2010-02-19 12:26:14.305244 [DEBUG] sofia_glue.c:1377 sofia/backend/
XXXXXXXXXX at 10.10.1.110:7065 Patched SDP
---
v=0
o=- 3779949069 5166785777234550622 IN IP4 206.132.232.212
s=session
t=0 0
m=audio 21108 RTP/AVP 18 4 8 0
c=IN IP4 10.10.1.110
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
m=image 21108 udptl t38
c=IN IP4 10.10.1.110
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxUdpEC:t38UDPRedundancy
a=T38FaxRateManagement:transferredTCF

+++
v=0
o=- 3779949069 5166785777234550622 IN IP4 206.132.232.212
s=session
t=0 0
m=audio 17058 RTP/AVP 18 4 8 0
c=IN IP4 10.10.1.100
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
m=image 17058 udptl t38
c=IN IP4 10.10.1.100
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxUdpEC:t38UDPRedundancy
a=T38FaxRateManagement:transferredTCF

2010-02-19 12:26:14.305244 [DEBUG] mod_sofia.c:117 (sofia/backend/
XXXXXXXXXX at 10.10.1.110:7065) State Change CS_INIT -> CS_ROUTING
......
2010-02-19 12:26:21.255201 [DEBUG] sofia.c:4135 Remote SDP:
v=0
o=FreeSWITCH 1266548331 1266548332 IN IP4 10.10.1.110
s=FreeSWITCH
c=IN IP4 10.10.1.110
t=0 0
*m=audio 26850 RTP/AVP 8*
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
*m=image 0 udptl 19*

2010-02-19 12:26:21.255201 [DEBUG] sofia.c:4124 Channel sofia/backend/
XXXXXXXXXX at 10.10.1.110:7065 entering state [ready][200]
2010-02-19 12:26:21.255201 [DEBUG] switch_channel.c:2285 Send signal
sofia/backend/YYYYYYYYYY at 10.10.1.110 [BREAK]
2010-02-19 12:26:21.255201 [NOTICE] sofia.c:4668 Channel [sofia/backend/
XXXXXXXXXX at 10.10.1.110:7065] has been answered
2010-02-19 12:26:21.255201 [DEBUG] sofia_glue.c:2288 Set Codec
sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 PROXY/8000 20 ms 160 samples
*2010-02-19 12:26:21.255201 [DEBUG] sofia_glue.c:2571 PROXY AUDIO RTP
[sofia/backend/XXXXXXXXXX at 10.10.1.110:7065]
10.10.1.100:17058->10.10.1.110:0codec: 0 ms: 20
2010-02-19 12:26:21.255201 [ERR] sofia_glue.c:2879 AUDIO RTP REPORTS ERROR:
[Missing remote port]
2010-02-19 12:26:21.255201 [NOTICE] sofia_glue.c:2880 Hangup sofia/backend/
XXXXXXXXXX at 10.10.1.110:7065 [CS_CONSUME_MEDIA] [DESTINATION_OUT_OF_ORDER]*
2010-02-19 12:26:21.255201 [DEBUG] switch_channel.c:2063 Send signal
sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [KILL]
2010-02-19 12:26:21.255201 [DEBUG] switch_core_session.c:1018 Send signal
sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [BREAK]
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:314
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) Running State Change CS_HANGUP
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:499
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State HANGUP
2010-02-19 12:26:21.255201 [DEBUG] mod_sofia.c:411 Channel sofia/backend/
XXXXXXXXXX at 10.10.1.110:7065 hanging up, cause: DESTINATION_OUT_OF_ORDER
2010-02-19 12:26:21.255201 [DEBUG] mod_sofia.c:454 Sending BYE to
sofia/backend/XXXXXXXXXX at 10.10.1.110:7065
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:46
sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 Standard HANGUP, cause:
DESTINATION_OUT_OF_ORDER
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:499
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State HANGUP going to sleep
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:333
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State Change CS_HANGUP ->
CS_REPORTING
2010-02-19 12:26:21.255201 [DEBUG] switch_core_session.c:1018 Send signal
sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [BREAK]
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:314
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) Running State Change
CS_REPORTING
2010-02-19 12:26:21.255201 [DEBUG] switch_ivr_originate.c:3185 Originate
Resulted in Error Cause: 27 [DESTINATION_OUT_OF_ORDER]
2010-02-19 12:26:21.255201 [INFO] mod_dptools.c:2353 Originate Failed.
Cause: DESTINATION_OUT_OF_ORDER
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:590
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State REPORTING


Thanks
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100219/bcc1fd79/attachment.html 

From gmaruzz at celliax.org  Fri Feb 19 02:44:03 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Fri, 19 Feb 2010 11:44:03 +0100
Subject: [Freeswitch-users] mod_fax undefined symbol
In-Reply-To: <7b197bef1002190241k7e061323nb663b14800e78232@mail.gmail.com>
References: <6FCE44FF-70C1-412D-B722-4C8D359381BA@ttnc.co.uk> 
	<0F7476C7-FA7E-441F-9E17-EB2D23CD6055@jerris.com>
	<EBF5F462-9A4F-4699-95FB-6947A9D2D55C@ttnc.co.uk> 
	<7b197bef1002190146x53492b85va88e4025c59e37f3@mail.gmail.com> 
	<496A4C3D-D185-4A86-BCA7-E4F5C05DA929@ttnc.co.uk>
	<20100219103128.GA30809@jdc.jasonjgw.net> 
	<7b197bef1002190241k7e061323nb663b14800e78232@mail.gmail.com>
Message-ID: <7b197bef1002190244m18aca4cfjb6a1c64b71e2a58@mail.gmail.com>

On Fri, Feb 19, 2010 at 11:41 AM, Giovanni Maruzzelli
<gmaruzz at celliax.org> wrote:
>
> oooops, I can confirm it too, in a non debian (as in "normal") build.

Double oooops, I had not built it ;)

I now built it, and loads flawlessly. So, "normal" as in non-debian,
build of mod_fax are OK



>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From technical at ttnc.co.uk  Fri Feb 19 02:58:46 2010
From: technical at ttnc.co.uk (TTNC - Technical)
Date: Fri, 19 Feb 2010 10:58:46 +0000
Subject: [Freeswitch-users] mod_fax undefined symbol
In-Reply-To: <7b197bef1002190241k7e061323nb663b14800e78232@mail.gmail.com>
References: <6FCE44FF-70C1-412D-B722-4C8D359381BA@ttnc.co.uk>
	<0F7476C7-FA7E-441F-9E17-EB2D23CD6055@jerris.com>
	<EBF5F462-9A4F-4699-95FB-6947A9D2D55C@ttnc.co.uk>
	<7b197bef1002190146x53492b85va88e4025c59e37f3@mail.gmail.com>
	<496A4C3D-D185-4A86-BCA7-E4F5C05DA929@ttnc.co.uk>
	<20100219103128.GA30809@jdc.jasonjgw.net>
	<7b197bef1002190241k7e061323nb663b14800e78232@mail.gmail.com>
Message-ID: <B05DBE65-CF57-4E13-9CE2-E490128F1C41@ttnc.co.uk>


On 19 Feb 2010, at 10:41, Giovanni Maruzzelli wrote:

> On Fri, Feb 19, 2010 at 11:31 AM, Jason White <jason at jasonjgw.net> wrote:
>> 
>> I can confirm this bug in my Debian build of R16654.
>> 
>> (I hadn't noticed it before, since I don't need and therefore don't load
>> mod_fax).
> 
> oooops, I can confirm it too, in a non debian (as in "normal") build.
> 

Hmm... that's a bit odd. You would have thought that if FreeSWITCH is compiled statically against it's own libtiff - then anything Debian-centric shouldn't affect it?

From gmaruzz at celliax.org  Fri Feb 19 03:08:25 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Fri, 19 Feb 2010 12:08:25 +0100
Subject: [Freeswitch-users] mod_fax undefined symbol
In-Reply-To: <B05DBE65-CF57-4E13-9CE2-E490128F1C41@ttnc.co.uk>
References: <6FCE44FF-70C1-412D-B722-4C8D359381BA@ttnc.co.uk> 
	<0F7476C7-FA7E-441F-9E17-EB2D23CD6055@jerris.com>
	<EBF5F462-9A4F-4699-95FB-6947A9D2D55C@ttnc.co.uk> 
	<7b197bef1002190146x53492b85va88e4025c59e37f3@mail.gmail.com> 
	<496A4C3D-D185-4A86-BCA7-E4F5C05DA929@ttnc.co.uk>
	<20100219103128.GA30809@jdc.jasonjgw.net> 
	<7b197bef1002190241k7e061323nb663b14800e78232@mail.gmail.com> 
	<B05DBE65-CF57-4E13-9CE2-E490128F1C41@ttnc.co.uk>
Message-ID: <7b197bef1002190308u375c16f0p9ed83d7e8e19cf89@mail.gmail.com>

On Fri, Feb 19, 2010 at 11:58 AM, TTNC - Technical <technical at ttnc.co.uk> wrote:
> Hmm... that's a bit odd. You would have thought that if FreeSWITCH is compiled statically against it's own libtiff - then anything Debian-centric shouldn't affect it?

yes, definitely

probably in your debian build libtiff is not compiled, or mod_fax is
not linked (statically) with it

"normal" build, as in non-debian, works fine
-gm


> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From technical at ttnc.co.uk  Fri Feb 19 04:04:21 2010
From: technical at ttnc.co.uk (TTNC - Technical)
Date: Fri, 19 Feb 2010 12:04:21 +0000
Subject: [Freeswitch-users] mod_fax undefined symbol
In-Reply-To: <7b197bef1002190234m58c7e8cbyd945af9ee0f5a3a@mail.gmail.com>
References: <6FCE44FF-70C1-412D-B722-4C8D359381BA@ttnc.co.uk>
	<0F7476C7-FA7E-441F-9E17-EB2D23CD6055@jerris.com>
	<EBF5F462-9A4F-4699-95FB-6947A9D2D55C@ttnc.co.uk>
	<7b197bef1002190146x53492b85va88e4025c59e37f3@mail.gmail.com>
	<496A4C3D-D185-4A86-BCA7-E4F5C05DA929@ttnc.co.uk>
	<8D506E69-CB30-40BD-8AB9-9524C18E4503@ttnc.co.uk>
	<7b197bef1002190234m58c7e8cbyd945af9ee0f5a3a@mail.gmail.com>
Message-ID: <D04CFC8A-4909-4E32-A480-567E712AE63A@ttnc.co.uk>

On 19 Feb 2010, at 10:34, Giovanni Maruzzelli wrote:

> On Fri, Feb 19, 2010 at 11:26 AM, TTNC - Technical <technical at ttnc.co.uk> wrote:
> 
> try first with a fresh checkout from svn (not the one you are using
> now, checkout in another dir), then ./bootstrap.sh and ./configure,
> then make && make install
> 
> So you'll be sure of the results for the report.
> 
> -gm

Totally clean build as suggested, checked out a new source tree, built the 'freeswitch' way:

./bootstrap.sh
./configure --prefix=/opt/freeswitch
make
make install

Same problem persists. 

Guess will have to wait to see what Michael can find out! :)

Russ



From rupa at rupa.com  Fri Feb 19 06:21:16 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Fri, 19 Feb 2010 08:21:16 -0600
Subject: [Freeswitch-users] 1.0.4 vs. trunk vs. bugs
In-Reply-To: <7716C919-BCD1-44FA-9ED2-5268F4C42100@jerris.com>
References: <5d3e0dc61002181741r57b91f86h4810de63c1a3d46d@mail.gmail.com>
	<20100219015118.GA12983@jdc.jasonjgw.net>
	<1266552605.7684.11.camel@local.freepabx.com>
	<616F57B5-182A-49A9-9825-2E5B72F5DB05@freeswitch.org>
	<7716C919-BCD1-44FA-9ED2-5268F4C42100@jerris.com>
Message-ID: <b39273531002190621l656f44a2s8b05bc4162aa5182@mail.gmail.com>

I would caution that maintaining a stable branch is going to be quite
challenging.  All commits against trunk will fall into 3 categories:

1) clearly bug fix
2) clearly new feature
3) a mix of the two

1 can probably be easily back ported and 2 would be not but what of 3?  We
don't split patches/commits based on a clear split between 1 and 2 so it
would be the job of the stable maintainer to figure it out, split it up and
then commit just the bug fix part.

I would argue that the churn in the stable branch would be sufficient to
make it a moving target just like trunk, just slower moving and one step
removed from tony ensuring everything is up to his standard.

I would also argue that at some point this project will clearly go from
"balls to the wall" development like now (lots of bug fixes and new features
all the time) to something more sane as it matures.  At some point going to
stable/dev might make sense.

Another thought.  Look at how the linux kernel is developed now.  There is
linus's branch which is essentially unstable.  It is the vendor's (distro)
job to pick a line in the sand and keep that kernel rev stable.  There is
help from people that have stepped up and maintain a "stable" kernel branch,
but that has nothing to do with the mainline development.

I can appreciate the pain that some people have with dealing with production
systems where you want stability above all else.  In reality you don't want
stability, you don't want surprising behavioral changes.  You want code that
doesn't change except in those areas that fix bugs in components that you
use.  But your component set and mine are different.  Once you are accepting
bug fixes for all components, the set that changes can churn quite a bit.

Anyway, just some food for thought.

I know that if I had to double commit (or at least consider
double committing)  every piece of code I'd get frustrated quickly.

On Fri, Feb 19, 2010 at 1:17 AM, Michael Jerris <mike at jerris.com> wrote:

> This seems a good time to note that we are still looking for volunteers to
> assist in maintaining a stable branch.  I can not do this without additional
> volunteer resources.  We have asked several times recently to fairly silent
> response.  If anyone is interested in assisting with this effort, please
> contact  me offlist and we can discuss further.
>
> Mike
>
> On Feb 18, 2010, at 11:16 PM, Brian West <brian at freeswitch.org> wrote:
>
> OK so I can sign you up for the stable team?  ;)  As per my previous email
> i'm 100% sure we would do a stable release if we had people tending to
> issues.  The only problem is you would have to be on IRC tending to issues
> because if tony sees someone asking about a problem he'll be diving in to
> fix it before they can say "I have this one".  This also means working in a
> similar manner we do already.  Our process is very chaotic at times but it
> has served us well so far.
>
> The goal is to leave Anthony alone so he can move forward and let the
> stable team manage the jira's and issues on the list related to stable.
>
> /b
>
> On Feb 18, 2010, at 10:10 PM, David Knell wrote:
>
>
> Lon Baker < <lon at kickasspixels.com>lon at kickasspixels.com> wrote:
>
>
> The development branch is where feature requests and non-critical bugs
>
> reports would be filed for the next production release.
>
>
> The current process leaves a gap between production ready and
>
> development code that may become greater over time.
>
>
> Going against the grain here, I agree with you.  The current way of
> doing things is, in my opinion, not well thought through - there's no
> reason to tag and release versions if the answer to any issue is 'make
> current', and support is not available unless that's been done.  Far
> better to either have meaningful releases with stable and devel
> branches, or not to have releases at all.
>
> --Dave
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100219/95e5af0a/attachment.html 

From mgg at giagnocavo.net  Fri Feb 19 06:23:16 2010
From: mgg at giagnocavo.net (Michael Giagnocavo)
Date: Fri, 19 Feb 2010 09:23:16 -0500
Subject: [Freeswitch-users] 1.0.4 vs. trunk vs. bugs
In-Reply-To: <7716C919-BCD1-44FA-9ED2-5268F4C42100@jerris.com>
References: <5d3e0dc61002181741r57b91f86h4810de63c1a3d46d@mail.gmail.com>
	<20100219015118.GA12983@jdc.jasonjgw.net>
	<1266552605.7684.11.camel@local.freepabx.com>
	<616F57B5-182A-49A9-9825-2E5B72F5DB05@freeswitch.org>
	<7716C919-BCD1-44FA-9ED2-5268F4C42100@jerris.com>
Message-ID: <6E8D2069C08AA84A83D336E996AE4C67032C9D5E98@mse17be1.mse17.exchange.ms>

Maybe something that could work is a simple way for people to record the SVN number and config they?re using. It might encourage people to not get so stuck up on a released version number. And once they?re more comfortable with SVN, perhaps they?ll try head more often.

-Michael

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris
Sent: Friday, February 19, 2010 12:18 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] 1.0.4 vs. trunk vs. bugs

This seems a good time to note that we are still looking for volunteers to assist in maintaining a stable branch.  I can not do this without additional volunteer resources.  We have asked several times recently to fairly silent response.  If anyone is interested in assisting with this effort, please contact  me offlist and we can discuss further.

Mike

On Feb 18, 2010, at 11:16 PM, Brian West <brian at freeswitch.org<mailto:brian at freeswitch.org>> wrote:
OK so I can sign you up for the stable team?  ;)  As per my previous email i'm 100% sure we would do a stable release if we had people tending to issues.  The only problem is you would have to be on IRC tending to issues because if tony sees someone asking about a problem he'll be diving in to fix it before they can say "I have this one".  This also means working in a similar manner we do already.  Our process is very chaotic at times but it has served us well so far.

The goal is to leave Anthony alone so he can move forward and let the stable team manage the jira's and issues on the list related to stable.

/b

On Feb 18, 2010, at 10:10 PM, David Knell wrote:



Lon Baker <lon at kickasspixels.com<mailto:lon at kickasspixels.com>> wrote:

The development branch is where feature requests and non-critical bugs
reports would be filed for the next production release.

The current process leaves a gap between production ready and
development code that may become greater over time.

Going against the grain here, I agree with you.  The current way of
doing things is, in my opinion, not well thought through - there's no
reason to tag and release versions if the answer to any issue is 'make
current', and support is not available unless that's been done.  Far
better to either have meaningful releases with stable and devel
branches, or not to have releases at all.

--Dave

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100219/28ca4ed7/attachment-0001.html 

From mike at jerris.com  Fri Feb 19 06:37:25 2010
From: mike at jerris.com (Michael Jerris)
Date: Fri, 19 Feb 2010 09:37:25 -0500
Subject: [Freeswitch-users] mod_fax undefined symbol
In-Reply-To: <D04CFC8A-4909-4E32-A480-567E712AE63A@ttnc.co.uk>
References: <6FCE44FF-70C1-412D-B722-4C8D359381BA@ttnc.co.uk>
	<0F7476C7-FA7E-441F-9E17-EB2D23CD6055@jerris.com>
	<EBF5F462-9A4F-4699-95FB-6947A9D2D55C@ttnc.co.uk>
	<7b197bef1002190146x53492b85va88e4025c59e37f3@mail.gmail.com>
	<496A4C3D-D185-4A86-BCA7-E4F5C05DA929@ttnc.co.uk>
	<8D506E69-CB30-40BD-8AB9-9524C18E4503@ttnc.co.uk>
	<7b197bef1002190234m58c7e8cbyd945af9ee0f5a3a@mail.gmail.com>
	<D04CFC8A-4909-4E32-A480-567E712AE63A@ttnc.co.uk>
Message-ID: <2547643E-D259-46CE-919D-51393A4151BC@jerris.com>

My only thought without actually looking is this could be related to  
libjpeg being there during build but not when running.  Do you have  
jpeg devel package but not lib installed?

Mike

On Feb 19, 2010, at 7:04 AM, TTNC - Technical <technical at ttnc.co.uk>  
wrote:

> On 19 Feb 2010, at 10:34, Giovanni Maruzzelli wrote:
>
>> On Fri, Feb 19, 2010 at 11:26 AM, TTNC - Technical <technical at ttnc.co.uk 
>> > wrote:
>>
>> try first with a fresh checkout from svn (not the one you are using
>> now, checkout in another dir), then ./bootstrap.sh and ./configure,
>> then make && make install
>>
>> So you'll be sure of the results for the report.
>>
>> -gm
>
> Totally clean build as suggested, checked out a new source tree,  
> built the 'freeswitch' way:
>
> ./bootstrap.sh
> ./configure --prefix=/opt/freeswitch
> make
> make install
>
> Same problem persists.
>
> Guess will have to wait to see what Michael can find out! :)
>
> Russ
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org


From helmut.kuper at ewetel.de  Fri Feb 19 07:08:04 2010
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Fri, 19 Feb 2010 16:08:04 +0100
Subject: [Freeswitch-users] Question about sofia_contact
Message-ID: <4B7EA954.30402@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hello,

I try to setup a FS sofia sip-profile which allows me to have multiple
sip-profiles but one registration database. So I set the following
parameters:

<param name="challenge-realm" value="auto_from"/>
<param name="force-register-domain" value="$${domain}"/>
<param name="force-subscription-domain" value="$${domain}"/>
<param name="force-register-db-domain" value="$${domain}"/>

where domain is set to "mydomain". "sofia status profile internal"
delivers the following:


Call-ID:        3c26705038e5-vwlg8u5q9cwe
User:           2701 at mydomain
Contact:        <sip:2701 at 85.16.245.208:1024;line=l2wksedo>
Agent:          snom370/8.2.22
Status:         Registered(UDP)(unknown) EXP(2010-02-19 16:13:31)
Host:           ippbx-prod-node0
IP:             85.16.245.208
Port:           1024
Auth-User:      2701
Auth-Realm:     mydomain
MWI-Account:    2701 at mydomain



sofia_contact internal/2701 at mydomain delivers this:
error/user_not_registered

The Phone is fully functional.

I use SVN trunk 16601

regards
Helmut
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.4.7 (MingW32)

iD8DBQFLfqlT4tZeNddg3dwRAgk1AJ4gtXoVLn8anWF6BX2nijuvApoJiwCbBYyT
jo6HuOqH62g8Mmia5PEXII8=
=fDxi
-----END PGP SIGNATURE-----


From helmut.kuper at ewetel.de  Fri Feb 19 07:26:52 2010
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Fri, 19 Feb 2010 16:26:52 +0100
Subject: [Freeswitch-users] Question about sofia_contact
In-Reply-To: <4B7EA954.30402@ewetel.de>
References: <4B7EA954.30402@ewetel.de>
Message-ID: <4B7EADBC.1040001@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hi,

an update:
The corresponding select statement looks for sip_user="2701" and
sip_host="internal" in registration table.
This delivers of course no result because 2701 is registered with
sip_host="mydomain".


Hm any workaround or am I going in a wrong direction?


regards
Helmut


On 19.02.2010 16:08, Helmut Kuper wrote:
> Hello,
> 
> I try to setup a FS sofia sip-profile which allows me to have multiple
> sip-profiles but one registration database. So I set the following
> parameters:
> 
> <param name="challenge-realm" value="auto_from"/>
> <param name="force-register-domain" value="$${domain}"/>
> <param name="force-subscription-domain" value="$${domain}"/>
> <param name="force-register-db-domain" value="$${domain}"/>
> 
> where domain is set to "mydomain". "sofia status profile internal"
> delivers the following:
> 
> 
> Call-ID:        3c26705038e5-vwlg8u5q9cwe
> User:           2701 at mydomain
> Contact:        <sip:2701 at 85.16.245.208:1024;line=l2wksedo>
> Agent:          snom370/8.2.22
> Status:         Registered(UDP)(unknown) EXP(2010-02-19 16:13:31)
> Host:           ippbx-prod-node0
> IP:             85.16.245.208
> Port:           1024
> Auth-User:      2701
> Auth-Realm:     mydomain
> MWI-Account:    2701 at mydomain
> 
> 
> 
> sofia_contact internal/2701 at mydomain delivers this:
> error/user_not_registered
> 
> The Phone is fully functional.
> 
> I use SVN trunk 16601
> 
> regards
> Helmut
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.4.7 (MingW32)

iD8DBQFLfq274tZeNddg3dwRAl/pAJ96PIV5s/sZTbcJ/Pq4v/VirYteygCfXsML
j/Wdb5ApZx+x0q0uilvqkEU=
=NdFk
-----END PGP SIGNATURE-----


From technical at ttnc.co.uk  Fri Feb 19 07:30:09 2010
From: technical at ttnc.co.uk (TTNC - Technical)
Date: Fri, 19 Feb 2010 15:30:09 +0000
Subject: [Freeswitch-users] mod_fax undefined symbol
In-Reply-To: <2547643E-D259-46CE-919D-51393A4151BC@jerris.com>
References: <6FCE44FF-70C1-412D-B722-4C8D359381BA@ttnc.co.uk>
	<0F7476C7-FA7E-441F-9E17-EB2D23CD6055@jerris.com>
	<EBF5F462-9A4F-4699-95FB-6947A9D2D55C@ttnc.co.uk>
	<7b197bef1002190146x53492b85va88e4025c59e37f3@mail.gmail.com>
	<496A4C3D-D185-4A86-BCA7-E4F5C05DA929@ttnc.co.uk>
	<8D506E69-CB30-40BD-8AB9-9524C18E4503@ttnc.co.uk>
	<7b197bef1002190234m58c7e8cbyd945af9ee0f5a3a@mail.gmail.com>
	<D04CFC8A-4909-4E32-A480-567E712AE63A@ttnc.co.uk>
	<2547643E-D259-46CE-919D-51393A4151BC@jerris.com>
Message-ID: <CAC92986-7EA7-446B-8A95-50BF2FC25FCB@ttnc.co.uk>

Hi Mike

libjpeg installed, both -dev library:

voipin1:~# dpkg -l | grep jpeg
ii  libjpeg62                            6b-15                    The Independent JPEG Group's JPEG runtime library
ii  libjpeg62-dev                        6b-15                    Development files for the IJG JPEG library

/usr/lib/libjpeg.a
/usr/lib/libjpeg.la
/usr/lib/libjpeg.so
/usr/lib/libjpeg.so.62
/usr/lib/libjpeg.so.62.0.0
/usr/include/jpeglib.h
/usr/include/jpegint.h
/usr/include/jconfig.h
/usr/include/jerror.h
/usr/include/jmorecfg.h

Any other ideas? :-/


On 19 Feb 2010, at 14:37, Michael Jerris wrote:

> My only thought without actually looking is this could be related to  
> libjpeg being there during build but not when running.  Do you have  
> jpeg devel package but not lib installed?
> 
> Mike
> 
> On Feb 19, 2010, at 7:04 AM, TTNC - Technical <technical at ttnc.co.uk>  
> wrote:
> 
>> On 19 Feb 2010, at 10:34, Giovanni Maruzzelli wrote:
>> 
>>> On Fri, Feb 19, 2010 at 11:26 AM, TTNC - Technical <technical at ttnc.co.uk 
>>>> wrote:
>>> 
>>> try first with a fresh checkout from svn (not the one you are using
>>> now, checkout in another dir), then ./bootstrap.sh and ./configure,
>>> then make && make install
>>> 
>>> So you'll be sure of the results for the report.
>>> 
>>> -gm
>> 
>> Totally clean build as suggested, checked out a new source tree,  
>> built the 'freeswitch' way:
>> 
>> ./bootstrap.sh
>> ./configure --prefix=/opt/freeswitch
>> make
>> make install
>> 
>> Same problem persists.
>> 
>> Guess will have to wait to see what Michael can find out! :)
>> 
>> Russ
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>> users
>> http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From anthony.minessale at gmail.com  Fri Feb 19 07:32:14 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 19 Feb 2010 09:32:14 -0600
Subject: [Freeswitch-users] 1.0.4 vs. trunk vs. bugs
In-Reply-To: <b39273531002190621l656f44a2s8b05bc4162aa5182@mail.gmail.com>
References: <5d3e0dc61002181741r57b91f86h4810de63c1a3d46d@mail.gmail.com>
	<20100219015118.GA12983@jdc.jasonjgw.net>
	<1266552605.7684.11.camel@local.freepabx.com>
	<616F57B5-182A-49A9-9825-2E5B72F5DB05@freeswitch.org>
	<7716C919-BCD1-44FA-9ED2-5268F4C42100@jerris.com>
	<b39273531002190621l656f44a2s8b05bc4162aa5182@mail.gmail.com>
Message-ID: <191c3a031002190732i3ff1598eob746072b5a55219d@mail.gmail.com>

Here's the deal.

This is a community project and its public but it only has a small group
individuals who are "all-in" committed to the project.  I am the one who
started the project and who had to spend a solid 2 years completely alone
working towards my goals before others even showed interest.

Now we are growing very fast, we have a lot of feedback and we listen to it
regardless of our position on the subject and I will decide and make any
policy that I choose and feel is the best interest of this project.

It's reasonable that someone who is using the software wants to have
wonderful stable stepping stones to migrate towards the future on.  It's
also customary that most software, even when you pay a premium price for it,
does not meet those standards every time.  If you do find software that has
these graceful releases, they probably have a lot of people getting paid to
work diligently on it.  There are also many successful open source projects
with shiny revision numbers and packaged up with a bow but that is because
they have a dedicated team of people.

So, we don't have that long list of people.  We invited people to do it and
we had nobody step up.  So, this is what i am planning to do:

We are going to move our development to a branch, work on it from there and
push them down to trunk when we think its the best time.  This might not
always end up perfect but this is what we are going to do.  The actual
release versions are still just fancy road signs in a long journey towards
perfection.  We are still on 1.0 for almost 2 years with 5 micro release
that contain a 12 page change log each time.  We are as careful as we can be
about releases and we have no time to try to back-port patches to 6 month
old code with more than 2000 revisions in between them.

When we feel we are happy with 1.0 we will then branch to 1.1 and all this
stuff everyone wants, *IF* we get enough volunteers by that time to dedicate
their time to maintaining it.  If not we will make the best of what we
have........

This is the final word on this subject, feel free to quote me.





On Fri, Feb 19, 2010 at 8:21 AM, Rupa Schomaker <rupa at rupa.com> wrote:

> I would caution that maintaining a stable branch is going to be quite
> challenging.  All commits against trunk will fall into 3 categories:
>
> 1) clearly bug fix
> 2) clearly new feature
> 3) a mix of the two
>
> 1 can probably be easily back ported and 2 would be not but what of 3?  We
> don't split patches/commits based on a clear split between 1 and 2 so it
> would be the job of the stable maintainer to figure it out, split it up and
> then commit just the bug fix part.
>
> I would argue that the churn in the stable branch would be sufficient to
> make it a moving target just like trunk, just slower moving and one step
> removed from tony ensuring everything is up to his standard.
>
> I would also argue that at some point this project will clearly go from
> "balls to the wall" development like now (lots of bug fixes and new features
> all the time) to something more sane as it matures.  At some point going to
> stable/dev might make sense.
>
> Another thought.  Look at how the linux kernel is developed now.  There is
> linus's branch which is essentially unstable.  It is the vendor's (distro)
> job to pick a line in the sand and keep that kernel rev stable.  There is
> help from people that have stepped up and maintain a "stable" kernel branch,
> but that has nothing to do with the mainline development.
>
> I can appreciate the pain that some people have with dealing with
> production systems where you want stability above all else.  In reality you
> don't want stability, you don't want surprising behavioral changes.  You
> want code that doesn't change except in those areas that fix bugs in
> components that you use.  But your component set and mine are different.
>  Once you are accepting bug fixes for all components, the set that changes
> can churn quite a bit.
>
> Anyway, just some food for thought.
>
> I know that if I had to double commit (or at least consider
> double committing)  every piece of code I'd get frustrated quickly.
>
>
> On Fri, Feb 19, 2010 at 1:17 AM, Michael Jerris <mike at jerris.com> wrote:
>
>> This seems a good time to note that we are still looking for volunteers to
>> assist in maintaining a stable branch.  I can not do this without additional
>> volunteer resources.  We have asked several times recently to fairly silent
>> response.  If anyone is interested in assisting with this effort, please
>> contact  me offlist and we can discuss further.
>>
>> Mike
>>
>> On Feb 18, 2010, at 11:16 PM, Brian West <brian at freeswitch.org> wrote:
>>
>> OK so I can sign you up for the stable team?  ;)  As per my previous email
>> i'm 100% sure we would do a stable release if we had people tending to
>> issues.  The only problem is you would have to be on IRC tending to issues
>> because if tony sees someone asking about a problem he'll be diving in to
>> fix it before they can say "I have this one".  This also means working in a
>> similar manner we do already.  Our process is very chaotic at times but it
>> has served us well so far.
>>
>> The goal is to leave Anthony alone so he can move forward and let the
>> stable team manage the jira's and issues on the list related to stable.
>>
>> /b
>>
>> On Feb 18, 2010, at 10:10 PM, David Knell wrote:
>>
>>
>> Lon Baker < <lon at kickasspixels.com>lon at kickasspixels.com> wrote:
>>
>>
>> The development branch is where feature requests and non-critical bugs
>>
>> reports would be filed for the next production release.
>>
>>
>> The current process leaves a gap between production ready and
>>
>> development code that may become greater over time.
>>
>>
>> Going against the grain here, I agree with you.  The current way of
>> doing things is, in my opinion, not well thought through - there's no
>> reason to tag and release versions if the answer to any issue is 'make
>> current', and support is not available unless that's been done.  Far
>> better to either have meaningful releases with stable and devel
>> branches, or not to have releases at all.
>>
>> --Dave
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100219/67e469b1/attachment-0001.html 

From testeador01 at gmail.com  Fri Feb 19 07:41:16 2010
From: testeador01 at gmail.com (Milena)
Date: Fri, 19 Feb 2010 10:41:16 -0500
Subject: [Freeswitch-users] 1.0.4 vs. trunk vs. bugs
In-Reply-To: <6E8D2069C08AA84A83D336E996AE4C67032C9D5E98@mse17be1.mse17.exchange.ms>
References: <5d3e0dc61002181741r57b91f86h4810de63c1a3d46d@mail.gmail.com>
	<20100219015118.GA12983@jdc.jasonjgw.net>
	<1266552605.7684.11.camel@local.freepabx.com>
	<616F57B5-182A-49A9-9825-2E5B72F5DB05@freeswitch.org>
	<7716C919-BCD1-44FA-9ED2-5268F4C42100@jerris.com>
	<6E8D2069C08AA84A83D336E996AE4C67032C9D5E98@mse17be1.mse17.exchange.ms>
Message-ID: <c29b930b1002190741k215db689v2fc20a0574db3659@mail.gmail.com>

there's no reason to tag and release versions if the answer to any
issue is 'make current', and support is not available unless that's
been done.  Far better to either have meaningful releases with  stable
and devel branches, or not to have releases at all.


Well, the response on a stable version about a bug report that's been
already fixed would be: "apply this patch and retest, come back to us
if problem persists"

Isn't that exactly what make current does?
This way of developing may be not broken-trunk-proof, but from your
comment I think an easier way to approach Lon's suggestion would be
making a more understandable way for simple users (not fs devs) to
check what modules have been altered from their version to the latest,
in order to check if an issue has been solved or not, as of those
clients that only want "released versions", that's the hard part.

ps: hmmm Anthony's message makes my post outdated but i spent a lil
bit of thought on it so i post anyways ^^

-Milena

2010/2/19 Michael Giagnocavo <mgg at giagnocavo.net>

> Maybe something that could work is a simple way for people to record the
> SVN number and config they?re using. It might encourage people to not get so
> stuck up on a released version number. And once they?re more comfortable
> with SVN, perhaps they?ll try head more often.
>
>
>
> -Michael
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Jerris
> *Sent:* Friday, February 19, 2010 12:18 AM
>
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] 1.0.4 vs. trunk vs. bugs
>
>
>
> This seems a good time to note that we are still looking for volunteers to
> assist in maintaining a stable branch.  I can not do this without additional
> volunteer resources.  We have asked several times recently to fairly silent
> response.  If anyone is interested in assisting with this effort, please
> contact  me offlist and we can discuss further.
>
>
>
> Mike
>
>
> On Feb 18, 2010, at 11:16 PM, Brian West <brian at freeswitch.org> wrote:
>
> OK so I can sign you up for the stable team?  ;)  As per my previous email
> i'm 100% sure we would do a stable release if we had people tending to
> issues.  The only problem is you would have to be on IRC tending to issues
> because if tony sees someone asking about a problem he'll be diving in to
> fix it before they can say "I have this one".  This also means working in a
> similar manner we do already.  Our process is very chaotic at times but it
> has served us well so far.
>
>
>
> The goal is to leave Anthony alone so he can move forward and let the
> stable team manage the jira's and issues on the list related to stable.
>
>
>
> /b
>
>
>
> On Feb 18, 2010, at 10:10 PM, David Knell wrote:
>
>
>
>
> Lon Baker <lon at kickasspixels.com> wrote:
>
>
>
> The development branch is where feature requests and non-critical bugs
>
> reports would be filed for the next production release.
>
>
>
> The current process leaves a gap between production ready and
>
> development code that may become greater over time.
>
>
> Going against the grain here, I agree with you.  The current way of
> doing things is, in my opinion, not well thought through - there's no
> reason to tag and release versions if the answer to any issue is 'make
> current', and support is not available unless that's been done.  Far
> better to either have meaningful releases with stable and devel
> branches, or not to have releases at all.
>
> --Dave
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100219/c8ea5705/attachment.html 

From jalsot at gmail.com  Fri Feb 19 08:12:27 2010
From: jalsot at gmail.com (Tamas)
Date: Fri, 19 Feb 2010 17:12:27 +0100
Subject: [Freeswitch-users] mod_fax undefined symbol
In-Reply-To: <CAC92986-7EA7-446B-8A95-50BF2FC25FCB@ttnc.co.uk>
References: <6FCE44FF-70C1-412D-B722-4C8D359381BA@ttnc.co.uk>	<0F7476C7-FA7E-441F-9E17-EB2D23CD6055@jerris.com>	<EBF5F462-9A4F-4699-95FB-6947A9D2D55C@ttnc.co.uk>	<7b197bef1002190146x53492b85va88e4025c59e37f3@mail.gmail.com>	<496A4C3D-D185-4A86-BCA7-E4F5C05DA929@ttnc.co.uk>	<8D506E69-CB30-40BD-8AB9-9524C18E4503@ttnc.co.uk>	<7b197bef1002190234m58c7e8cbyd945af9ee0f5a3a@mail.gmail.com>	<D04CFC8A-4909-4E32-A480-567E712AE63A@ttnc.co.uk>	<2547643E-D259-46CE-919D-51393A4151BC@jerris.com>
	<CAC92986-7EA7-446B-8A95-50BF2FC25FCB@ttnc.co.uk>
Message-ID: <4B7EB86B.9010002@gmail.com>

An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100219/397af4cf/attachment.html 

From t.mahe at telemaque.fr  Fri Feb 19 08:19:37 2010
From: t.mahe at telemaque.fr (=?ISO-8859-1?Q?Tristan_Mah=E9?=)
Date: Fri, 19 Feb 2010 17:19:37 +0100
Subject: [Freeswitch-users] 1.0.4 vs. trunk vs. bugs
In-Reply-To: <191c3a031002190732i3ff1598eob746072b5a55219d@mail.gmail.com>
References: <5d3e0dc61002181741r57b91f86h4810de63c1a3d46d@mail.gmail.com>	<20100219015118.GA12983@jdc.jasonjgw.net>	<1266552605.7684.11.camel@local.freepabx.com>	<616F57B5-182A-49A9-9825-2E5B72F5DB05@freeswitch.org>	<7716C919-BCD1-44FA-9ED2-5268F4C42100@jerris.com>	<b39273531002190621l656f44a2s8b05bc4162aa5182@mail.gmail.com>
	<191c3a031002190732i3ff1598eob746072b5a55219d@mail.gmail.com>
Message-ID: <4B7EBA19.8000507@telemaque.fr>

Hi Anthony,

As I said to Mike offlist, You can count on me for some help, not full
time badly, as I only have very little free time ( yep you
did/freeswitch is a much too awesome piece of code ;) ),:
backports/automated tests/some resources like boxes, I already said on
IRC that you just have to ask...

I follow carefully every commit to the project ( viva fs-trunk ML ), so
helping on identifying bugfixes and backporting them ( or at least
keeping track of what should be backported ) is a task I can help on...

For my part, I must say that I LOVE the 'make current' way, when I need
stable, I just stick to a specific svn rev I've fully tested.
I understand the need for a stable release for some people in the other
hand...

My 2cents on this subject:
As long as you and the people involved don't focus on maintaining a
'stable' branch, which would obviously slow down future improvements,
and keeps working like you did for all these years, adding a 'test field
branch' for instant commits is great to ensure trunk don't get broken
the time you fix it. This is a great compromise, with few overhead of work.

Regarding the labelling versions, it might also be quite a nightmare:
When do you branch ? new features ? new behaviour in code ? That would
require strict labelling rules that might add you some overhead of work
again, slowing things down.
Hope a team will arise for those people needing labels, that would carry
this work, and let you follow your journey on trunk without bothering
with theses issues, I would definitively give a hand to help on this if
needed...


Regards,

Gled

Anthony Minessale a ?crit :
> Here's the deal.
> 
> This is a community project and its public but it only has a small group
> individuals who are "all-in" committed to the project.  I am the one who
> started the project and who had to spend a solid 2 years completely
> alone working towards my goals before others even showed interest. 
> 
> Now we are growing very fast, we have a lot of feedback and we listen to
> it regardless of our position on the subject and I will decide and make
> any policy that I choose and feel is the best interest of this project.
> 
> It's reasonable that someone who is using the software wants to have
> wonderful stable stepping stones to migrate towards the future on.  It's
> also customary that most software, even when you pay a premium price for
> it, does not meet those standards every time.  If you do find software
> that has these graceful releases, they probably have a lot of people
> getting paid to work diligently on it.  There are also many successful
> open source projects with shiny revision numbers and packaged up with a
> bow but that is because they have a dedicated team of people.
> 
> So, we don't have that long list of people.  We invited people to do it
> and we had nobody step up.  So, this is what i am planning to do:
> 
> We are going to move our development to a branch, work on it from there
> and push them down to trunk when we think its the best time.  This might
> not always end up perfect but this is what we are going to do.  The
> actual release versions are still just fancy road signs in a long
> journey towards perfection.  We are still on 1.0 for almost 2 years with
> 5 micro release that contain a 12 page change log each time.  We are as
> careful as we can be about releases and we have no time to try to
> back-port patches to 6 month old code with more than 2000 revisions in
> between them.
> 
> When we feel we are happy with 1.0 we will then branch to 1.1 and all
> this stuff everyone wants, *IF* we get enough volunteers by that time to
> dedicate their time to maintaining it.  If not we will make the best of
> what we have........
> 
> This is the final word on this subject, feel free to quote me.
> 
> 
> 
> 
> 
> On Fri, Feb 19, 2010 at 8:21 AM, Rupa Schomaker <rupa at rupa.com
> <mailto:rupa at rupa.com>> wrote:
> 
>     I would caution that maintaining a stable branch is going to be
>     quite challenging.  All commits against trunk will fall into 3
>     categories:
> 
>     1) clearly bug fix
>     2) clearly new feature
>     3) a mix of the two
> 
>     1 can probably be easily back ported and 2 would be not but what of
>     3?  We don't split patches/commits based on a clear split between 1
>     and 2 so it would be the job of the stable maintainer to figure it
>     out, split it up and then commit just the bug fix part.
> 
>     I would argue that the churn in the stable branch would be
>     sufficient to make it a moving target just like trunk, just slower
>     moving and one step removed from tony ensuring everything is up to
>     his standard.
> 
>     I would also argue that at some point this project will clearly go
>     from "balls to the wall" development like now (lots of bug fixes and
>     new features all the time) to something more sane as it matures.  At
>     some point going to stable/dev might make sense.
> 
>     Another thought.  Look at how the linux kernel is developed now.
>      There is linus's branch which is essentially unstable.  It is the
>     vendor's (distro) job to pick a line in the sand and keep that
>     kernel rev stable.  There is help from people that have stepped up
>     and maintain a "stable" kernel branch, but that has nothing to do
>     with the mainline development.
> 
>     I can appreciate the pain that some people have with dealing with
>     production systems where you want stability above all else.  In
>     reality you don't want stability, you don't want
>     surprising behavioral changes.  You want code that doesn't change
>     except in those areas that fix bugs in components that you use.  But
>     your component set and mine are different.  Once you are accepting
>     bug fixes for all components, the set that changes can churn quite a
>     bit.  
> 
>     Anyway, just some food for thought.
> 
>     I know that if I had to double commit (or at least consider
>     double committing)  every piece of code I'd get frustrated quickly.
> 
> 
>     On Fri, Feb 19, 2010 at 1:17 AM, Michael Jerris <mike at jerris.com
>     <mailto:mike at jerris.com>> wrote:
> 
>         This seems a good time to note that we are still looking for
>         volunteers to assist in maintaining a stable branch.  I can not
>         do this without additional volunteer resources.  We have asked
>         several times recently to fairly silent response.  If anyone is
>         interested in assisting with this effort, please contact  me
>         offlist and we can discuss further.
> 
>         Mike
> 
>         On Feb 18, 2010, at 11:16 PM, Brian West <brian at freeswitch.org
>         <mailto:brian at freeswitch.org>> wrote:
> 
>>         OK so I can sign you up for the stable team?  ;)  As per my
>>         previous email i'm 100% sure we would do a stable release if
>>         we had people tending to issues.  The only problem is you
>>         would have to be on IRC tending to issues because if tony sees
>>         someone asking about a problem he'll be diving in to fix it
>>         before they can say "I have this one".  This also means
>>         working in a similar manner we do already.  Our process is
>>         very chaotic at times but it has served us well so far.  
>>
>>         The goal is to leave Anthony alone so he can move forward and
>>         let the stable team manage the jira's and issues on the list
>>         related to stable.
>>
>>         /b
>>
>>         On Feb 18, 2010, at 10:10 PM, David Knell wrote:
>>
>>>>
>>>>         Lon Baker <
>>>>         <mailto:lon at kickasspixels.com>lon at kickasspixels.com
>>>>         <mailto:lon at kickasspixels.com>> wrote:
>>>>
>>>>>         The development branch is where feature requests and
>>>>>         non-critical bugs
>>>>>         reports would be filed for the next production release.
>>>>>
>>>>>         The current process leaves a gap between production ready and
>>>>>         development code that may become greater over time.
>>>
>>>         Going against the grain here, I agree with you.  The current
>>>         way of
>>>         doing things is, in my opinion, not well thought through -
>>>         there's no
>>>         reason to tag and release versions if the answer to any issue
>>>         is 'make
>>>         current', and support is not available unless that's been
>>>         done.  Far
>>>         better to either have meaningful releases with stable and devel
>>>         branches, or not to have releases at all.
>>>
>>>         --Dave
>>
>>         _______________________________________________
>>         FreeSWITCH-users mailing list
>>         FreeSWITCH-users at lists.freeswitch.org
>>         <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>         http://www.freeswitch.org
> 
>         _______________________________________________
>         FreeSWITCH-users mailing list
>         FreeSWITCH-users at lists.freeswitch.org
>         <mailto:FreeSWITCH-users at lists.freeswitch.org>
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         http://www.freeswitch.org
> 
> 
> 
> 
>     -- 
>     -Rupa
> 
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
> 
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> <mailto:MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> <mailto:sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> <http://iax:guest at conference.freeswitch.org/888>
> googletalk:conf+888 at conference.freeswitch.org
> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From msc at freeswitch.org  Fri Feb 19 09:07:29 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 19 Feb 2010 09:07:29 -0800
Subject: [Freeswitch-users] FreeSWITCH Conference Call Starting Shortly!
Message-ID: <87f2f3b91002190907p19860ef7m461cc1957031d80e@mail.gmail.com>

Come on in!

http://wiki.freeswitch.org/wiki/FS_weekly_2010_02_19

-Michael
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100219/00657107/attachment.html 

From anthony.minessale at gmail.com  Fri Feb 19 09:08:37 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 19 Feb 2010 11:08:37 -0600
Subject: [Freeswitch-users] Announcement: FreeSWITCH Add Broadsoft SCA
	Support
In-Reply-To: <d361ca7b1002182346k71fd602fh6e6725834a6de528@mail.gmail.com>
References: <87f2f3b91001111742k691966cfs82515db77c13d4d4@mail.gmail.com>
	<191c3a031002170556p5dfd4c55le3c279f194808d92@mail.gmail.com>
	<191c3a031002170557g5d8c9df9xf40d1b74b9a6f19d@mail.gmail.com>
	<d361ca7b1002170700x15630ab2tabc6f3d91ebc3303@mail.gmail.com>
	<ADC8B813-4161-481C-877E-93FC012858B4@freeswitch.org>
	<d361ca7b1002182101p60b4e8f8hf4dc95e1d2251e30@mail.gmail.com>
	<8b1c9cda1002182144m65ccf979w71384a1e7a55e2ea@mail.gmail.com>
	<d361ca7b1002182154o52352c62h8c0ec02fa66e8444@mail.gmail.com>
	<2F87B95C-EA23-4072-B845-3F7AE0617BE3@jerris.com>
	<d361ca7b1002182346k71fd602fh6e6725834a6de528@mail.gmail.com>
Message-ID: <191c3a031002190908r3922efccqc6d223a5fabd458b@mail.gmail.com>

go see my comments on that bug note.
be prepared to give us ssh access and call or irc so we can can see you
reproducing it.

If you are not on the latest firmware on all the phones, we will not
continue with this process.



On Fri, Feb 19, 2010 at 1:46 AM, Yehavi Bourvine
<yehavi.bourvine at gmail.com>wrote:

> A jira issue has been created: *MODSOFIA-61*<http://jira.freeswitch.org/browse/MODSOFIA-61>
> .
>
>                   Thanks, __Yehavi:
>
>  2010/2/19 Michael Jerris <mike at jerris.com>
>
>>  If this issue is not already on jira could you please make sure it gets
>> added?
>>
>> Mike
>>
>>
>> On Feb 19, 2010, at 12:54 AM, Yehavi Bourvine <yehavi.bourvine at gmail.com>
>> wrote:
>>
>>     Hello Gabe,
>>
>>   As you can see - Brian is actively investigating it, so you can expect
>> for some fix soon...
>>
>>                        Regards, __Yehavi:
>>
>> 2010/2/19 Gabriel Kuri < <gkuri at ieee.org>gkuri at ieee.org>
>>
>>> >  When a call arrives, both ring; the one that did not answer gets only
>>> a
>>> > cancel mesage without any further notification that the extension is in
>>> use
>>> > by the other phone.
>>>
>>> These are the same exact symptoms I posted about earlier this week,
>>> with the Cisco SPA-5xx series phones. I still have yet to figure out
>>> why this is happening, if you find out what's going on, please post
>>> back the solution, I'd like to know the resolution.
>>>
>>> Thanks,
>>> Gabe
>>>
>>>
>>>
>>> On Thu, Feb 18, 2010 at 9:01 PM, Yehavi Bourvine
>>> < <yehavi.bourvine at gmail.com>yehavi.bourvine at gmail.com> wrote:
>>> > Thanks Brian. It now works better, but not fully (using 16659M).
>>> >
>>> > What happens is:
>>> >
>>> > When one of the Polycoms seize the line it is ok - the other phone gets
>>> > notification and the extension status is "in use".
>>> > When one of the Polycom phones initiates a call - all is ok:
>>> >
>>> > The other side sees that the extension is in use.
>>> > When it is put to hold all phones who share this extension see it and
>>> can
>>> > pick the call.
>>> >
>>>
>>> >
>>>  >                          Thanks! __Yehavi:
>>> >
>>> > 2010/2/17 Brian West < <brian at freeswitch.org>brian at freeswitch.org>
>>> >>
>>> >> Step 1. Enable manage-shared-appearance=true
>>> >>
>>> >> Step 2. Now in the phone's config Configure the phone as usually, set
>>> the
>>> >> line shared and DO NOT set the third party name.
>>> >>
>>> >> Step 3. Reboot
>>> >>
>>> >> It should work.
>>> >>
>>> >> I wish someone that has this working would write some wiki docs these
>>> >> threads about it not working are getting rather old when I know for a
>>> fact
>>> >> they work fine.
>>> >>
>>> >> The gateway info missing is a gateway you have configured getting a
>>> >> notify.  It has nothing to do with SCA.
>>> >>
>>> >> /b
>>> >>
>>> >> On Feb 17, 2010, at 9:00 AM, Yehavi Bourvine wrote:
>>> >>
>>> >> > .
>>> >>
>>> >>
>>> >> _______________________________________________
>>> >> FreeSWITCH-users mailing list
>>> >> <FreeSWITCH-users at lists.freeswitch.org>
>>> FreeSWITCH-users at lists.freeswitch.org
>>> >> <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> <http://www.freeswitch.org/>http://www.freeswitch.org
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > <FreeSWITCH-users at lists.freeswitch.org>
>>> FreeSWITCH-users at lists.freeswitch.org
>>> > <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > <http://www.freeswitch.org/>http://www.freeswitch.org
>>> >
>>> >
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>> FreeSWITCH-users at lists.freeswitch.org
>>> <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>  <http://www.freeswitch.org/>http://www.freeswitch.org
>>>
>>  _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100219/ef0dac4c/attachment.html 

From errotan at gmail.com  Fri Feb 19 09:25:50 2010
From: errotan at gmail.com (=?iso-8859-1?q?Pusk=E1s_Zsolt?=)
Date: Fri, 19 Feb 2010 18:25:50 +0100
Subject: [Freeswitch-users] mod_fax undefined symbol
In-Reply-To: <7b197bef1002190308u375c16f0p9ed83d7e8e19cf89@mail.gmail.com>
References: <6FCE44FF-70C1-412D-B722-4C8D359381BA@ttnc.co.uk>
	<B05DBE65-CF57-4E13-9CE2-E490128F1C41@ttnc.co.uk>
	<7b197bef1002190308u375c16f0p9ed83d7e8e19cf89@mail.gmail.com>
Message-ID: <201002191825.50957.errotan@gmail.com>

2010. febru?r 19. 12.08.25 Giovanni Maruzzelli d?tummal ezt ?rta:
> On Fri, Feb 19, 2010 at 11:58 AM, TTNC - Technical <technical at ttnc.co.uk> 
wrote:
> > Hmm... that's a bit odd. You would have thought that if FreeSWITCH is
> > compiled statically against it's own libtiff - then anything
> > Debian-centric shouldn't affect it?
> 
> yes, definitely
> 
> probably in your debian build libtiff is not compiled, or mod_fax is
> not linked (statically) with it
> 
> "normal" build, as in non-debian, works fine
> -gm
> 
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 

Just compiled svn16700 on Debian "lenny" x86, mod_fax loads and works 
perfectly. I have an ongoing compile on another machine (amd64) if It don't 
works i will send a mail (in 1 hour) otherwise consider it working.



From technical at ttnc.co.uk  Fri Feb 19 09:41:22 2010
From: technical at ttnc.co.uk (TTNC - Technical)
Date: Fri, 19 Feb 2010 17:41:22 +0000
Subject: [Freeswitch-users] mod_fax undefined symbol
In-Reply-To: <4B7EB86B.9010002@gmail.com>
References: <6FCE44FF-70C1-412D-B722-4C8D359381BA@ttnc.co.uk>	<0F7476C7-FA7E-441F-9E17-EB2D23CD6055@jerris.com>	<EBF5F462-9A4F-4699-95FB-6947A9D2D55C@ttnc.co.uk>	<7b197bef1002190146x53492b85va88e4025c59e37f3@mail.gmail.com>	<496A4C3D-D185-4A86-BCA7-E4F5C05DA929@ttnc.co.uk>	<8D506E69-CB30-40BD-8AB9-9524C18E4503@ttnc.co.uk>	<7b197bef1002190234m58c7e8cbyd945af9ee0f5a3a@mail.gmail.com>	<D04CFC8A-4909-4E32-A480-567E712AE63A@ttnc.co.uk>	<2547643E-D259-46CE-919D-51393A4151BC@jerris.com>
	<CAC92986-7EA7-446B-8A95-50BF2FC25FCB@ttnc.co.uk>
	<4B7EB86B.9010002@gmail.com>
Message-ID: <743BF178-CB48-41EC-A16B-31188ECE9496@ttnc.co.uk>


On 19 Feb 2010, at 16:12, Tamas wrote:

> Hi,
> 
> What does 'ldd mod_fax.so' say?

voipin2:/opt/freeswitch/mod# ldd mod_fax.so 
	linux-gate.so.1 =>  (0xb8015000)
	libm.so.6 => /lib/i686/cmov/libm.so.6 (0xb7f75000)
	libfreeswitch.so.1 => /opt/freeswitch/lib/libfreeswitch.so.1 (0xb7da1000)
	libpthread.so.0 => /lib/i686/cmov/libpthread.so.0 (0xb7d87000)
	libc.so.6 => /lib/i686/cmov/libc.so.6 (0xb7c40000)
	/lib/ld-linux.so.2 (0xb8016000)
	libssl.so.0.9.8 => /usr/lib/i686/cmov/libssl.so.0.9.8 (0xb7bf9000)
	libcrypto.so.0.9.8 => /usr/lib/i686/cmov/libcrypto.so.0.9.8 (0xb7aa2000)
	libncurses.so.5 => /lib/libncurses.so.5 (0xb7a6a000)
	libstdc++.so.6 => /usr/lib/libstdc++.so.6 (0xb7978000)
	libgcc_s.so.1 => /lib/libgcc_s.so.1 (0xb795b000)
	libodbc.so.1 => /usr/lib/libodbc.so.1 (0xb78f9000)
	libdl.so.2 => /lib/i686/cmov/libdl.so.2 (0xb78f5000)
	libz.so.1 => /usr/lib/libz.so.1 (0xb78e1000)
	libltdl.so.7 => /usr/lib/libltdl.so.7 (0xb78d8000)

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100219/000d8e27/attachment-0001.html 

From technical at ttnc.co.uk  Fri Feb 19 09:44:32 2010
From: technical at ttnc.co.uk (TTNC - Technical)
Date: Fri, 19 Feb 2010 17:44:32 +0000
Subject: [Freeswitch-users] mod_fax undefined symbol
In-Reply-To: <201002191825.50957.errotan@gmail.com>
References: <6FCE44FF-70C1-412D-B722-4C8D359381BA@ttnc.co.uk>
	<B05DBE65-CF57-4E13-9CE2-E490128F1C41@ttnc.co.uk>
	<7b197bef1002190308u375c16f0p9ed83d7e8e19cf89@mail.gmail.com>
	<201002191825.50957.errotan@gmail.com>
Message-ID: <AB3AD9C8-C162-4DB5-8095-97CA83EDDE57@ttnc.co.uk>


On 19 Feb 2010, at 17:25, Pusk?s Zsolt wrote:
> Just compiled svn16700 on Debian "lenny" x86, mod_fax loads and works 
> perfectly. I have an ongoing compile on another machine (amd64) if It don't 
> works i will send a mail (in 1 hour) otherwise consider it working.
> 

How did you compile it? Using dpkg-buildpackage or via make/make install?

Do you have any debian versions of libtiff4(-dev) installed?
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100219/49daefa6/attachment.html 

From yehavi.bourvine at gmail.com  Fri Feb 19 09:50:42 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Fri, 19 Feb 2010 19:50:42 +0200
Subject: [Freeswitch-users] Announcement: FreeSWITCH Add Broadsoft SCA
	Support
In-Reply-To: <191c3a031002190908r3922efccqc6d223a5fabd458b@mail.gmail.com>
References: <87f2f3b91001111742k691966cfs82515db77c13d4d4@mail.gmail.com>
	<191c3a031002170557g5d8c9df9xf40d1b74b9a6f19d@mail.gmail.com>
	<d361ca7b1002170700x15630ab2tabc6f3d91ebc3303@mail.gmail.com>
	<ADC8B813-4161-481C-877E-93FC012858B4@freeswitch.org>
	<d361ca7b1002182101p60b4e8f8hf4dc95e1d2251e30@mail.gmail.com>
	<8b1c9cda1002182144m65ccf979w71384a1e7a55e2ea@mail.gmail.com>
	<d361ca7b1002182154o52352c62h8c0ec02fa66e8444@mail.gmail.com>
	<2F87B95C-EA23-4072-B845-3F7AE0617BE3@jerris.com>
	<d361ca7b1002182346k71fd602fh6e6725834a6de528@mail.gmail.com>
	<191c3a031002190908r3922efccqc6d223a5fabd458b@mail.gmail.com>
Message-ID: <d361ca7b1002190950t7b4da3ccy245c1667f82092e0@mail.gmail.com>

I see  now that Polycom released newer versions of firmware for the phones
recently. On Sunday's mornning I'll download them and retest with the latest
FreeSwitch snapshot.

                   __Yehavi:

 2010/2/19 Anthony Minessale <anthony.minessale at gmail.com>

> go see my comments on that bug note.
> be prepared to give us ssh access and call or irc so we can can see you
> reproducing it.
>
> If you are not on the latest firmware on all the phones, we will not
> continue with this process.
>
>
>
>
> On Fri, Feb 19, 2010 at 1:46 AM, Yehavi Bourvine <
> yehavi.bourvine at gmail.com> wrote:
>
>>  A jira issue has been created: *MODSOFIA-61*<http://jira.freeswitch.org/browse/MODSOFIA-61>
>> .
>>
>>                   Thanks, __Yehavi:
>>
>>  2010/2/19 Michael Jerris <mike at jerris.com>
>>
>>>  If this issue is not already on jira could you please make sure it gets
>>> added?
>>>
>>> Mike
>>>
>>>
>>> On Feb 19, 2010, at 12:54 AM, Yehavi Bourvine <yehavi.bourvine at gmail.com>
>>> wrote:
>>>
>>>     Hello Gabe,
>>>
>>>   As you can see - Brian is actively investigating it, so you can expect
>>> for some fix soon...
>>>
>>>                        Regards, __Yehavi:
>>>
>>> 2010/2/19 Gabriel Kuri < <gkuri at ieee.org>gkuri at ieee.org>
>>>
>>>> >  When a call arrives, both ring; the one that did not answer gets only
>>>> a
>>>> > cancel mesage without any further notification that the extension is
>>>> in use
>>>> > by the other phone.
>>>>
>>>> These are the same exact symptoms I posted about earlier this week,
>>>> with the Cisco SPA-5xx series phones. I still have yet to figure out
>>>> why this is happening, if you find out what's going on, please post
>>>> back the solution, I'd like to know the resolution.
>>>>
>>>> Thanks,
>>>> Gabe
>>>>
>>>>
>>>>
>>>> On Thu, Feb 18, 2010 at 9:01 PM, Yehavi Bourvine
>>>> < <yehavi.bourvine at gmail.com>yehavi.bourvine at gmail.com> wrote:
>>>> > Thanks Brian. It now works better, but not fully (using 16659M).
>>>> >
>>>> > What happens is:
>>>> >
>>>> > When one of the Polycoms seize the line it is ok - the other phone
>>>> gets
>>>> > notification and the extension status is "in use".
>>>> > When one of the Polycom phones initiates a call - all is ok:
>>>> >
>>>> > The other side sees that the extension is in use.
>>>> > When it is put to hold all phones who share this extension see it and
>>>> can
>>>> > pick the call.
>>>> >
>>>>
>>>> >
>>>>  >                          Thanks! __Yehavi:
>>>> >
>>>> > 2010/2/17 Brian West < <brian at freeswitch.org>brian at freeswitch.org>
>>>> >>
>>>> >> Step 1. Enable manage-shared-appearance=true
>>>> >>
>>>> >> Step 2. Now in the phone's config Configure the phone as usually, set
>>>> the
>>>> >> line shared and DO NOT set the third party name.
>>>> >>
>>>> >> Step 3. Reboot
>>>> >>
>>>> >> It should work.
>>>> >>
>>>> >> I wish someone that has this working would write some wiki docs these
>>>> >> threads about it not working are getting rather old when I know for a
>>>> fact
>>>> >> they work fine.
>>>> >>
>>>> >> The gateway info missing is a gateway you have configured getting a
>>>> >> notify.  It has nothing to do with SCA.
>>>> >>
>>>> >> /b
>>>> >>
>>>> >> On Feb 17, 2010, at 9:00 AM, Yehavi Bourvine wrote:
>>>> >>
>>>> >> > .
>>>> >>
>>>> >>
>>>> >> _______________________________________________
>>>> >> FreeSWITCH-users mailing list
>>>> >> <FreeSWITCH-users at lists.freeswitch.org>
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> >> <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> >> <http://www.freeswitch.org/>http://www.freeswitch.org
>>>> >
>>>> > _______________________________________________
>>>> > FreeSWITCH-users mailing list
>>>> > <FreeSWITCH-users at lists.freeswitch.org>
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> > <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> > UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> > <http://www.freeswitch.org/>http://www.freeswitch.org
>>>> >
>>>> >
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> <FreeSWITCH-users at lists.freeswitch.org>
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> <http://www.freeswitch.org/>http://www.freeswitch.org
>>>>
>>>  _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100219/bd79ca33/attachment.html 

From errotan at gmail.com  Fri Feb 19 10:04:39 2010
From: errotan at gmail.com (=?utf-8?q?Pusk=C3=A1s_Zsolt?=)
Date: Fri, 19 Feb 2010 19:04:39 +0100
Subject: [Freeswitch-users] mod_fax undefined symbol
In-Reply-To: <AB3AD9C8-C162-4DB5-8095-97CA83EDDE57@ttnc.co.uk>
References: <6FCE44FF-70C1-412D-B722-4C8D359381BA@ttnc.co.uk>
	<201002191825.50957.errotan@gmail.com>
	<AB3AD9C8-C162-4DB5-8095-97CA83EDDE57@ttnc.co.uk>
Message-ID: <201002191904.39081.errotan@gmail.com>

2010. febru?r 19. 18.44.32 TTNC - Technical d?tummal ezt ?rta:
> On 19 Feb 2010, at 17:25, Pusk?s Zsolt wrote:
> > Just compiled svn16700 on Debian "lenny" x86, mod_fax loads and works
> > perfectly. I have an ongoing compile on another machine (amd64) if It
> > don't works i will send a mail (in 1 hour) otherwise consider it working.
> 
> How did you compile it? Using dpkg-buildpackage or via make/make install?
> 
> Do you have any debian versions of libtiff4(-dev) installed?
> 

Okay i can confirm that mod_fax works on Debian "lenny" amd64 but it don't work 
on Debian "testing,squeeze" amd64.

2010-02-19 18:49:14.610297 [CRIT] switch_loadable_module.c:882 Error Loading 
module /usr/local/freeswitch/mod/mod_fax.so
**/usr/local/freeswitch/mod/mod_fax.so: undefined symbol: 
TIFFDefaultStripSize**

I haven't tried to compile mod_fax on testing before so i don't know what is 
causeing the problem :(

# ldd mod_fax.so
        linux-vdso.so.1 =>  (0x00007fff106f6000)
        libm.so.6 => /lib/libm.so.6 (0x00007f506b711000)
        libfreeswitch.so.1 => /usr/local/freeswitch/lib/libfreeswitch.so.1 
(0x00007f506b345000)
        libpthread.so.0 => /lib/libpthread.so.0 (0x00007f506b128000)
        libc.so.6 => /lib/libc.so.6 (0x00007f506add4000)
        libssl.so.0.9.8 => /usr/lib/libssl.so.0.9.8 (0x00007f506ab82000)
        libcrypto.so.0.9.8 => /usr/lib/libcrypto.so.0.9.8 (0x00007f506a7e2000)
        libncurses.so.5 => /lib/libncurses.so.5 (0x00007f506a59d000)
        libstdc++.so.6 => /usr/lib/libstdc++.so.6 (0x00007f506a28d000)
        libgcc_s.so.1 => /lib/libgcc_s.so.1 (0x00007f506a076000)
        libodbc.so.1 => /usr/lib/libodbc.so.1 (0x00007f5069e17000)
        /lib64/ld-linux-x86-64.so.2 (0x00007f506bc31000)
        libdl.so.2 => /lib/libdl.so.2 (0x00007f5069c13000)
        libz.so.1 => /usr/lib/libz.so.1 (0x00007f50699fb000)
        libltdl.so.7 => /usr/lib/libltdl.so.7 (0x00007f50697f2000)

Recently in debian "testing" libtiff4 and libjpeg is upgraded:
libtiff 3.9.2-3+b1
libjpeg62 6b-16.1
libjeg8 8-2.1

Q&A:
Q: How did you compile it? Using dpkg-buildpackage or via make/make install?
A: svn-clean ./bootsrap ./configure make etc.

Q: Do you have any debian versions of libtiff4(-dev) installed?
A: Yes:3.8.2-11.2

I open a jira for this.


From scottferri09 at gmail.com  Fri Feb 19 12:01:34 2010
From: scottferri09 at gmail.com (Scott Fernandez)
Date: Sat, 20 Feb 2010 01:31:34 +0530
Subject: [Freeswitch-users] Establishing a Call from .Net based
	application
In-Reply-To: <922191.64755.qm@web33507.mail.mud.yahoo.com>
References: <6E8D2069C08AA84A83D336E996AE4C67032C9D5BAD@mse17be1.mse17.exchange.ms>
	<922191.64755.qm@web33507.mail.mud.yahoo.com>
Message-ID: <df97b9c51002191201n16183ebdk858fe50a0d479f0b@mail.gmail.com>

Thanks a lot Deigo and Michael.

Will work on this and let you know if I face any problems.

Thanks,
Scott

On Fri, Feb 19, 2010 at 5:04 AM, Diego Toro <dftoro at yahoo.com> wrote:

> The managed module is loaded as a module during the startup of FreeSWITCH
> if set in modules.conf.xml or through the command "load mod_managed" must
> keep in mind that there is a directory "mod/managed. As mod_managed is
> loaded into FreeSWITCH process to take control of the call must be running
> FreeSWITCH. So to "talk" with FreeSWITCH is not necessary to know the IP,
> the IP depends on the profile you've defined in the configuration of the
> module sofia. If you need the local address of the box running FreeSWITCH
> try expand variable $${local_ip_v4} which is assigned automatically by
> FreeSWITCH.
>
> Being more clear, when you use mod_managed including <action
> application="managed" data="yourclassname"/>
>  in a dialplan already have way to run your C# code.
>
> Now, if you need is to have control of the call to answer, originate, etc,
> without the application run inside FreeSWITCH process, you can use managed
> ESL (see examples in libs/esl/managed) this library allows your code using
> events "talk" with FreeSWITCH.
>
> Diego Toro
> http://lacarretade.blogspot.com/
>
>
> --- On Thu, 2/18/10, Michael Giagnocavo <mgg at giagnocavo.net> wrote:
>
> > From: Michael Giagnocavo <mgg at giagnocavo.net>
> > Subject: Re: [Freeswitch-users] Establishing a Call from .Net based
> application
> > To: "freeswitch-users at lists.freeswitch.org" <
> freeswitch-users at lists.freeswitch.org>
> > Date: Thursday, February 18, 2010, 6:12 AM
> > I?m not sure what the
> > FreeSWITCH APIs are to figure out what IP Sofia SIP has
> > bound to. Whatever it is, you?d call the same thing in
> > C#. What do you want to do with the API?  mod_managed.dll or .so is the
> > FreeSWITCH native code module that loads the CLR or Mono
> > into the FreeSWITCH process and loads
> > FreeSWITCH.Managed.dll. The managed DLL contains the bulk of
> > the managed-unmanaged interop code (.NET definitions of all
> > the FS C functions).   -Michael  From:
> > freeswitch-users-bounces at lists.freeswitch.org
> > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On
> > Behalf Of Scott Fernandez
> > Sent: Thursday, February 18, 2010 1:12 AM
> > To: freeswitch-users at lists.freeswitch.org
> > Subject: Re: [Freeswitch-users] Establishing a Call
> > from .Net based application
> >  Hi Diego & Michael,
> >
> > Thanks for your reply and support.
> >
> > However, I have some clarifications required from both of
> > you.
> >
> > 1. Here is the question for Diego,
> >
> > Simple Example:
> >
> > using FreeSWITCH;
> > using FreeSWITCH.Native;
> >
> > namespace BITS.Ivr.Bp.Server
> > {
> >  public class CIV_BPFSProcess : FreeSWITCH.IAppPlugin
> > {
> >  public void Run(AppContext context)
> >  {
> >   //answer call
> >   context.Session.Answer();
> >   //sleep 2 seconds
> >   context.Session.sleep(2000, 1);
> >   //hangup call
> >   context.Session.Hangup("NORMAL_CLEARING");
> >  }
> >  }
> > }
> > I understand that the concept of your example code.
> > However, would like to know as to how would my .NET C# know the
> > IP address of Freeswitch to talk to it as there is no
> > indication for that?. If not here, where would we need to
> > reference the IP address of FS in .NET code?
> >
> > I guess the IP address of FS needs to be mentioned in the
> > Target section of the below web.config file in .NET. If I am
> > right, how to specify the IP address over here. If I am
> > wrong, please let me know where do we need to mention the IP
> > address of FS.
> >
> >     <configuration>
> >
> > <dllmap dll="mod_managed.dll"
> > target="mod_managed.so"/>
> >     </configuration>
> >
> >
> > 2. Here is the question for Michael,
> >
> > You mentioned that "mod_managed.so will
> > be in your freeswitch mod directory". This is
> > very clear and what is mod_managed.dll in my .NET
> > application and the purpose of it?
> >
> > Thanks for all your help.
> >
> > Regards,
> > Scott.
> >
> >
> > On Sun, Feb 14, 2010 at 1:15
> > AM, Michael Giagnocavo <mgg at giagnocavo.net>
> > wrote:
> > 2. There is a configuration settings required to Map the
> > "DLL" to ".so" object in CentOS.
> > Now, the question is which DLL and .so file to be made
> > available and where??If you are
> > experiencing NullReferenceExceptions with any plugin run
> > through the dialplan, make sure you have included the
> > appropriate entry in your dllmap
> > configuration:  <dllmap dll="mod_managed.dll"
> > target="mod_managed.so"
> > os="!windows"/>?mod_managed.so will
> > be in your freeswitch mod directory.
> > All I need is to initiate a call from .NET application and
> > then it should talk to mod_managed module and establish a
> > call. Secondly, I need to know the status of the call such
> > as Ringing, Active, Hangup etc. To initiate a
> > call, try ManagedSession.Originate.-Michael
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> > -----Inline Attachment Follows-----
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100220/dcb92b1b/attachment-0001.html 

From ledoktre at meanie.us  Fri Feb 19 13:30:42 2010
From: ledoktre at meanie.us (Doc)
Date: Fri, 19 Feb 2010 15:30:42 -0600
Subject: [Freeswitch-users] Skypiax- how to know which card to use?
Message-ID: <4B7F0302.3060303@meanie.us>

Hey guys,

I am attempting to setup Skypiax and Freeswitch on Ubuntu Hardy.  I've 
had a couple of previous problems which updating to SVN trunk release 
seemed to resolve, but I've got one lingering one to ask about.

When a call comes in to my SkypeIn number (PSTN that rings my Skype), FS 
picks up and rings the desired extension.  No problems there.  What 
happens though is, the person that is on the extension in the office can 
hear the calling party just fine, but the calling party hears stuttered 
choppy sound.

I've already tested my internet connection, and through QOS have 
assigned my test box 512K up and 512K down, so on one test call 
bandwidth should not be an issue.

What do you think causes the stuttering on my audio for the calling party?

Do you think it might be a codec?  I have everything setup right now to 
use g711u.  I'd like to use something smaller and higher quality, but 
I'm not sure what is a better option.

I had made some phone calls as listed above and they tested out just 
fine, but in that same time frame I also recompiled Alsa drivers with 
the Skypiax dummy file, so do you think there might be something in that 
causing the stutter?

Looking for some advice.

Thanks,

Doc


From brian at microcomaustralia.com.au  Fri Feb 19 14:38:36 2010
From: brian at microcomaustralia.com.au (Brian May)
Date: Sat, 20 Feb 2010 09:38:36 +1100
Subject: [Freeswitch-users] building for Lenny
In-Reply-To: <4C13BCBF-1DA0-47AE-A749-738AD6D8FE48@jerris.com>
References: <3c5cf5261002180046vb0015e7t787599c9903aedc@mail.gmail.com>
	<33c87fa31002180056g51ef8496qa9a35746d0fe75ec@mail.gmail.com>
	<3c5cf5261002180110he0ae815uc713f77f289ac625@mail.gmail.com>
	<a9f4977a70b3fcf1b30413ea562ca962.squirrel@www.webcontracts.co.uk>
	<20100218161741.GC4236@base.carmickle.com>
	<3c5cf5261002181542y43adb7a7ofe2a15c38baf0327@mail.gmail.com>
	<4C13BCBF-1DA0-47AE-A749-738AD6D8FE48@jerris.com>
Message-ID: <3c5cf5261002191438o47aabde9tb588fe3ef23f27bc@mail.gmail.com>

On 19 February 2010 18:10, Michael Jerris <mike at jerris.com> wrote:
> Please create me a bug on http://jira.freeswitch.org for this issue.

Ok, hope I did that right. Now done.

Thanks
-- 
Brian May <brian at microcomaustralia.com.au>


From gmaruzz at celliax.org  Fri Feb 19 14:41:41 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Fri, 19 Feb 2010 23:41:41 +0100
Subject: [Freeswitch-users] Skypiax- how to know which card to use?
In-Reply-To: <4B7F0302.3060303@meanie.us>
References: <4B7F0302.3060303@meanie.us>
Message-ID: <7b197bef1002191441u455e8d8an4eb03c460254c6d3@mail.gmail.com>

On Fri, Feb 19, 2010 at 10:30 PM, Doc <ledoktre at meanie.us> wrote:
>
> I am attempting to setup Skypiax and Freeswitch on Ubuntu Hardy. ?I've
> had a couple of previous problems which updating to SVN trunk release
> seemed to resolve, but I've got one lingering one to ask about.

Hi Doc,

before to delve in the troubleshooting, I have to say that I'm
modifying the audio skypiax code in svn, so maybe it's just my fault
;).

please be patient for a little while, I hope to have done with it in a
couple days.

I'll announce to the mailing list when done.

In the mean time, at least one good news for you user of SkypeIn
service: a new feature of mod_skypiax is intended to recognize the
DTMFs coming from SkypeIn, so the incoming calls will be able to use
ivr, voicemail, etc.

>
> When a call comes in to my SkypeIn number (PSTN that rings my Skype), FS
> picks up and rings the desired extension. ?No problems there. ?What
> happens though is, the person that is on the extension in the office can
> hear the calling party just fine, but the calling party hears stuttered
> choppy sound.
> What do you think causes the stuttering on my audio for the calling party?
>
> Do you think it might be a codec? ?I have everything setup right now to
> use g711u. ?I'd like to use something smaller and higher quality, but
> I'm not sure what is a better option.

I would exclude is a codec problem

>
> I had made some phone calls as listed above and they tested out just
> fine, but in that same time frame I also recompiled Alsa drivers with
> the Skypiax dummy file, so do you think there might be something in that
> causing the stutter?

more probably is some change I made to the code, please be patient for
a while, couple days.

-giovanni


-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From ledoktre at meanie.us  Fri Feb 19 14:47:41 2010
From: ledoktre at meanie.us (Doc)
Date: Fri, 19 Feb 2010 16:47:41 -0600
Subject: [Freeswitch-users] Skypiax- how to know which card to use?
In-Reply-To: <7b197bef1002191441u455e8d8an4eb03c460254c6d3@mail.gmail.com>
References: <4B7F0302.3060303@meanie.us>
	<7b197bef1002191441u455e8d8an4eb03c460254c6d3@mail.gmail.com>
Message-ID: <4B7F150D.2040204@meanie.us>

Giovanni,

Awesome to hear from the creator of Skypiax.  You might be right, maybe 
it is the code in SVN.  I had recently updated SVN (at the time that I 
re-compiled with the skypiax dummy file).  I will look forward to an 
update on this.  I am also excited to hear that you have some new things 
coming that will allow for DTMF and IVR's, etc.  Yay!

Thanks for your reply,

Doc

> Hi Doc,
>
> before to delve in the troubleshooting, I have to say that I'm
> modifying the audio skypiax code in svn, so maybe it's just my fault
> ;).
>
> please be patient for a little while, I hope to have done with it in a
> couple days.
>
> I'll announce to the mailing list when done.
>
> In the mean time, at least one good news for you user of SkypeIn
> service: a new feature of mod_skypiax is intended to recognize the
> DTMFs coming from SkypeIn, so the incoming calls will be able to use
> ivr, voicemail, etc.
>
>   
>> When a call comes in to my SkypeIn number (PSTN that rings my Skype), FS
>> picks up and rings the desired extension.  No problems there.  What
>> happens though is, the person that is on the extension in the office can
>> hear the calling party just fine, but the calling party hears stuttered
>> choppy sound.
>> What do you think causes the stuttering on my audio for the calling party?
>>
>> Do you think it might be a codec?  I have everything setup right now to
>> use g711u.  I'd like to use something smaller and higher quality, but
>> I'm not sure what is a better option.
>>     
>
> I would exclude is a codec problem
>
>   
>> I had made some phone calls as listed above and they tested out just
>> fine, but in that same time frame I also recompiled Alsa drivers with
>> the Skypiax dummy file, so do you think there might be something in that
>> causing the stutter?
>>     
>
> more probably is some change I made to the code, please be patient for
> a while, couple days.
>
> -giovanni
>
>
>   


From spiritonly at gmail.com  Fri Feb 19 17:19:50 2010
From: spiritonly at gmail.com (=?UTF-8?B?5p2o5rGf6aqF?=)
Date: Sat, 20 Feb 2010 09:19:50 +0800
Subject: [Freeswitch-users] How to exchange media when I developed new
	endpoint module?
In-Reply-To: <ebbc2d551002091932u52dbf1b6jfecf2dfdb43b8a7c@mail.gmail.com>
References: <93b0f8ce1002091914q581f12fvd1de01e2181f0f3a@mail.gmail.com>
	<ebbc2d551002091932u52dbf1b6jfecf2dfdb43b8a7c@mail.gmail.com>
Message-ID: <93b0f8ce1002191719j71a7de5j374de41836ee2923@mail.gmail.com>

I have read mod_loopback and other endpoint modules, I found that it use
'switch_ivr_uuid_bridge' to bridge two session in mod_loopback, but in
others there are not 'switch_ivr_uuid_bridge' or bridge functions. So what
is different and which endpoint module should I consult?


2010/2/10 Jo?o Mesquita <jmesquita at freeswitch.org>

> You should look at read_frame and write_frame implementations of other
> endpoint modules.
>
> This should pretty much tell you how things work...
>
> Jo?o Mesquita
>
>
> On Wed, Feb 10, 2010 at 1:14 AM, ??? <spiritonly at gmail.com> wrote:
>
>> Hi,
>>     I am developping a new endpoint module, now I can make an inbound call
>> and execute IVR.
>> When I make an outbound call and bridge the inbound leg and outbound leg,
>> I receive remote alerting and pickup remote phone but there isn't
>> any voice exchange.
>>     So how to exchange media next?
>> ----------------------------------------------------------------------
>> gtalk:    spiritonly at gmail.com
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100220/c6050d87/attachment.html 

From spiritonly at gmail.com  Fri Feb 19 17:22:34 2010
From: spiritonly at gmail.com (=?UTF-8?B?5p2o5rGf6aqF?=)
Date: Sat, 20 Feb 2010 09:22:34 +0800
Subject: [Freeswitch-users] How to exchange media when I developed new
	endpoint module?
In-Reply-To: <4021C303-1AC3-48F2-A17B-63C02C1C54D4@freeswitch.org>
References: <93b0f8ce1002091914q581f12fvd1de01e2181f0f3a@mail.gmail.com>
	<ebbc2d551002091932u52dbf1b6jfecf2dfdb43b8a7c@mail.gmail.com>
	<4021C303-1AC3-48F2-A17B-63C02C1C54D4@freeswitch.org>
Message-ID: <93b0f8ce1002191722h5782b787qe56a06dd0cb94e76@mail.gmail.com>

Do you know mod_khomp? You can found it in FS wiki. I am developing an
endpoint module like it.
So you can give me some advice to bridge two session?

On Wed, Feb 10, 2010 at 11:44 AM, Brian West <brian at freeswitch.org> wrote:

> But the bigger question is what protocol are you doing that you have to
> create your own endpoint module?
>
> /b
>
> On Feb 9, 2010, at 9:32 PM, Jo?o Mesquita wrote:
>
> > You should look at read_frame and write_frame implementations of other
> endpoint modules.
> >
> > This should pretty much tell you how things work...
> >
> > Jo?o Mesquita
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100220/e7583a54/attachment.html 

From gamar at center.com  Fri Feb 19 06:07:33 2010
From: gamar at center.com (Gilbert Amar)
Date: Fri, 19 Feb 2010 15:07:33 +0100
Subject: [Freeswitch-users] Troubles bridging calls with mod_opal
Message-ID: <F9E933AA84A243688042045839DDEDFE@gamar>

Hello,

I am having troubles to make FS/opal works. Please do not ask me to use SIP 
instead of H323 it is out of scope.
I am trying here to bridge an incoming h323 call to another h323 party.
The result is that most of the time (sometime it actually works) the bridge 
is done but the callee cannot be heard by the caller.

If I dial directly the callee without FS/opal it works.
If I dial from a sip client like xlite then it works.

Here is our setup:
 FreeSWITCH Version 1.0.trunk (16659M) (at 2010-02-19 12:57 UTC)
 FS runs on a CentOs 5.2
 The calling party is a soft OpenPhone
 The called party is a Swissvoice IP10S


I try to enable/disable several codecs with no success.
Any idea where I should look ?

Please find attach the FS log and the Openphone log


Thanks
Gilbert 
-------------- next part --------------
A non-text attachment was scrubbed...
Name: ko.zip
Type: application/octet-stream
Size: 45635 bytes
Desc: not available
Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100219/5f0df6c2/attachment-0001.obj 

From gamar at center.com  Fri Feb 19 09:53:32 2010
From: gamar at center.com (Gilbert Amar)
Date: Fri, 19 Feb 2010 18:53:32 +0100
Subject: [Freeswitch-users] Troubles bridging calls with mod_opal
Message-ID: <91C2F3BC25F34E7AB6DA0441D390EA96@gamar>

Hello,

I am having troubles to make FS/opal works. Please do not ask me to use SIP 
instead of H323 it is out of scope.
I am trying here to bridge an incoming h323 call to another h323 party.
The result is that most of the time (sometime it actually works) the bridge 
is done but the callee cannot be heard by the caller.

If I dial directly the callee without FS/opal it works.
If I dial from a sip client like xlite then it works.

Here is our setup:
 FreeSWITCH Version 1.0.trunk (16659M) (at 2010-02-19 12:57 UTC)
 FS runs on a CentOs 5.2
 The calling party is a soft OpenPhone
 The called party is a Swissvoice IP10S


I try to enable/disable several codecs with no success.
Any idea where I should look ?

Please find attach the FS log and the Openphone log


Thanks
Gilbert 
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100219/3842edb1/attachment-0001.html 
-------------- next part --------------
A non-text attachment was scrubbed...
Name: ko.zip
Type: application/octet-stream
Size: 45635 bytes
Desc: not available
Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100219/3842edb1/attachment-0001.obj 

From mike at jerris.com  Fri Feb 19 18:00:31 2010
From: mike at jerris.com (Michael Jerris)
Date: Fri, 19 Feb 2010 21:00:31 -0500
Subject: [Freeswitch-users] mod_fax undefined symbol
In-Reply-To: <201002191904.39081.errotan@gmail.com>
References: <6FCE44FF-70C1-412D-B722-4C8D359381BA@ttnc.co.uk>
	<201002191825.50957.errotan@gmail.com>
	<AB3AD9C8-C162-4DB5-8095-97CA83EDDE57@ttnc.co.uk>
	<201002191904.39081.errotan@gmail.com>
Message-ID: <E6568E57-A3E2-4650-A8CD-70B9C2846AF4@jerris.com>

replying with more details on jira.


On Feb 19, 2010, at 1:04 PM, Pusk?s Zsolt wrote:

> 2010. febru?r 19. 18.44.32 TTNC - Technical d?tummal ezt ?rta:
>> On 19 Feb 2010, at 17:25, Pusk?s Zsolt wrote:
>>> Just compiled svn16700 on Debian "lenny" x86, mod_fax loads and works
>>> perfectly. I have an ongoing compile on another machine (amd64) if It
>>> don't works i will send a mail (in 1 hour) otherwise consider it working.
>> 
>> How did you compile it? Using dpkg-buildpackage or via make/make install?
>> 
>> Do you have any debian versions of libtiff4(-dev) installed?
>> 
> 
> Okay i can confirm that mod_fax works on Debian "lenny" amd64 but it don't work 
> on Debian "testing,squeeze" amd64.
> 
> 2010-02-19 18:49:14.610297 [CRIT] switch_loadable_module.c:882 Error Loading 
> module /usr/local/freeswitch/mod/mod_fax.so
> **/usr/local/freeswitch/mod/mod_fax.so: undefined symbol: 
> TIFFDefaultStripSize**
> 
> I haven't tried to compile mod_fax on testing before so i don't know what is 
> causeing the problem :(
> 
> # ldd mod_fax.so
>        linux-vdso.so.1 =>  (0x00007fff106f6000)
>        libm.so.6 => /lib/libm.so.6 (0x00007f506b711000)
>        libfreeswitch.so.1 => /usr/local/freeswitch/lib/libfreeswitch.so.1 
> (0x00007f506b345000)
>        libpthread.so.0 => /lib/libpthread.so.0 (0x00007f506b128000)
>        libc.so.6 => /lib/libc.so.6 (0x00007f506add4000)
>        libssl.so.0.9.8 => /usr/lib/libssl.so.0.9.8 (0x00007f506ab82000)
>        libcrypto.so.0.9.8 => /usr/lib/libcrypto.so.0.9.8 (0x00007f506a7e2000)
>        libncurses.so.5 => /lib/libncurses.so.5 (0x00007f506a59d000)
>        libstdc++.so.6 => /usr/lib/libstdc++.so.6 (0x00007f506a28d000)
>        libgcc_s.so.1 => /lib/libgcc_s.so.1 (0x00007f506a076000)
>        libodbc.so.1 => /usr/lib/libodbc.so.1 (0x00007f5069e17000)
>        /lib64/ld-linux-x86-64.so.2 (0x00007f506bc31000)
>        libdl.so.2 => /lib/libdl.so.2 (0x00007f5069c13000)
>        libz.so.1 => /usr/lib/libz.so.1 (0x00007f50699fb000)
>        libltdl.so.7 => /usr/lib/libltdl.so.7 (0x00007f50697f2000)
> 
> Recently in debian "testing" libtiff4 and libjpeg is upgraded:
> libtiff 3.9.2-3+b1
> libjpeg62 6b-16.1
> libjeg8 8-2.1
> 
> Q&A:
> Q: How did you compile it? Using dpkg-buildpackage or via make/make install?
> A: svn-clean ./bootsrap ./configure make etc.
> 
> Q: Do you have any debian versions of libtiff4(-dev) installed?
> A: Yes:3.8.2-11.2



From infos at madovsky.org  Fri Feb 19 21:08:25 2010
From: infos at madovsky.org (Madovsky)
Date: Sat, 20 Feb 2010 00:08:25 -0500
Subject: [Freeswitch-users] codecs transcoding
Message-ID: <DA78008AFA4A414480CDB572EE602D7D@MOBILEE1705>

Hello,

is it need proxy_media on true to transcode codecs ?

Thanks

Farnck
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100220/4ba33482/attachment.html 

From jason at jasonjgw.net  Fri Feb 19 21:42:43 2010
From: jason at jasonjgw.net (Jason White)
Date: Sat, 20 Feb 2010 16:42:43 +1100
Subject: [Freeswitch-users] Troubles bridging calls with mod_opal
In-Reply-To: <91C2F3BC25F34E7AB6DA0441D390EA96@gamar>
References: <91C2F3BC25F34E7AB6DA0441D390EA96@gamar>
Message-ID: <20100220054243.GA4160@jdc.jasonjgw.net>

Gilbert Amar <gamar at center.com> wrote:
> I am having troubles to make FS/opal works. 

My impression is that more work is being devoted to mod_h323 than to mod_opal
to provide H323 support.

Try mod_h323 and see if it solves your problem.
If not, there are H323 users on the list who might be able to help. (I don't
use H323 personally.)



From pmhshz at gmail.com  Fri Feb 19 21:50:13 2010
From: pmhshz at gmail.com (MohammedShehzad)
Date: Sat, 20 Feb 2010 11:20:13 +0530
Subject: [Freeswitch-users] stream a file multicast with mod_esf
In-Reply-To: <a792304b1002190002u50ec6912g1905ffc63a119174@mail.gmail.com>
References: <B98B6921358D0C49BD21CD023F1F52C302623B4B@hbghs04a.ww300.siemens.net>
	<7C996136-963A-42D2-B0FE-D2729F968E52@freeswitch.org>
	<27071973.post@talk.nabble.com>
	<378CDEC3-5A41-4EF9-927A-311F2531E6AB@freeswitch.org>
	<a792304b1001080725y19ff28camd37425927550f311@mail.gmail.com>
	<191c3a031001081848w9036207yf6a4b473ec233243@mail.gmail.com>
	<85C74329-0D7C-48B2-B7A8-D4AA67C6E3F2@freeswitch.org>
	<a792304b1002182209m43ada13cp8ed3d91ad509f0a2@mail.gmail.com>
	<DBA4E291-9377-489A-841A-1A3D83880DF2@jerris.com>
	<a792304b1002190002u50ec6912g1905ffc63a119174@mail.gmail.com>
Message-ID: <a792304b1002192150n44cc500fnd5b2ecb7cff9dd1a@mail.gmail.com>

On Fri, Feb 19, 2010 at 1:32 PM, MohammedShehzad <pmhshz at gmail.com> wrote:

>
>
> On Fri, Feb 19, 2010 at 12:59 PM, Michael Jerris <mike at jerris.com> wrote:
>
>> Listening on multicast is noting special for multicast, it is just
>> like reading any other udp socket
>>
>> Mike
>>
>> Correct, but I have to play those audio stream back to caller taking care
> of the audio codec  and other things, do anybody have any idea in that part?
> Please let me know that.
> --
>
> -MohammedShehzad
>

I am able to receive the play the multicasted RAW PCMU RTP (modified the
skel of format provided by brian), so that caller can hear the multicast
which done by other Freeswitch server using mod_esf application, but when i
change the caller's codec from PCMU to something else, it breaks.

-- 

-MohammedShehzad
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100220/4cd80cf6/attachment.html 

From brian at microcomaustralia.com.au  Fri Feb 19 21:57:21 2010
From: brian at microcomaustralia.com.au (Brian May)
Date: Sat, 20 Feb 2010 16:57:21 +1100
Subject: [Freeswitch-users] openzap TDM400 card
Message-ID: <3c5cf5261002192157t25de5798xc10ab2a7755e67d0@mail.gmail.com>

Hello,

I think I have the config correct, and not confused FXO/FXS anywhere.

voyage:/opt/freeswitch/conf# ztcfg  -vv

Zaptel Version: 1.4.11
Echo Canceller: MG2
Configuration
======================


Channel map:

Channel 01: FXO Kewlstart (Default) (Slaves: 01)
Channel 02: FXO Kewlstart (Default) (Slaves: 02)
Channel 03: FXS Kewlstart (Default) (Slaves: 03)
Channel 04: FXS Kewlstart (Default) (Slaves: 04)

4 channels to configure.

voyage:/opt/freeswitch/conf# cat openzap.conf
[span zt FXO1]
name => OpenZAP-FXO1
number => 1
fxo-channel => 1

[span zt FXO2]
name => OpenZAP-FXO2
number => 2
fxo-channel => 2

[span zt FXS1]
name => OpenZAP-FXS1
number => 3
fxs-channel => 3

[span zt FXS2]
name => OpenZAP-FXS2
number => 4
fxs-channel => 4

voyage:/opt/freeswitch/conf# cat autoload_configs/openzap.conf.xml
<configuration name="openzap.conf" description="OpenZAP Configuration">
  <settings>
    <param name="debug" value="0"/>
    <!--<param name="hold-music" value="$${moh_uri}"/>-->
    <!--<param name="enable-analog-option" value="call-swap"/>-->
    <!--<param name="enable-analog-option" value="3-way"/>-->
  </settings>
  <!-- one entry here per openzap span -->
  <analog_spans>
    <span id="1">
      <param name="tonegroup" value="au"/>
      <param name="digit-timeout" value="2000"/>
      <param name="max-digits" value="11"/>
      <param name="dialplan" value="XML"/>
      <param name="context" value="default"/>
      <param name="enable-analog-option" value="3-way"/>
<!--      <param name="moh" value="$${base_dir}/sounds/Dardedisco.wav"/> -->
    </span>
    <span id="2">
      <param name="tonegroup" value="au"/>
      <param name="digit-timeout" value="2000"/>
      <param name="max-digits" value="11"/>
      <param name="dialplan" value="XML"/>
      <param name="context" value="default"/>
      <param name="enable-analog-option" value="3-way"/>
<!--      <param name="moh" value="$${base_dir}/sounds/Dardedisco.wav"/> -->
    </span>
    <span id="3">
      <param name="tonegroup" value="au"/>
      <param name="digit-timeout" value="2000"/>
      <param name="max-digits" value="11"/>
      <param name="dialplan" value="XML"/>
      <param name="context" value="default"/>
    </span>
    <span id="4">
      <param name="tonegroup" value="au"/>
      <param name="digit-timeout" value="2000"/>
      <param name="max-digits" value="11"/>
      <param name="dialplan" value="XML"/>
      <param name="context" value="default"/>
    </span>
  </analog_spans>
</configuration>


freeswitch at voyage> reload mod_openzap
2010-02-20 16:45:53.312813 [NOTICE] switch_loadable_module.c:464
Deleting Endpoint 'openzap'
2010-02-20 16:45:53.312813 [NOTICE] switch_loadable_module.c:545
Deleting Application 'disable_ec'
2010-02-20 16:45:53.312813 [NOTICE] switch_loadable_module.c:572
Deleting API Function 'oz'
2010-02-20 16:45:53.312813 [CONSOLE] switch_loadable_module.c:1277
Stopping: mod_openzap
2010-02-20 16:45:53.312813 [INFO] zap_io.c:269 Closing channel zt:1:1 fd:36
2010-02-20 16:45:53.312813 [INFO] zap_io.c:269 Closing channel zt:2:1 fd:40
2010-02-20 16:45:53.312813 [INFO] zap_io.c:269 Closing channel zt:3:1 fd:41
2010-02-20 16:45:53.312813 [INFO] zap_io.c:269 Closing channel zt:4:1 fd:42
2010-02-20 16:45:53.612813 [ERR] ozmod_analog.c:951 Failure Polling
event! [no matching descriptor]
2010-02-20 16:45:54.162813 [ERR] ozmod_analog.c:951 Failure Polling
event! [no matching descriptor]
2010-02-20 16:45:54.162813 [ERR] ozmod_analog.c:951 Failure Polling
event! [no matching descriptor]
2010-02-20 16:45:54.162813 [ERR] ozmod_analog.c:951 Failure Polling
event! [no matching descriptor]
2010-02-20 16:45:54.322813 [INFO] zap_io.c:2694 Unloading
/opt/freeswitch/mod/ozmod_analog.so
2010-02-20 16:45:54.322813 [INFO] zap_io.c:2679 Unloading IO zt
2010-02-20 16:45:54.322813 [INFO] zap_io.c:2694 Unloading
/opt/freeswitch/mod/ozmod_zt.so
2010-02-20 16:45:54.322813 [CONSOLE] switch_loadable_module.c:1297
mod_openzap unloaded.
2010-02-20 16:45:54.322813 [NOTICE] zap_io.c:2778 Modules configured: 1
2010-02-20 16:45:54.322813 [NOTICE] ozmod_zt.c:1161 Using Zaptel control device
2010-02-20 16:45:54.322813 [INFO] zap_io.c:2579 Loading IO from
/opt/freeswitch/mod/ozmod_zt.so [zt]
2010-02-20 16:45:54.322813 [INFO] ozmod_zt.c:556 Setting rxgain val to 0.000000
2010-02-20 16:45:54.322813 [INFO] ozmod_zt.c:565 Setting txgain val to 0.000000
2010-02-20 16:45:54.322813 [INFO] zap_io.c:2379 auto-loaded 'zt'
2010-02-20 16:45:54.322813 [WARNING] ozmod_zt.c:333 this ioctl fails
on older zaptel but is harmless if you used ztcfg
[device /dev/zap/channel chan 1 fd 27 (Invalid argument)]
2010-02-20 16:45:54.322813 [INFO] ozmod_zt.c:385 configuring device
/dev/zap/channel channel 1 as OpenZAP device 1:1 fd:36
2010-02-20 16:45:54.322813 [WARNING] ozmod_zt.c:333 this ioctl fails
on older zaptel but is harmless if you used ztcfg
[device /dev/zap/channel chan 2 fd 27 (Invalid argument)]
2010-02-20 16:45:54.322813 [INFO] ozmod_zt.c:385 configuring device
/dev/zap/channel channel 2 as OpenZAP device 2:1 fd:40
2010-02-20 16:45:54.322813 [WARNING] ozmod_zt.c:333 this ioctl fails
on older zaptel but is harmless if you used ztcfg
[device /dev/zap/channel chan 3 fd 27 (Invalid argument)]
2010-02-20 16:45:54.322813 [INFO] ozmod_zt.c:385 configuring device
/dev/zap/channel channel 3 as OpenZAP device 3:1 fd:41
2010-02-20 16:45:54.322813 [WARNING] ozmod_zt.c:333 this ioctl fails
on older zaptel but is harmless if you used ztcfg
[device /dev/zap/channel chan 4 fd 27 (Invalid argument)]
2010-02-20 16:45:54.322813 [INFO] ozmod_zt.c:385 configuring device
/dev/zap/channel channel 4 as OpenZAP device 4:1 fd:42
2010-02-20 16:45:54.322813 [INFO] zap_io.c:2502 Configured 4 channel(s)
2010-02-20 16:45:54.322813 [INFO] zap_io.c:2596 Loading SIG from
/opt/freeswitch/mod/ozmod_analog.so
2010-02-20 16:45:54.322813 [INFO] zap_io.c:2712 auto-loaded 'analog'
2010-02-20 16:45:54.358813 [CONSOLE] switch_loadable_module.c:900
Successfully Loaded [mod_openzap]
2010-02-20 16:45:54.362813 [NOTICE] switch_loadable_module.c:144
Adding Endpoint 'openzap'
2010-02-20 16:45:54.362813 [NOTICE] switch_loadable_module.c:250
Adding Application 'disable_ec'
2010-02-20 16:45:54.362813 [NOTICE] switch_loadable_module.c:272
Adding API Function 'oz'

+OK module unloaded
+OK module loaded

freeswitch at voyage> oz list

+OK
span: 1 (FXO1)
type: analog
chan_count: 1
dialplan: XML
context: default
dial_regex:
fail_dial_regex:
hold_music:
analog_options 3way
+OK
span: 2 (FXO2)
type: analog
chan_count: 1
dialplan: XML
context: default
dial_regex:
fail_dial_regex:
hold_music:
analog_options 3way
+OK
span: 3 (FXS1)
type: analog
chan_count: 1
dialplan: XML
context: default
dial_regex:
fail_dial_regex:
hold_music:
analog_options none
+OK
span: 4 (FXS2)
type: analog
chan_count: 1
dialplan: XML
context: default
dial_regex:
fail_dial_regex:
hold_music:
analog_options none


Yet, when I lift the handset on port 1 or port 2 I don't get a dial tone :-(

Instead I get this message:

2010-02-20 16:52:26.332813 [INFO] ozmod_zt.c:640 Setting echo cancel
to 64 taps for 2:1
2010-02-20 16:52:26.332813 [ERR] zap_io.c:1599 I/O backend does not
support command 24!

Am I doing something wrong?
-- 
Brian May <brian at microcomaustralia.com.au>


From brian at microcomaustralia.com.au  Sat Feb 20 01:48:36 2010
From: brian at microcomaustralia.com.au (Brian May)
Date: Sat, 20 Feb 2010 20:48:36 +1100
Subject: [Freeswitch-users] openzap TDM400 card
In-Reply-To: <3c5cf5261002192157t25de5798xc10ab2a7755e67d0@mail.gmail.com>
References: <3c5cf5261002192157t25de5798xc10ab2a7755e67d0@mail.gmail.com>
Message-ID: <3c5cf5261002200148k1bb3a1f7mf97fabccfbd1c178@mail.gmail.com>

On 20 February 2010 16:57, Brian May <brian at microcomaustralia.com.au> wrote:
> I think I have the config correct, and not confused FXO/FXS anywhere.

I was confused. The messages after the incoming call gave it away.

2010-02-20 20:42:52.339813 [ERR] ozmod_analog.c:798 Cannot get a
RING_START event on a non-fxo channel, please check your config.
2010-02-20 20:42:52.339813 [WARNING] ozmod_analog.c:799 Why bother
changing state on 3:1 from DOWN to DOWN
2010-02-20 20:42:53.189813 [INFO] ozmod_zt.c:640 Setting echo cancel
to 64 taps for 3:1
2010-02-20 20:42:55.189813 [ERR] ozmod_analog.c:798 Cannot get a
RING_START event on a non-fxo channel, please check your config.
2010-02-20 20:42:56.449813 [INFO] ozmod_zt.c:640 Setting echo cancel
to 64 taps for 3:1
2010-02-20 20:42:58.239813 [ERR] ozmod_analog.c:798 Cannot get a
RING_START event on a non-fxo channel, please check your config.
2010-02-20 20:42:59.499813 [INFO] ozmod_zt.c:640 Setting echo cancel
to 64 taps for 3:1
2010-02-20 20:43:01.299813 [ERR] ozmod_analog.c:798 Cannot get a
RING_START event on a non-fxo channel, please check your config.
2010-02-20 20:43:02.569813 [INFO] ozmod_zt.c:640 Setting echo cancel
to 64 taps for 3:1
2010-02-20 20:43:04.379813 [ERR] ozmod_analog.c:798 Cannot get a
RING_START event on a non-fxo channel, please check your config.
2010-02-20 20:43:05.659813 [INFO] ozmod_zt.c:640 Setting echo cancel
to 64 taps for 3:1

It appears that a port listed as fxoks in zaptel.conf becomes fxs in
openzap.conf, and a port listed as fxsks becomes fxo in openzap.conf -
it would be nice if this were documented somewhere...


Working configuration:


voyage:/opt/freeswitch/conf# cat /etc/zaptel.conf
# Autogenerated by /usr/sbin/genzaptelconf -- do not hand edit
# Zaptel Configuration File
#
# This file is parsed by the Zaptel Configurator, ztcfg
#

# It must be in the module loading order


# Span 1: WCTDM/0 "Wildcard TDM400P REV I Board 1" (MASTER)
fxoks=1
fxoks=2
fxsks=3
fxsks=4

# Global data

loadzone	= au
defaultzone	= au
voyage:/opt/freeswitch/conf# vim openzap.conf
voyage:/opt/freeswitch/conf# cat /etc/zaptel.conf
# Autogenerated by /usr/sbin/genzaptelconf -- do not hand edit
# Zaptel Configuration File
#
# This file is parsed by the Zaptel Configurator, ztcfg
#

# It must be in the module loading order


# Span 1: WCTDM/0 "Wildcard TDM400P REV I Board 1" (MASTER)
fxoks=1
fxoks=2
fxsks=3
fxsks=4

# Global data

loadzone	= au
defaultzone	= au
voyage:/opt/freeswitch/conf# cat openzap.conf
[span zt FXS1]
name => OpenZAP-FXS1
number => 1
fxs-channel => 1

[span zt FXS2]
name => OpenZAP-FXS2
number => 2
fxs-channel => 2

[span zt FXO1]
name => OpenZAP-FXO1
number => 3
fxo-channel => 3

[span zt FXO2]
name => OpenZAP-FXO2
number => 4
fxo-channel => 4


-- 
Brian May <brian at microcomaustralia.com.au>


From Russell.Mosemann at cune.org  Sat Feb 20 03:25:06 2010
From: Russell.Mosemann at cune.org (Russell Mosemann)
Date: Sat, 20 Feb 2010 05:25:06 -0600
Subject: [Freeswitch-users] openzap TDM400 card
In-Reply-To: <3c5cf5261002200148k1bb3a1f7mf97fabccfbd1c178@mail.gmail.com>
References: <3c5cf5261002192157t25de5798xc10ab2a7755e67d0@mail.gmail.com>
	<3c5cf5261002200148k1bb3a1f7mf97fabccfbd1c178@mail.gmail.com>
Message-ID: <C8B78F95DB344ADDA010C52AE0AC8B66@cune.pri>

> It appears that a port listed as fxoks in zaptel.conf becomes fxs in
> openzap.conf, and a port listed as fxsks becomes fxo in openzap.conf -
> it would be nice if this were documented somewhere...

I believe it is, with the specific error you mention, "Cannot get a RING_START event on a non-fxo channel".

http://wiki.freeswitch.org/wiki/OpenZAP#Symptom:_.22Why_bother_changing_state_on_1:1_from_UP_to_UP.22_or_.22Cannot_get_a_RING_START_event_on_a_non-fxo_channel.22

If you don't think it is clear, please add more details to the wiki so that others will be able to more easily solve this problem when it happens.

--
Russell Mosemann



From rob4manhere at gmail.com  Sat Feb 20 05:48:48 2010
From: rob4manhere at gmail.com (Rob Forman)
Date: Sat, 20 Feb 2010 07:48:48 -0600
Subject: [Freeswitch-users] codecs transcoding
In-Reply-To: <DA78008AFA4A414480CDB572EE602D7D@MOBILEE1705>
References: <DA78008AFA4A414480CDB572EE602D7D@MOBILEE1705>
Message-ID: <73A3511B-EC7F-465B-BA51-9A7067303AFF@gmail.com>

No, proxy media is the opposite.  Its for staying in the middle of the  
RTP stream, yet *not* transcoding or processing the packets.

http://wiki.freeswitch.org/wiki/Proxy_Media

If you want to transcode, and both are supported codecs, just bridge  
the two channels.

Rob

On Feb 19, 2010, at 11:08 PM, Madovsky wrote:

> Hello,
>
> is it need proxy_media on true to transcode codecs ?
>
> Thanks
>
> Farnck
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100220/0112f237/attachment.html 

From infos at madovsky.org  Sat Feb 20 08:26:13 2010
From: infos at madovsky.org (Madovsky)
Date: Sat, 20 Feb 2010 11:26:13 -0500
Subject: [Freeswitch-users] codecs transcoding
References: <DA78008AFA4A414480CDB572EE602D7D@MOBILEE1705>
	<73A3511B-EC7F-465B-BA51-9A7067303AFF@gmail.com>
Message-ID: <789ED5715DFF4072A38363B654C5AD19@MOBILEE1705>


  ----- Original Message ----- 
  From: Rob Forman 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Saturday, February 20, 2010 8:48 AM
  Subject: Re: [Freeswitch-users] codecs transcoding


  No, proxy media is the opposite.  Its for staying in the middle of the RTP stream, yet *not* transcoding or processing the packets.


  http://wiki.freeswitch.org/wiki/Proxy_Media


  If you want to transcode, and both are supported codecs, just bridge the two channels.


  Rob


  On Feb 19, 2010, at 11:08 PM, Madovsky wrote:


    Hello,

    is it need proxy_media on true to transcode codecs ?

    Thanks

    Farnck
    _______________________________________________
    FreeSWITCH-users mailing list
    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    http://www.freeswitch.org





------------------------------------------------------------------------------


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  http://www.freeswitch.org

  Thanks Rob, I got it yesterday.

  Franck
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100220/9f5b8eaf/attachment.html 

From mbsip at gazeta.pl  Sat Feb 20 09:48:44 2010
From: mbsip at gazeta.pl (Maciej Bylica)
Date: Sat, 20 Feb 2010 18:48:44 +0100
Subject: [Freeswitch-users] LUA script providing dynamic directory
	information
Message-ID: <28f27f5d1002200948j33d818e1h9fc8c8f99a506656@mail.gmail.com>

Hello,

I am trying to use mod_lua to provide dynamic directory information
(binding in mod_lua.conf.xml)
Here is my script.
#!/usr/local/bin/lua
-- load driver
require "luasql.odbc"
-- create environment object
env = luasql.odbc();
-- connect to data source
conn = env:connect("freeswitch","root");
-- reset our table
if ( conn ~= nil ) then
cur = conn:execute(string.format("SELECT email from VM where
called_num='48112223344'"));

if ( cur ~= nil ) then
row = cur:fetch({}, "a");
if ( row ~= nil ) then

freeswitch.consoleLog("info", " Email fetched from DB is = "..
row.email .."\n");

cur:close(); conn:close(); env:close();

mydialplan = [[
<?xml version="1.0" encoding="UTF-8" standalone="no"?>
<document type="freeswitch/xml">
  <section name="directory">
    <domain name="10.10.10.1">
      <user id="48112223344">
        <params>
          <param name="vm-email-all-messages" value="true"/>
          <param name="vm-attach-file" value="true"/>
          <param name="vm-disk-quota" value="0"/>
          <param name="vm-email-all-messages" value="true"/>
          <param name="vm-mailto" value=".. row.email .."/>
          <param name="vm-keep-local-after-email" value="false"/>
        </params>
      </user>
    </domain>
  </section>
</document>
]]

XML_STRING = mydialplan
end
end
end


I've encountered a problem how to pass row.email gathered from DB
directly to XML (vm-mailto).
As you can see below configuration I have does not work properly.

2010-02-20 20:37:03.267071 [DEBUG] mod_voicemail.c:2358 Deliver VM to
48112223344 at 10.10.10.1
2010-02-20 20:37:03.276533 [INFO] switch_cpp.cpp:1129  Email fetched
from DB is = hereis at MyEmaill.com
2010-02-20 20:37:03.435902 [DEBUG] switch_utils.c:631 Emailed file
[/tmp/mail.1266694623f261] to [.. row.email ..]
2010-02-20 20:37:03.435902 [DEBUG] mod_voicemail.c:2526 Sending
message to .. row.email ..

I tried with:
<param name="vm-mailto" value=.. row.email ../>
<param name="vm-mailto" value="".. row.email ..""/>
Both of them do not produce any "Sending message" output at all.


Any thoughts?
Thanks in advance.
Maciej


From msc at freeswitch.org  Sat Feb 20 10:10:21 2010
From: msc at freeswitch.org (Michael S Collins)
Date: Sat, 20 Feb 2010 10:10:21 -0800
Subject: [Freeswitch-users] openzap TDM400 card
In-Reply-To: <3c5cf5261002192157t25de5798xc10ab2a7755e67d0@mail.gmail.com>
References: <3c5cf5261002192157t25de5798xc10ab2a7755e67d0@mail.gmail.com>
Message-ID: <E94B1537-7943-4328-B090-AEF26C781673@freeswitch.org>

Ports 1 and 2 are FXO which need a phone line. Try port 3 or 4. Also  
pastebin your zaptel.conf file.

-MC

Sent from my iPhone

On Feb 19, 2010, at 9:57 PM, Brian May  
<brian at microcomaustralia.com.au> wrote:

> Hello,
>
> I think I have the config correct, and not confused FXO/FXS anywhere.
>
> voyage:/opt/freeswitch/conf# ztcfg  -vv
>
> Zaptel Version: 1.4.11
> Echo Canceller: MG2
> Configuration
> ======================
>
>
> Channel map:
>
> Channel 01: FXO Kewlstart (Default) (Slaves: 01)
> Channel 02: FXO Kewlstart (Default) (Slaves: 02)
> Channel 03: FXS Kewlstart (Default) (Slaves: 03)
> Channel 04: FXS Kewlstart (Default) (Slaves: 04)
>
> 4 channels to configure.
>
> voyage:/opt/freeswitch/conf# cat openzap.conf
> [span zt FXO1]
> name => OpenZAP-FXO1
> number => 1
> fxo-channel => 1
>
> [span zt FXO2]
> name => OpenZAP-FXO2
> number => 2
> fxo-channel => 2
>
> [span zt FXS1]
> name => OpenZAP-FXS1
> number => 3
> fxs-channel => 3
>
> [span zt FXS2]
> name => OpenZAP-FXS2
> number => 4
> fxs-channel => 4
>
> voyage:/opt/freeswitch/conf# cat autoload_configs/openzap.conf.xml
> <configuration name="openzap.conf" description="OpenZAP  
> Configuration">
>  <settings>
>    <param name="debug" value="0"/>
>    <!--<param name="hold-music" value="$${moh_uri}"/>-->
>    <!--<param name="enable-analog-option" value="call-swap"/>-->
>    <!--<param name="enable-analog-option" value="3-way"/>-->
>  </settings>
>  <!-- one entry here per openzap span -->
>  <analog_spans>
>    <span id="1">
>      <param name="tonegroup" value="au"/>
>      <param name="digit-timeout" value="2000"/>
>      <param name="max-digits" value="11"/>
>      <param name="dialplan" value="XML"/>
>      <param name="context" value="default"/>
>      <param name="enable-analog-option" value="3-way"/>
> <!--      <param name="moh" value="$${base_dir}/sounds/ 
> Dardedisco.wav"/> -->
>    </span>
>    <span id="2">
>      <param name="tonegroup" value="au"/>
>      <param name="digit-timeout" value="2000"/>
>      <param name="max-digits" value="11"/>
>      <param name="dialplan" value="XML"/>
>      <param name="context" value="default"/>
>      <param name="enable-analog-option" value="3-way"/>
> <!--      <param name="moh" value="$${base_dir}/sounds/ 
> Dardedisco.wav"/> -->
>    </span>
>    <span id="3">
>      <param name="tonegroup" value="au"/>
>      <param name="digit-timeout" value="2000"/>
>      <param name="max-digits" value="11"/>
>      <param name="dialplan" value="XML"/>
>      <param name="context" value="default"/>
>    </span>
>    <span id="4">
>      <param name="tonegroup" value="au"/>
>      <param name="digit-timeout" value="2000"/>
>      <param name="max-digits" value="11"/>
>      <param name="dialplan" value="XML"/>
>      <param name="context" value="default"/>
>    </span>
>  </analog_spans>
> </configuration>
>
>
> freeswitch at voyage> reload mod_openzap
> 2010-02-20 16:45:53.312813 [NOTICE] switch_loadable_module.c:464
> Deleting Endpoint 'openzap'
> 2010-02-20 16:45:53.312813 [NOTICE] switch_loadable_module.c:545
> Deleting Application 'disable_ec'
> 2010-02-20 16:45:53.312813 [NOTICE] switch_loadable_module.c:572
> Deleting API Function 'oz'
> 2010-02-20 16:45:53.312813 [CONSOLE] switch_loadable_module.c:1277
> Stopping: mod_openzap
> 2010-02-20 16:45:53.312813 [INFO] zap_io.c:269 Closing channel zt: 
> 1:1 fd:36
> 2010-02-20 16:45:53.312813 [INFO] zap_io.c:269 Closing channel zt: 
> 2:1 fd:40
> 2010-02-20 16:45:53.312813 [INFO] zap_io.c:269 Closing channel zt: 
> 3:1 fd:41
> 2010-02-20 16:45:53.312813 [INFO] zap_io.c:269 Closing channel zt: 
> 4:1 fd:42
> 2010-02-20 16:45:53.612813 [ERR] ozmod_analog.c:951 Failure Polling
> event! [no matching descriptor]
> 2010-02-20 16:45:54.162813 [ERR] ozmod_analog.c:951 Failure Polling
> event! [no matching descriptor]
> 2010-02-20 16:45:54.162813 [ERR] ozmod_analog.c:951 Failure Polling
> event! [no matching descriptor]
> 2010-02-20 16:45:54.162813 [ERR] ozmod_analog.c:951 Failure Polling
> event! [no matching descriptor]
> 2010-02-20 16:45:54.322813 [INFO] zap_io.c:2694 Unloading
> /opt/freeswitch/mod/ozmod_analog.so
> 2010-02-20 16:45:54.322813 [INFO] zap_io.c:2679 Unloading IO zt
> 2010-02-20 16:45:54.322813 [INFO] zap_io.c:2694 Unloading
> /opt/freeswitch/mod/ozmod_zt.so
> 2010-02-20 16:45:54.322813 [CONSOLE] switch_loadable_module.c:1297
> mod_openzap unloaded.
> 2010-02-20 16:45:54.322813 [NOTICE] zap_io.c:2778 Modules  
> configured: 1
> 2010-02-20 16:45:54.322813 [NOTICE] ozmod_zt.c:1161 Using Zaptel  
> control device
> 2010-02-20 16:45:54.322813 [INFO] zap_io.c:2579 Loading IO from
> /opt/freeswitch/mod/ozmod_zt.so [zt]
> 2010-02-20 16:45:54.322813 [INFO] ozmod_zt.c:556 Setting rxgain val  
> to 0.000000
> 2010-02-20 16:45:54.322813 [INFO] ozmod_zt.c:565 Setting txgain val  
> to 0.000000
> 2010-02-20 16:45:54.322813 [INFO] zap_io.c:2379 auto-loaded 'zt'
> 2010-02-20 16:45:54.322813 [WARNING] ozmod_zt.c:333 this ioctl fails
> on older zaptel but is harmless if you used ztcfg
> [device /dev/zap/channel chan 1 fd 27 (Invalid argument)]
> 2010-02-20 16:45:54.322813 [INFO] ozmod_zt.c:385 configuring device
> /dev/zap/channel channel 1 as OpenZAP device 1:1 fd:36
> 2010-02-20 16:45:54.322813 [WARNING] ozmod_zt.c:333 this ioctl fails
> on older zaptel but is harmless if you used ztcfg
> [device /dev/zap/channel chan 2 fd 27 (Invalid argument)]
> 2010-02-20 16:45:54.322813 [INFO] ozmod_zt.c:385 configuring device
> /dev/zap/channel channel 2 as OpenZAP device 2:1 fd:40
> 2010-02-20 16:45:54.322813 [WARNING] ozmod_zt.c:333 this ioctl fails
> on older zaptel but is harmless if you used ztcfg
> [device /dev/zap/channel chan 3 fd 27 (Invalid argument)]
> 2010-02-20 16:45:54.322813 [INFO] ozmod_zt.c:385 configuring device
> /dev/zap/channel channel 3 as OpenZAP device 3:1 fd:41
> 2010-02-20 16:45:54.322813 [WARNING] ozmod_zt.c:333 this ioctl fails
> on older zaptel but is harmless if you used ztcfg
> [device /dev/zap/channel chan 4 fd 27 (Invalid argument)]
> 2010-02-20 16:45:54.322813 [INFO] ozmod_zt.c:385 configuring device
> /dev/zap/channel channel 4 as OpenZAP device 4:1 fd:42
> 2010-02-20 16:45:54.322813 [INFO] zap_io.c:2502 Configured 4 channel 
> (s)
> 2010-02-20 16:45:54.322813 [INFO] zap_io.c:2596 Loading SIG from
> /opt/freeswitch/mod/ozmod_analog.so
> 2010-02-20 16:45:54.322813 [INFO] zap_io.c:2712 auto-loaded 'analog'
> 2010-02-20 16:45:54.358813 [CONSOLE] switch_loadable_module.c:900
> Successfully Loaded [mod_openzap]
> 2010-02-20 16:45:54.362813 [NOTICE] switch_loadable_module.c:144
> Adding Endpoint 'openzap'
> 2010-02-20 16:45:54.362813 [NOTICE] switch_loadable_module.c:250
> Adding Application 'disable_ec'
> 2010-02-20 16:45:54.362813 [NOTICE] switch_loadable_module.c:272
> Adding API Function 'oz'
>
> +OK module unloaded
> +OK module loaded
>
> freeswitch at voyage> oz list
>
> +OK
> span: 1 (FXO1)
> type: analog
> chan_count: 1
> dialplan: XML
> context: default
> dial_regex:
> fail_dial_regex:
> hold_music:
> analog_options 3way
> +OK
> span: 2 (FXO2)
> type: analog
> chan_count: 1
> dialplan: XML
> context: default
> dial_regex:
> fail_dial_regex:
> hold_music:
> analog_options 3way
> +OK
> span: 3 (FXS1)
> type: analog
> chan_count: 1
> dialplan: XML
> context: default
> dial_regex:
> fail_dial_regex:
> hold_music:
> analog_options none
> +OK
> span: 4 (FXS2)
> type: analog
> chan_count: 1
> dialplan: XML
> context: default
> dial_regex:
> fail_dial_regex:
> hold_music:
> analog_options none
>
>
> Yet, when I lift the handset on port 1 or port 2 I don't get a dial  
> tone :-(
>
> Instead I get this message:
>
> 2010-02-20 16:52:26.332813 [INFO] ozmod_zt.c:640 Setting echo cancel
> to 64 taps for 2:1
> 2010-02-20 16:52:26.332813 [ERR] zap_io.c:1599 I/O backend does not
> support command 24!
>
> Am I doing something wrong?
> -- 
> Brian May <brian at microcomaustralia.com.au>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org


From infos at madovsky.org  Sat Feb 20 10:36:01 2010
From: infos at madovsky.org (Madovsky)
Date: Sat, 20 Feb 2010 13:36:01 -0500
Subject: [Freeswitch-users] outbound calls
Message-ID: <C9E83EA051BC49D8AB9425D5D1225C27@MOBILEE1705>

Hello,

I'm able to transcode a cal between 2 local legs,
but when a local user call an oubound call,
the call hangs up saying "not acceptable here",
so it doesn't transcode.

Any idea ?

Thanks

Franck
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100220/73f2ebd5/attachment.html 

From ederwander at gmail.com  Sat Feb 20 10:51:40 2010
From: ederwander at gmail.com (Eder Souza)
Date: Sat, 20 Feb 2010 16:51:40 -0200
Subject: [Freeswitch-users] openzap TDM400 card
In-Reply-To: <E94B1537-7943-4328-B090-AEF26C781673@freeswitch.org>
References: <3c5cf5261002192157t25de5798xc10ab2a7755e67d0@mail.gmail.com>
	<E94B1537-7943-4328-B090-AEF26C781673@freeswitch.org>
Message-ID: <622bedea1002201051y4de06dcdh9c1c1681e05c0d3b@mail.gmail.com>

Try set  loadzone  and defaultzone for your country in zaptel.conf

then do this:

ztcfg -vv

and

zttol -vvv  (see if status of your card is OK)


Att

Eng Eder de Souza




On Sat, Feb 20, 2010 at 4:10 PM, Michael S Collins <msc at freeswitch.org>wrote:

> Ports 1 and 2 are FXO which need a phone line. Try port 3 or 4. Also
> pastebin your zaptel.conf file.
>
> -MC
>
> Sent from my iPhone
>
> On Feb 19, 2010, at 9:57 PM, Brian May
> <brian at microcomaustralia.com.au> wrote:
>
> > Hello,
> >
> > I think I have the config correct, and not confused FXO/FXS anywhere.
> >
> > voyage:/opt/freeswitch/conf# ztcfg  -vv
> >
> > Zaptel Version: 1.4.11
> > Echo Canceller: MG2
> > Configuration
> > ======================
> >
> >
> > Channel map:
> >
> > Channel 01: FXO Kewlstart (Default) (Slaves: 01)
> > Channel 02: FXO Kewlstart (Default) (Slaves: 02)
> > Channel 03: FXS Kewlstart (Default) (Slaves: 03)
> > Channel 04: FXS Kewlstart (Default) (Slaves: 04)
> >
> > 4 channels to configure.
> >
> > voyage:/opt/freeswitch/conf# cat openzap.conf
> > [span zt FXO1]
> > name => OpenZAP-FXO1
> > number => 1
> > fxo-channel => 1
> >
> > [span zt FXO2]
> > name => OpenZAP-FXO2
> > number => 2
> > fxo-channel => 2
> >
> > [span zt FXS1]
> > name => OpenZAP-FXS1
> > number => 3
> > fxs-channel => 3
> >
> > [span zt FXS2]
> > name => OpenZAP-FXS2
> > number => 4
> > fxs-channel => 4
> >
> > voyage:/opt/freeswitch/conf# cat autoload_configs/openzap.conf.xml
> > <configuration name="openzap.conf" description="OpenZAP
> > Configuration">
> >  <settings>
> >    <param name="debug" value="0"/>
> >    <!--<param name="hold-music" value="$${moh_uri}"/>-->
> >    <!--<param name="enable-analog-option" value="call-swap"/>-->
> >    <!--<param name="enable-analog-option" value="3-way"/>-->
> >  </settings>
> >  <!-- one entry here per openzap span -->
> >  <analog_spans>
> >    <span id="1">
> >      <param name="tonegroup" value="au"/>
> >      <param name="digit-timeout" value="2000"/>
> >      <param name="max-digits" value="11"/>
> >      <param name="dialplan" value="XML"/>
> >      <param name="context" value="default"/>
> >      <param name="enable-analog-option" value="3-way"/>
> > <!--      <param name="moh" value="$${base_dir}/sounds/
> > Dardedisco.wav"/> -->
> >    </span>
> >    <span id="2">
> >      <param name="tonegroup" value="au"/>
> >      <param name="digit-timeout" value="2000"/>
> >      <param name="max-digits" value="11"/>
> >      <param name="dialplan" value="XML"/>
> >      <param name="context" value="default"/>
> >      <param name="enable-analog-option" value="3-way"/>
> > <!--      <param name="moh" value="$${base_dir}/sounds/
> > Dardedisco.wav"/> -->
> >    </span>
> >    <span id="3">
> >      <param name="tonegroup" value="au"/>
> >      <param name="digit-timeout" value="2000"/>
> >      <param name="max-digits" value="11"/>
> >      <param name="dialplan" value="XML"/>
> >      <param name="context" value="default"/>
> >    </span>
> >    <span id="4">
> >      <param name="tonegroup" value="au"/>
> >      <param name="digit-timeout" value="2000"/>
> >      <param name="max-digits" value="11"/>
> >      <param name="dialplan" value="XML"/>
> >      <param name="context" value="default"/>
> >    </span>
> >  </analog_spans>
> > </configuration>
> >
> >
> > freeswitch at voyage> reload mod_openzap
> > 2010-02-20 16:45:53.312813 [NOTICE] switch_loadable_module.c:464
> > Deleting Endpoint 'openzap'
> > 2010-02-20 16:45:53.312813 [NOTICE] switch_loadable_module.c:545
> > Deleting Application 'disable_ec'
> > 2010-02-20 16:45:53.312813 [NOTICE] switch_loadable_module.c:572
> > Deleting API Function 'oz'
> > 2010-02-20 16:45:53.312813 [CONSOLE] switch_loadable_module.c:1277
> > Stopping: mod_openzap
> > 2010-02-20 16:45:53.312813 [INFO] zap_io.c:269 Closing channel zt:
> > 1:1 fd:36
> > 2010-02-20 16:45:53.312813 [INFO] zap_io.c:269 Closing channel zt:
> > 2:1 fd:40
> > 2010-02-20 16:45:53.312813 [INFO] zap_io.c:269 Closing channel zt:
> > 3:1 fd:41
> > 2010-02-20 16:45:53.312813 [INFO] zap_io.c:269 Closing channel zt:
> > 4:1 fd:42
> > 2010-02-20 16:45:53.612813 [ERR] ozmod_analog.c:951 Failure Polling
> > event! [no matching descriptor]
> > 2010-02-20 16:45:54.162813 [ERR] ozmod_analog.c:951 Failure Polling
> > event! [no matching descriptor]
> > 2010-02-20 16:45:54.162813 [ERR] ozmod_analog.c:951 Failure Polling
> > event! [no matching descriptor]
> > 2010-02-20 16:45:54.162813 [ERR] ozmod_analog.c:951 Failure Polling
> > event! [no matching descriptor]
> > 2010-02-20 16:45:54.322813 [INFO] zap_io.c:2694 Unloading
> > /opt/freeswitch/mod/ozmod_analog.so
> > 2010-02-20 16:45:54.322813 [INFO] zap_io.c:2679 Unloading IO zt
> > 2010-02-20 16:45:54.322813 [INFO] zap_io.c:2694 Unloading
> > /opt/freeswitch/mod/ozmod_zt.so
> > 2010-02-20 16:45:54.322813 [CONSOLE] switch_loadable_module.c:1297
> > mod_openzap unloaded.
> > 2010-02-20 16:45:54.322813 [NOTICE] zap_io.c:2778 Modules
> > configured: 1
> > 2010-02-20 16:45:54.322813 [NOTICE] ozmod_zt.c:1161 Using Zaptel
> > control device
> > 2010-02-20 16:45:54.322813 [INFO] zap_io.c:2579 Loading IO from
> > /opt/freeswitch/mod/ozmod_zt.so [zt]
> > 2010-02-20 16:45:54.322813 [INFO] ozmod_zt.c:556 Setting rxgain val
> > to 0.000000
> > 2010-02-20 16:45:54.322813 [INFO] ozmod_zt.c:565 Setting txgain val
> > to 0.000000
> > 2010-02-20 16:45:54.322813 [INFO] zap_io.c:2379 auto-loaded 'zt'
> > 2010-02-20 16:45:54.322813 [WARNING] ozmod_zt.c:333 this ioctl fails
> > on older zaptel but is harmless if you used ztcfg
> > [device /dev/zap/channel chan 1 fd 27 (Invalid argument)]
> > 2010-02-20 16:45:54.322813 [INFO] ozmod_zt.c:385 configuring device
> > /dev/zap/channel channel 1 as OpenZAP device 1:1 fd:36
> > 2010-02-20 16:45:54.322813 [WARNING] ozmod_zt.c:333 this ioctl fails
> > on older zaptel but is harmless if you used ztcfg
> > [device /dev/zap/channel chan 2 fd 27 (Invalid argument)]
> > 2010-02-20 16:45:54.322813 [INFO] ozmod_zt.c:385 configuring device
> > /dev/zap/channel channel 2 as OpenZAP device 2:1 fd:40
> > 2010-02-20 16:45:54.322813 [WARNING] ozmod_zt.c:333 this ioctl fails
> > on older zaptel but is harmless if you used ztcfg
> > [device /dev/zap/channel chan 3 fd 27 (Invalid argument)]
> > 2010-02-20 16:45:54.322813 [INFO] ozmod_zt.c:385 configuring device
> > /dev/zap/channel channel 3 as OpenZAP device 3:1 fd:41
> > 2010-02-20 16:45:54.322813 [WARNING] ozmod_zt.c:333 this ioctl fails
> > on older zaptel but is harmless if you used ztcfg
> > [device /dev/zap/channel chan 4 fd 27 (Invalid argument)]
> > 2010-02-20 16:45:54.322813 [INFO] ozmod_zt.c:385 configuring device
> > /dev/zap/channel channel 4 as OpenZAP device 4:1 fd:42
> > 2010-02-20 16:45:54.322813 [INFO] zap_io.c:2502 Configured 4 channel
> > (s)
> > 2010-02-20 16:45:54.322813 [INFO] zap_io.c:2596 Loading SIG from
> > /opt/freeswitch/mod/ozmod_analog.so
> > 2010-02-20 16:45:54.322813 [INFO] zap_io.c:2712 auto-loaded 'analog'
> > 2010-02-20 16:45:54.358813 [CONSOLE] switch_loadable_module.c:900
> > Successfully Loaded [mod_openzap]
> > 2010-02-20 16:45:54.362813 [NOTICE] switch_loadable_module.c:144
> > Adding Endpoint 'openzap'
> > 2010-02-20 16:45:54.362813 [NOTICE] switch_loadable_module.c:250
> > Adding Application 'disable_ec'
> > 2010-02-20 16:45:54.362813 [NOTICE] switch_loadable_module.c:272
> > Adding API Function 'oz'
> >
> > +OK module unloaded
> > +OK module loaded
> >
> > freeswitch at voyage> oz list
> >
> > +OK
> > span: 1 (FXO1)
> > type: analog
> > chan_count: 1
> > dialplan: XML
> > context: default
> > dial_regex:
> > fail_dial_regex:
> > hold_music:
> > analog_options 3way
> > +OK
> > span: 2 (FXO2)
> > type: analog
> > chan_count: 1
> > dialplan: XML
> > context: default
> > dial_regex:
> > fail_dial_regex:
> > hold_music:
> > analog_options 3way
> > +OK
> > span: 3 (FXS1)
> > type: analog
> > chan_count: 1
> > dialplan: XML
> > context: default
> > dial_regex:
> > fail_dial_regex:
> > hold_music:
> > analog_options none
> > +OK
> > span: 4 (FXS2)
> > type: analog
> > chan_count: 1
> > dialplan: XML
> > context: default
> > dial_regex:
> > fail_dial_regex:
> > hold_music:
> > analog_options none
> >
> >
> > Yet, when I lift the handset on port 1 or port 2 I don't get a dial
> > tone :-(
> >
> > Instead I get this message:
> >
> > 2010-02-20 16:52:26.332813 [INFO] ozmod_zt.c:640 Setting echo cancel
> > to 64 taps for 2:1
> > 2010-02-20 16:52:26.332813 [ERR] zap_io.c:1599 I/O backend does not
> > support command 24!
> >
> > Am I doing something wrong?
> > --
> > Brian May <brian at microcomaustralia.com.au>
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> > users
> > http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100220/8757e35f/attachment-0001.html 

From jmesquita at freeswitch.org  Sat Feb 20 13:27:00 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Sat, 20 Feb 2010 19:27:00 -0200
Subject: [Freeswitch-users] How to exchange media when I developed new
	endpoint module?
In-Reply-To: <93b0f8ce1002191722h5782b787qe56a06dd0cb94e76@mail.gmail.com>
References: <93b0f8ce1002091914q581f12fvd1de01e2181f0f3a@mail.gmail.com>
	<ebbc2d551002091932u52dbf1b6jfecf2dfdb43b8a7c@mail.gmail.com>
	<4021C303-1AC3-48F2-A17B-63C02C1C54D4@freeswitch.org>
	<93b0f8ce1002191722h5782b787qe56a06dd0cb94e76@mail.gmail.com>
Message-ID: <ebbc2d551002201327o7a7bad26g48cac48f610aed55@mail.gmail.com>

I developed the current implementation of mod_khomp. I wouldn't take it as
an example for anything since there has been no activity there for the past
4 months. If you care to share a snippet of your code, maybe we can help
better.


JM


On Fri, Feb 19, 2010 at 11:22 PM, ??? <spiritonly at gmail.com> wrote:

> Do you know mod_khomp? You can found it in FS wiki. I am developing an
> endpoint module like it.
> So you can give me some advice to bridge two session?
>
>
> On Wed, Feb 10, 2010 at 11:44 AM, Brian West <brian at freeswitch.org> wrote:
>
>> But the bigger question is what protocol are you doing that you have to
>> create your own endpoint module?
>>
>> /b
>>
>> On Feb 9, 2010, at 9:32 PM, Jo?o Mesquita wrote:
>>
>> > You should look at read_frame and write_frame implementations of other
>> endpoint modules.
>> >
>> > This should pretty much tell you how things work...
>> >
>> > Jo?o Mesquita
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100220/86320003/attachment.html 

From dave at 3c.co.uk  Sat Feb 20 13:39:56 2010
From: dave at 3c.co.uk (David Knell)
Date: Sat, 20 Feb 2010 14:39:56 -0700
Subject: [Freeswitch-users] LUA script providing dynamic
	directoryinformation
References: <28f27f5d1002200948j33d818e1h9fc8c8f99a506656@mail.gmail.com>
Message-ID: <A80748C5894146E594F157A9811E1011@DELL9>

Hi Maciej,

Your problem is that Lua won't automatically substitute variables inside a 
string - which is why you're just seeing ..row.email.. passed directly 
through.

If you replace that line in the source with something like
<param name="vm-mailto" value="]] .. row.email .. [["/>

the it's more likely to work.

Cheers --

Dave

----- Original Message ----- 
From: "Maciej Bylica" <mbsip at gazeta.pl>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Saturday, February 20, 2010 10:48 AM
Subject: [Freeswitch-users] LUA script providing dynamic 
directoryinformation


> Hello,
>
> I am trying to use mod_lua to provide dynamic directory information
> (binding in mod_lua.conf.xml)
> Here is my script.
> #!/usr/local/bin/lua
> -- load driver
> require "luasql.odbc"
> -- create environment object
> env = luasql.odbc();
> -- connect to data source
> conn = env:connect("freeswitch","root");
> -- reset our table
> if ( conn ~= nil ) then
> cur = conn:execute(string.format("SELECT email from VM where
> called_num='48112223344'"));
>
> if ( cur ~= nil ) then
> row = cur:fetch({}, "a");
> if ( row ~= nil ) then
>
> freeswitch.consoleLog("info", " Email fetched from DB is = "..
> row.email .."\n");
>
> cur:close(); conn:close(); env:close();
>
> mydialplan = [[
> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
> <document type="freeswitch/xml">
>  <section name="directory">
>    <domain name="10.10.10.1">
>      <user id="48112223344">
>        <params>
>          <param name="vm-email-all-messages" value="true"/>
>          <param name="vm-attach-file" value="true"/>
>          <param name="vm-disk-quota" value="0"/>
>          <param name="vm-email-all-messages" value="true"/>
>          <param name="vm-mailto" value=".. row.email .."/>
>          <param name="vm-keep-local-after-email" value="false"/>
>        </params>
>      </user>
>    </domain>
>  </section>
> </document>
> ]]
>
> XML_STRING = mydialplan
> end
> end
> end
>
>
> I've encountered a problem how to pass row.email gathered from DB
> directly to XML (vm-mailto).
> As you can see below configuration I have does not work properly.
>
> 2010-02-20 20:37:03.267071 [DEBUG] mod_voicemail.c:2358 Deliver VM to
> 48112223344 at 10.10.10.1
> 2010-02-20 20:37:03.276533 [INFO] switch_cpp.cpp:1129  Email fetched
> from DB is = hereis at MyEmaill.com
> 2010-02-20 20:37:03.435902 [DEBUG] switch_utils.c:631 Emailed file
> [/tmp/mail.1266694623f261] to [.. row.email ..]
> 2010-02-20 20:37:03.435902 [DEBUG] mod_voicemail.c:2526 Sending
> message to .. row.email ..
>
> I tried with:
> <param name="vm-mailto" value=.. row.email ../>
> <param name="vm-mailto" value="".. row.email ..""/>
> Both of them do not produce any "Sending message" output at all.
>
>
> Any thoughts?
> Thanks in advance.
> Maciej
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> 



From brian at microcomaustralia.com.au  Sat Feb 20 14:08:05 2010
From: brian at microcomaustralia.com.au (Brian May)
Date: Sun, 21 Feb 2010 09:08:05 +1100
Subject: [Freeswitch-users] openzap TDM400 card
In-Reply-To: <C8B78F95DB344ADDA010C52AE0AC8B66@cune.pri>
References: <3c5cf5261002192157t25de5798xc10ab2a7755e67d0@mail.gmail.com>
	<3c5cf5261002200148k1bb3a1f7mf97fabccfbd1c178@mail.gmail.com>
	<C8B78F95DB344ADDA010C52AE0AC8B66@cune.pri>
Message-ID: <3c5cf5261002201408v63c150d3v419493ab5e4e2089@mail.gmail.com>

On 20 February 2010 22:25, Russell Mosemann <Russell.Mosemann at cune.org> wrote:
> I believe it is, with the specific error you mention, "Cannot get a RING_START event on a non-fxo channel".

That error was clear. I was able to solve the issue when I saw that,
without having to look up the documentation in fact.

Only I spent ages before I received this error, trying to debug the
other ports with the phone lines connected. These ports didn't give
any errors, except for stuff I can ignore. Based on the other
responses I have got here, it appears other people were confused too.

Lets see if I have got this correct:

Ports 1 & 2 are FXS, but use FXO signalling. So, yes I do plug a phone
into these.
Ports 3 & 4 are FXO, but use FXS signalling. So I plug the phone line
into these.


I will update the wiki, but there is still one part I don't
understand, see the example at the end here:

<http://wiki.freeswitch.org/wiki/Openzap.conf.xml_Examples>

I assume ports 1&2 (with 3-way and moh set) are the FXS ports that use
the FXO signalling? i.e. the same numbering as what I have?

Do I really need to set moh for every line in this file?


Looking at the openzap part of the wiki, the only thing I see related
is "I didn't understand why we have to swap the channel numbers for
FXS and FXO." There is also a link to
http://unixtoys.ca/wordpress/2008/09/freeswitch-and-tdm-hardware-pots-fxofxs/
which looks like it would have helped but unfortunately the link
appears to be broken. Am going to try and update these issues now.
-- 
Brian May <brian at microcomaustralia.com.au>


From mbsip at gazeta.pl  Sat Feb 20 14:39:25 2010
From: mbsip at gazeta.pl (Maciej Bylica)
Date: Sat, 20 Feb 2010 23:39:25 +0100
Subject: [Freeswitch-users] LUA script providing dynamic
	directoryinformation
In-Reply-To: <A80748C5894146E594F157A9811E1011@DELL9>
References: <28f27f5d1002200948j33d818e1h9fc8c8f99a506656@mail.gmail.com>
	<A80748C5894146E594F157A9811E1011@DELL9>
Message-ID: <28f27f5d1002201439j30d4b88cn14ad07ae83067ed5@mail.gmail.com>

Hi Dave,

Yesssss it works :P
I have been struggling with this for hours, so I appreciate Your help.

Thx,
Maciej.

> Hi Maciej,
>
> Your problem is that Lua won't automatically substitute variables inside a
> string - which is why you're just seeing ..row.email.. passed directly
> through.
>
> If you replace that line in the source with something like
> <param name="vm-mailto" value="]] .. row.email .. [["/>
>
> the it's more likely to work.
>
> Cheers --
>
> Dave
>
> ----- Original Message -----
> From: "Maciej Bylica" <mbsip at gazeta.pl>
> To: <freeswitch-users at lists.freeswitch.org>
> Sent: Saturday, February 20, 2010 10:48 AM
> Subject: [Freeswitch-users] LUA script providing dynamic
> directoryinformation
>
>
>> Hello,
>>
>> I am trying to use mod_lua to provide dynamic directory information
>> (binding in mod_lua.conf.xml)
>> Here is my script.
>> #!/usr/local/bin/lua
>> -- load driver
>> require "luasql.odbc"
>> -- create environment object
>> env = luasql.odbc();
>> -- connect to data source
>> conn = env:connect("freeswitch","root");
>> -- reset our table
>> if ( conn ~= nil ) then
>> cur = conn:execute(string.format("SELECT email from VM where
>> called_num='48112223344'"));
>>
>> if ( cur ~= nil ) then
>> row = cur:fetch({}, "a");
>> if ( row ~= nil ) then
>>
>> freeswitch.consoleLog("info", " Email fetched from DB is = "..
>> row.email .."\n");
>>
>> cur:close(); conn:close(); env:close();
>>
>> mydialplan = [[
>> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
>> <document type="freeswitch/xml">
>> ?<section name="directory">
>> ? ?<domain name="10.10.10.1">
>> ? ? ?<user id="48112223344">
>> ? ? ? ?<params>
>> ? ? ? ? ?<param name="vm-email-all-messages" value="true"/>
>> ? ? ? ? ?<param name="vm-attach-file" value="true"/>
>> ? ? ? ? ?<param name="vm-disk-quota" value="0"/>
>> ? ? ? ? ?<param name="vm-email-all-messages" value="true"/>
>> ? ? ? ? ?<param name="vm-mailto" value=".. row.email .."/>
>> ? ? ? ? ?<param name="vm-keep-local-after-email" value="false"/>
>> ? ? ? ?</params>
>> ? ? ?</user>
>> ? ?</domain>
>> ?</section>
>> </document>
>> ]]
>>
>> XML_STRING = mydialplan
>> end
>> end
>> end
>>
>>
>> I've encountered a problem how to pass row.email gathered from DB
>> directly to XML (vm-mailto).
>> As you can see below configuration I have does not work properly.
>>
>> 2010-02-20 20:37:03.267071 [DEBUG] mod_voicemail.c:2358 Deliver VM to
>> 48112223344 at 10.10.10.1
>> 2010-02-20 20:37:03.276533 [INFO] switch_cpp.cpp:1129 ?Email fetched
>> from DB is = hereis at MyEmaill.com
>> 2010-02-20 20:37:03.435902 [DEBUG] switch_utils.c:631 Emailed file
>> [/tmp/mail.1266694623f261] to [.. row.email ..]
>> 2010-02-20 20:37:03.435902 [DEBUG] mod_voicemail.c:2526 Sending
>> message to .. row.email ..
>>
>> I tried with:
>> <param name="vm-mailto" value=.. row.email ../>
>> <param name="vm-mailto" value="".. row.email ..""/>
>> Both of them do not produce any "Sending message" output at all.
>>
>>
>> Any thoughts?
>> Thanks in advance.
>> Maciej
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From brian at microcomaustralia.com.au  Sat Feb 20 16:32:46 2010
From: brian at microcomaustralia.com.au (Brian May)
Date: Sun, 21 Feb 2010 11:32:46 +1100
Subject: [Freeswitch-users] building for Lenny
In-Reply-To: <3c5cf5261002191438o47aabde9tb588fe3ef23f27bc@mail.gmail.com>
References: <3c5cf5261002180046vb0015e7t787599c9903aedc@mail.gmail.com>
	<33c87fa31002180056g51ef8496qa9a35746d0fe75ec@mail.gmail.com>
	<3c5cf5261002180110he0ae815uc713f77f289ac625@mail.gmail.com>
	<a9f4977a70b3fcf1b30413ea562ca962.squirrel@www.webcontracts.co.uk>
	<20100218161741.GC4236@base.carmickle.com>
	<3c5cf5261002181542y43adb7a7ofe2a15c38baf0327@mail.gmail.com>
	<4C13BCBF-1DA0-47AE-A749-738AD6D8FE48@jerris.com>
	<3c5cf5261002191438o47aabde9tb588fe3ef23f27bc@mail.gmail.com>
Message-ID: <3c5cf5261002201632j33256190ve36b667db928eedb@mail.gmail.com>

Just noticed the latest efforts for getting this packaged and into
Debian proper are described here:

http://bugs.debian.org/cgi-bin/bugreport.cgi?bug=389591
-- 
Brian May <brian at microcomaustralia.com.au>


From Prometheus001 at gmx.net  Sat Feb 20 16:44:22 2010
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Sun, 21 Feb 2010 01:44:22 +0100
Subject: [Freeswitch-users] more than 1 profile for sofia/user ?
In-Reply-To: <ebbc2d551002181929m72a17b7bj1fa1cff45b02607b@mail.gmail.com>
References: <4B7D9494.8050208@gmx.net>	<191c3a031002181140y44135106yb34e6a6f73a04773@mail.gmail.com>	<4B7DADC8.1060405@gmx.net>	<191c3a031002181353r2dbf15c6h206509f9b1399148@mail.gmail.com>	<4B7DC202.7090409@gmx.net>
	<ebbc2d551002181929m72a17b7bj1fa1cff45b02607b@mail.gmail.com>
Message-ID: <4B8081E6.8060806@gmx.net>

Thanks Jo?o, this solved my problem.

Just for the records how it works:

    * created a new profile "internalnat" as a copy of "internal"
    * modified ports 5060 and 5061 to 5065 and 5066
    * added parameters external-rtp-ip and external-sip-ip with external IPs
    * modified the dialstring as proposed below
    * Phone registers, phone can dial and can be called.

Freeswitch rocks!

Best regards
Peter

Jo?o Mesquita schrieb:
> I would:
>
> {presence_id=${dialed_user}@${dialed_domain},transfer_fallback_extension=${dialed_user}}${sofia_contact(${dialed_user}@${dialed_domain}),sofia/other_profile/${dialed_user}}
>
> You could toy with that a bit. The dialstring is really just an
> origination string that is generated by the user/<extension> ...
>
> Hope that clears it up a bit.
>
> Regards,
> Jo?o Mesquita
>
>
>
> On Thu, Feb 18, 2010 at 7:41 PM, Peter P GMX <Prometheus001 at gmx.net
> <mailto:Prometheus001 at gmx.net>> wrote:
>
>     Hello Anthony,
>
>     >add on a , then another dial string to reflect the other profile too
>     I really tried to understand this, but
>     can you give me an example?
>
>     Best regards
>     Peter
>
>     Anthony Minessale schrieb:
>     > add on a , then another dial string to reflect the other profile too
>     >
>     > On Thu, Feb 18, 2010 at 3:14 PM, Peter P GMX
>     <Prometheus001 at gmx.net <mailto:Prometheus001 at gmx.net>
>     > <mailto:Prometheus001 at gmx.net <mailto:Prometheus001 at gmx.net>>>
>     wrote:
>     >
>     >     Any idea how to do this?
>     >
>     >     currently I have
>     >    
>     {presence_id=${dialed_user}@${dialed_domain},transfer_fallback_extension=${dialed_user}}${sofia_contact(${dialed_user}@${dialed_domain})}
>     >
>     >
>     >     Best regards
>     >     Peter
>     >
>     >     Anthony Minessale schrieb:
>     >     > edit the dial-string for that user in the directory xml to
>     try the
>     >     > extension on both profile at once
>     >     >
>     >     > On Thu, Feb 18, 2010 at 1:27 PM, Peter P GMX
>     >     <Prometheus001 at gmx.net <mailto:Prometheus001 at gmx.net>
>     <mailto:Prometheus001 at gmx.net <mailto:Prometheus001 at gmx.net>>
>     >     > <mailto:Prometheus001 at gmx.net
>     <mailto:Prometheus001 at gmx.net> <mailto:Prometheus001 at gmx.net
>     <mailto:Prometheus001 at gmx.net>>>>
>     >     wrote:
>     >     >
>     >     >     Hello,
>     >     >
>     >     >     in the standard setup - if a phone is registering to port
>     >     5060 - it is
>     >     >     bound to the "internal" profile. And I can dial it via
>     >     >     sofia/user/xxxx then.
>     >     >
>     >     >     However due to NAT issues I would like to have to 2
>     seperate
>     >     profiles
>     >     >     for SIP phones. For example I have a "local" profile
>     for all
>     >     devices
>     >     >     inside the LAN (e.g. Pattons und in future: local phones)
>     >     and another
>     >     >     "internal" profile which allows also external phones via
>     >     >     external-xxx-ip. That way I would like to ensure that
>     local
>     >     phones
>     >     >     have
>     >     >     nothing to do with natted adresses and that external
>     phones can
>     >     >     register
>     >     >     via external IPs.
>     >     >
>     >     >     Question How do I manage that I can register a phone
>     to the
>     >     "local"
>     >     >     profile and being able to dial that phone via
>     sofia/user/xxxxx?
>     >     >
>     >     >     Or do I think too complicated and there is simply nothing
>     >     special
>     >     >     to do?
>     >     >
>     >     >     Best regards
>     >     >     Peter
>     >     >
>     >     >     _______________________________________________
>     >     >     FreeSWITCH-users mailing list
>     >     >     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     >     <mailto:FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>>
>     >     >     <mailto:FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     >     <mailto:FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>>>
>     >     >    
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     >     >
>     >    
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     >     >     http://www.freeswitch.org
>     >     >
>     >     >
>     >     >
>     >     >
>     >     > --
>     >     > Anthony Minessale II
>     >     >
>     >     > FreeSWITCH http://www.freeswitch.org/
>     >     > ClueCon http://www.cluecon.com/
>     >     > Twitter: http://twitter.com/FreeSWITCH_wire
>     >     >
>     >     > AIM: anthm
>     >     > MSN:anthony_minessale at hotmail.com
>     <mailto:MSN%3Aanthony_minessale at hotmail.com>
>     >     <mailto:MSN%3Aanthony_minessale at hotmail.com
>     <mailto:MSN%253Aanthony_minessale at hotmail.com>>
>     >     > <mailto:MSN%3Aanthony_minessale at hotmail.com
>     <mailto:MSN%253Aanthony_minessale at hotmail.com>
>     >     <mailto:MSN%253Aanthony_minessale at hotmail.com
>     <mailto:MSN%25253Aanthony_minessale at hotmail.com>>>
>     >     > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>     <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
>     >     <mailto:PAYPAL%3Aanthony.minessale at gmail.com
>     <mailto:PAYPAL%253Aanthony.minessale at gmail.com>>
>     >     > <mailto:PAYPAL%3Aanthony.minessale at gmail.com
>     <mailto:PAYPAL%253Aanthony.minessale at gmail.com>
>     >     <mailto:PAYPAL%253Aanthony.minessale at gmail.com
>     <mailto:PAYPAL%25253Aanthony.minessale at gmail.com>>>
>     >     > IRC: irc.freenode.net <http://irc.freenode.net>
>     <http://irc.freenode.net>
>     >     <http://irc.freenode.net> #freeswitch
>     >     >
>     >     > FreeSWITCH Developer Conference
>     >     > sip:888 at conference.freeswitch.org
>     <mailto:sip%3A888 at conference.freeswitch.org>
>     >     <mailto:sip%3A888 at conference.freeswitch.org
>     <mailto:sip%253A888 at conference.freeswitch.org>>
>     >     > <mailto:sip%3A888 at conference.freeswitch.org
>     <mailto:sip%253A888 at conference.freeswitch.org>
>     >     <mailto:sip%253A888 at conference.freeswitch.org
>     <mailto:sip%25253A888 at conference.freeswitch.org>>>
>     >     > iax:guest at conference.freeswitch.org/888
>     <http://iax:guest at conference.freeswitch.org/888>
>     >     <http://iax:guest at conference.freeswitch.org/888>
>     >     > <http://iax:guest at conference.freeswitch.org/888>
>     >     > googletalk:conf+888 at conference.freeswitch.org
>     <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
>     >     <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org
>     <mailto:googletalk%253Aconf%252B888 at conference.freeswitch.org>>
>     >     > <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org
>     <mailto:googletalk%253Aconf%252B888 at conference.freeswitch.org>
>     >    
>     <mailto:googletalk%253Aconf%252B888 at conference.freeswitch.org
>     <mailto:googletalk%25253Aconf%25252B888 at conference.freeswitch.org>>>
>     >     > pstn:+19193869900
>     >     >
>     >    
>     ------------------------------------------------------------------------
>     >     >
>     >     > _______________________________________________
>     >     > FreeSWITCH-users mailing list
>     >     > FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     >     <mailto:FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>>
>     >     > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     >     >
>     >    
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     >     > http://www.freeswitch.org
>     >     >
>     >
>     >     _______________________________________________
>     >     FreeSWITCH-users mailing list
>     >     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     >     <mailto:FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>>
>     >     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     >    
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     >     http://www.freeswitch.org
>     >
>     >
>     >
>     >
>     > --
>     > Anthony Minessale II
>     >
>     > FreeSWITCH http://www.freeswitch.org/
>     > ClueCon http://www.cluecon.com/
>     > Twitter: http://twitter.com/FreeSWITCH_wire
>     >
>     > AIM: anthm
>     > MSN:anthony_minessale at hotmail.com
>     <mailto:MSN%3Aanthony_minessale at hotmail.com>
>     > <mailto:MSN%3Aanthony_minessale at hotmail.com
>     <mailto:MSN%253Aanthony_minessale at hotmail.com>>
>     > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>     <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
>     > <mailto:PAYPAL%3Aanthony.minessale at gmail.com
>     <mailto:PAYPAL%253Aanthony.minessale at gmail.com>>
>     > IRC: irc.freenode.net <http://irc.freenode.net>
>     <http://irc.freenode.net> #freeswitch
>     >
>     > FreeSWITCH Developer Conference
>     > sip:888 at conference.freeswitch.org
>     <mailto:sip%3A888 at conference.freeswitch.org>
>     > <mailto:sip%3A888 at conference.freeswitch.org
>     <mailto:sip%253A888 at conference.freeswitch.org>>
>     > iax:guest at conference.freeswitch.org/888
>     <http://iax:guest at conference.freeswitch.org/888>
>     > <http://iax:guest at conference.freeswitch.org/888>
>     > googletalk:conf+888 at conference.freeswitch.org
>     <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
>     > <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org
>     <mailto:googletalk%253Aconf%252B888 at conference.freeswitch.org>>
>     > pstn:+19193869900
>     >
>     ------------------------------------------------------------------------
>     >
>     > _______________________________________________
>     > FreeSWITCH-users mailing list
>     > FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     >
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     > http://www.freeswitch.org
>     >
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   


From will.traenkle at yahoo.com  Fri Feb 19 22:26:18 2010
From: will.traenkle at yahoo.com (William Traenkle)
Date: Fri, 19 Feb 2010 22:26:18 -0800 (PST)
Subject: [Freeswitch-users] freeswitch.serial
Message-ID: <921026.87533.qm@web57602.mail.re1.yahoo.com>

A couple of quick questions:

1) what is the freeswitch.serial file under the &base_dir/conf directory?

2) can I change its location?

Thanks,

-Will


      
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100219/60761da9/attachment.html 

From mike at jerris.com  Sat Feb 20 17:47:17 2010
From: mike at jerris.com (Michael Jerris)
Date: Sat, 20 Feb 2010 20:47:17 -0500
Subject: [Freeswitch-users] stream a file multicast with mod_esf
In-Reply-To: <a792304b1002192150n44cc500fnd5b2ecb7cff9dd1a@mail.gmail.com>
References: <B98B6921358D0C49BD21CD023F1F52C302623B4B@hbghs04a.ww300.siemens.net>
	<7C996136-963A-42D2-B0FE-D2729F968E52@freeswitch.org>
	<27071973.post@talk.nabble.com>
	<378CDEC3-5A41-4EF9-927A-311F2531E6AB@freeswitch.org>
	<a792304b1001080725y19ff28camd37425927550f311@mail.gmail.com>
	<191c3a031001081848w9036207yf6a4b473ec233243@mail.gmail.com>
	<85C74329-0D7C-48B2-B7A8-D4AA67C6E3F2@freeswitch.org>
	<a792304b1002182209m43ada13cp8ed3d91ad509f0a2@mail.gmail.com>
	<DBA4E291-9377-489A-841A-1A3D83880DF2@jerris.com>
	<a792304b1002190002u50ec6912g1905ffc63a119174@mail.gmail.com>
	<a792304b1002192150n44cc500fnd5b2ecb7cff9dd1a@mail.gmail.com>
Message-ID: <795CD2F3-1395-4C08-A0E7-F5A7BA85F938@jerris.com>

You will need to create the codec for what you need, I think it is hardcoded in there to PCMU at the moment, correct?  This will of course need to match the stream its reading.

Mike

On Feb 20, 2010, at 12:50 AM, MohammedShehzad wrote:

> 
> 
> On Fri, Feb 19, 2010 at 1:32 PM, MohammedShehzad <pmhshz at gmail.com> wrote:
> 
> 
> On Fri, Feb 19, 2010 at 12:59 PM, Michael Jerris <mike at jerris.com> wrote:
> Listening on multicast is noting special for multicast, it is just
> like reading any other udp socket
> 
> Mike
> 
> Correct, but I have to play those audio stream back to caller taking care of the audio codec  and other things, do anybody have any idea in that part? Please let me know that.
> -- 
> 
> -MohammedShehzad
> 
> I am able to receive the play the multicasted RAW PCMU RTP (modified the skel of format provided by brian), so that caller can hear the multicast which done by other Freeswitch server using mod_esf application, but when i change the caller's codec from PCMU to something else, it breaks.

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100220/753626b2/attachment.html 

From mike at jerris.com  Sat Feb 20 17:48:39 2010
From: mike at jerris.com (Michael Jerris)
Date: Sat, 20 Feb 2010 20:48:39 -0500
Subject: [Freeswitch-users] outbound calls
In-Reply-To: <C9E83EA051BC49D8AB9425D5D1225C27@MOBILEE1705>
References: <C9E83EA051BC49D8AB9425D5D1225C27@MOBILEE1705>
Message-ID: <F0495877-E2B4-4830-AAA0-889D95CC2EEA@jerris.com>

Sounds like the remote end does not like the codecs we are offering.  There are some vars to adjust this, by default we only offer the 1 codec chosen with the a leg of the call.

Mike

On Feb 20, 2010, at 1:36 PM, Madovsky wrote:

> Hello,
>  
> I'm able to transcode a cal between 2 local legs,
> but when a local user call an oubound call,
> the call hangs up saying "not acceptable here",
> so it doesn't transcode.
>  

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100220/b7289bac/attachment.html 

From mike at jerris.com  Sat Feb 20 17:50:37 2010
From: mike at jerris.com (Michael Jerris)
Date: Sat, 20 Feb 2010 20:50:37 -0500
Subject: [Freeswitch-users] How to exchange media when I developed new
	endpoint module?
In-Reply-To: <ebbc2d551002201327o7a7bad26g48cac48f610aed55@mail.gmail.com>
References: <93b0f8ce1002091914q581f12fvd1de01e2181f0f3a@mail.gmail.com>
	<ebbc2d551002091932u52dbf1b6jfecf2dfdb43b8a7c@mail.gmail.com>
	<4021C303-1AC3-48F2-A17B-63C02C1C54D4@freeswitch.org>
	<93b0f8ce1002191722h5782b787qe56a06dd0cb94e76@mail.gmail.com>
	<ebbc2d551002201327o7a7bad26g48cac48f610aed55@mail.gmail.com>
Message-ID: <77784B66-AF0B-47FE-9DDB-D274F22E65E9@jerris.com>

If you are developing a module for hardware and you do not already have and want to use code for all the signaling, (pri, analog, etc) then take a look down at openzap.  This has all that for you already, and should be fairly trivial to implement a new piece of hardware.

Mike

On Feb 20, 2010, at 4:27 PM, Jo?o Mesquita wrote:

> I developed the current implementation of mod_khomp. I wouldn't take it as an example for anything since there has been no activity there for the past 4 months. If you care to share a snippet of your code, maybe we can help better.
> 
> On Fri, Feb 19, 2010 at 11:22 PM, ??? <spiritonly at gmail.com> wrote:
> Do you know mod_khomp? You can found it in FS wiki. I am developing an endpoint module like it.
> So you can give me some advice to bridge two session?
> 
> 
> On Wed, Feb 10, 2010 at 11:44 AM, Brian West <brian at freeswitch.org> wrote:
> But the bigger question is what protocol are you doing that you have to create your own endpoint module?
> 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100220/c514a77a/attachment.html 

From rupa at rupa.com  Sat Feb 20 17:53:31 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Sat, 20 Feb 2010 19:53:31 -0600
Subject: [Freeswitch-users] freeswitch.serial
In-Reply-To: <921026.87533.qm@web57602.mail.re1.yahoo.com>
References: <921026.87533.qm@web57602.mail.re1.yahoo.com>
Message-ID: <b39273531002201753x16c7be04n1a73d181141bee64@mail.gmail.com>

1) The only current user of it is zrtp

2) it is configured to be in the conf directory, can't move it without
changing src.

On Sat, Feb 20, 2010 at 12:26 AM, William Traenkle
<will.traenkle at yahoo.com>wrote:

> A couple of quick questions:
>
> 1) what is the freeswitch.serial file under the &base_dir/conf directory?
>
> 2) can I change its location?
>
> Thanks,
>
> -Will
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100220/ab86ecb2/attachment.html 

From mike at jerris.com  Sat Feb 20 17:53:58 2010
From: mike at jerris.com (Michael Jerris)
Date: Sat, 20 Feb 2010 20:53:58 -0500
Subject: [Freeswitch-users] freeswitch.serial
In-Reply-To: <921026.87533.qm@web57602.mail.re1.yahoo.com>
References: <921026.87533.qm@web57602.mail.re1.yahoo.com>
Message-ID: <BCB79C3E-4EFB-4FFA-B4D9-B4CB69A8F947@jerris.com>

On Feb 20, 2010, at 1:26 AM, William Traenkle wrote:

> A couple of quick questions:
> 
> 1) what is the freeswitch.serial file under the &base_dir/conf directory?

It is a unique number for that instance of FreeSWITCH, it is mostly used for zrtp

> 2) can I change its location?

No


Mike
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100220/345f450e/attachment.html 

From infos at madovsky.org  Sat Feb 20 17:57:44 2010
From: infos at madovsky.org (Madovsky)
Date: Sat, 20 Feb 2010 20:57:44 -0500
Subject: [Freeswitch-users] outbound calls
References: <C9E83EA051BC49D8AB9425D5D1225C27@MOBILEE1705>
	<F0495877-E2B4-4830-AAA0-889D95CC2EEA@jerris.com>
Message-ID: <634B5DC983EC4267993346BB172A4FD1@MOBILEE1705>

yes I know it is codec problem,
but what vars it needs to force transcoding when
B leg doesn't match any A leg  codec ?
in vars.xml example I can see only global_codecs_prefs and outbound_codecs prefs correctly set

Thanks
  ----- OriginaleMessage ----- 
  From: Michael Jerris 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Saturday, February 20, 2010 8:48 PM
  Subject: Re: [Freeswitch-users] outbound callsh


  Sounds like the remote end does not like the codecs we are offering.  There are some vars to adjust this, by default we only offer the 1 codec chosen with the a leg of the call.


  Mike


  On Feb 20, 2010, at 1:36 PM, Madovsky wrote:


    Hello,

    I'm able to transcode a cal between 2 local legs,
    but when a local user call an oubound call,
    the call hangs up saying "not acceptable here",
    so it doesn't transcode.





------------------------------------------------------------------------------


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100220/ebd19340/attachment-0001.html 

From frank at carmickle.com  Sat Feb 20 18:00:38 2010
From: frank at carmickle.com (Frank Carmickle)
Date: Sat, 20 Feb 2010 21:00:38 -0500
Subject: [Freeswitch-users] outbound calls
In-Reply-To: <C9E83EA051BC49D8AB9425D5D1225C27@MOBILEE1705>
References: <C9E83EA051BC49D8AB9425D5D1225C27@MOBILEE1705>
Message-ID: <20100221013429.GA9832@base.carmickle.com>

On Sat, Feb 20, Madovsky wrote:
> Hello,
> 
> I'm able to transcode a cal between 2 local legs,
> but when a local user call an oubound call,
> the call hangs up saying "not acceptable here",
> so it doesn't transcode.
> 
> Any idea ?

What are your outbound_codec_prefs set to in your vars.xml?

--FC


From spiritonly at gmail.com  Sat Feb 20 18:12:52 2010
From: spiritonly at gmail.com (=?UTF-8?B?5p2o5rGf6aqF?=)
Date: Sun, 21 Feb 2010 10:12:52 +0800
Subject: [Freeswitch-users] How to exchange media when I developed new
	endpoint module?
In-Reply-To: <ebbc2d551002201327o7a7bad26g48cac48f610aed55@mail.gmail.com>
References: <93b0f8ce1002091914q581f12fvd1de01e2181f0f3a@mail.gmail.com>
	<ebbc2d551002091932u52dbf1b6jfecf2dfdb43b8a7c@mail.gmail.com>
	<4021C303-1AC3-48F2-A17B-63C02C1C54D4@freeswitch.org>
	<93b0f8ce1002191722h5782b787qe56a06dd0cb94e76@mail.gmail.com>
	<ebbc2d551002201327o7a7bad26g48cac48f610aed55@mail.gmail.com>
Message-ID: <93b0f8ce1002201812x24cf3853u920bdba87f938b43@mail.gmail.com>

Oh, My God! You are the developer of mod_khomp! I read your blog, checked
out mod_khomp from google code, and found it had not update for a long time.
So what is your roadmap about mod_khomp?

I share my snippet in FS pastebin, you can find it with '
http://pastebin.freeswitch.org/12192 '.

My MSN is spiritonly at live.cn.
My Gtalk is spiritonly at gmail.com.
Hope we can keep connected.


2010/2/21 Jo?o Mesquita <jmesquita at freeswitch.org>

> I developed the current implementation of mod_khomp. I wouldn't take it as
> an example for anything since there has been no activity there for the past
> 4 months. If you care to share a snippet of your code, maybe we can help
> better.
>
>
> JM
>
>
>
> On Fri, Feb 19, 2010 at 11:22 PM, ??? <spiritonly at gmail.com> wrote:
>
>> Do you know mod_khomp? You can found it in FS wiki. I am developing an
>> endpoint module like it.
>> So you can give me some advice to bridge two session?
>>
>>
>> On Wed, Feb 10, 2010 at 11:44 AM, Brian West <brian at freeswitch.org>wrote:
>>
>>> But the bigger question is what protocol are you doing that you have to
>>> create your own endpoint module?
>>>
>>> /b
>>>
>>> On Feb 9, 2010, at 9:32 PM, Jo?o Mesquita wrote:
>>>
>>> > You should look at read_frame and write_frame implementations of other
>>> endpoint modules.
>>> >
>>> > This should pretty much tell you how things work...
>>> >
>>> > Jo?o Mesquita
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100221/e9a0626b/attachment.html 

From frank at carmickle.com  Sat Feb 20 18:15:35 2010
From: frank at carmickle.com (Frank Carmickle)
Date: Sat, 20 Feb 2010 21:15:35 -0500
Subject: [Freeswitch-users] outbound calls
In-Reply-To: <634B5DC983EC4267993346BB172A4FD1@MOBILEE1705>
References: <C9E83EA051BC49D8AB9425D5D1225C27@MOBILEE1705>
	<F0495877-E2B4-4830-AAA0-889D95CC2EEA@jerris.com>
	<634B5DC983EC4267993346BB172A4FD1@MOBILEE1705>
Message-ID: <20100221021535.GB9832@base.carmickle.com>

On Sat, Feb 20, Madovsky wrote:
> yes I know it is codec problem,
> but what vars it needs to force transcoding when
> B leg doesn't match any A leg  codec ?
> in vars.xml example I can see only global_codecs_prefs and outbound_codecs prefs correctly set

You know the codec you want to match on the B leg so in the dialplan

>From the wiki http://wiki.freeswitch.org/wiki/Codec_Negotiation
<action application="bridge" data="{absolute_codec_string='PCMA,PCMU'}sofia/gateway/mygateway/mynumber"/>
 
HTH
--FC


From infos at madovsky.org  Sat Feb 20 18:19:47 2010
From: infos at madovsky.org (Madovsky)
Date: Sat, 20 Feb 2010 21:19:47 -0500
Subject: [Freeswitch-users] outbound calls
References: <C9E83EA051BC49D8AB9425D5D1225C27@MOBILEE1705>
	<20100221013429.GA9832@base.carmickle.com>
Message-ID: <235403598AE24079969DAAE9A61E823B@MOBILEE1705>


----- Original Message ----- 
From: "Frank Carmickle" <frank at carmickle.com>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Saturday, February 20, 2010 9:00 PM
Subject: Re: [Freeswitch-users] outbound calls


> On Sat, Feb 20, Madovsky wrote:
>> Hello,
>>
>> I'm able to transcode a cal between 2 local legs,
>> but when a local user call an oubound call,
>> the call hangs up saying "not acceptable here",
>> so it doesn't transcode.
>>
>> Any idea ?
>
> What are your outbound_codec_prefs set to in your vars.xml?
>
> --FC
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

Hi Frank,

global
<X-PRE-PROCESS cmd="set" 
data="global_codec_prefs=speex at 16000k,PCMU,PCMA,GSM"/>

outbound

<X-PRE-PROCESS cmd="set" 
data="outbound_codec_prefs=speex at 16000k,PCMU,PCMA,GSM"/>


Thanks

Franck 



From brian at microcomaustralia.com.au  Sat Feb 20 18:21:34 2010
From: brian at microcomaustralia.com.au (Brian May)
Date: Sun, 21 Feb 2010 13:21:34 +1100
Subject: [Freeswitch-users] list all users
Message-ID: <3c5cf5261002201821q7b9f024bt7ab32b10946a5370@mail.gmail.com>

Hello,

How do I get a list of all users? Including users that are not registered yet?

(note: I have already changed the domain to microcomaustralia.com.au)

I have tried:

freeswitch at voyage> user_exists brian at microcomaustralia.com.au
false

i believe this user should be defined in
conf/directory/default/brian.xml, however it is like nothing in
conf/directory/default.xml is being read. I want to prove if this is
the case or not.

When I try to register I get this error:

2010-02-21 13:09:02.568081 [WARNING] sofia_reg.c:1019 SIP auth failure
(REGISTER) on sofia profile 'internal' for
[brian at microcomaustralia.com.au] from ip 192.168.87.14

which I think might be because it can't find the matching user entry.

Thanks
-- 
Brian May <brian at microcomaustralia.com.au>


From infos at madovsky.org  Sat Feb 20 18:36:41 2010
From: infos at madovsky.org (Madovsky)
Date: Sat, 20 Feb 2010 21:36:41 -0500
Subject: [Freeswitch-users] outbound calls
References: <C9E83EA051BC49D8AB9425D5D1225C27@MOBILEE1705><F0495877-E2B4-4830-AAA0-889D95CC2EEA@jerris.com><634B5DC983EC4267993346BB172A4FD1@MOBILEE1705>
	<20100221021535.GB9832@base.carmickle.com>
Message-ID: <849A848F7F7142639B883B5F8471B5AB@MOBILEE1705>


----- Original Message ----- 
From: "Frank Carmickle" <frank at carmickle.com>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Saturday, February 20, 2010 9:15 PM
Subject: Re: [Freeswitch-users] outbound calls


> On Sat, Feb 20, Madovsky wrote:
>> yes I know it is codec problem,
>> but what vars it needs to force transcoding when
>> B leg doesn't match any A leg  codec ?
>> in vars.xml example I can see only global_codecs_prefs and 
>> outbound_codecs prefs correctly set
>
> You know the codec you want to match on the B leg so in the dialplan
>
>>From the wiki http://wiki.freeswitch.org/wiki/Codec_Negotiation
> <action application="bridge" 
> data="{absolute_codec_string='PCMA,PCMU'}sofia/gateway/mygateway/mynumber"/>
>
> HTH
> --FC
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

No, in fact I don't know which codec will come from B leg,
but in my test I did the codec of B leg matches one codec in the outbound 
list but doesn't  transcode and
the call fails. but if A leg and B leg are local so it transcodes 
correctly...

Thanks 



From jason at jasonjgw.net  Sat Feb 20 18:37:41 2010
From: jason at jasonjgw.net (Jason White)
Date: Sun, 21 Feb 2010 13:37:41 +1100
Subject: [Freeswitch-users] list all users
In-Reply-To: <3c5cf5261002201821q7b9f024bt7ab32b10946a5370@mail.gmail.com>
References: <3c5cf5261002201821q7b9f024bt7ab32b10946a5370@mail.gmail.com>
Message-ID: <20100221023741.GA15005@jdc.jasonjgw.net>

Brian May <brian at microcomaustralia.com.au> wrote:
> (note: I have already changed the domain to microcomaustralia.com.au)
> 
> I have tried:
> 
> freeswitch at voyage> user_exists brian at microcomaustralia.com.au
> false
> 
> i believe this user should be defined in
> conf/directory/default/brian.xml, however it is like nothing in
> conf/directory/default.xml is being read. I want to prove if this is
> the case or not.

Have a look at /opt/freeswitch/log/freeswitch.xml.fsxml
This is the compilation of all the XML files, and it's the file which is
actually consulted to do the configuration lookup. If your user entry is in
there, then you know that FreeSWITCH has it.

Also, make sure the user-agent is configured to register to the domain, not to
an IP address.



From infos at madovsky.org  Sat Feb 20 18:40:57 2010
From: infos at madovsky.org (Madovsky)
Date: Sat, 20 Feb 2010 21:40:57 -0500
Subject: [Freeswitch-users] outbound calls
References: <C9E83EA051BC49D8AB9425D5D1225C27@MOBILEE1705><F0495877-E2B4-4830-AAA0-889D95CC2EEA@jerris.com><634B5DC983EC4267993346BB172A4FD1@MOBILEE1705>
	<20100221021535.GB9832@base.carmickle.com>
Message-ID: <9954CEB3764E4A11BEBE0DC14B2C968A@MOBILEE1705>


----- Original Message ----- 
From: "Frank Carmickle" <frank at carmickle.com>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Saturday, February 20, 2010 9:15 PM
Subject: Re: [Freeswitch-users] outbound calls


> On Sat, Feb 20, Madovsky wrote:
>> yes I know it is codec problem,
>> but what vars it needs to force transcoding when
>> B leg doesn't match any A leg  codec ?
>> in vars.xml example I can see only global_codecs_prefs and 
>> outbound_codecs prefs correctly set
>
> You know the codec you want to match on the B leg so in the dialplan
>
>>From the wiki http://wiki.freeswitch.org/wiki/Codec_Negotiation
> <action application="bridge" 
> data="{absolute_codec_string='PCMA,PCMU'}sofia/gateway/mygateway/mynumber"/>
>
> HTH
> --FC
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

I retried your suggestion (that I already did 3 days ago)
but no work, it's the same 



From mcampbellsmith at gmail.com  Sat Feb 20 19:08:45 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Sun, 21 Feb 2010 14:08:45 +1100
Subject: [Freeswitch-users] list all users
In-Reply-To: <3c5cf5261002201821q7b9f024bt7ab32b10946a5370@mail.gmail.com>
References: <3c5cf5261002201821q7b9f024bt7ab32b10946a5370@mail.gmail.com>
Message-ID: <33c87fa31002201908x55e43e62g32b340872df9863d@mail.gmail.com>

Try

user_exists id <domain>

Where <domain> is the output from the command 'eval ${domain}'

On Sun, Feb 21, 2010 at 1:21 PM, Brian May
<brian at microcomaustralia.com.au> wrote:
> Hello,
>
> How do I get a list of all users? Including users that are not registered yet?
>
> (note: I have already changed the domain to microcomaustralia.com.au)
>
> I have tried:
>
> freeswitch at voyage> user_exists brian at microcomaustralia.com.au
> false
>
> i believe this user should be defined in
> conf/directory/default/brian.xml, however it is like nothing in
> conf/directory/default.xml is being read. I want to prove if this is
> the case or not.
>
> When I try to register I get this error:
>
> 2010-02-21 13:09:02.568081 [WARNING] sofia_reg.c:1019 SIP auth failure
> (REGISTER) on sofia profile 'internal' for
> [brian at microcomaustralia.com.au] from ip 192.168.87.14
>
> which I think might be because it can't find the matching user entry.
>
> Thanks
> --
> Brian May <brian at microcomaustralia.com.au>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From frank at carmickle.com  Sat Feb 20 19:10:02 2010
From: frank at carmickle.com (Frank Carmickle)
Date: Sat, 20 Feb 2010 22:10:02 -0500
Subject: [Freeswitch-users] outbound calls
In-Reply-To: <9954CEB3764E4A11BEBE0DC14B2C968A@MOBILEE1705>
References: <20100221021535.GB9832@base.carmickle.com>
	<9954CEB3764E4A11BEBE0DC14B2C968A@MOBILEE1705>
Message-ID: <20100221031001.GC9832@base.carmickle.com>

On Sat, Feb 20, Madovsky wrote:
> 
> ----- Original Message ----- 
> From: "Frank Carmickle" <frank at carmickle.com>
> To: <freeswitch-users at lists.freeswitch.org>
> Sent: Saturday, February 20, 2010 9:15 PM
> Subject: Re: [Freeswitch-users] outbound calls
> 
> 
> > On Sat, Feb 20, Madovsky wrote:
> >> yes I know it is codec problem,
> >> but what vars it needs to force transcoding when
> >> B leg doesn't match any A leg  codec ?
> >> in vars.xml example I can see only global_codecs_prefs and 
> >> outbound_codecs prefs correctly set
> >
> > You know the codec you want to match on the B leg so in the dialplan
> >
> >>From the wiki http://wiki.freeswitch.org/wiki/Codec_Negotiation
> > <action application="bridge" 
> > data="{absolute_codec_string='PCMA,PCMU'}sofia/gateway/mygateway/mynumber"/>
> >
> > HTH
> > --FC
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 
> I retried your suggestion (that I already did 3 days ago)
> but no work, it's the same 

Is the external phone using the external profile?  Do the different profiles have differing codec settings?

--FC


From infos at madovsky.org  Sat Feb 20 19:17:56 2010
From: infos at madovsky.org (Madovsky)
Date: Sat, 20 Feb 2010 22:17:56 -0500
Subject: [Freeswitch-users] outbound calls
References: <C9E83EA051BC49D8AB9425D5D1225C27@MOBILEE1705><F0495877-E2B4-4830-AAA0-889D95CC2EEA@jerris.com><634B5DC983EC4267993346BB172A4FD1@MOBILEE1705>
	<20100221021535.GB9832@base.carmickle.com>
Message-ID: <793742B9BB1D4EB3990A6A7DA72A1A4B@MOBILEE1705>


----- Original Message ----- 
From: "Frank Carmickle" <frank at carmickle.com>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Saturday, February 20, 2010 9:15 PM
Subject: Re: [Freeswitch-users] outbound calls


> On Sat, Feb 20, Madovsky wrote:
>> yes I know it is codec problem,
>> but what vars it needs to force transcoding when
>> B leg doesn't match any A leg  codec ?
>> in vars.xml example I can see only global_codecs_prefs and 
>> outbound_codecs prefs correctly set
>
> You know the codec you want to match on the B leg so in the dialplan
>
>>From the wiki http://wiki.freeswitch.org/wiki/Codec_Negotiation
> <action application="bridge" 
> data="{absolute_codec_string='PCMA,PCMU'}sofia/gateway/mygateway/mynumber"/>
>
> HTH
> --FC
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

Ok now I can call from an external sip account to a local FS sip account and 
it
transcodes correctly (I made the same mistake as hundred users I imagine 
like I forgot that the exterrnal port is 5080, so I changed to 5060). but if 
I do the contrary (internal A leg to ext B leg) it doesn't .
is outbound-late-negociation exists ?



From infos at madovsky.org  Sat Feb 20 19:22:32 2010
From: infos at madovsky.org (Madovsky)
Date: Sat, 20 Feb 2010 22:22:32 -0500
Subject: [Freeswitch-users] outbound calls
References: <20100221021535.GB9832@base.carmickle.com><9954CEB3764E4A11BEBE0DC14B2C968A@MOBILEE1705>
	<20100221031001.GC9832@base.carmickle.com>
Message-ID: <74283D2E3F924DC5800BAFBAE2BF7F57@MOBILEE1705>


----- Original Message ----- 
From: "Frank Carmickle" <frank at carmickle.com>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Saturday, February 20, 2010 10:10 PM
Subject: Re: [Freeswitch-users] outbound calls


> On Sat, Feb 20, Madovsky wrote:
>>
>> ----- Original Message ----- 
>> From: "Frank Carmickle" <frank at carmickle.com>
>> To: <freeswitch-users at lists.freeswitch.org>
>> Sent: Saturday, February 20, 2010 9:15 PM
>> Subject: Re: [Freeswitch-users] outbound calls
>>
>>
>> > On Sat, Feb 20, Madovsky wrote:
>> >> yes I know it is codec problem,
>> >> but what vars it needs to force transcoding when
>> >> B leg doesn't match any A leg  codec ?
>> >> in vars.xml example I can see only global_codecs_prefs and
>> >> outbound_codecs prefs correctly set
>> >
>> > You know the codec you want to match on the B leg so in the dialplan
>> >
>> >>From the wiki http://wiki.freeswitch.org/wiki/Codec_Negotiation
>> > <action application="bridge"
>> > data="{absolute_codec_string='PCMA,PCMU'}sofia/gateway/mygateway/mynumber"/>
>> >
>> > HTH
>> > --FC
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>> I retried your suggestion (that I already did 3 days ago)
>> but no work, it's the same
>
> Is the external phone using the external profile?  Do the different 
> profiles have differing codec settings?
>
> --FC
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


FIrst it was the same vars, now I succeed to transcode from a call from Bleg 
to Aleg in listing
the only Aleg codec available ...




From brian at microcomaustralia.com.au  Sat Feb 20 20:00:15 2010
From: brian at microcomaustralia.com.au (Brian May)
Date: Sun, 21 Feb 2010 15:00:15 +1100
Subject: [Freeswitch-users] list all users
In-Reply-To: <33c87fa31002201908x55e43e62g32b340872df9863d@mail.gmail.com>
References: <3c5cf5261002201821q7b9f024bt7ab32b10946a5370@mail.gmail.com>
	<33c87fa31002201908x55e43e62g32b340872df9863d@mail.gmail.com>
Message-ID: <3c5cf5261002202000h12b9d6b2ybefd5422fd4f9ee5@mail.gmail.com>

Ok, I solved one problem. Yes, the user really was there, the problem
was I followed the wiki documentation for generating a password hash:

echo "username:domain:password" | openssl dgst -md5

This won't work because echo will append a new line; the correct version is:

echo -n "username:domain:password" | openssl dgst -md5

I have updated the wiki. Now the client will register, and can make
outgoing calls.

Incoming calls don't work however, I get the messages:

2010-02-21 14:45:51.400081 [ERR] switch_ivr_originate.c:2387 Cannot
create outgoing channel of type [error] cause: [USER_NOT_REGISTERED]
2010-02-21 14:45:51.410081 [ERR] switch_ivr_originate.c:2387 Cannot
create outgoing channel of type [user] cause: [USER_NOT_REGISTERED]

As per one suggestion (as far as I can tell this shouldn't be
required) I tried changing this (in dialplan/default.xml):

<action application="bridge" data="user/${dialed_extension}@${domain_name}"/>

with:

<action application="bridge"
data="sofia/internal/${dialed_extension}@${domain_name}"/>

However that isn't using the registered IP address for the
brian at microcomaustralia.com.au user; rather it does a DNS lookup for
$(domain_name) and tries to contact that address instead, this is
wrong; that
DNS address resolves to my webserver, not the jabber client.


On 21 February 2010 14:08, Mark Campbell-Smith <mcampbellsmith at gmail.com> wrote:
> Try
>
> user_exists id <domain>
>
> Where <domain> is the output from the command 'eval ${domain}'

Curiously that still doesn't work, even though I registered:

freeswitch at voyage> eval ${domain}

microcomaustralia.com.au
freeswitch at voyage>
freeswitch at voyage> user_exists id brian at microcomaustralia.com.au

false


freeswitch at voyage> sofia status profile microcomaustralia.com.au

=================================================================================================
Name             	microcomaustralia.com.au
Domain Name      	N/A
Alias Of         	internal
Auto-NAT         	false
DBName           	sofia_reg_internal
Pres Hosts       	
Dialplan         	XML
Context          	public
Challenge Realm  	auto_from
RTP-IP           	192.168.86.4
SIP-IP           	192.168.86.4
URL              	sip:mod_sofia at 192.168.86.4:5060
BIND-URL         	sip:mod_sofia at 192.168.86.4:5060
HOLD-MUSIC       	local_stream://moh
OUTBOUND-PROXY   	N/A
CODECS IN        	G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM
CODECS OUT       	G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM
TEL-EVENT        	101
DTMF-MODE        	rfc2833
CNG              	13
SESSION-TO       	0
MAX-DIALOG       	0
NOMEDIA          	false
LATE-NEG         	false
PROXY-MEDIA      	false
AGGRESSIVENAT    	false
STUN-ENABLED     	true
STUN-AUTO-DISABLE	false
CALLS-IN         	0
FAILED-CALLS-IN  	0
CALLS-OUT        	4
FAILED-CALLS-OUT 	4

Registrations:
=================================================================================================
Call-ID:    	advktkbqoyvqmwm at andean.pri
User:       	brian at microcomaustralia.com.au
Contact:    	"Brian May"
<sip:brian at 192.168.87.14;fs_nat=yes;fs_path=sip%3Abrian%40192.168.87.14%3A5060>
Agent:      	Twinkle/1.4.2
Status:     	Registered(UDP-NAT)(unknown) EXP(2010-02-21 16:07:39)
Host:       	voyage
IP:         	192.168.87.14
Port:       	5060
Auth-User:  	brian
Auth-Realm: 	microcomaustralia.com.au
MWI-Account:	brian at microcomaustralia.com.au

=================================================================================================

-- 
Brian May <brian at microcomaustralia.com.au>


From jason at jasonjgw.net  Sat Feb 20 20:15:59 2010
From: jason at jasonjgw.net (Jason White)
Date: Sun, 21 Feb 2010 15:15:59 +1100
Subject: [Freeswitch-users] list all users
In-Reply-To: <3c5cf5261002202000h12b9d6b2ybefd5422fd4f9ee5@mail.gmail.com>
References: <3c5cf5261002201821q7b9f024bt7ab32b10946a5370@mail.gmail.com>
	<33c87fa31002201908x55e43e62g32b340872df9863d@mail.gmail.com>
	<3c5cf5261002202000h12b9d6b2ybefd5422fd4f9ee5@mail.gmail.com>
Message-ID: <20100221041559.GA21387@jdc.jasonjgw.net>

Brian May <brian at microcomaustralia.com.au> wrote:
 
> Incoming calls don't work however, I get the messages:
> 
> 2010-02-21 14:45:51.400081 [ERR] switch_ivr_originate.c:2387 Cannot
> create outgoing channel of type [error] cause: [USER_NOT_REGISTERED]
> 2010-02-21 14:45:51.410081 [ERR] switch_ivr_originate.c:2387 Cannot
> create outgoing channel of type [user] cause: [USER_NOT_REGISTERED]

what does sofia_contact username at microcomaustralia.com.au
show?

Maybe turning on debug logging at this point would help.
/log debug



From brian at microcomaustralia.com.au  Sat Feb 20 20:30:03 2010
From: brian at microcomaustralia.com.au (Brian May)
Date: Sun, 21 Feb 2010 15:30:03 +1100
Subject: [Freeswitch-users] list all users
In-Reply-To: <20100221041559.GA21387@jdc.jasonjgw.net>
References: <3c5cf5261002201821q7b9f024bt7ab32b10946a5370@mail.gmail.com>
	<33c87fa31002201908x55e43e62g32b340872df9863d@mail.gmail.com>
	<3c5cf5261002202000h12b9d6b2ybefd5422fd4f9ee5@mail.gmail.com>
	<20100221041559.GA21387@jdc.jasonjgw.net>
Message-ID: <3c5cf5261002202030l74935bbr1013547a315aaf6a@mail.gmail.com>

On 21 February 2010 15:15, Jason White <jason at jasonjgw.net> wrote:
> what does sofia_contact username at microcomaustralia.com.au
> show?

freeswitch at voyage> sofia_contact brian at microcomaustralia.com.au

sofia/internal/sip:brian at 192.168.87.14;fs_nat=yes;fs_path=sip%3Abrian%40192.168.87.14%3A5060


looks good to me...


> Maybe turning on debug logging at this point would help.
> /log debug


freeswitch at voyage> /log debug
Unknown Command: /log debug


Only debug I have found is:

freeswitch at voyage> sofia loglevel all 9

Sofia log level for component [all] has been set to [9]


however that doesn't display any extra information for this test case.
-- 
Brian May <brian at microcomaustralia.com.au>


From jason at jasonjgw.net  Sat Feb 20 21:40:30 2010
From: jason at jasonjgw.net (Jason White)
Date: Sun, 21 Feb 2010 16:40:30 +1100
Subject: [Freeswitch-users] list all users
In-Reply-To: <3c5cf5261002202030l74935bbr1013547a315aaf6a@mail.gmail.com>
References: <3c5cf5261002201821q7b9f024bt7ab32b10946a5370@mail.gmail.com>
	<33c87fa31002201908x55e43e62g32b340872df9863d@mail.gmail.com>
	<3c5cf5261002202000h12b9d6b2ybefd5422fd4f9ee5@mail.gmail.com>
	<20100221041559.GA21387@jdc.jasonjgw.net>
	<3c5cf5261002202030l74935bbr1013547a315aaf6a@mail.gmail.com>
Message-ID: <20100221054030.GA25531@jdc.jasonjgw.net>

Brian May <brian at microcomaustralia.com.au> wrote:
 
> freeswitch at voyage> sofia_contact brian at microcomaustralia.com.au
> 
> sofia/internal/sip:brian at 192.168.87.14;fs_nat=yes;fs_path=sip%3Abrian%40192.168.87.14%3A5060
> 
> 
> looks good to me...

It does.
> 
> 
> > Maybe turning on debug logging at this point would help.
> > /log debug
> 
> 
> freeswitch at voyage> /log debug
> Unknown Command: /log debug

I was assuming you were running fs_cli with FreeSWITCH in daemon mode, as is
usual. At the console it's a different command to enable debugging, but I
never run FreeSWITCH that way unless I'm trying to deal with a segfault or
other startup issue.



From brian at microcomaustralia.com.au  Sat Feb 20 21:46:41 2010
From: brian at microcomaustralia.com.au (Brian May)
Date: Sun, 21 Feb 2010 16:46:41 +1100
Subject: [Freeswitch-users] list all users
In-Reply-To: <3c5cf5261002202030l74935bbr1013547a315aaf6a@mail.gmail.com>
References: <3c5cf5261002201821q7b9f024bt7ab32b10946a5370@mail.gmail.com>
	<33c87fa31002201908x55e43e62g32b340872df9863d@mail.gmail.com>
	<3c5cf5261002202000h12b9d6b2ybefd5422fd4f9ee5@mail.gmail.com>
	<20100221041559.GA21387@jdc.jasonjgw.net>
	<3c5cf5261002202030l74935bbr1013547a315aaf6a@mail.gmail.com>
Message-ID: <3c5cf5261002202146sd1dc48drb274951b4be687c8@mail.gmail.com>

On 21 February 2010 15:30, Brian May <brian at microcomaustralia.com.au> wrote:
> freeswitch at voyage> sofia_contact brian at microcomaustralia.com.au
>
> sofia/internal/sip:brian at 192.168.87.14;fs_nat=yes;fs_path=sip%3Abrian%40192.168.87.14%3A5060

Just managed to disable NAT. Just in case. No NAT in use here.

freeswitch at voyage> sofia_contact brian at microcomaustralia.com.au
sofia/internal/sip:brian at 192.168.87.14


I know it is locating my user record, because the voice message I get
is "The person at the extension b-r-i-a-n is not available" when I
dialled the extension of 1000.

However, for some reason it doesn't seem to locate the registration record:

=== cut ===
freeswitch at voyage> sofia status profile internal reg brian


Registrations:
=================================================================================================
Call-ID:    	advktkbqoyvqmwm at andean.pri
User:       	brian at microcomaustralia.com.au
Contact:    	"Brian May" <sip:brian at 192.168.87.14>
Agent:      	Twinkle/1.4.2
Status:     	Registered(UDP)(unknown) EXP(2010-02-21 18:38:40)
Host:       	voyage
IP:         	192.168.87.14
Port:       	5060
Auth-User:  	brian
Auth-Realm: 	microcomaustralia.com.au
MWI-Account:	brian at microcomaustralia.com.au

=================================================================================================

freeswitch at voyage> sofia status profile internal reg
brian at microcomaustralia.com.au


Registrations:
=================================================================================================
=================================================================================================

=== cut ===


Ok then, afraid I can't explain this behaviour. Am sure 2nd command
was working recently...
-- 
Brian May <brian at microcomaustralia.com.au>


From brian at microcomaustralia.com.au  Sat Feb 20 21:56:25 2010
From: brian at microcomaustralia.com.au (Brian May)
Date: Sun, 21 Feb 2010 16:56:25 +1100
Subject: [Freeswitch-users] list all users
In-Reply-To: <20100221054030.GA25531@jdc.jasonjgw.net>
References: <3c5cf5261002201821q7b9f024bt7ab32b10946a5370@mail.gmail.com>
	<33c87fa31002201908x55e43e62g32b340872df9863d@mail.gmail.com>
	<3c5cf5261002202000h12b9d6b2ybefd5422fd4f9ee5@mail.gmail.com>
	<20100221041559.GA21387@jdc.jasonjgw.net>
	<3c5cf5261002202030l74935bbr1013547a315aaf6a@mail.gmail.com>
	<20100221054030.GA25531@jdc.jasonjgw.net>
Message-ID: <3c5cf5261002202156u2f2e455jbf1e864c738b77a3@mail.gmail.com>

On 21 February 2010 16:40, Jason White <jason at jasonjgw.net> wrote:
> I was assuming you were running fs_cli with FreeSWITCH in daemon mode, as is
> usual. At the console it's a different command to enable debugging, but I
> never run FreeSWITCH that way unless I'm trying to deal with a segfault or
> other startup issue.

Yes, I should do that too. Seems to confirm what I already know,
nothing new though :-(

EXECUTE OpenZAP/1:1/1000 bridge(user/1000 at microcomaustralia.com.au)
2010-02-21 16:59:56.790081 [DEBUG] switch_ivr_originate.c:1859
variable string 0 = [presence_id=1000 at microcomaustralia.com.au]
2010-02-21 16:59:56.790081 [ERR] switch_ivr_originate.c:2387 Cannot
create outgoing channel of type [error] cause: [USER_NOT_REGISTERED]
2010-02-21 16:59:56.790081 [DEBUG] switch_ivr_originate.c:3185
Originate Resulted in Error Cause: 606 [USER_NOT_REGISTERED]
2010-02-21 16:59:56.800081 [ERR] switch_ivr_originate.c:2387 Cannot
create outgoing channel of type [user] cause: [USER_NOT_REGISTERED]
2010-02-21 16:59:56.800081 [DEBUG] switch_ivr_originate.c:3185
Originate Resulted in Error Cause: 606 [USER_NOT_REGISTERED]
2010-02-21 16:59:56.800081 [INFO] mod_dptools.c:2353 Originate Failed.
 Cause: USER_NOT_REGISTERED

-- 
Brian May <brian at microcomaustralia.com.au>


From brian at microcomaustralia.com.au  Sat Feb 20 22:28:54 2010
From: brian at microcomaustralia.com.au (Brian May)
Date: Sun, 21 Feb 2010 17:28:54 +1100
Subject: [Freeswitch-users] list all users
In-Reply-To: <3c5cf5261002202156u2f2e455jbf1e864c738b77a3@mail.gmail.com>
References: <3c5cf5261002201821q7b9f024bt7ab32b10946a5370@mail.gmail.com>
	<33c87fa31002201908x55e43e62g32b340872df9863d@mail.gmail.com>
	<3c5cf5261002202000h12b9d6b2ybefd5422fd4f9ee5@mail.gmail.com>
	<20100221041559.GA21387@jdc.jasonjgw.net>
	<3c5cf5261002202030l74935bbr1013547a315aaf6a@mail.gmail.com>
	<20100221054030.GA25531@jdc.jasonjgw.net>
	<3c5cf5261002202156u2f2e455jbf1e864c738b77a3@mail.gmail.com>
Message-ID: <3c5cf5261002202228n7605936uf517fc608b38d2a2@mail.gmail.com>

Ok, not sure what best practise is here...

Previously, I had defined the user as such:

<user id="brian" number-alias="1000">

This seems to work a lot better if I just register it as the number:

<user id="1000">
-- 
Brian May <brian at microcomaustralia.com.au>


From jason at jasonjgw.net  Sat Feb 20 22:45:20 2010
From: jason at jasonjgw.net (Jason White)
Date: Sun, 21 Feb 2010 17:45:20 +1100
Subject: [Freeswitch-users] list all users
In-Reply-To: <3c5cf5261002202156u2f2e455jbf1e864c738b77a3@mail.gmail.com>
References: <3c5cf5261002201821q7b9f024bt7ab32b10946a5370@mail.gmail.com>
	<33c87fa31002201908x55e43e62g32b340872df9863d@mail.gmail.com>
	<3c5cf5261002202000h12b9d6b2ybefd5422fd4f9ee5@mail.gmail.com>
	<20100221041559.GA21387@jdc.jasonjgw.net>
	<3c5cf5261002202030l74935bbr1013547a315aaf6a@mail.gmail.com>
	<20100221054030.GA25531@jdc.jasonjgw.net>
	<3c5cf5261002202156u2f2e455jbf1e864c738b77a3@mail.gmail.com>
Message-ID: <20100221064520.GA26593@jdc.jasonjgw.net>

Brian May <brian at microcomaustralia.com.au> wrote:
 
> EXECUTE OpenZAP/1:1/1000 bridge(user/1000 at microcomaustralia.com.au)
> 2010-02-21 16:59:56.790081 [DEBUG] switch_ivr_originate.c:1859
> variable string 0 = [presence_id=1000 at microcomaustralia.com.au]
> 2010-02-21 16:59:56.790081 [ERR] switch_ivr_originate.c:2387 Cannot
> create outgoing channel of type [error] cause: [USER_NOT_REGISTERED]

I think it's looking for 1000 rather than the user name, which of course isn't
registered.

In the extension for the user in your dial-plan, you could always write:
<action application="bridge"
data="$sofia_contact(brian at microcomaustralia.com.au)"/>
or you could use a variable instead of writing the domain name explicitly, of
course, which would be better practice.



From jason at jasonjgw.net  Sat Feb 20 22:51:38 2010
From: jason at jasonjgw.net (Jason White)
Date: Sun, 21 Feb 2010 17:51:38 +1100
Subject: [Freeswitch-users] list all users
In-Reply-To: <20100221064520.GA26593@jdc.jasonjgw.net>
References: <3c5cf5261002201821q7b9f024bt7ab32b10946a5370@mail.gmail.com>
	<33c87fa31002201908x55e43e62g32b340872df9863d@mail.gmail.com>
	<3c5cf5261002202000h12b9d6b2ybefd5422fd4f9ee5@mail.gmail.com>
	<20100221041559.GA21387@jdc.jasonjgw.net>
	<3c5cf5261002202030l74935bbr1013547a315aaf6a@mail.gmail.com>
	<20100221054030.GA25531@jdc.jasonjgw.net>
	<3c5cf5261002202156u2f2e455jbf1e864c738b77a3@mail.gmail.com>
	<20100221064520.GA26593@jdc.jasonjgw.net>
Message-ID: <20100221065138.GA26756@jdc.jasonjgw.net>

Jason White <jason at jasonjgw.net> wrote:
 
> In the extension for the user in your dial-plan, you could always write:
> <action application="bridge"
> data="$sofia_contact(brian at microcomaustralia.com.au)"/>
> or you could use a variable instead of writing the domain name explicitly, of
> course, which would be better practice.

Sorry - there should have been braces in the above:
${sofia_contact(user at domain)}
it's the same function you called from the console in response to my earlier
post, but this time executed from the dial-plan to perform the lookup.



From brian at microcomaustralia.com.au  Sun Feb 21 00:30:39 2010
From: brian at microcomaustralia.com.au (Brian May)
Date: Sun, 21 Feb 2010 19:30:39 +1100
Subject: [Freeswitch-users] outgoing dialplans for australia
Message-ID: <3c5cf5261002210030n12bd743fh69f0c862eae0db8@mail.gmail.com>

Hello again!

Ok, after spending several hours trying to debug why incoming calls
didn't work, for two separate issues (2nd time I accidentally set "Do
not disturb" on my VOIP client. Duh!), I am feeling slightly lazy...

Anyone got a set of rules for outgoing calls (Australia) that they are
willing to share?

Even if I don't use it directly, it may help me understand this new
XML based syntax (I am only really familiar with the Asterisk dialplan
syntax).

As a specific example of something I am unsure of, is it possible to
have it try and dial using a sip provider, and if that fails try the
zap port? I don't want it to fall back to the zap port though if the
person doesn't answer or is engaged, only if there is a problem with
the SIP connection (e.g. Internet connection down).

Thanks.
-- 
Brian May <brian at microcomaustralia.com.au>


From brian at microcomaustralia.com.au  Sun Feb 21 00:50:48 2010
From: brian at microcomaustralia.com.au (Brian May)
Date: Sun, 21 Feb 2010 19:50:48 +1100
Subject: [Freeswitch-users] groups
Message-ID: <3c5cf5261002210050s235142d6o3f8be42146399c57@mail.gmail.com>

Hello,

Can I add a analogue FXS port to a group that is then accessed using
group_call(...)?

If so, how? I can only see support for SIP members, as defined in
directory/default.xml

Thanks
-- 
Brian May <brian at microcomaustralia.com.au>


From jason at jasonjgw.net  Sun Feb 21 01:10:11 2010
From: jason at jasonjgw.net (Jason White)
Date: Sun, 21 Feb 2010 20:10:11 +1100
Subject: [Freeswitch-users] outgoing dialplans for australia
In-Reply-To: <3c5cf5261002210030n12bd743fh69f0c862eae0db8@mail.gmail.com>
References: <3c5cf5261002210030n12bd743fh69f0c862eae0db8@mail.gmail.com>
Message-ID: <20100221091011.GA28429@jdc.jasonjgw.net>

Brian May <brian at microcomaustralia.com.au> wrote:
 
> As a specific example of something I am unsure of, is it possible to
> have it try and dial using a sip provider, and if that fails try the
> zap port? I don't want it to fall back to the zap port though if the
> person doesn't answer or is engaged, only if there is a problem with
> the SIP connection (e.g. Internet connection down).

http://wiki.freeswitch.org/wiki/Extension_Status_Example
is similar to what you want, except that instead of invoking a Javascript
application if the call fails (as in the example), you'll need to test the
value of originate_disposition to decide how to handle the call if the bridge
is unsuccessful and it falls through to the next action in the dial-plan.

(I'm assuming based on that page that originate_disposition is the correct
variable in this kind of scenario.)

Someone (perhaps you after you've implemented and debugged it) should document
the result on the wiki, as I'm sure this is a common use case, but there is no
documentation currently other than the above-mentioned page.




From Russell.Mosemann at cune.org  Sun Feb 21 03:55:28 2010
From: Russell.Mosemann at cune.org (Russell Mosemann)
Date: Sun, 21 Feb 2010 05:55:28 -0600
Subject: [Freeswitch-users] openzap TDM400 card
In-Reply-To: <3c5cf5261002201408v63c150d3v419493ab5e4e2089@mail.gmail.com>
References: <3c5cf5261002192157t25de5798xc10ab2a7755e67d0@mail.gmail.com><3c5cf5261002200148k1bb3a1f7mf97fabccfbd1c178@mail.gmail.com><C8B78F95DB344ADDA010C52AE0AC8B66@cune.pri>
	<3c5cf5261002201408v63c150d3v419493ab5e4e2089@mail.gmail.com>
Message-ID: <5C49F61510CA4156A574A1CCFA743B94@cune.pri>

Brian May wrote:

> Ports 1 & 2 are FXS, but use FXO signalling. So, yes I do plug a phone
> into these.
> Ports 3 & 4 are FXO, but use FXS signalling. So I plug the phone line
> into these.

Maybe a better way to think about it is that Zaptel is looking at the remote end, because it is interested in what signals might be received. FS is looking at the local end, because that is what FS has to manage. If Zaptel is expecting the remote end to be FXO (office phone), then FS is managing an FXS (subscriber signaling).

--
Russell Mosemann



From technical at ttnc.co.uk  Sun Feb 21 05:15:53 2010
From: technical at ttnc.co.uk (TTNC - Technical)
Date: Sun, 21 Feb 2010 13:15:53 +0000
Subject: [Freeswitch-users] mod_fax undefined symbol
In-Reply-To: <E6568E57-A3E2-4650-A8CD-70B9C2846AF4@jerris.com>
References: <6FCE44FF-70C1-412D-B722-4C8D359381BA@ttnc.co.uk>
	<201002191825.50957.errotan@gmail.com>
	<AB3AD9C8-C162-4DB5-8095-97CA83EDDE57@ttnc.co.uk>
	<201002191904.39081.errotan@gmail.com>
	<E6568E57-A3E2-4650-A8CD-70B9C2846AF4@jerris.com>
Message-ID: <EFB211AD-9B65-4CCE-9C5E-BC0A9ED55B7D@ttnc.co.uk>

Hi Guys

Without meaning to hassle (I know I am), has there been any progress with this bug yet? Anything else I can do to assist?

Please let me know.

Thanks

Russ

On 20 Feb 2010, at 02:00, Michael Jerris wrote:

> replying with more details on jira.
> 
> 
> On Feb 19, 2010, at 1:04 PM, Pusk?s Zsolt wrote:
> 
>> 2010. febru?r 19. 18.44.32 TTNC - Technical d?tummal ezt ?rta:
>>> On 19 Feb 2010, at 17:25, Pusk?s Zsolt wrote:
>>>> Just compiled svn16700 on Debian "lenny" x86, mod_fax loads and works
>>>> perfectly. I have an ongoing compile on another machine (amd64) if It
>>>> don't works i will send a mail (in 1 hour) otherwise consider it working.
>>> 
>>> How did you compile it? Using dpkg-buildpackage or via make/make install?
>>> 
>>> Do you have any debian versions of libtiff4(-dev) installed?
>>> 
>> 
>> Okay i can confirm that mod_fax works on Debian "lenny" amd64 but it don't work 
>> on Debian "testing,squeeze" amd64.
>> 
>> 2010-02-19 18:49:14.610297 [CRIT] switch_loadable_module.c:882 Error Loading 
>> module /usr/local/freeswitch/mod/mod_fax.so
>> **/usr/local/freeswitch/mod/mod_fax.so: undefined symbol: 
>> TIFFDefaultStripSize**
>> 
>> I haven't tried to compile mod_fax on testing before so i don't know what is 
>> causeing the problem :(
>> 
>> # ldd mod_fax.so
>>       linux-vdso.so.1 =>  (0x00007fff106f6000)
>>       libm.so.6 => /lib/libm.so.6 (0x00007f506b711000)
>>       libfreeswitch.so.1 => /usr/local/freeswitch/lib/libfreeswitch.so.1 
>> (0x00007f506b345000)
>>       libpthread.so.0 => /lib/libpthread.so.0 (0x00007f506b128000)
>>       libc.so.6 => /lib/libc.so.6 (0x00007f506add4000)
>>       libssl.so.0.9.8 => /usr/lib/libssl.so.0.9.8 (0x00007f506ab82000)
>>       libcrypto.so.0.9.8 => /usr/lib/libcrypto.so.0.9.8 (0x00007f506a7e2000)
>>       libncurses.so.5 => /lib/libncurses.so.5 (0x00007f506a59d000)
>>       libstdc++.so.6 => /usr/lib/libstdc++.so.6 (0x00007f506a28d000)
>>       libgcc_s.so.1 => /lib/libgcc_s.so.1 (0x00007f506a076000)
>>       libodbc.so.1 => /usr/lib/libodbc.so.1 (0x00007f5069e17000)
>>       /lib64/ld-linux-x86-64.so.2 (0x00007f506bc31000)
>>       libdl.so.2 => /lib/libdl.so.2 (0x00007f5069c13000)
>>       libz.so.1 => /usr/lib/libz.so.1 (0x00007f50699fb000)
>>       libltdl.so.7 => /usr/lib/libltdl.so.7 (0x00007f50697f2000)
>> 
>> Recently in debian "testing" libtiff4 and libjpeg is upgraded:
>> libtiff 3.9.2-3+b1
>> libjpeg62 6b-16.1
>> libjeg8 8-2.1
>> 
>> Q&A:
>> Q: How did you compile it? Using dpkg-buildpackage or via make/make install?
>> A: svn-clean ./bootsrap ./configure make etc.
>> 
>> Q: Do you have any debian versions of libtiff4(-dev) installed?
>> A: Yes:3.8.2-11.2
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From steveu at coppice.org  Sun Feb 21 05:47:17 2010
From: steveu at coppice.org (Steve Underwood)
Date: Sun, 21 Feb 2010 21:47:17 +0800
Subject: [Freeswitch-users] mod_fax undefined symbol
In-Reply-To: <EFB211AD-9B65-4CCE-9C5E-BC0A9ED55B7D@ttnc.co.uk>
References: <6FCE44FF-70C1-412D-B722-4C8D359381BA@ttnc.co.uk>	<201002191825.50957.errotan@gmail.com>	<AB3AD9C8-C162-4DB5-8095-97CA83EDDE57@ttnc.co.uk>	<201002191904.39081.errotan@gmail.com>	<E6568E57-A3E2-4650-A8CD-70B9C2846AF4@jerris.com>
	<EFB211AD-9B65-4CCE-9C5E-BC0A9ED55B7D@ttnc.co.uk>
Message-ID: <4B813965.1030704@coppice.org>

Hi Russ,

The only place in FS where TIFFDefaulyStripSize is used is in the file 
t4_rx.c, and you probably won't actually be calling it. Try commenting 
out that line, and see if there are any other stumbling blocks. Often 
there is a mass of errors, and the system just tells you about them one 
by one.

Steve

On 02/21/2010 09:15 PM, TTNC - Technical wrote:
> Hi Guys
>
> Without meaning to hassle (I know I am), has there been any progress with this bug yet? Anything else I can do to assist?
>
> Please let me know.
>
> Thanks
>
> Russ
>
> On 20 Feb 2010, at 02:00, Michael Jerris wrote:
>
>    
>> replying with more details on jira.
>>
>>
>> On Feb 19, 2010, at 1:04 PM, Pusk?s Zsolt wrote:
>>
>>      
>>> 2010. febru?r 19. 18.44.32 TTNC - Technical d?tummal ezt ?rta:
>>>        
>>>> On 19 Feb 2010, at 17:25, Pusk?s Zsolt wrote:
>>>>          
>>>>> Just compiled svn16700 on Debian "lenny" x86, mod_fax loads and works
>>>>> perfectly. I have an ongoing compile on another machine (amd64) if It
>>>>> don't works i will send a mail (in 1 hour) otherwise consider it working.
>>>>>            
>>>> How did you compile it? Using dpkg-buildpackage or via make/make install?
>>>>
>>>> Do you have any debian versions of libtiff4(-dev) installed?
>>>>
>>>>          
>>> Okay i can confirm that mod_fax works on Debian "lenny" amd64 but it don't work
>>> on Debian "testing,squeeze" amd64.
>>>
>>> 2010-02-19 18:49:14.610297 [CRIT] switch_loadable_module.c:882 Error Loading
>>> module /usr/local/freeswitch/mod/mod_fax.so
>>> **/usr/local/freeswitch/mod/mod_fax.so: undefined symbol:
>>> TIFFDefaultStripSize**
>>>
>>> I haven't tried to compile mod_fax on testing before so i don't know what is
>>> causeing the problem :(
>>>
>>> # ldd mod_fax.so
>>>        linux-vdso.so.1 =>   (0x00007fff106f6000)
>>>        libm.so.6 =>  /lib/libm.so.6 (0x00007f506b711000)
>>>        libfreeswitch.so.1 =>  /usr/local/freeswitch/lib/libfreeswitch.so.1
>>> (0x00007f506b345000)
>>>        libpthread.so.0 =>  /lib/libpthread.so.0 (0x00007f506b128000)
>>>        libc.so.6 =>  /lib/libc.so.6 (0x00007f506add4000)
>>>        libssl.so.0.9.8 =>  /usr/lib/libssl.so.0.9.8 (0x00007f506ab82000)
>>>        libcrypto.so.0.9.8 =>  /usr/lib/libcrypto.so.0.9.8 (0x00007f506a7e2000)
>>>        libncurses.so.5 =>  /lib/libncurses.so.5 (0x00007f506a59d000)
>>>        libstdc++.so.6 =>  /usr/lib/libstdc++.so.6 (0x00007f506a28d000)
>>>        libgcc_s.so.1 =>  /lib/libgcc_s.so.1 (0x00007f506a076000)
>>>        libodbc.so.1 =>  /usr/lib/libodbc.so.1 (0x00007f5069e17000)
>>>        /lib64/ld-linux-x86-64.so.2 (0x00007f506bc31000)
>>>        libdl.so.2 =>  /lib/libdl.so.2 (0x00007f5069c13000)
>>>        libz.so.1 =>  /usr/lib/libz.so.1 (0x00007f50699fb000)
>>>        libltdl.so.7 =>  /usr/lib/libltdl.so.7 (0x00007f50697f2000)
>>>
>>> Recently in debian "testing" libtiff4 and libjpeg is upgraded:
>>> libtiff 3.9.2-3+b1
>>> libjpeg62 6b-16.1
>>> libjeg8 8-2.1
>>>
>>> Q&A:
>>> Q: How did you compile it? Using dpkg-buildpackage or via make/make install?
>>> A: svn-clean ./bootsrap ./configure make etc.
>>>
>>> Q: Do you have any debian versions of libtiff4(-dev) installed?
>>> A: Yes:3.8.2-11.2
>>>        
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>      
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>    



From technical at ttnc.co.uk  Sun Feb 21 06:12:28 2010
From: technical at ttnc.co.uk (TTNC - Technical)
Date: Sun, 21 Feb 2010 14:12:28 +0000
Subject: [Freeswitch-users] mod_fax undefined symbol
In-Reply-To: <4B813965.1030704@coppice.org>
References: <6FCE44FF-70C1-412D-B722-4C8D359381BA@ttnc.co.uk>	<201002191825.50957.errotan@gmail.com>	<AB3AD9C8-C162-4DB5-8095-97CA83EDDE57@ttnc.co.uk>	<201002191904.39081.errotan@gmail.com>	<E6568E57-A3E2-4650-A8CD-70B9C2846AF4@jerris.com>
	<EFB211AD-9B65-4CCE-9C5E-BC0A9ED55B7D@ttnc.co.uk>
	<4B813965.1030704@coppice.org>
Message-ID: <1656F554-374C-40A7-8E67-A311CB9BEC19@ttnc.co.uk>

Hi Steve

Just tried that, changed:

TIFFSetField(t->tiff_file,
                     TIFFTAG_ROWSPERSTRIP,
                     TIFFDefaultStripSize(t->tiff_file, 0));
to:

TIFFSetField(t->tiff_file,
                     TIFFTAG_ROWSPERSTRIP,
                     0);

Compiled OK, but then got the following error when trying to 'load mod_fax':

2010-02-21 14:08:30.242671 [CRIT] switch_loadable_module.c:882 Error Loading module /opt/freeswitch/mod/mod_fax.so
**/opt/freeswitch/mod/mod_fax.so: undefined symbol: TIFFSetDirectory**

It seems TIFFSetDirectory appears in quite a few places throughout t4_rx.c along with t4.c - so I doubt going through and commenting them out will really work?

Hopefully this maybe of some use to anyone looking into the problem though?

Anything else I can do to help then please let me know.

Russ


On 21 Feb 2010, at 13:47, Steve Underwood wrote:

> Hi Russ,
> 
> The only place in FS where TIFFDefaulyStripSize is used is in the file 
> t4_rx.c, and you probably won't actually be calling it. Try commenting 
> out that line, and see if there are any other stumbling blocks. Often 
> there is a mass of errors, and the system just tells you about them one 
> by one.
> 
> Steve
> 
> On 02/21/2010 09:15 PM, TTNC - Technical wrote:
>> Hi Guys
>> 
>> Without meaning to hassle (I know I am), has there been any progress with this bug yet? Anything else I can do to assist?
>> 
>> Please let me know.
>> 
>> Thanks
>> 
>> Russ
>> 
>> On 20 Feb 2010, at 02:00, Michael Jerris wrote:
>> 
>> 
>>> replying with more details on jira.
>>> 
>>> 
>>> On Feb 19, 2010, at 1:04 PM, Pusk?s Zsolt wrote:
>>> 
>>> 
>>>> 2010. febru?r 19. 18.44.32 TTNC - Technical d?tummal ezt ?rta:
>>>> 
>>>>> On 19 Feb 2010, at 17:25, Pusk?s Zsolt wrote:
>>>>> 
>>>>>> Just compiled svn16700 on Debian "lenny" x86, mod_fax loads and works
>>>>>> perfectly. I have an ongoing compile on another machine (amd64) if It
>>>>>> don't works i will send a mail (in 1 hour) otherwise consider it working.
>>>>>> 
>>>>> How did you compile it? Using dpkg-buildpackage or via make/make install?
>>>>> 
>>>>> Do you have any debian versions of libtiff4(-dev) installed?
>>>>> 
>>>>> 
>>>> Okay i can confirm that mod_fax works on Debian "lenny" amd64 but it don't work
>>>> on Debian "testing,squeeze" amd64.
>>>> 
>>>> 2010-02-19 18:49:14.610297 [CRIT] switch_loadable_module.c:882 Error Loading
>>>> module /usr/local/freeswitch/mod/mod_fax.so
>>>> **/usr/local/freeswitch/mod/mod_fax.so: undefined symbol:
>>>> TIFFDefaultStripSize**
>>>> 
>>>> I haven't tried to compile mod_fax on testing before so i don't know what is
>>>> causeing the problem :(
>>>> 
>>>> # ldd mod_fax.so
>>>>       linux-vdso.so.1 =>   (0x00007fff106f6000)
>>>>       libm.so.6 =>  /lib/libm.so.6 (0x00007f506b711000)
>>>>       libfreeswitch.so.1 =>  /usr/local/freeswitch/lib/libfreeswitch.so.1
>>>> (0x00007f506b345000)
>>>>       libpthread.so.0 =>  /lib/libpthread.so.0 (0x00007f506b128000)
>>>>       libc.so.6 =>  /lib/libc.so.6 (0x00007f506add4000)
>>>>       libssl.so.0.9.8 =>  /usr/lib/libssl.so.0.9.8 (0x00007f506ab82000)
>>>>       libcrypto.so.0.9.8 =>  /usr/lib/libcrypto.so.0.9.8 (0x00007f506a7e2000)
>>>>       libncurses.so.5 =>  /lib/libncurses.so.5 (0x00007f506a59d000)
>>>>       libstdc++.so.6 =>  /usr/lib/libstdc++.so.6 (0x00007f506a28d000)
>>>>       libgcc_s.so.1 =>  /lib/libgcc_s.so.1 (0x00007f506a076000)
>>>>       libodbc.so.1 =>  /usr/lib/libodbc.so.1 (0x00007f5069e17000)
>>>>       /lib64/ld-linux-x86-64.so.2 (0x00007f506bc31000)
>>>>       libdl.so.2 =>  /lib/libdl.so.2 (0x00007f5069c13000)
>>>>       libz.so.1 =>  /usr/lib/libz.so.1 (0x00007f50699fb000)
>>>>       libltdl.so.7 =>  /usr/lib/libltdl.so.7 (0x00007f50697f2000)
>>>> 
>>>> Recently in debian "testing" libtiff4 and libjpeg is upgraded:
>>>> libtiff 3.9.2-3+b1
>>>> libjpeg62 6b-16.1
>>>> libjeg8 8-2.1
>>>> 
>>>> Q&A:
>>>> Q: How did you compile it? Using dpkg-buildpackage or via make/make install?
>>>> A: svn-clean ./bootsrap ./configure make etc.
>>>> 
>>>> Q: Do you have any debian versions of libtiff4(-dev) installed?
>>>> A: Yes:3.8.2-11.2
>>>> 
>>> 
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From steveu at coppice.org  Sun Feb 21 06:28:48 2010
From: steveu at coppice.org (Steve Underwood)
Date: Sun, 21 Feb 2010 22:28:48 +0800
Subject: [Freeswitch-users] mod_fax undefined symbol
In-Reply-To: <1656F554-374C-40A7-8E67-A311CB9BEC19@ttnc.co.uk>
References: <6FCE44FF-70C1-412D-B722-4C8D359381BA@ttnc.co.uk>	<201002191825.50957.errotan@gmail.com>	<AB3AD9C8-C162-4DB5-8095-97CA83EDDE57@ttnc.co.uk>	<201002191904.39081.errotan@gmail.com>	<E6568E57-A3E2-4650-A8CD-70B9C2846AF4@jerris.com>	<EFB211AD-9B65-4CCE-9C5E-BC0A9ED55B7D@ttnc.co.uk>	<4B813965.1030704@coppice.org>
	<1656F554-374C-40A7-8E67-A311CB9BEC19@ttnc.co.uk>
Message-ID: <4B814320.7040306@coppice.org>

Hi Russ,

It sounds like this is nothing to do with the libtiff version, but that 
you just don't have libtiff there at all. Having no familiarity with 
Debian, I'll leave further analysis to someone else.

Regards,
Steve

On 02/21/2010 10:12 PM, TTNC - Technical wrote:
> Hi Steve
>
> Just tried that, changed:
>
> TIFFSetField(t->tiff_file,
>                       TIFFTAG_ROWSPERSTRIP,
>                       TIFFDefaultStripSize(t->tiff_file, 0));
> to:
>
> TIFFSetField(t->tiff_file,
>                       TIFFTAG_ROWSPERSTRIP,
>                       0);
>
> Compiled OK, but then got the following error when trying to 'load mod_fax':
>
> 2010-02-21 14:08:30.242671 [CRIT] switch_loadable_module.c:882 Error Loading module /opt/freeswitch/mod/mod_fax.so
> **/opt/freeswitch/mod/mod_fax.so: undefined symbol: TIFFSetDirectory**
>
> It seems TIFFSetDirectory appears in quite a few places throughout t4_rx.c along with t4.c - so I doubt going through and commenting them out will really work?
>
> Hopefully this maybe of some use to anyone looking into the problem though?
>
> Anything else I can do to help then please let me know.
>
> Russ
>
>
> On 21 Feb 2010, at 13:47, Steve Underwood wrote:
>
>    
>> Hi Russ,
>>
>> The only place in FS where TIFFDefaulyStripSize is used is in the file
>> t4_rx.c, and you probably won't actually be calling it. Try commenting
>> out that line, and see if there are any other stumbling blocks. Often
>> there is a mass of errors, and the system just tells you about them one
>> by one.
>>
>> Steve
>>
>> On 02/21/2010 09:15 PM, TTNC - Technical wrote:
>>      
>>> Hi Guys
>>>
>>> Without meaning to hassle (I know I am), has there been any progress with this bug yet? Anything else I can do to assist?
>>>
>>> Please let me know.
>>>
>>> Thanks
>>>
>>> Russ
>>>
>>> On 20 Feb 2010, at 02:00, Michael Jerris wrote:
>>>
>>>
>>>        
>>>> replying with more details on jira.
>>>>
>>>>
>>>> On Feb 19, 2010, at 1:04 PM, Pusk?s Zsolt wrote:
>>>>
>>>>
>>>>          
>>>>> 2010. febru?r 19. 18.44.32 TTNC - Technical d?tummal ezt ?rta:
>>>>>
>>>>>            
>>>>>> On 19 Feb 2010, at 17:25, Pusk?s Zsolt wrote:
>>>>>>
>>>>>>              
>>>>>>> Just compiled svn16700 on Debian "lenny" x86, mod_fax loads and works
>>>>>>> perfectly. I have an ongoing compile on another machine (amd64) if It
>>>>>>> don't works i will send a mail (in 1 hour) otherwise consider it working.
>>>>>>>
>>>>>>>                
>>>>>> How did you compile it? Using dpkg-buildpackage or via make/make install?
>>>>>>
>>>>>> Do you have any debian versions of libtiff4(-dev) installed?
>>>>>>
>>>>>>
>>>>>>              
>>>>> Okay i can confirm that mod_fax works on Debian "lenny" amd64 but it don't work
>>>>> on Debian "testing,squeeze" amd64.
>>>>>
>>>>> 2010-02-19 18:49:14.610297 [CRIT] switch_loadable_module.c:882 Error Loading
>>>>> module /usr/local/freeswitch/mod/mod_fax.so
>>>>> **/usr/local/freeswitch/mod/mod_fax.so: undefined symbol:
>>>>> TIFFDefaultStripSize**
>>>>>
>>>>> I haven't tried to compile mod_fax on testing before so i don't know what is
>>>>> causeing the problem :(
>>>>>
>>>>> # ldd mod_fax.so
>>>>>        linux-vdso.so.1 =>    (0x00007fff106f6000)
>>>>>        libm.so.6 =>   /lib/libm.so.6 (0x00007f506b711000)
>>>>>        libfreeswitch.so.1 =>   /usr/local/freeswitch/lib/libfreeswitch.so.1
>>>>> (0x00007f506b345000)
>>>>>        libpthread.so.0 =>   /lib/libpthread.so.0 (0x00007f506b128000)
>>>>>        libc.so.6 =>   /lib/libc.so.6 (0x00007f506add4000)
>>>>>        libssl.so.0.9.8 =>   /usr/lib/libssl.so.0.9.8 (0x00007f506ab82000)
>>>>>        libcrypto.so.0.9.8 =>   /usr/lib/libcrypto.so.0.9.8 (0x00007f506a7e2000)
>>>>>        libncurses.so.5 =>   /lib/libncurses.so.5 (0x00007f506a59d000)
>>>>>        libstdc++.so.6 =>   /usr/lib/libstdc++.so.6 (0x00007f506a28d000)
>>>>>        libgcc_s.so.1 =>   /lib/libgcc_s.so.1 (0x00007f506a076000)
>>>>>        libodbc.so.1 =>   /usr/lib/libodbc.so.1 (0x00007f5069e17000)
>>>>>        /lib64/ld-linux-x86-64.so.2 (0x00007f506bc31000)
>>>>>        libdl.so.2 =>   /lib/libdl.so.2 (0x00007f5069c13000)
>>>>>        libz.so.1 =>   /usr/lib/libz.so.1 (0x00007f50699fb000)
>>>>>        libltdl.so.7 =>   /usr/lib/libltdl.so.7 (0x00007f50697f2000)
>>>>>
>>>>> Recently in debian "testing" libtiff4 and libjpeg is upgraded:
>>>>> libtiff 3.9.2-3+b1
>>>>> libjpeg62 6b-16.1
>>>>> libjeg8 8-2.1
>>>>>
>>>>> Q&A:
>>>>> Q: How did you compile it? Using dpkg-buildpackage or via make/make install?
>>>>> A: svn-clean ./bootsrap ./configure make etc.
>>>>>
>>>>> Q: Do you have any debian versions of libtiff4(-dev) installed?
>>>>> A: Yes:3.8.2-11.2
>>>>>
>>>>>            



From moizchinoy at gmail.com  Sun Feb 21 08:00:59 2010
From: moizchinoy at gmail.com (Moiz Chinoy)
Date: Sun, 21 Feb 2010 20:00:59 +0400
Subject: [Freeswitch-users] Dingaling - external rtp supported?
In-Reply-To: <191c3a031002171136q7424e9bai34379abb542daf93@mail.gmail.com>
References: <29b888f81002160835y6e290319ub0d04c088caa1b79@mail.gmail.com>
	<DEA39AD1-0EE9-473A-B78C-CDDF3F100578@freeswitch.org>
	<29b888f81002170318p63622f3fhc8d26ad0e85602d2@mail.gmail.com>
	<191c3a031002170538r3b029515i8a000812a70ad558@mail.gmail.com>
	<29b888f81002170741y3d13cb57ycb4436a372c4eded@mail.gmail.com>
	<191c3a031002171136q7424e9bai34379abb542daf93@mail.gmail.com>
Message-ID: <29b888f81002210800v617bca9dk4bf1abf1bd791ee0@mail.gmail.com>

Guys,

To make things simple gtalk client is entirely on different network.

Call comes from outside through external Sip profile.

If gtalk answers the call after 3-4 rings both parties can hear each other.
If gtalk answers the call after 2 rings both parties no one can hear each other.
If gtalk answers the call immediately FS crashes.

Attached is the screen shot of the error...

Here is the FS log...
--------------------------------
http://pastebin.freeswitch.org/12197

External Sip Profile has following lines:
---------------------------------------------------------
    <param name="rtp-ip" value="$${local_ip_v4}"/>
    <param name="sip-ip" value="$${local_ip_v4}"/>
    <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
    <param name="ext-sip-ip" value="$${external_sip_ip}"/>

Jingle Client.xml has following lines:
-----------------------------------------------------
    <param name="rtp-ip" value="$${bind_server_ip}"/>
    <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
    <param name="disable-rtp-auto-adjust" value="true"/>
    <!--<param name="avatar" value="/path/to/tiny.jpg"/>-->
    <!--<param name="candidate-acl" value="wan.auto"/>-->
    <!--<param name="local-network-acl" value="localnet.auto"/>-->

Vars.xml has following lines:
-------------------------------------------
<X-PRE-PROCESS cmd="set" data="external_rtp_ip=stun:stun.freeswitch.org"/>
<X-PRE-PROCESS cmd="set" data="external_sip_ip=stun:stun.freeswitch.org"/>


Please advise me how can I provide more of the required data.

On Wed, Feb 17, 2010 at 11:36 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> you cant combine stun and gtalk and boxes in the same lan very easily if you
> do need to do that you will need to mess with
>
> <param name="candidate-acl" value="wan.auto"/>
> <param name="local-network-acl" value="localnet.auto"/>
>
>
>
>
>
> On Wed, Feb 17, 2010 at 9:41 AM, Moiz Chinoy <moizchinoy at gmail.com> wrote:
>>
>> Guys I am unable to produce the crash but now both parties cannot hear
>> each other!
>>
>> Vars.xml has following lines:
>> ?<X-PRE-PROCESS cmd="set"
>> data="external_rtp_ip=stun:stun.freeswitch.org"/>
>> ?<X-PRE-PROCESS cmd="set"
>> data="external_sip_ip=stun:stun.freeswitch.org"/>
>>
>> Jingle Client.xml has following lines:
>> ? ?<param name="rtp-ip" value="$${bind_server_ip}"/>
>> ? ?<!-- <param name="ext-rtp-ip" value="auto-nat"/> -->
>> ? ?<param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
>> ? ?<param name="disable-rtp-auto-adjust" value="true"/>
>> ? ?<param name="local-network-acl" value="localnet.auto"/>
>>
>> On Wed, Feb 17, 2010 at 5:38 PM, Anthony Minessale
>> <anthony.minessale at gmail.com> wrote:
>> > Obtain a stack trace from the crash.
>> >
>> > On Feb 17, 2010 5:26 AM, "Moiz Chinoy" <moizchinoy at gmail.com> wrote:
>> >
>> > Hi,
>> >
>> > FS rev: 16673
>> > Platform: Windows
>> >
>> > More details:
>> >
>> > FS is behind NAT and machine is running a VPN connection.
>> >
>> > FS and GTalk client on the same machine:
>> >
>> > --------------------------------------------------------------------------------------------------
>> > jingle profile client.xml has following line:
>> > <param name="rtp-ip" value="$${bind_server_ip}"/>
>> >
>> > External SIP call is successfully bridged to GTalk client.
>> >
>> >
>> > FS and GTalk client on the different machine:
>> >
>> > --------------------------------------------------------------------------------------------------
>> > jingle profile client.xml has following lines:
>> > <param name="rtp-ip" value="$${bind_server_ip}"/>
>> >
>> > <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
>> >
>> > As soon as external SIP call land and I try to bridge the call to
>> > GTalk client, FS crashes.
>> >
>> >
>> > NAT Details:
>> > ---------------------------
>> > I think my NAT does not support UpNP or PMP. The reason I say it
>> > because when FS starts following message is displayed:
>> >
>> > 2010-02-17 09:15:42.809674 [INFO] switch_nat.c:409 Scanning for NAT
>> > 2010-02-17 09:15:42.809674 [ERR] switch_nat.c:197 Error checking for
>> > PMP [init failed]
>> > 2010-02-17 09:15:42.809674 [DEBUG] switch_nat.c:414 Checking for UPnP
>> > 2010-02-17 09:15:56.825388 [DEBUG] switch_nat.c:114 No
>> > InternetGatewayDevice, using first entry as default
>> > (http://192.168.16.17:50144/).
>> > 2010-02-17 09:15:56.887889 [INFO] switch_nat.c:429 No PMP or UPnP NAT
>> > devices detected!
>> >
>> >
>> >
>> > On Tue, Feb 16, 2010 at 8:41 PM, Brian West <brian at freeswitch.org>
>> > wrote:
>> >> can you please update...
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>>
>>
>> --
>> Regards,
>> Moiz Chinoy.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Regards,
Moiz Chinoy.
-------------- next part --------------
A non-text attachment was scrubbed...
Name: mutex_error.JPG
Type: image/jpeg
Size: 33069 bytes
Desc: not available
Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100221/d942434c/attachment-0001.jpe 

From infos at madovsky.org  Sun Feb 21 09:43:34 2010
From: infos at madovsky.org (Madovsky)
Date: Sun, 21 Feb 2010 12:43:34 -0500
Subject: [Freeswitch-users] codec negociations
Message-ID: <E16A52EDBA634BDB98A00F938612E526@MOBILEE1705>

Hi,

is it possible for FS to take the callee codec from an interant call to external as reference ?
example:
userA (with only GSM) from FS calls whoeverUser at whateverdomain, this whoeverUser has only PCMU codec,
so FS has a list of GSM,PCMU,PCMA and so starts to transcode from PCMU to GSM.

I succeed to do the contrary (it transcodes from external to internal phone)

Thanks

Franck
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100221/04a36de7/attachment.html 

From oseslija at gmail.com  Sun Feb 21 10:03:05 2010
From: oseslija at gmail.com (Ognjen Seslija)
Date: Sun, 21 Feb 2010 19:03:05 +0100
Subject: [Freeswitch-users] ASTPP For FreeSWITCH
In-Reply-To: <4B799839.2090008@aleph-com.net>
References: <87f2f3b91002150959q6accece5od67b262fb38ec41e@mail.gmail.com>
	<4B799839.2090008@aleph-com.net>
Message-ID: <4468a6771002211003o41fdb86en72b9c7a669a17ab1@mail.gmail.com>

I installed it. Don't see the way to switch from current asterisk menus to
FS ones.

Is there a irc channel to ask questions?

Regards,
Ognjen

On Mon, Feb 15, 2010 at 7:53 PM, Darren Wiebe <darren at aleph-com.net> wrote:

>  I will comment.  We've been using ASTPP for rating freeswitch cdrs for
> some time already.  It provides lcr from a database as well as sip user
> management.  It uses the mod_xml_curl and mod_xml_cdr modules for routing as
> well as realtime rating.  It also has an application that can listen to
> freeswitch and rate calls in realtime that way.  I patched a couple of bugs
> earlier this morning and I would not say that it's bug free but it's
> certainly in testing.
>
> Darren Wiebe
> darren at aleph-com.net
>
>
>
> On 02/15/2010 10:59 AM, Michael Collins wrote:
>
> Hey all,
>
> Here's a quick story <http://www.freeswitch.org/node/239> about ASTPP and
> FreeSWITCH. If you are using ASTPP with FreeSWITCH please let me know how it
> works. I didn't see any information on our wiki about ASTPP. If ASTPP is
> viable then we should document it as best we can.
>
> Thanks!
> -Michael
>
>
> _______________________________________________
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100221/ecadfb4b/attachment.html 

From darren at aleph-com.net  Sun Feb 21 11:37:44 2010
From: darren at aleph-com.net (Darren Wiebe)
Date: Sun, 21 Feb 2010 12:37:44 -0700
Subject: [Freeswitch-users] ASTPP For FreeSWITCH
In-Reply-To: <4468a6771002211003o41fdb86en72b9c7a669a17ab1@mail.gmail.com>
References: <87f2f3b91002150959q6accece5od67b262fb38ec41e@mail.gmail.com>	<4B799839.2090008@aleph-com.net>
	<4468a6771002211003o41fdb86en72b9c7a669a17ab1@mail.gmail.com>
Message-ID: <4B818B88.3030806@aleph-com.net>

Look in System->Configuration.  Set "users_dids_rt" to 0 and 
"users_dids_freeswitch" to 1.  There's not currently an irc channel but 
that's a good idea.

Darren Wiebe
darren at aleph-com.net


On 21/02/2010 11:03 AM, Ognjen Seslija wrote:
> I installed it. Don't see the way to switch from current asterisk 
> menus to FS ones.
>
> Is there a irc channel to ask questions?
>
> Regards,
> Ognjen
>
> On Mon, Feb 15, 2010 at 7:53 PM, Darren Wiebe <darren at aleph-com.net 
> <mailto:darren at aleph-com.net>> wrote:
>
>     I will comment.  We've been using ASTPP for rating freeswitch cdrs
>     for some time already.  It provides lcr from a database as well as
>     sip user management.  It uses the mod_xml_curl and mod_xml_cdr
>     modules for routing as well as realtime rating.  It also has an
>     application that can listen to freeswitch and rate calls in
>     realtime that way.  I patched a couple of bugs earlier this
>     morning and I would not say that it's bug free but it's certainly
>     in testing.
>
>     Darren Wiebe
>     darren at aleph-com.net <mailto:darren at aleph-com.net>
>
>
>
>     On 02/15/2010 10:59 AM, Michael Collins wrote:
>>     Hey all,
>>
>>     Here's a quick story <http://www.freeswitch.org/node/239> about
>>     ASTPP and FreeSWITCH. If you are using ASTPP with FreeSWITCH
>>     please let me know how it works. I didn't see any information on
>>     our wiki about ASTPP. If ASTPP is viable then we should document
>>     it as best we can.
>>
>>     Thanks!
>>     -Michael
>>
>>
>>     _______________________________________________
>>     FreeSWITCH-users mailing list
>>     FreeSWITCH-users at lists.freeswitch.org  <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     http://www.freeswitch.org
>>        
>
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>    

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100221/6bd70949/attachment.html 

From oseslija at gmail.com  Sun Feb 21 12:04:31 2010
From: oseslija at gmail.com (Ognjen Seslija)
Date: Sun, 21 Feb 2010 21:04:31 +0100
Subject: [Freeswitch-users] ASTPP For FreeSWITCH
In-Reply-To: <4B818B88.3030806@aleph-com.net>
References: <87f2f3b91002150959q6accece5od67b262fb38ec41e@mail.gmail.com>
	<4B799839.2090008@aleph-com.net>
	<4468a6771002211003o41fdb86en72b9c7a669a17ab1@mail.gmail.com>
	<4B818B88.3030806@aleph-com.net>
Message-ID: <4468a6771002211204y1aeb09b0tbe7980da9b15da19@mail.gmail.com>

Thanks.

On Sun, Feb 21, 2010 at 8:37 PM, Darren Wiebe <darren at aleph-com.net> wrote:

>  Look in System->Configuration.  Set "users_dids_rt" to 0 and
> "users_dids_freeswitch" to 1.  There's not currently an irc channel but
> that's a good idea.
>
>
> Darren Wiebe
> darren at aleph-com.net
>
>
> On 21/02/2010 11:03 AM, Ognjen Seslija wrote:
>
> I installed it. Don't see the way to switch from current asterisk menus to
> FS ones.
>
> Is there a irc channel to ask questions?
>
> Regards,
> Ognjen
>
> On Mon, Feb 15, 2010 at 7:53 PM, Darren Wiebe <darren at aleph-com.net>wrote:
>
>>  I will comment.  We've been using ASTPP for rating freeswitch cdrs for
>> some time already.  It provides lcr from a database as well as sip user
>> management.  It uses the mod_xml_curl and mod_xml_cdr modules for routing as
>> well as realtime rating.  It also has an application that can listen to
>> freeswitch and rate calls in realtime that way.  I patched a couple of bugs
>> earlier this morning and I would not say that it's bug free but it's
>> certainly in testing.
>>
>> Darren Wiebe
>> darren at aleph-com.net
>>
>>
>>
>> On 02/15/2010 10:59 AM, Michael Collins wrote:
>>
>>  Hey all,
>>
>> Here's a quick story <http://www.freeswitch.org/node/239> about ASTPP and
>> FreeSWITCH. If you are using ASTPP with FreeSWITCH please let me know how it
>> works. I didn't see any information on our wiki about ASTPP. If ASTPP is
>> viable then we should document it as best we can.
>>
>> Thanks!
>> -Michael
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100221/0226f863/attachment.html 

From technical at ttnc.co.uk  Sun Feb 21 13:25:05 2010
From: technical at ttnc.co.uk (TTNC - Technical)
Date: Sun, 21 Feb 2010 21:25:05 +0000
Subject: [Freeswitch-users] mod_fax undefined symbol
In-Reply-To: <201002191904.39081.errotan@gmail.com>
References: <6FCE44FF-70C1-412D-B722-4C8D359381BA@ttnc.co.uk>
	<201002191825.50957.errotan@gmail.com>
	<AB3AD9C8-C162-4DB5-8095-97CA83EDDE57@ttnc.co.uk>
	<201002191904.39081.errotan@gmail.com>
Message-ID: <41F9D585-2116-4219-9AF7-E8E944D43362@ttnc.co.uk>

Out of interest, I downgraded my versions of libtiff and libjpeg to the versions shipped with Lenny:

voipin1:/opt# dpkg -l | egrep 'libtiff|libjpeg'
ii  libjpeg62                            6b-14                    The Independent JPEG Group's JPEG runtime library
ii  libjpeg62-dev                        6b-14                    Development files for the IJG JPEG library
ii  libtiff4                             3.8.2-11.2               Tag Image File Format (TIFF) library
ii  libtiff4-dev                         3.8.2-11.2               Tag Image File Format library (TIFF), development files
ii  libtiffxx0c2                         3.8.2-11.2               Tag Image File Format (TIFF) library -- C++ interface

Everything else stayed at the 'squeeze' version.

Still didn't make any different, **/opt/freeswitch/mod/mod_fax.so: undefined symbol: TIFFDefaultStripSize**

I'm guessing that points to it being a problem outside of these packages and somewhere else in Debian?

Russ


On 19 Feb 2010, at 18:04, Pusk?s Zsolt wrote:

> 2010. febru?r 19. 18.44.32 TTNC - Technical d?tummal ezt ?rta:
>> On 19 Feb 2010, at 17:25, Pusk?s Zsolt wrote:
>>> Just compiled svn16700 on Debian "lenny" x86, mod_fax loads and works
>>> perfectly. I have an ongoing compile on another machine (amd64) if It
>>> don't works i will send a mail (in 1 hour) otherwise consider it working.
>> 
>> How did you compile it? Using dpkg-buildpackage or via make/make install?
>> 
>> Do you have any debian versions of libtiff4(-dev) installed?
>> 
> 
> Okay i can confirm that mod_fax works on Debian "lenny" amd64 but it don't work 
> on Debian "testing,squeeze" amd64.
> 
> 2010-02-19 18:49:14.610297 [CRIT] switch_loadable_module.c:882 Error Loading 
> module /usr/local/freeswitch/mod/mod_fax.so
> **/usr/local/freeswitch/mod/mod_fax.so: undefined symbol: 
> TIFFDefaultStripSize**
> 
> I haven't tried to compile mod_fax on testing before so i don't know what is 
> causeing the problem :(
> 
> # ldd mod_fax.so
>        linux-vdso.so.1 =>  (0x00007fff106f6000)
>        libm.so.6 => /lib/libm.so.6 (0x00007f506b711000)
>        libfreeswitch.so.1 => /usr/local/freeswitch/lib/libfreeswitch.so.1 
> (0x00007f506b345000)
>        libpthread.so.0 => /lib/libpthread.so.0 (0x00007f506b128000)
>        libc.so.6 => /lib/libc.so.6 (0x00007f506add4000)
>        libssl.so.0.9.8 => /usr/lib/libssl.so.0.9.8 (0x00007f506ab82000)
>        libcrypto.so.0.9.8 => /usr/lib/libcrypto.so.0.9.8 (0x00007f506a7e2000)
>        libncurses.so.5 => /lib/libncurses.so.5 (0x00007f506a59d000)
>        libstdc++.so.6 => /usr/lib/libstdc++.so.6 (0x00007f506a28d000)
>        libgcc_s.so.1 => /lib/libgcc_s.so.1 (0x00007f506a076000)
>        libodbc.so.1 => /usr/lib/libodbc.so.1 (0x00007f5069e17000)
>        /lib64/ld-linux-x86-64.so.2 (0x00007f506bc31000)
>        libdl.so.2 => /lib/libdl.so.2 (0x00007f5069c13000)
>        libz.so.1 => /usr/lib/libz.so.1 (0x00007f50699fb000)
>        libltdl.so.7 => /usr/lib/libltdl.so.7 (0x00007f50697f2000)
> 
> Recently in debian "testing" libtiff4 and libjpeg is upgraded:
> libtiff 3.9.2-3+b1
> libjpeg62 6b-16.1
> libjeg8 8-2.1
> 
> Q&A:
> Q: How did you compile it? Using dpkg-buildpackage or via make/make install?
> A: svn-clean ./bootsrap ./configure make etc.
> 
> Q: Do you have any debian versions of libtiff4(-dev) installed?
> A: Yes:3.8.2-11.2
> 
> I open a jira for this.
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From matt at webcontracts.co.uk  Sun Feb 21 15:26:08 2010
From: matt at webcontracts.co.uk (Matthew Law)
Date: Sun, 21 Feb 2010 23:26:08 -0000
Subject: [Freeswitch-users] Dialplan question
Message-ID: <2dac3814a041c79208af8a433b279566.squirrel@www.webcontracts.co.uk>

I have FS installed and I can make outgoing calls through my SIP provider.
 I can also call other extensions (FS is running on a small Xen domU on
the internet), but I am having problems getting the dialplan for incoming
calls to work.

What I want to do is have incoming calls on my number ring all extensions,
e.g. 1000 - 1005 for 10 seconds and then go to voicemail for extension
1000.  If there are no logged-on users, then it should go straight to
voicemail.  Rather than bite off too much, I thought I would try and get a
very basic setup working and take it from there...

At the moment it goes straight to voicemail for extension 1000 even if
1000 is logged in.  Here are the dialplan files I have (everything else is
default from the trunk install):

dialplan/public/00_inbound_did.xml:

<include>
    <extension name="08433xxxxxxx">
        <condition field="destination_number" expression="^(0843xxxxxxx)$">
            <action application="transfer" data="$1 XML default"/>
        </condition>
    </extension>
</include>

dialplan/default/12_voiptalk.xml:

<include>
  <extension name="public_did">
    <condition field="destination_number" expression="^(0843xxxxxxx)$">
      <action application="transfer" data="1000 XML default"/>
    </condition>
  </extension>
  <extension name="outgoing">
    <condition field="destination_number" expression="^9([0-9]+)$">
      <action application="bridge" data="sofia/gateway/voiptalk/$1" />
    </condition>
  </extension>
</include>

I would be very grateful if someone could tell me where I am going wrong. 
I've been looking at various FS wiki pages for hours as well as the
example configs and can't seem to make any headway.  My other question is
what command should I be run after changing the dialplan? is it just
'reloadxml'?


Many thanks,

Matt.



From brian at freeswitch.org  Sun Feb 21 15:32:18 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 21 Feb 2010 17:32:18 -0600
Subject: [Freeswitch-users] Dialplan question
In-Reply-To: <2dac3814a041c79208af8a433b279566.squirrel@www.webcontracts.co.uk>
References: <2dac3814a041c79208af8a433b279566.squirrel@www.webcontracts.co.uk>
Message-ID: <FD94E01E-580B-457D-A26C-1641A680A76A@freeswitch.org>

What do your logs say??? Press F8 and make a call.  Then check the green lines.. maybe its not matching the right thing.

/b

On Feb 21, 2010, at 5:26 PM, Matthew Law wrote:

> I would be very grateful if someone could tell me where I am going wrong. 
> I've been looking at various FS wiki pages for hours as well as the
> example configs and can't seem to make any headway.  My other question is
> what command should I be run after changing the dialplan? is it just
> 'reloadxml'?



From brian at microcomaustralia.com.au  Sun Feb 21 15:44:03 2010
From: brian at microcomaustralia.com.au (Brian May)
Date: Mon, 22 Feb 2010 10:44:03 +1100
Subject: [Freeswitch-users] list all users
In-Reply-To: <3c5cf5261002202000h12b9d6b2ybefd5422fd4f9ee5@mail.gmail.com>
References: <3c5cf5261002201821q7b9f024bt7ab32b10946a5370@mail.gmail.com>
	<33c87fa31002201908x55e43e62g32b340872df9863d@mail.gmail.com>
	<3c5cf5261002202000h12b9d6b2ybefd5422fd4f9ee5@mail.gmail.com>
Message-ID: <3c5cf5261002211544m79832c1cwfe12001dc8b411e9@mail.gmail.com>

On 21 February 2010 15:00, Brian May <brian at microcomaustralia.com.au> wrote:
> As per one suggestion (as far as I can tell this shouldn't be
> required) I tried changing this (in dialplan/default.xml):
>
> <action application="bridge" data="user/${dialed_extension}@${domain_name}"/>
>
> with:
>
> <action application="bridge" data="sofia/internal/${dialed_extension}@${domain_name}"/>

Just noticed why this change didn't work. According to the example in
http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML I needed to
use

<action application="bridge"
data="sofia/internal/${dialed_extension}%${domain_name}"/>

as "The % behind the username tells FS to lookup the user in it's
local sip_registration database" and "If you don't want to bridge a
call to a local registered user, but to a SIP URI, use the @ instead"

disclaimer: not tested! However this matches exactly was was happening.
-- 
Brian May <brian at microcomaustralia.com.au>


From brian at microcomaustralia.com.au  Sun Feb 21 16:51:53 2010
From: brian at microcomaustralia.com.au (Brian May)
Date: Mon, 22 Feb 2010 11:51:53 +1100
Subject: [Freeswitch-users] outgoing dialplans for australia
In-Reply-To: <20100221091011.GA28429@jdc.jasonjgw.net>
References: <3c5cf5261002210030n12bd743fh69f0c862eae0db8@mail.gmail.com>
	<20100221091011.GA28429@jdc.jasonjgw.net>
Message-ID: <3c5cf5261002211651v7962c0afj30e055778b59b1e7@mail.gmail.com>

On 21 February 2010 20:10, Jason White <jason at jasonjgw.net> wrote:
> http://wiki.freeswitch.org/wiki/Extension_Status_Example

Actually suspect the solution might be even simpler. e.g something like:

<action application="set" data="hangup_after_bridge=true"/>
<action application="set"
data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,TIMEOUT,NO_ROUTE_DESTINATION,NORMAL_CIRCUIT_CONGESTION,NETWORK_OUT_OF_ORDER,etc"/>
<action application="bridge" data="sofia/.../$1"/>
<action application="bridge" data="openzap/.../$1"/>
error tone if all else fails

Only I am not in a position to test it just yet, and not sure yet how
to generate the error tone if everything fails either.
-- 
Brian May <brian at microcomaustralia.com.au>


From jason at jasonjgw.net  Sun Feb 21 17:13:29 2010
From: jason at jasonjgw.net (Jason White)
Date: Mon, 22 Feb 2010 12:13:29 +1100
Subject: [Freeswitch-users] outgoing dialplans for australia
In-Reply-To: <3c5cf5261002211651v7962c0afj30e055778b59b1e7@mail.gmail.com>
References: <3c5cf5261002210030n12bd743fh69f0c862eae0db8@mail.gmail.com>
	<20100221091011.GA28429@jdc.jasonjgw.net>
	<3c5cf5261002211651v7962c0afj30e055778b59b1e7@mail.gmail.com>
Message-ID: <20100222011329.GA14787@jdc.jasonjgw.net>

Brian May <brian at microcomaustralia.com.au> wrote:
> On 21 February 2010 20:10, Jason White <jason at jasonjgw.net> wrote:
> > http://wiki.freeswitch.org/wiki/Extension_Status_Example
> 
> Actually suspect the solution might be even simpler. e.g something like:
> 
> <action application="set" data="hangup_after_bridge=true"/>
> <action application="set"
> data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,TIMEOUT,NO_ROUTE_DESTINATION,NORMAL_CIRCUIT_CONGESTION,NETWORK_OUT_OF_ORDER,etc"/>
> <action application="bridge" data="sofia/.../$1"/>
> <action application="bridge" data="openzap/.../$1"/>
> error tone if all else fails

That looks promising.



From brian at microcomaustralia.com.au  Sun Feb 21 18:13:01 2010
From: brian at microcomaustralia.com.au (Brian May)
Date: Mon, 22 Feb 2010 13:13:01 +1100
Subject: [Freeswitch-users] variable substitutions
Message-ID: <3c5cf5261002211813m1835b2b6oc246518a057b578c@mail.gmail.com>

In the sample dialplan, I see the syntax ${...} and the syntax
$${...}. Are both these correct? Using eval suggests that the later
simply prefix the result with a dollar sign, I am not sure if this
intended...

examples - copied from random non-sequential lines in default.xml:

        <action application="set" data="ringback=${us-ring}"/>
        <action application="bridge"
data="user/${dialed_extension}@${domain_name}"/>

and for the $${...} syntax:

        <action application="set" data="transfer_ringback=$${hold_music}"/>
        <action application="set" data="ringback=$${uk-ring}"/>
        <action application="bridge" data="sofia/$${domain}/2000 at bkw.org"/>

Maybe both mean the same thing?
-- 
Brian May <brian at microcomaustralia.com.au>


From jason at jasonjgw.net  Sun Feb 21 18:24:02 2010
From: jason at jasonjgw.net (Jason White)
Date: Mon, 22 Feb 2010 13:24:02 +1100
Subject: [Freeswitch-users] variable substitutions
In-Reply-To: <3c5cf5261002211813m1835b2b6oc246518a057b578c@mail.gmail.com>
References: <3c5cf5261002211813m1835b2b6oc246518a057b578c@mail.gmail.com>
Message-ID: <20100222022402.GA15634@jdc.jasonjgw.net>

Brian May <brian at microcomaustralia.com.au> wrote:
> In the sample dialplan, I see the syntax ${...} and the syntax
> $${...}. Are both these correct? 

Yes. $$ is a preprocessor variable which is expanded when the XML
configuration is parsed. There is a wiki page on the subject.



From Russell.Mosemann at cune.org  Sun Feb 21 18:31:39 2010
From: Russell.Mosemann at cune.org (Russell Mosemann)
Date: Sun, 21 Feb 2010 20:31:39 -0600
Subject: [Freeswitch-users] variable substitutions
In-Reply-To: <3c5cf5261002211813m1835b2b6oc246518a057b578c@mail.gmail.com>
References: <3c5cf5261002211813m1835b2b6oc246518a057b578c@mail.gmail.com>
Message-ID: <6E9F8E83BBE74F2B8DF116B86AE05EDD@cune.pri>

Brian May wrote:

> In the sample dialplan, I see the syntax ${...} and the syntax
> $${...}. Are both these correct?

http://wiki.freeswitch.org/wiki/Channel_Variables#.24.7Bvariable.7D_vs._.24.24.7Bvariable.7D

--
Russell Mosemann



From brian at microcomaustralia.com.au  Sun Feb 21 18:41:35 2010
From: brian at microcomaustralia.com.au (Brian May)
Date: Mon, 22 Feb 2010 13:41:35 +1100
Subject: [Freeswitch-users] variable substitutions
In-Reply-To: <20100222022402.GA15634@jdc.jasonjgw.net>
References: <3c5cf5261002211813m1835b2b6oc246518a057b578c@mail.gmail.com>
	<20100222022402.GA15634@jdc.jasonjgw.net>
Message-ID: <3c5cf5261002211841p2d2626d7j6cfba3b063e05e5c@mail.gmail.com>

On 22 February 2010 13:24, Jason White <jason at jasonjgw.net> wrote:
> Yes. $$ is a preprocessor variable which is expanded when the XML
> configuration is parsed. There is a wiki page on the subject.

Oh, Ok. This looks like a good reference:

http://wiki.freeswitch.org/wiki/Channel_Variables

Thanks.
-- 
Brian May <brian at microcomaustralia.com.au>


From brian at microcomaustralia.com.au  Sun Feb 21 18:52:30 2010
From: brian at microcomaustralia.com.au (Brian May)
Date: Mon, 22 Feb 2010 13:52:30 +1100
Subject: [Freeswitch-users] altering callerid
Message-ID: <3c5cf5261002211852k577d4d78x32e43c0b4d29a1c4@mail.gmail.com>

Ok, now for something maybe a little bit dodgy. How do I alter the
callerid for an incoming call?

Example, the SIP provider I use provides callerid in the format
613XXXXXXXX, however some of the analogue phones have fixed width
displays and cannot display this long number correctly, as such, I
would like to change that to 03XXXXXXXX - compatible with what the
local telephone company uses.

Is this possible?

In asterisk I used:

exten => number/_61NXXXXXXXX,1,Set(CALLERID(num)=0${CALLERID(num):2})
exten => number/_X.,1,Set(CALLERID(num)=+${CALLERID(num)})

(disclaimer - 2nd line not tested with analogue phones)
-- 
Brian May <brian at microcomaustralia.com.au>


From brian at freeswitch.org  Sun Feb 21 18:58:30 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 21 Feb 2010 20:58:30 -0600
Subject: [Freeswitch-users] altering callerid
In-Reply-To: <3c5cf5261002211852k577d4d78x32e43c0b4d29a1c4@mail.gmail.com>
References: <3c5cf5261002211852k577d4d78x32e43c0b4d29a1c4@mail.gmail.com>
Message-ID: <8C9F2C93-DDF3-4283-8015-82062AF794CD@freeswitch.org>

read the variables page ;)

/b

On Feb 21, 2010, at 8:52 PM, Brian May wrote:

> Is this possible?



From brian at microcomaustralia.com.au  Sun Feb 21 19:34:52 2010
From: brian at microcomaustralia.com.au (Brian May)
Date: Mon, 22 Feb 2010 14:34:52 +1100
Subject: [Freeswitch-users] altering callerid
In-Reply-To: <8C9F2C93-DDF3-4283-8015-82062AF794CD@freeswitch.org>
References: <3c5cf5261002211852k577d4d78x32e43c0b4d29a1c4@mail.gmail.com>
	<8C9F2C93-DDF3-4283-8015-82062AF794CD@freeswitch.org>
Message-ID: <3c5cf5261002211934t12f79a5dh5564003af0f4b74d@mail.gmail.com>

On 22 February 2010 13:58, Brian West <brian at freeswitch.org> wrote:
> read the variables page ;)

Which variables? For caller_id_number it says "Practically it is read
only." What does this mean?
-- 
Brian May <brian at microcomaustralia.com.au>


From brian at freeswitch.org  Sun Feb 21 19:39:14 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 21 Feb 2010 21:39:14 -0600
Subject: [Freeswitch-users] altering callerid
In-Reply-To: <3c5cf5261002211934t12f79a5dh5564003af0f4b74d@mail.gmail.com>
References: <3c5cf5261002211852k577d4d78x32e43c0b4d29a1c4@mail.gmail.com>
	<8C9F2C93-DDF3-4283-8015-82062AF794CD@freeswitch.org>
	<3c5cf5261002211934t12f79a5dh5564003af0f4b74d@mail.gmail.com>
Message-ID: <973F4A98-1461-483A-BB9D-8C1746C7DFE8@freeswitch.org>

effective_caller_id_*
origination_caller_id_*

/b

On Feb 21, 2010, at 9:34 PM, Brian May wrote:

> Which variables? For caller_id_number it says "Practically it is read
> only." What does this mean?



From rupa at rupa.com  Sun Feb 21 19:46:01 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Sun, 21 Feb 2010 21:46:01 -0600
Subject: [Freeswitch-users] altering callerid
In-Reply-To: <3c5cf5261002211934t12f79a5dh5564003af0f4b74d@mail.gmail.com>
References: <3c5cf5261002211852k577d4d78x32e43c0b4d29a1c4@mail.gmail.com>
	<8C9F2C93-DDF3-4283-8015-82062AF794CD@freeswitch.org>
	<3c5cf5261002211934t12f79a5dh5564003af0f4b74d@mail.gmail.com>
Message-ID: <b39273531002211946y5361be9cvce49b9aaa8d10d26@mail.gmail.com>

caller_id is slightly tricky.  Use the dptool set_profile_var
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_set_profile_var.
 Though that is under documented.

But really, if you are just bridging then do what Brian said.  That is the
"appropriate" method in most cases.

On Sun, Feb 21, 2010 at 9:34 PM, Brian May
<brian at microcomaustralia.com.au>wrote:

> On 22 February 2010 13:58, Brian West <brian at freeswitch.org> wrote:
> > read the variables page ;)
>
> Which variables? For caller_id_number it says "Practically it is read
> only." What does this mean?
> --
> Brian May <brian at microcomaustralia.com.au>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
-Rupa
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100221/92374dcd/attachment.html 

From brian at microcomaustralia.com.au  Sun Feb 21 19:50:15 2010
From: brian at microcomaustralia.com.au (Brian May)
Date: Mon, 22 Feb 2010 14:50:15 +1100
Subject: [Freeswitch-users] altering callerid
In-Reply-To: <973F4A98-1461-483A-BB9D-8C1746C7DFE8@freeswitch.org>
References: <3c5cf5261002211852k577d4d78x32e43c0b4d29a1c4@mail.gmail.com>
	<8C9F2C93-DDF3-4283-8015-82062AF794CD@freeswitch.org>
	<3c5cf5261002211934t12f79a5dh5564003af0f4b74d@mail.gmail.com>
	<973F4A98-1461-483A-BB9D-8C1746C7DFE8@freeswitch.org>
Message-ID: <3c5cf5261002211950t1bbbb34i20e6509ed345a1a5@mail.gmail.com>

On 22 February 2010 14:39, Brian West <brian at freeswitch.org> wrote:
> effective_caller_id_*
> origination_caller_id_*

How do I generate the new number? I need to be able to test if it
starts with '61', and if so, replace the first two digits with '0',
otherwise just prefix the number with '+'.
-- 
Brian May <brian at microcomaustralia.com.au>


From Russell.Mosemann at cune.org  Sun Feb 21 19:57:47 2010
From: Russell.Mosemann at cune.org (Russell Mosemann)
Date: Sun, 21 Feb 2010 21:57:47 -0600
Subject: [Freeswitch-users] altering callerid
In-Reply-To: <3c5cf5261002211950t1bbbb34i20e6509ed345a1a5@mail.gmail.com>
References: <3c5cf5261002211852k577d4d78x32e43c0b4d29a1c4@mail.gmail.com><8C9F2C93-DDF3-4283-8015-82062AF794CD@freeswitch.org><3c5cf5261002211934t12f79a5dh5564003af0f4b74d@mail.gmail.com><973F4A98-1461-483A-BB9D-8C1746C7DFE8@freeswitch.org>
	<3c5cf5261002211950t1bbbb34i20e6509ed345a1a5@mail.gmail.com>
Message-ID: <B93F97C3D14E4BEA833E14D920F5FCFC@cune.pri>

Brian May asked:
> How do I generate the new number? I need to be able to test if it
> starts with '61', and if so, replace the first two digits with '0',
> otherwise just prefix the number with '+'.

http://wiki.freeswitch.org/wiki/Regular_Expression
http://wiki.freeswitch.org/wiki/Dialplan_XML

--
Russell Mosemann



From freeswitch at cartissolutions.com  Sun Feb 21 20:00:05 2010
From: freeswitch at cartissolutions.com (Yossi Neiman)
Date: Sun, 21 Feb 2010 22:00:05 -0600
Subject: [Freeswitch-users] Doxygen help
Message-ID: <4B820145.2090109@cartissolutions.com>

I am not sure how many folks make use of the Doxygen documentation.  I 
know I do all the time.  I find that it provides a nice conceptual view 
of FreeSWITCH's API, which can make it very easy to find the functions 
and data types needed for writing modules for FreeSWITCH.

A person by the name of Mohammad Shahzad (apologies if I misspelled it) 
started to do a lot of work on revamping the Doxygen configuration for 
FreeSWITCH a few months back.  The problem with the work that he did was 
that he used Doxygen 1.6.x specific configuration parameters that are 
not understood by the Doxygen 1.4.x tree (which is what we currently use 
in FreeSWITCH) or even the 1.5.x tree which is what is in Slackware.  In 
my discussions with Michael Collins we have decided that it might not 
hurt to go ahead and move the Doxygen version forward to match the 1.6.x 
tree.

We are looking for somebody who has interest in working with the Doxygen 
configuration to continue the work that Mohammad had started and to help 
the project out.  If you are such a person, please contact me off-list 
and I can provide further information.

Thanks!

Yossi Neiman


From Russell.Mosemann at cune.org  Sun Feb 21 20:05:59 2010
From: Russell.Mosemann at cune.org (Russell Mosemann)
Date: Sun, 21 Feb 2010 22:05:59 -0600
Subject: [Freeswitch-users] altering callerid
In-Reply-To: <B93F97C3D14E4BEA833E14D920F5FCFC@cune.pri>
References: <3c5cf5261002211852k577d4d78x32e43c0b4d29a1c4@mail.gmail.com><8C9F2C93-DDF3-4283-8015-82062AF794CD@freeswitch.org><3c5cf5261002211934t12f79a5dh5564003af0f4b74d@mail.gmail.com><973F4A98-1461-483A-BB9D-8C1746C7DFE8@freeswitch.org><3c5cf5261002211950t1bbbb34i20e6509ed345a1a5@mail.gmail.com>
	<B93F97C3D14E4BEA833E14D920F5FCFC@cune.pri>
Message-ID: <69D702BA87A54ABA93281A8DA621BB12@cune.pri>

> Brian May asked:
> > How do I generate the new number? I need to be able to test if it
> > starts with '61', and if so, replace the first two digits with '0',
> > otherwise just prefix the number with '+'.
> 
> http://wiki.freeswitch.org/wiki/Regular_Expression
> http://wiki.freeswitch.org/wiki/Dialplan_XML

And

http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_set

--
Russell Mosemann



From jason at jasonjgw.net  Sun Feb 21 20:28:27 2010
From: jason at jasonjgw.net (Jason White)
Date: Mon, 22 Feb 2010 15:28:27 +1100
Subject: [Freeswitch-users] altering callerid
In-Reply-To: <69D702BA87A54ABA93281A8DA621BB12@cune.pri>
References: <3c5cf5261002211852k577d4d78x32e43c0b4d29a1c4@mail.gmail.com>
	<8C9F2C93-DDF3-4283-8015-82062AF794CD@freeswitch.org>
	<3c5cf5261002211934t12f79a5dh5564003af0f4b74d@mail.gmail.com>
	<973F4A98-1461-483A-BB9D-8C1746C7DFE8@freeswitch.org>
	<3c5cf5261002211950t1bbbb34i20e6509ed345a1a5@mail.gmail.com>
	<B93F97C3D14E4BEA833E14D920F5FCFC@cune.pri>
	<69D702BA87A54ABA93281A8DA621BB12@cune.pri>
Message-ID: <20100222042827.GA21444@jdc.jasonjgw.net>

Russell Mosemann <Russell.Mosemann at cune.org> wrote:
 > 
> > http://wiki.freeswitch.org/wiki/Regular_Expression
> > http://wiki.freeswitch.org/wiki/Dialplan_XML
> 
> And
> 
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_set

And as a further hint, regular expressions and variable substitutions ($1 $2
etc.).



From brian at microcomaustralia.com.au  Sun Feb 21 20:37:10 2010
From: brian at microcomaustralia.com.au (Brian May)
Date: Mon, 22 Feb 2010 15:37:10 +1100
Subject: [Freeswitch-users] altering callerid
In-Reply-To: <69D702BA87A54ABA93281A8DA621BB12@cune.pri>
References: <3c5cf5261002211852k577d4d78x32e43c0b4d29a1c4@mail.gmail.com>
	<8C9F2C93-DDF3-4283-8015-82062AF794CD@freeswitch.org>
	<3c5cf5261002211934t12f79a5dh5564003af0f4b74d@mail.gmail.com>
	<973F4A98-1461-483A-BB9D-8C1746C7DFE8@freeswitch.org>
	<3c5cf5261002211950t1bbbb34i20e6509ed345a1a5@mail.gmail.com>
	<B93F97C3D14E4BEA833E14D920F5FCFC@cune.pri>
	<69D702BA87A54ABA93281A8DA621BB12@cune.pri>
Message-ID: <3c5cf5261002212037l3576f9d7s33e32c310799aaf7@mail.gmail.com>

On 22 February 2010 15:05, Russell Mosemann >>
>> http://wiki.freeswitch.org/wiki/Regular_Expression
>> http://wiki.freeswitch.org/wiki/Dialplan_XML
>
> And
>
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_set

So would something like this work?

  <extension name="oztel_cid" continue="true">
    <condition field="destination_number" expression="^(12345678)$"/>
    <condition field="caller_id_number"
expression="^61([2-9]\d\d\d\d\d\d\d\d)$" break="on-true">
      <action application="set" data="effective_caller_id_number=0$1"/>
    </condition>
    <condition field="caller_id_number" expression="^(\d+)$" break="on-true">
      <action application="set" data="effective_caller_id_number=$1"/>
    </condition>
  </extension>
  <extension name="oztel_cid">
    <condition field="destination_number" expression="^(12345678)$">
      <action application="set" data="domain_name=$${domain}"/>
      <action application="transfer" data="2000 XML default"/>
    </condition>
  </extension>

-- 
Brian May <brian at microcomaustralia.com.au>


From brian at freeswitch.org  Sun Feb 21 20:43:02 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 21 Feb 2010 22:43:02 -0600
Subject: [Freeswitch-users] altering callerid
In-Reply-To: <3c5cf5261002212037l3576f9d7s33e32c310799aaf7@mail.gmail.com>
References: <3c5cf5261002211852k577d4d78x32e43c0b4d29a1c4@mail.gmail.com>
	<8C9F2C93-DDF3-4283-8015-82062AF794CD@freeswitch.org>
	<3c5cf5261002211934t12f79a5dh5564003af0f4b74d@mail.gmail.com>
	<973F4A98-1461-483A-BB9D-8C1746C7DFE8@freeswitch.org>
	<3c5cf5261002211950t1bbbb34i20e6509ed345a1a5@mail.gmail.com>
	<B93F97C3D14E4BEA833E14D920F5FCFC@cune.pri>
	<69D702BA87A54ABA93281A8DA621BB12@cune.pri>
	<3c5cf5261002212037l3576f9d7s33e32c310799aaf7@mail.gmail.com>
Message-ID: <C7134664-A3C3-4B9A-BCBA-745EE3B1817A@freeswitch.org>

regex lesson:

^61([2-9]\d{8})$

/b


On Feb 21, 2010, at 10:37 PM, Brian May wrote:

> "^61([2-9]\d\d\d\d\d\d\d\d)$"



From brian at microcomaustralia.com.au  Sun Feb 21 20:57:31 2010
From: brian at microcomaustralia.com.au (Brian May)
Date: Mon, 22 Feb 2010 15:57:31 +1100
Subject: [Freeswitch-users] altering callerid
In-Reply-To: <C7134664-A3C3-4B9A-BCBA-745EE3B1817A@freeswitch.org>
References: <3c5cf5261002211852k577d4d78x32e43c0b4d29a1c4@mail.gmail.com>
	<8C9F2C93-DDF3-4283-8015-82062AF794CD@freeswitch.org>
	<3c5cf5261002211934t12f79a5dh5564003af0f4b74d@mail.gmail.com>
	<973F4A98-1461-483A-BB9D-8C1746C7DFE8@freeswitch.org>
	<3c5cf5261002211950t1bbbb34i20e6509ed345a1a5@mail.gmail.com>
	<B93F97C3D14E4BEA833E14D920F5FCFC@cune.pri>
	<69D702BA87A54ABA93281A8DA621BB12@cune.pri>
	<3c5cf5261002212037l3576f9d7s33e32c310799aaf7@mail.gmail.com>
	<C7134664-A3C3-4B9A-BCBA-745EE3B1817A@freeswitch.org>
Message-ID: <3c5cf5261002212057s6fab56a7o403a3e84763d1ad3@mail.gmail.com>

On 22 February 2010 15:43, Brian West <brian at freeswitch.org> wrote:
> regex lesson:
>
> ^61([2-9]\d{8})$

Much better. Thanks for this.
-- 
Brian May <brian at microcomaustralia.com.au>


From feeswitch.ml at hez.ca  Sun Feb 21 12:11:58 2010
From: feeswitch.ml at hez.ca (Hez Ronningen)
Date: Sun, 21 Feb 2010 12:11:58 -0800
Subject: [Freeswitch-users] dingaling module failing to load with gnutls
	error
Message-ID: <C1FC75D9-29E0-4233-8613-2E23551045B6@hez.ca>

Hello,

Installed freeswitch on ubuntu and enabled the dingaling module but when it boots I get the following error

2010-02-21 11:59:55.213568 [CRIT] switch_loadable_module.c:882 Error Loading module /opt/freeswitch/mod/mod_dingaling.so
**/opt/freeswitch/mod/mod_dingaling.so: undefined symbol: gnutls_global_init**

I have the following libraries installed

ii  libgnutls-dev                        2.8.3-2                                                the GNU TLS library - development files
ii  libgnutls26                          2.8.3-2                                                the GNU TLS library - runtime library


Is there a library I am missing or an incompatibility?

I've checked around on the web and the mailing list archives and no one else seems to have run in to this problem.

Any help is much appreciated,
Hez

From rperry at madisonip.com  Sun Feb 21 21:14:21 2010
From: rperry at madisonip.com (Ryan Perry)
Date: Sun, 21 Feb 2010 23:14:21 -0600
Subject: [Freeswitch-users] Using FS with Asterisk as a PBX
Message-ID: <ea35b6851002212114kd90a5b5h8ebde347d7b0feae@mail.gmail.com>

I'm new to FS.  I am trying to get started with implementing a phone system
to manage 12+ small companies.  I'd planned to use Asterisk, but I've come
to understand the problems with it on a larger scale.  My question is will I
avoid potential problems by using FS to manage 12 Asterisk PBXs?  OR is it
advantageous to use FS's PBX abilities?

Thanks for your opinions and expertise.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100221/67ec9dce/attachment.html 

From mike at jerris.com  Sun Feb 21 22:12:45 2010
From: mike at jerris.com (Michael Jerris)
Date: Mon, 22 Feb 2010 01:12:45 -0500
Subject: [Freeswitch-users] groups
In-Reply-To: <3c5cf5261002210050s235142d6o3f8be42146399c57@mail.gmail.com>
References: <3c5cf5261002210050s235142d6o3f8be42146399c57@mail.gmail.com>
Message-ID: <F7769DC5-6497-4031-BF3E-8E69288D3C43@jerris.com>

You can use any endpoints you want in a group.  This is defined by the dial string for each user.

On Feb 21, 2010, at 3:50 AM, Brian May wrote:

> Hello,
> 
> Can I add a analogue FXS port to a group that is then accessed using
> group_call(...)?
> 
> If so, how? I can only see support for SIP members, as defined in
> directory/default.xml



From mike at jerris.com  Sun Feb 21 22:21:35 2010
From: mike at jerris.com (Michael Jerris)
Date: Mon, 22 Feb 2010 01:21:35 -0500
Subject: [Freeswitch-users] altering callerid
In-Reply-To: <b39273531002211946y5361be9cvce49b9aaa8d10d26@mail.gmail.com>
References: <3c5cf5261002211852k577d4d78x32e43c0b4d29a1c4@mail.gmail.com>
	<8C9F2C93-DDF3-4283-8015-82062AF794CD@freeswitch.org>
	<3c5cf5261002211934t12f79a5dh5564003af0f4b74d@mail.gmail.com>
	<b39273531002211946y5361be9cvce49b9aaa8d10d26@mail.gmail.com>
Message-ID: <A8955E1D-72FF-4385-AE73-3240F8492653@jerris.com>

As a note, this method overwrites what is in the caller profile.  The vars are really the right way to do this.  If you ever find yourself actually using set profile var you are almost definitely doing the wrong thing unless you are 100% sure you are not.  This function was originally left completely undocumented because you should not be using it.  If any documentation is added for this other than completely removing documentation, it should be a bold warning that says you should not under any circumstances use this.

Mike

On Feb 21, 2010, at 10:46 PM, Rupa Schomaker wrote:

> caller_id is slightly tricky.  Use the dptool set_profile_var http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_set_profile_var.  Though that is under documented.  
> 
> But really, if you are just bridging then do what Brian said.  That is the "appropriate" method in most cases.
> 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100222/3f981eb1/attachment.html 

From mike at jerris.com  Sun Feb 21 22:32:40 2010
From: mike at jerris.com (Michael Jerris)
Date: Mon, 22 Feb 2010 01:32:40 -0500
Subject: [Freeswitch-users] Using FS with Asterisk as a PBX
In-Reply-To: <ea35b6851002212114kd90a5b5h8ebde347d7b0feae@mail.gmail.com>
References: <ea35b6851002212114kd90a5b5h8ebde347d7b0feae@mail.gmail.com>
Message-ID: <27ED0182-15CC-41AB-BA9D-864C29B11BDD@jerris.com>

I don't understand the question.  FreeSWITCH does not provide any functionality to manage asterisk instances.

Mike

On Feb 22, 2010, at 12:14 AM, Ryan Perry wrote:

> I'm new to FS.  I am trying to get started with implementing a phone system to manage 12+ small companies.  I'd planned to use Asterisk, but I've come to understand the problems with it on a larger scale.  My question is will I avoid potential problems by using FS to manage 12 Asterisk PBXs?  OR is it advantageous to use FS's PBX abilities?



From pmhshz at gmail.com  Sun Feb 21 22:47:20 2010
From: pmhshz at gmail.com (MohammedShehzad)
Date: Mon, 22 Feb 2010 12:17:20 +0530
Subject: [Freeswitch-users] stream a file multicast with mod_esf
In-Reply-To: <795CD2F3-1395-4C08-A0E7-F5A7BA85F938@jerris.com>
References: <B98B6921358D0C49BD21CD023F1F52C302623B4B@hbghs04a.ww300.siemens.net>
	<378CDEC3-5A41-4EF9-927A-311F2531E6AB@freeswitch.org>
	<a792304b1001080725y19ff28camd37425927550f311@mail.gmail.com>
	<191c3a031001081848w9036207yf6a4b473ec233243@mail.gmail.com>
	<85C74329-0D7C-48B2-B7A8-D4AA67C6E3F2@freeswitch.org>
	<a792304b1002182209m43ada13cp8ed3d91ad509f0a2@mail.gmail.com>
	<DBA4E291-9377-489A-841A-1A3D83880DF2@jerris.com>
	<a792304b1002190002u50ec6912g1905ffc63a119174@mail.gmail.com>
	<a792304b1002192150n44cc500fnd5b2ecb7cff9dd1a@mail.gmail.com>
	<795CD2F3-1395-4C08-A0E7-F5A7BA85F938@jerris.com>
Message-ID: <a792304b1002212247q7cc27ecy5e74368acd68b99c@mail.gmail.com>

Yes, PCMU is hardcoded currently from multicaster. I looked into mod_sndfile
for decoding PCMU to other codec, but it seems that module is using
libsndfile, which reads sound file directly and decode them to L16. If
something similar to libsndfile is available, which work on stream instead
of file io, then it would surely work.

I don't know how exactly Freeswitch's codec structures & functions work, I
am sure decoding can be done by using that, but don't know how to use them.

On Sun, Feb 21, 2010 at 7:17 AM, Michael Jerris <mike at jerris.com> wrote:

> You will need to create the codec for what you need, I think it is
> hardcoded in there to PCMU at the moment, correct?  This will of course need
> to match the stream its reading.
>
> Mike
>
> On Feb 20, 2010, at 12:50 AM, MohammedShehzad wrote:
>
>
>
> On Fri, Feb 19, 2010 at 1:32 PM, MohammedShehzad <pmhshz at gmail.com> wrote:
>
>>
>>
>> On Fri, Feb 19, 2010 at 12:59 PM, Michael Jerris <mike at jerris.com> wrote:
>>
>>> Listening on multicast is noting special for multicast, it is just
>>> like reading any other udp socket
>>>
>>> Mike
>>>
>>> Correct, but I have to play those audio stream back to caller taking care
>> of the audio codec  and other things, do anybody have any idea in that part?
>> Please let me know that.
>>  --
>>
>> -MohammedShehzad
>>
>
> I am able to receive the play the multicasted RAW PCMU RTP (modified the
> skel of format provided by brian), so that caller can hear the multicast
> which done by other Freeswitch server using mod_esf application, but when i
> change the caller's codec from PCMU to something else, it breaks.
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 

-MohammedShehzad
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100222/99c3fef0/attachment.html 

From mike at jerris.com  Sun Feb 21 23:19:23 2010
From: mike at jerris.com (Michael Jerris)
Date: Mon, 22 Feb 2010 02:19:23 -0500
Subject: [Freeswitch-users] stream a file multicast with mod_esf
In-Reply-To: <a792304b1002212247q7cc27ecy5e74368acd68b99c@mail.gmail.com>
References: <B98B6921358D0C49BD21CD023F1F52C302623B4B@hbghs04a.ww300.siemens.net>
	<378CDEC3-5A41-4EF9-927A-311F2531E6AB@freeswitch.org>
	<a792304b1001080725y19ff28camd37425927550f311@mail.gmail.com>
	<191c3a031001081848w9036207yf6a4b473ec233243@mail.gmail.com>
	<85C74329-0D7C-48B2-B7A8-D4AA67C6E3F2@freeswitch.org>
	<a792304b1002182209m43ada13cp8ed3d91ad509f0a2@mail.gmail.com>
	<DBA4E291-9377-489A-841A-1A3D83880DF2@jerris.com>
	<a792304b1002190002u50ec6912g1905ffc63a119174@mail.gmail.com>
	<a792304b1002192150n44cc500fnd5b2ecb7cff9dd1a@mail.gmail.com>
	<795CD2F3-1395-4C08-A0E7-F5A7BA85F938@jerris.com>
	<a792304b1002212247q7cc27ecy5e74368acd68b99c@mail.gmail.com>
Message-ID: <8A3C6D64-9215-47A5-8FCD-7A328770772D@jerris.com>

You would just change the PCMU to whatever codec you want.  This should have nothing to do with file io, take a look at the line that has PCMU hardcoded, thats all you should need to change.

Mike

On Feb 22, 2010, at 1:47 AM, MohammedShehzad wrote:

> Yes, PCMU is hardcoded currently from multicaster. I looked into mod_sndfile for decoding PCMU to other codec, but it seems that module is using libsndfile, which reads sound file directly and decode them to L16. If something similar to libsndfile is available, which work on stream instead of file io, then it would surely work.
> 
> I don't know how exactly Freeswitch's codec structures & functions work, I am sure decoding can be done by using that, but don't know how to use them.
> 
> On Sun, Feb 21, 2010 at 7:17 AM, Michael Jerris <mike at jerris.com> wrote:
> You will need to create the codec for what you need, I think it is hardcoded in there to PCMU at the moment, correct?  This will of course need to match the stream its reading.
> 
> Mike
> 
> On Feb 20, 2010, at 12:50 AM, MohammedShehzad wrote:
> 
>> 
>> 
>> On Fri, Feb 19, 2010 at 1:32 PM, MohammedShehzad <pmhshz at gmail.com> wrote:
>> 
>> 
>> On Fri, Feb 19, 2010 at 12:59 PM, Michael Jerris <mike at jerris.com> wrote:
>> Listening on multicast is noting special for multicast, it is just
>> like reading any other udp socket
>> 
>> Mike
>> 
>> Correct, but I have to play those audio stream back to caller taking care of the audio codec  and other things, do anybody have any idea in that part? Please let me know that.
>> -- 
>> 
>> -MohammedShehzad
>> 
>> I am able to receive the play the multicasted RAW PCMU RTP (modified the skel of format provided by brian), so that caller can hear the multicast which done by other Freeswitch server using mod_esf application, but when i change the caller's codec from PCMU to something else, it breaks.
> 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100222/7489f43d/attachment.html 

From gamar at center.com  Mon Feb 22 00:52:08 2010
From: gamar at center.com (Gilbert Amar)
Date: Mon, 22 Feb 2010 09:52:08 +0100
Subject: [Freeswitch-users] Freeswitch and mod_opal
Message-ID: <4A424236C3C44A8FBED67E818845BCAD@gamar>

Hello,


I try Freeswitch and mod_opal on CenTos and on Windows XP
Calling FS IVR from Openphone or a regular H323 phone works
But on those two platforms I could not bridge two calls using h323. There is 
always a mute or deaf leg.
I also try to build mod_h323 with no success.
Did anyone have tried this on the svn trunk and succeeded.
If yes I will be glad to know how you did and what parameters you choose 
regarding faststart, h245 tunneling, codecs, etc.


Gilbert



From helmut.kuper at ewetel.de  Mon Feb 22 01:09:19 2010
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Mon, 22 Feb 2010 10:09:19 +0100
Subject: [Freeswitch-users] Question about sofia_contact
In-Reply-To: <4B7EADBC.1040001@ewetel.de>
References: <4B7EA954.30402@ewetel.de> <4B7EADBC.1040001@ewetel.de>
Message-ID: <4B8249BF.3090708@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hello,

has anybody an idea?

regards
helmut


On 19.02.2010 16:26, Helmut Kuper wrote:
> Hi,
> 
> an update:
> The corresponding select statement looks for sip_user="2701" and
> sip_host="internal" in registration table.
> This delivers of course no result because 2701 is registered with
> sip_host="mydomain".
> 
> 
> Hm any workaround or am I going in a wrong direction?
> 
> 
> regards
> Helmut
> 
> 
> On 19.02.2010 16:08, Helmut Kuper wrote:
>> Hello,
> 
>> I try to setup a FS sofia sip-profile which allows me to have multiple
>> sip-profiles but one registration database. So I set the following
>> parameters:
> 
>> <param name="challenge-realm" value="auto_from"/>
>> <param name="force-register-domain" value="$${domain}"/>
>> <param name="force-subscription-domain" value="$${domain}"/>
>> <param name="force-register-db-domain" value="$${domain}"/>
> 
>> where domain is set to "mydomain". "sofia status profile internal"
>> delivers the following:
> 
> 
>> Call-ID:        3c26705038e5-vwlg8u5q9cwe
>> User:           2701 at mydomain
>> Contact:        <sip:2701 at 85.16.245.208:1024;line=l2wksedo>
>> Agent:          snom370/8.2.22
>> Status:         Registered(UDP)(unknown) EXP(2010-02-19 16:13:31)
>> Host:           ippbx-prod-node0
>> IP:             85.16.245.208
>> Port:           1024
>> Auth-User:      2701
>> Auth-Realm:     mydomain
>> MWI-Account:    2701 at mydomain
> 
> 
> 
>> sofia_contact internal/2701 at mydomain delivers this:
>> error/user_not_registered
> 
>> The Phone is fully functional.
> 
>> I use SVN trunk 16601
> 
>> regards
>> Helmut

-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.4.7 (MingW32)

iD8DBQFLgkm/4tZeNddg3dwRApfoAKCiX8fX/WNrZ7GXRrBJA54+VTThmACfT0d3
fBzQlyVObkJaLHxJbfUjZG4=
=uhHR
-----END PGP SIGNATURE-----


From oseslija at gmail.com  Mon Feb 22 01:12:47 2010
From: oseslija at gmail.com (Ognjen Seslija)
Date: Mon, 22 Feb 2010 10:12:47 +0100
Subject: [Freeswitch-users] Using FS with Asterisk as a PBX
In-Reply-To: <ea35b6851002212114kd90a5b5h8ebde347d7b0feae@mail.gmail.com>
References: <ea35b6851002212114kd90a5b5h8ebde347d7b0feae@mail.gmail.com>
Message-ID: <4468a6771002220112k2b656607g4638602582fb5cc1@mail.gmail.com>

You can safely replace those asterisks with a single FS configured for
multitenant/multidomain PBXes.

Ognjen

On Mon, Feb 22, 2010 at 6:14 AM, Ryan Perry <rperry at madisonip.com> wrote:

> I'm new to FS.  I am trying to get started with implementing a phone system
> to manage 12+ small companies.  I'd planned to use Asterisk, but I've come
> to understand the problems with it on a larger scale.  My question is will I
> avoid potential problems by using FS to manage 12 Asterisk PBXs?  OR is it
> advantageous to use FS's PBX abilities?
>
> Thanks for your opinions and expertise.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100222/98b545b5/attachment.html 

From shaheryarkh at googlemail.com  Mon Feb 22 02:00:21 2010
From: shaheryarkh at googlemail.com (Muhammad Shahzad)
Date: Mon, 22 Feb 2010 15:00:21 +0500
Subject: [Freeswitch-users] Doxygen help
In-Reply-To: <4B820145.2090109@cartissolutions.com>
References: <4B820145.2090109@cartissolutions.com>
Message-ID: <b16156851002220200u3736acbdre1a1b82db0e82651@mail.gmail.com>

Hi,

Its a quite embarrassing to admit that i could completed that work, not even
able to submit what was done to FS trunk due to my own stupidity. Actually,
we are having major power crisis in the country for last 1 years and most of
power cuts are so long that even best UPS systems can't ensure 24/7 up time
for our servers. About 3 months ago the development server i was working on
for FS Docs was hit by this problem and its hard disk crashed, destroying
all my hard work on FS Docs and mod_msn. It was my mistake that i didn't
arrange for proper backup for my work due to shortage of resources and
didn't inform the FS developers community about the loss.

I want to restart this work but don't want to take the lead on this due to
issues described above. If you can arrange a common server for FS docs
development then i can submit my work to you which you can test on this dev
server before committing it to FS Trunk.

Second option is that i work on a single file, test it on my laptop and
immediately submit it to FS trunk. But this could cause problem for end
users (developers using FS doxygen docs) as many hyper links connected to
undocumented files won't work till their docs are done and uploaded.

We also need to decide on,

1. What Doxygen version to use? i use Archlinux which has the latest 1.6.x
version, I can downgrade it to 1.4.x.
2. What format of documentation to adopt? I think currently we have HTML
format only, but Doxygen can also generate PDF and CHM formats.

Thank you.


On Mon, Feb 22, 2010 at 9:00 AM, Yossi Neiman <
freeswitch at cartissolutions.com> wrote:

> I am not sure how many folks make use of the Doxygen documentation.  I
> know I do all the time.  I find that it provides a nice conceptual view
> of FreeSWITCH's API, which can make it very easy to find the functions
> and data types needed for writing modules for FreeSWITCH.
>
> A person by the name of Mohammad Shahzad (apologies if I misspelled it)
> started to do a lot of work on revamping the Doxygen configuration for
> FreeSWITCH a few months back.  The problem with the work that he did was
> that he used Doxygen 1.6.x specific configuration parameters that are
> not understood by the Doxygen 1.4.x tree (which is what we currently use
> in FreeSWITCH) or even the 1.5.x tree which is what is in Slackware.  In
> my discussions with Michael Collins we have decided that it might not
> hurt to go ahead and move the Doxygen version forward to match the 1.6.x
> tree.
>
> We are looking for somebody who has interest in working with the Doxygen
> configuration to continue the work that Mohammad had started and to help
> the project out.  If you are such a person, please contact me off-list
> and I can provide further information.
>
> Thanks!
>
> Yossi Neiman
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Muhammad Shahzad
-----------------------------------
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +92 334 422 40 88
MSN: shari_786pk at hotmail.com
Email: shaheryarkh at googlemail.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100222/c7b55b40/attachment.html 

From pmhshz at gmail.com  Mon Feb 22 03:56:55 2010
From: pmhshz at gmail.com (MohammedShehzad)
Date: Mon, 22 Feb 2010 17:26:55 +0530
Subject: [Freeswitch-users] stream a file multicast with mod_esf
In-Reply-To: <8A3C6D64-9215-47A5-8FCD-7A328770772D@jerris.com>
References: <B98B6921358D0C49BD21CD023F1F52C302623B4B@hbghs04a.ww300.siemens.net>
	<191c3a031001081848w9036207yf6a4b473ec233243@mail.gmail.com>
	<85C74329-0D7C-48B2-B7A8-D4AA67C6E3F2@freeswitch.org>
	<a792304b1002182209m43ada13cp8ed3d91ad509f0a2@mail.gmail.com>
	<DBA4E291-9377-489A-841A-1A3D83880DF2@jerris.com>
	<a792304b1002190002u50ec6912g1905ffc63a119174@mail.gmail.com>
	<a792304b1002192150n44cc500fnd5b2ecb7cff9dd1a@mail.gmail.com>
	<795CD2F3-1395-4C08-A0E7-F5A7BA85F938@jerris.com>
	<a792304b1002212247q7cc27ecy5e74368acd68b99c@mail.gmail.com>
	<8A3C6D64-9215-47A5-8FCD-7A328770772D@jerris.com>
Message-ID: <a792304b1002220356t18b1fb5exfdeb02a75e425976@mail.gmail.com>

Actually I don't want to change anything from Multicaster, I am talking
about the changes required on Listener side (the format module I am going to
develo). I think I should discuss further on developer's mailing list,
please let me know you ideas there.

Thanks for your response.

On Mon, Feb 22, 2010 at 12:49 PM, Michael Jerris <mike at jerris.com> wrote:

> You would just change the PCMU to whatever codec you want.  This should
> have nothing to do with file io, take a look at the line that has PCMU
> hardcoded, thats all you should need to change.
>
> Mike
>
> On Feb 22, 2010, at 1:47 AM, MohammedShehzad wrote:
>
> Yes, PCMU is hardcoded currently from multicaster. I looked into
> mod_sndfile for decoding PCMU to other codec, but it seems that module is
> using libsndfile, which reads sound file directly and decode them to L16. If
> something similar to libsndfile is available, which work on stream instead
> of file io, then it would surely work.
>
> I don't know how exactly Freeswitch's codec structures & functions work, I
> am sure decoding can be done by using that, but don't know how to use them.
>
> On Sun, Feb 21, 2010 at 7:17 AM, Michael Jerris <mike at jerris.com> wrote:
>
>> You will need to create the codec for what you need, I think it is
>> hardcoded in there to PCMU at the moment, correct?  This will of course need
>> to match the stream its reading.
>>
>> Mike
>>
>> On Feb 20, 2010, at 12:50 AM, MohammedShehzad wrote:
>>
>>
>>
>> On Fri, Feb 19, 2010 at 1:32 PM, MohammedShehzad <pmhshz at gmail.com>wrote:
>>
>>>
>>>
>>> On Fri, Feb 19, 2010 at 12:59 PM, Michael Jerris <mike at jerris.com>wrote:
>>>
>>>> Listening on multicast is noting special for multicast, it is just
>>>> like reading any other udp socket
>>>>
>>>> Mike
>>>>
>>>> Correct, but I have to play those audio stream back to caller taking
>>> care of the audio codec  and other things, do anybody have any idea in that
>>> part? Please let me know that.
>>>  --
>>>
>>> -MohammedShehzad
>>>
>>
>> I am able to receive the play the multicasted RAW PCMU RTP (modified the
>> skel of format provided by brian), so that caller can hear the multicast
>> which done by other Freeswitch server using mod_esf application, but when i
>> change the caller's codec from PCMU to something else, it breaks.
>>
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 

-MohammedShehzad
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100222/7da0738f/attachment.html 

From tculjaga at gmail.com  Mon Feb 22 04:33:34 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Mon, 22 Feb 2010 13:33:34 +0100
Subject: [Freeswitch-users] Freeswitch and mod_opal
In-Reply-To: <4A424236C3C44A8FBED67E818845BCAD@gamar>
References: <4A424236C3C44A8FBED67E818845BCAD@gamar>
Message-ID: <65d96fc81002220433j25e428d9m2ddff5c0b40ef344@mail.gmail.com>

mod_h323

you need to build h323plus & ptlib before you build the module itself.

check http://wiki.freeswitch.org/wiki/Mod_h323 it works out of the box.


regarding you call setup, faststart = true, h245tuneling = true, h245InSetup
= false

It has to work.... if not post the logs here.

T.



On Mon, Feb 22, 2010 at 9:52 AM, Gilbert Amar <gamar at center.com> wrote:

> Hello,
>
>
> I try Freeswitch and mod_opal on CenTos and on Windows XP
> Calling FS IVR from Openphone or a regular H323 phone works
> But on those two platforms I could not bridge two calls using h323. There
> is
> always a mute or deaf leg.
> I also try to build mod_h323 with no success.
> Did anyone have tried this on the svn trunk and succeeded.
> If yes I will be glad to know how you did and what parameters you choose
> regarding faststart, h245 tunneling, codecs, etc.
>
>
> Gilbert
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100222/1970fca9/attachment.html 

From ivdreg at gmail.com  Mon Feb 22 04:48:57 2010
From: ivdreg at gmail.com (ivdreg ivdreg)
Date: Mon, 22 Feb 2010 14:48:57 +0200
Subject: [Freeswitch-users] SDP With T.38 in INVITE Problem
In-Reply-To: <b437a0e1002190242y2aa64696j91e8bc76f7ae2a92@mail.gmail.com>
References: <b437a0e1002190242y2aa64696j91e8bc76f7ae2a92@mail.gmail.com>
Message-ID: <b437a0e1002220448g3e3bda9fpa283e93a11d319f1@mail.gmail.com>

Hi All,

Actually while seeking the solution in internet I see some people having
this problem with sofia library. I'm not sure that SIP reply in this case
contains a valid SDP (I think that teminating endpoint is broken) but in my
opinion if we have at least one valid media type in SDP (video, audio, image
...) call must be established. Can someone comment and/or help me with this
issue.

Regards.

2010/2/19 ivdreg ivdreg <ivdreg at gmail.com>

> Hi all,
>
> Dose someone have a problem that if there T.38 in coming from gateway
> FreeSwitch drops the call because of media error ? As I see from log only
> T.38 port is zero and SDP has also media port. Is it possible to configure
> FS to do not break a call but if media is OK.
>
> 2010-02-19 12:26:14.305244 [NOTICE] switch_channel.c:666 New Channel
> sofia/backend/XXXXXXXXXX at 10.10.1.110:7065[6cd9f634-411d-df11-99ca-003048bb99cc]
> 2010-02-19 12:26:14.305244 [DEBUG] mod_sofia.c:3369 (sofia/backend/
> XXXXXXXXXX at 10.10.1.110:7065) State Change CS_NEW -> CS_INIT
> 2010-02-19 12:26:14.305244 [DEBUG] switch_core_session.c:1018 Send signal
> sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [BREAK]
> 2010-02-19 12:26:14.305244 [DEBUG] switch_core_state_machine.c:314
> (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) Running State Change CS_INIT
> 2010-02-19 12:26:14.305244 [DEBUG] switch_core_state_machine.c:338
> (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State INIT
> 2010-02-19 12:26:14.305244 [DEBUG] mod_sofia.c:83 sofia/backend/
> XXXXXXXXXX at 10.10.1.110:7065 SOFIA INIT
> 2010-02-19 12:26:14.305244 [DEBUG] sofia_glue.c:1377 sofia/backend/
> XXXXXXXXXX at 10.10.1.110:7065 Patched SDP
> ---
> v=0
> o=- 3779949069 5166785777234550622 IN IP4 206.132.232.212
> s=session
> t=0 0
> m=audio 21108 RTP/AVP 18 4 8 0
> c=IN IP4 10.10.1.110
> a=rtpmap:18 G729/8000
> a=rtpmap:4 G723/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:0 PCMU/8000
> m=image 21108 udptl t38
> c=IN IP4 10.10.1.110
> a=T38FaxVersion:0
> a=T38MaxBitRate:14400
> a=T38FaxUdpEC:t38UDPRedundancy
> a=T38FaxRateManagement:transferredTCF
>
> +++
> v=0
> o=- 3779949069 5166785777234550622 IN IP4 206.132.232.212
> s=session
> t=0 0
> m=audio 17058 RTP/AVP 18 4 8 0
> c=IN IP4 10.10.1.100
> a=rtpmap:18 G729/8000
> a=rtpmap:4 G723/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:0 PCMU/8000
> m=image 17058 udptl t38
> c=IN IP4 10.10.1.100
> a=T38FaxVersion:0
> a=T38MaxBitRate:14400
> a=T38FaxUdpEC:t38UDPRedundancy
> a=T38FaxRateManagement:transferredTCF
>
> 2010-02-19 12:26:14.305244 [DEBUG] mod_sofia.c:117 (sofia/backend/
> XXXXXXXXXX at 10.10.1.110:7065) State Change CS_INIT -> CS_ROUTING
> ......
> 2010-02-19 12:26:21.255201 [DEBUG] sofia.c:4135 Remote SDP:
> v=0
> o=FreeSWITCH 1266548331 1266548332 IN IP4 10.10.1.110
> s=FreeSWITCH
> c=IN IP4 10.10.1.110
> t=0 0
> *m=audio 26850 RTP/AVP 8*
> a=rtpmap:8 PCMA/8000
> a=silenceSupp:off - - - -
> a=ptime:20
> *m=image 0 udptl 19*
>
> 2010-02-19 12:26:21.255201 [DEBUG] sofia.c:4124 Channel sofia/backend/
> XXXXXXXXXX at 10.10.1.110:7065 entering state [ready][200]
> 2010-02-19 12:26:21.255201 [DEBUG] switch_channel.c:2285 Send signal
> sofia/backend/YYYYYYYYYY at 10.10.1.110 [BREAK]
> 2010-02-19 12:26:21.255201 [NOTICE] sofia.c:4668 Channel [sofia/backend/
> XXXXXXXXXX at 10.10.1.110:7065] has been answered
> 2010-02-19 12:26:21.255201 [DEBUG] sofia_glue.c:2288 Set Codec
> sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 PROXY/8000 20 ms 160 samples
> *2010-02-19 12:26:21.255201 [DEBUG] sofia_glue.c:2571 PROXY AUDIO RTP
> [sofia/backend/XXXXXXXXXX at 10.10.1.110:7065] 10.10.1.100:17058->
> 10.10.1.110:0 codec: 0 ms: 20
> 2010-02-19 12:26:21.255201 [ERR] sofia_glue.c:2879 AUDIO RTP REPORTS ERROR:
> [Missing remote port]
> 2010-02-19 12:26:21.255201 [NOTICE] sofia_glue.c:2880 Hangup sofia/backend/
> XXXXXXXXXX at 10.10.1.110:7065 [CS_CONSUME_MEDIA] [DESTINATION_OUT_OF_ORDER]*
> 2010-02-19 12:26:21.255201 [DEBUG] switch_channel.c:2063 Send signal
> sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [KILL]
> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_session.c:1018 Send signal
> sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [BREAK]
> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:314
> (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) Running State Change CS_HANGUP
> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:499
> (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State HANGUP
> 2010-02-19 12:26:21.255201 [DEBUG] mod_sofia.c:411 Channel sofia/backend/
> XXXXXXXXXX at 10.10.1.110:7065 hanging up, cause: DESTINATION_OUT_OF_ORDER
> 2010-02-19 12:26:21.255201 [DEBUG] mod_sofia.c:454 Sending BYE to
> sofia/backend/XXXXXXXXXX at 10.10.1.110:7065
> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:46
> sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 Standard HANGUP, cause:
> DESTINATION_OUT_OF_ORDER
> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:499
> (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State HANGUP going to sleep
> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:333
> (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State Change CS_HANGUP ->
> CS_REPORTING
> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_session.c:1018 Send signal
> sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [BREAK]
> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:314
> (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) Running State Change
> CS_REPORTING
> 2010-02-19 12:26:21.255201 [DEBUG] switch_ivr_originate.c:3185 Originate
> Resulted in Error Cause: 27 [DESTINATION_OUT_OF_ORDER]
> 2010-02-19 12:26:21.255201 [INFO] mod_dptools.c:2353 Originate Failed.
> Cause: DESTINATION_OUT_OF_ORDER
> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:590
> (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State REPORTING
>
>
> Thanks
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100222/38410808/attachment-0001.html 

From dftoro at yahoo.com  Mon Feb 22 05:11:29 2010
From: dftoro at yahoo.com (Diego Toro)
Date: Mon, 22 Feb 2010 05:11:29 -0800 (PST)
Subject: [Freeswitch-users] Dialplan question
In-Reply-To: <2dac3814a041c79208af8a433b279566.squirrel@www.webcontracts.co.uk>
Message-ID: <240053.46379.qm@web33504.mail.mud.yahoo.com>

Hi, check <extension name="Local_Extension"> on conf/dialplan/default.xml, You can get an idea of how to do what you need.


Diego Toro
http://lacarretade.blogspot.com/


--- On Sun, 2/21/10, Matthew Law <matt at webcontracts.co.uk> wrote:

> From: Matthew Law <matt at webcontracts.co.uk>
> Subject: [Freeswitch-users] Dialplan question
> To: freeswitch-users at lists.freeswitch.org
> Date: Sunday, February 21, 2010, 6:26 PM
> I have FS installed and I can make
> outgoing calls through my SIP provider.
>  I can also call other extensions (FS is running on a small
> Xen domU on
> the internet), but I am having problems getting the
> dialplan for incoming
> calls to work.
> 
> What I want to do is have incoming calls on my number ring
> all extensions,
> e.g. 1000 - 1005 for 10 seconds and then go to voicemail
> for extension
> 1000.? If there are no logged-on users, then it should
> go straight to
> voicemail.? Rather than bite off too much, I thought I
> would try and get a
> very basic setup working and take it from there...
> 
> At the moment it goes straight to voicemail for extension
> 1000 even if
> 1000 is logged in.? Here are the dialplan files I have
> (everything else is
> default from the trunk install):
> 
> dialplan/public/00_inbound_did.xml:
> 
> <include>
> ? ? <extension name="08433xxxxxxx">
> ? ? ? ? <condition
> field="destination_number" expression="^(0843xxxxxxx)$">
> ? ? ? ? ? ? <action
> application="transfer" data="$1 XML default"/>
> ? ? ? ? </condition>
> ? ? </extension>
> </include>
> 
> dialplan/default/12_voiptalk.xml:
> 
> <include>
> ? <extension name="public_did">
> ? ? <condition field="destination_number"
> expression="^(0843xxxxxxx)$">
> ? ? ? <action application="transfer"
> data="1000 XML default"/>
> ? ? </condition>
> ? </extension>
> ? <extension name="outgoing">
> ? ? <condition field="destination_number"
> expression="^9([0-9]+)$">
> ? ? ? <action application="bridge"
> data="sofia/gateway/voiptalk/$1" />
> ? ? </condition>
> ? </extension>
> </include>
> 
> I would be very grateful if someone could tell me where I
> am going wrong. 
> I've been looking at various FS wiki pages for hours as
> well as the
> example configs and can't seem to make any headway.?
> My other question is
> what command should I be run after changing the dialplan?
> is it just
> 'reloadxml'?
> 
> 
> Many thanks,
> 
> Matt.
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 


      


From gamar at center.com  Mon Feb 22 05:30:22 2010
From: gamar at center.com (Gilbert Amar)
Date: Mon, 22 Feb 2010 14:30:22 +0100
Subject: [Freeswitch-users] Freeswitch and mod_opal
Message-ID: <BDB26458EDF74CF6A382375739AF86BD@gamar>

Thank you Tihomir


We tried to build the svn trunk of mod_h323 and succeded but got a core dump
a the first call hanging up.

>mod_h323
>you need to build h323plus & ptlib before you build the module itself.
>check http://wiki.freeswitch.org/wiki/Mod_h323 it works out of the box.
>regarding you call setup, faststart = true, h245tuneling = true,
>h245InSetup = false
>It has to work.... if not post the logs here.

>T.



From matt at webcontracts.co.uk  Mon Feb 22 06:16:48 2010
From: matt at webcontracts.co.uk (Matthew Law)
Date: Mon, 22 Feb 2010 14:16:48 -0000
Subject: [Freeswitch-users] Dialplan question
In-Reply-To: <240053.46379.qm@web33504.mail.mud.yahoo.com>
References: <240053.46379.qm@web33504.mail.mud.yahoo.com>
Message-ID: <a8f8fa302ad239a2d6c8332838785448.squirrel@www.webcontracts.co.uk>

On Mon, February 22, 2010 1:11 pm, Diego Toro wrote:
> Hi, check <extension name="Local_Extension"> on conf/dialplan/default.xml,
> You can get an idea of how to do what you need.
>
>
> Diego Toro
> http://lacarretade.blogspot.com/

Thank you all.

It seems it was working in it's current configuration, I just had to
figure out how to get FS to reload the internal config.

Matt.





From mike at jerris.com  Mon Feb 22 06:45:08 2010
From: mike at jerris.com (Michael Jerris)
Date: Mon, 22 Feb 2010 09:45:08 -0500
Subject: [Freeswitch-users] SDP With T.38 in INVITE Problem
In-Reply-To: <b437a0e1002220448g3e3bda9fpa283e93a11d319f1@mail.gmail.com>
References: <b437a0e1002190242y2aa64696j91e8bc76f7ae2a92@mail.gmail.com>
	<b437a0e1002220448g3e3bda9fpa283e93a11d319f1@mail.gmail.com>
Message-ID: <24736FFE-BEEF-4D34-BBA7-77B6B6AE0EBD@jerris.com>

If the endpoint does not correctly follow the sdp o/a model its not going to work.  This is not a "problem" with the sofia library, this is intended behavior and what we are supposed to do.  What is the device?

Mike

On Feb 22, 2010, at 7:48 AM, ivdreg ivdreg wrote:

> Hi All,
> 
> Actually while seeking the solution in internet I see some people having this problem with sofia library. I'm not sure that SIP reply in this case contains a valid SDP (I think that teminating endpoint is broken) but in my opinion if we have at least one valid media type in SDP (video, audio, image ...) call must be established. Can someone comment and/or help me with this issue.
> 
> Regards.
> 
> 2010/2/19 ivdreg ivdreg <ivdreg at gmail.com>
> Hi all,
> 
> Dose someone have a problem that if there T.38 in coming from gateway FreeSwitch drops the call because of media error ? As I see from log only T.38 port is zero and SDP has also media port. Is it possible to configure FS to do not break a call but if media is OK.
> 
> 2010-02-19 12:26:14.305244 [NOTICE] switch_channel.c:666 New Channel sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [6cd9f634-411d-df11-99ca-003048bb99cc]
> 2010-02-19 12:26:14.305244 [DEBUG] mod_sofia.c:3369 (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State Change CS_NEW -> CS_INIT
> 2010-02-19 12:26:14.305244 [DEBUG] switch_core_session.c:1018 Send signal sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [BREAK]
> 2010-02-19 12:26:14.305244 [DEBUG] switch_core_state_machine.c:314 (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) Running State Change CS_INIT
> 2010-02-19 12:26:14.305244 [DEBUG] switch_core_state_machine.c:338 (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State INIT
> 2010-02-19 12:26:14.305244 [DEBUG] mod_sofia.c:83 sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 SOFIA INIT
> 2010-02-19 12:26:14.305244 [DEBUG] sofia_glue.c:1377 sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 Patched SDP
> ---
> v=0
> o=- 3779949069 5166785777234550622 IN IP4 206.132.232.212
> s=session
> t=0 0
> m=audio 21108 RTP/AVP 18 4 8 0
> c=IN IP4 10.10.1.110
> a=rtpmap:18 G729/8000
> a=rtpmap:4 G723/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:0 PCMU/8000
> m=image 21108 udptl t38
> c=IN IP4 10.10.1.110
> a=T38FaxVersion:0
> a=T38MaxBitRate:14400
> a=T38FaxUdpEC:t38UDPRedundancy
> a=T38FaxRateManagement:transferredTCF
> 
> +++
> v=0
> o=- 3779949069 5166785777234550622 IN IP4 206.132.232.212
> s=session
> t=0 0
> m=audio 17058 RTP/AVP 18 4 8 0
> c=IN IP4 10.10.1.100
> a=rtpmap:18 G729/8000
> a=rtpmap:4 G723/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:0 PCMU/8000
> m=image 17058 udptl t38
> c=IN IP4 10.10.1.100
> a=T38FaxVersion:0
> a=T38MaxBitRate:14400
> a=T38FaxUdpEC:t38UDPRedundancy
> a=T38FaxRateManagement:transferredTCF
> 
> 2010-02-19 12:26:14.305244 [DEBUG] mod_sofia.c:117 (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State Change CS_INIT -> CS_ROUTING
> ......
> 2010-02-19 12:26:21.255201 [DEBUG] sofia.c:4135 Remote SDP:
> v=0
> o=FreeSWITCH 1266548331 1266548332 IN IP4 10.10.1.110
> s=FreeSWITCH
> c=IN IP4 10.10.1.110
> t=0 0
> m=audio 26850 RTP/AVP 8
> a=rtpmap:8 PCMA/8000
> a=silenceSupp:off - - - -
> a=ptime:20
> m=image 0 udptl 19
> 
> 2010-02-19 12:26:21.255201 [DEBUG] sofia.c:4124 Channel sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 entering state [ready][200]
> 2010-02-19 12:26:21.255201 [DEBUG] switch_channel.c:2285 Send signal sofia/backend/YYYYYYYYYY at 10.10.1.110 [BREAK]
> 2010-02-19 12:26:21.255201 [NOTICE] sofia.c:4668 Channel [sofia/backend/XXXXXXXXXX at 10.10.1.110:7065] has been answered
> 2010-02-19 12:26:21.255201 [DEBUG] sofia_glue.c:2288 Set Codec sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 PROXY/8000 20 ms 160 samples
> 2010-02-19 12:26:21.255201 [DEBUG] sofia_glue.c:2571 PROXY AUDIO RTP [sofia/backend/XXXXXXXXXX at 10.10.1.110:7065] 10.10.1.100:17058->10.10.1.110:0 codec: 0 ms: 20
> 2010-02-19 12:26:21.255201 [ERR] sofia_glue.c:2879 AUDIO RTP REPORTS ERROR: [Missing remote port]
> 2010-02-19 12:26:21.255201 [NOTICE] sofia_glue.c:2880 Hangup sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [CS_CONSUME_MEDIA] [DESTINATION_OUT_OF_ORDER]
> 2010-02-19 12:26:21.255201 [DEBUG] switch_channel.c:2063 Send signal sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [KILL]
> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_session.c:1018 Send signal sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [BREAK]
> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:314 (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) Running State Change CS_HANGUP
> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:499 (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State HANGUP
> 2010-02-19 12:26:21.255201 [DEBUG] mod_sofia.c:411 Channel sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 hanging up, cause: DESTINATION_OUT_OF_ORDER
> 2010-02-19 12:26:21.255201 [DEBUG] mod_sofia.c:454 Sending BYE to sofia/backend/XXXXXXXXXX at 10.10.1.110:7065
> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:46 sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 Standard HANGUP, cause: DESTINATION_OUT_OF_ORDER
> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:499 (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State HANGUP going to sleep
> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:333 (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State Change CS_HANGUP -> CS_REPORTING
> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_session.c:1018 Send signal sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [BREAK]
> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:314 (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) Running State Change CS_REPORTING
> 2010-02-19 12:26:21.255201 [DEBUG] switch_ivr_originate.c:3185 Originate Resulted in Error Cause: 27 [DESTINATION_OUT_OF_ORDER]
> 2010-02-19 12:26:21.255201 [INFO] mod_dptools.c:2353 Originate Failed.  Cause: DESTINATION_OUT_OF_ORDER
> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:590 (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State REPORTING
> 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100222/10e660e2/attachment.html 

From phunk000 at hotmail.com  Mon Feb 22 06:19:13 2010
From: phunk000 at hotmail.com (phunk000)
Date: Mon, 22 Feb 2010 06:19:13 -0800 (PST)
Subject: [Freeswitch-users] mod_nibblebill
Message-ID: <1266848353835-4612298.post@n2.nabble.com>


Hello there!  I am using the fusionPBX freeSWITCH installation.  I am
attempting to setup the mod_nibblebill module to keep track of billing using
a database.  I have installed ODBC, mod_spidermonkey, and nibblebill appears
to be working, except for the blank line after "last successful billing time
was____" based on the freeSWITCH log as follows:


2010-02-22 08:46:55.535673 [DEBUG] switch_ivr_bridge.c:131
sofia/internal/sip:3008 at 192.168.15.176:42866 receive message [DISPLAY]
2010-02-22 08:46:55.555674 [DEBUG] switch_ivr_bridge.c:131
sofia/internal/3007 at 192.168.15.177 receive message [DISPLAY]
2010-02-22 08:47:23.735442 [DEBUG] mod_nibblebill.c:529 Received request via
SESSION_HEARTBEAT!
2010-02-22 08:47:23.735442 [DEBUG] mod_nibblebill.c:397 Attempting to bill
at $1.0 per minute to account 3007
2010-02-22 08:47:23.735442 [INFO] mod_nibblebill.c:447 Beginning new billing
on 68cde83c-26f4-44e3-8060-4188a106ff51
2010-02-22 08:47:23.735442 [DEBUG] mod_nibblebill.c:455 28 seconds passed
since last bill time of 2010-02-22 08:46:55
2010-02-22 08:47:23.735442 [DEBUG] mod_nibblebill.c:461 Billing $0.474338 to
3007 (Call: 68cde83c-26f4-44e3-8060-4188a106ff51 / 0.000000 so far)
2010-02-22 08:47:42.701637 [DEBUG] switch_core_sqldb.c:111 Dropping idle DB
connection db="core";user="";pass="";thread="3070618512"
2010-02-22 08:47:42.701637 [DEBUG] switch_core_sqldb.c:111 Dropping idle DB
connection db="sofia_reg_internal-ipv6";user="";pass="";thread="3070618512"
2010-02-22 08:47:42.701637 [DEBUG] switch_core_sqldb.c:111 Dropping idle DB
connection db="sofia_reg_external";user="";pass="";thread="3070372752"
2010-02-22 08:47:53.875324 [DEBUG] mod_nibblebill.c:529 Received request via
SESSION_HEARTBEAT!
2010-02-22 08:47:53.875324 [DEBUG] mod_nibblebill.c:397 Attempting to bill
at $1.0 per minute to account 3007
2010-02-22 08:47:53.875324 [DEBUG] mod_nibblebill.c:449 Last successful
billing time was 
2010-02-22 08:47:53.875324 [DEBUG] mod_nibblebill.c:455 30 seconds passed
since last bill time of 2010-02-22 08:47:23
2010-02-22 08:47:53.875324 [DEBUG] mod_nibblebill.c:461 Billing $0.502331 to
3007 (Call: 68cde83c-26f4-44e3-8060-4188a106ff51 / 0.474338 so far)
2010-02-22 08:48:23.995212 [DEBUG] mod_nibblebill.c:529 Received request via
SESSION_HEARTBEAT!
2010-02-22 08:48:23.995212 [DEBUG] mod_nibblebill.c:397 Attempting to bill
at $1.0 per minute to account 3007
2010-02-22 08:48:23.995212 [DEBUG] mod_nibblebill.c:449 Last successful
billing time was 
2010-02-22 08:48:23.995212 [DEBUG] mod_nibblebill.c:455 30 seconds passed
since last bill time of 2010-02-22 08:47:53
2010-02-22 08:48:23.995212 [DEBUG] mod_nibblebill.c:461 Billing $0.501998 to
3007 (Call: 68cde83c-26f4-44e3-8060-4188a106ff51 / 0.976670 so far)
2010-02-22 08:48:54.115099 [DEBUG] mod_nibblebill.c:529 Received request via
SESSION_HEARTBEAT!
2010-02-22 08:48:54.115099 [DEBUG] mod_nibblebill.c:397 Attempting to bill
at $1.0 per minute to account 3007
2010-02-22 08:48:54.115099 [DEBUG] mod_nibblebill.c:449 Last successful
billing time was 
2010-02-22 08:48:54.115099 [DEBUG] mod_nibblebill.c:455 30 seconds passed
since last bill time of 2010-02-22 08:48:23
2010-02-22 08:48:54.115099 [DEBUG] mod_nibblebill.c:461 Billing $0.501998 to
3007 (Call: 68cde83c-26f4-44e3-8060-4188a106ff51 / 1.478668 so far)
2010-02-22 08:49:12.271231 [NOTICE] sofia.c:329 Hangup
sofia/internal/3007 at 192.168.15.177 [CS_EXECUTE] [NORMAL_CLEARING]

Any help in regards of how to get nibblebill to access the database properly
would be great.  Again I have installed ODBC and created the accounts table
with the required fields.  Thanks a ton


-----
Todd who is to learn
-- 
View this message in context: http://n2.nabble.com/mod-nibblebill-tp4612298p4612298.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From ivdreg at gmail.com  Mon Feb 22 07:11:26 2010
From: ivdreg at gmail.com (ivdreg ivdreg)
Date: Mon, 22 Feb 2010 17:11:26 +0200
Subject: [Freeswitch-users] SDP With T.38 in INVITE Problem
In-Reply-To: <24736FFE-BEEF-4D34-BBA7-77B6B6AE0EBD@jerris.com>
References: <b437a0e1002190242y2aa64696j91e8bc76f7ae2a92@mail.gmail.com>
	<b437a0e1002220448g3e3bda9fpa283e93a11d319f1@mail.gmail.com>
	<24736FFE-BEEF-4D34-BBA7-77B6B6AE0EBD@jerris.com>
Message-ID: <b437a0e1002220711u1bfd138l8a67b4e0b024d856@mail.gmail.com>

Hi Michael,

This happens when ONLY IF initial INVITE is coming with T.38 from a GW (this
is ITSP equipment and I don't know vendor) to our SIP subscribers with ATCOM
ATA and IP Phone. We use now in production YATE for terminating and
originating GWs to ITSPs and FS as main routing logic (backend). We want to
switch YATE to FS for a GW also but we faced this problem. This not happens
if initial INVITE nave no T.38 offered and later re-INVITE with T.38 with
valid SDP port.

Thanks

2010/2/22 Michael Jerris <mike at jerris.com>

> If the endpoint does not correctly follow the sdp o/a model its not going
> to work.  This is not a "problem" with the sofia library, this is intended
> behavior and what we are supposed to do.  What is the device?
>
> Mike
>
> On Feb 22, 2010, at 7:48 AM, ivdreg ivdreg wrote:
>
> Hi All,
>
> Actually while seeking the solution in internet I see some people having
> this problem with sofia library. I'm not sure that SIP reply in this case
> contains a valid SDP (I think that teminating endpoint is broken) but in my
> opinion if we have at least one valid media type in SDP (video, audio, image
> ...) call must be established. Can someone comment and/or help me with this
> issue.
>
> Regards.
>
> 2010/2/19 ivdreg ivdreg <ivdreg at gmail.com>
>
>> Hi all,
>>
>> Dose someone have a problem that if there T.38 in coming from gateway
>> FreeSwitch drops the call because of media error ? As I see from log only
>> T.38 port is zero and SDP has also media port. Is it possible to configure
>> FS to do not break a call but if media is OK.
>>
>> 2010-02-19 12:26:14.305244 [NOTICE] switch_channel.c:666 New Channel
>> sofia/backend/XXXXXXXXXX at 10.10.1.110:7065[6cd9f634-411d-df11-99ca-003048bb99cc]
>> 2010-02-19 12:26:14.305244 [DEBUG] mod_sofia.c:3369 (sofia/backend/
>> XXXXXXXXXX at 10.10.1.110:7065) State Change CS_NEW -> CS_INIT
>> 2010-02-19 12:26:14.305244 [DEBUG] switch_core_session.c:1018 Send signal
>> sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [BREAK]
>> 2010-02-19 12:26:14.305244 [DEBUG] switch_core_state_machine.c:314
>> (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) Running State Change CS_INIT
>> 2010-02-19 12:26:14.305244 [DEBUG] switch_core_state_machine.c:338
>> (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State INIT
>> 2010-02-19 12:26:14.305244 [DEBUG] mod_sofia.c:83 sofia/backend/
>> XXXXXXXXXX at 10.10.1.110:7065 SOFIA INIT
>> 2010-02-19 12:26:14.305244 [DEBUG] sofia_glue.c:1377 sofia/backend/
>> XXXXXXXXXX at 10.10.1.110:7065 Patched SDP
>> ---
>> v=0
>> o=- 3779949069 5166785777234550622 IN IP4 206.132.232.212
>> s=session
>> t=0 0
>> m=audio 21108 RTP/AVP 18 4 8 0
>> c=IN IP4 10.10.1.110
>> a=rtpmap:18 G729/8000
>> a=rtpmap:4 G723/8000
>> a=rtpmap:8 PCMA/8000
>> a=rtpmap:0 PCMU/8000
>> m=image 21108 udptl t38
>> c=IN IP4 10.10.1.110
>> a=T38FaxVersion:0
>> a=T38MaxBitRate:14400
>> a=T38FaxUdpEC:t38UDPRedundancy
>> a=T38FaxRateManagement:transferredTCF
>>
>> +++
>> v=0
>> o=- 3779949069 5166785777234550622 IN IP4 206.132.232.212
>> s=session
>> t=0 0
>> m=audio 17058 RTP/AVP 18 4 8 0
>> c=IN IP4 10.10.1.100
>> a=rtpmap:18 G729/8000
>> a=rtpmap:4 G723/8000
>> a=rtpmap:8 PCMA/8000
>> a=rtpmap:0 PCMU/8000
>> m=image 17058 udptl t38
>> c=IN IP4 10.10.1.100
>> a=T38FaxVersion:0
>> a=T38MaxBitRate:14400
>> a=T38FaxUdpEC:t38UDPRedundancy
>> a=T38FaxRateManagement:transferredTCF
>>
>> 2010-02-19 12:26:14.305244 [DEBUG] mod_sofia.c:117 (sofia/backend/
>> XXXXXXXXXX at 10.10.1.110:7065) State Change CS_INIT -> CS_ROUTING
>> ......
>> 2010-02-19 12:26:21.255201 [DEBUG] sofia.c:4135 Remote SDP:
>> v=0
>> o=FreeSWITCH 1266548331 1266548332 IN IP4 10.10.1.110
>> s=FreeSWITCH
>> c=IN IP4 10.10.1.110
>> t=0 0
>> *m=audio 26850 RTP/AVP 8*
>> a=rtpmap:8 PCMA/8000
>> a=silenceSupp:off - - - -
>> a=ptime:20
>> *m=image 0 udptl 19*
>>
>> 2010-02-19 12:26:21.255201 [DEBUG] sofia.c:4124 Channel sofia/backend/
>> XXXXXXXXXX at 10.10.1.110:7065 entering state [ready][200]
>> 2010-02-19 12:26:21.255201 [DEBUG] switch_channel.c:2285 Send signal
>> sofia/backend/YYYYYYYYYY at 10.10.1.110 [BREAK]
>> 2010-02-19 12:26:21.255201 [NOTICE] sofia.c:4668 Channel [sofia/backend/
>> XXXXXXXXXX at 10.10.1.110:7065] has been answered
>> 2010-02-19 12:26:21.255201 [DEBUG] sofia_glue.c:2288 Set Codec
>> sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 PROXY/8000 20 ms 160 samples
>> *2010-02-19 12:26:21.255201 [DEBUG] sofia_glue.c:2571 PROXY AUDIO RTP
>> [sofia/backend/XXXXXXXXXX at 10.10.1.110:7065] 10.10.1.100:17058->
>> 10.10.1.110:0 codec: 0 ms: 20
>> 2010-02-19 12:26:21.255201 [ERR] sofia_glue.c:2879 AUDIO RTP REPORTS
>> ERROR: [Missing remote port]
>> 2010-02-19 12:26:21.255201 [NOTICE] sofia_glue.c:2880 Hangup
>> sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [CS_CONSUME_MEDIA]
>> [DESTINATION_OUT_OF_ORDER]*
>> 2010-02-19 12:26:21.255201 [DEBUG] switch_channel.c:2063 Send signal
>> sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [KILL]
>> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_session.c:1018 Send signal
>> sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [BREAK]
>> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:314
>> (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) Running State Change
>> CS_HANGUP
>> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:499
>> (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State HANGUP
>> 2010-02-19 12:26:21.255201 [DEBUG] mod_sofia.c:411 Channel sofia/backend/
>> XXXXXXXXXX at 10.10.1.110:7065 hanging up, cause: DESTINATION_OUT_OF_ORDER
>> 2010-02-19 12:26:21.255201 [DEBUG] mod_sofia.c:454 Sending BYE to
>> sofia/backend/XXXXXXXXXX at 10.10.1.110:7065
>> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:46
>> sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 Standard HANGUP, cause:
>> DESTINATION_OUT_OF_ORDER
>> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:499
>> (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State HANGUP going to sleep
>> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:333
>> (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State Change CS_HANGUP ->
>> CS_REPORTING
>> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_session.c:1018 Send signal
>> sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [BREAK]
>> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:314
>> (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) Running State Change
>> CS_REPORTING
>> 2010-02-19 12:26:21.255201 [DEBUG] switch_ivr_originate.c:3185 Originate
>> Resulted in Error Cause: 27 [DESTINATION_OUT_OF_ORDER]
>> 2010-02-19 12:26:21.255201 [INFO] mod_dptools.c:2353 Originate Failed.
>> Cause: DESTINATION_OUT_OF_ORDER
>> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:590
>> (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State REPORTING
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100222/94b5f65c/attachment.html 

From tculjaga at gmail.com  Mon Feb 22 07:23:22 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Mon, 22 Feb 2010 16:23:22 +0100
Subject: [Freeswitch-users] Freeswitch and mod_opal
In-Reply-To: <BDB26458EDF74CF6A382375739AF86BD@gamar>
References: <BDB26458EDF74CF6A382375739AF86BD@gamar>
Message-ID: <65d96fc81002220723o7186d29dr89edeb1be7ead647@mail.gmail.com>

On Mon, Feb 22, 2010 at 2:30 PM, Gilbert Amar <gamar at center.com> wrote:

> Thank you Tihomir
>
>
> We tried to build the svn trunk of mod_h323 and succeded but got a core
> dump
> a the first call hanging up.
>


strange, what h323plus and ptlib version are you using ?

can you post some logs on pastebin and add a link here ?


>
> >mod_h323
> >you need to build h323plus & ptlib before you build the module itself.
> >check http://wiki.freeswitch.org/wiki/Mod_h323 it works out of the box.
> >regarding you call setup, faststart = true, h245tuneling = true,
> >h245InSetup = false
> >It has to work.... if not post the logs here.
>
> >T.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100222/306ec349/attachment.html 

From mike at jerris.com  Mon Feb 22 07:29:22 2010
From: mike at jerris.com (Michael Jerris)
Date: Mon, 22 Feb 2010 10:29:22 -0500
Subject: [Freeswitch-users] SDP With T.38 in INVITE Problem
In-Reply-To: <b437a0e1002220711u1bfd138l8a67b4e0b024d856@mail.gmail.com>
References: <b437a0e1002190242y2aa64696j91e8bc76f7ae2a92@mail.gmail.com>
	<b437a0e1002220448g3e3bda9fpa283e93a11d319f1@mail.gmail.com>
	<24736FFE-BEEF-4D34-BBA7-77B6B6AE0EBD@jerris.com>
	<b437a0e1002220711u1bfd138l8a67b4e0b024d856@mail.gmail.com>
Message-ID: <93769c21002220729q5bf36eadk80726aa8c00e3bd@mail.gmail.com>

if you want to see what is going on, crank up the debug in freeswitch and
sofia and you should see exactly what is going on.

Mike

On Mon, Feb 22, 2010 at 10:11 AM, ivdreg ivdreg <ivdreg at gmail.com> wrote:

> Hi Michael,
>
> This happens when ONLY IF initial INVITE is coming with T.38 from a GW
> (this is ITSP equipment and I don't know vendor) to our SIP subscribers with
> ATCOM ATA and IP Phone. We use now in production YATE for terminating and
> originating GWs to ITSPs and FS as main routing logic (backend). We want to
> switch YATE to FS for a GW also but we faced this problem. This not happens
> if initial INVITE nave no T.38 offered and later re-INVITE with T.38 with
> valid SDP port.
>
> Thanks
>
> 2010/2/22 Michael Jerris <mike at jerris.com>
>
>> If the endpoint does not correctly follow the sdp o/a model its not going
>> to work.  This is not a "problem" with the sofia library, this is intended
>> behavior and what we are supposed to do.  What is the device?
>>
>> Mike
>>
>> On Feb 22, 2010, at 7:48 AM, ivdreg ivdreg wrote:
>>
>> Hi All,
>>
>> Actually while seeking the solution in internet I see some people having
>> this problem with sofia library. I'm not sure that SIP reply in this case
>> contains a valid SDP (I think that teminating endpoint is broken) but in my
>> opinion if we have at least one valid media type in SDP (video, audio, image
>> ...) call must be established. Can someone comment and/or help me with this
>> issue.
>>
>> Regards.
>>
>> 2010/2/19 ivdreg ivdreg <ivdreg at gmail.com>
>>
>>> Hi all,
>>>
>>> Dose someone have a problem that if there T.38 in coming from gateway
>>> FreeSwitch drops the call because of media error ? As I see from log only
>>> T.38 port is zero and SDP has also media port. Is it possible to configure
>>> FS to do not break a call but if media is OK.
>>>
>>> 2010-02-19 12:26:14.305244 [NOTICE] switch_channel.c:666 New Channel
>>> sofia/backend/XXXXXXXXXX at 10.10.1.110:7065[6cd9f634-411d-df11-99ca-003048bb99cc]
>>> 2010-02-19 12:26:14.305244 [DEBUG] mod_sofia.c:3369 (sofia/backend/
>>> XXXXXXXXXX at 10.10.1.110:7065) State Change CS_NEW -> CS_INIT
>>> 2010-02-19 12:26:14.305244 [DEBUG] switch_core_session.c:1018 Send signal
>>> sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [BREAK]
>>> 2010-02-19 12:26:14.305244 [DEBUG] switch_core_state_machine.c:314
>>> (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) Running State Change CS_INIT
>>> 2010-02-19 12:26:14.305244 [DEBUG] switch_core_state_machine.c:338
>>> (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State INIT
>>> 2010-02-19 12:26:14.305244 [DEBUG] mod_sofia.c:83 sofia/backend/
>>> XXXXXXXXXX at 10.10.1.110:7065 SOFIA INIT
>>> 2010-02-19 12:26:14.305244 [DEBUG] sofia_glue.c:1377 sofia/backend/
>>> XXXXXXXXXX at 10.10.1.110:7065 Patched SDP
>>> ---
>>> v=0
>>> o=- 3779949069 5166785777234550622 IN IP4 206.132.232.212
>>> s=session
>>> t=0 0
>>> m=audio 21108 RTP/AVP 18 4 8 0
>>> c=IN IP4 10.10.1.110
>>> a=rtpmap:18 G729/8000
>>> a=rtpmap:4 G723/8000
>>> a=rtpmap:8 PCMA/8000
>>> a=rtpmap:0 PCMU/8000
>>> m=image 21108 udptl t38
>>> c=IN IP4 10.10.1.110
>>> a=T38FaxVersion:0
>>> a=T38MaxBitRate:14400
>>> a=T38FaxUdpEC:t38UDPRedundancy
>>> a=T38FaxRateManagement:transferredTCF
>>>
>>> +++
>>> v=0
>>> o=- 3779949069 5166785777234550622 IN IP4 206.132.232.212
>>> s=session
>>> t=0 0
>>> m=audio 17058 RTP/AVP 18 4 8 0
>>> c=IN IP4 10.10.1.100
>>> a=rtpmap:18 G729/8000
>>> a=rtpmap:4 G723/8000
>>> a=rtpmap:8 PCMA/8000
>>> a=rtpmap:0 PCMU/8000
>>> m=image 17058 udptl t38
>>> c=IN IP4 10.10.1.100
>>> a=T38FaxVersion:0
>>> a=T38MaxBitRate:14400
>>> a=T38FaxUdpEC:t38UDPRedundancy
>>> a=T38FaxRateManagement:transferredTCF
>>>
>>> 2010-02-19 12:26:14.305244 [DEBUG] mod_sofia.c:117 (sofia/backend/
>>> XXXXXXXXXX at 10.10.1.110:7065) State Change CS_INIT -> CS_ROUTING
>>> ......
>>> 2010-02-19 12:26:21.255201 [DEBUG] sofia.c:4135 Remote SDP:
>>> v=0
>>> o=FreeSWITCH 1266548331 1266548332 IN IP4 10.10.1.110
>>> s=FreeSWITCH
>>> c=IN IP4 10.10.1.110
>>> t=0 0
>>> *m=audio 26850 RTP/AVP 8*
>>> a=rtpmap:8 PCMA/8000
>>> a=silenceSupp:off - - - -
>>> a=ptime:20
>>> *m=image 0 udptl 19*
>>>
>>> 2010-02-19 12:26:21.255201 [DEBUG] sofia.c:4124 Channel sofia/backend/
>>> XXXXXXXXXX at 10.10.1.110:7065 entering state [ready][200]
>>> 2010-02-19 12:26:21.255201 [DEBUG] switch_channel.c:2285 Send signal
>>> sofia/backend/YYYYYYYYYY at 10.10.1.110 [BREAK]
>>> 2010-02-19 12:26:21.255201 [NOTICE] sofia.c:4668 Channel [sofia/backend/
>>> XXXXXXXXXX at 10.10.1.110:7065] has been answered
>>> 2010-02-19 12:26:21.255201 [DEBUG] sofia_glue.c:2288 Set Codec
>>> sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 PROXY/8000 20 ms 160 samples
>>> *2010-02-19 12:26:21.255201 [DEBUG] sofia_glue.c:2571 PROXY AUDIO RTP
>>> [sofia/backend/XXXXXXXXXX at 10.10.1.110:7065] 10.10.1.100:17058->
>>> 10.10.1.110:0 codec: 0 ms: 20
>>> 2010-02-19 12:26:21.255201 [ERR] sofia_glue.c:2879 AUDIO RTP REPORTS
>>> ERROR: [Missing remote port]
>>> 2010-02-19 12:26:21.255201 [NOTICE] sofia_glue.c:2880 Hangup
>>> sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [CS_CONSUME_MEDIA]
>>> [DESTINATION_OUT_OF_ORDER]*
>>> 2010-02-19 12:26:21.255201 [DEBUG] switch_channel.c:2063 Send signal
>>> sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [KILL]
>>> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_session.c:1018 Send signal
>>> sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [BREAK]
>>> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:314
>>> (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) Running State Change
>>> CS_HANGUP
>>> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:499
>>> (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State HANGUP
>>> 2010-02-19 12:26:21.255201 [DEBUG] mod_sofia.c:411 Channel sofia/backend/
>>> XXXXXXXXXX at 10.10.1.110:7065 hanging up, cause: DESTINATION_OUT_OF_ORDER
>>> 2010-02-19 12:26:21.255201 [DEBUG] mod_sofia.c:454 Sending BYE to
>>> sofia/backend/XXXXXXXXXX at 10.10.1.110:7065
>>> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:46
>>> sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 Standard HANGUP, cause:
>>> DESTINATION_OUT_OF_ORDER
>>> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:499
>>> (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State HANGUP going to sleep
>>> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:333
>>> (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State Change CS_HANGUP ->
>>> CS_REPORTING
>>> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_session.c:1018 Send signal
>>> sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [BREAK]
>>> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:314
>>> (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) Running State Change
>>> CS_REPORTING
>>> 2010-02-19 12:26:21.255201 [DEBUG] switch_ivr_originate.c:3185 Originate
>>> Resulted in Error Cause: 27 [DESTINATION_OUT_OF_ORDER]
>>> 2010-02-19 12:26:21.255201 [INFO] mod_dptools.c:2353 Originate Failed.
>>> Cause: DESTINATION_OUT_OF_ORDER
>>> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:590
>>> (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State REPORTING
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100222/0825dfa9/attachment-0001.html 

From ederwander at gmail.com  Mon Feb 22 07:58:18 2010
From: ederwander at gmail.com (Eder Souza)
Date: Mon, 22 Feb 2010 12:58:18 -0300
Subject: [Freeswitch-users] Dial String Inject in FreeSwitch
Message-ID: <622bedea1002220758r3fb59040g4a12af7996d289b5@mail.gmail.com>

http://ederwander.wordpress.com/2010/02/22/dial-string-inject-in-freeswitch/

just for yours informations i write this article my test for injections in
freesitch

version of my tests

freeswitch at internal> version
FreeSWITCH Version 1.0.5-20100218-0400 (hacked)
freeswitch at internal>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100222/8c929f9a/attachment.html 

From anthony.minessale at gmail.com  Mon Feb 22 08:19:21 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 22 Feb 2010 10:19:21 -0600
Subject: [Freeswitch-users] Dial String Inject in FreeSwitch
In-Reply-To: <622bedea1002220758r3fb59040g4a12af7996d289b5@mail.gmail.com>
References: <622bedea1002220758r3fb59040g4a12af7996d289b5@mail.gmail.com>
Message-ID: <191c3a031002220819j462ee4c1v3e573124bf161d5b@mail.gmail.com>

Please do not use our project to try to make your blog more popular.

Your example requires you to prepare an intentional specific extension on
the FreeSWITCH custom made for your attack. It?s like saying if you leave
your door wide open at your house and call and tell someone, they can come
and rob you at 8:30.

This extension is also vulnerable ?by virtue of the stupidity of the
composer?

<extension name=?please-hack-me?/>
  <condition>
   <action application=?system? data=?${destination_number}?/>
  </condition>
</extension>

You should not allow tainted data from outside system to be fed directly
into your code. There is a regex system in place to extract legitimate data
from the user tainted input and safeguard against this.





On Mon, Feb 22, 2010 at 9:58 AM, Eder Souza <ederwander at gmail.com> wrote:

>
> http://ederwander.wordpress.com/2010/02/22/dial-string-inject-in-freeswitch/
>
> just for yours informations i write this article my test for injections in
> freesitch
>
> version of my tests
>
> freeswitch at internal> version
> FreeSWITCH Version 1.0.5-20100218-0400 (hacked)
> freeswitch at internal>
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100222/9052a7e0/attachment.html 

From ederwander at gmail.com  Mon Feb 22 08:33:32 2010
From: ederwander at gmail.com (Eder Souza)
Date: Mon, 22 Feb 2010 13:33:32 -0300
Subject: [Freeswitch-users] Dial String Inject in FreeSwitch
In-Reply-To: <191c3a031002220819j462ee4c1v3e573124bf161d5b@mail.gmail.com>
References: <622bedea1002220758r3fb59040g4a12af7996d289b5@mail.gmail.com>
	<191c3a031002220819j462ee4c1v3e573124bf161d5b@mail.gmail.com>
Message-ID: <622bedea1002220833q2a14a9fbh280d0944f5b86fee@mail.gmail.com>

Antony i dont see why ??


this is just one alert for all comunity of danger in the use of regular
expression (.*) or (.*) ...

many peoples can make dialplans witch use of this expressions ...






On Mon, Feb 22, 2010 at 1:19 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> Please do not use our project to try to make your blog more popular.
>
> Your example requires you to prepare an intentional specific extension on
> the FreeSWITCH custom made for your attack. It?s like saying if you leave
> your door wide open at your house and call and tell someone, they can come
> and rob you at 8:30.
>
> This extension is also vulnerable ?by virtue of the stupidity of the
> composer?
>
> <extension name=?please-hack-me?/>
>   <condition>
>    <action application=?system? data=?${destination_number}?/>
>   </condition>
> </extension>
>
> You should not allow tainted data from outside system to be fed directly
> into your code. There is a regex system in place to extract legitimate data
> from the user tainted input and safeguard against this.
>
>
>
>
>
>   On Mon, Feb 22, 2010 at 9:58 AM, Eder Souza <ederwander at gmail.com>wrote:
>
>>
>> http://ederwander.wordpress.com/2010/02/22/dial-string-inject-in-freeswitch/
>>
>> just for yours informations i write this article my test for injections in
>> freesitch
>>
>> version of my tests
>>
>> freeswitch at internal> version
>> FreeSWITCH Version 1.0.5-20100218-0400 (hacked)
>> freeswitch at internal>
>>
>>
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100222/9aa03884/attachment.html 

From brian at freeswitch.org  Mon Feb 22 08:39:37 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 22 Feb 2010 10:39:37 -0600
Subject: [Freeswitch-users] Dial String Inject in FreeSwitch
In-Reply-To: <622bedea1002220833q2a14a9fbh280d0944f5b86fee@mail.gmail.com>
References: <622bedea1002220758r3fb59040g4a12af7996d289b5@mail.gmail.com>
	<191c3a031002220819j462ee4c1v3e573124bf161d5b@mail.gmail.com>
	<622bedea1002220833q2a14a9fbh280d0944f5b86fee@mail.gmail.com>
Message-ID: <B2819BAB-7337-460C-BFF0-E0F7242A6D70@freeswitch.org>

And many people that own guns end up shooting themselves too.

/b

On Feb 22, 2010, at 10:33 AM, Eder Souza wrote:

> many peoples can make dialplans witch use of this expressions ...



From steveu at coppice.org  Mon Feb 22 08:41:39 2010
From: steveu at coppice.org (Steve Underwood)
Date: Tue, 23 Feb 2010 00:41:39 +0800
Subject: [Freeswitch-users] Dial String Inject in FreeSwitch
In-Reply-To: <622bedea1002220758r3fb59040g4a12af7996d289b5@mail.gmail.com>
References: <622bedea1002220758r3fb59040g4a12af7996d289b5@mail.gmail.com>
Message-ID: <4B82B3C3.9090007@coppice.org>

On 02/22/2010 11:58 PM, Eder Souza wrote:
> http://ederwander.wordpress.com/2010/02/22/dial-string-inject-in-freeswitch/
> just for yours informations i write this article my test for 
> injections in freesitch
> version of my tests
> freeswitch at internal <mailto:freeswitch at internal>> version
> FreeSWITCH Version 1.0.5-20100218-0400 (hacked)
> freeswitch at internal <mailto:freeswitch at internal>>
>
Leaving your car unlocked, the keys in the ignition, and a big sign on 
the windscreen saying "THIS CAR IS UNLOCKED" has a comparable effect. 
This is a sleazy way to get page hits. Someone really should create a 
wordpress blacklist.

Steve



From anthony.minessale at gmail.com  Mon Feb 22 08:42:13 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 22 Feb 2010 10:42:13 -0600
Subject: [Freeswitch-users] Question about sofia_contact
In-Reply-To: <4B8249BF.3090708@ewetel.de>
References: <4B7EA954.30402@ewetel.de> <4B7EADBC.1040001@ewetel.de>
	<4B8249BF.3090708@ewetel.de>
Message-ID: <191c3a031002220842j5bec442an5f1ea89cb0e8a6ff@mail.gmail.com>

it's mad at you for asking twice before waiting for a reply, so it's not
working on purpose.

Actually it's mad at you because your domain does not contain a . so it is
assuming you are specifying a profile name as the domain.  if your domain
was mydomain.com instead it would work.


On Mon, Feb 22, 2010 at 3:09 AM, Helmut Kuper <helmut.kuper at ewetel.de>wrote:

> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> Hello,
>
> has anybody an idea?
>
> regards
> helmut
>
>
> On 19.02.2010 16:26, Helmut Kuper wrote:
> > Hi,
> >
> > an update:
> > The corresponding select statement looks for sip_user="2701" and
> > sip_host="internal" in registration table.
> > This delivers of course no result because 2701 is registered with
> > sip_host="mydomain".
> >
> >
> > Hm any workaround or am I going in a wrong direction?
> >
> >
> > regards
> > Helmut
> >
> >
> > On 19.02.2010 16:08, Helmut Kuper wrote:
> >> Hello,
> >
> >> I try to setup a FS sofia sip-profile which allows me to have multiple
> >> sip-profiles but one registration database. So I set the following
> >> parameters:
> >
> >> <param name="challenge-realm" value="auto_from"/>
> >> <param name="force-register-domain" value="$${domain}"/>
> >> <param name="force-subscription-domain" value="$${domain}"/>
> >> <param name="force-register-db-domain" value="$${domain}"/>
> >
> >> where domain is set to "mydomain". "sofia status profile internal"
> >> delivers the following:
> >
> >
> >> Call-ID:        3c26705038e5-vwlg8u5q9cwe
> >> User:           2701 at mydomain
> >> Contact:        <sip:2701 at 85.16.245.208:1024;line=l2wksedo>
> >> Agent:          snom370/8.2.22
> >> Status:         Registered(UDP)(unknown) EXP(2010-02-19 16:13:31)
> >> Host:           ippbx-prod-node0
> >> IP:             85.16.245.208
> >> Port:           1024
> >> Auth-User:      2701
> >> Auth-Realm:     mydomain
> >> MWI-Account:    2701 at mydomain
> >
> >
> >
> >> sofia_contact internal/2701 at mydomain delivers this:
> >> error/user_not_registered
> >
> >> The Phone is fully functional.
> >
> >> I use SVN trunk 16601
> >
> >> regards
> >> Helmut
>
> -----BEGIN PGP SIGNATURE-----
> Version: GnuPG v1.4.7 (MingW32)
>
> iD8DBQFLgkm/4tZeNddg3dwRApfoAKCiX8fX/WNrZ7GXRrBJA54+VTThmACfT0d3
> fBzQlyVObkJaLHxJbfUjZG4=
> =uhHR
> -----END PGP SIGNATURE-----
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100222/fd4e782f/attachment-0001.html 

From anthony.minessale at gmail.com  Mon Feb 22 08:47:43 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 22 Feb 2010 10:47:43 -0600
Subject: [Freeswitch-users] Dial String Inject in FreeSwitch
In-Reply-To: <622bedea1002220833q2a14a9fbh280d0944f5b86fee@mail.gmail.com>
References: <622bedea1002220758r3fb59040g4a12af7996d289b5@mail.gmail.com>
	<191c3a031002220819j462ee4c1v3e573124bf161d5b@mail.gmail.com>
	<622bedea1002220833q2a14a9fbh280d0944f5b86fee@mail.gmail.com>
Message-ID: <191c3a031002220847pa6ca94ey94fa526dad3583d5@mail.gmail.com>

To me it sounds like a way to sound the alarms and bring negative attention.

For instance, if you were sincerely concerned, you could have told us about
your discovery privately first, and we could feature a story on our own site
warning people of this danger and reminding them how to compose extension
properly.

The posting was instead made like a big public announcement calling our
software "imperfect".
Yes it is imperfect, It can't properly detect someone being a moron 100% of
the time but it sure tries it's darndest.




On Mon, Feb 22, 2010 at 10:33 AM, Eder Souza <ederwander at gmail.com> wrote:

> Antony i dont see why ??
>
>
> this is just one alert for all comunity of danger in the use of regular
> expression (.*) or (.*) ...
>
> many peoples can make dialplans witch use of this expressions ...
>
>
>
>
>
>
> On Mon, Feb 22, 2010 at 1:19 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> Please do not use our project to try to make your blog more popular.
>>
>> Your example requires you to prepare an intentional specific extension on
>> the FreeSWITCH custom made for your attack. It?s like saying if you leave
>> your door wide open at your house and call and tell someone, they can come
>> and rob you at 8:30.
>>
>> This extension is also vulnerable ?by virtue of the stupidity of the
>> composer?
>>
>> <extension name=?please-hack-me?/>
>>   <condition>
>>    <action application=?system? data=?${destination_number}?/>
>>   </condition>
>> </extension>
>>
>> You should not allow tainted data from outside system to be fed directly
>> into your code. There is a regex system in place to extract legitimate data
>> from the user tainted input and safeguard against this.
>>
>>
>>
>>
>>
>>   On Mon, Feb 22, 2010 at 9:58 AM, Eder Souza <ederwander at gmail.com>wrote:
>>
>>>
>>> http://ederwander.wordpress.com/2010/02/22/dial-string-inject-in-freeswitch/
>>>
>>> just for yours informations i write this article my test for injections
>>> in freesitch
>>>
>>> version of my tests
>>>
>>> freeswitch at internal> version
>>> FreeSWITCH Version 1.0.5-20100218-0400 (hacked)
>>> freeswitch at internal>
>>>
>>>
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100222/187bcb50/attachment.html 

From ivdreg at gmail.com  Mon Feb 22 08:49:07 2010
From: ivdreg at gmail.com (ivdreg ivdreg)
Date: Mon, 22 Feb 2010 18:49:07 +0200
Subject: [Freeswitch-users] SDP With T.38 in INVITE Problem
In-Reply-To: <93769c21002220729q5bf36eadk80726aa8c00e3bd@mail.gmail.com>
References: <b437a0e1002190242y2aa64696j91e8bc76f7ae2a92@mail.gmail.com>
	<b437a0e1002220448g3e3bda9fpa283e93a11d319f1@mail.gmail.com>
	<24736FFE-BEEF-4D34-BBA7-77B6B6AE0EBD@jerris.com>
	<b437a0e1002220711u1bfd138l8a67b4e0b024d856@mail.gmail.com>
	<93769c21002220729q5bf36eadk80726aa8c00e3bd@mail.gmail.com>
Message-ID: <b437a0e1002220849k16fd87ffl5a7257d80e43b036@mail.gmail.com>

Hi Michael,

As I said in a previous mails I know exactly what is happening.
In working setup:

ITSP ---> YATE (GW - Frontend) ---> FreeSwitch (routing server/xml_curl)
---> YATE (GW - SIP Interop) ---> OpenSIPS ---> Subscriber.

I prefer to change YATE (GW - Frontend) and YATE (GW - SIP Interop) with
FreeSwitch for some reasons. The problem is:

INVITE comes from ITSP and goes to subscriber via OpenSIPS. INIVITE between
FreeSwitch (routing server) and YATE (GW - SIP Interop) contains SDP:
m=audio 21108 RTP/AVP 18 4 8 0
c=IN IP4 10.10.1.110
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
m=image 21108 udptl t38
c=IN IP4 10.10.1.110
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxUdpEC:t38UDPRedundancy
a=T38FaxRateManagement:
transferredTCF

And reply 200 OK contains in SDP:
*m=audio 34788 RTP/AVP 8*
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20

Reply 200 OK SDP between YATE (GW - SIP Interop) and OpenSIPS contains in
SDP:
*m=audio 16330 RTP/AVP 8*
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
*m=image 0 udptl 19*

In this case everything works fine. Line *m=image 0 udptl 19 *is removed by
YATE.
But same test with FreeSWITCH on the place of YATE (GW - SIP Interop) *"m=image
0 udptl 19" *call brakes as you can see in my first mail.

I don't want to compare or discus YATE and FS functionality or something
else. I just see difference in behavior and because I'm not a big expert
don't know witch implementation is more accurate according standards. And
second: Is it impossible for me to upgrade all CPE so only thing I can do is
to fix it on server side. That is because I ask for a help.


Thanks to all.


2010/2/22 Michael Jerris <mike at jerris.com>

> if you want to see what is going on, crank up the debug in freeswitch and
> sofia and you should see exactly what is going on.
>
> Mike
>
>
> On Mon, Feb 22, 2010 at 10:11 AM, ivdreg ivdreg <ivdreg at gmail.com> wrote:
>
>> Hi Michael,
>>
>> This happens when ONLY IF initial INVITE is coming with T.38 from a GW
>> (this is ITSP equipment and I don't know vendor) to our SIP subscribers with
>> ATCOM ATA and IP Phone. We use now in production YATE for terminating and
>> originating GWs to ITSPs and FS as main routing logic (backend). We want to
>> switch YATE to FS for a GW also but we faced this problem. This not happens
>> if initial INVITE nave no T.38 offered and later re-INVITE with T.38 with
>> valid SDP port.
>>
>> Thanks
>>
>> 2010/2/22 Michael Jerris <mike at jerris.com>
>>
>>>  If the endpoint does not correctly follow the sdp o/a model its not
>>> going to work.  This is not a "problem" with the sofia library, this is
>>> intended behavior and what we are supposed to do.  What is the device?
>>>
>>> Mike
>>>
>>> On Feb 22, 2010, at 7:48 AM, ivdreg ivdreg wrote:
>>>
>>> Hi All,
>>>
>>> Actually while seeking the solution in internet I see some people having
>>> this problem with sofia library. I'm not sure that SIP reply in this case
>>> contains a valid SDP (I think that teminating endpoint is broken) but in my
>>> opinion if we have at least one valid media type in SDP (video, audio, image
>>> ...) call must be established. Can someone comment and/or help me with this
>>> issue.
>>>
>>> Regards.
>>>
>>> 2010/2/19 ivdreg ivdreg <ivdreg at gmail.com>
>>>
>>>> Hi all,
>>>>
>>>> Dose someone have a problem that if there T.38 in coming from gateway
>>>> FreeSwitch drops the call because of media error ? As I see from log only
>>>> T.38 port is zero and SDP has also media port. Is it possible to configure
>>>> FS to do not break a call but if media is OK.
>>>>
>>>> 2010-02-19 12:26:14.305244 [NOTICE] switch_channel.c:666 New Channel
>>>> sofia/backend/XXXXXXXXXX at 10.10.1.110:7065[6cd9f634-411d-df11-99ca-003048bb99cc]
>>>> 2010-02-19 12:26:14.305244 [DEBUG] mod_sofia.c:3369 (sofia/backend/
>>>> XXXXXXXXXX at 10.10.1.110:7065) State Change CS_NEW -> CS_INIT
>>>> 2010-02-19 12:26:14.305244 [DEBUG] switch_core_session.c:1018 Send
>>>> signal sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [BREAK]
>>>> 2010-02-19 12:26:14.305244 [DEBUG] switch_core_state_machine.c:314
>>>> (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) Running State Change
>>>> CS_INIT
>>>> 2010-02-19 12:26:14.305244 [DEBUG] switch_core_state_machine.c:338
>>>> (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State INIT
>>>> 2010-02-19 12:26:14.305244 [DEBUG] mod_sofia.c:83 sofia/backend/
>>>> XXXXXXXXXX at 10.10.1.110:7065 SOFIA INIT
>>>> 2010-02-19 12:26:14.305244 [DEBUG] sofia_glue.c:1377 sofia/backend/
>>>> XXXXXXXXXX at 10.10.1.110:7065 Patched SDP
>>>> ---
>>>> v=0
>>>> o=- 3779949069 5166785777234550622 IN IP4 206.132.232.212
>>>> s=session
>>>> t=0 0
>>>> m=audio 21108 RTP/AVP 18 4 8 0
>>>> c=IN IP4 10.10.1.110
>>>> a=rtpmap:18 G729/8000
>>>> a=rtpmap:4 G723/8000
>>>> a=rtpmap:8 PCMA/8000
>>>> a=rtpmap:0 PCMU/8000
>>>> m=image 21108 udptl t38
>>>> c=IN IP4 10.10.1.110
>>>> a=T38FaxVersion:0
>>>> a=T38MaxBitRate:14400
>>>> a=T38FaxUdpEC:t38UDPRedundancy
>>>> a=T38FaxRateManagement:transferredTCF
>>>>
>>>> +++
>>>> v=0
>>>> o=- 3779949069 5166785777234550622 IN IP4 206.132.232.212
>>>> s=session
>>>> t=0 0
>>>> m=audio 17058 RTP/AVP 18 4 8 0
>>>> c=IN IP4 10.10.1.100
>>>> a=rtpmap:18 G729/8000
>>>> a=rtpmap:4 G723/8000
>>>> a=rtpmap:8 PCMA/8000
>>>> a=rtpmap:0 PCMU/8000
>>>> m=image 17058 udptl t38
>>>> c=IN IP4 10.10.1.100
>>>> a=T38FaxVersion:0
>>>> a=T38MaxBitRate:14400
>>>> a=T38FaxUdpEC:t38UDPRedundancy
>>>> a=T38FaxRateManagement:transferredTCF
>>>>
>>>> 2010-02-19 12:26:14.305244 [DEBUG] mod_sofia.c:117 (sofia/backend/
>>>> XXXXXXXXXX at 10.10.1.110:7065) State Change CS_INIT -> CS_ROUTING
>>>> ......
>>>> 2010-02-19 12:26:21.255201 [DEBUG] sofia.c:4135 Remote SDP:
>>>> v=0
>>>> o=FreeSWITCH 1266548331 1266548332 IN IP4 10.10.1.110
>>>> s=FreeSWITCH
>>>> c=IN IP4 10.10.1.110
>>>> t=0 0
>>>> *m=audio 26850 RTP/AVP 8*
>>>> a=rtpmap:8 PCMA/8000
>>>> a=silenceSupp:off - - - -
>>>> a=ptime:20
>>>> *m=image 0 udptl 19*
>>>>
>>>> 2010-02-19 12:26:21.255201 [DEBUG] sofia.c:4124 Channel sofia/backend/
>>>> XXXXXXXXXX at 10.10.1.110:7065 entering state [ready][200]
>>>> 2010-02-19 12:26:21.255201 [DEBUG] switch_channel.c:2285 Send signal
>>>> sofia/backend/YYYYYYYYYY at 10.10.1.110 [BREAK]
>>>> 2010-02-19 12:26:21.255201 [NOTICE] sofia.c:4668 Channel [sofia/backend/
>>>> XXXXXXXXXX at 10.10.1.110:7065] has been answered
>>>> 2010-02-19 12:26:21.255201 [DEBUG] sofia_glue.c:2288 Set Codec
>>>> sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 PROXY/8000 20 ms 160 samples
>>>> *2010-02-19 12:26:21.255201 [DEBUG] sofia_glue.c:2571 PROXY AUDIO RTP
>>>> [sofia/backend/XXXXXXXXXX at 10.10.1.110:7065] 10.10.1.100:17058->
>>>> 10.10.1.110:0 codec: 0 ms: 20
>>>> 2010-02-19 12:26:21.255201 [ERR] sofia_glue.c:2879 AUDIO RTP REPORTS
>>>> ERROR: [Missing remote port]
>>>> 2010-02-19 12:26:21.255201 [NOTICE] sofia_glue.c:2880 Hangup
>>>> sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [CS_CONSUME_MEDIA]
>>>> [DESTINATION_OUT_OF_ORDER]*
>>>> 2010-02-19 12:26:21.255201 [DEBUG] switch_channel.c:2063 Send signal
>>>> sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [KILL]
>>>> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_session.c:1018 Send
>>>> signal sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [BREAK]
>>>> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:314
>>>> (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) Running State Change
>>>> CS_HANGUP
>>>> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:499
>>>> (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State HANGUP
>>>> 2010-02-19 12:26:21.255201 [DEBUG] mod_sofia.c:411 Channel
>>>> sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 hanging up, cause:
>>>> DESTINATION_OUT_OF_ORDER
>>>> 2010-02-19 12:26:21.255201 [DEBUG] mod_sofia.c:454 Sending BYE to
>>>> sofia/backend/XXXXXXXXXX at 10.10.1.110:7065
>>>> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:46
>>>> sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 Standard HANGUP, cause:
>>>> DESTINATION_OUT_OF_ORDER
>>>> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:499
>>>> (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State HANGUP going to sleep
>>>> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:333
>>>> (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State Change CS_HANGUP ->
>>>> CS_REPORTING
>>>> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_session.c:1018 Send
>>>> signal sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [BREAK]
>>>> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:314
>>>> (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) Running State Change
>>>> CS_REPORTING
>>>> 2010-02-19 12:26:21.255201 [DEBUG] switch_ivr_originate.c:3185 Originate
>>>> Resulted in Error Cause: 27 [DESTINATION_OUT_OF_ORDER]
>>>> 2010-02-19 12:26:21.255201 [INFO] mod_dptools.c:2353 Originate Failed.
>>>> Cause: DESTINATION_OUT_OF_ORDER
>>>> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:590
>>>> (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State REPORTING
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100222/e5b81361/attachment-0001.html 

From ederwander at gmail.com  Mon Feb 22 09:09:41 2010
From: ederwander at gmail.com (Eder Souza)
Date: Mon, 22 Feb 2010 14:09:41 -0300
Subject: [Freeswitch-users] Dial String Inject in FreeSwitch
In-Reply-To: <191c3a031002220847pa6ca94ey94fa526dad3583d5@mail.gmail.com>
References: <622bedea1002220758r3fb59040g4a12af7996d289b5@mail.gmail.com>
	<191c3a031002220819j462ee4c1v3e573124bf161d5b@mail.gmail.com>
	<622bedea1002220833q2a14a9fbh280d0944f5b86fee@mail.gmail.com>
	<191c3a031002220847pa6ca94ey94fa526dad3583d5@mail.gmail.com>
Message-ID: <622bedea1002220909x6694fdfdr6dd43f30b056edb2@mail.gmail.com>

i prefer FreeSwitch im left Asterisk

FreeSwitch is Very Very betther then Asterisk in my option !!


my intention is just say dont use (.*), (.+)  or combinations of this
regular expressions, for me FreeSwitch is the betther  !!




On Mon, Feb 22, 2010 at 1:47 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> To me it sounds like a way to sound the alarms and bring negative
> attention.
>
> For instance, if you were sincerely concerned, you could have told us about
> your discovery privately first, and we could feature a story on our own site
> warning people of this danger and reminding them how to compose extension
> properly.
>
> The posting was instead made like a big public announcement calling our
> software "imperfect".
> Yes it is imperfect, It can't properly detect someone being a moron 100% of
> the time but it sure tries it's darndest.
>
>
>
>
>
> On Mon, Feb 22, 2010 at 10:33 AM, Eder Souza <ederwander at gmail.com> wrote:
>
>> Antony i dont see why ??
>>
>>
>> this is just one alert for all comunity of danger in the use of regular
>> expression (.*) or (.*) ...
>>
>> many peoples can make dialplans witch use of this expressions ...
>>
>>
>>
>>
>>
>>
>> On Mon, Feb 22, 2010 at 1:19 PM, Anthony Minessale <
>> anthony.minessale at gmail.com> wrote:
>>
>>> Please do not use our project to try to make your blog more popular.
>>>
>>> Your example requires you to prepare an intentional specific extension on
>>> the FreeSWITCH custom made for your attack. It?s like saying if you leave
>>> your door wide open at your house and call and tell someone, they can come
>>> and rob you at 8:30.
>>>
>>> This extension is also vulnerable ?by virtue of the stupidity of the
>>> composer?
>>>
>>> <extension name=?please-hack-me?/>
>>>   <condition>
>>>    <action application=?system? data=?${destination_number}?/>
>>>   </condition>
>>> </extension>
>>>
>>> You should not allow tainted data from outside system to be fed directly
>>> into your code. There is a regex system in place to extract legitimate data
>>> from the user tainted input and safeguard against this.
>>>
>>>
>>>
>>>
>>>
>>>   On Mon, Feb 22, 2010 at 9:58 AM, Eder Souza <ederwander at gmail.com>wrote:
>>>
>>>>
>>>> http://ederwander.wordpress.com/2010/02/22/dial-string-inject-in-freeswitch/
>>>>
>>>> just for yours informations i write this article my test for injections
>>>> in freesitch
>>>>
>>>> version of my tests
>>>>
>>>> freeswitch at internal> version
>>>> FreeSWITCH Version 1.0.5-20100218-0400 (hacked)
>>>> freeswitch at internal>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>> iax:guest at conference.freeswitch.org/888
>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> pstn:+19193869900
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100222/ec4f6533/attachment.html 

From steveu at coppice.org  Mon Feb 22 09:10:40 2010
From: steveu at coppice.org (Steve Underwood)
Date: Tue, 23 Feb 2010 01:10:40 +0800
Subject: [Freeswitch-users] SDP With T.38 in INVITE Problem
In-Reply-To: <b437a0e1002220849k16fd87ffl5a7257d80e43b036@mail.gmail.com>
References: <b437a0e1002190242y2aa64696j91e8bc76f7ae2a92@mail.gmail.com>	<b437a0e1002220448g3e3bda9fpa283e93a11d319f1@mail.gmail.com>	<24736FFE-BEEF-4D34-BBA7-77B6B6AE0EBD@jerris.com>	<b437a0e1002220711u1bfd138l8a67b4e0b024d856@mail.gmail.com>	<93769c21002220729q5bf36eadk80726aa8c00e3bd@mail.gmail.com>
	<b437a0e1002220849k16fd87ffl5a7257d80e43b036@mail.gmail.com>
Message-ID: <4B82BA90.10709@coppice.org>

Hi Michael,

On 02/23/2010 12:49 AM, ivdreg ivdreg wrote:
> Hi Michael,
>
> As I said in a previous mails I know exactly what is happening.
> In working setup:
>
> ITSP ---> YATE (GW - Frontend) ---> FreeSwitch (routing 
> server/xml_curl) ---> YATE (GW - SIP Interop) ---> OpenSIPS ---> 
> Subscriber.
>
> I prefer to change YATE (GW - Frontend) and YATE (GW - SIP Interop) 
> with FreeSwitch for some reasons. The problem is:
>
> INVITE comes from ITSP and goes to subscriber via OpenSIPS. INIVITE 
> between FreeSwitch (routing server) and YATE (GW - SIP Interop) 
> contains SDP:
> m=audio 21108 RTP/AVP 18 4 8 0
> c=IN IP4 10.10.1.110
> a=rtpmap:18 G729/8000
> a=rtpmap:4 G723/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:0 PCMU/8000
> m=image 21108 udptl t38
> c=IN IP4 10.10.1.110
> a=T38FaxVersion:0
> a=T38MaxBitRate:14400
> a=T38FaxUdpEC:t38UDPRedundancy
> a=T38FaxRateManagement:
> transferredTCF
>
> And reply 200 OK contains in SDP:
> *m=audio 34788 RTP/AVP 8*
> a=rtpmap:8 PCMA/8000
> a=silenceSupp:off - - - -
> a=ptime:20
>
> Reply 200 OK SDP between YATE (GW - SIP Interop) and OpenSIPS contains 
> in SDP:
> *m=audio 16330 RTP/AVP 8*
> a=rtpmap:8 PCMA/8000
> a=silenceSupp:off - - - -
> a=ptime:20
> *m=image 0 udptl 19*
>
> In this case everything works fine. Line *m=image 0 udptl 19 *is 
> removed by YATE.
> But same test with FreeSWITCH on the place of YATE (GW - SIP Interop) 
> *"m=image 0 udptl 19" *call brakes as you can see in my first mail.
>
> I don't want to compare or discus YATE and FS functionality or 
> something else. I just see difference in behavior and because I'm not 
> a big expert don't know witch implementation is more accurate 
> according standards. And second: Is it impossible for me to upgrade 
> all CPE so only thing I can do is to fix it on server side. That is 
> because I ask for a help.
You said the the broken line "m=image 0 udptl 19" goes from OpenSIPS to 
YATE. Do you know if it originates from the OpenSIPS box or the 
subscriber? If it originates from the OpenSIPS box it should be reported 
to them. If its from the subscriber, well...... your chances of getting 
anything fixed are usually small.

Steve



From christian.loeschenkohl at xpirio.com  Mon Feb 22 09:16:43 2010
From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=)
Date: Mon, 22 Feb 2010 18:16:43 +0100
Subject: [Freeswitch-users] sending a sip notify with sendevent
Message-ID: <4B82BBFB.1040804@xpirio.com>

hi

i try to send a sip notify message to a registered sip device
"sofia profile nat flush_inbound_reg 10 at vts.vie.xpirio.net reboot" works, but i need
to send "check-sync;reboot=false" - so the device does a resync and don't do a reboot

my message looks like this

sendevent NOTIFY
profile: nat
event-string: check-sync;reboot=false
user: 10
host: vts.vie.xpirio.net
content-type: application/simple-message-summary

if i listen on the loopback interface i do see

##
T 2010/02/22 18:11:59.083204 127.0.0.1:51840 -> 127.0.0.1:8021 [AP]
sendevent NOTIFY
profile: nat
event-string: check-sync;reboot=false
user: 10
host: vts.vie.xpirio.net
content-type: application/simple-message-summary

##
T 2010/02/22 18:11:59.084032 127.0.0.1:8021 -> 127.0.0.1:51840 [AP]
Content-Type: command/reply
Reply-Text: -ERR invalid

--------
i don't get what it is wrong. i also rechecked the registered user in the sqlite database and this
looks good to me.

no message is send to the user.

we do use multiple domains, so user could also be 10 at somedomain.com - or am i wrong on this?
could somebody please bring some light in this.

we do use trunk rev. 16631

br



-- 
Ing. Christian L?schenkohl
Technische Leitung, Forschung & Entwicklung VoIP

xpirio
Telekommunikation & Service GmbH
Lakeside B04
9020 Klagenfurt
Austria

T  +43 (0) 5 77 11 - 1000
F  +43 (0) 5 77 11 - 1002
E  christian.loeschenkohl at xpirio.com


From m.sobkow at marketelsystems.com  Mon Feb 22 09:21:40 2010
From: m.sobkow at marketelsystems.com (Mark Sobkow)
Date: Mon, 22 Feb 2010 11:21:40 -0600
Subject: [Freeswitch-users] 8000 rate .wav files
Message-ID: <4B82BD24.2030108@marketelsystems.com>

I've got the 8000 sample rate .wav files installed for Freeswitch.  
According to the logs, my SIP phone is connecting with an 8000 rate.  
However, when I try to play_and_get_digits using those sound files, I 
get errors:

2010-02-22 11:15:22.029347 [ERR] switch_core_file.c:122 Invalid file 
format [wav] for 
[/opt/freeswitch/sounds/en/us/callie/conference/8000/conf-pin.wav]!
2010-02-22 11:15:22.029347 [ERR] switch_core_file.c:122 Invalid file 
format [wav] for 
[/opt/freeswitch/sounds/en/us/callie/conference/8000/conf-bad-pin.wav]!

Aren't .wav files supposed to be compatible with all codecs for 
playback?  If not, what do I have to do to convert them to the proper 
formats?  How do I find out what the proper formats are?

-- 
Mark Sobkow
Senior Developer
MarkeTel Multi-Line Dialing Systems LTD.
428 Victoria Ave
Regina, SK S4N-0P6
Toll-Free: 800-289-8616-X533
Local: 306-359-6893-X533
Fax: 306-359-6879
Email: m.sobkow at marketelsystems.com
Web: http://www.marketelsystems.com 



From anthony.minessale at gmail.com  Mon Feb 22 09:24:37 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 22 Feb 2010 11:24:37 -0600
Subject: [Freeswitch-users] Dial String Inject in FreeSwitch
In-Reply-To: <622bedea1002220909x6694fdfdr6dd43f30b056edb2@mail.gmail.com>
References: <622bedea1002220758r3fb59040g4a12af7996d289b5@mail.gmail.com>
	<191c3a031002220819j462ee4c1v3e573124bf161d5b@mail.gmail.com>
	<622bedea1002220833q2a14a9fbh280d0944f5b86fee@mail.gmail.com>
	<191c3a031002220847pa6ca94ey94fa526dad3583d5@mail.gmail.com>
	<622bedea1002220909x6694fdfdr6dd43f30b056edb2@mail.gmail.com>
Message-ID: <191c3a031002220924m465a74b9w7e8b60678d7a3de@mail.gmail.com>

correct,

You could write a CGI for apache too that could let someone figure out how
to download the root password.

By default, nobody should trust the data supplied by the outside user.
FreeSWITCH cannot do this for you or the limitations would impair desired
functionality.

All you have to do is look for a digit sequence in your dial string.
Moreover you need to make sure even then that it's safe to pass this digit
string to the provider.
Here in USA we share the 1 country code with several other countries that
could cost 50 cents to a dollar a minute.
So you are not even safe when you made sure it's a number.



On Mon, Feb 22, 2010 at 11:09 AM, Eder Souza <ederwander at gmail.com> wrote:

> i prefer FreeSwitch im left Asterisk
>
> FreeSwitch is Very Very betther then Asterisk in my option !!
>
>
> my intention is just say dont use (.*), (.+)  or combinations of this
> regular expressions, for me FreeSwitch is the betther  !!
>
>
>
>
> On Mon, Feb 22, 2010 at 1:47 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> To me it sounds like a way to sound the alarms and bring negative
>> attention.
>>
>> For instance, if you were sincerely concerned, you could have told us
>> about your discovery privately first, and we could feature a story on our
>> own site warning people of this danger and reminding them how to compose
>> extension properly.
>>
>> The posting was instead made like a big public announcement calling our
>> software "imperfect".
>> Yes it is imperfect, It can't properly detect someone being a moron 100%
>> of the time but it sure tries it's darndest.
>>
>>
>>
>>
>>
>> On Mon, Feb 22, 2010 at 10:33 AM, Eder Souza <ederwander at gmail.com>wrote:
>>
>>> Antony i dont see why ??
>>>
>>>
>>> this is just one alert for all comunity of danger in the use of regular
>>> expression (.*) or (.*) ...
>>>
>>> many peoples can make dialplans witch use of this expressions ...
>>>
>>>
>>>
>>>
>>>
>>>
>>> On Mon, Feb 22, 2010 at 1:19 PM, Anthony Minessale <
>>> anthony.minessale at gmail.com> wrote:
>>>
>>>> Please do not use our project to try to make your blog more popular.
>>>>
>>>> Your example requires you to prepare an intentional specific extension
>>>> on the FreeSWITCH custom made for your attack. It?s like saying if you leave
>>>> your door wide open at your house and call and tell someone, they can come
>>>> and rob you at 8:30.
>>>>
>>>> This extension is also vulnerable ?by virtue of the stupidity of the
>>>> composer?
>>>>
>>>> <extension name=?please-hack-me?/>
>>>>   <condition>
>>>>    <action application=?system? data=?${destination_number}?/>
>>>>   </condition>
>>>> </extension>
>>>>
>>>> You should not allow tainted data from outside system to be fed directly
>>>> into your code. There is a regex system in place to extract legitimate data
>>>> from the user tainted input and safeguard against this.
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>   On Mon, Feb 22, 2010 at 9:58 AM, Eder Souza <ederwander at gmail.com>wrote:
>>>>
>>>>>
>>>>> http://ederwander.wordpress.com/2010/02/22/dial-string-inject-in-freeswitch/
>>>>>
>>>>> just for yours informations i write this article my test for injections
>>>>> in freesitch
>>>>>
>>>>> version of my tests
>>>>>
>>>>> freeswitch at internal> version
>>>>> FreeSWITCH Version 1.0.5-20100218-0400 (hacked)
>>>>> freeswitch at internal>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>>
>>>> --
>>>> Anthony Minessale II
>>>>
>>>> FreeSWITCH http://www.freeswitch.org/
>>>> ClueCon http://www.cluecon.com/
>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>
>>>> AIM: anthm
>>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>> IRC: irc.freenode.net #freeswitch
>>>>
>>>> FreeSWITCH Developer Conference
>>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>>> iax:guest at conference.freeswitch.org/888
>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>> pstn:+19193869900
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100222/6c7fa474/attachment-0001.html 

From gmaruzz at celliax.org  Mon Feb 22 09:26:00 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Mon, 22 Feb 2010 18:26:00 +0100
Subject: [Freeswitch-users] Dial String Inject in FreeSwitch
In-Reply-To: <622bedea1002220909x6694fdfdr6dd43f30b056edb2@mail.gmail.com>
References: <622bedea1002220758r3fb59040g4a12af7996d289b5@mail.gmail.com> 
	<191c3a031002220819j462ee4c1v3e573124bf161d5b@mail.gmail.com> 
	<622bedea1002220833q2a14a9fbh280d0944f5b86fee@mail.gmail.com> 
	<191c3a031002220847pa6ca94ey94fa526dad3583d5@mail.gmail.com> 
	<622bedea1002220909x6694fdfdr6dd43f30b056edb2@mail.gmail.com>
Message-ID: <7b197bef1002220926k5c659d67y1b2e6b843a97afee@mail.gmail.com>

Eder,

If you fear people can do such *really stupid* things, and this is
nice from you, please add something to the wiki, for example a
paragraph in the dialplan page, or whatever, explaining why this is a
stupid thing.

If you publish a page in your blog, that look like a security alert,
or that you found a security flaw in FS, people will rightly think
that you are just looking for some attention in the search engines,
and to bring viewers to your page.

Also, in doing so, you push non technical people to think there is a
security problem in FS, and this is really a big damage to the
project. Because it is not true, it is just how it look like in your
page.

So, delete that page, and add something to the wiki, if you care about
telling people not to do stupid things.

But please, be aware that your page, the page you published, is really
something that do a damage and put a bad light on a project, and there
is no one reason for doing this.

-giovanni



On Mon, Feb 22, 2010 at 6:09 PM, Eder Souza <ederwander at gmail.com> wrote:
> i prefer FreeSwitch im left Asterisk
>
> FreeSwitch is Very Very betther then Asterisk in my option !!
>
>
> my intention is just say dont use (.*),?(.+)? or combinations of this
> regular expressions, for me FreeSwitch?is the betther??!!
>
>
>
> On Mon, Feb 22, 2010 at 1:47 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>>
>> To me it sounds like a way to sound the alarms and bring negative
>> attention.
>>
>> For instance, if you were sincerely concerned, you could have told us
>> about your discovery privately first, and we could feature a story on our
>> own site warning people of this danger and reminding them how to compose
>> extension properly.
>>
>> The posting was instead made like a big public announcement calling our
>> software "imperfect".
>> Yes it is imperfect, It can't properly detect someone being a moron 100%
>> of the time but it sure tries it's darndest.
>>
>>
>>
>>
>> On Mon, Feb 22, 2010 at 10:33 AM, Eder Souza <ederwander at gmail.com> wrote:
>>>
>>> Antony i dont see why ??
>>>
>>>
>>> this is just one alert for all comunity of danger in the use of regular
>>> expression (.*) or (.*) ...
>>>
>>> many peoples can make dialplans?witch use of this expressions ...
>>>
>>>
>>>
>>>
>>>
>>> On Mon, Feb 22, 2010 at 1:19 PM, Anthony Minessale
>>> <anthony.minessale at gmail.com> wrote:
>>>>
>>>> Please do not use our project to try to make your blog more popular.
>>>>
>>>> Your example requires you to prepare an intentional specific extension
>>>> on the FreeSWITCH custom made for your attack. It?s like saying if you leave
>>>> your door wide open at your house and call and tell someone, they can come
>>>> and rob you at 8:30.
>>>>
>>>> This extension is also vulnerable ?by virtue of the stupidity of the
>>>> composer?
>>>>
>>>> <extension name=?please-hack-me?/>
>>>> ? <condition>
>>>> ?? <action application=?system? data=?${destination_number}?/>
>>>> ? </condition>
>>>> </extension>
>>>>
>>>> You should not allow tainted data from outside system to be fed directly
>>>> into your code. There is a regex system in place to extract legitimate data
>>>> from the user tainted input and safeguard against this.
>>>>
>>>>
>>>>
>>>> On Mon, Feb 22, 2010 at 9:58 AM, Eder Souza <ederwander at gmail.com>
>>>> wrote:
>>>>>
>>>>>
>>>>> http://ederwander.wordpress.com/2010/02/22/dial-string-inject-in-freeswitch/
>>>>>
>>>>> just for yours informations i?write this article my test for injections
>>>>> in freesitch
>>>>>
>>>>> version of my tests
>>>>>
>>>>> freeswitch at internal> version
>>>>> FreeSWITCH Version 1.0.5-20100218-0400 (hacked)
>>>>> freeswitch at internal>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> Anthony Minessale II
>>>>
>>>> FreeSWITCH http://www.freeswitch.org/
>>>> ClueCon http://www.cluecon.com/
>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>
>>>> AIM: anthm
>>>> MSN:anthony_minessale at hotmail.com
>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>>> IRC: irc.freenode.net #freeswitch
>>>>
>>>> FreeSWITCH Developer Conference
>>>> sip:888 at conference.freeswitch.org
>>>> iax:guest at conference.freeswitch.org/888
>>>> googletalk:conf+888 at conference.freeswitch.org
>>>> pstn:+19193869900
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From ivdreg at gmail.com  Mon Feb 22 09:28:43 2010
From: ivdreg at gmail.com (ivdreg ivdreg)
Date: Mon, 22 Feb 2010 19:28:43 +0200
Subject: [Freeswitch-users] SDP With T.38 in INVITE Problem
In-Reply-To: <4B82BA90.10709@coppice.org>
References: <b437a0e1002190242y2aa64696j91e8bc76f7ae2a92@mail.gmail.com>
	<b437a0e1002220448g3e3bda9fpa283e93a11d319f1@mail.gmail.com>
	<24736FFE-BEEF-4D34-BBA7-77B6B6AE0EBD@jerris.com>
	<b437a0e1002220711u1bfd138l8a67b4e0b024d856@mail.gmail.com>
	<93769c21002220729q5bf36eadk80726aa8c00e3bd@mail.gmail.com>
	<b437a0e1002220849k16fd87ffl5a7257d80e43b036@mail.gmail.com>
	<4B82BA90.10709@coppice.org>
Message-ID: <b437a0e1002220928o2df6b326m6b928d78e025293e@mail.gmail.com>

It comes form subscriber (ATCOM CPEs). As you know OpenSIPS is just a proxy
so cannot generate or rewrite SDP (generally speaking). Yes, I cannot fix
CPEs only server :(

Regards

2010/2/22 Steve Underwood <steveu at coppice.org>

> Hi Michael,
>
> On 02/23/2010 12:49 AM, ivdreg ivdreg wrote:
> > Hi Michael,
> >
> > As I said in a previous mails I know exactly what is happening.
> > In working setup:
> >
> > ITSP ---> YATE (GW - Frontend) ---> FreeSwitch (routing
> > server/xml_curl) ---> YATE (GW - SIP Interop) ---> OpenSIPS --->
> > Subscriber.
> >
> > I prefer to change YATE (GW - Frontend) and YATE (GW - SIP Interop)
> > with FreeSwitch for some reasons. The problem is:
> >
> > INVITE comes from ITSP and goes to subscriber via OpenSIPS. INIVITE
> > between FreeSwitch (routing server) and YATE (GW - SIP Interop)
> > contains SDP:
> > m=audio 21108 RTP/AVP 18 4 8 0
> > c=IN IP4 10.10.1.110
> > a=rtpmap:18 G729/8000
> > a=rtpmap:4 G723/8000
> > a=rtpmap:8 PCMA/8000
> > a=rtpmap:0 PCMU/8000
> > m=image 21108 udptl t38
> > c=IN IP4 10.10.1.110
> > a=T38FaxVersion:0
> > a=T38MaxBitRate:14400
> > a=T38FaxUdpEC:t38UDPRedundancy
> > a=T38FaxRateManagement:
> > transferredTCF
> >
> > And reply 200 OK contains in SDP:
> > *m=audio 34788 RTP/AVP 8*
> > a=rtpmap:8 PCMA/8000
> > a=silenceSupp:off - - - -
> > a=ptime:20
> >
> > Reply 200 OK SDP between YATE (GW - SIP Interop) and OpenSIPS contains
> > in SDP:
> > *m=audio 16330 RTP/AVP 8*
> > a=rtpmap:8 PCMA/8000
> > a=silenceSupp:off - - - -
> > a=ptime:20
> > *m=image 0 udptl 19*
> >
> > In this case everything works fine. Line *m=image 0 udptl 19 *is
> > removed by YATE.
> > But same test with FreeSWITCH on the place of YATE (GW - SIP Interop)
> > *"m=image 0 udptl 19" *call brakes as you can see in my first mail.
> >
> > I don't want to compare or discus YATE and FS functionality or
> > something else. I just see difference in behavior and because I'm not
> > a big expert don't know witch implementation is more accurate
> > according standards. And second: Is it impossible for me to upgrade
> > all CPE so only thing I can do is to fix it on server side. That is
> > because I ask for a help.
> You said the the broken line "m=image 0 udptl 19" goes from OpenSIPS to
> YATE. Do you know if it originates from the OpenSIPS box or the
> subscriber? If it originates from the OpenSIPS box it should be reported
> to them. If its from the subscriber, well...... your chances of getting
> anything fixed are usually small.
>
> Steve
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100222/a5f50c8b/attachment.html 

From mike at jerris.com  Mon Feb 22 09:29:22 2010
From: mike at jerris.com (Michael Jerris)
Date: Mon, 22 Feb 2010 12:29:22 -0500
Subject: [Freeswitch-users] SDP With T.38 in INVITE Problem
In-Reply-To: <4B82BA90.10709@coppice.org>
References: <b437a0e1002190242y2aa64696j91e8bc76f7ae2a92@mail.gmail.com>
	<b437a0e1002220448g3e3bda9fpa283e93a11d319f1@mail.gmail.com>
	<24736FFE-BEEF-4D34-BBA7-77B6B6AE0EBD@jerris.com>
	<b437a0e1002220711u1bfd138l8a67b4e0b024d856@mail.gmail.com>
	<93769c21002220729q5bf36eadk80726aa8c00e3bd@mail.gmail.com>
	<b437a0e1002220849k16fd87ffl5a7257d80e43b036@mail.gmail.com>
	<4B82BA90.10709@coppice.org>
Message-ID: <93769c21002220929t1dbba5bcm3d9200f68a9e1800@mail.gmail.com>

the port 0 with PT of 19 is sofia rejecting the sdp becuase we don't support
it.

On Mon, Feb 22, 2010 at 12:10 PM, Steve Underwood <steveu at coppice.org>wrote:

> Hi Michael,
>
> On 02/23/2010 12:49 AM, ivdreg ivdreg wrote:
> > Hi Michael,
> >
> > As I said in a previous mails I know exactly what is happening.
> > In working setup:
> >
> > ITSP ---> YATE (GW - Frontend) ---> FreeSwitch (routing
> > server/xml_curl) ---> YATE (GW - SIP Interop) ---> OpenSIPS --->
> > Subscriber.
> >
> > I prefer to change YATE (GW - Frontend) and YATE (GW - SIP Interop)
> > with FreeSwitch for some reasons. The problem is:
> >
> > INVITE comes from ITSP and goes to subscriber via OpenSIPS. INIVITE
> > between FreeSwitch (routing server) and YATE (GW - SIP Interop)
> > contains SDP:
> > m=audio 21108 RTP/AVP 18 4 8 0
> > c=IN IP4 10.10.1.110
> > a=rtpmap:18 G729/8000
> > a=rtpmap:4 G723/8000
> > a=rtpmap:8 PCMA/8000
> > a=rtpmap:0 PCMU/8000
> > m=image 21108 udptl t38
> > c=IN IP4 10.10.1.110
> > a=T38FaxVersion:0
> > a=T38MaxBitRate:14400
> > a=T38FaxUdpEC:t38UDPRedundancy
> > a=T38FaxRateManagement:
> > transferredTCF
> >
> > And reply 200 OK contains in SDP:
> > *m=audio 34788 RTP/AVP 8*
> > a=rtpmap:8 PCMA/8000
> > a=silenceSupp:off - - - -
> > a=ptime:20
> >
> > Reply 200 OK SDP between YATE (GW - SIP Interop) and OpenSIPS contains
> > in SDP:
> > *m=audio 16330 RTP/AVP 8*
> > a=rtpmap:8 PCMA/8000
> > a=silenceSupp:off - - - -
> > a=ptime:20
> > *m=image 0 udptl 19*
> >
> > In this case everything works fine. Line *m=image 0 udptl 19 *is
> > removed by YATE.
> > But same test with FreeSWITCH on the place of YATE (GW - SIP Interop)
> > *"m=image 0 udptl 19" *call brakes as you can see in my first mail.
> >
> > I don't want to compare or discus YATE and FS functionality or
> > something else. I just see difference in behavior and because I'm not
> > a big expert don't know witch implementation is more accurate
> > according standards. And second: Is it impossible for me to upgrade
> > all CPE so only thing I can do is to fix it on server side. That is
> > because I ask for a help.
> You said the the broken line "m=image 0 udptl 19" goes from OpenSIPS to
> YATE. Do you know if it originates from the OpenSIPS box or the
> subscriber? If it originates from the OpenSIPS box it should be reported
> to them. If its from the subscriber, well...... your chances of getting
> anything fixed are usually small.
>
> Steve
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100222/8f3184e4/attachment.html 

From mike at jerris.com  Mon Feb 22 09:32:45 2010
From: mike at jerris.com (Michael Jerris)
Date: Mon, 22 Feb 2010 12:32:45 -0500
Subject: [Freeswitch-users] Dial String Inject in FreeSwitch
In-Reply-To: <622bedea1002220909x6694fdfdr6dd43f30b056edb2@mail.gmail.com>
References: <622bedea1002220758r3fb59040g4a12af7996d289b5@mail.gmail.com>
	<191c3a031002220819j462ee4c1v3e573124bf161d5b@mail.gmail.com>
	<622bedea1002220833q2a14a9fbh280d0944f5b86fee@mail.gmail.com>
	<191c3a031002220847pa6ca94ey94fa526dad3583d5@mail.gmail.com>
	<622bedea1002220909x6694fdfdr6dd43f30b056edb2@mail.gmail.com>
Message-ID: <93769c21002220932l40e26365u33b034636dc44949@mail.gmail.com>

a good response to this would be to put appropriate notes on the wiki about
what is good practice in this respect.  Perhaps a patch to the default
configs to add notes with an extra warning would be good as well.

Mike

On Mon, Feb 22, 2010 at 12:09 PM, Eder Souza <ederwander at gmail.com> wrote:

> i prefer FreeSwitch im left Asterisk
>
> FreeSwitch is Very Very betther then Asterisk in my option !!
>
>
> my intention is just say dont use (.*), (.+)  or combinations of this
> regular expressions, for me FreeSwitch is the betther  !!
>
> On Mon, Feb 22, 2010 at 1:47 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> To me it sounds like a way to sound the alarms and bring negative
>> attention.
>>
>> For instance, if you were sincerely concerned, you could have told us
>> about your discovery privately first, and we could feature a story on our
>> own site warning people of this danger and reminding them how to compose
>> extension properly.
>>
>> The posting was instead made like a big public announcement calling our
>> software "imperfect".
>> Yes it is imperfect, It can't properly detect someone being a moron 100%
>> of the time but it sure tries it's darndest.
>>
>>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100222/ed2d65da/attachment-0001.html 

From null at invalid.name  Mon Feb 22 09:35:53 2010
From: null at invalid.name (Dan Lane)
Date: Mon, 22 Feb 2010 17:35:53 +0000
Subject: [Freeswitch-users] Dial String Inject in FreeSwitch
In-Reply-To: <622bedea1002220758r3fb59040g4a12af7996d289b5@mail.gmail.com>
References: <622bedea1002220758r3fb59040g4a12af7996d289b5@mail.gmail.com>
Message-ID: <b042b23b1002220935k9e9c281jefc8202e21159b20@mail.gmail.com>

On Mon, Feb 22, 2010 at 3:58 PM, Eder Souza <ederwander at gmail.com> wrote:
> http://ederwander.wordpress.com/2010/02/22/dial-string-inject-in-freeswitch/
>
> just for yours informations i?write this article my test for injections in
> freesitch
>
> version of my tests

I look forward to next week's blog post where you reveal that "rm -rf
/" results in unexpected data loss.


From null at invalid.name  Mon Feb 22 09:39:20 2010
From: null at invalid.name (Dan Lane)
Date: Mon, 22 Feb 2010 17:39:20 +0000
Subject: [Freeswitch-users] 8000 rate .wav files
In-Reply-To: <4B82BD24.2030108@marketelsystems.com>
References: <4B82BD24.2030108@marketelsystems.com>
Message-ID: <b042b23b1002220939v42633482j1084100413f8c52f@mail.gmail.com>

On Mon, Feb 22, 2010 at 5:21 PM, Mark Sobkow
<m.sobkow at marketelsystems.com> wrote:
> I've got the 8000 sample rate .wav files installed for Freeswitch.
> According to the logs, my SIP phone is connecting with an 8000 rate.
> However, when I try to play_and_get_digits using those sound files, I
> get errors:
>
> 2010-02-22 11:15:22.029347 [ERR] switch_core_file.c:122 Invalid file
> format [wav] for
> [/opt/freeswitch/sounds/en/us/callie/conference/8000/conf-pin.wav]!
> 2010-02-22 11:15:22.029347 [ERR] switch_core_file.c:122 Invalid file
> format [wav] for
> [/opt/freeswitch/sounds/en/us/callie/conference/8000/conf-bad-pin.wav]!
>
> Aren't .wav files supposed to be compatible with all codecs for
> playback? ?If not, what do I have to do to convert them to the proper
> formats? ?How do I find out what the proper formats are?

Try resampling the file using sox... something like the following
should do the trick:

sox input.wav -r 8000 -c 1 -s -w output.wav resample -ql


From ederwander at gmail.com  Mon Feb 22 09:39:43 2010
From: ederwander at gmail.com (Eder Souza)
Date: Mon, 22 Feb 2010 14:39:43 -0300
Subject: [Freeswitch-users] Dial String Inject in FreeSwitch
In-Reply-To: <7b197bef1002220926k5c659d67y1b2e6b843a97afee@mail.gmail.com>
References: <622bedea1002220758r3fb59040g4a12af7996d289b5@mail.gmail.com>
	<191c3a031002220819j462ee4c1v3e573124bf161d5b@mail.gmail.com>
	<622bedea1002220833q2a14a9fbh280d0944f5b86fee@mail.gmail.com>
	<191c3a031002220847pa6ca94ey94fa526dad3583d5@mail.gmail.com>
	<622bedea1002220909x6694fdfdr6dd43f30b056edb2@mail.gmail.com>
	<7b197bef1002220926k5c659d67y1b2e6b843a97afee@mail.gmail.com>
Message-ID: <622bedea1002220939y393688d2sb25161c162a04795@mail.gmail.com>

yeah can somebody make one wiki for this alert??

im make down my link page now to prevent thes problems !!

OK

On Mon, Feb 22, 2010 at 2:26 PM, Giovanni Maruzzelli <gmaruzz at celliax.org>wrote:

> Eder,
>
> If you fear people can do such *really stupid* things, and this is
> nice from you, please add something to the wiki, for example a
> paragraph in the dialplan page, or whatever, explaining why this is a
> stupid thing.
>
> If you publish a page in your blog, that look like a security alert,
> or that you found a security flaw in FS, people will rightly think
> that you are just looking for some attention in the search engines,
> and to bring viewers to your page.
>
> Also, in doing so, you push non technical people to think there is a
> security problem in FS, and this is really a big damage to the
> project. Because it is not true, it is just how it look like in your
> page.
>
> So, delete that page, and add something to the wiki, if you care about
> telling people not to do stupid things.
>
> But please, be aware that your page, the page you published, is really
> something that do a damage and put a bad light on a project, and there
> is no one reason for doing this.
>
> -giovanni
>
>
>
> On Mon, Feb 22, 2010 at 6:09 PM, Eder Souza <ederwander at gmail.com> wrote:
> > i prefer FreeSwitch im left Asterisk
> >
> > FreeSwitch is Very Very betther then Asterisk in my option !!
> >
> >
> > my intention is just say dont use (.*), (.+)  or combinations of this
> > regular expressions, for me FreeSwitch is the betther  !!
> >
> >
> >
> > On Mon, Feb 22, 2010 at 1:47 PM, Anthony Minessale
> > <anthony.minessale at gmail.com> wrote:
> >>
> >> To me it sounds like a way to sound the alarms and bring negative
> >> attention.
> >>
> >> For instance, if you were sincerely concerned, you could have told us
> >> about your discovery privately first, and we could feature a story on
> our
> >> own site warning people of this danger and reminding them how to compose
> >> extension properly.
> >>
> >> The posting was instead made like a big public announcement calling our
> >> software "imperfect".
> >> Yes it is imperfect, It can't properly detect someone being a moron 100%
> >> of the time but it sure tries it's darndest.
> >>
> >>
> >>
> >>
> >> On Mon, Feb 22, 2010 at 10:33 AM, Eder Souza <ederwander at gmail.com>
> wrote:
> >>>
> >>> Antony i dont see why ??
> >>>
> >>>
> >>> this is just one alert for all comunity of danger in the use of regular
> >>> expression (.*) or (.*) ...
> >>>
> >>> many peoples can make dialplans witch use of this expressions ...
> >>>
> >>>
> >>>
> >>>
> >>>
> >>> On Mon, Feb 22, 2010 at 1:19 PM, Anthony Minessale
> >>> <anthony.minessale at gmail.com> wrote:
> >>>>
> >>>> Please do not use our project to try to make your blog more popular.
> >>>>
> >>>> Your example requires you to prepare an intentional specific extension
> >>>> on the FreeSWITCH custom made for your attack. It?s like saying if you
> leave
> >>>> your door wide open at your house and call and tell someone, they can
> come
> >>>> and rob you at 8:30.
> >>>>
> >>>> This extension is also vulnerable ?by virtue of the stupidity of the
> >>>> composer?
> >>>>
> >>>> <extension name=?please-hack-me?/>
> >>>>   <condition>
> >>>>    <action application=?system? data=?${destination_number}?/>
> >>>>   </condition>
> >>>> </extension>
> >>>>
> >>>> You should not allow tainted data from outside system to be fed
> directly
> >>>> into your code. There is a regex system in place to extract legitimate
> data
> >>>> from the user tainted input and safeguard against this.
> >>>>
> >>>>
> >>>>
> >>>> On Mon, Feb 22, 2010 at 9:58 AM, Eder Souza <ederwander at gmail.com>
> >>>> wrote:
> >>>>>
> >>>>>
> >>>>>
> http://ederwander.wordpress.com/2010/02/22/dial-string-inject-in-freeswitch/
> >>>>>
> >>>>> just for yours informations i write this article my test for
> injections
> >>>>> in freesitch
> >>>>>
> >>>>> version of my tests
> >>>>>
> >>>>> freeswitch at internal> version
> >>>>> FreeSWITCH Version 1.0.5-20100218-0400 (hacked)
> >>>>> freeswitch at internal>
> >>>>>
> >>>>>
> >>>>>
> >>>>>
> >>>>> _______________________________________________
> >>>>> FreeSWITCH-users mailing list
> >>>>> FreeSWITCH-users at lists.freeswitch.org
> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>>
> >>>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>> http://www.freeswitch.org
> >>>>>
> >>>>
> >>>>
> >>>>
> >>>> --
> >>>> Anthony Minessale II
> >>>>
> >>>> FreeSWITCH http://www.freeswitch.org/
> >>>> ClueCon http://www.cluecon.com/
> >>>> Twitter: http://twitter.com/FreeSWITCH_wire
> >>>>
> >>>> AIM: anthm
> >>>> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
> >>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >>>> IRC: irc.freenode.net #freeswitch
> >>>>
> >>>> FreeSWITCH Developer Conference
> >>>> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> >>>> iax:guest at conference.freeswitch.org/888
> >>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >>>> pstn:+19193869900
> >>>>
> >>>> _______________________________________________
> >>>> FreeSWITCH-users mailing list
> >>>> FreeSWITCH-users at lists.freeswitch.org
> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>> http://www.freeswitch.org
> >>>>
> >>>
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>>
> >>
> >>
> >>
> >> --
> >> Anthony Minessale II
> >>
> >> FreeSWITCH http://www.freeswitch.org/
> >> ClueCon http://www.cluecon.com/
> >> Twitter: http://twitter.com/FreeSWITCH_wire
> >>
> >> AIM: anthm
> >> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >> IRC: irc.freenode.net #freeswitch
> >>
> >> FreeSWITCH Developer Conference
> >> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> >> iax:guest at conference.freeswitch.org/888
> >> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >> pstn:+19193869900
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100222/9d2fd1e4/attachment.html 

From ederwander at gmail.com  Mon Feb 22 09:45:49 2010
From: ederwander at gmail.com (Eder Souza)
Date: Mon, 22 Feb 2010 14:45:49 -0300
Subject: [Freeswitch-users] Dial String Inject in FreeSwitch
In-Reply-To: <622bedea1002220939y393688d2sb25161c162a04795@mail.gmail.com>
References: <622bedea1002220758r3fb59040g4a12af7996d289b5@mail.gmail.com>
	<191c3a031002220819j462ee4c1v3e573124bf161d5b@mail.gmail.com>
	<622bedea1002220833q2a14a9fbh280d0944f5b86fee@mail.gmail.com>
	<191c3a031002220847pa6ca94ey94fa526dad3583d5@mail.gmail.com>
	<622bedea1002220909x6694fdfdr6dd43f30b056edb2@mail.gmail.com>
	<7b197bef1002220926k5c659d67y1b2e6b843a97afee@mail.gmail.com>
	<622bedea1002220939y393688d2sb25161c162a04795@mail.gmail.com>
Message-ID: <622bedea1002220945r5fd10a4cp19ed1244a67374d9@mail.gmail.com>

Link Down :-)

i thaks if somebody create one wiki witch this alert


Eng Eder de Souza

On Mon, Feb 22, 2010 at 2:39 PM, Eder Souza <ederwander at gmail.com> wrote:

> yeah can somebody make one wiki for this alert??
>
> im make down my link page now to prevent thes problems !!
>
> OK
>
>   On Mon, Feb 22, 2010 at 2:26 PM, Giovanni Maruzzelli <
> gmaruzz at celliax.org> wrote:
>
>> Eder,
>>
>> If you fear people can do such *really stupid* things, and this is
>> nice from you, please add something to the wiki, for example a
>> paragraph in the dialplan page, or whatever, explaining why this is a
>> stupid thing.
>>
>> If you publish a page in your blog, that look like a security alert,
>> or that you found a security flaw in FS, people will rightly think
>> that you are just looking for some attention in the search engines,
>> and to bring viewers to your page.
>>
>> Also, in doing so, you push non technical people to think there is a
>> security problem in FS, and this is really a big damage to the
>> project. Because it is not true, it is just how it look like in your
>> page.
>>
>> So, delete that page, and add something to the wiki, if you care about
>> telling people not to do stupid things.
>>
>> But please, be aware that your page, the page you published, is really
>> something that do a damage and put a bad light on a project, and there
>> is no one reason for doing this.
>>
>> -giovanni
>>
>>
>>
>> On Mon, Feb 22, 2010 at 6:09 PM, Eder Souza <ederwander at gmail.com> wrote:
>> > i prefer FreeSwitch im left Asterisk
>> >
>> > FreeSwitch is Very Very betther then Asterisk in my option !!
>> >
>> >
>> > my intention is just say dont use (.*), (.+)  or combinations of this
>> > regular expressions, for me FreeSwitch is the betther  !!
>> >
>> >
>> >
>> > On Mon, Feb 22, 2010 at 1:47 PM, Anthony Minessale
>> > <anthony.minessale at gmail.com> wrote:
>> >>
>> >> To me it sounds like a way to sound the alarms and bring negative
>> >> attention.
>> >>
>> >> For instance, if you were sincerely concerned, you could have told us
>> >> about your discovery privately first, and we could feature a story on
>> our
>> >> own site warning people of this danger and reminding them how to
>> compose
>> >> extension properly.
>> >>
>> >> The posting was instead made like a big public announcement calling our
>> >> software "imperfect".
>> >> Yes it is imperfect, It can't properly detect someone being a moron
>> 100%
>> >> of the time but it sure tries it's darndest.
>> >>
>> >>
>> >>
>> >>
>> >> On Mon, Feb 22, 2010 at 10:33 AM, Eder Souza <ederwander at gmail.com>
>> wrote:
>> >>>
>> >>> Antony i dont see why ??
>> >>>
>> >>>
>> >>> this is just one alert for all comunity of danger in the use of
>> regular
>> >>> expression (.*) or (.*) ...
>> >>>
>> >>> many peoples can make dialplans witch use of this expressions ...
>> >>>
>> >>>
>> >>>
>> >>>
>> >>>
>> >>> On Mon, Feb 22, 2010 at 1:19 PM, Anthony Minessale
>> >>> <anthony.minessale at gmail.com> wrote:
>> >>>>
>> >>>> Please do not use our project to try to make your blog more popular.
>> >>>>
>> >>>> Your example requires you to prepare an intentional specific
>> extension
>> >>>> on the FreeSWITCH custom made for your attack. It?s like saying if
>> you leave
>> >>>> your door wide open at your house and call and tell someone, they can
>> come
>> >>>> and rob you at 8:30.
>> >>>>
>> >>>> This extension is also vulnerable ?by virtue of the stupidity of the
>> >>>> composer?
>> >>>>
>> >>>> <extension name=?please-hack-me?/>
>> >>>>   <condition>
>> >>>>    <action application=?system? data=?${destination_number}?/>
>> >>>>   </condition>
>> >>>> </extension>
>> >>>>
>> >>>> You should not allow tainted data from outside system to be fed
>> directly
>> >>>> into your code. There is a regex system in place to extract
>> legitimate data
>> >>>> from the user tainted input and safeguard against this.
>> >>>>
>> >>>>
>> >>>>
>> >>>> On Mon, Feb 22, 2010 at 9:58 AM, Eder Souza <ederwander at gmail.com>
>> >>>> wrote:
>> >>>>>
>> >>>>>
>> >>>>>
>> http://ederwander.wordpress.com/2010/02/22/dial-string-inject-in-freeswitch/
>> >>>>>
>> >>>>> just for yours informations i write this article my test for
>> injections
>> >>>>> in freesitch
>> >>>>>
>> >>>>> version of my tests
>> >>>>>
>> >>>>> freeswitch at internal> version
>> >>>>> FreeSWITCH Version 1.0.5-20100218-0400 (hacked)
>> >>>>> freeswitch at internal>
>> >>>>>
>> >>>>>
>> >>>>>
>> >>>>>
>> >>>>> _______________________________________________
>> >>>>> FreeSWITCH-users mailing list
>> >>>>> FreeSWITCH-users at lists.freeswitch.org
>> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>>>
>> >>>>> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>>>> http://www.freeswitch.org
>> >>>>>
>> >>>>
>> >>>>
>> >>>>
>> >>>> --
>> >>>> Anthony Minessale II
>> >>>>
>> >>>> FreeSWITCH http://www.freeswitch.org/
>> >>>> ClueCon http://www.cluecon.com/
>> >>>> Twitter: http://twitter.com/FreeSWITCH_wire
>> >>>>
>> >>>> AIM: anthm
>> >>>> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
>> >>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> >>>> IRC: irc.freenode.net #freeswitch
>> >>>>
>> >>>> FreeSWITCH Developer Conference
>> >>>> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>> >>>> iax:guest at conference.freeswitch.org/888
>> >>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> >>>> pstn:+19193869900
>> >>>>
>> >>>> _______________________________________________
>> >>>> FreeSWITCH-users mailing list
>> >>>> FreeSWITCH-users at lists.freeswitch.org
>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>>> http://www.freeswitch.org
>> >>>>
>> >>>
>> >>>
>> >>> _______________________________________________
>> >>> FreeSWITCH-users mailing list
>> >>> FreeSWITCH-users at lists.freeswitch.org
>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> http://www.freeswitch.org
>> >>>
>> >>
>> >>
>> >>
>> >> --
>> >> Anthony Minessale II
>> >>
>> >> FreeSWITCH http://www.freeswitch.org/
>> >> ClueCon http://www.cluecon.com/
>> >> Twitter: http://twitter.com/FreeSWITCH_wire
>> >>
>> >> AIM: anthm
>> >> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
>> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> >> IRC: irc.freenode.net #freeswitch
>> >>
>> >> FreeSWITCH Developer Conference
>> >> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>> >> iax:guest at conference.freeswitch.org/888
>> >> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> >> pstn:+19193869900
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>>
>>
>> --
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> Cell : +39-347-2665618
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100222/a44409e9/attachment-0001.html 

From niall.crosby at gmail.com  Mon Feb 22 10:07:42 2010
From: niall.crosby at gmail.com (Niall Crosby)
Date: Mon, 22 Feb 2010 18:07:42 +0000
Subject: [Freeswitch-users] sending custom events
Message-ID: <4aec92831002221007r3084f75l3db5c9d129625539@mail.gmail.com>

Dear List,

Can someone give an example of sending a custom event use the Event Socket
Layer directly, and not one of the wrapper APIs?

I'm writing in Java and have my own ESL wrapper implementation, but can't
work out how to send custom events.

Thanks in advance,
Niall.

-- 
-- 

The information transmitted is intended only for the person or entity to
which it is addressed and may contain confidential and/or privileged
material. Statements and opinions expressed in this e-mail may not represent
those of the sender. Any review, retransmission, dissemination or other use
of, or taking of any action in reliance upon, this information by persons or
entities other than the intended recipient is prohibited. If you received
this in error, please contact the sender immediately and delete the material
from any computer.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100222/075265fa/attachment.html 

From gmaruzz at celliax.org  Mon Feb 22 10:20:37 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Mon, 22 Feb 2010 19:20:37 +0100
Subject: [Freeswitch-users] Dial String Inject in FreeSwitch
In-Reply-To: <622bedea1002220945r5fd10a4cp19ed1244a67374d9@mail.gmail.com>
References: <622bedea1002220758r3fb59040g4a12af7996d289b5@mail.gmail.com> 
	<191c3a031002220819j462ee4c1v3e573124bf161d5b@mail.gmail.com> 
	<622bedea1002220833q2a14a9fbh280d0944f5b86fee@mail.gmail.com> 
	<191c3a031002220847pa6ca94ey94fa526dad3583d5@mail.gmail.com> 
	<622bedea1002220909x6694fdfdr6dd43f30b056edb2@mail.gmail.com> 
	<7b197bef1002220926k5c659d67y1b2e6b843a97afee@mail.gmail.com> 
	<622bedea1002220939y393688d2sb25161c162a04795@mail.gmail.com> 
	<622bedea1002220945r5fd10a4cp19ed1244a67374d9@mail.gmail.com>
Message-ID: <7b197bef1002221020u7b6587f6l4c25b992f2ac01a3@mail.gmail.com>

Thanks a lot, Eder.

If you feel like, you can add a paragraph yourself, then we'll edit if
necessary.

Just let us know.

-giovanni

On Mon, Feb 22, 2010 at 6:45 PM, Eder Souza <ederwander at gmail.com> wrote:
> Link Down :-)
>
> i thaks if somebody create one wiki witch this alert
>
>
> Eng Eder de Souza
>
> On Mon, Feb 22, 2010 at 2:39 PM, Eder Souza <ederwander at gmail.com> wrote:
>>
>> yeah?can somebody make one wiki for this alert??
>>
>>
>> im make down my link page now?to prevent thes problems !!
>>
>> OK
>>
>> On Mon, Feb 22, 2010 at 2:26 PM, Giovanni Maruzzelli <gmaruzz at celliax.org>
>> wrote:
>>>
>>> Eder,
>>>
>>> If you fear people can do such *really stupid* things, and this is
>>> nice from you, please add something to the wiki, for example a
>>> paragraph in the dialplan page, or whatever, explaining why this is a
>>> stupid thing.
>>>
>>> If you publish a page in your blog, that look like a security alert,
>>> or that you found a security flaw in FS, people will rightly think
>>> that you are just looking for some attention in the search engines,
>>> and to bring viewers to your page.
>>>
>>> Also, in doing so, you push non technical people to think there is a
>>> security problem in FS, and this is really a big damage to the
>>> project. Because it is not true, it is just how it look like in your
>>> page.
>>>
>>> So, delete that page, and add something to the wiki, if you care about
>>> telling people not to do stupid things.
>>>
>>> But please, be aware that your page, the page you published, is really
>>> something that do a damage and put a bad light on a project, and there
>>> is no one reason for doing this.
>>>
>>> -giovanni
>>>
>>>
>>>
>>> On Mon, Feb 22, 2010 at 6:09 PM, Eder Souza <ederwander at gmail.com> wrote:
>>> > i prefer FreeSwitch im left Asterisk
>>> >
>>> > FreeSwitch is Very Very betther then Asterisk in my option !!
>>> >
>>> >
>>> > my intention is just say dont use (.*),?(.+)? or combinations of this
>>> > regular expressions, for me FreeSwitch?is the betther??!!
>>> >
>>> >
>>> >
>>> > On Mon, Feb 22, 2010 at 1:47 PM, Anthony Minessale
>>> > <anthony.minessale at gmail.com> wrote:
>>> >>
>>> >> To me it sounds like a way to sound the alarms and bring negative
>>> >> attention.
>>> >>
>>> >> For instance, if you were sincerely concerned, you could have told us
>>> >> about your discovery privately first, and we could feature a story on
>>> >> our
>>> >> own site warning people of this danger and reminding them how to
>>> >> compose
>>> >> extension properly.
>>> >>
>>> >> The posting was instead made like a big public announcement calling
>>> >> our
>>> >> software "imperfect".
>>> >> Yes it is imperfect, It can't properly detect someone being a moron
>>> >> 100%
>>> >> of the time but it sure tries it's darndest.
>>> >>
>>> >>
>>> >>
>>> >>
>>> >> On Mon, Feb 22, 2010 at 10:33 AM, Eder Souza <ederwander at gmail.com>
>>> >> wrote:
>>> >>>
>>> >>> Antony i dont see why ??
>>> >>>
>>> >>>
>>> >>> this is just one alert for all comunity of danger in the use of
>>> >>> regular
>>> >>> expression (.*) or (.*) ...
>>> >>>
>>> >>> many peoples can make dialplans?witch use of this expressions ...
>>> >>>
>>> >>>
>>> >>>
>>> >>>
>>> >>>
>>> >>> On Mon, Feb 22, 2010 at 1:19 PM, Anthony Minessale
>>> >>> <anthony.minessale at gmail.com> wrote:
>>> >>>>
>>> >>>> Please do not use our project to try to make your blog more popular.
>>> >>>>
>>> >>>> Your example requires you to prepare an intentional specific
>>> >>>> extension
>>> >>>> on the FreeSWITCH custom made for your attack. It?s like saying if
>>> >>>> you leave
>>> >>>> your door wide open at your house and call and tell someone, they
>>> >>>> can come
>>> >>>> and rob you at 8:30.
>>> >>>>
>>> >>>> This extension is also vulnerable ?by virtue of the stupidity of the
>>> >>>> composer?
>>> >>>>
>>> >>>> <extension name=?please-hack-me?/>
>>> >>>> ? <condition>
>>> >>>> ?? <action application=?system? data=?${destination_number}?/>
>>> >>>> ? </condition>
>>> >>>> </extension>
>>> >>>>
>>> >>>> You should not allow tainted data from outside system to be fed
>>> >>>> directly
>>> >>>> into your code. There is a regex system in place to extract
>>> >>>> legitimate data
>>> >>>> from the user tainted input and safeguard against this.
>>> >>>>
>>> >>>>
>>> >>>>
>>> >>>> On Mon, Feb 22, 2010 at 9:58 AM, Eder Souza <ederwander at gmail.com>
>>> >>>> wrote:
>>> >>>>>
>>> >>>>>
>>> >>>>>
>>> >>>>> http://ederwander.wordpress.com/2010/02/22/dial-string-inject-in-freeswitch/
>>> >>>>>
>>> >>>>> just for yours informations i?write this article my test for
>>> >>>>> injections
>>> >>>>> in freesitch
>>> >>>>>
>>> >>>>> version of my tests
>>> >>>>>
>>> >>>>> freeswitch at internal> version
>>> >>>>> FreeSWITCH Version 1.0.5-20100218-0400 (hacked)
>>> >>>>> freeswitch at internal>
>>> >>>>>
>>> >>>>>
>>> >>>>>
>>> >>>>>
>>> >>>>> _______________________________________________
>>> >>>>> FreeSWITCH-users mailing list
>>> >>>>> FreeSWITCH-users at lists.freeswitch.org
>>> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>>>>
>>> >>>>>
>>> >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >>>>> http://www.freeswitch.org
>>> >>>>>
>>> >>>>
>>> >>>>
>>> >>>>
>>> >>>> --
>>> >>>> Anthony Minessale II
>>> >>>>
>>> >>>> FreeSWITCH http://www.freeswitch.org/
>>> >>>> ClueCon http://www.cluecon.com/
>>> >>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>> >>>>
>>> >>>> AIM: anthm
>>> >>>> MSN:anthony_minessale at hotmail.com
>>> >>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>> >>>> IRC: irc.freenode.net #freeswitch
>>> >>>>
>>> >>>> FreeSWITCH Developer Conference
>>> >>>> sip:888 at conference.freeswitch.org
>>> >>>> iax:guest at conference.freeswitch.org/888
>>> >>>> googletalk:conf+888 at conference.freeswitch.org
>>> >>>> pstn:+19193869900
>>> >>>>
>>> >>>> _______________________________________________
>>> >>>> FreeSWITCH-users mailing list
>>> >>>> FreeSWITCH-users at lists.freeswitch.org
>>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>>>
>>> >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >>>> http://www.freeswitch.org
>>> >>>>
>>> >>>
>>> >>>
>>> >>> _______________________________________________
>>> >>> FreeSWITCH-users mailing list
>>> >>> FreeSWITCH-users at lists.freeswitch.org
>>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>>
>>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >>> http://www.freeswitch.org
>>> >>>
>>> >>
>>> >>
>>> >>
>>> >> --
>>> >> Anthony Minessale II
>>> >>
>>> >> FreeSWITCH http://www.freeswitch.org/
>>> >> ClueCon http://www.cluecon.com/
>>> >> Twitter: http://twitter.com/FreeSWITCH_wire
>>> >>
>>> >> AIM: anthm
>>> >> MSN:anthony_minessale at hotmail.com
>>> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>> >> IRC: irc.freenode.net #freeswitch
>>> >>
>>> >> FreeSWITCH Developer Conference
>>> >> sip:888 at conference.freeswitch.org
>>> >> iax:guest at conference.freeswitch.org/888
>>> >> googletalk:conf+888 at conference.freeswitch.org
>>> >> pstn:+19193869900
>>> >>
>>> >> _______________________________________________
>>> >> FreeSWITCH-users mailing list
>>> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>
>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> http://www.freeswitch.org
>>> >>
>>> >
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >
>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>> >
>>>
>>>
>>>
>>> --
>>> Sincerely,
>>>
>>> Giovanni Maruzzelli
>>> Cell : +39-347-2665618
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From Russell.Mosemann at cune.org  Mon Feb 22 10:27:40 2010
From: Russell.Mosemann at cune.org (Russell.Mosemann at cune.org)
Date: Mon, 22 Feb 2010 18:27:40 -0000
Subject: [Freeswitch-users] Dial String Inject in FreeSwitch
In-Reply-To: <622bedea1002220945r5fd10a4cp19ed1244a67374d9@mail.gmail.com>
Message-ID: <20100222182741.05F7829BF68@cuneorg-email.cune.pri>

> i thaks if somebody create one wiki witch this alert

A place to change is Example 1 of the dialplan XML examples. You can tell
people not to use the catchall expressions, because you cannot trust
information from the sender.

http://wiki.freeswitch.org/wiki/Dialplan_XML

A word of caution could also be added to

http://wiki.freeswitch.org/wiki/Regular_Expression

-- 
Russell Mosemann



________________________________________________________
Concordia University, Nebraska
See http://www.cune.edu/ for the latest news and events!



From dftoro at yahoo.com  Mon Feb 22 10:30:04 2010
From: dftoro at yahoo.com (Diego Toro)
Date: Mon, 22 Feb 2010 10:30:04 -0800 (PST)
Subject: [Freeswitch-users] sending a sip notify with sendevent
In-Reply-To: <4B82BBFB.1040804@xpirio.com>
Message-ID: <126320.178.qm@web33502.mail.mud.yahoo.com>

Read "case SWITCH_EVENT_NOTIFY:" in mod_sofia.c. I suggest you assign call-id in the header of the event.


Diego Toro
http://lacarretade.blogspot.com/


--- On Mon, 2/22/10, Christian L?schenkohl <christian.loeschenkohl at xpirio.com> wrote:

> From: Christian L?schenkohl <christian.loeschenkohl at xpirio.com>
> Subject: [Freeswitch-users] sending a sip notify with sendevent
> To: freeswitch-users at lists.freeswitch.org
> Date: Monday, February 22, 2010, 12:16 PM
> hi
> 
> i try to send a sip notify message to a registered sip
> device
> "sofia profile nat flush_inbound_reg 10 at vts.vie.xpirio.net
> reboot" works, but i need
> to send "check-sync;reboot=false" - so the device does a
> resync and don't do a reboot
> 
> my message looks like this
> 
> sendevent NOTIFY
> profile: nat
> event-string: check-sync;reboot=false
> user: 10
> host: vts.vie.xpirio.net
> content-type: application/simple-message-summary
> 
> if i listen on the loopback interface i do see
> 
> ##
> T 2010/02/22 18:11:59.083204 127.0.0.1:51840 ->
> 127.0.0.1:8021 [AP]
> sendevent NOTIFY
> profile: nat
> event-string: check-sync;reboot=false
> user: 10
> host: vts.vie.xpirio.net
> content-type: application/simple-message-summary
> 
> ##
> T 2010/02/22 18:11:59.084032 127.0.0.1:8021 ->
> 127.0.0.1:51840 [AP]
> Content-Type: command/reply
> Reply-Text: -ERR invalid
> 
> --------
> i don't get what it is wrong. i also rechecked the
> registered user in the sqlite database and this
> looks good to me.
> 
> no message is send to the user.
> 
> we do use multiple domains, so user could also be 10 at somedomain.com
> - or am i wrong on this?
> could somebody please bring some light in this.
> 
> we do use trunk rev. 16631
> 
> br
> 
> 
> 
> -- 
> Ing. Christian L?schenkohl
> Technische Leitung, Forschung & Entwicklung VoIP
> 
> xpirio
> Telekommunikation & Service GmbH
> Lakeside B04
> 9020 Klagenfurt
> Austria
> 
> T? +43 (0) 5 77 11 - 1000
> F? +43 (0) 5 77 11 - 1002
> E? christian.loeschenkohl at xpirio.com
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 


      


From lawwton at gmail.com  Mon Feb 22 10:31:21 2010
From: lawwton at gmail.com (Alfredo Quiroga-Villamil)
Date: Mon, 22 Feb 2010 13:31:21 -0500
Subject: [Freeswitch-users] sending custom events
In-Reply-To: <4aec92831002221007r3084f75l3db5c9d129625539@mail.gmail.com>
References: <4aec92831002221007r3084f75l3db5c9d129625539@mail.gmail.com>
Message-ID: <5fe6fa8f1002221031m79b6fefase512629c7e13fb8a@mail.gmail.com>

Niall:

I am really new to FS so I apologize before hand if my response is not
correct; but here is what I understand so far for Events. Some modules
have events associated with them. Some of the modules have Custom
Events.

a) All the way to the bottom of the link shown below you'll see the
modules that have custom events:

http://wiki.freeswitch.org/wiki/Event_list

b) All events listed shown below:

http://wiki.freeswitch.org/wiki/Event_list

c) In my case for example I am also using Java and opening a
connection using my own wrapper as well to ESL. For a conference, we
then send the following custom event via the socket.
An example of a custom event:

event plain CUSTOM conference::maintenance

Java Sample Code:

            Socket socket = new Socket("API URL", PORT);
            String cmd = String.format("event plain %s %s
conference::maintenance",
                                       FreeSwitchEvent.SHUTDOWN.toString(),
                                       FreeSwitchEvent.CUSTOM.toString());

So there I am registering to a couple of events. One of them being a
custom event for the conference in this case.

Like I said I am really new to FS, so I might not be 100% on the money here.

Regards,

Alfredo


On Mon, Feb 22, 2010 at 1:07 PM, Niall Crosby <niall.crosby at gmail.com> wrote:
>
> Dear List,
> Can someone give an example of sending a custom event use the Event Socket
> Layer directly, and not one of the wrapper APIs?
> I'm writing in Java and have my own ESL wrapper implementation, but can't
> work out how to send custom events.
> Thanks in advance,
> Niall.
>
> --
> --
>
> The information transmitted is intended only for the person or entity to
> which it is addressed and may contain confidential and/or privileged
> material. Statements and opinions expressed in this e-mail may not represent
> those of the sender. Any review, retransmission, dissemination or other use
> of, or taking of any action in reliance upon, this information by persons or
> entities other than the intended recipient is prohibited. If you received
> this in error, please contact the sender immediately and delete the material
> from any computer.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From brian at freeswitch.org  Mon Feb 22 10:35:51 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 22 Feb 2010 12:35:51 -0600
Subject: [Freeswitch-users] sending a sip notify with sendevent
In-Reply-To: <126320.178.qm@web33502.mail.mud.yahoo.com>
References: <126320.178.qm@web33502.mail.mud.yahoo.com>
Message-ID: <3959C202-34BE-4DDF-B387-45C1F702377D@freeswitch.org>

Is this not documented on the wiki yet?

/b

On Feb 22, 2010, at 12:30 PM, Diego Toro wrote:

> Read "case SWITCH_EVENT_NOTIFY:" in mod_sofia.c. I suggest you assign call-id in the header of the event.
> 
> 
> Diego Toro
> http://lacarretade.blogspot.com/

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100222/4eb1eacc/attachment.html 

From ederwander at gmail.com  Mon Feb 22 10:38:53 2010
From: ederwander at gmail.com (Eder Souza)
Date: Mon, 22 Feb 2010 15:38:53 -0300
Subject: [Freeswitch-users] Dial String Inject in FreeSwitch
In-Reply-To: <20100222182741.05F7829BF68@cuneorg-email.cune.pri>
References: <622bedea1002220945r5fd10a4cp19ed1244a67374d9@mail.gmail.com>
	<20100222182741.05F7829BF68@cuneorg-email.cune.pri>
Message-ID: <622bedea1002221038u2a2e4232qd1d15cf006abb7db@mail.gmail.com>

Perfect place lol


On Mon, Feb 22, 2010 at 3:27 PM, <Russell.Mosemann at cune.org> wrote:

> > i thaks if somebody create one wiki witch this alert
>
> A place to change is Example 1 of the dialplan XML examples. You can tell
> people not to use the catchall expressions, because you cannot trust
> information from the sender.
>
> http://wiki.freeswitch.org/wiki/Dialplan_XML
>
> A word of caution could also be added to
>
> http://wiki.freeswitch.org/wiki/Regular_Expression
>
> --
> Russell Mosemann
>
>
>
> ________________________________________________________
> Concordia University, Nebraska
> See http://www.cune.edu/ for the latest news and events!
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100222/6ed6336b/attachment-0001.html 

From anthony.minessale at gmail.com  Mon Feb 22 11:05:24 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 22 Feb 2010 13:05:24 -0600
Subject: [Freeswitch-users] sending a sip notify with sendevent
In-Reply-To: <3959C202-34BE-4DDF-B387-45C1F702377D@freeswitch.org>
References: <126320.178.qm@web33502.mail.mud.yahoo.com>
	<3959C202-34BE-4DDF-B387-45C1F702377D@freeswitch.org>
Message-ID: <191c3a031002221105wa86cdfeq845a2d7cb00a4931@mail.gmail.com>

its at the very least missing the profile name

On Mon, Feb 22, 2010 at 12:35 PM, Brian West <brian at freeswitch.org> wrote:

> Is this not documented on the wiki yet?
>
> /b
>
> On Feb 22, 2010, at 12:30 PM, Diego Toro wrote:
>
> Read "case SWITCH_EVENT_NOTIFY:" in mod_sofia.c. I suggest you assign
> call-id in the header of the event.
>
>
> Diego Toro
> http://lacarretade.blogspot.com/
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100222/86bd786e/attachment.html 

From ederwander at gmail.com  Mon Feb 22 11:13:04 2010
From: ederwander at gmail.com (Eder Souza)
Date: Mon, 22 Feb 2010 16:13:04 -0300
Subject: [Freeswitch-users] Dial String Inject in FreeSwitch
In-Reply-To: <622bedea1002221038u2a2e4232qd1d15cf006abb7db@mail.gmail.com>
References: <622bedea1002220945r5fd10a4cp19ed1244a67374d9@mail.gmail.com>
	<20100222182741.05F7829BF68@cuneorg-email.cune.pri>
	<622bedea1002221038u2a2e4232qd1d15cf006abb7db@mail.gmail.com>
Message-ID: <622bedea1002221113j5ac06477jb24ac51eedcd8d8f@mail.gmail.com>

http://wiki.freeswitch.org/wiki/User:Ederwander




On Mon, Feb 22, 2010 at 3:38 PM, Eder Souza <ederwander at gmail.com> wrote:

> Perfect place lol
>
>
> On Mon, Feb 22, 2010 at 3:27 PM, <Russell.Mosemann at cune.org> wrote:
>
>> > i thaks if somebody create one wiki witch this alert
>>
>> A place to change is Example 1 of the dialplan XML examples. You can tell
>> people not to use the catchall expressions, because you cannot trust
>> information from the sender.
>>
>> http://wiki.freeswitch.org/wiki/Dialplan_XML
>>
>> A word of caution could also be added to
>>
>> http://wiki.freeswitch.org/wiki/Regular_Expression
>>
>> --
>> Russell Mosemann
>>
>>
>>
>> ________________________________________________________
>> Concordia University, Nebraska
>> See http://www.cune.edu/ for the latest news and events!
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100222/302a85da/attachment.html 

From andrew at hijacked.us  Mon Feb 22 11:20:42 2010
From: andrew at hijacked.us (Andrew Thompson)
Date: Mon, 22 Feb 2010 14:20:42 -0500
Subject: [Freeswitch-users] 8000 rate .wav files
In-Reply-To: <4B82BD24.2030108@marketelsystems.com>
References: <4B82BD24.2030108@marketelsystems.com>
Message-ID: <20100222192042.GI8518@hijacked.us>

On Mon, Feb 22, 2010 at 11:21:40AM -0600, Mark Sobkow wrote:
> I've got the 8000 sample rate .wav files installed for Freeswitch.  
> According to the logs, my SIP phone is connecting with an 8000 rate.  
> However, when I try to play_and_get_digits using those sound files, I 
> get errors:
> 
> 2010-02-22 11:15:22.029347 [ERR] switch_core_file.c:122 Invalid file 
> format [wav] for 
> [/opt/freeswitch/sounds/en/us/callie/conference/8000/conf-pin.wav]!
> 2010-02-22 11:15:22.029347 [ERR] switch_core_file.c:122 Invalid file 
> format [wav] for 
> [/opt/freeswitch/sounds/en/us/callie/conference/8000/conf-bad-pin.wav]!
> 
> Aren't .wav files supposed to be compatible with all codecs for 
> playback?  If not, what do I have to do to convert them to the proper 
> formats?  How do I find out what the proper formats are?
> 

Do you have mod_sndfile loaded?

Andrew


From m.sobkow at marketelsystems.com  Mon Feb 22 11:45:19 2010
From: m.sobkow at marketelsystems.com (Mark Sobkow)
Date: Mon, 22 Feb 2010 13:45:19 -0600
Subject: [Freeswitch-users] 8000 rate .wav files
In-Reply-To: <20100222192042.GI8518@hijacked.us>
References: <4B82BD24.2030108@marketelsystems.com>
	<20100222192042.GI8518@hijacked.us>
Message-ID: <4B82DECF.5000300@marketelsystems.com>

An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100222/3bbe847c/attachment.html 

From m.sobkow at marketelsystems.com  Mon Feb 22 12:20:10 2010
From: m.sobkow at marketelsystems.com (Mark Sobkow)
Date: Mon, 22 Feb 2010 14:20:10 -0600
Subject: [Freeswitch-users] Is there any way to loop a dialplan?
Message-ID: <4B82E6FA.3090008@marketelsystems.com>

Let me explain what it is I'm trying to do.  Maybe there's another way 
to achieve it.

When an operator dials in to the log-in line (e.g. Extension 6000), I 
use play_and_get_digits to collect the operator's PIN.  I then need to 
be able to fire up some Erlang (or Javascript) to verify the PIN, and 
after verification, put the call into a park state, collecting the 
UUID.  I then need to fire an event to Erlang passing along the parked 
UUID and the operator's PIN so that Erlang can direct received customer 
calls to the operators based on relatively complex criteria that won't 
fit in a dialplan.

The catch is that when I get a customer call, I collect their info via 
IVR menus, park the call, and fire an event to Erlang with the UUID of 
the parked call and info collected from the IVR.  Erlang analyses the 
info, selects an operator who is free, and bridges the calls.

The problem I'm having is figuring out a way to get the operator leg to 
go back into a park state after handling the customer's call.  Ideally I 
want the operator to be presented with a short IVR to collect info about 
how the call was handled, but I can do that through a custom application 
GUI if I need to.  Regardless, once an operator logs in, they need to 
_stay_ logged in until they explicityly log out, but I can't figure out 
how I'm supposed to do that without some sort of looping capability.

One thing I was thinking that might work is to set up a set of "dummy" 
extensions that I can have Erlang dial and bridge which contain a 
dialplan fragment to collect the IVR call result, park the call, and 
issue the operator PIN and parked UUID again to Erlang.  That way 
between Erlang events and the dialplan fragment I end up with an 
effective "loop".  (Though I've yet to figure out how I can break out of 
that loop.  Maybe it'll have to be an IVR option for logging out.)

Sample code/dialplans would be good, but for now I'll settle for knowing 
whether I'm at least on the right track for how to implement this beast.

Note that I only want to drop into Javascript if I can't figure out how 
to do it with dialplans and Erlang.

Thanks for any ideas and suggestions.

-- 
Mark Sobkow
Senior Developer
MarkeTel Multi-Line Dialing Systems LTD.
428 Victoria Ave
Regina, SK S4N-0P6
Toll-Free: 800-289-8616-X533
Local: 306-359-6893-X533
Fax: 306-359-6879
Email: m.sobkow at marketelsystems.com
Web: http://www.marketelsystems.com 



From ivdreg at gmail.com  Mon Feb 22 12:23:00 2010
From: ivdreg at gmail.com (ivdreg ivdreg)
Date: Mon, 22 Feb 2010 22:23:00 +0200
Subject: [Freeswitch-users] SDP With T.38 in INVITE Problem
In-Reply-To: <93769c21002220929t1dbba5bcm3d9200f68a9e1800@mail.gmail.com>
References: <b437a0e1002190242y2aa64696j91e8bc76f7ae2a92@mail.gmail.com>
	<b437a0e1002220448g3e3bda9fpa283e93a11d319f1@mail.gmail.com>
	<24736FFE-BEEF-4D34-BBA7-77B6B6AE0EBD@jerris.com>
	<b437a0e1002220711u1bfd138l8a67b4e0b024d856@mail.gmail.com>
	<93769c21002220729q5bf36eadk80726aa8c00e3bd@mail.gmail.com>
	<b437a0e1002220849k16fd87ffl5a7257d80e43b036@mail.gmail.com>
	<4B82BA90.10709@coppice.org>
	<93769c21002220929t1dbba5bcm3d9200f68a9e1800@mail.gmail.com>
Message-ID: <b437a0e1002221223w53ef522asc807e56a13d8bc70@mail.gmail.com>

Thanks Michael,

I will try to find some solution/workaround.

2010/2/22 Michael Jerris <mike at jerris.com>

> the port 0 with PT of 19 is sofia rejecting the sdp becuase we don't
> support it.
>
>
> On Mon, Feb 22, 2010 at 12:10 PM, Steve Underwood <steveu at coppice.org>wrote:
>
>> Hi Michael,
>>
>> On 02/23/2010 12:49 AM, ivdreg ivdreg wrote:
>> > Hi Michael,
>> >
>> > As I said in a previous mails I know exactly what is happening.
>> > In working setup:
>> >
>> > ITSP ---> YATE (GW - Frontend) ---> FreeSwitch (routing
>> > server/xml_curl) ---> YATE (GW - SIP Interop) ---> OpenSIPS --->
>> > Subscriber.
>> >
>> > I prefer to change YATE (GW - Frontend) and YATE (GW - SIP Interop)
>> > with FreeSwitch for some reasons. The problem is:
>> >
>> > INVITE comes from ITSP and goes to subscriber via OpenSIPS. INIVITE
>> > between FreeSwitch (routing server) and YATE (GW - SIP Interop)
>> > contains SDP:
>> > m=audio 21108 RTP/AVP 18 4 8 0
>> > c=IN IP4 10.10.1.110
>> > a=rtpmap:18 G729/8000
>> > a=rtpmap:4 G723/8000
>> > a=rtpmap:8 PCMA/8000
>> > a=rtpmap:0 PCMU/8000
>> > m=image 21108 udptl t38
>> > c=IN IP4 10.10.1.110
>> > a=T38FaxVersion:0
>> > a=T38MaxBitRate:14400
>> > a=T38FaxUdpEC:t38UDPRedundancy
>> > a=T38FaxRateManagement:
>> > transferredTCF
>> >
>> > And reply 200 OK contains in SDP:
>> > *m=audio 34788 RTP/AVP 8*
>> > a=rtpmap:8 PCMA/8000
>> > a=silenceSupp:off - - - -
>> > a=ptime:20
>> >
>> > Reply 200 OK SDP between YATE (GW - SIP Interop) and OpenSIPS contains
>> > in SDP:
>> > *m=audio 16330 RTP/AVP 8*
>> > a=rtpmap:8 PCMA/8000
>> > a=silenceSupp:off - - - -
>> > a=ptime:20
>> > *m=image 0 udptl 19*
>> >
>> > In this case everything works fine. Line *m=image 0 udptl 19 *is
>> > removed by YATE.
>> > But same test with FreeSWITCH on the place of YATE (GW - SIP Interop)
>> > *"m=image 0 udptl 19" *call brakes as you can see in my first mail.
>> >
>> > I don't want to compare or discus YATE and FS functionality or
>> > something else. I just see difference in behavior and because I'm not
>> > a big expert don't know witch implementation is more accurate
>> > according standards. And second: Is it impossible for me to upgrade
>> > all CPE so only thing I can do is to fix it on server side. That is
>> > because I ask for a help.
>> You said the the broken line "m=image 0 udptl 19" goes from OpenSIPS to
>> YATE. Do you know if it originates from the OpenSIPS box or the
>> subscriber? If it originates from the OpenSIPS box it should be reported
>> to them. If its from the subscriber, well...... your chances of getting
>> anything fixed are usually small.
>>
>> Steve
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100222/5cd510a7/attachment-0001.html 

From lfurrea at gmail.com  Mon Feb 22 13:05:11 2010
From: lfurrea at gmail.com (Luis F Urrea)
Date: Mon, 22 Feb 2010 15:05:11 -0600
Subject: [Freeswitch-users] ESF_PAGE_GROUP in a box with 2 interfaces
Message-ID: <f120f4c81002221305r7c9b4dedw13ccf5f72eb75d05@mail.gmail.com>

 Hi all,
I have a FS process running on a fw with 2 ethernet
interfaces, the FS process is bound to the internal iface but
when I use esf_page_group it tries to forward multicast
packets through the external iface, is there a config
parameter maybe to be able to control this?

TIA
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100222/ead3f499/attachment.html 

From anthony.minessale at gmail.com  Mon Feb 22 13:37:43 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 22 Feb 2010 15:37:43 -0600
Subject: [Freeswitch-users] ESF_PAGE_GROUP in a box with 2 interfaces
In-Reply-To: <f120f4c81002221305r7c9b4dedw13ccf5f72eb75d05@mail.gmail.com>
References: <f120f4c81002221305r7c9b4dedw13ccf5f72eb75d05@mail.gmail.com>
Message-ID: <191c3a031002221337t3c49cda7r14d80c71d510a8e9@mail.gmail.com>

yes you can control the IP and by virtue of your routing table which
interface.


On Mon, Feb 22, 2010 at 3:05 PM, Luis F Urrea <lfurrea at gmail.com> wrote:

>  Hi all,
> I have a FS process running on a fw with 2 ethernet
> interfaces, the FS process is bound to the internal iface but
> when I use esf_page_group it tries to forward multicast
> packets through the external iface, is there a config
> parameter maybe to be able to control this?
>
> TIA
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100222/e75a6eeb/attachment.html 

From christian.loeschenkohl at xpirio.com  Mon Feb 22 13:40:18 2010
From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=)
Date: Mon, 22 Feb 2010 22:40:18 +0100
Subject: [Freeswitch-users] sending a sip notify with sendevent
In-Reply-To: <126320.178.qm@web33502.mail.mud.yahoo.com>
References: <126320.178.qm@web33502.mail.mud.yahoo.com>
Message-ID: <4B82F9C2.2040002@xpirio.com>

thank you for this advise

i read the section "case SWITCH_EVENT_NOTIFY" carefully, debugged my script (i messed something
up, with telnet the command works - returns Reply-Text: +OK)

sql is executed and returns 1 row
select sip_user,sip_host,contact,profile_name,'application/simple-message-summary','check-sync;reboot=true',''
from sip_registrations where sip_user='10' and sip_host='vts.vie.xpirio.net'

however no notify message is send to the device

i can't use call-id because i simply don't know it

br


Diego Toro wrote:

> Read "case SWITCH_EVENT_NOTIFY:" in mod_sofia.c. I suggest you assign call-id in the header of the event.
> 
> 
> Diego Toro
> http://lacarretade.blogspot.com/
> 
> 
> --- On Mon, 2/22/10, Christian L?schenkohl <christian.loeschenkohl at xpirio.com> wrote:
> 
>> From: Christian L?schenkohl <christian.loeschenkohl at xpirio.com>
>> Subject: [Freeswitch-users] sending a sip notify with sendevent
>> To: freeswitch-users at lists.freeswitch.org
>> Date: Monday, February 22, 2010, 12:16 PM
>> hi
>>
>> i try to send a sip notify message to a registered sip
>> device
>> "sofia profile nat flush_inbound_reg 10 at vts.vie.xpirio.net
>> reboot" works, but i need
>> to send "check-sync;reboot=false" - so the device does a
>> resync and don't do a reboot
>>
>> my message looks like this
>>
>> sendevent NOTIFY
>> profile: nat
>> event-string: check-sync;reboot=false
>> user: 10
>> host: vts.vie.xpirio.net
>> content-type: application/simple-message-summary
>>
>> if i listen on the loopback interface i do see
>>
>> ##
>> T 2010/02/22 18:11:59.083204 127.0.0.1:51840 ->
>> 127.0.0.1:8021 [AP]
>> sendevent NOTIFY
>> profile: nat
>> event-string: check-sync;reboot=false
>> user: 10
>> host: vts.vie.xpirio.net
>> content-type: application/simple-message-summary
>>
>> ##
>> T 2010/02/22 18:11:59.084032 127.0.0.1:8021 ->
>> 127.0.0.1:51840 [AP]
>> Content-Type: command/reply
>> Reply-Text: -ERR invalid
>>
>> --------
>> i don't get what it is wrong. i also rechecked the
>> registered user in the sqlite database and this
>> looks good to me.
>>
>> no message is send to the user.
>>
>> we do use multiple domains, so user could also be 10 at somedomain.com
>> - or am i wrong on this?
>> could somebody please bring some light in this.
>>
>> we do use trunk rev. 16631
>>
>> br
>>
>>
>>
>> -- 
>> Ing. Christian L?schenkohl
>> Technische Leitung, Forschung & Entwicklung VoIP
>>
>> xpirio
>> Telekommunikation & Service GmbH
>> Lakeside B04
>> 9020 Klagenfurt
>> Austria
>>
>> T  +43 (0) 5 77 11 - 1000
>> F  +43 (0) 5 77 11 - 1002
>> E  christian.loeschenkohl at xpirio.com
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
> 
> 
>       
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


-- 
Ing. Christian L?schenkohl
Technische Leitung, Forschung & Entwicklung VoIP

xpirio
Telekommunikation & Service GmbH
Lakeside B04
9020 Klagenfurt
Austria

T  +43 (0) 5 77 11 - 1000
F  +43 (0) 5 77 11 - 1002
E  christian.loeschenkohl at xpirio.com


From anthony.minessale at gmail.com  Mon Feb 22 13:48:45 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 22 Feb 2010 15:48:45 -0600
Subject: [Freeswitch-users] sending a sip notify with sendevent
In-Reply-To: <4B82F9C2.2040002@xpirio.com>
References: <126320.178.qm@web33502.mail.mud.yahoo.com>
	<4B82F9C2.2040002@xpirio.com>
Message-ID: <191c3a031002221348m7b474499rea4d8ab6e806d05a@mail.gmail.com>

compare that sql stmt to your db manually with the sqlite3 app

sqlite3 /usr/local/freeswitch/db/sofia_reg_internal.db


2010/2/22 Christian L?schenkohl <christian.loeschenkohl at xpirio.com>

> thank you for this advise
>
> i read the section "case SWITCH_EVENT_NOTIFY" carefully, debugged my script
> (i messed something
> up, with telnet the command works - returns Reply-Text: +OK)
>
> sql is executed and returns 1 row
> select
> sip_user,sip_host,contact,profile_name,'application/simple-message-summary','check-sync;reboot=true',''
> from sip_registrations where sip_user='10' and sip_host='
> vts.vie.xpirio.net'
>
> however no notify message is send to the device
>
> i can't use call-id because i simply don't know it
>
> br
>
>
> Diego Toro wrote:
>
> > Read "case SWITCH_EVENT_NOTIFY:" in mod_sofia.c. I suggest you assign
> call-id in the header of the event.
> >
> >
> > Diego Toro
> > http://lacarretade.blogspot.com/
> >
> >
> > --- On Mon, 2/22/10, Christian L?schenkohl <
> christian.loeschenkohl at xpirio.com> wrote:
> >
> >> From: Christian L?schenkohl <christian.loeschenkohl at xpirio.com>
> >> Subject: [Freeswitch-users] sending a sip notify with sendevent
> >> To: freeswitch-users at lists.freeswitch.org
> >> Date: Monday, February 22, 2010, 12:16 PM
> >> hi
> >>
> >> i try to send a sip notify message to a registered sip
> >> device
> >> "sofia profile nat flush_inbound_reg 10 at vts.vie.xpirio.net
> >> reboot" works, but i need
> >> to send "check-sync;reboot=false" - so the device does a
> >> resync and don't do a reboot
> >>
> >> my message looks like this
> >>
> >> sendevent NOTIFY
> >> profile: nat
> >> event-string: check-sync;reboot=false
> >> user: 10
> >> host: vts.vie.xpirio.net
> >> content-type: application/simple-message-summary
> >>
> >> if i listen on the loopback interface i do see
> >>
> >> ##
> >> T 2010/02/22 18:11:59.083204 127.0.0.1:51840 ->
> >> 127.0.0.1:8021 [AP]
> >> sendevent NOTIFY
> >> profile: nat
> >> event-string: check-sync;reboot=false
> >> user: 10
> >> host: vts.vie.xpirio.net
> >> content-type: application/simple-message-summary
> >>
> >> ##
> >> T 2010/02/22 18:11:59.084032 127.0.0.1:8021 ->
> >> 127.0.0.1:51840 [AP]
> >> Content-Type: command/reply
> >> Reply-Text: -ERR invalid
> >>
> >> --------
> >> i don't get what it is wrong. i also rechecked the
> >> registered user in the sqlite database and this
> >> looks good to me.
> >>
> >> no message is send to the user.
> >>
> >> we do use multiple domains, so user could also be 10 at somedomain.com
> >> - or am i wrong on this?
> >> could somebody please bring some light in this.
> >>
> >> we do use trunk rev. 16631
> >>
> >> br
> >>
> >>
> >>
> >> --
> >> Ing. Christian L?schenkohl
> >> Technische Leitung, Forschung & Entwicklung VoIP
> >>
> >> xpirio
> >> Telekommunikation & Service GmbH
> >> Lakeside B04
> >> 9020 Klagenfurt
> >> Austria
> >>
> >> T  +43 (0) 5 77 11 - 1000
> >> F  +43 (0) 5 77 11 - 1002
> >> E  christian.loeschenkohl at xpirio.com
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> --
> Ing. Christian L?schenkohl
> Technische Leitung, Forschung & Entwicklung VoIP
>
> xpirio
> Telekommunikation & Service GmbH
> Lakeside B04
> 9020 Klagenfurt
> Austria
>
> T  +43 (0) 5 77 11 - 1000
> F  +43 (0) 5 77 11 - 1002
> E  christian.loeschenkohl at xpirio.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100222/a4526a0e/attachment.html 

From m.sobkow at marketelsystems.com  Mon Feb 22 14:06:12 2010
From: m.sobkow at marketelsystems.com (Mark Sobkow)
Date: Mon, 22 Feb 2010 16:06:12 -0600
Subject: [Freeswitch-users] Is there any way to loop a dialplan?
Message-ID: <4B82FFD4.40309@marketelsystems.com>

Apparently you can't have that first call waiting for the FIFO to be 
picked up.  It'll bridge a FIFO, but the remainder of the dial plan 
never executes. *sigh*

What is INTENDED to happen is that the operator dials in to extension 
6000 and enters their PIN, then gets put in the FIFO.  The idea is that 
after they've handled a call from the FIFO, the dialplan for extension 
6000 resumes, collects the call result, and then enters the looping 
dialplan for extension 6001, which just keeps putting them back in the 
queue, collecting a result code, and repeating ad-nauseum.

I've also tried calling a loopback extension so I could properly bridge 
the call, thinking that would respect the set hangup_after_bridge=false, 
but that doesn't work either.

I may be frustrated, but I'm having fun...

<extension name=\"TestBridging\">
\t<condition field=\"destination_number\" expression=\"^6000$\">
\t\t<action application=\"answer\" />
\t\t<action application=\"play_and_get_digits\" data=\"4 4 1 5000 # 
/opt/freeswitch/sounds/en/us/callie/conference/8000/conf-pin.wav 
/opt/freeswitch/sounds/en/us/callie/conference/8000/conf-bad-pin.wav 
operator_pin \\d+\" />
\t\t<action application=\"log\" data=\"INFO operator_pin is 
${operator_pin}\" />
\t\t<action application=\"log\" data=\"INFO Queing operator to fifo\" />
\t\t<action application=\"set\" data=\"hangup_after_bridge=false\" />
\t\t<action application=\"bridge\" data=\"loopback/6003\" />
\t\t<action application=\"log\" data=\"INFO Handled first call\" />
\t\t<action application=\"play_and_get_digits\" data=\"1 2 1 5000 # 
/opt/freeswitch/sounds/en/us/callie/conference/8000/conf-pin.wav 
/opt/freeswitch/sounds/en/us/callie/conference/8000/conf-bad-pin.wav 
call_result \\d+\" />
\t\t<action application=\"log\" data=\"INFO call_result is 
${call_result}\" />
\t\t<action application=\"set\" data=\"hangup_after_bridge=true\" />
\t\t<action application=\"bridge\" data=\"loopback/6001\" />
\t</condition>
</extension>
<extension name=\"TestLoop\">
\t<condition field=\"destination_number\" expression=\"^6001$\">
\t\t<action application=\"answer\" />
\t\t<action application=\"log\" data=\"INFO Requeing operator to fifo\" />
\t\t<action application=\"set\" data=\"hangup_after_bridge=false\" />
\t\t<action application=\"bridge\" data=\"loopback/6003\" />
\t\t<action application=\"log\" data=\"INFO Handled another call\" />
\t\t<action application=\"play_and_get_digits\" data=\"1 2 1 5000 # 
/opt/freeswitch/sounds/en/us/callie/conference/8000/conf-pin.wav 
/opt/freeswitch/sounds/en/us/callie/conference/8000/conf-bad-pin.wav 
call_result \\d+\" />
\t\t<action application=\"log\" data=\"INFO call_result is 
${call_result}\" />
\t\t<action application=\"set\" data=\"hangup_after_bridge=true\" />
\t\t<action application=\"bridge\" data=\"loopback/6001\" />
\t</condition>
</extension>
<extension name=\"TestCallIn\">
\t<condition field=\"destination_number\" expression=\"^6002$\">
\t\t<action application=\"answer\" />
\t\t<action application=\"log\" data=\"INFO Pulling operator\" />
\t\t<action application=\"set\" data=\"hangup_after_bridge=true\" />
\t\t<action application=\"fifo\" data=\"6003@${domain_name} out nowait\" />
\t\t<action application=\"log\" data=\"INFO Done call\" />
\t</condition>
</extension>
<extension name=\"park\">
\t<condition field=\"destination_number\" expression=\"^6003$\">
\t\t<action application=\"fifo\" data=\"6003@${domain_name} in\" />
\t</condition>
</extension>
<extension name=\"unpark\">
\t<condition field=\"destination_number\" expression=\"^6004$\">
\t\t<action application=\"answer\" />
\t\t<action application=\"fifo\" data=\"6003@${domain_name} out nowait\" />
\t</condition>
</extension>

-- 
Mark Sobkow
Senior Developer
MarkeTel Multi-Line Dialing Systems LTD.
428 Victoria Ave
Regina, SK S4N-0P6
Toll-Free: 800-289-8616-X533
Local: 306-359-6893-X533
Fax: 306-359-6879
Email: m.sobkow at marketelsystems.com
Web: http://www.marketelsystems.com 



From lfurrea at gmail.com  Mon Feb 22 14:27:19 2010
From: lfurrea at gmail.com (Luis F Urrea)
Date: Mon, 22 Feb 2010 16:27:19 -0600
Subject: [Freeswitch-users] ESF_PAGE_GROUP in a box with 2 interfaces
In-Reply-To: <191c3a031002221337t3c49cda7r14d80c71d510a8e9@mail.gmail.com>
References: <f120f4c81002221305r7c9b4dedw13ccf5f72eb75d05@mail.gmail.com>
	<191c3a031002221337t3c49cda7r14d80c71d510a8e9@mail.gmail.com>
Message-ID: <f120f4c81002221427x78956438oc8b54cd1b85cd76d@mail.gmail.com>

Well the rest of the story is that I'm running a FreeBSD box and when the
box has only one interface then it's not necessary to setup multicast
routing which by the way is not built in a FreeBSD generic kernel.

FreebsD docs state:

"FreeBSD supports multicast host operations by default"

Since this would be in the category of a host operation and there would be
no need to forward multicast traffic between interfaces I thought that maybe
there could be code in the esf application that would choose the IP it would
bound to.

Just wondering if that is the case but if you Anthony can confirm that it is
totally left out to the OS routing rules I'll take my inquiry somewhere else
to clarify further.

Thx


On Mon, Feb 22, 2010 at 3:37 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> yes you can control the IP and by virtue of your routing table which
> interface.
>
>
> On Mon, Feb 22, 2010 at 3:05 PM, Luis F Urrea <lfurrea at gmail.com> wrote:
>
>>  Hi all,
>> I have a FS process running on a fw with 2 ethernet
>> interfaces, the FS process is bound to the internal iface but
>> when I use esf_page_group it tries to forward multicast
>> packets through the external iface, is there a config
>> parameter maybe to be able to control this?
>>
>> TIA
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
<img width="300" height="160" border="0" _moz_dirty="" src="
http://www.simplecs.net/firmas/calufa.jpg" alt="firma"
_moz_resizing="true"/>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100222/84e8160a/attachment.html 

From christian.loeschenkohl at xpirio.com  Mon Feb 22 14:28:39 2010
From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=)
Date: Mon, 22 Feb 2010 23:28:39 +0100
Subject: [Freeswitch-users] sending a sip notify with sendevent
In-Reply-To: <191c3a031002221348m7b474499rea4d8ab6e806d05a@mail.gmail.com>
References: <126320.178.qm@web33502.mail.mud.yahoo.com>	<4B82F9C2.2040002@xpirio.com>
	<191c3a031002221348m7b474499rea4d8ab6e806d05a@mail.gmail.com>
Message-ID: <4B830517.2070402@xpirio.com>

hi anthony

i did it

my profile is actually named nat, "sofia status profile nat" shows me presence_nat as the db name
so i had a look in the file presence_nat.db

i execute
select sip_user,sip_host,contact,profile_name,'application/simple-message-summary','check-sync;reboot=false','' from sip_registrations where sip_user='10' and sip_host='vts.vie.xpirio.net';

and it returns
10|vts.vie.xpirio.net|"10" <sip:10 at 81.94.53.5:1032>|nat|application/simple-message-summary|check-sync;reboot=false|

looks good to me so far, but as i said no sip notify message is send to the client

br

Anthony Minessale wrote:

> compare that sql stmt to your db manually with the sqlite3 app
> 
> sqlite3 /usr/local/freeswitch/db/sofia_reg_internal.db
> 
> 
> 2010/2/22 Christian L?schenkohl <christian.loeschenkohl at xpirio.com 
> <mailto:christian.loeschenkohl at xpirio.com>>
> 
>     thank you for this advise
> 
>     i read the section "case SWITCH_EVENT_NOTIFY" carefully, debugged my
>     script (i messed something
>     up, with telnet the command works - returns Reply-Text: +OK)
> 
>     sql is executed and returns 1 row
>     select
>     sip_user,sip_host,contact,profile_name,'application/simple-message-summary','check-sync;reboot=true',''
>     from sip_registrations where sip_user='10' and
>     sip_host='vts.vie.xpirio.net <http://vts.vie.xpirio.net>'
> 
>     however no notify message is send to the device
> 
>     i can't use call-id because i simply don't know it
> 
>     br
> 
> 
>     Diego Toro wrote:
> 
>      > Read "case SWITCH_EVENT_NOTIFY:" in mod_sofia.c. I suggest you
>     assign call-id in the header of the event.
>      >
>      >
>      > Diego Toro
>      > http://lacarretade.blogspot.com/
>      >
>      >
>      > --- On Mon, 2/22/10, Christian L?schenkohl
>     <christian.loeschenkohl at xpirio.com
>     <mailto:christian.loeschenkohl at xpirio.com>> wrote:
>      >
>      >> From: Christian L?schenkohl <christian.loeschenkohl at xpirio.com
>     <mailto:christian.loeschenkohl at xpirio.com>>
>      >> Subject: [Freeswitch-users] sending a sip notify with sendevent
>      >> To: freeswitch-users at lists.freeswitch.org
>     <mailto:freeswitch-users at lists.freeswitch.org>
>      >> Date: Monday, February 22, 2010, 12:16 PM
>      >> hi
>      >>
>      >> i try to send a sip notify message to a registered sip
>      >> device
>      >> "sofia profile nat flush_inbound_reg 10 at vts.vie.xpirio.net
>     <mailto:10 at vts.vie.xpirio.net>
>      >> reboot" works, but i need
>      >> to send "check-sync;reboot=false" - so the device does a
>      >> resync and don't do a reboot
>      >>
>      >> my message looks like this
>      >>
>      >> sendevent NOTIFY
>      >> profile: nat
>      >> event-string: check-sync;reboot=false
>      >> user: 10
>      >> host: vts.vie.xpirio.net <http://vts.vie.xpirio.net>
>      >> content-type: application/simple-message-summary
>      >>
>      >> if i listen on the loopback interface i do see
>      >>
>      >> ##
>      >> T 2010/02/22 18:11:59.083204 127.0.0.1:51840
>     <http://127.0.0.1:51840> ->
>      >> 127.0.0.1:8021 <http://127.0.0.1:8021> [AP]
>      >> sendevent NOTIFY
>      >> profile: nat
>      >> event-string: check-sync;reboot=false
>      >> user: 10
>      >> host: vts.vie.xpirio.net <http://vts.vie.xpirio.net>
>      >> content-type: application/simple-message-summary
>      >>
>      >> ##
>      >> T 2010/02/22 18:11:59.084032 127.0.0.1:8021
>     <http://127.0.0.1:8021> ->
>      >> 127.0.0.1:51840 <http://127.0.0.1:51840> [AP]
>      >> Content-Type: command/reply
>      >> Reply-Text: -ERR invalid
>      >>
>      >> --------
>      >> i don't get what it is wrong. i also rechecked the
>      >> registered user in the sqlite database and this
>      >> looks good to me.
>      >>
>      >> no message is send to the user.
>      >>
>      >> we do use multiple domains, so user could also be
>     10 at somedomain.com <mailto:10 at somedomain.com>
>      >> - or am i wrong on this?
>      >> could somebody please bring some light in this.
>      >>
>      >> we do use trunk rev. 16631
>      >>
>      >> br
>      >>
>      >>
>      >>
>      >> --
>      >> Ing. Christian L?schenkohl
>      >> Technische Leitung, Forschung & Entwicklung VoIP
>      >>
>      >> xpirio
>      >> Telekommunikation & Service GmbH
>      >> Lakeside B04
>      >> 9020 Klagenfurt
>      >> Austria
>      >>
>      >> T  +43 (0) 5 77 11 - 1000
>      >> F  +43 (0) 5 77 11 - 1002
>      >> E  christian.loeschenkohl at xpirio.com
>     <mailto:christian.loeschenkohl at xpirio.com>
>      >>
>      >> _______________________________________________
>      >> FreeSWITCH-users mailing list
>      >> FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>      >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>      >>
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>      >> http://www.freeswitch.org
>      >>
>      >
>      >
>      >
>      >
>      > _______________________________________________
>      > FreeSWITCH-users mailing list
>      > FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>      > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>      >
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>      > http://www.freeswitch.org
> 
> 
>     --
>     Ing. Christian L?schenkohl
>     Technische Leitung, Forschung & Entwicklung VoIP
> 
>     xpirio
>     Telekommunikation & Service GmbH
>     Lakeside B04
>     9020 Klagenfurt
>     Austria
> 
>     T  +43 (0) 5 77 11 - 1000
>     F  +43 (0) 5 77 11 - 1002
>     E  christian.loeschenkohl at xpirio.com
>     <mailto:christian.loeschenkohl at xpirio.com>
> 
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
> 
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com 
> <mailto:MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com 
> <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org 
> <mailto:sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888 
> <http://iax:guest at conference.freeswitch.org/888>
> googletalk:conf+888 at conference.freeswitch.org 
> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


-- 
Ing. Christian L?schenkohl
Technische Leitung, Forschung & Entwicklung VoIP

xpirio
Telekommunikation & Service GmbH
Lakeside B04
9020 Klagenfurt
Austria

T  +43 (0) 5 77 11 - 1000
F  +43 (0) 5 77 11 - 1002
E  christian.loeschenkohl at xpirio.com


From brian at freeswitch.org  Mon Feb 22 14:35:36 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 22 Feb 2010 16:35:36 -0600
Subject: [Freeswitch-users] ESF_PAGE_GROUP in a box with 2 interfaces
In-Reply-To: <f120f4c81002221427x78956438oc8b54cd1b85cd76d@mail.gmail.com>
References: <f120f4c81002221305r7c9b4dedw13ccf5f72eb75d05@mail.gmail.com>
	<191c3a031002221337t3c49cda7r14d80c71d510a8e9@mail.gmail.com>
	<f120f4c81002221427x78956438oc8b54cd1b85cd76d@mail.gmail.com>
Message-ID: <328BF8BA-3BBB-4C57-8C79-73BC681E8C87@freeswitch.org>

The args to the app are the ip and port to send on... 

/b

On Feb 22, 2010, at 4:27 PM, Luis F Urrea wrote:

> Since this would be in the category of a host operation and there would be no need to forward multicast traffic between interfaces I thought that maybe there could be code in the esf application that would choose the IP it would bound to.



From dftoro at yahoo.com  Mon Feb 22 14:47:25 2010
From: dftoro at yahoo.com (Diego Toro)
Date: Mon, 22 Feb 2010 14:47:25 -0800 (PST)
Subject: [Freeswitch-users] sending a sip notify with sendevent
In-Reply-To: <4B830517.2070402@xpirio.com>
Message-ID: <959367.79243.qm@web33508.mail.mud.yahoo.com>

Hi, Read "case SWITCH_EVENT_NOTIFY:" in mod_sofia.c. I suggest you assign call-id in the header of the event.


Diego Toro
http://lacarretade.blogspot.com/



--- On Mon, 2/22/10, Christian L?schenkohl <christian.loeschenkohl at xpirio.com> wrote:

> From: Christian L?schenkohl <christian.loeschenkohl at xpirio.com>
> Subject: Re: [Freeswitch-users] sending a sip notify with sendevent
> To: freeswitch-users at lists.freeswitch.org
> Date: Monday, February 22, 2010, 5:28 PM
> hi anthony
> 
> i did it
> 
> my profile is actually named nat, "sofia status profile
> nat" shows me presence_nat as the db name
> so i had a look in the file presence_nat.db
> 
> i execute
> select
> sip_user,sip_host,contact,profile_name,'application/simple-message-summary','check-sync;reboot=false',''
> from sip_registrations where sip_user='10' and
> sip_host='vts.vie.xpirio.net';
> 
> and it returns
> 10|vts.vie.xpirio.net|"10"
> <sip:10 at 81.94.53.5:1032>|nat|application/simple-message-summary|check-sync;reboot=false|
> 
> looks good to me so far, but as i said no sip notify
> message is send to the client
> 
> br
> 
> Anthony Minessale wrote:
> 
> > compare that sql stmt to your db manually with the
> sqlite3 app
> > 
> > sqlite3
> /usr/local/freeswitch/db/sofia_reg_internal.db
> > 
> > 
> > 2010/2/22 Christian L?schenkohl <christian.loeschenkohl at xpirio.com
> 
> > <mailto:christian.loeschenkohl at xpirio.com>>
> > 
> >? ???thank you for this advise
> > 
> >? ???i read the section "case
> SWITCH_EVENT_NOTIFY" carefully, debugged my
> >? ???script (i messed something
> >? ???up, with telnet the command
> works - returns Reply-Text: +OK)
> > 
> >? ???sql is executed and returns 1
> row
> >? ???select
> >?
> ???sip_user,sip_host,contact,profile_name,'application/simple-message-summary','check-sync;reboot=true',''
> >? ???from sip_registrations where
> sip_user='10' and
> >? ???sip_host='vts.vie.xpirio.net
> <http://vts.vie.xpirio.net>'
> > 
> >? ???however no notify message is
> send to the device
> > 
> >? ???i can't use call-id because i
> simply don't know it
> > 
> >? ???br
> > 
> > 
> >? ???Diego Toro wrote:
> > 
> >? ? ? > Read "case
> SWITCH_EVENT_NOTIFY:" in mod_sofia.c. I suggest you
> >? ???assign call-id in the header
> of the event.
> >? ? ? >
> >? ? ? >
> >? ? ? > Diego Toro
> >? ? ? > http://lacarretade.blogspot.com/
> >? ? ? >
> >? ? ? >
> >? ? ? > --- On Mon, 2/22/10,
> Christian L?schenkohl
> >? ???<christian.loeschenkohl at xpirio.com
> >? ???<mailto:christian.loeschenkohl at xpirio.com>>
> wrote:
> >? ? ? >
> >? ? ? >> From: Christian
> L?schenkohl <christian.loeschenkohl at xpirio.com
> >? ???<mailto:christian.loeschenkohl at xpirio.com>>
> >? ? ? >> Subject:
> [Freeswitch-users] sending a sip notify with sendevent
> >? ? ? >> To: freeswitch-users at lists.freeswitch.org
> >? ???<mailto:freeswitch-users at lists.freeswitch.org>
> >? ? ? >> Date: Monday, February
> 22, 2010, 12:16 PM
> >? ? ? >> hi
> >? ? ? >>
> >? ? ? >> i try to send a sip
> notify message to a registered sip
> >? ? ? >> device
> >? ? ? >> "sofia profile nat
> flush_inbound_reg 10 at vts.vie.xpirio.net
> >? ???<mailto:10 at vts.vie.xpirio.net>
> >? ? ? >> reboot" works, but i
> need
> >? ? ? >> to send
> "check-sync;reboot=false" - so the device does a
> >? ? ? >> resync and don't do a
> reboot
> >? ? ? >>
> >? ? ? >> my message looks like
> this
> >? ? ? >>
> >? ? ? >> sendevent NOTIFY
> >? ? ? >> profile: nat
> >? ? ? >> event-string:
> check-sync;reboot=false
> >? ? ? >> user: 10
> >? ? ? >> host: vts.vie.xpirio.net
> <http://vts.vie.xpirio.net>
> >? ? ? >> content-type:
> application/simple-message-summary
> >? ? ? >>
> >? ? ? >> if i listen on the
> loopback interface i do see
> >? ? ? >>
> >? ? ? >> ##
> >? ? ? >> T 2010/02/22
> 18:11:59.083204 127.0.0.1:51840
> >? ???<http://127.0.0.1:51840> ->
> >? ? ? >> 127.0.0.1:8021 <http://127.0.0.1:8021> [AP]
> >? ? ? >> sendevent NOTIFY
> >? ? ? >> profile: nat
> >? ? ? >> event-string:
> check-sync;reboot=false
> >? ? ? >> user: 10
> >? ? ? >> host: vts.vie.xpirio.net
> <http://vts.vie.xpirio.net>
> >? ? ? >> content-type:
> application/simple-message-summary
> >? ? ? >>
> >? ? ? >> ##
> >? ? ? >> T 2010/02/22
> 18:11:59.084032 127.0.0.1:8021
> >? ???<http://127.0.0.1:8021> ->
> >? ? ? >> 127.0.0.1:51840 <http://127.0.0.1:51840> [AP]
> >? ? ? >> Content-Type:
> command/reply
> >? ? ? >> Reply-Text: -ERR invalid
> >? ? ? >>
> >? ? ? >> --------
> >? ? ? >> i don't get what it is
> wrong. i also rechecked the
> >? ? ? >> registered user in the
> sqlite database and this
> >? ? ? >> looks good to me.
> >? ? ? >>
> >? ? ? >> no message is send to the
> user.
> >? ? ? >>
> >? ? ? >> we do use multiple
> domains, so user could also be
> >? ???10 at somedomain.com
> <mailto:10 at somedomain.com>
> >? ? ? >> - or am i wrong on this?
> >? ? ? >> could somebody please
> bring some light in this.
> >? ? ? >>
> >? ? ? >> we do use trunk rev.
> 16631
> >? ? ? >>
> >? ? ? >> br
> >? ? ? >>
> >? ? ? >>
> >? ? ? >>
> >? ? ? >> --
> >? ? ? >> Ing. Christian
> L?schenkohl
> >? ? ? >> Technische Leitung,
> Forschung & Entwicklung VoIP
> >? ? ? >>
> >? ? ? >> xpirio
> >? ? ? >> Telekommunikation &
> Service GmbH
> >? ? ? >> Lakeside B04
> >? ? ? >> 9020 Klagenfurt
> >? ? ? >> Austria
> >? ? ? >>
> >? ? ? >> T? +43 (0) 5 77 11 -
> 1000
> >? ? ? >> F? +43 (0) 5 77 11 -
> 1002
> >? ? ? >> E? christian.loeschenkohl at xpirio.com
> >? ???<mailto:christian.loeschenkohl at xpirio.com>
> >? ? ? >>
> >? ? ? >>
> _______________________________________________
> >? ? ? >> FreeSWITCH-users mailing
> list
> >? ? ? >> FreeSWITCH-users at lists.freeswitch.org
> >? ???<mailto:FreeSWITCH-users at lists.freeswitch.org>
> >? ? ? >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >? ? ? >>
> >? ???UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >? ? ? >> http://www.freeswitch.org
> >? ? ? >>
> >? ? ? >
> >? ? ? >
> >? ? ? >
> >? ? ? >
> >? ? ? >
> _______________________________________________
> >? ? ? > FreeSWITCH-users mailing
> list
> >? ? ? > FreeSWITCH-users at lists.freeswitch.org
> >? ???<mailto:FreeSWITCH-users at lists.freeswitch.org>
> >? ? ? > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >? ? ? >
> >? ???UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >? ? ? > http://www.freeswitch.org
> > 
> > 
> >? ???--
> >? ???Ing. Christian L?schenkohl
> >? ???Technische Leitung, Forschung
> & Entwicklung VoIP
> > 
> >? ???xpirio
> >? ???Telekommunikation &
> Service GmbH
> >? ???Lakeside B04
> >? ???9020 Klagenfurt
> >? ???Austria
> > 
> >? ???T? +43 (0) 5 77 11 -
> 1000
> >? ???F? +43 (0) 5 77 11 -
> 1002
> >? ???E? christian.loeschenkohl at xpirio.com
> >? ???<mailto:christian.loeschenkohl at xpirio.com>
> > 
> >?
> ???_______________________________________________
> >? ???FreeSWITCH-users mailing list
> >? ???FreeSWITCH-users at lists.freeswitch.org
> >? ???<mailto:FreeSWITCH-users at lists.freeswitch.org>
> >? ???http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >? ???UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >? ???http://www.freeswitch.org
> > 
> > 
> > 
> > 
> > -- 
> > Anthony Minessale II
> > 
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> > Twitter: http://twitter.com/FreeSWITCH_wire
> > 
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com
> 
> > <mailto:MSN%3Aanthony_minessale at hotmail.com>
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> 
> > <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
> > IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
> > 
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org
> 
> > <mailto:sip%3A888 at conference.freeswitch.org>
> > iax:guest at conference.freeswitch.org/888
> 
> > <http://iax:guest at conference.freeswitch.org/888>
> > googletalk:conf+888 at conference.freeswitch.org
> 
> > <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > pstn:+19193869900
> > 
> > 
> >
> ------------------------------------------------------------------------
> > 
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 
> 
> -- 
> Ing. Christian L?schenkohl
> Technische Leitung, Forschung & Entwicklung VoIP
> 
> xpirio
> Telekommunikation & Service GmbH
> Lakeside B04
> 9020 Klagenfurt
> Austria
> 
> T? +43 (0) 5 77 11 - 1000
> F? +43 (0) 5 77 11 - 1002
> E? christian.loeschenkohl at xpirio.com
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 


      


From lfurrea at gmail.com  Mon Feb 22 16:12:16 2010
From: lfurrea at gmail.com (Luis F Urrea)
Date: Mon, 22 Feb 2010 18:12:16 -0600
Subject: [Freeswitch-users] ESF_PAGE_GROUP in a box with 2 interfaces
In-Reply-To: <328BF8BA-3BBB-4C57-8C79-73BC681E8C87@freeswitch.org>
References: <f120f4c81002221305r7c9b4dedw13ccf5f72eb75d05@mail.gmail.com>
	<191c3a031002221337t3c49cda7r14d80c71d510a8e9@mail.gmail.com>
	<f120f4c81002221427x78956438oc8b54cd1b85cd76d@mail.gmail.com>
	<328BF8BA-3BBB-4C57-8C79-73BC681E8C87@freeswitch.org>
Message-ID: <f120f4c81002221612q6313de01h2bd4c6a53022e41f@mail.gmail.com>

Thx Brian,

I understand that you can set the *destination* IP:Port via variables, but I
was concerned with the source interface of the multicast traffic.

And I just confirmed that on FreeBSD you do not need to specify a route in
the single interface case because:

"the default multicast route is via the interface
with the default route; setting a route isn't necessary unless you need to
force multicast to go via a particular interface by default, this is done
by longest-prefix matching like all other IPv4 routing activities."

They also state that:

"An unprivileged userland application is also able to control where it is
sending its multicast traffic (without mucking with the routing table)
by using the sockopt IP_MULTICAST_IF.  It can specify the address of
any interface on the
machine"

But this would really be a hassle for the programmer :)

Thx for your help!


On Mon, Feb 22, 2010 at 4:35 PM, Brian West <brian at freeswitch.org> wrote:

> The args to the app are the ip and port to send on...
>
> /b
>
> On Feb 22, 2010, at 4:27 PM, Luis F Urrea wrote:
>
> > Since this would be in the category of a host operation and there would
> be no need to forward multicast traffic between interfaces I thought that
> maybe there could be code in the esf application that would choose the IP it
> would bound to.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100222/14f37b15/attachment.html 

From lfurrea at gmail.com  Mon Feb 22 16:22:27 2010
From: lfurrea at gmail.com (Luis F Urrea)
Date: Mon, 22 Feb 2010 18:22:27 -0600
Subject: [Freeswitch-users] ESF_PAGE_GROUP in a box with 2 interfaces
In-Reply-To: <f120f4c81002221612q6313de01h2bd4c6a53022e41f@mail.gmail.com>
References: <f120f4c81002221305r7c9b4dedw13ccf5f72eb75d05@mail.gmail.com>
	<191c3a031002221337t3c49cda7r14d80c71d510a8e9@mail.gmail.com>
	<f120f4c81002221427x78956438oc8b54cd1b85cd76d@mail.gmail.com>
	<328BF8BA-3BBB-4C57-8C79-73BC681E8C87@freeswitch.org>
	<f120f4c81002221612q6313de01h2bd4c6a53022e41f@mail.gmail.com>
Message-ID: <f120f4c81002221622j6f77473asd1279adfbb7c97f8@mail.gmail.com>

wiki updated

On Mon, Feb 22, 2010 at 6:12 PM, Luis F Urrea <lfurrea at gmail.com> wrote:

> Thx Brian,
>
> I understand that you can set the *destination* IP:Port via variables, but
> I was concerned with the source interface of the multicast traffic.
>
> And I just confirmed that on FreeBSD you do not need to specify a route in
> the single interface case because:
>
> "the default multicast route is via the interface
> with the default route; setting a route isn't necessary unless you need to
> force multicast to go via a particular interface by default, this is done
>
> by longest-prefix matching like all other IPv4 routing activities."
>
> They also state that:
>
> "An unprivileged userland application is also able to control where it is
> sending its multicast traffic (without mucking with the routing table)
>
> by using the sockopt IP_MULTICAST_IF.  It can specify the address of any interface on the
> machine"
>
> But this would really be a hassle for the programmer :)
>
> Thx for your help!
>
>
> On Mon, Feb 22, 2010 at 4:35 PM, Brian West <brian at freeswitch.org> wrote:
>
>> The args to the app are the ip and port to send on...
>>
>> /b
>>
>> On Feb 22, 2010, at 4:27 PM, Luis F Urrea wrote:
>>
>> > Since this would be in the category of a host operation and there would
>> be no need to forward multicast traffic between interfaces I thought that
>> maybe there could be code in the esf application that would choose the IP it
>> would bound to.
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100222/0174a341/attachment.html 

From joseph.puchalski at personalcyberspace.com  Mon Feb 22 16:24:41 2010
From: joseph.puchalski at personalcyberspace.com (Joseph Puchalski)
Date: Tue, 23 Feb 2010 00:24:41 +0000
Subject: [Freeswitch-users] How to set outbound caller id info
	for	multiple users/extensions
In-Reply-To: <3FF62C88-2423-43F0-B8A3-C64EF4BC80AC@freeswitch.org>
References: <093DD565390C1E4FB15D7B383E86BB05AF05AB@Goose.personalcyberspace.net>
	<191c3a031002130707v5f1d5ee0lb9e791e9a0aab956@mail.gmail.com>
	<3FF62C88-2423-43F0-B8A3-C64EF4BC80AC@freeswitch.org>
Message-ID: <093DD565390C1E4FB15D7B383E86BB05AF15CD@Goose.personalcyberspace.net>

Thanks for the replies. Since then I've poked around in the wiki and experimented with my config files.

It seems as if I was setting the caller ID for all traffic outbound through my trunk provider:

>From default.xml in /opt/freeswitch/conf/dialplan/

  <extension name="vitel-outbound">
   <condition field="destination_number" expression="^(1{0,1}\d{10})$">
   <action application="set" data="effective_caller_id_number=1234567890"/>
     <action application="bridge" data="sofia/gateway/vitelity-outbound/$1"/>
    </condition>
  </extension>

This worked, insofar as outbound calls carried the caller ID as configured.

I removed the line:

   <action application="set" data="effective_caller_id_number=1234567890"/>

I had hoped that this would allow the value set in 5859.xml to take effect. It didn't. Instead my calls go out without any Caller ID.

I'm obviously missing something.

Is it possible to explicitly set the outbound caller ID for an extension when configuring it? I've tried to do so as follows:

>From 5859.xml

<include>
  <user id="5859">
    <params>
      <param name="password" value="XXX"/>
      <param name="vm-password" value="XXX"/>
    </params>
    <variables>
..
      <variable name="user_context" value="default"/>
      <variable name="effective_caller_id_name" value="2034135859"/>
      <variable name="effective_caller_id_number" value="2034135859"/>
..
    </variables>
  </user>
</include>

Or should I be doing this via <action application="set" data="effective_caller_id_number=${outbound_caller_id_number}"/>
Somewhere else?

I did try to add a <condition> line in my vitelity config area to set "effective_caller_id"  based on originating number. I had no success, possibly because I was checking the  wrong variable, possibly because that's the wrong place to do it.

I've gone back and gotten a much better grounding in XML, but there are still more than enough simultaneously moving parts in FreeSWITCH to make me feel pretty clueless at the moment . I apologize if I'm totally missing something obvious.

Thanks again for any help. The capabilities of  FreeSWITCH continue to amaze me, sufficiently so that I won't be happy until I've got my head wrapped at least part way around it.

Joe P.


From: Brian West [mailto:brian at freeswitch.org]
Sent: Saturday, February 13, 2010 10:17 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] How to set outbound caller id info for multiple users/extensions

I also have to point out their is no such official variable for "outbound_caller_id_name" or "outbound_caller_id_number", Those are just made up variables I used in the default config.

01_example.com.xml

You'll notice I use these lines.

      <action application="set" data="effective_caller_id_number=${outbound_caller_id_number}"/>
      <action application="set" data="effective_caller_id_name=${outbound_caller_id_name}"/>

Its just a way to set the users default outbound caller ID .

/b


On Feb 13, 2010, at 9:07 AM, Anthony Minessale wrote:



It should be covered on the wiki http://wiki.freeswitch.org<http://wiki.freeswitch.org/>

On Feb 12, 2010 6:23 PM, "Joseph Puchalski" <joseph.puchalski at personalcyberspace.com<mailto:joseph.puchalski at personalcyberspace.com>> wrote:
I'm having problems setting different outbound caller id info for different extensions/users.

I've set up a small system with two active users. I set up my users by copying and modifying existing entries from the dialplan files that come with freeSWITCH

Here are my two extensions:

<include>
  <user id="5859">
    <params>
      <param name="password" value="xxxxxx"/>
      <param name="vm-password" value="xxxxxx"/>
    </params>
    <variables>
      <variable name="toll_allow" value="domestic,international,local"/>
      <variable name="accountcode" value="5859"/>
      <variable name="user_context" value="default"/>
      <variable name="effective_caller_id_name" value="5859"/>
      <variable name="effective_caller_id_number" value="5859"/>
      <variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/>
      <variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/>
      <variable name="callgroup" value="techsupport"/>
    </variables>
  </user>
</include>

<include>
  <user id="5515">
    <params>
      <param name="password" value="xxxxxxx"/>
      <param name="vm-password" value="xxxxxxx"/>
    </params>
    <variables>
      <variable name="toll_allow" value="domestic,international,local"/>
      <variable name="accountcode" value="5515"/>
      <variable name="user_context" value="default"/>
      <variable name="effective_caller_id_name" value="5515"/>
      <variable name="effective_caller_id_number" value="5515"/>
      <variable name="outbound_caller_id_name" value="Jane Doe"/>
      <variable name="outbound_caller_id_number" value="5558395515"/>
      <variable name="callgroup" value="techsupport"/>
    </variables>
  </user>
</include>

These extensions are in files named 5859.xml and 5515.xml respectively.

I'm using a SIP trunk from Vitelity (in and out) with two DIDs corresponding to the extensions above.

Inbound and outbound calling work as needed with one exception: Calls originating from user/extension 5515 go out with the caller ID of extension/user 5859.

Extension 5859 was the first that I created.

Where should I be setting the outbound caller id number for my second extension? I've been trying to track this down in the available documentation but have been unable to do so.

I apologize ahead of time if this is answered somewhere obvious that I've missed.

Thanks for any help.

Joe (FreeSWITCH newbie)

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org<http://www.freeswitch.org/>

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100223/b7d83763/attachment-0001.html 

From matt at webcontracts.co.uk  Mon Feb 22 16:55:12 2010
From: matt at webcontracts.co.uk (Matthew Law)
Date: Tue, 23 Feb 2010 00:55:12 -0000
Subject: [Freeswitch-users] How to debug time-based routing?
Message-ID: <eb2506f03e9045c509a6974bcc4a3a49.squirrel@www.webcontracts.co.uk>


After some playing around I now have a working config but it appears to be
routing calls straight to voicemail based on time of day.  I did see the
example of this in the default config, commented it out and reloaded but I
cannot see anything in the log output to verify this.

Looking at the log output, I can see it hit the regex for the outside
number, match that and then get sent to extension 1000 (which is correct)
but at that point it is sent straight to voicemail and I'm stumped.

Can someone please explain how to go about debugging this?  I have the log
level set to 7, if that helps.


Many thanks,

Matt.



From brian at freeswitch.org  Mon Feb 22 17:00:45 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 22 Feb 2010 19:00:45 -0600
Subject: [Freeswitch-users] How to debug time-based routing?
In-Reply-To: <eb2506f03e9045c509a6974bcc4a3a49.squirrel@www.webcontracts.co.uk>
References: <eb2506f03e9045c509a6974bcc4a3a49.squirrel@www.webcontracts.co.uk>
Message-ID: <74B270D5-E134-4221-A0FE-8275B05826A5@freeswitch.org>

Lets start with how about you pastebin your extension and logs... or better join #freeswitch on irc.freenode.net?  ;)

/b

On Feb 22, 2010, at 6:55 PM, Matthew Law wrote:

> Can someone please explain how to go about debugging this?  I have the log
> level set to 7, if that helps.



From andrewkt at aktzero.com  Mon Feb 22 19:10:17 2010
From: andrewkt at aktzero.com (Andrew Thompson)
Date: Mon, 22 Feb 2010 22:10:17 -0500
Subject: [Freeswitch-users] How to set outbound caller id info for
 multiple users/extensions
In-Reply-To: <093DD565390C1E4FB15D7B383E86BB05AF15CD@Goose.personalcyberspace.net>
References: <093DD565390C1E4FB15D7B383E86BB05AF05AB@Goose.personalcyberspace.net>	<191c3a031002130707v5f1d5ee0lb9e791e9a0aab956@mail.gmail.com>	<3FF62C88-2423-43F0-B8A3-C64EF4BC80AC@freeswitch.org>
	<093DD565390C1E4FB15D7B383E86BB05AF15CD@Goose.personalcyberspace.net>
Message-ID: <4B834719.3000505@aktzero.com>

On 2/22/2010 7:24 PM, Joseph Puchalski wrote:
>
> Or should I be doing this via <action application="set" 
> data="effective_caller_id_number=${outbound_caller_id_number}"/>
>
> Somewhere else?
>

I have the following set on my own extension, in 1000.xml:

<variable name="effective_caller_id_name" value="Andrew Thompson"/>
<variable name="effective_caller_id_number" value="1000"/>
<variable name="outbound_caller_id_name" value="MyCompany"/>
<variable name="outbound_caller_id_number" value="9109991234"/>

When I dial extensions internally, the effective_* name/number show up.

When I dial outbound via my SIP provider, I set the following before the 
bridge so that it passes externally valid info:

<action application="set" 
data="effective_caller_id_number=${outbound_caller_id_number}"/>
<action application="set" 
data="effective_caller_id_name=${outbound_caller_id_name}"/>


In my setup, if I don't explicitly overide the effective_* with 
outbound_*, I actually see 1000 as my callerid when I call my cell from 
my extension, so if you're not getting at least that much, something 
else might be wrong. (I have used vitelity, and they do pass callerid 
properly most of the time.)

-- 
Andrew Thompson

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100222/650f5e1b/attachment.html 

From dftoro at yahoo.com  Mon Feb 22 19:18:03 2010
From: dftoro at yahoo.com (Diego Toro)
Date: Mon, 22 Feb 2010 19:18:03 -0800 (PST)
Subject: [Freeswitch-users] sending a sip notify with sendevent
In-Reply-To: <4B830517.2070402@xpirio.com>
Message-ID: <12033.93456.qm@web33507.mail.mud.yahoo.com>

hi, I'm using 

sendevent NOTIFY
profile: internal
event-string: check-sync;reboot=false
user: 1001
content-type: application/simple-message-sumary
profile: internal
event-string: check-sync;reboot=false
host: 192.168.7.3


where:
user 1001 is a registered user
host: IP of FreeSwitch

I see a sip notify message sent to the client.

Diego Toro 
http://lacarretade.blogspot.com/


--- On Mon, 2/22/10, Christian L?schenkohl <christian.loeschenkohl at xpirio.com> wrote:

> From: Christian L?schenkohl <christian.loeschenkohl at xpirio.com>
> Subject: Re: [Freeswitch-users] sending a sip notify with sendevent
> To: freeswitch-users at lists.freeswitch.org
> Date: Monday, February 22, 2010, 5:28 PM
> hi anthony
> 
> i did it
> 
> my profile is actually named nat, "sofia status profile
> nat" shows me presence_nat as the db name
> so i had a look in the file presence_nat.db
> 
> i execute
> select
> sip_user,sip_host,contact,profile_name,'application/simple-message-summary','check-sync;reboot=false',''
> from sip_registrations where sip_user='10' and
> sip_host='vts.vie.xpirio.net';
> 
> and it returns
> 10|vts.vie.xpirio.net|"10"
> <sip:10 at 81.94.53.5:1032>|nat|application/simple-message-summary|check-sync;reboot=false|
> 
> looks good to me so far, but as i said no sip notify
> message is send to the client
> 
> br
> 
> Anthony Minessale wrote:
> 
> > compare that sql stmt to your db manually with the
> sqlite3 app
> > 
> > sqlite3
> /usr/local/freeswitch/db/sofia_reg_internal.db
> > 
> > 
> > 2010/2/22 Christian L?schenkohl <christian.loeschenkohl at xpirio.com
> 
> > <mailto:christian.loeschenkohl at xpirio.com>>
> > 
> >? ???thank you for this advise
> > 
> >? ???i read the section "case
> SWITCH_EVENT_NOTIFY" carefully, debugged my
> >? ???script (i messed something
> >? ???up, with telnet the command
> works - returns Reply-Text: +OK)
> > 
> >? ???sql is executed and returns 1
> row
> >? ???select
> >?
> ???sip_user,sip_host,contact,profile_name,'application/simple-message-summary','check-sync;reboot=true',''
> >? ???from sip_registrations where
> sip_user='10' and
> >? ???sip_host='vts.vie.xpirio.net
> <http://vts.vie.xpirio.net>'
> > 
> >? ???however no notify message is
> send to the device
> > 
> >? ???i can't use call-id because i
> simply don't know it
> > 
> >? ???br
> > 
> > 
> >? ???Diego Toro wrote:
> > 
> >? ? ? > Read "case
> SWITCH_EVENT_NOTIFY:" in mod_sofia.c. I suggest you
> >? ???assign call-id in the header
> of the event.
> >? ? ? >
> >? ? ? >
> >? ? ? > Diego Toro
> >? ? ? > http://lacarretade.blogspot.com/
> >? ? ? >
> >? ? ? >
> >? ? ? > --- On Mon, 2/22/10,
> Christian L?schenkohl
> >? ???<christian.loeschenkohl at xpirio.com
> >? ???<mailto:christian.loeschenkohl at xpirio.com>>
> wrote:
> >? ? ? >
> >? ? ? >> From: Christian
> L?schenkohl <christian.loeschenkohl at xpirio.com
> >? ???<mailto:christian.loeschenkohl at xpirio.com>>
> >? ? ? >> Subject:
> [Freeswitch-users] sending a sip notify with sendevent
> >? ? ? >> To: freeswitch-users at lists.freeswitch.org
> >? ???<mailto:freeswitch-users at lists.freeswitch.org>
> >? ? ? >> Date: Monday, February
> 22, 2010, 12:16 PM
> >? ? ? >> hi
> >? ? ? >>
> >? ? ? >> i try to send a sip
> notify message to a registered sip
> >? ? ? >> device
> >? ? ? >> "sofia profile nat
> flush_inbound_reg 10 at vts.vie.xpirio.net
> >? ???<mailto:10 at vts.vie.xpirio.net>
> >? ? ? >> reboot" works, but i
> need
> >? ? ? >> to send
> "check-sync;reboot=false" - so the device does a
> >? ? ? >> resync and don't do a
> reboot
> >? ? ? >>
> >? ? ? >> my message looks like
> this
> >? ? ? >>
> >? ? ? >> sendevent NOTIFY
> >? ? ? >> profile: nat
> >? ? ? >> event-string:
> check-sync;reboot=false
> >? ? ? >> user: 10
> >? ? ? >> host: vts.vie.xpirio.net
> <http://vts.vie.xpirio.net>
> >? ? ? >> content-type:
> application/simple-message-summary
> >? ? ? >>
> >? ? ? >> if i listen on the
> loopback interface i do see
> >? ? ? >>
> >? ? ? >> ##
> >? ? ? >> T 2010/02/22
> 18:11:59.083204 127.0.0.1:51840
> >? ???<http://127.0.0.1:51840> ->
> >? ? ? >> 127.0.0.1:8021 <http://127.0.0.1:8021> [AP]
> >? ? ? >> sendevent NOTIFY
> >? ? ? >> profile: nat
> >? ? ? >> event-string:
> check-sync;reboot=false
> >? ? ? >> user: 10
> >? ? ? >> host: vts.vie.xpirio.net
> <http://vts.vie.xpirio.net>
> >? ? ? >> content-type:
> application/simple-message-summary
> >? ? ? >>
> >? ? ? >> ##
> >? ? ? >> T 2010/02/22
> 18:11:59.084032 127.0.0.1:8021
> >? ???<http://127.0.0.1:8021> ->
> >? ? ? >> 127.0.0.1:51840 <http://127.0.0.1:51840> [AP]
> >? ? ? >> Content-Type:
> command/reply
> >? ? ? >> Reply-Text: -ERR invalid
> >? ? ? >>
> >? ? ? >> --------
> >? ? ? >> i don't get what it is
> wrong. i also rechecked the
> >? ? ? >> registered user in the
> sqlite database and this
> >? ? ? >> looks good to me.
> >? ? ? >>
> >? ? ? >> no message is send to the
> user.
> >? ? ? >>
> >? ? ? >> we do use multiple
> domains, so user could also be
> >? ???10 at somedomain.com
> <mailto:10 at somedomain.com>
> >? ? ? >> - or am i wrong on this?
> >? ? ? >> could somebody please
> bring some light in this.
> >? ? ? >>
> >? ? ? >> we do use trunk rev.
> 16631
> >? ? ? >>
> >? ? ? >> br
> >? ? ? >>
> >? ? ? >>
> >? ? ? >>
> >? ? ? >> --
> >? ? ? >> Ing. Christian
> L?schenkohl
> >? ? ? >> Technische Leitung,
> Forschung & Entwicklung VoIP
> >? ? ? >>
> >? ? ? >> xpirio
> >? ? ? >> Telekommunikation &
> Service GmbH
> >? ? ? >> Lakeside B04
> >? ? ? >> 9020 Klagenfurt
> >? ? ? >> Austria
> >? ? ? >>
> >? ? ? >> T? +43 (0) 5 77 11 -
> 1000
> >? ? ? >> F? +43 (0) 5 77 11 -
> 1002
> >? ? ? >> E? christian.loeschenkohl at xpirio.com
> >? ???<mailto:christian.loeschenkohl at xpirio.com>
> >? ? ? >>
> >? ? ? >>
> _______________________________________________
> >? ? ? >> FreeSWITCH-users mailing
> list
> >? ? ? >> FreeSWITCH-users at lists.freeswitch.org
> >? ???<mailto:FreeSWITCH-users at lists.freeswitch.org>
> >? ? ? >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >? ? ? >>
> >? ???UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >? ? ? >> http://www.freeswitch.org
> >? ? ? >>
> >? ? ? >
> >? ? ? >
> >? ? ? >
> >? ? ? >
> >? ? ? >
> _______________________________________________
> >? ? ? > FreeSWITCH-users mailing
> list
> >? ? ? > FreeSWITCH-users at lists.freeswitch.org
> >? ???<mailto:FreeSWITCH-users at lists.freeswitch.org>
> >? ? ? > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >? ? ? >
> >? ???UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >? ? ? > http://www.freeswitch.org
> > 
> > 
> >? ???--
> >? ???Ing. Christian L?schenkohl
> >? ???Technische Leitung, Forschung
> & Entwicklung VoIP
> > 
> >? ???xpirio
> >? ???Telekommunikation &
> Service GmbH
> >? ???Lakeside B04
> >? ???9020 Klagenfurt
> >? ???Austria
> > 
> >? ???T? +43 (0) 5 77 11 -
> 1000
> >? ???F? +43 (0) 5 77 11 -
> 1002
> >? ???E? christian.loeschenkohl at xpirio.com
> >? ???<mailto:christian.loeschenkohl at xpirio.com>
> > 
> >?
> ???_______________________________________________
> >? ???FreeSWITCH-users mailing list
> >? ???FreeSWITCH-users at lists.freeswitch.org
> >? ???<mailto:FreeSWITCH-users at lists.freeswitch.org>
> >? ???http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >? ???UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >? ???http://www.freeswitch.org
> > 
> > 
> > 
> > 
> > -- 
> > Anthony Minessale II
> > 
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> > Twitter: http://twitter.com/FreeSWITCH_wire
> > 
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com
> 
> > <mailto:MSN%3Aanthony_minessale at hotmail.com>
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> 
> > <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
> > IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
> > 
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org
> 
> > <mailto:sip%3A888 at conference.freeswitch.org>
> > iax:guest at conference.freeswitch.org/888
> 
> > <http://iax:guest at conference.freeswitch.org/888>
> > googletalk:conf+888 at conference.freeswitch.org
> 
> > <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > pstn:+19193869900
> > 
> > 
> >
> ------------------------------------------------------------------------
> > 
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 
> 
> -- 
> Ing. Christian L?schenkohl
> Technische Leitung, Forschung & Entwicklung VoIP
> 
> xpirio
> Telekommunikation & Service GmbH
> Lakeside B04
> 9020 Klagenfurt
> Austria
> 
> T? +43 (0) 5 77 11 - 1000
> F? +43 (0) 5 77 11 - 1002
> E? christian.loeschenkohl at xpirio.com
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 


      


From infos at madovsky.org  Mon Feb 22 19:29:01 2010
From: infos at madovsky.org (Madovsky)
Date: Mon, 22 Feb 2010 22:29:01 -0500
Subject: [Freeswitch-users] call from an internal extension to external
	number
Message-ID: <D616CD3E2BFC4FDDA7ABEE7A3F268A4B@MOBILEE1705>

Hi,

day after I undertand a littlee more all these xml hell files (not friendly to read ;)),
but to be a PERl developer since 1999 understand regex and PERL language make life more easy...
However, I don't understand yet the concept of internal exterenal.
is it for phone registration AND outbound calls ?
for now I try to make an external call from 1000 ext (registered on port 5060)
so I added an extension in dialplan/default.xml 

    <extension name="outbound_europe">
        <condition field="destination_number" expression="^00?(\d+)$">
              <action application="bridge" data="sofia/gateway/myprovider_europe/00$1"/>
        </condition>
    </extension>

so if call starts with "00" it redirects to my provider that manage outbound calls, is it correct ?
I put the myprovider.xml account into sip_profiles/external/myprovider.xml.

Thanks for your help

Franck
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100222/166da759/attachment.html 

From tim at communicatefreely.net  Mon Feb 22 20:06:15 2010
From: tim at communicatefreely.net (Tim St. Pierre)
Date: Mon, 22 Feb 2010 23:06:15 -0500
Subject: [Freeswitch-users] Is there any way to loop a dialplan?
In-Reply-To: <4B82E6FA.3090008@marketelsystems.com>
References: <4B82E6FA.3090008@marketelsystems.com>
Message-ID: <4B835437.2070803@communicatefreely.net>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Why not just use transfer?

Break your dialplan up into segments - one that does the authentication, and another that has the
call flow to the parking pool, and the post call work.  You can use the transfer application to
connect these segments together.  Variables are preserved across transfers, so things like the agent
ID and their authenticated status can be set in a variable.  You can also make routing decisions
based on the value of a variable.  In your condition statement, use the variable name
"${my_variable}" as the field, and then a very simple pattern match that decides if it's valid or
not.  You can also use the execute_extension application as a sort of "macro", to execute another
dialplan block, but return when it completes.  At the bottom of your dialplan, transfer back to the top.

Hope that's helpful.  Sorry I don't have any example code.  I'm generating XML dynamically in PHP,
but the above concept seems to work well.

- -Tim

Mark Sobkow wrote:
> Let me explain what it is I'm trying to do.  Maybe there's another way 
> to achieve it.
> 
> When an operator dials in to the log-in line (e.g. Extension 6000), I 
> use play_and_get_digits to collect the operator's PIN.  I then need to 
> be able to fire up some Erlang (or Javascript) to verify the PIN, and 
> after verification, put the call into a park state, collecting the 
> UUID.  I then need to fire an event to Erlang passing along the parked 
> UUID and the operator's PIN so that Erlang can direct received customer 
> calls to the operators based on relatively complex criteria that won't 
> fit in a dialplan.
> 
> The catch is that when I get a customer call, I collect their info via 
> IVR menus, park the call, and fire an event to Erlang with the UUID of 
> the parked call and info collected from the IVR.  Erlang analyses the 
> info, selects an operator who is free, and bridges the calls.
> 
> The problem I'm having is figuring out a way to get the operator leg to 
> go back into a park state after handling the customer's call.  Ideally I 
> want the operator to be presented with a short IVR to collect info about 
> how the call was handled, but I can do that through a custom application 
> GUI if I need to.  Regardless, once an operator logs in, they need to 
> _stay_ logged in until they explicityly log out, but I can't figure out 
> how I'm supposed to do that without some sort of looping capability.
> 
> One thing I was thinking that might work is to set up a set of "dummy" 
> extensions that I can have Erlang dial and bridge which contain a 
> dialplan fragment to collect the IVR call result, park the call, and 
> issue the operator PIN and parked UUID again to Erlang.  That way 
> between Erlang events and the dialplan fragment I end up with an 
> effective "loop".  (Though I've yet to figure out how I can break out of 
> that loop.  Maybe it'll have to be an IVR option for logging out.)
> 
> Sample code/dialplans would be good, but for now I'll settle for knowing 
> whether I'm at least on the right track for how to implement this beast.
> 
> Note that I only want to drop into Javascript if I can't figure out how 
> to do it with dialplans and Erlang.
> 
> Thanks for any ideas and suggestions.
> 

- --
Tim St. Pierre
IP Voice technician
Communicate Freely
1-877-291-8647 x5101
sip:5101 at communicatefreely.net
tim at communicatefreely.net
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v2.0.4 (FreeBSD)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org

iQCVAwUBS4NUN4qVcvNCnHOrAQIDcgP/SpzpLUpsnFjGaamy4EbUw95l2mDHrEYa
ay1cbciSV5qICRLoDvTrleqYkrMhgRlvzxvkLRRzFIOPjm4+cFQMojmMS5HZZQiJ
TWndXAiZGgtlKqEDfgqr1ea2BcXi/oozsJIk0iePgPLIGlMUa/O2p3kaizQzPMc7
fbMNwcSYSc8=
=KtCw
-----END PGP SIGNATURE-----


From troy at tlainvestments.com  Mon Feb 22 20:30:14 2010
From: troy at tlainvestments.com (Troy Anderson)
Date: Mon, 22 Feb 2010 21:30:14 -0700
Subject: [Freeswitch-users] Hook Flash
Message-ID: <915F9D0D-FB13-4D02-995B-FB7F5EB488D2@tlainvestments.com>

Is there a way to cause a hook-flash on a zap channel via the dial plan?  e.g. VoIP phone <=> FS <=> POTS line, and I want to flash via some star code on the phone.

I'm happy to document on the openzap page on the wiki if so.

Thanks,
Troy



From infos at madovsky.org  Mon Feb 22 22:41:04 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 23 Feb 2010 01:41:04 -0500
Subject: [Freeswitch-users] FScomm
Message-ID: <157FDB96FB88476FBAEC69C1309A0A5F@MOBILEE1705>

Hi,

is http://wiki.freeswitch.org/wiki/FSComm
available yet ? because QT framework link has broken
and some instruction don't work with last svn trunk (for example to compile FScomm in FS svn root)

Thanks

Franck
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100223/81c1d0f7/attachment.html 

From infos at madovsky.org  Mon Feb 22 23:06:12 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 23 Feb 2010 02:06:12 -0500
Subject: [Freeswitch-users] rtp timeout and call hangs up
Message-ID: <3F9354A277B544A98E35F8C3F5900496@MOBILEE1705>

Hello,

A leg local extension (codec GSM)  -----> B leg local extension (codec PCMU)
rtp timeout after 300 sec the call hangs up,
I can ear audio and no problem of configuration.
did I forget to set anything ?

below the sip trace :

ACK sip:1001 at 67.xx.xx.138:5060;transport=udp SIP/2.0
Content-Length: 0
Via: SIP/2.0/UDP 67.xx.xx.138:62690;rport;branch=z9hG4bK4t7xGwrintJDYr4HPfz0UQ..
From: "1000" <sip:1000 at 67.xx.xx.138:5080>;tag=52834814554
To: "Extension 1001" <sip:1001 at 67.xx.xx.138>;tag=3SKarSKZB63DF
Contact: <sip:1000 at 67.xx.xx.138:62690>
CSeq: 1 ACK
Max-Forwards: 70
Call-ID: 2053375473 at 67.xx.xx.138

------------------------------------------------------------------------
2010-02-23 02:02:19.649316 [NOTICE] mod_sofia.c:853 Hangup sofia/external/1000 at 67.xx.xx.138:5080 [CS_EXECUTE] [MEDIA_TIMEOUT]
2010-02-23 02:02:19.649316 [NOTICE] switch_ivr_bridge.c:634 Hangup sofia/internal/sip:1001 at 70.81.84.218:2249 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
send 622 bytes to udp/[67.xx.xx.138]:62690 at 07:02:19.653239:
------------------------------------------------------------------------
BYE sip:1000 at 67.xx.xx.138:62690 SIP/2.0
Via: SIP/2.0/UDP 67.xx.xx.138;rport;branch=z9hG4bK790g187j50eNF
Max-Forwards: 70
From: "Extension 1001" <sip:1001 at 67.xx.xx.138>;tag=3SKarSKZB63DF
To: "1000" <sip:1000 at 67.xx.xx.138:5080>;tag=52834814554
Call-ID: 2053375473 at 67.xx.xx.138
CSeq: 127335933 BYE
Contact: <sip:1001 at 67.xx.xx.138:5060;transport=udp>
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16676M
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY
Supported: timer, precondition, path, replaces
Reason: FreeSWITCH;cause=604;text="MEDIA_TIMEOUT"
Content-Length: 0

------------------------------------------------------------------------
2010-02-23 02:02:19.649316 [NOTICE] switch_core_session.c:1179 Session 5 (sofia/external/1000 at 67.xx.xx.138:5080) Ended
2010-02-23 02:02:19.649316 [NOTICE] switch_core_session.c:1181 Close Channel sofia/external/1000 at 67.xx.xx.138:5080 [CS_DESTROY]
recv 289 bytes from udp/[67.xx.xx.138]:62690 at 07:02:19.656927:
------------------------------------------------------------------------
SIP/2.0 200 OK
Content-Length: 0
Via: SIP/2.0/UDP 67.xx.xx.138;rport=5060;branch=z9hG4bK790g187j50eNF
From: "Extension 1001" <sip:1001 at 67.xx.xx.138>;tag=3SKarSKZB63DF
To: "1000" <sip:1000 at 67.xx.xx.138:5080>;tag=52834814554
CSeq: 127335933 BYE
Call-ID: 2053375473 at 67.xx.xx.138

------------------------------------------------------------------------
recv 391 bytes from udp/[67.xx.xx.138]:62690 at 07:02:19.660173:
------------------------------------------------------------------------
BYE sip:1001 at 67.xx.xx.138:5060;transport=udp SIP/2.0
Content-Length: 0
Via: SIP/2.0/UDP 67.xx.xx.138:62690;rport;branch=z9hG4bKGTIQmsc5dUvSdP0ZtJFKcA..
From: "1000" <sip:1000 at 67.xx.xx.138:5080>;tag=52834814554
To: "Extension 1001" <sip:1001 at 67.xx.xx.138>;tag=3SKarSKZB63DF
Contact: <sip:1000 at 67.xx.xx.138:62690>
CSeq: 2 BYE
Max-Forwards: 70
Call-ID: 2053375473 at 67.xx.xx.138

------------------------------------------------------------------------
send 505 bytes to udp/[67.xx.xx.138]:62690 at 07:02:19.660607:
------------------------------------------------------------------------
SIP/2.0 481 Call Does Not Exist
Via: SIP/2.0/UDP 67.xx.xx.138:62690;rport=62690;branch=z9hG4bKGTIQmsc5dUvSdP0ZtJFKcA..
From: "1000" <sip:1000 at 67.xx.xx.138:5080>;tag=52834814554
To: "Extension 1001" <sip:1001 at 67.xx.xx.138>;tag=3SKarSKZB63DF
Call-ID: 2053375473 at 67.xx.xx.138
CSeq: 2 BYE
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16676M
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY
Supported: timer, precondition, path, replaces
Content-Length: 0

------------------------------------------------------------------------
send 669 bytes to udp/[70.81.84.218]:2249 at 07:02:19.869656:
------------------------------------------------------------------------
BYE sip:1001 at 70.81.84.218:2249;rinstance=f17f8a2dd028c23d SIP/2.0
Via: SIP/2.0/UDP 67.xx.xx.138:5080;rport;branch=z9hG4bKQ1y6QNQ56aDDS
Max-Forwards: 70
From: "1000" <sip:1000 at 67.xx.xx.138>;tag=SK3p6Frt5pa7D
To: <sip:1001 at 70.81.84.218:2249;rinstance=f17f8a2dd028c23d>;tag=df099549
Call-ID: 263d4030-9aec-122d-6d87-00e0ed0b00c2
CSeq: 127335918 BYE
Contact: <sip:mod_sofia at 67.xx.xx.138:5080>
User-Agent: FreeSWITCH and Rock!
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, precondition, path, replaces
Reason: Q.850;cause=16;text="NORMAL_CLEARING"
Content-Length: 0

------------------------------------------------------------------------
2010-02-23 02:02:19.869666 [NOTICE] switch_core_session.c:1179 Session 6 (sofia/internal/sip:1001 at 70.81.84.218:2249) Ended
2010-02-23 02:02:19.869666 [NOTICE] switch_core_session.c:1181 Close Channel sofia/internal/sip:1001 at 70.81.84.218:2249 [CS_DESTROY]
recv 411 bytes from udp/[70.81.84.218]:2249 at 07:02:19.986415:
------------------------------------------------------------------------
SIP/2.0 200 OK
Via: SIP/2.0/UDP 67.xx.xx.138:5080;rport=5080;branch=z9hG4bKQ1y6QNQ56aDDS
Contact: <sip:1001 at 70.81.84.218:2249;rinstance=f17f8a2dd028c23d>
To: <sip:1001 at 70.81.84.218:2249;rinstance=f17f8a2dd028c23d>;tag=df099549
From: "1000"<sip:1000 at 67.xx.xx.138>;tag=SK3p6Frt5pa7D
Call-ID: 263d4030-9aec-122d-6d87-00e0ed0b00c2
CSeq: 127335918 BYE
User-Agent: 3CXPhone 4.0.10858.0
Content-Length: 0

------------------------------------------------------------------------





Thanks


Franck
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100223/578bfa97/attachment.html 

From moizchinoy at gmail.com  Mon Feb 22 23:59:04 2010
From: moizchinoy at gmail.com (Moiz Chinoy)
Date: Tue, 23 Feb 2010 11:59:04 +0400
Subject: [Freeswitch-users] Dingaling - external rtp supported?
In-Reply-To: <29b888f81002210800v617bca9dk4bf1abf1bd791ee0@mail.gmail.com>
References: <29b888f81002160835y6e290319ub0d04c088caa1b79@mail.gmail.com>
	<DEA39AD1-0EE9-473A-B78C-CDDF3F100578@freeswitch.org>
	<29b888f81002170318p63622f3fhc8d26ad0e85602d2@mail.gmail.com>
	<191c3a031002170538r3b029515i8a000812a70ad558@mail.gmail.com>
	<29b888f81002170741y3d13cb57ycb4436a372c4eded@mail.gmail.com>
	<191c3a031002171136q7424e9bai34379abb542daf93@mail.gmail.com>
	<29b888f81002210800v617bca9dk4bf1abf1bd791ee0@mail.gmail.com>
Message-ID: <29b888f81002222359w6ae4cd8ej944d453a5b099af4@mail.gmail.com>

Moreover, if I gtalk client is on the same machine as FS and i have
following settings, FS crashes with the same mutex error.

External Sip Profile has following lines:
---------------------------------------------------------
   <param name="rtp-ip" value="$${local_ip_v4}"/>
   <param name="sip-ip" value="$${local_ip_v4}"/>
   <param name="ext-rtp-ip" value="auto-nat"/>
   <param name="ext-sip-ip" value="auto-nat"/>

Jingle Client.xml has following lines:
-----------------------------------------------------
   <param name="rtp-ip" value="$${bind_server_ip}"/>
   <!--<param name="ext-rtp-ip" value="$${external_rtp_ip}"/>-->
   <!--<param name="disable-rtp-auto-adjust" value="true"/>-->
   <!--<param name="avatar" value="/path/to/tiny.jpg"/>-->
   <!--<param name="candidate-acl" value="wan.auto"/>-->
   <!--<param name="local-network-acl" value="localnet.auto"/>-->

If I uncomment the following line in client.xml (Jingle profile)
   <!--<param name="local-network-acl" value="localnet.auto"/>-->
then exception does not happen.

Is this a known issue or do I need to post it in JIRA?

Tell me if more logs are needed...


On Sun, Feb 21, 2010 at 8:00 PM, Moiz Chinoy <moizchinoy at gmail.com> wrote:
> Guys,
>
> To make things simple gtalk client is entirely on different network.
>
> Call comes from outside through external Sip profile.
>
> If gtalk answers the call after 3-4 rings both parties can hear each other.
> If gtalk answers the call after 2 rings both parties no one can hear each other.
> If gtalk answers the call immediately FS crashes.
>
> Attached is the screen shot of the error...
>
> Here is the FS log...
> --------------------------------
> http://pastebin.freeswitch.org/12197
>
> External Sip Profile has following lines:
> ---------------------------------------------------------
> ? ?<param name="rtp-ip" value="$${local_ip_v4}"/>
> ? ?<param name="sip-ip" value="$${local_ip_v4}"/>
> ? ?<param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
> ? ?<param name="ext-sip-ip" value="$${external_sip_ip}"/>
>
> Jingle Client.xml has following lines:
> -----------------------------------------------------
> ? ?<param name="rtp-ip" value="$${bind_server_ip}"/>
> ? ?<param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
> ? ?<param name="disable-rtp-auto-adjust" value="true"/>
> ? ?<!--<param name="avatar" value="/path/to/tiny.jpg"/>-->
> ? ?<!--<param name="candidate-acl" value="wan.auto"/>-->
> ? ?<!--<param name="local-network-acl" value="localnet.auto"/>-->
>
> Vars.xml has following lines:
> -------------------------------------------
> <X-PRE-PROCESS cmd="set" data="external_rtp_ip=stun:stun.freeswitch.org"/>
> <X-PRE-PROCESS cmd="set" data="external_sip_ip=stun:stun.freeswitch.org"/>
>
>
> Please advise me how can I provide more of the required data.
>
> On Wed, Feb 17, 2010 at 11:36 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>> you cant combine stun and gtalk and boxes in the same lan very easily if you
>> do need to do that you will need to mess with
>>
>> <param name="candidate-acl" value="wan.auto"/>
>> <param name="local-network-acl" value="localnet.auto"/>
>>
>>
>>
>>
>>
>> On Wed, Feb 17, 2010 at 9:41 AM, Moiz Chinoy <moizchinoy at gmail.com> wrote:
>>>
>>> Guys I am unable to produce the crash but now both parties cannot hear
>>> each other!
>>>
>>> Vars.xml has following lines:
>>> ?<X-PRE-PROCESS cmd="set"
>>> data="external_rtp_ip=stun:stun.freeswitch.org"/>
>>> ?<X-PRE-PROCESS cmd="set"
>>> data="external_sip_ip=stun:stun.freeswitch.org"/>
>>>
>>> Jingle Client.xml has following lines:
>>> ? ?<param name="rtp-ip" value="$${bind_server_ip}"/>
>>> ? ?<!-- <param name="ext-rtp-ip" value="auto-nat"/> -->
>>> ? ?<param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
>>> ? ?<param name="disable-rtp-auto-adjust" value="true"/>
>>> ? ?<param name="local-network-acl" value="localnet.auto"/>
>>>
>>> On Wed, Feb 17, 2010 at 5:38 PM, Anthony Minessale
>>> <anthony.minessale at gmail.com> wrote:
>>> > Obtain a stack trace from the crash.
>>> >
>>> > On Feb 17, 2010 5:26 AM, "Moiz Chinoy" <moizchinoy at gmail.com> wrote:
>>> >
>>> > Hi,
>>> >
>>> > FS rev: 16673
>>> > Platform: Windows
>>> >
>>> > More details:
>>> >
>>> > FS is behind NAT and machine is running a VPN connection.
>>> >
>>> > FS and GTalk client on the same machine:
>>> >
>>> > --------------------------------------------------------------------------------------------------
>>> > jingle profile client.xml has following line:
>>> > <param name="rtp-ip" value="$${bind_server_ip}"/>
>>> >
>>> > External SIP call is successfully bridged to GTalk client.
>>> >
>>> >
>>> > FS and GTalk client on the different machine:
>>> >
>>> > --------------------------------------------------------------------------------------------------
>>> > jingle profile client.xml has following lines:
>>> > <param name="rtp-ip" value="$${bind_server_ip}"/>
>>> >
>>> > <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
>>> >
>>> > As soon as external SIP call land and I try to bridge the call to
>>> > GTalk client, FS crashes.
>>> >
>>> >
>>> > NAT Details:
>>> > ---------------------------
>>> > I think my NAT does not support UpNP or PMP. The reason I say it
>>> > because when FS starts following message is displayed:
>>> >
>>> > 2010-02-17 09:15:42.809674 [INFO] switch_nat.c:409 Scanning for NAT
>>> > 2010-02-17 09:15:42.809674 [ERR] switch_nat.c:197 Error checking for
>>> > PMP [init failed]
>>> > 2010-02-17 09:15:42.809674 [DEBUG] switch_nat.c:414 Checking for UPnP
>>> > 2010-02-17 09:15:56.825388 [DEBUG] switch_nat.c:114 No
>>> > InternetGatewayDevice, using first entry as default
>>> > (http://192.168.16.17:50144/).
>>> > 2010-02-17 09:15:56.887889 [INFO] switch_nat.c:429 No PMP or UPnP NAT
>>> > devices detected!
>>> >
>>> >
>>> >
>>> > On Tue, Feb 16, 2010 at 8:41 PM, Brian West <brian at freeswitch.org>
>>> > wrote:
>>> >> can you please update...
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>> >
>>>
>>>
>>>
>>> --
>>> Regards,
>>> Moiz Chinoy.
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
>
> --
> Regards,
> Moiz Chinoy.
>



-- 
Regards,
Moiz Chinoy.


From helmut.kuper at ewetel.de  Tue Feb 23 01:41:59 2010
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Tue, 23 Feb 2010 10:41:59 +0100
Subject: [Freeswitch-users] Question about sofia_contact
In-Reply-To: <191c3a031002220842j5bec442an5f1ea89cb0e8a6ff@mail.gmail.com>
References: <4B7EA954.30402@ewetel.de>
	<4B7EADBC.1040001@ewetel.de>	<4B8249BF.3090708@ewetel.de>
	<191c3a031002220842j5bec442an5f1ea89cb0e8a6ff@mail.gmail.com>
Message-ID: <4B83A2E7.1060905@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hello Anthony,


you are right, I'm quite unpatient, sorry 4 that. Your solution works
fine. I thought the sip domain could be any string and must not be a
valid domain format.

Thanks to you, board and community for this fantastic project!

regards from rainy germany
Helmut



On 22.02.2010 17:42, Anthony Minessale wrote:
> it's mad at you for asking twice before waiting for a reply, so it's not
> working on purpose.
> 
> Actually it's mad at you because your domain does not contain a . so it
> is assuming you are specifying a profile name as the domain.  if your
> domain was mydomain.com <http://mydomain.com> instead it would work.
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.4.7 (MingW32)

iD8DBQFLg6Ln4tZeNddg3dwRAvj6AJ9ruybNpbL8mdUlx1jVtLPYVbCSDACfQJLo
zfieJnHZdp2Xv3OS6HTZE/k=
=ESgY
-----END PGP SIGNATURE-----


From nagalenoj at gmail.com  Tue Feb 23 02:02:46 2010
From: nagalenoj at gmail.com (Nagalenoj H.)
Date: Tue, 23 Feb 2010 15:32:46 +0530
Subject: [Freeswitch-users] Error loading
	/usr/local/freeswitch/mod/ozmod_sangoma_boost.so
Message-ID: <fe67a3341002230202y1d45a8f3t3a2cda07de2240a2@mail.gmail.com>

Dear friends,
    I've installed freeswitch trunk - 16729 and tried to configure with
wanpipe for sangoma A102 pri card.

Followed the steps given in
http://wiki.sangoma.com/wanpipe-freeswitch-install

When loading the freeswitch, I've got the following error.

2010-02-23 14:49:58.545726 [INFO] ozmod_wanpipe.c:335 configuring device
s2c15 as OpenZAP device 1:30 fd:57 DTMF: software
2010-02-23 14:49:58.546004 [ERR] ozmod_wanpipe.c:436 No valid wanpipe span
and channel was specified
2010-02-23 14:49:58.546275 [INFO] zap_io.c:2500 Configured 30 channel(s)
2010-02-23 14:49:58.560724 [ERR] zap_io.c:2560 Error loading
/usr/local/freeswitch/mod/ozmod_sangoma_boost.so
[/usr/local/freeswitch/mod/ozmod_sangoma_boost.so: cannot open shared object
file: No such file or directory]
2010-02-23 14:49:58.561048 [ERR] zap_io.c:2720 can't find 'sangoma_boost'
2010-02-23 14:49:58.561266 [ERR] mod_openzap.c:2458 Error starting OpenZAP
span 1 error:
2010-02-23 14:49:58.561712 [NOTICE] switch_loadable_module.c:144 Adding
Endpoint 'openzap'

Configuration and log files are pasted to pastebin. Kindly someone help me
to solve this issue.

openzap.conf and openzap.conf.xml
   http://pastebin.freeswitch.org/12214

freeswitch log
   http://pastebin.freeswitch.org/12216

smg_pri.conf
   http://pastebin.freeswitch.org/12217

-- 
Regards,
Nagalenoj H.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100223/35cf0dcb/attachment.html 

From steveayre at gmail.com  Tue Feb 23 02:54:52 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 23 Feb 2010 10:54:52 +0000
Subject: [Freeswitch-users] rtp timeout and call hangs up
In-Reply-To: <3F9354A277B544A98E35F8C3F5900496@MOBILEE1705>
References: <3F9354A277B544A98E35F8C3F5900496@MOBILEE1705>
Message-ID: <eafa772f1002230254v7fcc8ae8mce35a19339a36170@mail.gmail.com>

Hi Franck,

Sorry, you haven't provided enough of the trace or logs to see the
reason for the timeout.

To hazard a guess though... are you using RTP in bypass media mode? If
so RTP won't go via the switch, so it won't see any RTP for the call
even though media's working - FS will then end the call if
rtp-timeout-sec is set in sofia.conf.xml.

Regards,
-Steve



On 23 February 2010 07:06, Madovsky <infos at madovsky.org> wrote:
> Hello,
>
> A leg local extension (codec GSM) ?-----> B leg local extension (codec PCMU)
> rtp timeout after 300 sec the call hangs up,
> I can ear audio and no problem of configuration.
> did I forget to set anything ?
>
> below the sip trace :
>
>
> ACK sip:1001 at 67.xx.xx.138:5060;transport=udp SIP/2.0
> Content-Length: 0
> Via: SIP/2.0/UDP
> 67.xx.xx.138:62690;rport;branch=z9hG4bK4t7xGwrintJDYr4HPfz0UQ..
> From: "1000" <sip:1000 at 67.xx.xx.138:5080>;tag=52834814554
> To: "Extension 1001" <sip:1001 at 67.xx.xx.138>;tag=3SKarSKZB63DF
> Contact: <sip:1000 at 67.xx.xx.138:62690>
> CSeq: 1 ACK
> Max-Forwards: 70
> Call-ID: 2053375473 at 67.xx.xx.138
>
> ------------------------------------------------------------------------
> 2010-02-23 02:02:19.649316 [NOTICE] mod_sofia.c:853 Hangup
> sofia/external/1000 at 67.xx.xx.138:5080 [CS_EXECUTE] [MEDIA_TIMEOUT]
> 2010-02-23 02:02:19.649316 [NOTICE] switch_ivr_bridge.c:634 Hangup
> sofia/internal/sip:1001 at 70.81.84.218:2249 [CS_EXCHANGE_MEDIA]
> [NORMAL_CLEARING]
> send 622 bytes to udp/[67.xx.xx.138]:62690 at 07:02:19.653239:
> ------------------------------------------------------------------------
> BYE sip:1000 at 67.xx.xx.138:62690 SIP/2.0
> Via: SIP/2.0/UDP 67.xx.xx.138;rport;branch=z9hG4bK790g187j50eNF
> Max-Forwards: 70
> From: "Extension 1001" <sip:1001 at 67.xx.xx.138>;tag=3SKarSKZB63DF
> To: "1000" <sip:1000 at 67.xx.xx.138:5080>;tag=52834814554
> Call-ID: 2053375473 at 67.xx.xx.138
> CSeq: 127335933 BYE
> Contact: <sip:1001 at 67.xx.xx.138:5060;transport=udp>
> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16676M
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER,
> REFER, NOTIFY
> Supported: timer, precondition, path, replaces
> Reason: FreeSWITCH;cause=604;text="MEDIA_TIMEOUT"
> Content-Length: 0
>
> ------------------------------------------------------------------------
> 2010-02-23 02:02:19.649316 [NOTICE] switch_core_session.c:1179 Session 5
> (sofia/external/1000 at 67.xx.xx.138:5080) Ended
> 2010-02-23 02:02:19.649316 [NOTICE] switch_core_session.c:1181 Close Channel
> sofia/external/1000 at 67.xx.xx.138:5080 [CS_DESTROY]
> recv 289 bytes from udp/[67.xx.xx.138]:62690 at 07:02:19.656927:
> ------------------------------------------------------------------------
> SIP/2.0 200 OK
> Content-Length: 0
> Via: SIP/2.0/UDP 67.xx.xx.138;rport=5060;branch=z9hG4bK790g187j50eNF
> From: "Extension 1001" <sip:1001 at 67.xx.xx.138>;tag=3SKarSKZB63DF
> To: "1000" <sip:1000 at 67.xx.xx.138:5080>;tag=52834814554
> CSeq: 127335933 BYE
> Call-ID: 2053375473 at 67.xx.xx.138
>
> ------------------------------------------------------------------------
> recv 391 bytes from udp/[67.xx.xx.138]:62690 at 07:02:19.660173:
> ------------------------------------------------------------------------
> BYE sip:1001 at 67.xx.xx.138:5060;transport=udp SIP/2.0
> Content-Length: 0
> Via: SIP/2.0/UDP
> 67.xx.xx.138:62690;rport;branch=z9hG4bKGTIQmsc5dUvSdP0ZtJFKcA..
> From: "1000" <sip:1000 at 67.xx.xx.138:5080>;tag=52834814554
> To: "Extension 1001" <sip:1001 at 67.xx.xx.138>;tag=3SKarSKZB63DF
> Contact: <sip:1000 at 67.xx.xx.138:62690>
> CSeq: 2 BYE
> Max-Forwards: 70
> Call-ID: 2053375473 at 67.xx.xx.138
>
> ------------------------------------------------------------------------
> send 505 bytes to udp/[67.xx.xx.138]:62690 at 07:02:19.660607:
> ------------------------------------------------------------------------
> SIP/2.0 481 Call Does Not Exist
> Via: SIP/2.0/UDP
> 67.xx.xx.138:62690;rport=62690;branch=z9hG4bKGTIQmsc5dUvSdP0ZtJFKcA..
> From: "1000" <sip:1000 at 67.xx.xx.138:5080>;tag=52834814554
> To: "Extension 1001" <sip:1001 at 67.xx.xx.138>;tag=3SKarSKZB63DF
> Call-ID: 2053375473 at 67.xx.xx.138
> CSeq: 2 BYE
> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16676M
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER,
> REFER, NOTIFY
> Supported: timer, precondition, path, replaces
> Content-Length: 0
>
> ------------------------------------------------------------------------
> send 669 bytes to udp/[70.81.84.218]:2249 at 07:02:19.869656:
> ------------------------------------------------------------------------
> BYE sip:1001 at 70.81.84.218:2249;rinstance=f17f8a2dd028c23d SIP/2.0
> Via: SIP/2.0/UDP 67.xx.xx.138:5080;rport;branch=z9hG4bKQ1y6QNQ56aDDS
> Max-Forwards: 70
> From: "1000" <sip:1000 at 67.xx.xx.138>;tag=SK3p6Frt5pa7D
> To: <sip:1001 at 70.81.84.218:2249;rinstance=f17f8a2dd028c23d>;tag=df099549
> Call-ID: 263d4030-9aec-122d-6d87-00e0ed0b00c2
> CSeq: 127335918 BYE
> Contact: <sip:mod_sofia at 67.xx.xx.138:5080>
> User-Agent: FreeSWITCH and Rock!
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER,
> REFER, NOTIFY, PUBLISH, SUBSCRIBE
> Supported: timer, precondition, path, replaces
> Reason: Q.850;cause=16;text="NORMAL_CLEARING"
> Content-Length: 0
>
> ------------------------------------------------------------------------
> 2010-02-23 02:02:19.869666 [NOTICE] switch_core_session.c:1179 Session 6
> (sofia/internal/sip:1001 at 70.81.84.218:2249) Ended
> 2010-02-23 02:02:19.869666 [NOTICE] switch_core_session.c:1181 Close Channel
> sofia/internal/sip:1001 at 70.81.84.218:2249 [CS_DESTROY]
> recv 411 bytes from udp/[70.81.84.218]:2249 at 07:02:19.986415:
> ------------------------------------------------------------------------
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 67.xx.xx.138:5080;rport=5080;branch=z9hG4bKQ1y6QNQ56aDDS
> Contact: <sip:1001 at 70.81.84.218:2249;rinstance=f17f8a2dd028c23d>
> To: <sip:1001 at 70.81.84.218:2249;rinstance=f17f8a2dd028c23d>;tag=df099549
> From: "1000"<sip:1000 at 67.xx.xx.138>;tag=SK3p6Frt5pa7D
> Call-ID: 263d4030-9aec-122d-6d87-00e0ed0b00c2
> CSeq: 127335918 BYE
> User-Agent: 3CXPhone 4.0.10858.0
> Content-Length: 0
>
> ------------------------------------------------------------------------
>
>
>
>
>
> Thanks
>
> Franck
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From technical at ttnc.co.uk  Tue Feb 23 04:21:02 2010
From: technical at ttnc.co.uk (TTNC - Technical)
Date: Tue, 23 Feb 2010 12:21:02 +0000
Subject: [Freeswitch-users] leg_timeout with ignore_early_media false
Message-ID: <F86E6E00-B774-487F-9F62-2B1BB00382FC@ttnc.co.uk>

Hi Guys

I'm trying to create a hunt group where ignore_early_media = false is set, so that the international ring tone is passed through to the caller. Setting ignore_early_media = false on the channel does what I want, but with this set leg_timeout is not honoured.

I've switched to use bridge_answer_timeout which is honoured if ignore_early_media = false and the call progresses through the different legs, but bridge_answer_timeout times out the call after the set period, even if the call has been successfully answered and bridged. I've tried all different combinations of group_confirm_cancel_timeout [1|2|3], none of them seem to affect bridge_answer_timeout.

Does anyone have a solution for timing out legs of a hunt group with ignore_early_media = false set?

This is my dial string:

'{caller-id-in-from=true,origination_caller_id_name=012345123123,origination_caller_id_number= 012345123123}[bridge_answer_timeout=20]sofia/internal/4412345123123 at sipipgw.siphost.net |sofia/internal/4412345123123 at sipipgw.siphost.net'

And in my lua application, I'm setting the following:

session:setVariable("group_confirm_cancel_timeout", "1"); -- substitute 1 for 1, 2 or 3, none work.
session:setVariable("ignore_early_media", "false");

I'm using the latest trunk revision - it's still happening.

Any suggestions welcome.

Thanks

Russ



From dftoro at yahoo.com  Tue Feb 23 05:33:12 2010
From: dftoro at yahoo.com (Diego Toro)
Date: Tue, 23 Feb 2010 05:33:12 -0800 (PST)
Subject: [Freeswitch-users] Hook Flash
In-Reply-To: <915F9D0D-FB13-4D02-995B-FB7F5EB488D2@tlainvestments.com>
Message-ID: <742756.71167.qm@web33501.mail.mud.yahoo.com>

hi, read http://jira.freeswitch.org/browse/OPENZAP-30

  <action application="send_dtmf" data="F"/>

Diego Toro
http://lacarretade.blogspot.com/


--- On Mon, 2/22/10, Troy Anderson <troy at tlainvestments.com> wrote:

> From: Troy Anderson <troy at tlainvestments.com>
> Subject: [Freeswitch-users] Hook Flash
> To: freeswitch-users at lists.freeswitch.org
> Date: Monday, February 22, 2010, 11:30 PM
> Is there a way to cause a hook-flash
> on a zap channel via the dial plan?? e.g. VoIP phone
> <=> FS <=> POTS line, and I want to flash via
> some star code on the phone.
> 
> I'm happy to document on the openzap page on the wiki if
> so.
> 
> Thanks,
> Troy
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 


      


From jeff at jefflenk.com  Tue Feb 23 06:23:54 2010
From: jeff at jefflenk.com (Jeff Lenk)
Date: Tue, 23 Feb 2010 08:23:54 -0600
Subject: [Freeswitch-users] FScomm
In-Reply-To: <157FDB96FB88476FBAEC69C1309A0A5F@MOBILEE1705>
References: <157FDB96FB88476FBAEC69C1309A0A5F@MOBILEE1705>
Message-ID: <COL117-W5563292C886257AFC56548B9420@phx.gbl>


What platform are you trying to build?
 


From: infos at madovsky.org
To: freeswitch-users at lists.freeswitch.org
Date: Tue, 23 Feb 2010 01:41:04 -0500
Subject: [Freeswitch-users] FScomm




Hi,
 
is http://wiki.freeswitch.org/wiki/FSComm
available yet ? because QT framework link has broken
and some instruction don't work with last svn trunk (for example to compile FScomm in FS svn root)
 
Thanks

Franck 		 	   		  
_________________________________________________________________
Your E-mail and More On-the-Go. Get Windows Live Hotmail Free.
http://clk.atdmt.com/GBL/go/201469229/direct/01/
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100223/8cc247d7/attachment.html 

From christian.loeschenkohl at xpirio.com  Tue Feb 23 06:56:28 2010
From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=)
Date: Tue, 23 Feb 2010 15:56:28 +0100
Subject: [Freeswitch-users] sending a sip notify with sendevent
In-Reply-To: <12033.93456.qm@web33507.mail.mud.yahoo.com>
References: <12033.93456.qm@web33507.mail.mud.yahoo.com>
Message-ID: <4B83EC9C.8030501@xpirio.com>

i think it was my fault
i'm not 100% sure why - but i works now as expected

thank you very much for your suggestions

br


On 2010-02-23 04:18, Diego Toro wrote:

> hi, I'm using
>
> sendevent NOTIFY
> profile: internal
> event-string: check-sync;reboot=false
> user: 1001
> content-type: application/simple-message-sumary
> profile: internal
> event-string: check-sync;reboot=false
> host: 192.168.7.3
>
>
> where:
> user 1001 is a registered user
> host: IP of FreeSwitch
>
> I see a sip notify message sent to the client.
>
> Diego Toro
> http://lacarretade.blogspot.com/
>
>
> --- On Mon, 2/22/10, Christian L?schenkohl<christian.loeschenkohl at xpirio.com>  wrote:
>
>> From: Christian L?schenkohl<christian.loeschenkohl at xpirio.com>
>> Subject: Re: [Freeswitch-users] sending a sip notify with sendevent
>> To: freeswitch-users at lists.freeswitch.org
>> Date: Monday, February 22, 2010, 5:28 PM
>> hi anthony
>>
>> i did it
>>
>> my profile is actually named nat, "sofia status profile
>> nat" shows me presence_nat as the db name
>> so i had a look in the file presence_nat.db
>>
>> i execute
>> select
>> sip_user,sip_host,contact,profile_name,'application/simple-message-summary','check-sync;reboot=false',''
>> from sip_registrations where sip_user='10' and
>> sip_host='vts.vie.xpirio.net';
>>
>> and it returns
>> 10|vts.vie.xpirio.net|"10"
>> <sip:10 at 81.94.53.5:1032>|nat|application/simple-message-summary|check-sync;reboot=false|
>>
>> looks good to me so far, but as i said no sip notify
>> message is send to the client
>>
>> br
>>
>> Anthony Minessale wrote:
>>
>>> compare that sql stmt to your db manually with the
>> sqlite3 app
>>>
>>> sqlite3
>> /usr/local/freeswitch/db/sofia_reg_internal.db
>>>
>>>
>>> 2010/2/22 Christian L?schenkohl<christian.loeschenkohl at xpirio.com
>>
>>> <mailto:christian.loeschenkohl at xpirio.com>>
>>>
>>>       thank you for this advise
>>>
>>>       i read the section "case
>> SWITCH_EVENT_NOTIFY" carefully, debugged my
>>>       script (i messed something
>>>       up, with telnet the command
>> works - returns Reply-Text: +OK)
>>>
>>>       sql is executed and returns 1
>> row
>>>       select
>>>
>>     sip_user,sip_host,contact,profile_name,'application/simple-message-summary','check-sync;reboot=true',''
>>>       from sip_registrations where
>> sip_user='10' and
>>>       sip_host='vts.vie.xpirio.net
>> <http://vts.vie.xpirio.net>'
>>>
>>>       however no notify message is
>> send to the device
>>>
>>>       i can't use call-id because i
>> simply don't know it
>>>
>>>       br
>>>
>>>
>>>       Diego Toro wrote:
>>>
>>>        >  Read "case
>> SWITCH_EVENT_NOTIFY:" in mod_sofia.c. I suggest you
>>>       assign call-id in the header
>> of the event.
>>>        >
>>>        >
>>>        >  Diego Toro
>>>        >  http://lacarretade.blogspot.com/
>>>        >
>>>        >
>>>        >  --- On Mon, 2/22/10,
>> Christian L?schenkohl
>>>       <christian.loeschenkohl at xpirio.com
>>>       <mailto:christian.loeschenkohl at xpirio.com>>
>> wrote:
>>>        >
>>>        >>  From: Christian
>> L?schenkohl<christian.loeschenkohl at xpirio.com
>>>       <mailto:christian.loeschenkohl at xpirio.com>>
>>>        >>  Subject:
>> [Freeswitch-users] sending a sip notify with sendevent
>>>        >>  To: freeswitch-users at lists.freeswitch.org
>>>       <mailto:freeswitch-users at lists.freeswitch.org>
>>>        >>  Date: Monday, February
>> 22, 2010, 12:16 PM
>>>        >>  hi
>>>        >>
>>>        >>  i try to send a sip
>> notify message to a registered sip
>>>        >>  device
>>>        >>  "sofia profile nat
>> flush_inbound_reg 10 at vts.vie.xpirio.net
>>>       <mailto:10 at vts.vie.xpirio.net>
>>>        >>  reboot" works, but i
>> need
>>>        >>  to send
>> "check-sync;reboot=false" - so the device does a
>>>        >>  resync and don't do a
>> reboot
>>>        >>
>>>        >>  my message looks like
>> this
>>>        >>
>>>        >>  sendevent NOTIFY
>>>        >>  profile: nat
>>>        >>  event-string:
>> check-sync;reboot=false
>>>        >>  user: 10
>>>        >>  host: vts.vie.xpirio.net
>> <http://vts.vie.xpirio.net>
>>>        >>  content-type:
>> application/simple-message-summary
>>>        >>
>>>        >>  if i listen on the
>> loopback interface i do see
>>>        >>
>>>        >>  ##
>>>        >>  T 2010/02/22
>> 18:11:59.083204 127.0.0.1:51840
>>>       <http://127.0.0.1:51840>  ->
>>>        >>  127.0.0.1:8021<http://127.0.0.1:8021>  [AP]
>>>        >>  sendevent NOTIFY
>>>        >>  profile: nat
>>>        >>  event-string:
>> check-sync;reboot=false
>>>        >>  user: 10
>>>        >>  host: vts.vie.xpirio.net
>> <http://vts.vie.xpirio.net>
>>>        >>  content-type:
>> application/simple-message-summary
>>>        >>
>>>        >>  ##
>>>        >>  T 2010/02/22
>> 18:11:59.084032 127.0.0.1:8021
>>>       <http://127.0.0.1:8021>  ->
>>>        >>  127.0.0.1:51840<http://127.0.0.1:51840>  [AP]
>>>        >>  Content-Type:
>> command/reply
>>>        >>  Reply-Text: -ERR invalid
>>>        >>
>>>        >>  --------
>>>        >>  i don't get what it is
>> wrong. i also rechecked the
>>>        >>  registered user in the
>> sqlite database and this
>>>        >>  looks good to me.
>>>        >>
>>>        >>  no message is send to the
>> user.
>>>        >>
>>>        >>  we do use multiple
>> domains, so user could also be
>>>       10 at somedomain.com
>> <mailto:10 at somedomain.com>
>>>        >>  - or am i wrong on this?
>>>        >>  could somebody please
>> bring some light in this.
>>>        >>
>>>        >>  we do use trunk rev.
>> 16631
>>>        >>
>>>        >>  br
>>>        >>
>>>        >>
>>>        >>
>>>        >>  --
>>>        >>  Ing. Christian
>> L?schenkohl
>>>        >>  Technische Leitung,
>> Forschung&  Entwicklung VoIP
>>>        >>
>>>        >>  xpirio
>>>        >>  Telekommunikation&
>> Service GmbH
>>>        >>  Lakeside B04
>>>        >>  9020 Klagenfurt
>>>        >>  Austria
>>>        >>
>>>        >>  T  +43 (0) 5 77 11 -
>> 1000
>>>        >>  F  +43 (0) 5 77 11 -
>> 1002
>>>        >>  E  christian.loeschenkohl at xpirio.com
>>>       <mailto:christian.loeschenkohl at xpirio.com>
>>>        >>
>>>        >>
>> _______________________________________________
>>>        >>  FreeSWITCH-users mailing
>> list
>>>        >>  FreeSWITCH-users at lists.freeswitch.org
>>>       <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>        >>  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>        >>
>>>       UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>        >>  http://www.freeswitch.org
>>>        >>
>>>        >
>>>        >
>>>        >
>>>        >
>>>        >
>> _______________________________________________
>>>        >  FreeSWITCH-users mailing
>> list
>>>        >  FreeSWITCH-users at lists.freeswitch.org
>>>       <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>        >  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>        >
>>>       UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>        >  http://www.freeswitch.org
>>>
>>>
>>>       --
>>>       Ing. Christian L?schenkohl
>>>       Technische Leitung, Forschung
>> &  Entwicklung VoIP
>>>
>>>       xpirio
>>>       Telekommunikation&
>> Service GmbH
>>>       Lakeside B04
>>>       9020 Klagenfurt
>>>       Austria
>>>
>>>       T  +43 (0) 5 77 11 -
>> 1000
>>>       F  +43 (0) 5 77 11 -
>> 1002
>>>       E  christian.loeschenkohl at xpirio.com
>>>       <mailto:christian.loeschenkohl at xpirio.com>
>>>
>>>
>>     _______________________________________________
>>>       FreeSWITCH-users mailing list
>>>       FreeSWITCH-users at lists.freeswitch.org
>>>       <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>       http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>       UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>       http://www.freeswitch.org
>>>
>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com
>>
>>> <mailto:MSN%3Aanthony_minessale at hotmail.com>
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>
>>> <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
>>> IRC: irc.freenode.net<http://irc.freenode.net>  #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org
>>
>>> <mailto:sip%3A888 at conference.freeswitch.org>
>>> iax:guest at conference.freeswitch.org/888
>>
>>> <http://iax:guest at conference.freeswitch.org/888>
>>> googletalk:conf+888 at conference.freeswitch.org
>>
>>> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> pstn:+19193869900
>>>
>>>
>>>
>> ------------------------------------------------------------------------
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> --
>> Ing. Christian L?schenkohl
>> Technische Leitung, Forschung&  Entwicklung VoIP
>>
>> xpirio
>> Telekommunikation&  Service GmbH
>> Lakeside B04
>> 9020 Klagenfurt
>> Austria
>>
>> T  +43 (0) 5 77 11 - 1000
>> F  +43 (0) 5 77 11 - 1002
>> E  christian.loeschenkohl at xpirio.com
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


-- 
Ing. Christian L?schenkohl
Technische Leitung, Forschung & Entwicklung VoIP

xpirio
Telekommunikation & Service GmbH
Lakeside B04
9020 Klagenfurt
Austria

T  +43 (0) 5 77 11 - 1000
F  +43 (0) 5 77 11 - 1002
E  christian.loeschenkohl at xpirio.com


From christian.loeschenkohl at xpirio.com  Tue Feb 23 07:12:00 2010
From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=)
Date: Tue, 23 Feb 2010 16:12:00 +0100
Subject: [Freeswitch-users] big thanks to all freeswitch developers and
	contributing users
Message-ID: <4B83F040.7040005@xpirio.com>

i want to say a big THANKY YOU to all contributing freeswitch community members.

over one year has passed since i did fall in love with this project.
it is getting better every day, one get's help and advices if needed.
the admins do care about nearly every problem - no matter if it's big or small.
i also did manage an opensource project and i wish i had done it with that much
heart and intense power that i see here.

i also hope that i can contribute back enough (questions, bug reports, wiki enhancements).

wishing you all the best
br

-- 
Ing. Christian L?schenkohl
Technische Leitung, Forschung & Entwicklung VoIP

xpirio
Telekommunikation & Service GmbH
Lakeside B04
9020 Klagenfurt
Austria

T  +43 (0) 5 77 11 - 1000
F  +43 (0) 5 77 11 - 1002
E  christian.loeschenkohl at xpirio.com


From technical at ttnc.co.uk  Tue Feb 23 07:31:55 2010
From: technical at ttnc.co.uk (TTNC - Technical)
Date: Tue, 23 Feb 2010 15:31:55 +0000
Subject: [Freeswitch-users] leg_timeout with ignore_early_media false
In-Reply-To: <F86E6E00-B774-487F-9F62-2B1BB00382FC@ttnc.co.uk>
References: <F86E6E00-B774-487F-9F62-2B1BB00382FC@ttnc.co.uk>
Message-ID: <F25F531B-C93C-4293-8400-BB82B3B3A2FC@ttnc.co.uk>

I've opened a Jira for this issue as I believe it's a bug, there's an example LUA script attached to the bug to replicate the issue.

http://jira.freeswitch.org/browse/FSCORE-556

Russ

On 23 Feb 2010, at 12:21, TTNC - Technical wrote:

> Hi Guys
> 
> I'm trying to create a hunt group where ignore_early_media = false is set, so that the international ring tone is passed through to the caller. Setting ignore_early_media = false on the channel does what I want, but with this set leg_timeout is not honoured.
> 
> I've switched to use bridge_answer_timeout which is honoured if ignore_early_media = false and the call progresses through the different legs, but bridge_answer_timeout times out the call after the set period, even if the call has been successfully answered and bridged. I've tried all different combinations of group_confirm_cancel_timeout [1|2|3], none of them seem to affect bridge_answer_timeout.
> 
> Does anyone have a solution for timing out legs of a hunt group with ignore_early_media = false set?
> 
> This is my dial string:
> 
> '{caller-id-in-from=true,origination_caller_id_name=012345123123,origination_caller_id_number= 012345123123}[bridge_answer_timeout=20]sofia/internal/4412345123123 at sipipgw.siphost.net |sofia/internal/4412345123123 at sipipgw.siphost.net'
> 
> And in my lua application, I'm setting the following:
> 
> session:setVariable("group_confirm_cancel_timeout", "1"); -- substitute 1 for 1, 2 or 3, none work.
> session:setVariable("ignore_early_media", "false");
> 
> I'm using the latest trunk revision - it's still happening.
> 
> Any suggestions welcome.
> 
> Thanks
> 
> Russ
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From infos at madovsky.org  Tue Feb 23 08:51:51 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 23 Feb 2010 11:51:51 -0500
Subject: [Freeswitch-users] FScomm
References: <157FDB96FB88476FBAEC69C1309A0A5F@MOBILEE1705>
	<COL117-W5563292C886257AFC56548B9420@phx.gbl>
Message-ID: <D64783281177440B9094B8181E55EFA7@MOBILEE1705>


  ----- Original Message ----- 
  From: Jeff Lenk 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Tuesday, February 23, 2010 9:23 AM
  Subject: Re: [Freeswitch-users] FScomm


  What platform are you trying to build?
   

------------------------------------------------------------------------------
  From: infos at madovsky.org
  To: freeswitch-users at lists.freeswitch.org
  Date: Tue, 23 Feb 2010 01:41:04 -0500
  Subject: [Freeswitch-users] FScomm


  Hi,

  is http://wiki.freeswitch.org/wiki/FSComm
  available yet ? because QT framework link has broken
  and some instruction don't work with last svn trunk (for example to compile FScomm in FS svn root)

  Thanks

  Franck


------------------------------------------------------------------------------
  Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. Sign up now. 


------------------------------------------------------------------------------


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  http://www.freeswitch.org


  FSComm on Linux fedora 10 64 bits
  It says FSComm can be built inside FS svn folder typing gmake make
  but there is no Makefile inside

  Thanks

  Franck
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100223/22489c2b/attachment.html 

From anthony.minessale at gmail.com  Tue Feb 23 09:44:03 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 23 Feb 2010 11:44:03 -0600
Subject: [Freeswitch-users] leg_timeout with ignore_early_media false
In-Reply-To: <F25F531B-C93C-4293-8400-BB82B3B3A2FC@ttnc.co.uk>
References: <F86E6E00-B774-487F-9F62-2B1BB00382FC@ttnc.co.uk>
	<F25F531B-C93C-4293-8400-BB82B3B3A2FC@ttnc.co.uk>
Message-ID: <191c3a031002230944h29741d21p58db0f9bfff57787@mail.gmail.com>

Bug is fixed and there is a note on your other bug awaiting your response.


On Tue, Feb 23, 2010 at 9:31 AM, TTNC - Technical <technical at ttnc.co.uk>wrote:

> I've opened a Jira for this issue as I believe it's a bug, there's an
> example LUA script attached to the bug to replicate the issue.
>
> http://jira.freeswitch.org/browse/FSCORE-556
>
> Russ
>
> On 23 Feb 2010, at 12:21, TTNC - Technical wrote:
>
> > Hi Guys
> >
> > I'm trying to create a hunt group where ignore_early_media = false is
> set, so that the international ring tone is passed through to the caller.
> Setting ignore_early_media = false on the channel does what I want, but with
> this set leg_timeout is not honoured.
> >
> > I've switched to use bridge_answer_timeout which is honoured if
> ignore_early_media = false and the call progresses through the different
> legs, but bridge_answer_timeout times out the call after the set period,
> even if the call has been successfully answered and bridged. I've tried all
> different combinations of group_confirm_cancel_timeout [1|2|3], none of them
> seem to affect bridge_answer_timeout.
> >
> > Does anyone have a solution for timing out legs of a hunt group with
> ignore_early_media = false set?
> >
> > This is my dial string:
> >
> >
> '{caller-id-in-from=true,origination_caller_id_name=012345123123,origination_caller_id_number=
> 012345123123}[bridge_answer_timeout=20]sofia/internal/
> 4412345123123 at sipipgw.siphost.net |sofia/internal/
> 4412345123123 at sipipgw.siphost.net'
> >
> > And in my lua application, I'm setting the following:
> >
> > session:setVariable("group_confirm_cancel_timeout", "1"); -- substitute 1
> for 1, 2 or 3, none work.
> > session:setVariable("ignore_early_media", "false");
> >
> > I'm using the latest trunk revision - it's still happening.
> >
> > Any suggestions welcome.
> >
> > Thanks
> >
> > Russ
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100223/7fd8c5b5/attachment.html 

From robert.hadley at teotech.com  Tue Feb 23 09:48:54 2010
From: robert.hadley at teotech.com (Robert Hadley)
Date: Tue, 23 Feb 2010 09:48:54 -0800
Subject: [Freeswitch-users] Possible typo on Openzap.conf wiki page
	specifying fxs/fxo-channels
Message-ID: <2C33D9D6C1444AAC80950CDF8D38A428@greyhawk.tonecommander.com>


On the http://wiki.freeswitch.org/wiki/Openzap.conf_Examples wiki page, is
there a typo in the wanpipe /usr/local/freeswitch/conf/openzap.conf example
concerning specifying the fxo-channel vs. fxs-channel?


In the [span wanpipe FXS] section the channels are shown on wiki page as
fxo-channels => 1:1 and 1:2


In the [span wanpipe FXO] section the channels are shown on wiki page as
fxs-channels => 1:3 and 1:4


Are specifying the fxs-channels and fxo-channels shown in the wrong
sections?  I had to specify fxs-channels in the FXS span and fxo-channels in
the FXO span to get it working on my hardware.


Except from wiki page:


Sangoma A200/A400


*	A200, A200D, A400, A400D series and variants 

The configuration depends on whether wanpipe is configured to use Zaptel TDM
Voice, or the Sangoma standalone TDM Voice API. This is determined in the
installation and configuration of the Sangoma wanpipe software. 

If wanpipe is using Zaptel, you need to configure openzap.conf with [span
zt] entries. For example, if ports 1 and 2 are FXS (e.g. configured to
accept analog phone connections), and ports 3 and 4 are FXO (e.g. configured
to accept PSTN analog lines) you will need: 

/usr/local/freeswitch/conf/openzap.conf: 

[span zt FXO]
name => OpenZAP
number => 3001
fxo-channel => 1
number => 3002
fxo-channel => 2
 
[span zt FXS]
name => OpenZAP
number => 4165551111
fxs-channel => 3
number => 4165552222
fxs-channel => 4

If wanpipe is standalone, you need to configure openzap.conf with [span
wanpipe] entries. For the same example as above this would be: 

/usr/local/freeswitch/conf/openzap.conf: 

[span wanpipe FXS]
# This is the value of the callerid_name variable that is raised in the
dialplan
name => Analog Phone 1
# This is the value of the callerid_number variable that is raised in the
dialplan
number => 3001
fxo-channel => 1:1
name => Analog Phone 2
number => 3002
fxo-channel => 1:2
 
[span wanpipe FXO]
# the chan_name variable will raised as "OpenZAP/2:1/4165551111" in dialplan
when an incoming call arrives on this port
name => PSTN line 1
number => 4165551111
fxs-channel => 1:3
name => PSTN line 2
number => 4165552222
fxs-channel => 1:4

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100223/a54600fc/attachment-0001.html 

From brian at freeswitch.org  Tue Feb 23 09:53:46 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 23 Feb 2010 11:53:46 -0600
Subject: [Freeswitch-users] Possible typo on Openzap.conf wiki page
	specifying fxs/fxo-channels
In-Reply-To: <2C33D9D6C1444AAC80950CDF8D38A428@greyhawk.tonecommander.com>
References: <2C33D9D6C1444AAC80950CDF8D38A428@greyhawk.tonecommander.com>
Message-ID: <71D261B7-ECBA-4582-8BBA-CC34258970D7@freeswitch.org>

You can edit the examples on the wiki and it should be good.

/b

On Feb 23, 2010, at 11:48 AM, Robert Hadley wrote:

> On the http://wiki.freeswitch.org/wiki/Openzap.conf_Examples wiki page, is there a typo in the wanpipe/usr/local/freeswitch/conf/openzap.conf example concerning specifying the fxo-channel vs. fxs-channel?
> 
> In the [span wanpipe FXS] section the channels are shown on wiki page as fxo-channels => 1:1 and 1:2
> 
> In the [span wanpipe FXO] section the channels are shown on wiki page as fxs-channels => 1:3 and 1:4
> 
> Are specifying the fxs-channels and fxo-channels shown in the wrong sections?  I had to specify fxs-channels in the FXS span and fxo-channels in the FXO span to get it working on my hardware.
> 
> Except from wiki page:
> Sangoma A200/A400
> A200, A200D, A400, A400D series and variants
> The configuration depends on whether wanpipe is configured to use Zaptel TDM Voice, or the Sangoma standalone TDM Voice API. This is determined in the installation and configuration of the Sangoma wanpipe software.
> 
> If wanpipe is using Zaptel, you need to configure openzap.conf with [span zt] entries. For example, if ports 1 and 2 are FXS (e.g. configured to accept analog phone connections), and ports 3 and 4 are FXO (e.g. configured to accept PSTN analog lines) you will need:
> 
> /usr/local/freeswitch/conf/openzap.conf:
> 
> [span zt FXO]
> name => OpenZAP
> number => 3001
> fxo-channel => 1
> number => 3002
> fxo-channel => 2
>  
> [span zt FXS]
> name => OpenZAP
> number => 4165551111
> fxs-channel => 3
> number => 4165552222
> fxs-channel => 4
> If wanpipe is standalone, you need to configure openzap.conf with [span wanpipe] entries. For the same example as above this would be:
> 
> /usr/local/freeswitch/conf/openzap.conf:
> 
> [span wanpipe FXS]
> # This is the value of the callerid_name variable that is raised in the dialplan
> name => Analog Phone 1
> # This is the value of the callerid_number variable that is raised in the dialplan
> number => 3001
> fxo-channel => 1:1
> name => Analog Phone 2
> number => 3002
> fxo-channel => 1:2
>  
> [span wanpipe FXO]
> # the chan_name variable will raised as "OpenZAP/2:1/4165551111" in dialplan when an incoming call arrives on this port
> name => PSTN line 1
> number => 4165551111
> fxs-channel => 1:3
> name => PSTN line 2
> number => 4165552222
> fxs-channel => 1:4
>  
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100223/891675db/attachment.html 

From srinivas.ksvreddy at gmail.com  Mon Feb 22 22:02:11 2010
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Tue, 23 Feb 2010 11:32:11 +0530
Subject: [Freeswitch-users] Freeswitch to another Freeswitch(or gateway)
Message-ID: <f8af5741002222202p3457e8efl2e683082dc597b47@mail.gmail.com>

Hi,
i want divert calls from my sipserver to another sipserver or third party
gateway, is there any way to achive this.

Regards
Srinivasula Reddy K
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100223/1660851f/attachment-0001.html 

From srinivas.ksvreddy at gmail.com  Tue Feb 23 06:00:07 2010
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Tue, 23 Feb 2010 19:30:07 +0530
Subject: [Freeswitch-users] Fwd: Freeswitch to another Freeswitch(or gateway)
In-Reply-To: <f8af5741002222202p3457e8efl2e683082dc597b47@mail.gmail.com>
References: <f8af5741002222202p3457e8efl2e683082dc597b47@mail.gmail.com>
Message-ID: <f8af5741002230600h6911e676iae6fc48375b31b1@mail.gmail.com>

 Hi,
i want divert calls from my sipserver to another sipserver or third party
gateway based on the host name, is there any way to achive this.

Regards
Srinivasula Reddy K



-- 
Srinivasula Reddy K
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100223/e25ca1ae/attachment-0001.html 

From phunk0000 at hotmail.com  Tue Feb 23 07:42:28 2010
From: phunk0000 at hotmail.com (Meg Stroodle)
Date: Tue, 23 Feb 2010 10:42:28 -0500
Subject: [Freeswitch-users] mod_nibblebill
Message-ID: <BLU110-W28EA8BD06A601D5D85268BA7420@phx.gbl>




Hello List!  I am trying to install mod_nibblebill on
my FS installation.  I get the following log entry form FS &
nibblebill, but the database table I setup remains unchanged.  Any help in
this matter would be greatly appreciated.  Following is an excerpt from
the FS log:

 

2010-02-23 10:32:52.869149 [DEBUG] switch_rtp.c:2004 Correct
ip/port confirmed.

2010-02-23 10:32:53.077162 [DEBUG] sofia.c:3727 Channel sofia/internal/3007 at 192.168.15.177
entering state [ready][200]

2010-02-23 10:32:53.149166 [DEBUG] switch_core_session.c:706
Send signal sofia/internal/sip:3008 at 192.168.15.176:21828 [BREAK]

2010-02-23 10:32:53.149166 [DEBUG] switch_core_session.c:706
Send signal sofia/internal/3007 at 192.168.15.177
[BREAK]

2010-02-23 10:32:53.149166 [DEBUG] switch_ivr_bridge.c:131
sofia/internal/sip:3008 at 192.168.15.176:21828 receive message [DISPLAY]

2010-02-23 10:32:53.169168 [DEBUG] switch_ivr_bridge.c:131 sofia/internal/3007 at 192.168.15.177
receive message [DISPLAY]

2010-02-23 10:33:21.388949 [DEBUG] mod_nibblebill.c:529
Received request via SESSION_HEARTBEAT!

2010-02-23 10:33:21.388949 [DEBUG] mod_nibblebill.c:397
Attempting to bill at $1.0 per minute to account 3007

2010-02-23 10:33:21.388949 [INFO] mod_nibblebill.c:447
Beginning new billing on 3ec17ffc-c94e-46ff-bb0d-7ad0ef27bdab

2010-02-23 10:33:21.388949 [DEBUG] mod_nibblebill.c:455 28
seconds passed since last bill time of 2010-02-23 10:32:52

2010-02-23 10:33:21.388949 [DEBUG] mod_nibblebill.c:461
Billing $0.475997 to 3007 (Call: 3ec17ffc-c94e-46ff-bb0d-7ad0ef27bdab /
0.000000 so far)

2010-02-23 10:33:51.508822 [DEBUG] mod_nibblebill.c:529
Received request via SESSION_HEARTBEAT!

2010-02-23 10:33:51.508822 [DEBUG] mod_nibblebill.c:397 Attempting
to bill at $1.0 per minute to account 3007

2010-02-23 10:33:51.508822 [DEBUG] mod_nibblebill.c:449 Last
successful billing time was 

2010-02-23 10:33:51.508822 [DEBUG] mod_nibblebill.c:455 30
seconds passed since last bill time of 2010-02-23 10:33:21

2010-02-23 10:33:51.508822 [DEBUG] mod_nibblebill.c:461
Billing $0.501998 to 3007 (Call: 3ec17ffc-c94e-46ff-bb0d-7ad0ef27bdab /
0.475997 so far)

2010-02-23 10:34:21.649708 [DEBUG] mod_nibblebill.c:529
Received request via SESSION_HEARTBEAT!

2010-02-23 10:34:21.649708 [DEBUG] mod_nibblebill.c:397
Attempting to bill at $1.0 per minute to account 3007

2010-02-23 10:34:21.649708 [DEBUG] mod_nibblebill.c:449 Last
successful billing time was 

2010-02-23 10:34:21.649708 [DEBUG] mod_nibblebill.c:455 30 seconds
passed since last bill time of 2010-02-23 10:33:51

2010-02-23 10:34:21.649708 [DEBUG] mod_nibblebill.c:461
Billing $0.502348 to 3007 (Call: 3ec17ffc-c94e-46ff-bb0d-7ad0ef27bdab /
0.977995 so far)

2010-02-23 10:34:26.254308 [NOTICE] sofia.c:329 Hangup sofia/internal/3007 at 192.168.15.177
[CS_EXECUTE] [NORMAL_CLEARING]

2010-02-23 10:34:26.254308 [DEBUG] switch_channel.c:1912
Send signal sofia/internal/3007 at 192.168.15.177
[KILL]

2010-02-23 10:34:26.254308 [DEBUG] switch_core_session.c:999
Send signal sofia/internal/3007 at 192.168.15.177
[BREAK]

2010-02-23 10:34:26.254308 [DEBUG]
switch_core_state_machine.c:459 sofia/internal/3007 at 192.168.15.177
thread mismatch skipping state handler.

2010-02-23 10:34:26.269008 [DEBUG] switch_ivr_bridge.c:470 sofia/internal/3007 at 192.168.15.177
ending bridge by request from read function

2010-02-23 10:34:26.269008 [DEBUG] switch_ivr_bridge.c:563
BRIDGE THREAD DONE [sofia/internal/3007 at 192.168.15.177]

2010-02-23 10:34:26.269008 [DEBUG] switch_ivr_bridge.c:565
Send signal sofia/internal/sip:3008 at 192.168.15.176:21828 [BREAK]

2010-02-23 10:34:26.269008 [DEBUG] switch_ivr_bridge.c:520
sofia/internal/sip:3008 at 192.168.15.176:21828 receive message [UNBRIDGE]

2010-02-23 10:34:26.269008 [DEBUG] switch_core_session.c:645
Send signal sofia/internal/sip:3008 at 192.168.15.176:21828 [BREAK]

2010-02-23 10:34:26.269008 [DEBUG] switch_ivr_bridge.c:563
BRIDGE THREAD DONE [sofia/internal/sip:3008 at 192.168.15.176:21828]

2010-02-23 10:34:26.269008 [DEBUG] switch_ivr_bridge.c:565
Send signal sofia/internal/3007 at 192.168.15.177
[BREAK]

2010-02-23 10:34:26.269008 [NOTICE] switch_ivr_bridge.c:617
Hangup sofia/internal/sip:3008 at 192.168.15.176:21828 [CS_EXCHANGE_MEDIA]
[NORMAL_CLEARING]

2010-02-23 10:34:26.269008 [DEBUG] switch_channel.c:1912
Send signal sofia/internal/sip:3008 at 192.168.15.176:21828 [KILL]

2010-02-23 10:34:26.269008 [DEBUG] switch_core_session.c:999
Send signal sofia/internal/sip:3008 at 192.168.15.176:21828 [BREAK]

2010-02-23 10:34:26.269008 [DEBUG]
switch_core_state_machine.c:488 (sofia/internal/sip:3008 at 192.168.15.176:21828)
State HANGUP

2010-02-23 10:34:26.269008 [DEBUG] mod_sofia.c:358 Channel
sofia/internal/sip:3008 at 192.168.15.176:21828 hanging up, cause: NORMAL_CLEARING

2010-02-23 10:34:26.269008 [DEBUG] mod_sofia.c:400 Sending
BYE to sofia/internal/sip:3008 at 192.168.15.176:21828

2010-02-23 10:34:26.269008 [DEBUG] mod_nibblebill.c:397
Attempting to bill at $2.30 per minute to account 3008

2010-02-23 10:34:26.269008 [INFO] mod_nibblebill.c:447
Beginning new billing on c1c42712-89e9-44c5-914f-11ed8f6aebb1

2010-02-23 10:34:26.269008 [DEBUG] mod_nibblebill.c:455 93
seconds passed since last bill time of 2010-02-23 10:32:52

2010-02-23 10:34:26.269008 [DEBUG] mod_nibblebill.c:461
Billing $3.582436 to 3008 (Call: c1c42712-89e9-44c5-914f-11ed8f6aebb1 / 0.000000
so far)

2010-02-23 10:34:26.269008 [DEBUG]
switch_core_state_machine.c:46 sofia/internal/sip:3008 at 192.168.15.176:21828
Standard HANGUP, cause: NORMAL_CLEARING

2010-02-23 10:34:26.269008 [DEBUG]
switch_core_state_machine.c:488 (sofia/internal/sip:3008 at 192.168.15.176:21828)
State HANGUP going to sleep

2010-02-23 10:34:26.269008 [INFO] switch_core_session.c:1108
sofia/internal/sip:3008 at 192.168.15.176:21828 setting session heartbeat to 30
second(s).

2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:351
(sofia/internal/sip:3008 at 192.168.15.176:21828) State EXCHANGE_MEDIA going to
sleep

2010-02-23 10:34:26.269008 [DEBUG]
switch_core_state_machine.c:314 (sofia/internal/sip:3008 at 192.168.15.176:21828)
Running State Change CS_HANGUP

2010-02-23 10:34:26.269008 [DEBUG]
switch_core_state_machine.c:465 sofia/internal/sip:3008 at 192.168.15.176:21828
handler already called, skipping state handler.

2010-02-23 10:34:26.269008 [DEBUG]
switch_core_state_machine.c:333 (sofia/internal/sip:3008 at 192.168.15.176:21828)
State Change CS_HANGUP -> CS_REPORTING

2010-02-23 10:34:26.269008 [DEBUG] switch_core_session.c:999
Send signal sofia/internal/sip:3008 at 192.168.15.176:21828 [BREAK]

2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/sip:3008 at 192.168.15.176:21828) Running State Change
CS_REPORTING

2010-02-23 10:34:26.269008 [DEBUG]
switch_core_state_machine.c:579 (sofia/internal/sip:3008 at 192.168.15.176:21828)
State REPORTING

2010-02-23 10:34:26.269008 [DEBUG]
switch_core_state_machine.c:53 sofia/internal/sip:3008 at 192.168.15.176:21828
Standard REPORTING, cause: NORMAL_CLEARING

2010-02-23 10:34:26.269008 [DEBUG]
switch_core_state_machine.c:579 (sofia/internal/sip:3008 at 192.168.15.176:21828)
State REPORTING going to sleep

2010-02-23 10:34:26.269008 [DEBUG]
switch_core_state_machine.c:327 (sofia/internal/sip:3008 at 192.168.15.176:21828)
State Change CS_REPORTING -> CS_DESTROY

2010-02-23 10:34:26.269008 [DEBUG] switch_core_session.c:999
Send signal sofia/internal/sip:3008 at 192.168.15.176:21828 [BREAK]

2010-02-23 10:34:26.269008 [DEBUG]
switch_core_session.c:1136 Session 6
(sofia/internal/sip:3008 at 192.168.15.176:21828) Locked, Waiting on external
entities

2010-02-23 10:34:26.272437 [DEBUG] switch_core_state_machine.c:348
(sofia/internal/3007 at 192.168.15.177)
State EXECUTE going to sleep

2010-02-23 10:34:26.272437 [DEBUG]
switch_core_state_machine.c:314 (sofia/internal/3007 at 192.168.15.177)
Running State Change CS_HANGUP

2010-02-23 10:34:26.272437 [DEBUG] switch_core_state_machine.c:488
(sofia/internal/3007 at 192.168.15.177)
State HANGUP

2010-02-23 10:34:26.272437 [DEBUG] mod_sofia.c:358 Channel sofia/internal/3007 at 192.168.15.177
hanging up, cause: NORMAL_CLEARING

2010-02-23 10:34:26.272437 [DEBUG] mod_nibblebill.c:397
Attempting to bill at $1.0 per minute to account 3007

2010-02-23 10:34:26.272437 [DEBUG] mod_nibblebill.c:449 Last
successful billing time was 

2010-02-23 10:34:26.272437 [DEBUG] mod_nibblebill.c:455 4
seconds passed since last bill time of 2010-02-23 10:34:21

2010-02-23 10:34:26.272437 [DEBUG] mod_nibblebill.c:461
Billing $0.077045 to 3007 (Call: 3ec17ffc-c94e-46ff-bb0d-7ad0ef27bdab /
1.480343 so far)

2010-02-23 10:34:26.272437 [DEBUG]
switch_core_state_machine.c:46 sofia/internal/3007 at 192.168.15.177
Standard HANGUP, cause: NORMAL_CLEARING

 

Anyhelp getting nibblebill to connect to the database would
be greatly appreciated.  Thanks

 

 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
http://clk.atdmt.com/GBL/go/201469226/direct/01/
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100223/12a2dc15/attachment-0001.html 

From infos at madovsky.org  Tue Feb 23 10:09:55 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 23 Feb 2010 13:09:55 -0500
Subject: [Freeswitch-users] RTP timeout
Message-ID: <A68022CC2AFE40E29FE73385397A665B@MOBILEE1705>

Hi,

thanks to answer me if I misunderstood something,
but if I run a softphone on the same IP as FS, there
is an RTP timeout.

Any idea ?

Thanks

Franck
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100223/de7f5a9c/attachment.html 

From anthony.minessale at gmail.com  Tue Feb 23 11:13:45 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 23 Feb 2010 13:13:45 -0600
Subject: [Freeswitch-users] Question about sofia_contact
In-Reply-To: <4B83A2E7.1060905@ewetel.de>
References: <4B7EA954.30402@ewetel.de> <4B7EADBC.1040001@ewetel.de>
	<4B8249BF.3090708@ewetel.de>
	<191c3a031002220842j5bec442an5f1ea89cb0e8a6ff@mail.gmail.com>
	<4B83A2E7.1060905@ewetel.de>
Message-ID: <191c3a031002231113i5e838cf8g6354977d3b361e09@mail.gmail.com>

I added a patch that I think will allow what you want by being more strict
in the code about deciding if a string was meant to be a domain or profile
name.

On Tue, Feb 23, 2010 at 3:41 AM, Helmut Kuper <helmut.kuper at ewetel.de>wrote:

> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> Hello Anthony,
>
>
> you are right, I'm quite unpatient, sorry 4 that. Your solution works
> fine. I thought the sip domain could be any string and must not be a
> valid domain format.
>
> Thanks to you, board and community for this fantastic project!
>
> regards from rainy germany
> Helmut
>
>
>
> On 22.02.2010 17:42, Anthony Minessale wrote:
> > it's mad at you for asking twice before waiting for a reply, so it's not
> > working on purpose.
> >
> > Actually it's mad at you because your domain does not contain a . so it
> > is assuming you are specifying a profile name as the domain.  if your
> > domain was mydomain.com <http://mydomain.com> instead it would work.
> -----BEGIN PGP SIGNATURE-----
> Version: GnuPG v1.4.7 (MingW32)
>
> iD8DBQFLg6Ln4tZeNddg3dwRAvj6AJ9ruybNpbL8mdUlx1jVtLPYVbCSDACfQJLo
> zfieJnHZdp2Xv3OS6HTZE/k=
> =ESgY
> -----END PGP SIGNATURE-----
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100223/6c487fbd/attachment.html 

From anthony.minessale at gmail.com  Tue Feb 23 11:31:49 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 23 Feb 2010 13:31:49 -0600
Subject: [Freeswitch-users] Dingaling - external rtp supported?
In-Reply-To: <29b888f81002222359w6ae4cd8ej944d453a5b099af4@mail.gmail.com>
References: <29b888f81002160835y6e290319ub0d04c088caa1b79@mail.gmail.com>
	<DEA39AD1-0EE9-473A-B78C-CDDF3F100578@freeswitch.org>
	<29b888f81002170318p63622f3fhc8d26ad0e85602d2@mail.gmail.com>
	<191c3a031002170538r3b029515i8a000812a70ad558@mail.gmail.com>
	<29b888f81002170741y3d13cb57ycb4436a372c4eded@mail.gmail.com>
	<191c3a031002171136q7424e9bai34379abb542daf93@mail.gmail.com>
	<29b888f81002210800v617bca9dk4bf1abf1bd791ee0@mail.gmail.com>
	<29b888f81002222359w6ae4cd8ej944d453a5b099af4@mail.gmail.com>
Message-ID: <191c3a031002231131x19c9791cgd986a6c1be18e09a@mail.gmail.com>

If you are modifying your build to add libgcrypt / libgnutls to win32, you
have chosen an incompatible version of one of these libs.  We do not support
manually adding this modification to the code, you will need to find someone

else who has done it successfully to help you.




On Tue, Feb 23, 2010 at 1:59 AM, Moiz Chinoy <moizchinoy at gmail.com> wrote:

> Moreover, if I gtalk client is on the same machine as FS and i have
> following settings, FS crashes with the same mutex error.
>
> External Sip Profile has following lines:
> ---------------------------------------------------------
>   <param name="rtp-ip" value="$${local_ip_v4}"/>
>   <param name="sip-ip" value="$${local_ip_v4}"/>
>    <param name="ext-rtp-ip" value="auto-nat"/>
>    <param name="ext-sip-ip" value="auto-nat"/>
>
> Jingle Client.xml has following lines:
> -----------------------------------------------------
>   <param name="rtp-ip" value="$${bind_server_ip}"/>
>   <!--<param name="ext-rtp-ip" value="$${external_rtp_ip}"/>-->
>   <!--<param name="disable-rtp-auto-adjust" value="true"/>-->
>   <!--<param name="avatar" value="/path/to/tiny.jpg"/>-->
>   <!--<param name="candidate-acl" value="wan.auto"/>-->
>   <!--<param name="local-network-acl" value="localnet.auto"/>-->
>
> If I uncomment the following line in client.xml (Jingle profile)
>    <!--<param name="local-network-acl" value="localnet.auto"/>-->
> then exception does not happen.
>
> Is this a known issue or do I need to post it in JIRA?
>
> Tell me if more logs are needed...
>
>
> On Sun, Feb 21, 2010 at 8:00 PM, Moiz Chinoy <moizchinoy at gmail.com> wrote:
> > Guys,
> >
> > To make things simple gtalk client is entirely on different network.
> >
> > Call comes from outside through external Sip profile.
> >
> > If gtalk answers the call after 3-4 rings both parties can hear each
> other.
> > If gtalk answers the call after 2 rings both parties no one can hear each
> other.
> > If gtalk answers the call immediately FS crashes.
> >
> > Attached is the screen shot of the error...
> >
> > Here is the FS log...
> > --------------------------------
> > http://pastebin.freeswitch.org/12197
> >
> > External Sip Profile has following lines:
> > ---------------------------------------------------------
> >    <param name="rtp-ip" value="$${local_ip_v4}"/>
> >    <param name="sip-ip" value="$${local_ip_v4}"/>
> >    <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
> >    <param name="ext-sip-ip" value="$${external_sip_ip}"/>
> >
> > Jingle Client.xml has following lines:
> > -----------------------------------------------------
> >    <param name="rtp-ip" value="$${bind_server_ip}"/>
> >    <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
> >    <param name="disable-rtp-auto-adjust" value="true"/>
> >    <!--<param name="avatar" value="/path/to/tiny.jpg"/>-->
> >    <!--<param name="candidate-acl" value="wan.auto"/>-->
> >    <!--<param name="local-network-acl" value="localnet.auto"/>-->
> >
> > Vars.xml has following lines:
> > -------------------------------------------
> > <X-PRE-PROCESS cmd="set" data="external_rtp_ip=stun:stun.freeswitch.org
> "/>
> > <X-PRE-PROCESS cmd="set" data="external_sip_ip=stun:stun.freeswitch.org
> "/>
> >
> >
> > Please advise me how can I provide more of the required data.
> >
> > On Wed, Feb 17, 2010 at 11:36 PM, Anthony Minessale
> > <anthony.minessale at gmail.com> wrote:
> >> you cant combine stun and gtalk and boxes in the same lan very easily if
> you
> >> do need to do that you will need to mess with
> >>
> >> <param name="candidate-acl" value="wan.auto"/>
> >> <param name="local-network-acl" value="localnet.auto"/>
> >>
> >>
> >>
> >>
> >>
> >> On Wed, Feb 17, 2010 at 9:41 AM, Moiz Chinoy <moizchinoy at gmail.com>
> wrote:
> >>>
> >>> Guys I am unable to produce the crash but now both parties cannot hear
> >>> each other!
> >>>
> >>> Vars.xml has following lines:
> >>>  <X-PRE-PROCESS cmd="set"
> >>> data="external_rtp_ip=stun:stun.freeswitch.org"/>
> >>>  <X-PRE-PROCESS cmd="set"
> >>> data="external_sip_ip=stun:stun.freeswitch.org"/>
> >>>
> >>> Jingle Client.xml has following lines:
> >>>    <param name="rtp-ip" value="$${bind_server_ip}"/>
> >>>    <!-- <param name="ext-rtp-ip" value="auto-nat"/> -->
> >>>    <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
> >>>    <param name="disable-rtp-auto-adjust" value="true"/>
> >>>    <param name="local-network-acl" value="localnet.auto"/>
> >>>
> >>> On Wed, Feb 17, 2010 at 5:38 PM, Anthony Minessale
> >>> <anthony.minessale at gmail.com> wrote:
> >>> > Obtain a stack trace from the crash.
> >>> >
> >>> > On Feb 17, 2010 5:26 AM, "Moiz Chinoy" <moizchinoy at gmail.com> wrote:
> >>> >
> >>> > Hi,
> >>> >
> >>> > FS rev: 16673
> >>> > Platform: Windows
> >>> >
> >>> > More details:
> >>> >
> >>> > FS is behind NAT and machine is running a VPN connection.
> >>> >
> >>> > FS and GTalk client on the same machine:
> >>> >
> >>> >
> --------------------------------------------------------------------------------------------------
> >>> > jingle profile client.xml has following line:
> >>> > <param name="rtp-ip" value="$${bind_server_ip}"/>
> >>> >
> >>> > External SIP call is successfully bridged to GTalk client.
> >>> >
> >>> >
> >>> > FS and GTalk client on the different machine:
> >>> >
> >>> >
> --------------------------------------------------------------------------------------------------
> >>> > jingle profile client.xml has following lines:
> >>> > <param name="rtp-ip" value="$${bind_server_ip}"/>
> >>> >
> >>> > <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
> >>> >
> >>> > As soon as external SIP call land and I try to bridge the call to
> >>> > GTalk client, FS crashes.
> >>> >
> >>> >
> >>> > NAT Details:
> >>> > ---------------------------
> >>> > I think my NAT does not support UpNP or PMP. The reason I say it
> >>> > because when FS starts following message is displayed:
> >>> >
> >>> > 2010-02-17 09:15:42.809674 [INFO] switch_nat.c:409 Scanning for NAT
> >>> > 2010-02-17 09:15:42.809674 [ERR] switch_nat.c:197 Error checking for
> >>> > PMP [init failed]
> >>> > 2010-02-17 09:15:42.809674 [DEBUG] switch_nat.c:414 Checking for UPnP
> >>> > 2010-02-17 09:15:56.825388 [DEBUG] switch_nat.c:114 No
> >>> > InternetGatewayDevice, using first entry as default
> >>> > (http://192.168.16.17:50144/).
> >>> > 2010-02-17 09:15:56.887889 [INFO] switch_nat.c:429 No PMP or UPnP NAT
> >>> > devices detected!
> >>> >
> >>> >
> >>> >
> >>> > On Tue, Feb 16, 2010 at 8:41 PM, Brian West <brian at freeswitch.org>
> >>> > wrote:
> >>> >> can you please update...
> >>> >
> >>> > _______________________________________________
> >>> > FreeSWITCH-users mailing list
> >>> > FreeSWITCH-users at lists.freeswitch.org
> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> > http://www.freeswitch.org
> >>> >
> >>> >
> >>>
> >>>
> >>>
> >>> --
> >>> Regards,
> >>> Moiz Chinoy.
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>
> >>
> >>
> >> --
> >> Anthony Minessale II
> >>
> >> FreeSWITCH http://www.freeswitch.org/
> >> ClueCon http://www.cluecon.com/
> >> Twitter: http://twitter.com/FreeSWITCH_wire
> >>
> >> AIM: anthm
> >> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >> IRC: irc.freenode.net #freeswitch
> >>
> >> FreeSWITCH Developer Conference
> >> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> >> iax:guest at conference.freeswitch.org/888
> >> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >> pstn:+19193869900
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >
> >
> >
> > --
> > Regards,
> > Moiz Chinoy.
> >
>
>
>
> --
> Regards,
> Moiz Chinoy.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100223/5c7e9c74/attachment-0001.html 

From rob4manhere at gmail.com  Tue Feb 23 11:51:22 2010
From: rob4manhere at gmail.com (Rob Forman)
Date: Tue, 23 Feb 2010 13:51:22 -0600
Subject: [Freeswitch-users] SIP provider recommendation for US termination
Message-ID: <F56BB50F-6584-4235-91C4-54E4C6EF4867@gmail.com>

Hey all,

I'm having on-going sporadic issues with one of my SIP providers (call  
quality, delayed or lost DTMFs, high random PPD).  Does anyone have  
some good experiences (for US termination) in terms of both quality  
and support?  There are so many bad ones out there; I don't want to  
switch blindly.

I don't know if we're supposed to share commercial endorsements on  
here.  If you have advice, would you mind dropping me a note off-list  
at rob4manhere (at) gmail.com.

Many thanks,
Rob


From joseph.puchalski at personalcyberspace.com  Tue Feb 23 12:12:41 2010
From: joseph.puchalski at personalcyberspace.com (Joseph Puchalski)
Date: Tue, 23 Feb 2010 20:12:41 +0000
Subject: [Freeswitch-users] How to set outbound caller id info
	for	multiple users/extensions
In-Reply-To: <4B834719.3000505@aktzero.com>
References: <093DD565390C1E4FB15D7B383E86BB05AF05AB@Goose.personalcyberspace.net>
	<191c3a031002130707v5f1d5ee0lb9e791e9a0aab956@mail.gmail.com>
	<3FF62C88-2423-43F0-B8A3-C64EF4BC80AC@freeswitch.org>
	<093DD565390C1E4FB15D7B383E86BB05AF15CD@Goose.personalcyberspace.net>
	<4B834719.3000505@aktzero.com>
Message-ID: <093DD565390C1E4FB15D7B383E86BB05AF1857@Goose.personalcyberspace.net>

Thanks, this helps. Setting "effective_caller_id*" in my extension xml file doesn't work for me at all.

Something must be wrong somewhere else in my config.

I think I'll probably go back and reinstall from the beginning. When I did this initially I made some xml changes late at night that seemed logical at the time.

Thanks again,

Joe

From: Andrew Thompson [mailto:andrewkt at aktzero.com]
Sent: Monday, February 22, 2010 10:10 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] How to set outbound caller id info for multiple users/extensions

On 2/22/2010 7:24 PM, Joseph Puchalski wrote:
Or should I be doing this via <action application="set" data="effective_caller_id_number=${outbound_caller_id_number}"/>
Somewhere else?

I have the following set on my own extension, in 1000.xml:

      <variable name="effective_caller_id_name" value="Andrew Thompson"/>
      <variable name="effective_caller_id_number" value="1000"/>
      <variable name="outbound_caller_id_name" value="MyCompany"/>
      <variable name="outbound_caller_id_number" value="9109991234"/>

When I dial extensions internally, the effective_* name/number show up.

When I dial outbound via my SIP provider, I set the following before the bridge so that it passes externally valid info:

      <action application="set" data="effective_caller_id_number=${outbound_caller_id_number}"/>
      <action application="set" data="effective_caller_id_name=${outbound_caller_id_name}"/>


In my setup, if I don't explicitly overide the effective_* with outbound_*, I actually see 1000 as my callerid when I call my cell from my extension, so if you're not getting at least that much, something else might be wrong. (I have used vitelity, and they do pass callerid properly most of the time.)



--

Andrew Thompson
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100223/b9450fcf/attachment.html 

From william.suffill at gmail.com  Tue Feb 23 12:43:59 2010
From: william.suffill at gmail.com (William Suffill)
Date: Tue, 23 Feb 2010 15:43:59 -0500
Subject: [Freeswitch-users] SIP provider recommendation for US
	termination
In-Reply-To: <F56BB50F-6584-4235-91C4-54E4C6EF4867@gmail.com>
References: <F56BB50F-6584-4235-91C4-54E4C6EF4867@gmail.com>
Message-ID: <6b65470d1002231243x4268de5di655831071c9a28ab@mail.gmail.com>

There is a freeswitch-biz list too. I'm sure more people are faced with this
issue as well so it might be a good topic for the biz list.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100223/2487f618/attachment.html 

From m.sobkow at marketelsystems.com  Tue Feb 23 13:00:46 2010
From: m.sobkow at marketelsystems.com (Mark Sobkow)
Date: Tue, 23 Feb 2010 15:00:46 -0600
Subject: [Freeswitch-users] mod_erlang_event
Message-ID: <4B8441FE.80506@marketelsystems.com>

It's become clear that I need to use Erlang event processing to do what 
I need to do with Freeswitch, but I can't even get the most basic of 
tasks working yet.  (i.e. Answer the call and collect the PIN code from 
the operator.)

The dialplan version of what I'm trying to do is:


<extension name=\"OpCallsIn\">
    <condition field=\"destination_number\" expression=\"^6100$\">
        <action application=\"answer\" />
        <action application=\"play_and_get_digits\" data=\"4 4 1 5000 # 
/opt/freeswitch/sounds/en/us/callie/conference/8000/conf-pin.wav 
/opt/freeswitch/sounds/en/us/callie/conference/8000/conf-bad-pin.wav 
operator_pin \\d+\" />
        <action application=\"log\" data=\"INFO operator_pin is 
${operator_pin}\" />
        <action application=\"log\" data=\"INFO Queueing operator to 
fifo\" />
        <action application=\"fifo\" data=\"6100@${domain_name} in\" />
    </condition>
</extension>

Attached is the Erlang that's attempting to do the same thing.  The 
Erlang is invoked by the following dialplan fragment:

<extension name=\"ErlangCallback\">
    <condition field=\"destination_number\" expression=\"^6200$\">
        <action application=\"erlang\" data=\"pbx_callback 
pursuit at testsrv\" />
    </condition>
</extension>

Any suggestions?

-- 
Mark Sobkow
Senior Developer
MarkeTel Multi-Line Dialing Systems LTD.
428 Victoria Ave
Regina, SK S4N-0P6
Toll-Free: 800-289-8616-X533
Local: 306-359-6893-X533
Fax: 306-359-6879
Email: m.sobkow at marketelsystems.com
Web: http://www.marketelsystems.com 

-------------- next part --------------
An embedded and charset-unspecified text was scrubbed...
Name: pbx_callback.erl
Url: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100223/55402511/attachment.pl 

From msc at freeswitch.org  Tue Feb 23 13:35:24 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 23 Feb 2010 13:35:24 -0800
Subject: [Freeswitch-users] Hook Flash
In-Reply-To: <742756.71167.qm@web33501.mail.mud.yahoo.com>
References: <915F9D0D-FB13-4D02-995B-FB7F5EB488D2@tlainvestments.com>
	<742756.71167.qm@web33501.mail.mud.yahoo.com>
Message-ID: <87f2f3b91002231335o8c0b6f4vee98dbcd4d2b994d@mail.gmail.com>

On Tue, Feb 23, 2010 at 5:33 AM, Diego Toro <dftoro at yahoo.com> wrote:

> hi, read http://jira.freeswitch.org/browse/OPENZAP-30
>
>  <action application="send_dtmf" data="F"/>
>
> Diego Toro
> http://lacarretade.blogspot.com/
>

FYI, I didn't see this on the wiki so I added it to the OpenZAP FAQ:
http://wiki.freeswitch.org/wiki/OpenZAP#FAQ

Thanks Diego,
MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100223/d66f268e/attachment.html 

From msc at freeswitch.org  Tue Feb 23 13:47:04 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 23 Feb 2010 13:47:04 -0800
Subject: [Freeswitch-users] call from an internal extension to external
	number
In-Reply-To: <D616CD3E2BFC4FDDA7ABEE7A3F268A4B@MOBILEE1705>
References: <D616CD3E2BFC4FDDA7ABEE7A3F268A4B@MOBILEE1705>
Message-ID: <87f2f3b91002231347k6fd1e5b8wc6543ba2696b1629@mail.gmail.com>

On Mon, Feb 22, 2010 at 7:29 PM, Madovsky <infos at madovsky.org> wrote:

>  Hi,
>
> day after I undertand a littlee more all these xml hell files (not friendly
> to read ;)),
>
Use a text editor that does syntax highlighting. :)


> but to be a PERl developer since 1999 understand regex and PERL language
> make life more easy...
>
Hint: don't say PERL, say Perl or perl instead. People who say PERL are
considered uneducated.


> However, I don't understand yet the concept of internal exterenal.
> is it for phone registration AND outbound calls ?
>
Internal and external are SIP profiles. Each SIP profile is a SIP user
agent, or UA. For a more complete discussion on this topic check out
http://en.wikipedia.org/wiki/User_agent

In short, the internal profile is listening on a particular IP and port and
usually it's to listen for registrations and calls from your telephones, as
well as to send calls out to your telephones. The external profile is
generally used just for outbound gateway registrations.


> for now I try to make an external call from 1000 ext (registered on port
> 5060)
> so I added an extension in dialplan/default.xml
>
>     <extension name="outbound_europe">
>         <condition field="destination_number" expression="^00?(\d+)$">
>               <action application="bridge"
> data="sofia/gateway/myprovider_europe/00$1"/>
>         </condition>
>     </extension>
> so if call starts with "00" it redirects to my provider that manage
> outbound calls, is it correct ?
> I put the myprovider.xml account into sip_profiles/external/myprovider.xml.
>
At first look this appears correct. You can make sure that the gateway is up
by typing "sofia status" at the fs_cli prompt. If you are having trouble
with making calls it is best to watch the debug output very carefully. It is
a lot of information to look at but eventually you will learn to focus on
the information that you need.

You're doing well! Just keep plugging away at it and you will figure it all
out and soon you will be helping others. :)
-MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100223/4a85626a/attachment-0001.html 

From msc at freeswitch.org  Tue Feb 23 13:57:51 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 23 Feb 2010 13:57:51 -0800
Subject: [Freeswitch-users] Error loading
	/usr/local/freeswitch/mod/ozmod_sangoma_boost.so
In-Reply-To: <fe67a3341002230202y1d45a8f3t3a2cda07de2240a2@mail.gmail.com>
References: <fe67a3341002230202y1d45a8f3t3a2cda07de2240a2@mail.gmail.com>
Message-ID: <87f2f3b91002231357t6c3e40cdld6a0a9861a9ec5a5@mail.gmail.com>

Does the file ozmod_sangoma_boost.so exist somewhere in your openzap build
tree? If not then something went wrong while compiling or you have an old
version. If it does exist, do a quick test: cp the file into
/usr/local/freeswitch/mod and restart FreeSWITCH and see if it finds the
file and loads OpenZAP properly. Let us know the results so we can determine
if it's a bug in the build system or not.

-MC

On Tue, Feb 23, 2010 at 2:02 AM, Nagalenoj H. <nagalenoj at gmail.com> wrote:

> Dear friends,
>     I've installed freeswitch trunk - 16729 and tried to configure with
> wanpipe for sangoma A102 pri card.
>
> Followed the steps given in
> http://wiki.sangoma.com/wanpipe-freeswitch-install
>
> When loading the freeswitch, I've got the following error.
>
> 2010-02-23 14:49:58.545726 [INFO] ozmod_wanpipe.c:335 configuring device
> s2c15 as OpenZAP device 1:30 fd:57 DTMF: software
> 2010-02-23 14:49:58.546004 [ERR] ozmod_wanpipe.c:436 No valid wanpipe span
> and channel was specified
> 2010-02-23 14:49:58.546275 [INFO] zap_io.c:2500 Configured 30 channel(s)
> 2010-02-23 14:49:58.560724 [ERR] zap_io.c:2560 Error loading
> /usr/local/freeswitch/mod/ozmod_sangoma_boost.so
> [/usr/local/freeswitch/mod/ozmod_sangoma_boost.so: cannot open shared object
> file: No such file or directory]
> 2010-02-23 14:49:58.561048 [ERR] zap_io.c:2720 can't find 'sangoma_boost'
> 2010-02-23 14:49:58.561266 [ERR] mod_openzap.c:2458 Error starting OpenZAP
> span 1 error:
> 2010-02-23 14:49:58.561712 [NOTICE] switch_loadable_module.c:144 Adding
> Endpoint 'openzap'
>
> Configuration and log files are pasted to pastebin. Kindly someone help me
> to solve this issue.
>
> openzap.conf and openzap.conf.xml
>    http://pastebin.freeswitch.org/12214
>
> freeswitch log
>    http://pastebin.freeswitch.org/12216
>
> smg_pri.conf
>    http://pastebin.freeswitch.org/12217
>
> --
> Regards,
> Nagalenoj H.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100223/10258572/attachment.html 

From msc at freeswitch.org  Tue Feb 23 14:00:09 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 23 Feb 2010 14:00:09 -0800
Subject: [Freeswitch-users] big thanks to all freeswitch developers and
	contributing users
In-Reply-To: <4B83F040.7040005@xpirio.com>
References: <4B83F040.7040005@xpirio.com>
Message-ID: <87f2f3b91002231400h146b48ckb26aa407945b5979@mail.gmail.com>

2010/2/23 Christian L?schenkohl <christian.loeschenkohl at xpirio.com>

> i want to say a big THANKY YOU to all contributing freeswitch community
> members.
>
> over one year has passed since i did fall in love with this project.
> it is getting better every day, one get's help and advices if needed.
> the admins do care about nearly every problem - no matter if it's big or
> small.
> i also did manage an opensource project and i wish i had done it with that
> much
> heart and intense power that i see here.
>
> i also hope that i can contribute back enough (questions, bug reports, wiki
> enhancements).
>
> Don't forget "sitting on IRC all day long helping newcomers!" :)
-MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100223/1d11fbfa/attachment.html 

From jeff at jefflenk.com  Tue Feb 23 14:05:33 2010
From: jeff at jefflenk.com (Jeff Lenk)
Date: Tue, 23 Feb 2010 16:05:33 -0600
Subject: [Freeswitch-users] FScomm
In-Reply-To: <D64783281177440B9094B8181E55EFA7@MOBILEE1705>
References: <157FDB96FB88476FBAEC69C1309A0A5F@MOBILEE1705>,
	<COL117-W5563292C886257AFC56548B9420@phx.gbl>,
	<D64783281177440B9094B8181E55EFA7@MOBILEE1705>
Message-ID: <COL117-W429F3BED9F781A58264238B9420@phx.gbl>


http://wiki.freeswitch.org/wiki/FSComm#Linux

 

you must run those from the FSComm directory
 


From: infos at madovsky.org
To: freeswitch-users at lists.freeswitch.org
Date: Tue, 23 Feb 2010 11:51:51 -0500
Subject: Re: [Freeswitch-users] FScomm




 

----- Original Message ----- 
From: Jeff Lenk 
To: freeswitch-users at lists.freeswitch.org 
Sent: Tuesday, February 23, 2010 9:23 AM
Subject: Re: [Freeswitch-users] FScomm

What platform are you trying to build?
 


From: infos at madovsky.org
To: freeswitch-users at lists.freeswitch.org
Date: Tue, 23 Feb 2010 01:41:04 -0500
Subject: [Freeswitch-users] FScomm




Hi,
 
is http://wiki.freeswitch.org/wiki/FSComm
available yet ? because QT framework link has broken
and some instruction don't work with last svn trunk (for example to compile FScomm in FS svn root)
 
Thanks

Franck


Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. Sign up now. 





_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

 
FSComm on Linux fedora 10 64 bits
It says FSComm can be built inside FS svn folder typing gmake make
but there is no Makefile inside
 
Thanks

Franck 		 	   		  
_________________________________________________________________
Hotmail: Powerful Free email with security by Microsoft.
http://clk.atdmt.com/GBL/go/201469230/direct/01/
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100223/842c5a3d/attachment.html 

From andrew at hijacked.us  Tue Feb 23 14:49:02 2010
From: andrew at hijacked.us (Andrew Thompson)
Date: Tue, 23 Feb 2010 17:49:02 -0500
Subject: [Freeswitch-users] mod_erlang_event
In-Reply-To: <4B8441FE.80506@marketelsystems.com>
References: <4B8441FE.80506@marketelsystems.com>
Message-ID: <20100223224902.GB1751@hijacked.us>

Comments inline.

On Tue, Feb 23, 2010 at 03:00:46PM -0600, Mark Sobkow wrote:
> It's become clear that I need to use Erlang event processing to do what 
> I need to do with Freeswitch, but I can't even get the most basic of 
> tasks working yet.  (i.e. Answer the call and collect the PIN code from 
> the operator.)
> 
> The dialplan version of what I'm trying to do is:
> 
> 
> <extension name=\"OpCallsIn\">
>    <condition field=\"destination_number\" expression=\"^6100$\">
>        <action application=\"answer\" />
>        <action application=\"play_and_get_digits\" data=\"4 4 1 5000 # 
> /opt/freeswitch/sounds/en/us/callie/conference/8000/conf-pin.wav 
> /opt/freeswitch/sounds/en/us/callie/conference/8000/conf-bad-pin.wav 
> operator_pin \\d+\" />
>        <action application=\"log\" data=\"INFO operator_pin is 
> ${operator_pin}\" />
>        <action application=\"log\" data=\"INFO Queueing operator to 
> fifo\" />
>        <action application=\"fifo\" data=\"6100@${domain_name} in\" />
>    </condition>
> </extension>
> 
> Attached is the Erlang that's attempting to do the same thing.  The 
> Erlang is invoked by the following dialplan fragment:
> 
> <extension name=\"ErlangCallback\">
>    <condition field=\"destination_number\" expression=\"^6200$\">
>        <action application=\"erlang\" data=\"pbx_callback 
> pursuit at testsrv\" />
>    </condition>
> </extension>
> 
> Any suggestions?

Why not request the pin in the dialplan and then yield call control to
erlang? That's what I do most of the time.
 

> %% Author: mark
> %% Created: Feb 23, 2010
> %% Description: TODO: Add description to pbx_callback
> -module(pbx_callback).
> 
> %%
> %% Include files
> %%
> 
> %%
> %% Exported Functions
> %%
> -export([start/0, run/0, launch/1]).
> 
> start() ->
> 	Pid = spawn( ?MODULE, run, [] ),
> 	register( ?MODULE, Pid ),
> 	{ ok, Pid }.
> 
> run() ->
> 	receive
> 		{ call, Data } ->
> 			{ event, [UUID | Rest]} = Data,
> 			syslog:debug( "pbx_callback:run() New call received, UUID=~p, Rest=~p~n", [UUID, Rest] ),
> 			AnswerResults = pbx:api( eval, "uuid:" ++ UUID ++ " answer" ),
> 			syslog:debug( "pbx_callback:run() AnswerResults=~p~n", [AnswerResults] ),
> 			GetPinResults = pbx:api( eval, "uuid:" ++ UUID ++ " play_and_get_digits 4 4 1 5000 # /opt/freeswitch/sounds/en/us/callie/conference/8000/conf-pin.wav /opt/freeswitch/sounds/en/us/callie/conference/8000/conf-bad-pin.wav operator_pin \\d+" ),
> 			syslog:debug( "pbx_callback:run() GetPinResults=~p~n", [GetPinResults] ),
> 			GetPinVarResults = pbx:api( uuid_getvar, UUID ++ " operator_pin" ),
> 			syslog:debug( "pbx_callback:run() GetPinVarResults=~p~n", [GetPinVarResults] ),
> 			run();
> 		{call_event, Data} ->
> 			{ event, [UUID | Rest]} = Data,
> 			Name = proplists:get_value( "Event-Name", Rest ),
> 			syslog:debug( "pbx_callback:run() call_event UUID=~p, Name=~p, Rest=~p~n", [UUID, Name, Rest] ),
> 			run();
> 		{get_pid, UUID, Ref, Pid} ->
> 			NewPid = spawn( ?MODULE, run, [] ),
> 			syslog:debug( "pbx_callback:run() Request to spawn new handler process, returning PID ~p~n", [NewPid] ),
> 			Pid ! { Ref, NewPid },
> 			run()
> 	end.
> 
> launch( Ref ) ->
> 	NewPid = spawn( ?MODULE, run, [] ),
> 	syslog:debug( "pbx_callback:launch() Returning new PID ~p~n", [NewPid] ),
> 	{Ref, NewPid}.

I don't know what your 'pbx' module is doing so I can't really help you
there. Are you doing a sendmsg for play_and_get_digits or what? You
should be using a uuid_getvar to get the result of the
play_and_get_digits in any case. How far does this code get before
failing?

Andrew


From infos at madovsky.org  Tue Feb 23 15:18:58 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 23 Feb 2010 18:18:58 -0500
Subject: [Freeswitch-users] FScomm
References: <157FDB96FB88476FBAEC69C1309A0A5F@MOBILEE1705>,
	<COL117-W5563292C886257AFC56548B9420@phx.gbl>,
	<D64783281177440B9094B8181E55EFA7@MOBILEE1705>
	<COL117-W429F3BED9F781A58264238B9420@phx.gbl>
Message-ID: <11F8492DFC2546598E6FDC3E538301CD@MOBILEE1705>


----- Original Message ----- 
From: Jeff Lenk
To: freeswitch-users at lists.freeswitch.org
Sent: Tuesday, February 23, 2010 5:05 PM
Subject: Re: [Freeswitch-users] FScomm


http://wiki.freeswitch.org/wiki/FSComm#Linux

you must run those from the FSComm directory



From: infos at madovsky.org
To: freeswitch-users at lists.freeswitch.org
Date: Tue, 23 Feb 2010 11:51:51 -0500
Subject: Re: [Freeswitch-users] FScomm



----- Original Message ----- 
From: Jeff Lenk
To: freeswitch-users at lists.freeswitch.org
Sent: Tuesday, February 23, 2010 9:23 AM
Subject: Re: [Freeswitch-users] FScomm


What platform are you trying to build?



From: infos at madovsky.org
To: freeswitch-users at lists.freeswitch.org
Date: Tue, 23 Feb 2010 01:41:04 -0500
Subject: [Freeswitch-users] FScomm


Hi,

is http://wiki.freeswitch.org/wiki/FSComm
available yet ? because QT framework link has broken
and some instruction don't work with last svn trunk (for example to compile 
FScomm in FS svn root)

Thanks

Franck



Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. Sign up now.




_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


FSComm on Linux fedora 10 64 bits
It says FSComm can be built inside FS svn folder typing gmake make
but there is no Makefile inside

Thanks

Franck



Hotmail: Powerful Free email with security by Microsoft. Get it now.



_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



It's what I did,
but from FS trunk, inside fscomm directory,
there s only

account.cpp  conf                fshost.h        mainwindow.ui 
resources.qrc
account.h    FSComm.2008.vcproj  main.cpp        mod_qsettings
call.cpp     FSComm.pro          mainwindow.cpp  preferences
call.h       fshost.cpp          mainwindow.h    resources



From brian at microcomaustralia.com.au  Tue Feb 23 15:24:31 2010
From: brian at microcomaustralia.com.au (Brian May)
Date: Wed, 24 Feb 2010 10:24:31 +1100
Subject: [Freeswitch-users] Possible typo on Openzap.conf wiki page
	specifying fxs/fxo-channels
In-Reply-To: <2C33D9D6C1444AAC80950CDF8D38A428@greyhawk.tonecommander.com>
References: <2C33D9D6C1444AAC80950CDF8D38A428@greyhawk.tonecommander.com>
Message-ID: <3c5cf5261002231524n4e9af307y80b3accb1ee4ff26@mail.gmail.com>

On 24 February 2010 04:48, Robert Hadley <robert.hadley at teotech.com> wrote:
> On the http://wiki.freeswitch.org/wiki/Openzap.conf_Examples wiki page, is
> there a typo in the wanpipe /usr/local/freeswitch/conf/openzap.conf example
> concerning specifying the fxo-channel vs. fxs-channel?

I agree. The first example looks correct to me; the 2nd example looks wrong.

See the table I created to try and explain what term to use where:

http://wiki.freeswitch.org/wiki/OpenZAP#FXO.2FFXS_Terminology

In the examples you quoted, ports 1 and 2 are extension ports, so are
FXS ports, but should be defined as FXO ports in openzap.conf.

Ports 3 and 4 are telephone line ports, so are FXO ports, but should
be defined as FXS ports in openzap.conf.

I have only used zaptel myself, however I suspect the same applies to wanpipe.
-- 
Brian May <brian at microcomaustralia.com.au>


From infos at madovsky.org  Tue Feb 23 15:29:29 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 23 Feb 2010 18:29:29 -0500
Subject: [Freeswitch-users] call from an internal extension to
	externalnumber
References: <D616CD3E2BFC4FDDA7ABEE7A3F268A4B@MOBILEE1705>
	<87f2f3b91002231347k6fd1e5b8wc6543ba2696b1629@mail.gmail.com>
Message-ID: <9130B3FED335446C85DB26E71489FB91@MOBILEE1705>


  ----- Original Message ----- 
  From: Michael Collins 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Tuesday, February 23, 2010 4:47 PM
  Subject: Re: [Freeswitch-users] call from an internal extension to externalnumber





  On Mon, Feb 22, 2010 at 7:29 PM, Madovsky <infos at madovsky.org> wrote:

    Hi,

    day after I undertand a littlee more all these xml hell files (not friendly to read ;)),
  Use a text editor that does syntax highlighting. :)
   

    but to be a PERl developer since 1999 understand regex and PERL language make life more easy...
  Hint: don't say PERL, say Perl or perl instead. People who say PERL are considered uneducated.
   

    However, I don't understand yet the concept of internal exterenal.
    is it for phone registration AND outbound calls ?
  Internal and external are SIP profiles. Each SIP profile is a SIP user agent, or UA. For a more complete discussion on this topic check out http://en.wikipedia.org/wiki/User_agent

  In short, the internal profile is listening on a particular IP and port and usually it's to listen for registrations and calls from your telephones, as well as to send calls out to your telephones. The external profile is generally used just for outbound gateway registrations.
   
    for now I try to make an external call from 1000 ext (registered on port 5060)
    so I added an extension in dialplan/default.xml 

        <extension name="outbound_europe">
            <condition field="destination_number" expression="^00?(\d+)$">
                  <action application="bridge" data="sofia/gateway/myprovider_europe/00$1"/>
            </condition>
        </extension>

    so if call starts with "00" it redirects to my provider that manage outbound calls, is it correct ?
    I put the myprovider.xml account into sip_profiles/external/myprovider.xml.
  At first look this appears correct. You can make sure that the gateway is up by typing "sofia status" at the fs_cli prompt. If you are having trouble with making calls it is best to watch the debug output very carefully. It is a lot of information to look at but eventually you will learn to focus on the information that you need.

  You're doing well! Just keep plugging away at it and you will figure it all out and soon you will be helping others. :)
  -MC





------------------------------------------------------------------------------


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  http://www.freeswitch.org


  Use a text editor that does syntax highlighting. :)

  black an white on my putty ssh ;)


  Thanks for your help !

  Franck
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100223/4903abf4/attachment.html 

From infos at madovsky.org  Tue Feb 23 15:37:02 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 23 Feb 2010 18:37:02 -0500
Subject: [Freeswitch-users] FreeSWITCH manual
Message-ID: <8E2F03C27AD4415BBDF1E9FF16F3DA98@MOBILEE1705>

Hi dev friends !

Is this manual yet available for the last trunk version ?

http://www.scribd.com/doc/17425068/Free-Switch-in-Real-Life

Thanks

Franck
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100223/99101e64/attachment.html 

From rupa at rupa.com  Tue Feb 23 15:37:41 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Tue, 23 Feb 2010 17:37:41 -0600
Subject: [Freeswitch-users] mod_nibblebill
In-Reply-To: <BLU110-W28EA8BD06A601D5D85268BA7420@phx.gbl>
References: <BLU110-W28EA8BD06A601D5D85268BA7420@phx.gbl>
Message-ID: <b39273531002231537k626d1698vd88549a7f59e73a@mail.gmail.com>

what database backend are you using?  Have you verified the SQL is going to
the right database backend?  I use mod_nibblebill against postgresql w/out
problems.

On Tue, Feb 23, 2010 at 9:42 AM, Meg Stroodle <phunk0000 at hotmail.com> wrote:

>  Hello List!  I am trying to install mod_nibblebill on my FS
> installation.  I get the following log entry form FS & nibblebill, but the
> database table I setup remains unchanged.  Any help in this matter would be
> greatly appreciated.  Following is an excerpt from the FS log:
>
>
>
> 2010-02-23 10:32:52.869149 [DEBUG] switch_rtp.c:2004 Correct ip/port
> confirmed.
>
> 2010-02-23 10:32:53.077162 [DEBUG] sofia.c:3727 Channel
> sofia/internal/3007 at 192.168.15.177 entering state [ready][200]
>
> 2010-02-23 10:32:53.149166 [DEBUG] switch_core_session.c:706 Send signal
> sofia/internal/sip:3008 at 192.168.15.176:21828 [BREAK]
>
> 2010-02-23 10:32:53.149166 [DEBUG] switch_core_session.c:706 Send signal
> sofia/internal/3007 at 192.168.15.177 [BREAK]
>
> 2010-02-23 10:32:53.149166 [DEBUG] switch_ivr_bridge.c:131 sofia/internal/
> sip:3008 at 192.168.15.176:21828 receive message [DISPLAY]
>
> 2010-02-23 10:32:53.169168 [DEBUG] switch_ivr_bridge.c:131
> sofia/internal/3007 at 192.168.15.177 receive message [DISPLAY]
>
> 2010-02-23 10:33:21.388949 [DEBUG] mod_nibblebill.c:529 Received request
> via SESSION_HEARTBEAT!
>
> 2010-02-23 10:33:21.388949 [DEBUG] mod_nibblebill.c:397 Attempting to bill
> at $1.0 per minute to account 3007
>
> 2010-02-23 10:33:21.388949 [INFO] mod_nibblebill.c:447 Beginning new
> billing on 3ec17ffc-c94e-46ff-bb0d-7ad0ef27bdab
>
> 2010-02-23 10:33:21.388949 [DEBUG] mod_nibblebill.c:455 28 seconds passed
> since last bill time of 2010-02-23 10:32:52
>
> 2010-02-23 10:33:21.388949 [DEBUG] mod_nibblebill.c:461 Billing $0.475997
> to 3007 (Call: 3ec17ffc-c94e-46ff-bb0d-7ad0ef27bdab / 0.000000 so far)
>
> 2010-02-23 10:33:51.508822 [DEBUG] mod_nibblebill.c:529 Received request
> via SESSION_HEARTBEAT!
>
> 2010-02-23 10:33:51.508822 [DEBUG] mod_nibblebill.c:397 Attempting to bill
> at $1.0 per minute to account 3007
>
> 2010-02-23 10:33:51.508822 [DEBUG] mod_nibblebill.c:449 Last successful
> billing time was
>
> 2010-02-23 10:33:51.508822 [DEBUG] mod_nibblebill.c:455 30 seconds passed
> since last bill time of 2010-02-23 10:33:21
>
> 2010-02-23 10:33:51.508822 [DEBUG] mod_nibblebill.c:461 Billing $0.501998
> to 3007 (Call: 3ec17ffc-c94e-46ff-bb0d-7ad0ef27bdab / 0.475997 so far)
>
> 2010-02-23 10:34:21.649708 [DEBUG] mod_nibblebill.c:529 Received request
> via SESSION_HEARTBEAT!
>
> 2010-02-23 10:34:21.649708 [DEBUG] mod_nibblebill.c:397 Attempting to bill
> at $1.0 per minute to account 3007
>
> 2010-02-23 10:34:21.649708 [DEBUG] mod_nibblebill.c:449 Last successful
> billing time was
>
> 2010-02-23 10:34:21.649708 [DEBUG] mod_nibblebill.c:455 30 seconds passed
> since last bill time of 2010-02-23 10:33:51
>
> 2010-02-23 10:34:21.649708 [DEBUG] mod_nibblebill.c:461 Billing $0.502348
> to 3007 (Call: 3ec17ffc-c94e-46ff-bb0d-7ad0ef27bdab / 0.977995 so far)
>
> 2010-02-23 10:34:26.254308 [NOTICE] sofia.c:329 Hangup
> sofia/internal/3007 at 192.168.15.177 [CS_EXECUTE] [NORMAL_CLEARING]
>
> 2010-02-23 10:34:26.254308 [DEBUG] switch_channel.c:1912 Send signal
> sofia/internal/3007 at 192.168.15.177 [KILL]
>
> 2010-02-23 10:34:26.254308 [DEBUG] switch_core_session.c:999 Send signal
> sofia/internal/3007 at 192.168.15.177 [BREAK]
>
> 2010-02-23 10:34:26.254308 [DEBUG] switch_core_state_machine.c:459
> sofia/internal/3007 at 192.168.15.177 thread mismatch skipping state handler.
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_ivr_bridge.c:470
> sofia/internal/3007 at 192.168.15.177 ending bridge by request from read
> function
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_ivr_bridge.c:563 BRIDGE THREAD
> DONE [sofia/internal/3007 at 192.168.15.177]
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_ivr_bridge.c:565 Send signal
> sofia/internal/sip:3008 at 192.168.15.176:21828 [BREAK]
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_ivr_bridge.c:520 sofia/internal/
> sip:3008 at 192.168.15.176:21828 receive message [UNBRIDGE]
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_session.c:645 Send signal
> sofia/internal/sip:3008 at 192.168.15.176:21828 [BREAK]
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_ivr_bridge.c:563 BRIDGE THREAD
> DONE [sofia/internal/sip:3008 at 192.168.15.176:21828]
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_ivr_bridge.c:565 Send signal
> sofia/internal/3007 at 192.168.15.177 [BREAK]
>
> 2010-02-23 10:34:26.269008 [NOTICE] switch_ivr_bridge.c:617 Hangup
> sofia/internal/sip:3008 at 192.168.15.176:21828 [CS_EXCHANGE_MEDIA]
> [NORMAL_CLEARING]
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_channel.c:1912 Send signal
> sofia/internal/sip:3008 at 192.168.15.176:21828 [KILL]
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_session.c:999 Send signal
> sofia/internal/sip:3008 at 192.168.15.176:21828 [BREAK]
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:488
> (sofia/internal/sip:3008 at 192.168.15.176:21828) State HANGUP
>
> 2010-02-23 10:34:26.269008 [DEBUG] mod_sofia.c:358 Channel sofia/internal/
> sip:3008 at 192.168.15.176:21828 hanging up, cause: NORMAL_CLEARING
>
> 2010-02-23 10:34:26.269008 [DEBUG] mod_sofia.c:400 Sending BYE to
> sofia/internal/sip:3008 at 192.168.15.176:21828
>
> 2010-02-23 10:34:26.269008 [DEBUG] mod_nibblebill.c:397 Attempting to bill
> at $2.30 per minute to account 3008
>
> 2010-02-23 10:34:26.269008 [INFO] mod_nibblebill.c:447 Beginning new
> billing on c1c42712-89e9-44c5-914f-11ed8f6aebb1
>
> 2010-02-23 10:34:26.269008 [DEBUG] mod_nibblebill.c:455 93 seconds passed
> since last bill time of 2010-02-23 10:32:52
>
> 2010-02-23 10:34:26.269008 [DEBUG] mod_nibblebill.c:461 Billing $3.582436
> to 3008 (Call: c1c42712-89e9-44c5-914f-11ed8f6aebb1 / 0.000000 so far)
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:46
> sofia/internal/sip:3008 at 192.168.15.176:21828 Standard HANGUP, cause:
> NORMAL_CLEARING
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:488
> (sofia/internal/sip:3008 at 192.168.15.176:21828) State HANGUP going to sleep
>
> 2010-02-23 10:34:26.269008 [INFO] switch_core_session.c:1108
> sofia/internal/sip:3008 at 192.168.15.176:21828 setting session heartbeat to
> 30 second(s).
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:351
> (sofia/internal/sip:3008 at 192.168.15.176:21828) State EXCHANGE_MEDIA going
> to sleep
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:314
> (sofia/internal/sip:3008 at 192.168.15.176:21828) Running State Change
> CS_HANGUP
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:465
> sofia/internal/sip:3008 at 192.168.15.176:21828 handler already called,
> skipping state handler.
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:333
> (sofia/internal/sip:3008 at 192.168.15.176:21828) State Change CS_HANGUP ->
> CS_REPORTING
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_session.c:999 Send signal
> sofia/internal/sip:3008 at 192.168.15.176:21828 [BREAK]
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:314
> (sofia/internal/sip:3008 at 192.168.15.176:21828) Running State Change
> CS_REPORTING
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:579
> (sofia/internal/sip:3008 at 192.168.15.176:21828) State REPORTING
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:53
> sofia/internal/sip:3008 at 192.168.15.176:21828 Standard REPORTING, cause:
> NORMAL_CLEARING
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:579
> (sofia/internal/sip:3008 at 192.168.15.176:21828) State REPORTING going to
> sleep
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:327
> (sofia/internal/sip:3008 at 192.168.15.176:21828) State Change CS_REPORTING
> -> CS_DESTROY
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_session.c:999 Send signal
> sofia/internal/sip:3008 at 192.168.15.176:21828 [BREAK]
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_session.c:1136 Session 6
> (sofia/internal/sip:3008 at 192.168.15.176:21828) Locked, Waiting on external
> entities
>
> 2010-02-23 10:34:26.272437 [DEBUG] switch_core_state_machine.c:348 (
> sofia/internal/3007 at 192.168.15.177) State EXECUTE going to sleep
>
> 2010-02-23 10:34:26.272437 [DEBUG] switch_core_state_machine.c:314 (
> sofia/internal/3007 at 192.168.15.177) Running State Change CS_HANGUP
>
> 2010-02-23 10:34:26.272437 [DEBUG] switch_core_state_machine.c:488 (
> sofia/internal/3007 at 192.168.15.177) State HANGUP
>
> 2010-02-23 10:34:26.272437 [DEBUG] mod_sofia.c:358 Channel
> sofia/internal/3007 at 192.168.15.177 hanging up, cause: NORMAL_CLEARING
>
> 2010-02-23 10:34:26.272437 [DEBUG] mod_nibblebill.c:397 Attempting to bill
> at $1.0 per minute to account 3007
>
> 2010-02-23 10:34:26.272437 [DEBUG] mod_nibblebill.c:449 Last successful
> billing time was
>
> 2010-02-23 10:34:26.272437 [DEBUG] mod_nibblebill.c:455 4 seconds passed
> since last bill time of 2010-02-23 10:34:21
>
> 2010-02-23 10:34:26.272437 [DEBUG] mod_nibblebill.c:461 Billing $0.077045
> to 3007 (Call: 3ec17ffc-c94e-46ff-bb0d-7ad0ef27bdab / 1.480343 so far)
>
> 2010-02-23 10:34:26.272437 [DEBUG] switch_core_state_machine.c:46
> sofia/internal/3007 at 192.168.15.177 Standard HANGUP, cause: NORMAL_CLEARING
>
>
>
> Anyhelp getting nibblebill to connect to the database would be greatly
> appreciated.  Thanks
>
>
>
> ------------------------------
> Hotmail: Trusted email with Microsoft?s powerful SPAM protection. Sign up
> now. <http://clk.atdmt.com/GBL/go/201469226/direct/01/>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100223/1e887229/attachment-0001.html 

From brian at microcomaustralia.com.au  Tue Feb 23 15:49:15 2010
From: brian at microcomaustralia.com.au (Brian May)
Date: Wed, 24 Feb 2010 10:49:15 +1100
Subject: [Freeswitch-users] Possible typo on Openzap.conf wiki page
	specifying fxs/fxo-channels
In-Reply-To: <2C33D9D6C1444AAC80950CDF8D38A428@greyhawk.tonecommander.com>
References: <2C33D9D6C1444AAC80950CDF8D38A428@greyhawk.tonecommander.com>
Message-ID: <3c5cf5261002231549s7e847c91l98529a95432b7175@mail.gmail.com>

On 24 February 2010 04:48, Robert Hadley <robert.hadley at teotech.com> wrote:
> [span zt FXS]
> name => OpenZAP
> number => 4165551111
>
> fxs-channel => 3
> number => 4165552222
> fxs-channel => 4

Two other details really confused me at first, and I don't think are
addressed in the documentation.

1. What is this "number" setting?  Some of the examples make it look
like it is a channel number:

=== cut ===
[span zt FXS1]
name => OpenZAP-FXS
number => 1
fxs-channel => 1

[span zt FXO1]
name => OpenZAP-FXO1
number => 2
fxo-channel => 3

[span zt FXO2]
name => OpenZAP-FXO2
number => 3
fxo-channel => 4
=== cut ===

It is not, it looks like on FXO ports it is the telephone number used
when looking up the dialplan for incoming calls; for FXS ports it is
the telephone number used for the callerid.


2. I have seen examples that use different formats for <span ...>:

<span id="FXO">
<span name="SPAN1" id="1">
<span name="PRI_1">
<span id="1">

Which syntax is correct? Which one should we be trying to use? If both
name= and  id= are specified, which one is used?
-- 
Brian May <brian at microcomaustralia.com.au>


From brian at microcomaustralia.com.au  Tue Feb 23 16:27:38 2010
From: brian at microcomaustralia.com.au (Brian May)
Date: Wed, 24 Feb 2010 11:27:38 +1100
Subject: [Freeswitch-users] internal/external profiles
Message-ID: <3c5cf5261002231627s1dfb04b2m64bf9bfeb3ff171d@mail.gmail.com>

Hello,

Why is it recommended to use separate profiles for internal and external SIP?

This page:
<http://wiki.freeswitch.org/wiki/External_profile>
suggests it is because of NAT.

However this page recommends using separate profiles even if NAT is
not an issue:
<http://wiki.freeswitch.org/wiki/Auto_NAT>: "NOTE: It is still
recommended that you use a second profile for your SIP providers. The
default conf/sip_profiles/external.xml is set up specifically for use
with providers."

However I am still left uncertain what this means.

Not trying to criticize here, just trying to learn.

Thanks.
-- 
Brian May <brian at microcomaustralia.com.au>


From msc at freeswitch.org  Tue Feb 23 16:47:18 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 23 Feb 2010 16:47:18 -0800
Subject: [Freeswitch-users] FreeSWITCH manual
In-Reply-To: <8E2F03C27AD4415BBDF1E9FF16F3DA98@MOBILEE1705>
References: <8E2F03C27AD4415BBDF1E9FF16F3DA98@MOBILEE1705>
Message-ID: <87f2f3b91002231647g4c216c9av8c9417c5ff37281f@mail.gmail.com>

On Tue, Feb 23, 2010 at 3:37 PM, Madovsky <infos at madovsky.org> wrote:

>  Hi dev friends !
>
> Is this manual yet available for the last trunk version ?
>
> http://www.scribd.com/doc/17425068/Free-Switch-in-Real-Life
>
> Thanks
>
> Franck
>
No, that is a very old document. There is, however, a FreeSWITCH book in the
works. It's almost drafted and still has to go through the editing process
before it will be published.
-MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100223/61657f5c/attachment.html 

From msc at freeswitch.org  Tue Feb 23 16:50:59 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 23 Feb 2010 16:50:59 -0800
Subject: [Freeswitch-users] REMINDER: FreeSWITCH Conf Call Moved to
	Wednesday!
Message-ID: <87f2f3b91002231650q6388c789v5f1287c7d2703204@mail.gmail.com>

Hi all,

Just a reminder that we are meeting up on Wednesay morning. Here's the
agenda:
http://wiki.freeswitch.org/wiki/FS_weekly_2010_02_24

It is light since we only had a few days since our meeting last Friday.
Remember that I won't be in right at the start of the meeting because I will
be taking my kids to school. Please feel free to use that time to mingle...

Thanks,
Michael
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100223/7ba1bb66/attachment.html 

From infos at madovsky.org  Tue Feb 23 17:04:42 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 23 Feb 2010 20:04:42 -0500
Subject: [Freeswitch-users] FS directories explaination
Message-ID: <4D4DE3DF3BD344809C43D68B3E410CF3@MOBILEE1705>

Hi,

Maybe a wiki page of directories description of conf directory would be great..

Thanks

Franck
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100223/893ff61f/attachment.html 

From brian at microcomaustralia.com.au  Tue Feb 23 17:17:24 2010
From: brian at microcomaustralia.com.au (Brian May)
Date: Wed, 24 Feb 2010 12:17:24 +1100
Subject: [Freeswitch-users] FS directories explaination
In-Reply-To: <4D4DE3DF3BD344809C43D68B3E410CF3@MOBILEE1705>
References: <4D4DE3DF3BD344809C43D68B3E410CF3@MOBILEE1705>
Message-ID: <3c5cf5261002231717k64418e34yf78fa24315652dbe@mail.gmail.com>

On 24 February 2010 12:04, Madovsky <infos at madovsky.org> wrote:
> Maybe a wiki page of directories description of conf
> directory would be great..

Not sure if I understand your question (do you want documentation on
/conf/directory/ or all of /conf/?), did you see this page?

http://wiki.freeswitch.org/wiki/XML_User_Directory_Guide
-- 
Brian May <brian at microcomaustralia.com.au>


From infos at madovsky.org  Tue Feb 23 18:44:08 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 23 Feb 2010 21:44:08 -0500
Subject: [Freeswitch-users] freeswitch minimum install
Message-ID: <219A4D36FE4A42A3B40A594F960461B9@MOBILEE1705>

Hi,

Is there a way to install freeswitch from source with the
strict minimum xml necassary in the conf dir to run freeswitch ?

Thanks

Franck
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100223/6546a80b/attachment.html 

From infos at madovsky.org  Tue Feb 23 18:53:45 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 23 Feb 2010 21:53:45 -0500
Subject: [Freeswitch-users] call from an internal extension to
	externalnumber
References: <D616CD3E2BFC4FDDA7ABEE7A3F268A4B@MOBILEE1705>
	<87f2f3b91002231347k6fd1e5b8wc6543ba2696b1629@mail.gmail.com>
Message-ID: <7AB1FDA0C14343DFB0FB30D32A0ED33C@MOBILEE1705>


    day after I undertand a littlee more all these xml hell files (not friendly to read ;)),
  Use a text editor that does syntax highlighting. :)
   

    but to be a PERl developer since 1999 understand regex and PERL language make life more easy...
  Hint: don't say PERL, say Perl or perl instead. People who say PERL are considered uneducated.
   

    mmhmm,  I said PERL as you say SIP, ti's an acronysm....
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100223/61d9573c/attachment.html 

From xanlich at gmail.com  Tue Feb 23 18:54:53 2010
From: xanlich at gmail.com (Chia-Yen Wu)
Date: Wed, 24 Feb 2010 10:54:53 +0800
Subject: [Freeswitch-users] Time condition in Lua
Message-ID: <314dc3f81002231854s7f20447brc48a1051bb83edbc@mail.gmail.com>

hello

is there anyway to do the time condition in lua script?
the only way i know is get the infomation by strftime() and compare it
but not all of them, like "*month of week" *doesnt support by strftime()
which dialplan XML does.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100224/8521bd6c/attachment.html 

From Russell.Mosemann at cune.org  Tue Feb 23 19:07:33 2010
From: Russell.Mosemann at cune.org (Russell Mosemann)
Date: Tue, 23 Feb 2010 21:07:33 -0600
Subject: [Freeswitch-users] call from an internal extension
	toexternalnumber
In-Reply-To: <7AB1FDA0C14343DFB0FB30D32A0ED33C@MOBILEE1705>
References: <D616CD3E2BFC4FDDA7ABEE7A3F268A4B@MOBILEE1705><87f2f3b91002231347k6fd1e5b8wc6543ba2696b1629@mail.gmail.com>
	<7AB1FDA0C14343DFB0FB30D32A0ED33C@MOBILEE1705>
Message-ID: <E5386A8959B24CE59F401F9ECEFF201C@cune.pri>

Madovsky said:

> 		mmhmm,  I said PERL as you say SIP, ti's an acronysm....

No, it is not an acronym. See the following.

http://en.wikipedia.org/wiki/Perl

" When referring to the language, the name is normally capitalized (Perl) as a proper noun, as you would a spoken language (e.g. English or French). When referring to the interpreter program itself, the name is often uncapitalized (perl) because most Unix-like file systems are case-sensitive. Before the release of the first edition of Programming Perl, it was common to refer to the language as perl; Randal L. Schwartz, however, capitalized the language's name in the book to make it stand out better when typeset. This case distinction was subsequently documented as canonical."

" There is some contention about the all-caps spelling "PERL," which the documentation declares incorrect and which some core community members consider a sign of outsiders. Although the name is occasionally taken as an acronym for Practical Extraction and Report Language (which appears at the top of the documentation and in some printed literature), this expansion actually came after the name; several others have been suggested as equally canonical, including Wall's own humorous Pathologically Eclectic Rubbish Lister. Indeed, Wall claims that the name was intended to inspire many different expansions."

--
Russell Mosemann



From infos at madovsky.org  Tue Feb 23 19:57:50 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 23 Feb 2010 22:57:50 -0500
Subject: [Freeswitch-users] call from an internal
	extensiontoexternalnumber
References: <D616CD3E2BFC4FDDA7ABEE7A3F268A4B@MOBILEE1705><87f2f3b91002231347k6fd1e5b8wc6543ba2696b1629@mail.gmail.com><7AB1FDA0C14343DFB0FB30D32A0ED33C@MOBILEE1705>
	<E5386A8959B24CE59F401F9ECEFF201C@cune.pri>
Message-ID: <E510B0579DFB435FB3C5E99FD3451AC1@MOBILEE1705>


----- Original Message ----- 
From: "Russell Mosemann" <Russell.Mosemann at cune.org>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Tuesday, February 23, 2010 10:07 PM
Subject: Re: [Freeswitch-users] call from an internal 
extensiontoexternalnumber


> Madovsky said:
>
>> mmhmm,  I said PERL as you say SIP, ti's an acronysm....
>
> No, it is not an acronym. See the following.
>
> http://en.wikipedia.org/wiki/Perl
>
> " When referring to the language, the name is normally capitalized (Perl) 
> as a proper noun, as you would a spoken language (e.g. English or French). 
> When referring to the interpreter program itself, the name is often 
> uncapitalized (perl) because most Unix-like file systems are 
> case-sensitive. Before the release of the first edition of Programming 
> Perl, it was common to refer to the language as perl; Randal L. Schwartz, 
> however, capitalized the language's name in the book to make it stand out 
> better when typeset. This case distinction was subsequently documented as 
> canonical."
>
> " There is some contention about the all-caps spelling "PERL," which the 
> documentation declares incorrect and which some core community members 
> consider a sign of outsiders. Although the name is occasionally taken as 
> an acronym for Practical Extraction and Report Language (which appears at 
> the top of the documentation and in some printed literature), this 
> expansion actually came after the name; several others have been suggested 
> as equally canonical, including Wall's own humorous Pathologically 
> Eclectic Rubbish Lister. Indeed, Wall claims that the name was intended to 
> inspire many different expansions."
>
> --
> Russell Mosemann
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

They Changed definition since they changed thei website...
old website it said
PERL - Practical Extraction Report Language

Thanks

Franck 



From gorand at donevtechconsulting.com  Tue Feb 23 20:23:58 2010
From: gorand at donevtechconsulting.com (Goran Donev)
Date: Tue, 23 Feb 2010 22:23:58 -0600
Subject: [Freeswitch-users] Are we close to final version 1.05
In-Reply-To: <mailman.6775.1266983891.2495.freeswitch-users@lists.freeswitch.org>
References: <mailman.6775.1266983891.2495.freeswitch-users@lists.freeswitch.org>
Message-ID: <12c101cab509$31893850$949ba8f0$@com>

Just checking if all you wonderful developers of this great project have an
ETA as to the final stable code of 1.05 to set into production environments.


It has been two weeks since the Feb 8th dinner and announcement of release
of 1.05. 

Thanks
Goran



From talk2ram at gmail.com  Tue Feb 23 20:34:02 2010
From: talk2ram at gmail.com (ram)
Date: Wed, 24 Feb 2010 10:04:02 +0530
Subject: [Freeswitch-users] Fwd: Freeswitch to another Freeswitch(or
	gateway)
In-Reply-To: <f8af5741002230600h6911e676iae6fc48375b31b1@mail.gmail.com>
References: <f8af5741002222202p3457e8efl2e683082dc597b47@mail.gmail.com>
	<f8af5741002230600h6911e676iae6fc48375b31b1@mail.gmail.com>
Message-ID: <b74751491002232034s7b32c9f1q96f1ade7c88ecbf8@mail.gmail.com>

On Tue, Feb 23, 2010 at 7:30 PM, srinivasula reddy <
srinivas.ksvreddy at gmail.com> wrote:

>
>  Hi,
> i want divert calls from my sipserver to another sipserver or third party
> gateway based on the host name, is there any way to achive this.
>

how about this examples

http://wiki.freeswitch.org/wiki/Dialplan_XML#Example_1

Ram
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100224/5b9aa9d7/attachment.html 

From troy at tlainvestments.com  Tue Feb 23 20:38:17 2010
From: troy at tlainvestments.com (Troy Anderson)
Date: Tue, 23 Feb 2010 21:38:17 -0700
Subject: [Freeswitch-users] Hook Flash
In-Reply-To: <87f2f3b91002231335o8c0b6f4vee98dbcd4d2b994d@mail.gmail.com>
References: <915F9D0D-FB13-4D02-995B-FB7F5EB488D2@tlainvestments.com>
	<742756.71167.qm@web33501.mail.mud.yahoo.com>
	<87f2f3b91002231335o8c0b6f4vee98dbcd4d2b994d@mail.gmail.com>
Message-ID: <84CAA6FA-F4C6-4BD5-83BB-9A19D782103F@tlainvestments.com>

Thanks, Diego!

-Troy


On Feb 23, 2010, at 2:35 PM, Michael Collins wrote:

> 
> 
> On Tue, Feb 23, 2010 at 5:33 AM, Diego Toro <dftoro at yahoo.com> wrote:
> hi, read http://jira.freeswitch.org/browse/OPENZAP-30
> 
>  <action application="send_dtmf" data="F"/>
> 
> Diego Toro
> http://lacarretade.blogspot.com/
> 
> FYI, I didn't see this on the wiki so I added it to the OpenZAP FAQ:
> http://wiki.freeswitch.org/wiki/OpenZAP#FAQ
> 
> Thanks Diego,
> MC
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100223/9accac2a/attachment.html 

From nagalenoj at gmail.com  Tue Feb 23 20:43:45 2010
From: nagalenoj at gmail.com (Nagalenoj H.)
Date: Wed, 24 Feb 2010 10:13:45 +0530
Subject: [Freeswitch-users] Error loading
	/usr/local/freeswitch/mod/ozmod_sangoma_boost.so
In-Reply-To: <87f2f3b91002231357t6c3e40cdld6a0a9861a9ec5a5@mail.gmail.com>
References: <fe67a3341002230202y1d45a8f3t3a2cda07de2240a2@mail.gmail.com>
	<87f2f3b91002231357t6c3e40cdld6a0a9861a9ec5a5@mail.gmail.com>
Message-ID: <fe67a3341002232043w23e5c04bh2598bdd9641d9c23@mail.gmail.com>

ozmod_sangoma_boost.so doesn't exist anywhere. It may not be a old version,
since I've checked out the source yesterday.

I've a doubt in the installation steps given. It is given to edit the
modules.conf after executing ./configure. Is it right? Do I need to edit the
modules.conf before ./configure??

On Wed, Feb 24, 2010 at 3:27 AM, Michael Collins <msc at freeswitch.org> wrote:

> Does the file ozmod_sangoma_boost.so exist somewhere in your openzap build
> tree? If not then something went wrong while compiling or you have an old
> version. If it does exist, do a quick test: cp the file into
> /usr/local/freeswitch/mod and restart FreeSWITCH and see if it finds the
> file and loads OpenZAP properly. Let us know the results so we can determine
> if it's a bug in the build system or not.
>
> -MC
>
> On Tue, Feb 23, 2010 at 2:02 AM, Nagalenoj H. <nagalenoj at gmail.com> wrote:
>
>> Dear friends,
>>     I've installed freeswitch trunk - 16729 and tried to configure with
>> wanpipe for sangoma A102 pri card.
>>
>> Followed the steps given in
>> http://wiki.sangoma.com/wanpipe-freeswitch-install
>>
>> When loading the freeswitch, I've got the following error.
>>
>> 2010-02-23 14:49:58.545726 [INFO] ozmod_wanpipe.c:335 configuring device
>> s2c15 as OpenZAP device 1:30 fd:57 DTMF: software
>> 2010-02-23 14:49:58.546004 [ERR] ozmod_wanpipe.c:436 No valid wanpipe span
>> and channel was specified
>> 2010-02-23 14:49:58.546275 [INFO] zap_io.c:2500 Configured 30 channel(s)
>> 2010-02-23 14:49:58.560724 [ERR] zap_io.c:2560 Error loading
>> /usr/local/freeswitch/mod/ozmod_sangoma_boost.so
>> [/usr/local/freeswitch/mod/ozmod_sangoma_boost.so: cannot open shared object
>> file: No such file or directory]
>> 2010-02-23 14:49:58.561048 [ERR] zap_io.c:2720 can't find 'sangoma_boost'
>> 2010-02-23 14:49:58.561266 [ERR] mod_openzap.c:2458 Error starting OpenZAP
>> span 1 error:
>> 2010-02-23 14:49:58.561712 [NOTICE] switch_loadable_module.c:144 Adding
>> Endpoint 'openzap'
>>
>> Configuration and log files are pasted to pastebin. Kindly someone help me
>> to solve this issue.
>>
>> openzap.conf and openzap.conf.xml
>>    http://pastebin.freeswitch.org/12214
>>
>> freeswitch log
>>    http://pastebin.freeswitch.org/12216
>>
>> smg_pri.conf
>>    http://pastebin.freeswitch.org/12217
>>
>> --
>> Regards,
>> Nagalenoj H.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Regards,
Nagalenoj H.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100224/1b8c3402/attachment.html 

From brian at freeswitch.org  Tue Feb 23 21:04:04 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 23 Feb 2010 23:04:04 -0600
Subject: [Freeswitch-users] Are we close to final version 1.05
In-Reply-To: <12c101cab509$31893850$949ba8f0$@com>
References: <mailman.6775.1266983891.2495.freeswitch-users@lists.freeswitch.org>
	<12c101cab509$31893850$949ba8f0$@com>
Message-ID: <3B5CB034-7DD5-406D-BC06-DA1B7A10F19F@freeswitch.org>

Its coming... we had a flood of issues we wanted to resolve before 1.0.5... I thank everyone that did donate for the dinner.

/b

On Feb 23, 2010, at 10:23 PM, Goran Donev wrote:

> It has been two weeks since the Feb 8th dinner and announcement of release
> of 1.05. 



From nazim.agabekov at gmail.com  Tue Feb 23 21:07:17 2010
From: nazim.agabekov at gmail.com (Nazim Agabekov)
Date: Wed, 24 Feb 2010 09:07:17 +0400
Subject: [Freeswitch-users] Time condition in Lua
In-Reply-To: <314dc3f81002231854s7f20447brc48a1051bb83edbc@mail.gmail.com>
References: <314dc3f81002231854s7f20447brc48a1051bb83edbc@mail.gmail.com>
Message-ID: <4B84B405.9020609@gmail.com>

Hello Wu

That's what I've found searching Lua Wiki:

http://lua-users.org/wiki/DayOfWeekAndDaysInMonthExample

There are a lot of helpful examples which could be used to create a
powerful calendar application.
Did you mean to say "week of month"?

Nazim.

On 02/24/2010 06:54 AM, Chia-Yen Wu wrote:
> hello
>
> is there anyway to do the time condition in lua script?
> the only way i know is get the infomation by strftime() and compare it
> but not all of them, like "*month of week" *doesnt support by
> strftime() which dialplan XML does.
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100224/34d81043/attachment.html 

From anthony.minessale at gmail.com  Tue Feb 23 21:36:51 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 23 Feb 2010 23:36:51 -0600
Subject: [Freeswitch-users] Are we close to final version 1.05
In-Reply-To: <191c3a031002232135s1e12d42evbbb87fd7ecfde505@mail.gmail.com>
References: <mailman.6775.1266983891.2495.freeswitch-users@lists.freeswitch.org>
	<12c101cab509$31893850$949ba8f0$@com>
	<3B5CB034-7DD5-406D-BC06-DA1B7A10F19F@freeswitch.org>
	<191c3a031002232135s1e12d42evbbb87fd7ecfde505@mail.gmail.com>
Message-ID: <191c3a031002232136t43fe155fs64eb5a4da4c69e4f@mail.gmail.com>

Just an fyi, there will probably be less bugs in trunk the day after we
release it than there was in that release.  =P

On Feb 23, 2010 11:10 PM, "Brian West" <brian at freeswitch.org> wrote:

Its coming... we had a flood of issues we wanted to resolve before 1.0.5...
I thank everyone that did donate for the dinner.

/b


On Feb 23, 2010, at 10:23 PM, Goran Donev wrote:

> It has been two weeks since the Feb 8th dinner ...

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists...
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100223/6fe457db/attachment-0001.html 

From dome at tel.co.th  Tue Feb 23 21:54:00 2010
From: dome at tel.co.th (Dome Charoenyost)
Date: Wed, 24 Feb 2010 12:54:00 +0700
Subject: [Freeswitch-users] Mod_h323 On openvz (64 bit)
Message-ID: <8ccbff061002232154u38bee079y3226a09d0f36bce6@mail.gmail.com>

Dear All,
I'm testing mod_h323 follow
http://wiki.freeswitch.org/wiki/Mod_h323

FS running on openvz 64 bit (kernel 2.6.18 pve 1.5 from proxmox)
After build mod_h323 sucessful  i found problem when load mod_h323 in FS CLI

2010-02-24 11:59:01.517104 [CRIT] switch_loadable_module.c:882 Error
Loading module /opt/freeswitch/mod/mod_h323.so
**/opt/freeswitch/mod/mod_h323.so: undefined symbol:
_ZN14H323Connection24OnUserInputInlineRFC2833ER15OpalRFC2833Infoi**

How to fix this problem ?


BG
Dome C.
2010/2/24 Chia-Yen Wu <xanlich at gmail.com>:
> hello
> is there anyway to do the time condition in lua script?
> the only way i know is get the infomation by?strftime() and compare it
> but not all of them, like "month of week"?doesnt support by strftime() which
> dialplan XML does.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From bekelemartins at gmail.com  Tue Feb 23 19:33:41 2010
From: bekelemartins at gmail.com (Bekele Martins)
Date: Tue, 23 Feb 2010 22:33:41 -0500
Subject: [Freeswitch-users] Streaming conference audio to a website
Message-ID: <5cc9e8f31002231933na50bd6o90e0c23a6667175a@mail.gmail.com>

Hello. I'm new to Freeswitch, but I have a question.
Is it possible to have a conference call and then stream the audio of the
conference to my website so people can listen to it over the Internet?
I read about mod_conference, but I couldn't find the answer to my question.
Thank you.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100223/eefb8a17/attachment.html 

From mike at jerris.com  Wed Feb 24 00:14:15 2010
From: mike at jerris.com (Michael Jerris)
Date: Wed, 24 Feb 2010 03:14:15 -0500
Subject: [Freeswitch-users] freeswitch minimum install
In-Reply-To: <219A4D36FE4A42A3B40A594F960461B9@MOBILEE1705>
References: <219A4D36FE4A42A3B40A594F960461B9@MOBILEE1705>
Message-ID: <8171AC69-88E7-4793-8C24-2594387C15E5@jerris.com>

no, other than manually creating that minimum conf

Mike
On Feb 23, 2010, at 9:44 PM, Madovsky wrote:

> Is there a way to install freeswitch from source with the
> strict minimum xml necassary in the conf dir to run FreeSWITCH ?

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100224/e368db19/attachment.html 

From mike at jerris.com  Wed Feb 24 00:23:54 2010
From: mike at jerris.com (Michael Jerris)
Date: Wed, 24 Feb 2010 03:23:54 -0500
Subject: [Freeswitch-users] Error loading
	/usr/local/freeswitch/mod/ozmod_sangoma_boost.so
In-Reply-To: <fe67a3341002232043w23e5c04bh2598bdd9641d9c23@mail.gmail.com>
References: <fe67a3341002230202y1d45a8f3t3a2cda07de2240a2@mail.gmail.com>
	<87f2f3b91002231357t6c3e40cdld6a0a9861a9ec5a5@mail.gmail.com>
	<fe67a3341002232043w23e5c04bh2598bdd9641d9c23@mail.gmail.com>
Message-ID: <D62532C0-53A1-4327-9BAA-DB2FB0C4BEF8@jerris.com>

you missed the second 1/2 of step 3 of Wanpipe TDM Installation


On Feb 23, 2010, at 11:43 PM, Nagalenoj H. wrote:

> ozmod_sangoma_boost.so doesn't exist anywhere. It may not be a old version, since I've checked out the source yesterday.
> 
> I've a doubt in the installation steps given. It is given to edit the modules.conf after executing ./configure. Is it right? Do I need to edit the modules.conf before ./configure??
> 
> On Wed, Feb 24, 2010 at 3:27 AM, Michael Collins <msc at freeswitch.org> wrote:
> Does the file ozmod_sangoma_boost.so exist somewhere in your openzap build tree? If not then something went wrong while compiling or you have an old version. If it does exist, do a quick test: cp the file into /usr/local/freeswitch/mod and restart FreeSWITCH and see if it finds the file and loads OpenZAP properly. Let us know the results so we can determine if it's a bug in the build system or not.
> 
> -MC
> 
> On Tue, Feb 23, 2010 at 2:02 AM, Nagalenoj H. <nagalenoj at gmail.com> wrote:
> Dear friends,
>     I've installed freeswitch trunk - 16729 and tried to configure with wanpipe for sangoma A102 pri card.
> 
> Followed the steps given in http://wiki.sangoma.com/wanpipe-freeswitch-install
> 
> When loading the freeswitch, I've got the following error.
> 
> 2010-02-23 14:49:58.545726 [INFO] ozmod_wanpipe.c:335 configuring device s2c15 as OpenZAP device 1:30 fd:57 DTMF: software
> 2010-02-23 14:49:58.546004 [ERR] ozmod_wanpipe.c:436 No valid wanpipe span and channel was specified
> 2010-02-23 14:49:58.546275 [INFO] zap_io.c:2500 Configured 30 channel(s)
> 2010-02-23 14:49:58.560724 [ERR] zap_io.c:2560 Error loading /usr/local/freeswitch/mod/ozmod_sangoma_boost.so [/usr/local/freeswitch/mod/ozmod_sangoma_boost.so: cannot open shared object file: No such file or directory]
> 2010-02-23 14:49:58.561048 [ERR] zap_io.c:2720 can't find 'sangoma_boost'
> 2010-02-23 14:49:58.561266 [ERR] mod_openzap.c:2458 Error starting OpenZAP span 1 error:
> 2010-02-23 14:49:58.561712 [NOTICE] switch_loadable_module.c:144 Adding Endpoint 'openzap'
> 
> Configuration and log files are pasted to pastebin. Kindly someone help me to solve this issue.
> 
> openzap.conf and openzap.conf.xml
>    http://pastebin.freeswitch.org/12214
> 
> freeswitch log
>    http://pastebin.freeswitch.org/12216
> 
> smg_pri.conf
>    http://pastebin.freeswitch.org/12217
> 
> -- 
> Regards,
> Nagalenoj H.
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> 
> -- 
> Regards,
> Nagalenoj H.
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100224/2d9ab798/attachment.html 

From mike at jerris.com  Wed Feb 24 00:28:25 2010
From: mike at jerris.com (Michael Jerris)
Date: Wed, 24 Feb 2010 03:28:25 -0500
Subject: [Freeswitch-users] Dial String Inject in FreeSwitch
In-Reply-To: <622bedea1002221113j5ac06477jb24ac51eedcd8d8f@mail.gmail.com>
References: <622bedea1002220945r5fd10a4cp19ed1244a67374d9@mail.gmail.com>
	<20100222182741.05F7829BF68@cuneorg-email.cune.pri>
	<622bedea1002221038u2a2e4232qd1d15cf006abb7db@mail.gmail.com>
	<622bedea1002221113j5ac06477jb24ac51eedcd8d8f@mail.gmail.com>
Message-ID: <FC38F850-32AF-4FBF-9BBE-626CBA2E48B3@jerris.com>

Why not actually edit the real wiki pages instead of hiding this information on your user page?

On Feb 22, 2010, at 2:13 PM, Eder Souza wrote:

> http://wiki.freeswitch.org/wiki/User:Ederwander
>  
> On Mon, Feb 22, 2010 at 3:38 PM, Eder Souza <ederwander at gmail.com> wrote:
> Perfect place lol
> 
>  
> On Mon, Feb 22, 2010 at 3:27 PM, <Russell.Mosemann at cune.org> wrote:
> > i thaks if somebody create one wiki witch this alert
> 
> A place to change is Example 1 of the dialplan XML examples. You can tell
> people not to use the catchall expressions, because you cannot trust
> information from the sender.
> 
> http://wiki.freeswitch.org/wiki/Dialplan_XML
> 
> A word of caution could also be added to
> 
> http://wiki.freeswitch.org/wiki/Regular_Expression
> 
> --
> Russell Mosemann
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100224/80680633/attachment.html 

From jason at jasonjgw.net  Wed Feb 24 00:31:36 2010
From: jason at jasonjgw.net (Jason White)
Date: Wed, 24 Feb 2010 19:31:36 +1100
Subject: [Freeswitch-users] Streaming conference audio to a website
In-Reply-To: <5cc9e8f31002231933na50bd6o90e0c23a6667175a@mail.gmail.com>
References: <5cc9e8f31002231933na50bd6o90e0c23a6667175a@mail.gmail.com>
Message-ID: <20100224083136.GA2549@jdc.jasonjgw.net>

Bekele Martins <bekelemartins at gmail.com> wrote:
> Is it possible to have a conference call and then stream the audio of the
> conference to my website so people can listen to it over the Internet?

You could turn on recording in the conference and have it written to a file
which is accessible to your Web server.
conference confname record filename



From mike at jerris.com  Wed Feb 24 00:32:15 2010
From: mike at jerris.com (Michael Jerris)
Date: Wed, 24 Feb 2010 03:32:15 -0500
Subject: [Freeswitch-users] SDP With T.38 in INVITE Problem
In-Reply-To: <b437a0e1002220849k16fd87ffl5a7257d80e43b036@mail.gmail.com>
References: <b437a0e1002190242y2aa64696j91e8bc76f7ae2a92@mail.gmail.com>
	<b437a0e1002220448g3e3bda9fpa283e93a11d319f1@mail.gmail.com>
	<24736FFE-BEEF-4D34-BBA7-77B6B6AE0EBD@jerris.com>
	<b437a0e1002220711u1bfd138l8a67b4e0b024d856@mail.gmail.com>
	<93769c21002220729q5bf36eadk80726aa8c00e3bd@mail.gmail.com>
	<b437a0e1002220849k16fd87ffl5a7257d80e43b036@mail.gmail.com>
Message-ID: <5ED9E41E-DFF1-4E03-B0F8-032309EA9A61@jerris.com>

if you want clarity on this, read the rfc for sdp offer answer.  You are not supposed to remove an m= line in an answer, if something is doing that, it is incorrect.

Mike

On Feb 22, 2010, at 11:49 AM, ivdreg ivdreg wrote:

> Hi Michael,
> 
> As I said in a previous mails I know exactly what is happening.
> In working setup:
> 
> ITSP ---> YATE (GW - Frontend) ---> FreeSwitch (routing server/xml_curl) ---> YATE (GW - SIP Interop) ---> OpenSIPS ---> Subscriber.
> 
> I prefer to change YATE (GW - Frontend) and YATE (GW - SIP Interop) with FreeSwitch for some reasons. The problem is:
> 
> INVITE comes from ITSP and goes to subscriber via OpenSIPS. INIVITE between FreeSwitch (routing server) and YATE (GW - SIP Interop) contains SDP: 
> m=audio 21108 RTP/AVP 18 4 8 0
> c=IN IP4 10.10.1.110
> a=rtpmap:18 G729/8000
> a=rtpmap:4 G723/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:0 PCMU/8000
> m=image 21108 udptl t38
> c=IN IP4 10.10.1.110
> a=T38FaxVersion:0
> a=T38MaxBitRate:14400
> a=T38FaxUdpEC:t38UDPRedundancy
> a=T38FaxRateManagement:
> transferredTCF
> 
> And reply 200 OK contains in SDP:
> m=audio 34788 RTP/AVP 8
> a=rtpmap:8 PCMA/8000
> a=silenceSupp:off - - - -
> a=ptime:20
> 
> Reply 200 OK SDP between YATE (GW - SIP Interop) and OpenSIPS contains in SDP:
> m=audio 16330 RTP/AVP 8
> a=rtpmap:8 PCMA/8000
> a=silenceSupp:off - - - -
> a=ptime:20
> m=image 0 udptl 19
> 
> In this case everything works fine. Line m=image 0 udptl 19 is removed by YATE. 
> But same test with FreeSWITCH on the place of YATE (GW - SIP Interop) "m=image 0 udptl 19" call brakes as you can see in my first mail.
> 
> I don't want to compare or discus YATE and FS functionality or something else. I just see difference in behavior and because I'm not a big expert don't know witch implementation is more accurate according standards. And second: Is it impossible for me to upgrade all CPE so only thing I can do is to fix it on server side. That is because I ask for a help.
> 
> 
> Thanks to all.
> 
> 
> 2010/2/22 Michael Jerris <mike at jerris.com>
> if you want to see what is going on, crank up the debug in freeswitch and sofia and you should see exactly what is going on.
> 
> Mike
> 
> 
> On Mon, Feb 22, 2010 at 10:11 AM, ivdreg ivdreg <ivdreg at gmail.com> wrote:
> Hi Michael,
> 
> This happens when ONLY IF initial INVITE is coming with T.38 from a GW (this is ITSP equipment and I don't know vendor) to our SIP subscribers with ATCOM ATA and IP Phone. We use now in production YATE for terminating and originating GWs to ITSPs and FS as main routing logic (backend). We want to switch YATE to FS for a GW also but we faced this problem. This not happens if initial INVITE nave no T.38 offered and later re-INVITE with T.38 with valid SDP port.
> 
> Thanks
> 
> 2010/2/22 Michael Jerris <mike at jerris.com>
> If the endpoint does not correctly follow the sdp o/a model its not going to work.  This is not a "problem" with the sofia library, this is intended behavior and what we are supposed to do.  What is the device?
> 
> Mike
> 
> On Feb 22, 2010, at 7:48 AM, ivdreg ivdreg wrote:
> 
>> Hi All,
>> 
>> Actually while seeking the solution in internet I see some people having this problem with sofia library. I'm not sure that SIP reply in this case contains a valid SDP (I think that teminating endpoint is broken) but in my opinion if we have at least one valid media type in SDP (video, audio, image ...) call must be established. Can someone comment and/or help me with this issue.
>> 
>> Regards.
>> 
>> 2010/2/19 ivdreg ivdreg <ivdreg at gmail.com>
>> Hi all,
>> 
>> Dose someone have a problem that if there T.38 in coming from gateway FreeSwitch drops the call because of media error ? As I see from log only T.38 port is zero and SDP has also media port. Is it possible to configure FS to do not break a call but if media is OK.
>> 
>> 2010-02-19 12:26:14.305244 [NOTICE] switch_channel.c:666 New Channel sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [6cd9f634-411d-df11-99ca-003048bb99cc]
>> 2010-02-19 12:26:14.305244 [DEBUG] mod_sofia.c:3369 (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State Change CS_NEW -> CS_INIT
>> 2010-02-19 12:26:14.305244 [DEBUG] switch_core_session.c:1018 Send signal sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [BREAK]
>> 2010-02-19 12:26:14.305244 [DEBUG] switch_core_state_machine.c:314 (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) Running State Change CS_INIT
>> 2010-02-19 12:26:14.305244 [DEBUG] switch_core_state_machine.c:338 (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State INIT
>> 2010-02-19 12:26:14.305244 [DEBUG] mod_sofia.c:83 sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 SOFIA INIT
>> 2010-02-19 12:26:14.305244 [DEBUG] sofia_glue.c:1377 sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 Patched SDP
>> ---
>> v=0
>> o=- 3779949069 5166785777234550622 IN IP4 206.132.232.212
>> s=session
>> t=0 0
>> m=audio 21108 RTP/AVP 18 4 8 0
>> c=IN IP4 10.10.1.110
>> a=rtpmap:18 G729/8000
>> a=rtpmap:4 G723/8000
>> a=rtpmap:8 PCMA/8000
>> a=rtpmap:0 PCMU/8000
>> m=image 21108 udptl t38
>> c=IN IP4 10.10.1.110
>> a=T38FaxVersion:0
>> a=T38MaxBitRate:14400
>> a=T38FaxUdpEC:t38UDPRedundancy
>> a=T38FaxRateManagement:transferredTCF
>> 
>> +++
>> v=0
>> o=- 3779949069 5166785777234550622 IN IP4 206.132.232.212
>> s=session
>> t=0 0
>> m=audio 17058 RTP/AVP 18 4 8 0
>> c=IN IP4 10.10.1.100
>> a=rtpmap:18 G729/8000
>> a=rtpmap:4 G723/8000
>> a=rtpmap:8 PCMA/8000
>> a=rtpmap:0 PCMU/8000
>> m=image 17058 udptl t38
>> c=IN IP4 10.10.1.100
>> a=T38FaxVersion:0
>> a=T38MaxBitRate:14400
>> a=T38FaxUdpEC:t38UDPRedundancy
>> a=T38FaxRateManagement:transferredTCF
>> 
>> 2010-02-19 12:26:14.305244 [DEBUG] mod_sofia.c:117 (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State Change CS_INIT -> CS_ROUTING
>> ......
>> 2010-02-19 12:26:21.255201 [DEBUG] sofia.c:4135 Remote SDP:
>> v=0
>> o=FreeSWITCH 1266548331 1266548332 IN IP4 10.10.1.110
>> s=FreeSWITCH
>> c=IN IP4 10.10.1.110
>> t=0 0
>> m=audio 26850 RTP/AVP 8
>> a=rtpmap:8 PCMA/8000
>> a=silenceSupp:off - - - -
>> a=ptime:20
>> m=image 0 udptl 19
>> 
>> 2010-02-19 12:26:21.255201 [DEBUG] sofia.c:4124 Channel sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 entering state [ready][200]
>> 2010-02-19 12:26:21.255201 [DEBUG] switch_channel.c:2285 Send signal sofia/backend/YYYYYYYYYY at 10.10.1.110 [BREAK]
>> 2010-02-19 12:26:21.255201 [NOTICE] sofia.c:4668 Channel [sofia/backend/XXXXXXXXXX at 10.10.1.110:7065] has been answered
>> 2010-02-19 12:26:21.255201 [DEBUG] sofia_glue.c:2288 Set Codec sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 PROXY/8000 20 ms 160 samples
>> 2010-02-19 12:26:21.255201 [DEBUG] sofia_glue.c:2571 PROXY AUDIO RTP [sofia/backend/XXXXXXXXXX at 10.10.1.110:7065] 10.10.1.100:17058->10.10.1.110:0 codec: 0 ms: 20
>> 2010-02-19 12:26:21.255201 [ERR] sofia_glue.c:2879 AUDIO RTP REPORTS ERROR: [Missing remote port]
>> 2010-02-19 12:26:21.255201 [NOTICE] sofia_glue.c:2880 Hangup sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [CS_CONSUME_MEDIA] [DESTINATION_OUT_OF_ORDER]
>> 2010-02-19 12:26:21.255201 [DEBUG] switch_channel.c:2063 Send signal sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [KILL]
>> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_session.c:1018 Send signal sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [BREAK]
>> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:314 (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) Running State Change CS_HANGUP
>> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:499 (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State HANGUP
>> 2010-02-19 12:26:21.255201 [DEBUG] mod_sofia.c:411 Channel sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 hanging up, cause: DESTINATION_OUT_OF_ORDER
>> 2010-02-19 12:26:21.255201 [DEBUG] mod_sofia.c:454 Sending BYE to sofia/backend/XXXXXXXXXX at 10.10.1.110:7065
>> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:46 sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 Standard HANGUP, cause: DESTINATION_OUT_OF_ORDER
>> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:499 (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State HANGUP going to sleep
>> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:333 (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State Change CS_HANGUP -> CS_REPORTING
>> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_session.c:1018 Send signal sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [BREAK]
>> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:314 (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) Running State Change CS_REPORTING
>> 2010-02-19 12:26:21.255201 [DEBUG] switch_ivr_originate.c:3185 Originate Resulted in Error Cause: 27 [DESTINATION_OUT_OF_ORDER]
>> 2010-02-19 12:26:21.255201 [INFO] mod_dptools.c:2353 Originate Failed.  Cause: DESTINATION_OUT_OF_ORDER
>> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:590 (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State REPORTING
>> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100224/df6bbcb5/attachment-0001.html 

From mike at jerris.com  Wed Feb 24 00:36:37 2010
From: mike at jerris.com (Michael Jerris)
Date: Wed, 24 Feb 2010 03:36:37 -0500
Subject: [Freeswitch-users] RTP timeout
In-Reply-To: <A68022CC2AFE40E29FE73385397A665B@MOBILEE1705>
References: <A68022CC2AFE40E29FE73385397A665B@MOBILEE1705>
Message-ID: <FD559959-D7F2-4BE7-B713-234478FFA72D@jerris.com>

If it hurts ? 

On Feb 23, 2010, at 1:09 PM, Madovsky wrote:

> Hi,
>  
> thanks to answer me if I misunderstood something,
> but if I run a softphone on the same IP as FS, there
> is an RTP timeout.
>  
> Any idea ?

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100224/12352730/attachment.html 

From mike at jerris.com  Wed Feb 24 00:40:32 2010
From: mike at jerris.com (Michael Jerris)
Date: Wed, 24 Feb 2010 03:40:32 -0500
Subject: [Freeswitch-users] FScomm
In-Reply-To: <11F8492DFC2546598E6FDC3E538301CD@MOBILEE1705>
References: <157FDB96FB88476FBAEC69C1309A0A5F@MOBILEE1705>,
	<COL117-W5563292C886257AFC56548B9420@phx.gbl>,
	<D64783281177440B9094B8181E55EFA7@MOBILEE1705>
	<COL117-W429F3BED9F781A58264238B9420@phx.gbl>
	<11F8492DFC2546598E6FDC3E538301CD@MOBILEE1705>
Message-ID: <BAA7D201-6E09-4EDE-BBA0-FDFA64E38391@jerris.com>


On Feb 23, 2010, at 6:18 PM, Madovsky wrote:

> 
> ----- Original Message ----- 
> From: Jeff Lenk
> To: freeswitch-users at lists.freeswitch.org
> Sent: Tuesday, February 23, 2010 5:05 PM
> Subject: Re: [Freeswitch-users] FScomm
> 
> 
> http://wiki.freeswitch.org/wiki/FSComm#Linux
> 
> you must run those from the FSComm directory
> 

?.

> 
> It's what I did,
> but from FS trunk, inside fscomm directory,
> there s only
> 
> account.cpp  conf                fshost.h        mainwindow.ui 
> resources.qrc
> account.h    FSComm.2008.vcproj  main.cpp        mod_qsettings
> call.cpp     FSComm.pro          mainwindow.cpp  preferences
> call.h       fshost.cpp          mainwindow.h    resources
> 


Read those installation instructions again and do them step by step, you skipped one.

From mike at jerris.com  Wed Feb 24 00:44:10 2010
From: mike at jerris.com (Michael Jerris)
Date: Wed, 24 Feb 2010 03:44:10 -0500
Subject: [Freeswitch-users] Streaming conference audio to a website
In-Reply-To: <5cc9e8f31002231933na50bd6o90e0c23a6667175a@mail.gmail.com>
References: <5cc9e8f31002231933na50bd6o90e0c23a6667175a@mail.gmail.com>
Message-ID: <F373C26A-1A75-45F0-9AFB-42269F64E9F0@jerris.com>

http://wiki.freeswitch.org/wiki/Mod_conference#.3Cprofiles.3E

see auto-record, note the example with mod_shout

http://wiki.freeswitch.org/wiki/Mod_shout

On Feb 23, 2010, at 10:33 PM, Bekele Martins wrote:

> Hello. I'm new to Freeswitch, but I have a question. 
> Is it possible to have a conference call and then stream the audio of the conference to my website so people can listen to it over the Internet?
> I read about mod_conference, but I couldn't find the answer to my question.

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100224/6568fcb7/attachment.html 

From xanlich at gmail.com  Wed Feb 24 01:14:45 2010
From: xanlich at gmail.com (Chia-Yen Wu)
Date: Wed, 24 Feb 2010 17:14:45 +0800
Subject: [Freeswitch-users] Time condition in Lua
In-Reply-To: <4B84B405.9020609@gmail.com>
References: <314dc3f81002231854s7f20447brc48a1051bb83edbc@mail.gmail.com>
	<4B84B405.9020609@gmail.com>
Message-ID: <314dc3f81002240114v30099971gdd64ecfc8d202c3a@mail.gmail.com>

yep , sorry i mean to say "week of month"

thank for help! that's what i needed!

2010/2/24 Nazim Agabekov <nazim.agabekov at gmail.com>

>  Hello Wu
>
> That's what I've found searching Lua Wiki:
>
> http://lua-users.org/wiki/DayOfWeekAndDaysInMonthExample
>
> There are a lot of helpful examples which could be used to create a
> powerful calendar application.
> Did you mean to say "week of month"?
>
> Nazim.
>
>
> On 02/24/2010 06:54 AM, Chia-Yen Wu wrote:
>
> hello
>
>  is there anyway to do the time condition in lua script?
> the only way i know is get the infomation by strftime() and compare it
> but not all of them, like "*month of week" *doesnt support by strftime()
> which dialplan XML does.
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100224/4d95aad9/attachment.html 

From devel at thom.fr.eu.org  Wed Feb 24 01:25:20 2010
From: devel at thom.fr.eu.org (=?UTF-8?Q?Fran=C3=A7ois_Legal?=)
Date: Wed, 24 Feb 2010 10:25:20 +0100
Subject: [Freeswitch-users] Possible typo on Openzap.conf wiki page
 specifying fxs/fxo-channels
In-Reply-To: <3c5cf5261002231524n4e9af307y80b3accb1ee4ff26@mail.gmail.com>
References: <2C33D9D6C1444AAC80950CDF8D38A428@greyhawk.tonecommander.com>
	<3c5cf5261002231524n4e9af307y80b3accb1ee4ff26@mail.gmail.com>
Message-ID: <a689f57fe216d671d7d1ff132c6b636e@thom.fr.eu.org>

Well, I am the guy who made the modification in the wiki.

I use sangoma card and the openzap file is generated by the Setup script
from sangoma driver.
It seems that the terminology used by zaptel is not used in wanpipe
configuration.

I have an A400 card with an FXO module (providing ports 11 and 12) and an
FXS module (providing ports 9 and 10)

My openzap.conf is like this :

[span wanpipe FXS]
name => Analog phone 1
number => 9000
fxs-channel => 1:9
name => Analog phone 2
number => 9001
fxs-channel => 1:10

[span wanpipe FXO]
name => POTS line 1
number => 1234567890
fxo-channel => 1:11
name => POTS line 2
number => 0987654321
fxo-channel => 1:12

About the name and number, this is what I get here by observation :
If I make a call from channel 1:9 and the diaplan do not modify the CID
variables, the called person see "Analog Phone 1" as CID name and 9000 as
CID number.
On the other hand, if I receive a call on channel 1:11, the "channel_name"
variable raised in the diaplan would be openzap/2/1/1234567890 (I guess
here that if number is not specified, I would get openzap/2/1). Moreover
(still this is by observation) if the carrier does not send CID (to be
precise, I mean no modulation is received) the CID name and number raised
in diaplan on incoming calls are set to name and number from openzap.conf

I hope this is clear enought

Fran?ois

On Wed, 24 Feb 2010 10:24:31 +1100, Brian May
<brian at microcomaustralia.com.au> wrote:
> On 24 February 2010 04:48, Robert Hadley <robert.hadley at teotech.com>
wrote:
>> On the http://wiki.freeswitch.org/wiki/Openzap.conf_Examples wiki page,
>> is
>> there a typo in the wanpipe /usr/local/freeswitch/conf/openzap.conf
>> example
>> concerning specifying the fxo-channel vs. fxs-channel?
> 
> I agree. The first example looks correct to me; the 2nd example looks
> wrong.
> 
> See the table I created to try and explain what term to use where:
> 
> http://wiki.freeswitch.org/wiki/OpenZAP#FXO.2FFXS_Terminology
> 
> In the examples you quoted, ports 1 and 2 are extension ports, so are
> FXS ports, but should be defined as FXO ports in openzap.conf.
> 
> Ports 3 and 4 are telephone line ports, so are FXO ports, but should
> be defined as FXS ports in openzap.conf.
> 
> I have only used zaptel myself, however I suspect the same applies to
> wanpipe.


From tculjaga at gmail.com  Wed Feb 24 01:53:49 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Wed, 24 Feb 2010 10:53:49 +0100
Subject: [Freeswitch-users] Mod_h323 On openvz (64 bit)
In-Reply-To: <8ccbff061002232154u38bee079y3226a09d0f36bce6@mail.gmail.com>
References: <8ccbff061002232154u38bee079y3226a09d0f36bce6@mail.gmail.com>
Message-ID: <65d96fc81002240153h213371dbpec400d651c6550@mail.gmail.com>

On Wed, Feb 24, 2010 at 6:54 AM, Dome Charoenyost <dome at tel.co.th> wrote:

> Dear All,
> I'm testing mod_h323 follow
> http://wiki.freeswitch.org/wiki/Mod_h323
>
> FS running on openvz 64 bit (kernel 2.6.18 pve 1.5 from proxmox)
> After build mod_h323 sucessful  i found problem when load mod_h323 in FS
> CLI
>
> 2010-02-24 11:59:01.517104 [CRIT] switch_loadable_module.c:882 Error
> Loading module /opt/freeswitch/mod/mod_h323.so
> **/opt/freeswitch/mod/mod_h323.so: undefined symbol:
> _ZN14H323Connection24OnUserInputInlineRFC2833ER15OpalRFC2833Infoi**
>
> How to fix this problem ?
>


can you start FS in debug mode (set debug level into switch.conf.xml)
without mod_h323 automatic load. Than, on console run "load mod_h323"
command and paste the output here ... just want to be sure of something.

If it is a larger log ... use pastebin with the url reference here.

T.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100224/3780d703/attachment.html 

From phunk0000 at hotmail.com  Wed Feb 24 05:03:42 2010
From: phunk0000 at hotmail.com (Todd)
Date: Wed, 24 Feb 2010 08:03:42 -0500
Subject: [Freeswitch-users] mod_nibblebill
In-Reply-To: <b39273531002231537k626d1698vd88549a7f59e73a@mail.gmail.com>
References: <BLU110-W28EA8BD06A601D5D85268BA7420@phx.gbl>
	<b39273531002231537k626d1698vd88549a7f59e73a@mail.gmail.com>
Message-ID: <BLU110-DS68941CBE9213A6DA338ACA7410@phx.gbl>

I am attempting to us MySQL.  I installed the spidermonkey mod, newest ODBC,
compiled FS with ODBC, configured xml's in FS.  not 100% sure I did it right
though..followed wiki directions as close as possible.  What is the best way
to verify the SQL is talking to MySQL. or perhaps the easiest way to switch
to postgresql?  Still kinda new to DB admin.  Thanks a ton.

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rupa
Schomaker
Sent: Tuesday, February 23, 2010 6:38 PM
To: freeswitch-users
Subject: Re: [Freeswitch-users] mod_nibblebill

 

what database backend are you using?  Have you verified the SQL is going to
the right database backend?  I use mod_nibblebill against postgresql w/out
problems.

On Tue, Feb 23, 2010 at 9:42 AM, Meg Stroodle <phunk0000 at hotmail.com> wrote:

Hello List!  I am trying to install mod_nibblebill on my FS installation.  I
get the following log entry form FS & nibblebill, but the database table I
setup remains unchanged.  Any help in this matter would be greatly
appreciated.  Following is an excerpt from the FS log:

 

2010-02-23 10:32:52.869149 [DEBUG] switch_rtp.c:2004 Correct ip/port
confirmed.

2010-02-23 10:32:53.077162 [DEBUG] sofia.c:3727 Channel
sofia/internal/3007 at 192.168.15.177 entering state [ready][200]

2010-02-23 10:32:53.149166 [DEBUG] switch_core_session.c:706 Send signal
sofia/internal/sip:3008 at 192.168.15.176:21828 [BREAK]

2010-02-23 10:32:53.149166 [DEBUG] switch_core_session.c:706 Send signal
sofia/internal/3007 at 192.168.15.177 [BREAK]

2010-02-23 10:32:53.149166 [DEBUG] switch_ivr_bridge.c:131
sofia/internal/sip:3008 at 192.168.15.176:21828 receive message [DISPLAY]

2010-02-23 10:32:53.169168 [DEBUG] switch_ivr_bridge.c:131
sofia/internal/3007 at 192.168.15.177 receive message [DISPLAY]

2010-02-23 10:33:21.388949 [DEBUG] mod_nibblebill.c:529 Received request via
SESSION_HEARTBEAT!

2010-02-23 10:33:21.388949 [DEBUG] mod_nibblebill.c:397 Attempting to bill
at $1.0 per minute to account 3007

2010-02-23 10:33:21.388949 [INFO] mod_nibblebill.c:447 Beginning new billing
on 3ec17ffc-c94e-46ff-bb0d-7ad0ef27bdab

2010-02-23 10:33:21.388949 [DEBUG] mod_nibblebill.c:455 28 seconds passed
since last bill time of 2010-02-23 10:32:52

2010-02-23 10:33:21.388949 [DEBUG] mod_nibblebill.c:461 Billing $0.475997 to
3007 (Call: 3ec17ffc-c94e-46ff-bb0d-7ad0ef27bdab / 0.000000 so far)

2010-02-23 10:33:51.508822 [DEBUG] mod_nibblebill.c:529 Received request via
SESSION_HEARTBEAT!

2010-02-23 10:33:51.508822 [DEBUG] mod_nibblebill.c:397 Attempting to bill
at $1.0 per minute to account 3007

2010-02-23 10:33:51.508822 [DEBUG] mod_nibblebill.c:449 Last successful
billing time was 

2010-02-23 10:33:51.508822 [DEBUG] mod_nibblebill.c:455 30 seconds passed
since last bill time of 2010-02-23 10:33:21

2010-02-23 10:33:51.508822 [DEBUG] mod_nibblebill.c:461 Billing $0.501998 to
3007 (Call: 3ec17ffc-c94e-46ff-bb0d-7ad0ef27bdab / 0.475997 so far)

2010-02-23 10:34:21.649708 [DEBUG] mod_nibblebill.c:529 Received request via
SESSION_HEARTBEAT!

2010-02-23 10:34:21.649708 [DEBUG] mod_nibblebill.c:397 Attempting to bill
at $1.0 per minute to account 3007

2010-02-23 10:34:21.649708 [DEBUG] mod_nibblebill.c:449 Last successful
billing time was 

2010-02-23 10:34:21.649708 [DEBUG] mod_nibblebill.c:455 30 seconds passed
since last bill time of 2010-02-23 10:33:51

2010-02-23 10:34:21.649708 [DEBUG] mod_nibblebill.c:461 Billing $0.502348 to
3007 (Call: 3ec17ffc-c94e-46ff-bb0d-7ad0ef27bdab / 0.977995 so far)

2010-02-23 10:34:26.254308 [NOTICE] sofia.c:329 Hangup
sofia/internal/3007 at 192.168.15.177 [CS_EXECUTE] [NORMAL_CLEARING]

2010-02-23 10:34:26.254308 [DEBUG] switch_channel.c:1912 Send signal
sofia/internal/3007 at 192.168.15.177 [KILL]

2010-02-23 10:34:26.254308 [DEBUG] switch_core_session.c:999 Send signal
sofia/internal/3007 at 192.168.15.177 [BREAK]

2010-02-23 10:34:26.254308 [DEBUG] switch_core_state_machine.c:459
sofia/internal/3007 at 192.168.15.177 thread mismatch skipping state handler.

2010-02-23 10:34:26.269008 [DEBUG] switch_ivr_bridge.c:470
sofia/internal/3007 at 192.168.15.177 ending bridge by request from read
function

2010-02-23 10:34:26.269008 [DEBUG] switch_ivr_bridge.c:563 BRIDGE THREAD
DONE [sofia/internal/3007 at 192.168.15.177]

2010-02-23 10:34:26.269008 [DEBUG] switch_ivr_bridge.c:565 Send signal
sofia/internal/sip:3008 at 192.168.15.176:21828 [BREAK]

2010-02-23 10:34:26.269008 [DEBUG] switch_ivr_bridge.c:520
sofia/internal/sip:3008 at 192.168.15.176:21828 receive message [UNBRIDGE]

2010-02-23 10:34:26.269008 [DEBUG] switch_core_session.c:645 Send signal
sofia/internal/sip:3008 at 192.168.15.176:21828 [BREAK]

2010-02-23 10:34:26.269008 [DEBUG] switch_ivr_bridge.c:563 BRIDGE THREAD
DONE [sofia/internal/sip:3008 at 192.168.15.176:21828]

2010-02-23 10:34:26.269008 [DEBUG] switch_ivr_bridge.c:565 Send signal
sofia/internal/3007 at 192.168.15.177 [BREAK]

2010-02-23 10:34:26.269008 [NOTICE] switch_ivr_bridge.c:617 Hangup
sofia/internal/sip:3008 at 192.168.15.176:21828 [CS_EXCHANGE_MEDIA]
[NORMAL_CLEARING]

2010-02-23 10:34:26.269008 [DEBUG] switch_channel.c:1912 Send signal
sofia/internal/sip:3008 at 192.168.15.176:21828 [KILL]

2010-02-23 10:34:26.269008 [DEBUG] switch_core_session.c:999 Send signal
sofia/internal/sip:3008 at 192.168.15.176:21828 [BREAK]

2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:488
(sofia/internal/sip:3008 at 192.168.15.176:21828) State HANGUP

2010-02-23 10:34:26.269008 [DEBUG] mod_sofia.c:358 Channel
sofia/internal/sip:3008 at 192.168.15.176:21828 hanging up, cause:
NORMAL_CLEARING

2010-02-23 10:34:26.269008 [DEBUG] mod_sofia.c:400 Sending BYE to
sofia/internal/sip:3008 at 192.168.15.176:21828

2010-02-23 10:34:26.269008 [DEBUG] mod_nibblebill.c:397 Attempting to bill
at $2.30 per minute to account 3008

2010-02-23 10:34:26.269008 [INFO] mod_nibblebill.c:447 Beginning new billing
on c1c42712-89e9-44c5-914f-11ed8f6aebb1

2010-02-23 10:34:26.269008 [DEBUG] mod_nibblebill.c:455 93 seconds passed
since last bill time of 2010-02-23 10:32:52

2010-02-23 10:34:26.269008 [DEBUG] mod_nibblebill.c:461 Billing $3.582436 to
3008 (Call: c1c42712-89e9-44c5-914f-11ed8f6aebb1 / 0.000000 so far)

2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:46
sofia/internal/sip:3008 at 192.168.15.176:21828 Standard HANGUP, cause:
NORMAL_CLEARING

2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:488
(sofia/internal/sip:3008 at 192.168.15.176:21828) State HANGUP going to sleep

2010-02-23 10:34:26.269008 [INFO] switch_core_session.c:1108
sofia/internal/sip:3008 at 192.168.15.176:21828 setting session heartbeat to 30
second(s).

2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:351
(sofia/internal/sip:3008 at 192.168.15.176:21828) State EXCHANGE_MEDIA going to
sleep

2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/sip:3008 at 192.168.15.176:21828) Running State Change
CS_HANGUP

2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:465
sofia/internal/sip:3008 at 192.168.15.176:21828 handler already called,
skipping state handler.

2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:333
(sofia/internal/sip:3008 at 192.168.15.176:21828) State Change CS_HANGUP ->
CS_REPORTING

2010-02-23 10:34:26.269008 [DEBUG] switch_core_session.c:999 Send signal
sofia/internal/sip:3008 at 192.168.15.176:21828 [BREAK]

2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/sip:3008 at 192.168.15.176:21828) Running State Change
CS_REPORTING

2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:579
(sofia/internal/sip:3008 at 192.168.15.176:21828) State REPORTING

2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:53
sofia/internal/sip:3008 at 192.168.15.176:21828 Standard REPORTING, cause:
NORMAL_CLEARING

2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:579
(sofia/internal/sip:3008 at 192.168.15.176:21828) State REPORTING going to
sleep

2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:327
(sofia/internal/sip:3008 at 192.168.15.176:21828) State Change CS_REPORTING ->
CS_DESTROY

2010-02-23 10:34:26.269008 [DEBUG] switch_core_session.c:999 Send signal
sofia/internal/sip:3008 at 192.168.15.176:21828 [BREAK]

2010-02-23 10:34:26.269008 [DEBUG] switch_core_session.c:1136 Session 6
(sofia/internal/sip:3008 at 192.168.15.176:21828) Locked, Waiting on external
entities

2010-02-23 10:34:26.272437 [DEBUG] switch_core_state_machine.c:348
(sofia/internal/3007 at 192.168.15.177) State EXECUTE going to sleep

2010-02-23 10:34:26.272437 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/3007 at 192.168.15.177) Running State Change CS_HANGUP

2010-02-23 10:34:26.272437 [DEBUG] switch_core_state_machine.c:488
(sofia/internal/3007 at 192.168.15.177) State HANGUP

2010-02-23 10:34:26.272437 [DEBUG] mod_sofia.c:358 Channel
sofia/internal/3007 at 192.168.15.177 hanging up, cause: NORMAL_CLEARING

2010-02-23 10:34:26.272437 [DEBUG] mod_nibblebill.c:397 Attempting to bill
at $1.0 per minute to account 3007

2010-02-23 10:34:26.272437 [DEBUG] mod_nibblebill.c:449 Last successful
billing time was 

2010-02-23 10:34:26.272437 [DEBUG] mod_nibblebill.c:455 4 seconds passed
since last bill time of 2010-02-23 10:34:21

2010-02-23 10:34:26.272437 [DEBUG] mod_nibblebill.c:461 Billing $0.077045 to
3007 (Call: 3ec17ffc-c94e-46ff-bb0d-7ad0ef27bdab / 1.480343 so far)

2010-02-23 10:34:26.272437 [DEBUG] switch_core_state_machine.c:46
sofia/internal/3007 at 192.168.15.177 Standard HANGUP, cause: NORMAL_CLEARING

 

Anyhelp getting nibblebill to connect to the database would be greatly
appreciated.  Thanks

 

 

  _____  

Hotmail: Trusted email with Microsoft's powerful SPAM protection. Sign up
now. <http://clk.atdmt.com/GBL/go/201469226/direct/01/> 


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




-- 
-Rupa

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100224/d3687737/attachment-0001.html 

From mayamatakeshi at gmail.com  Wed Feb 24 05:12:33 2010
From: mayamatakeshi at gmail.com (mayamatakeshi)
Date: Wed, 24 Feb 2010 22:12:33 +0900
Subject: [Freeswitch-users] Setting username in the header To
Message-ID: <15b9404e1002240512y47beb0afha555c24e24d9cc2d@mail.gmail.com>

Hello,
while doing a bridge or originate,
is it possible to send a username in the header To that is different from
the one in the Request-URI?
This is to interoperate with a GW that understands this as a request for
redirection (it will send a call to the PSTN with a parameter ISUP
RedirectingNumber).

br,
Takeshi
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100224/3a668e82/attachment.html 

From bekelemartins at gmail.com  Wed Feb 24 05:47:03 2010
From: bekelemartins at gmail.com (Bekele Martins)
Date: Wed, 24 Feb 2010 08:47:03 -0500
Subject: [Freeswitch-users] Streaming conference audio to a website
In-Reply-To: <F373C26A-1A75-45F0-9AFB-42269F64E9F0@jerris.com>
References: <5cc9e8f31002231933na50bd6o90e0c23a6667175a@mail.gmail.com>
	<F373C26A-1A75-45F0-9AFB-42269F64E9F0@jerris.com>
Message-ID: <5cc9e8f31002240547x6c3eab8lcf53486572b16d9d@mail.gmail.com>

I'm sorry, I meant how can I stream it live, so if someone joins the stream
from the website in the middle of the conference they will not hear the
beginning of the conversation, they will hear what's being said in real
time. Is this possible?


On Wed, Feb 24, 2010 at 3:44 AM, Michael Jerris <mike at jerris.com> wrote:

> http://wiki.freeswitch.org/wiki/Mod_conference#.3Cprofiles.3E
>
> see auto-record, note the example with mod_shout
>
> http://wiki.freeswitch.org/wiki/Mod_shout
>
> On Feb 23, 2010, at 10:33 PM, Bekele Martins wrote:
>
> Hello. I'm new to Freeswitch, but I have a question.
> Is it possible to have a conference call and then stream the audio of the
> conference to my website so people can listen to it over the Internet?
> I read about mod_conference, but I couldn't find the answer to my question.
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100224/a26ace5b/attachment.html 

From max.bridgewater at gmail.com  Wed Feb 24 05:53:12 2010
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Wed, 24 Feb 2010 05:53:12 -0800
Subject: [Freeswitch-users] Increasing call Volume
Message-ID: <c55d46951002240553k9bf14bak1b684096e2a09fff@mail.gmail.com>

Hi,

Is there a way to increase the call volume with FS? I'm getting a call
from Portech but with an echo. their suggestion to resolve the echo
issue is to reduce the RX Gain. But then Rx Gain  also impacts the
volume in the call received from portech. So I was wondering if there
could be way to "correct" the stream at FS level.

Thanks,
Max.


From rupa at rupa.com  Wed Feb 24 06:03:39 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Wed, 24 Feb 2010 08:03:39 -0600
Subject: [Freeswitch-users] Streaming conference audio to a website
In-Reply-To: <5cc9e8f31002240547x6c3eab8lcf53486572b16d9d@mail.gmail.com>
References: <5cc9e8f31002231933na50bd6o90e0c23a6667175a@mail.gmail.com>
	<F373C26A-1A75-45F0-9AFB-42269F64E9F0@jerris.com>
	<5cc9e8f31002240547x6c3eab8lcf53486572b16d9d@mail.gmail.com>
Message-ID: <b39273531002240603y4440307bv78ff5e49d93fdac0@mail.gmail.com>

That is exactly what Mike's suggestion would do -- live streaming.

On Wed, Feb 24, 2010 at 7:47 AM, Bekele Martins <bekelemartins at gmail.com>wrote:

> I'm sorry, I meant how can I stream it live, so if someone joins the stream
> from the website in the middle of the conference they will not hear the
> beginning of the conversation, they will hear what's being said in real
> time. Is this possible?
>
>
> On Wed, Feb 24, 2010 at 3:44 AM, Michael Jerris <mike at jerris.com> wrote:
>
>> http://wiki.freeswitch.org/wiki/Mod_conference#.3Cprofiles.3E
>>
>> see auto-record, note the example with mod_shout
>>
>> http://wiki.freeswitch.org/wiki/Mod_shout
>>
>> On Feb 23, 2010, at 10:33 PM, Bekele Martins wrote:
>>
>> Hello. I'm new to Freeswitch, but I have a question.
>> Is it possible to have a conference call and then stream the audio of the
>> conference to my website so people can listen to it over the Internet?
>> I read about mod_conference, but I couldn't find the answer to my
>> question.
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100224/6e237f42/attachment.html 

From rupa at rupa.com  Wed Feb 24 06:06:24 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Wed, 24 Feb 2010 08:06:24 -0600
Subject: [Freeswitch-users] mod_nibblebill
In-Reply-To: <BLU110-DS68941CBE9213A6DA338ACA7410@phx.gbl>
References: <BLU110-W28EA8BD06A601D5D85268BA7420@phx.gbl>
	<b39273531002231537k626d1698vd88549a7f59e73a@mail.gmail.com>
	<BLU110-DS68941CBE9213A6DA338ACA7410@phx.gbl>
Message-ID: <b39273531002240606p5bc03f34vbc8e934fedfe65a3@mail.gmail.com>

I am only passingly familiar with MySQL.   There must be a way for it to log
all sql statements sent to it?

Setting up postgresql would be the same (in broad terms) as mysql.  Install
packages, create database/user/tables, populate data, configure odbc dsn,
test.

On Wed, Feb 24, 2010 at 7:03 AM, Todd <phunk0000 at hotmail.com> wrote:

>  I am attempting to us MySQL.  I installed the spidermonkey mod, newest
> ODBC, compiled FS with ODBC, configured xml?s in FS?  not 100% sure I did it
> right though..followed wiki directions as close as possible.  What is the
> best way to verify the SQL is talking to MySQL? or perhaps the easiest way
> to switch  to postgresql?  Still kinda new to DB admin.  Thanks a ton.
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Rupa
> Schomaker
> *Sent:* Tuesday, February 23, 2010 6:38 PM
> *To:* freeswitch-users
> *Subject:* Re: [Freeswitch-users] mod_nibblebill
>
>
>
> what database backend are you using?  Have you verified the SQL is going to
> the right database backend?  I use mod_nibblebill against postgresql w/out
> problems.
>
> On Tue, Feb 23, 2010 at 9:42 AM, Meg Stroodle <phunk0000 at hotmail.com>
> wrote:
>
> Hello List!  I am trying to install mod_nibblebill on my FS installation.
> I get the following log entry form FS & nibblebill, but the database table I
> setup remains unchanged.  Any help in this matter would be greatly
> appreciated.  Following is an excerpt from the FS log:
>
>
>
> 2010-02-23 10:32:52.869149 [DEBUG] switch_rtp.c:2004 Correct ip/port
> confirmed.
>
> 2010-02-23 10:32:53.077162 [DEBUG] sofia.c:3727 Channel
> sofia/internal/3007 at 192.168.15.177 entering state [ready][200]
>
> 2010-02-23 10:32:53.149166 [DEBUG] switch_core_session.c:706 Send signal
> sofia/internal/sip:3008 at 192.168.15.176:21828 [BREAK]
>
> 2010-02-23 10:32:53.149166 [DEBUG] switch_core_session.c:706 Send signal
> sofia/internal/3007 at 192.168.15.177 [BREAK]
>
> 2010-02-23 10:32:53.149166 [DEBUG] switch_ivr_bridge.c:131 sofia/internal/
> sip:3008 at 192.168.15.176:21828 receive message [DISPLAY]
>
> 2010-02-23 10:32:53.169168 [DEBUG] switch_ivr_bridge.c:131
> sofia/internal/3007 at 192.168.15.177 receive message [DISPLAY]
>
> 2010-02-23 10:33:21.388949 [DEBUG] mod_nibblebill.c:529 Received request
> via SESSION_HEARTBEAT!
>
> 2010-02-23 10:33:21.388949 [DEBUG] mod_nibblebill.c:397 Attempting to bill
> at $1.0 per minute to account 3007
>
> 2010-02-23 10:33:21.388949 [INFO] mod_nibblebill.c:447 Beginning new
> billing on 3ec17ffc-c94e-46ff-bb0d-7ad0ef27bdab
>
> 2010-02-23 10:33:21.388949 [DEBUG] mod_nibblebill.c:455 28 seconds passed
> since last bill time of 2010-02-23 10:32:52
>
> 2010-02-23 10:33:21.388949 [DEBUG] mod_nibblebill.c:461 Billing $0.475997
> to 3007 (Call: 3ec17ffc-c94e-46ff-bb0d-7ad0ef27bdab / 0.000000 so far)
>
> 2010-02-23 10:33:51.508822 [DEBUG] mod_nibblebill.c:529 Received request
> via SESSION_HEARTBEAT!
>
> 2010-02-23 10:33:51.508822 [DEBUG] mod_nibblebill.c:397 Attempting to bill
> at $1.0 per minute to account 3007
>
> 2010-02-23 10:33:51.508822 [DEBUG] mod_nibblebill.c:449 Last successful
> billing time was
>
> 2010-02-23 10:33:51.508822 [DEBUG] mod_nibblebill.c:455 30 seconds passed
> since last bill time of 2010-02-23 10:33:21
>
> 2010-02-23 10:33:51.508822 [DEBUG] mod_nibblebill.c:461 Billing $0.501998
> to 3007 (Call: 3ec17ffc-c94e-46ff-bb0d-7ad0ef27bdab / 0.475997 so far)
>
> 2010-02-23 10:34:21.649708 [DEBUG] mod_nibblebill.c:529 Received request
> via SESSION_HEARTBEAT!
>
> 2010-02-23 10:34:21.649708 [DEBUG] mod_nibblebill.c:397 Attempting to bill
> at $1.0 per minute to account 3007
>
> 2010-02-23 10:34:21.649708 [DEBUG] mod_nibblebill.c:449 Last successful
> billing time was
>
> 2010-02-23 10:34:21.649708 [DEBUG] mod_nibblebill.c:455 30 seconds passed
> since last bill time of 2010-02-23 10:33:51
>
> 2010-02-23 10:34:21.649708 [DEBUG] mod_nibblebill.c:461 Billing $0.502348
> to 3007 (Call: 3ec17ffc-c94e-46ff-bb0d-7ad0ef27bdab / 0.977995 so far)
>
> 2010-02-23 10:34:26.254308 [NOTICE] sofia.c:329 Hangup
> sofia/internal/3007 at 192.168.15.177 [CS_EXECUTE] [NORMAL_CLEARING]
>
> 2010-02-23 10:34:26.254308 [DEBUG] switch_channel.c:1912 Send signal
> sofia/internal/3007 at 192.168.15.177 [KILL]
>
> 2010-02-23 10:34:26.254308 [DEBUG] switch_core_session.c:999 Send signal
> sofia/internal/3007 at 192.168.15.177 [BREAK]
>
> 2010-02-23 10:34:26.254308 [DEBUG] switch_core_state_machine.c:459
> sofia/internal/3007 at 192.168.15.177 thread mismatch skipping state handler.
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_ivr_bridge.c:470
> sofia/internal/3007 at 192.168.15.177 ending bridge by request from read
> function
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_ivr_bridge.c:563 BRIDGE THREAD
> DONE [sofia/internal/3007 at 192.168.15.177]
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_ivr_bridge.c:565 Send signal
> sofia/internal/sip:3008 at 192.168.15.176:21828 [BREAK]
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_ivr_bridge.c:520 sofia/internal/
> sip:3008 at 192.168.15.176:21828 receive message [UNBRIDGE]
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_session.c:645 Send signal
> sofia/internal/sip:3008 at 192.168.15.176:21828 [BREAK]
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_ivr_bridge.c:563 BRIDGE THREAD
> DONE [sofia/internal/sip:3008 at 192.168.15.176:21828]
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_ivr_bridge.c:565 Send signal
> sofia/internal/3007 at 192.168.15.177 [BREAK]
>
> 2010-02-23 10:34:26.269008 [NOTICE] switch_ivr_bridge.c:617 Hangup
> sofia/internal/sip:3008 at 192.168.15.176:21828 [CS_EXCHANGE_MEDIA]
> [NORMAL_CLEARING]
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_channel.c:1912 Send signal
> sofia/internal/sip:3008 at 192.168.15.176:21828 [KILL]
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_session.c:999 Send signal
> sofia/internal/sip:3008 at 192.168.15.176:21828 [BREAK]
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:488
> (sofia/internal/sip:3008 at 192.168.15.176:21828) State HANGUP
>
> 2010-02-23 10:34:26.269008 [DEBUG] mod_sofia.c:358 Channel sofia/internal/
> sip:3008 at 192.168.15.176:21828 hanging up, cause: NORMAL_CLEARING
>
> 2010-02-23 10:34:26.269008 [DEBUG] mod_sofia.c:400 Sending BYE to
> sofia/internal/sip:3008 at 192.168.15.176:21828
>
> 2010-02-23 10:34:26.269008 [DEBUG] mod_nibblebill.c:397 Attempting to bill
> at $2.30 per minute to account 3008
>
> 2010-02-23 10:34:26.269008 [INFO] mod_nibblebill.c:447 Beginning new
> billing on c1c42712-89e9-44c5-914f-11ed8f6aebb1
>
> 2010-02-23 10:34:26.269008 [DEBUG] mod_nibblebill.c:455 93 seconds passed
> since last bill time of 2010-02-23 10:32:52
>
> 2010-02-23 10:34:26.269008 [DEBUG] mod_nibblebill.c:461 Billing $3.582436
> to 3008 (Call: c1c42712-89e9-44c5-914f-11ed8f6aebb1 / 0.000000 so far)
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:46
> sofia/internal/sip:3008 at 192.168.15.176:21828 Standard HANGUP, cause:
> NORMAL_CLEARING
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:488
> (sofia/internal/sip:3008 at 192.168.15.176:21828) State HANGUP going to sleep
>
> 2010-02-23 10:34:26.269008 [INFO] switch_core_session.c:1108
> sofia/internal/sip:3008 at 192.168.15.176:21828 setting session heartbeat to
> 30 second(s).
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:351
> (sofia/internal/sip:3008 at 192.168.15.176:21828) State EXCHANGE_MEDIA going
> to sleep
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:314
> (sofia/internal/sip:3008 at 192.168.15.176:21828) Running State Change
> CS_HANGUP
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:465
> sofia/internal/sip:3008 at 192.168.15.176:21828 handler already called,
> skipping state handler.
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:333
> (sofia/internal/sip:3008 at 192.168.15.176:21828) State Change CS_HANGUP ->
> CS_REPORTING
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_session.c:999 Send signal
> sofia/internal/sip:3008 at 192.168.15.176:21828 [BREAK]
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:314
> (sofia/internal/sip:3008 at 192.168.15.176:21828) Running State Change
> CS_REPORTING
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:579
> (sofia/internal/sip:3008 at 192.168.15.176:21828) State REPORTING
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:53
> sofia/internal/sip:3008 at 192.168.15.176:21828 Standard REPORTING, cause:
> NORMAL_CLEARING
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:579
> (sofia/internal/sip:3008 at 192.168.15.176:21828) State REPORTING going to
> sleep
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:327
> (sofia/internal/sip:3008 at 192.168.15.176:21828) State Change CS_REPORTING
> -> CS_DESTROY
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_session.c:999 Send signal
> sofia/internal/sip:3008 at 192.168.15.176:21828 [BREAK]
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_session.c:1136 Session 6
> (sofia/internal/sip:3008 at 192.168.15.176:21828) Locked, Waiting on external
> entities
>
> 2010-02-23 10:34:26.272437 [DEBUG] switch_core_state_machine.c:348 (
> sofia/internal/3007 at 192.168.15.177) State EXECUTE going to sleep
>
> 2010-02-23 10:34:26.272437 [DEBUG] switch_core_state_machine.c:314 (
> sofia/internal/3007 at 192.168.15.177) Running State Change CS_HANGUP
>
> 2010-02-23 10:34:26.272437 [DEBUG] switch_core_state_machine.c:488 (
> sofia/internal/3007 at 192.168.15.177) State HANGUP
>
> 2010-02-23 10:34:26.272437 [DEBUG] mod_sofia.c:358 Channel
> sofia/internal/3007 at 192.168.15.177 hanging up, cause: NORMAL_CLEARING
>
> 2010-02-23 10:34:26.272437 [DEBUG] mod_nibblebill.c:397 Attempting to bill
> at $1.0 per minute to account 3007
>
> 2010-02-23 10:34:26.272437 [DEBUG] mod_nibblebill.c:449 Last successful
> billing time was
>
> 2010-02-23 10:34:26.272437 [DEBUG] mod_nibblebill.c:455 4 seconds passed
> since last bill time of 2010-02-23 10:34:21
>
> 2010-02-23 10:34:26.272437 [DEBUG] mod_nibblebill.c:461 Billing $0.077045
> to 3007 (Call: 3ec17ffc-c94e-46ff-bb0d-7ad0ef27bdab / 1.480343 so far)
>
> 2010-02-23 10:34:26.272437 [DEBUG] switch_core_state_machine.c:46
> sofia/internal/3007 at 192.168.15.177 Standard HANGUP, cause: NORMAL_CLEARING
>
>
>
> Anyhelp getting nibblebill to connect to the database would be greatly
> appreciated.  Thanks
>
>
>
>
>  ------------------------------
>
> Hotmail: Trusted email with Microsoft?s powerful SPAM protection. Sign up
> now. <http://clk.atdmt.com/GBL/go/201469226/direct/01/>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100224/ab8b4cca/attachment-0001.html 

From jeff at jefflenk.com  Wed Feb 24 06:07:19 2010
From: jeff at jefflenk.com (Jeff Lenk)
Date: Wed, 24 Feb 2010 08:07:19 -0600
Subject: [Freeswitch-users] Increasing call Volume
In-Reply-To: <c55d46951002240553k9bf14bak1b684096e2a09fff@mail.gmail.com>
References: <c55d46951002240553k9bf14bak1b684096e2a09fff@mail.gmail.com>
Message-ID: <COL117-W7D89C1D1F7F15CDFBF1D3B9410@phx.gbl>


You can use: this is not the recommended way to solve this problem!

 

http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_set_audio_level

 

 

 


 
> Date: Wed, 24 Feb 2010 05:53:12 -0800
> From: max.bridgewater at gmail.com
> To: freeswitch-users at lists.freeswitch.org
> Subject: [Freeswitch-users] Increasing call Volume
> 
> Hi,
> 
> Is there a way to increase the call volume with FS? I'm getting a call
> from Portech but with an echo. their suggestion to resolve the echo
> issue is to reduce the RX Gain. But then Rx Gain also impacts the
> volume in the call received from portech. So I was wondering if there
> could be way to "correct" the stream at FS level.
> 
> Thanks,
> Max.
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
 		 	   		  
_________________________________________________________________
Hotmail: Free, trusted and rich email service.
http://clk.atdmt.com/GBL/go/201469228/direct/01/
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100224/6130ec74/attachment.html 

From andrew at hijacked.us  Wed Feb 24 06:19:55 2010
From: andrew at hijacked.us (Andrew Thompson)
Date: Wed, 24 Feb 2010 09:19:55 -0500
Subject: [Freeswitch-users] Increasing call Volume
In-Reply-To: <COL117-W7D89C1D1F7F15CDFBF1D3B9410@phx.gbl>
References: <c55d46951002240553k9bf14bak1b684096e2a09fff@mail.gmail.com>
	<COL117-W7D89C1D1F7F15CDFBF1D3B9410@phx.gbl>
Message-ID: <20100224141955.GE1751@hijacked.us>

On Wed, Feb 24, 2010 at 08:07:19AM -0600, Jeff Lenk wrote:
> 
> You can use: this is not the recommended way to solve this problem!
> 
>  
> 
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_set_audio_level
>

There's also the uuid_audio API command (again, maybe not the best tool
to solve the problem).

Andrew


From max.bridgewater at gmail.com  Wed Feb 24 06:20:11 2010
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Wed, 24 Feb 2010 06:20:11 -0800
Subject: [Freeswitch-users] Increasing call Volume
In-Reply-To: <COL117-W7D89C1D1F7F15CDFBF1D3B9410@phx.gbl>
References: <c55d46951002240553k9bf14bak1b684096e2a09fff@mail.gmail.com>
	<COL117-W7D89C1D1F7F15CDFBF1D3B9410@phx.gbl>
Message-ID: <c55d46951002240620s7d95789bidc564488d580ec0e@mail.gmail.com>

Thanks jeff. so what would be the recommended way for solving this problem?

Max.

On Wed, Feb 24, 2010 at 6:07 AM, Jeff Lenk <jeff at jefflenk.com> wrote:
> You can use: this is not the recommended way to solve this problem!
>
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_set_audio_level
>
>
>
>
>
>> Date: Wed, 24 Feb 2010 05:53:12 -0800
>> From: max.bridgewater at gmail.com
>> To: freeswitch-users at lists.freeswitch.org
>> Subject: [Freeswitch-users] Increasing call Volume
>>
>> Hi,
>>
>> Is there a way to increase the call volume with FS? I'm getting a call
>> from Portech but with an echo. their suggestion to resolve the echo
>> issue is to reduce the RX Gain. But then Rx Gain also impacts the
>> volume in the call received from portech. So I was wondering if there
>> could be way to "correct" the stream at FS level.
>>
>> Thanks,
>> Max.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> ________________________________
> Hotmail: Free, trusted and rich email service. Get it now.
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From matt at webcontracts.co.uk  Wed Feb 24 06:20:29 2010
From: matt at webcontracts.co.uk (Matthew Law)
Date: Wed, 24 Feb 2010 14:20:29 -0000
Subject: [Freeswitch-users] How to debug time-based routing?
In-Reply-To: <74B270D5-E134-4221-A0FE-8275B05826A5@freeswitch.org>
References: <eb2506f03e9045c509a6974bcc4a3a49.squirrel@www.webcontracts.co.uk>
	<74B270D5-E134-4221-A0FE-8275B05826A5@freeswitch.org>
Message-ID: <a797462218a492c72e11a575534c18e0.squirrel@www.webcontracts.co.uk>


On Tue, February 23, 2010 1:00 am, Brian West wrote:
> Lets start with how about you pastebin your extension and logs... or
> better join #freeswitch on irc.freenode.net?  ;)
>
> /b

Thanks, Brian.

It turns out that Zoiper, the softphone I was using on my Mac seems to be
a little hit and miss.  If I switch to windows and use X-lite or the 3CX
softphone it works perfectly.

Matt.



From Suneel.Papineni at mettoni.com  Wed Feb 24 06:35:49 2010
From: Suneel.Papineni at mettoni.com (Suneel Papineni)
Date: Wed, 24 Feb 2010 14:35:49 -0000
Subject: [Freeswitch-users] Issue with making calls through FSComm and Build
	issues with Freeswitch 1.0.5 latest updated on 24-Feb-2010 04:05
Message-ID: <3181A30B8C35AB4AA8577B78DDF4613806810AEF@nickel.mettonigroup.com>

Hi,

 

I am trying to make FSComm work with Freeswitch (1.0.5 latest), but
failed to make any calls. FSComm is getting registered properly and I
can see SIP messages at Freeswitch. When a call is tried to make from
one FSComm to another, there is no INVITE message seen in Freeswitch and
also in wireshark traces as well. (OS environment is Windows XP 32-bit
). Could someone let me know if there is any specific config changes to
be made to FSComm to work. (I am trying initially with FSComm pre-build
binary version. Results are same).

 

 

As per previous suggestions I tried to work with the latest version,
downloaded Freeswitch 1.0.5 Latest updated on 24th February 2010 at
4:00am from "http://latest.freeswitch.org/". When I tried to build this,
it is giving 4 errors. So build is failing. Errors are as follows:

 

Error      201         error C2491: 'spandsp_stop_inband_dtmf_session' :
definition of dllimport function not allowed
d:\FS\freeswitch-1.0.5-latest24022010\freeswitch-1.0.5-20100224-0400\src
\mod\applications\mod_fax\mod_fax.c             812         mod_fax

Error      202         error C2491: 'spandsp_inband_dtmf_session' :
definition of dllimport function not allowed
d:\FS\freeswitch-1.0.5-latest24022010\freeswitch-1.0.5-20100224-0400\src
\mod\applications\mod_fax\mod_fax.c             825         mod_fax

 

 

Error      213         error LNK2019: unresolved external symbol
_sip_dig_function referenced in function _mod_sofia_load
mod_sofia.obj  mod_sofia

Error      214         fatal error LNK1120: 1 unresolved externals
D:\FS\freeswitch-1.0.5-latest24022010\freeswitch-1.0.5-20100224-0400\Deb
ug\mod\mod_sofia.dll         mod_sofia

 

I am trying to build Freeswitch for a windows 32-bit system. Tried to
build both Debug and Release versions but failed with the above errors.

 

Could someone let me know from where I need to download the latest
version (possibly without errors).

 

Thanks & Regards

Suneel


*************************************************************************
Please consider the environment before printing this e-mail
*************************************************************************
This email and any files transmitted with it are confidential and
intended solely for the use of the individual or entity to whom they
are addressed. If you have received this email in error please notify
the system manager.  http://www.mettoni.com

Mettoni Ltd
Registered in England and Wales: 4485956
9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN
*************************************************************************

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100224/e62f441c/attachment.html 

From ivdreg at gmail.com  Wed Feb 24 06:52:21 2010
From: ivdreg at gmail.com (ivdreg ivdreg)
Date: Wed, 24 Feb 2010 16:52:21 +0200
Subject: [Freeswitch-users] SDP With T.38 in INVITE Problem
In-Reply-To: <5ED9E41E-DFF1-4E03-B0F8-032309EA9A61@jerris.com>
References: <b437a0e1002190242y2aa64696j91e8bc76f7ae2a92@mail.gmail.com>
	<b437a0e1002220448g3e3bda9fpa283e93a11d319f1@mail.gmail.com>
	<24736FFE-BEEF-4D34-BBA7-77B6B6AE0EBD@jerris.com>
	<b437a0e1002220711u1bfd138l8a67b4e0b024d856@mail.gmail.com>
	<93769c21002220729q5bf36eadk80726aa8c00e3bd@mail.gmail.com>
	<b437a0e1002220849k16fd87ffl5a7257d80e43b036@mail.gmail.com>
	<5ED9E41E-DFF1-4E03-B0F8-032309EA9A61@jerris.com>
Message-ID: <b437a0e1002240652m70497461g165eb8e7d4a47bca@mail.gmail.com>

Hi Michael,

Thanks for clarifying. Unfortunately we don't live in prefect world. I was
fixed that by disabling T.38 in codec negotiation and everything works fine.

Thanks Again.

2010/2/24 Michael Jerris <mike at jerris.com>

> if you want clarity on this, read the rfc for sdp offer answer.  You are
> not supposed to remove an m= line in an answer, if something is doing that,
> it is incorrect.
>
> Mike
>
> On Feb 22, 2010, at 11:49 AM, ivdreg ivdreg wrote:
>
> Hi Michael,
>
> As I said in a previous mails I know exactly what is happening.
> In working setup:
>
> ITSP ---> YATE (GW - Frontend) ---> FreeSwitch (routing server/xml_curl)
> ---> YATE (GW - SIP Interop) ---> OpenSIPS ---> Subscriber.
>
> I prefer to change YATE (GW - Frontend) and YATE (GW - SIP Interop) with
> FreeSwitch for some reasons. The problem is:
>
> INVITE comes from ITSP and goes to subscriber via OpenSIPS. INIVITE between
> FreeSwitch (routing server) and YATE (GW - SIP Interop) contains SDP:
> m=audio 21108 RTP/AVP 18 4 8 0
> c=IN IP4 10.10.1.110
> a=rtpmap:18 G729/8000
> a=rtpmap:4 G723/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:0 PCMU/8000
> m=image 21108 udptl t38
> c=IN IP4 10.10.1.110
> a=T38FaxVersion:0
> a=T38MaxBitRate:14400
> a=T38FaxUdpEC:t38UDPRedundancy
> a=T38FaxRateManagement:
> transferredTCF
>
> And reply 200 OK contains in SDP:
> *m=audio 34788 RTP/AVP 8*
> a=rtpmap:8 PCMA/8000
> a=silenceSupp:off - - - -
> a=ptime:20
>
> Reply 200 OK SDP between YATE (GW - SIP Interop) and OpenSIPS contains in
> SDP:
> *m=audio 16330 RTP/AVP 8*
> a=rtpmap:8 PCMA/8000
> a=silenceSupp:off - - - -
> a=ptime:20
> *m=image 0 udptl 19*
>
> In this case everything works fine. Line *m=image 0 udptl 19 *is removed
> by YATE.
> But same test with FreeSWITCH on the place of YATE (GW - SIP Interop) *"m=image
> 0 udptl 19" *call brakes as you can see in my first mail.
>
> I don't want to compare or discus YATE and FS functionality or something
> else. I just see difference in behavior and because I'm not a big expert
> don't know witch implementation is more accurate according standards. And
> second: Is it impossible for me to upgrade all CPE so only thing I can do is
> to fix it on server side. That is because I ask for a help.
>
>
> Thanks to all.
>
>
> 2010/2/22 Michael Jerris <mike at jerris.com>
>
>> if you want to see what is going on, crank up the debug in freeswitch and
>> sofia and you should see exactly what is going on.
>>
>> Mike
>>
>>
>> On Mon, Feb 22, 2010 at 10:11 AM, ivdreg ivdreg <ivdreg at gmail.com> wrote:
>>
>>> Hi Michael,
>>>
>>> This happens when ONLY IF initial INVITE is coming with T.38 from a GW
>>> (this is ITSP equipment and I don't know vendor) to our SIP subscribers with
>>> ATCOM ATA and IP Phone. We use now in production YATE for terminating and
>>> originating GWs to ITSPs and FS as main routing logic (backend). We want to
>>> switch YATE to FS for a GW also but we faced this problem. This not happens
>>> if initial INVITE nave no T.38 offered and later re-INVITE with T.38 with
>>> valid SDP port.
>>>
>>> Thanks
>>>
>>> 2010/2/22 Michael Jerris <mike at jerris.com>
>>>
>>>>  If the endpoint does not correctly follow the sdp o/a model its not
>>>> going to work.  This is not a "problem" with the sofia library, this is
>>>> intended behavior and what we are supposed to do.  What is the device?
>>>>
>>>> Mike
>>>>
>>>> On Feb 22, 2010, at 7:48 AM, ivdreg ivdreg wrote:
>>>>
>>>> Hi All,
>>>>
>>>> Actually while seeking the solution in internet I see some people having
>>>> this problem with sofia library. I'm not sure that SIP reply in this case
>>>> contains a valid SDP (I think that teminating endpoint is broken) but in my
>>>> opinion if we have at least one valid media type in SDP (video, audio, image
>>>> ...) call must be established. Can someone comment and/or help me with this
>>>> issue.
>>>>
>>>> Regards.
>>>>
>>>> 2010/2/19 ivdreg ivdreg <ivdreg at gmail.com>
>>>>
>>>>> Hi all,
>>>>>
>>>>> Dose someone have a problem that if there T.38 in coming from gateway
>>>>> FreeSwitch drops the call because of media error ? As I see from log only
>>>>> T.38 port is zero and SDP has also media port. Is it possible to configure
>>>>> FS to do not break a call but if media is OK.
>>>>>
>>>>> 2010-02-19 12:26:14.305244 [NOTICE] switch_channel.c:666 New Channel
>>>>> sofia/backend/XXXXXXXXXX at 10.10.1.110:7065[6cd9f634-411d-df11-99ca-003048bb99cc]
>>>>> 2010-02-19 12:26:14.305244 [DEBUG] mod_sofia.c:3369 (sofia/backend/
>>>>> XXXXXXXXXX at 10.10.1.110:7065) State Change CS_NEW -> CS_INIT
>>>>> 2010-02-19 12:26:14.305244 [DEBUG] switch_core_session.c:1018 Send
>>>>> signal sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [BREAK]
>>>>> 2010-02-19 12:26:14.305244 [DEBUG] switch_core_state_machine.c:314
>>>>> (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) Running State Change
>>>>> CS_INIT
>>>>> 2010-02-19 12:26:14.305244 [DEBUG] switch_core_state_machine.c:338
>>>>> (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State INIT
>>>>> 2010-02-19 12:26:14.305244 [DEBUG] mod_sofia.c:83 sofia/backend/
>>>>> XXXXXXXXXX at 10.10.1.110:7065 SOFIA INIT
>>>>> 2010-02-19 12:26:14.305244 [DEBUG] sofia_glue.c:1377 sofia/backend/
>>>>> XXXXXXXXXX at 10.10.1.110:7065 Patched SDP
>>>>> ---
>>>>> v=0
>>>>> o=- 3779949069 5166785777234550622 IN IP4 206.132.232.212
>>>>> s=session
>>>>> t=0 0
>>>>> m=audio 21108 RTP/AVP 18 4 8 0
>>>>> c=IN IP4 10.10.1.110
>>>>> a=rtpmap:18 G729/8000
>>>>> a=rtpmap:4 G723/8000
>>>>> a=rtpmap:8 PCMA/8000
>>>>> a=rtpmap:0 PCMU/8000
>>>>> m=image 21108 udptl t38
>>>>> c=IN IP4 10.10.1.110
>>>>> a=T38FaxVersion:0
>>>>> a=T38MaxBitRate:14400
>>>>> a=T38FaxUdpEC:t38UDPRedundancy
>>>>> a=T38FaxRateManagement:transferredTCF
>>>>>
>>>>> +++
>>>>> v=0
>>>>> o=- 3779949069 5166785777234550622 IN IP4 206.132.232.212
>>>>> s=session
>>>>> t=0 0
>>>>> m=audio 17058 RTP/AVP 18 4 8 0
>>>>> c=IN IP4 10.10.1.100
>>>>> a=rtpmap:18 G729/8000
>>>>> a=rtpmap:4 G723/8000
>>>>> a=rtpmap:8 PCMA/8000
>>>>> a=rtpmap:0 PCMU/8000
>>>>> m=image 17058 udptl t38
>>>>> c=IN IP4 10.10.1.100
>>>>> a=T38FaxVersion:0
>>>>> a=T38MaxBitRate:14400
>>>>> a=T38FaxUdpEC:t38UDPRedundancy
>>>>> a=T38FaxRateManagement:transferredTCF
>>>>>
>>>>> 2010-02-19 12:26:14.305244 [DEBUG] mod_sofia.c:117 (sofia/backend/
>>>>> XXXXXXXXXX at 10.10.1.110:7065) State Change CS_INIT -> CS_ROUTING
>>>>> ......
>>>>> 2010-02-19 12:26:21.255201 [DEBUG] sofia.c:4135 Remote SDP:
>>>>> v=0
>>>>> o=FreeSWITCH 1266548331 1266548332 IN IP4 10.10.1.110
>>>>> s=FreeSWITCH
>>>>> c=IN IP4 10.10.1.110
>>>>> t=0 0
>>>>> *m=audio 26850 RTP/AVP 8*
>>>>> a=rtpmap:8 PCMA/8000
>>>>> a=silenceSupp:off - - - -
>>>>> a=ptime:20
>>>>> *m=image 0 udptl 19*
>>>>>
>>>>> 2010-02-19 12:26:21.255201 [DEBUG] sofia.c:4124 Channel sofia/backend/
>>>>> XXXXXXXXXX at 10.10.1.110:7065 entering state [ready][200]
>>>>> 2010-02-19 12:26:21.255201 [DEBUG] switch_channel.c:2285 Send signal
>>>>> sofia/backend/YYYYYYYYYY at 10.10.1.110 [BREAK]
>>>>> 2010-02-19 12:26:21.255201 [NOTICE] sofia.c:4668 Channel
>>>>> [sofia/backend/XXXXXXXXXX at 10.10.1.110:7065] has been answered
>>>>> 2010-02-19 12:26:21.255201 [DEBUG] sofia_glue.c:2288 Set Codec
>>>>> sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 PROXY/8000 20 ms 160 samples
>>>>> *2010-02-19 12:26:21.255201 [DEBUG] sofia_glue.c:2571 PROXY AUDIO RTP
>>>>> [sofia/backend/XXXXXXXXXX at 10.10.1.110:7065] 10.10.1.100:17058->
>>>>> 10.10.1.110:0 codec: 0 ms: 20
>>>>> 2010-02-19 12:26:21.255201 [ERR] sofia_glue.c:2879 AUDIO RTP REPORTS
>>>>> ERROR: [Missing remote port]
>>>>> 2010-02-19 12:26:21.255201 [NOTICE] sofia_glue.c:2880 Hangup
>>>>> sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [CS_CONSUME_MEDIA]
>>>>> [DESTINATION_OUT_OF_ORDER]*
>>>>> 2010-02-19 12:26:21.255201 [DEBUG] switch_channel.c:2063 Send signal
>>>>> sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [KILL]
>>>>> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_session.c:1018 Send
>>>>> signal sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [BREAK]
>>>>> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:314
>>>>> (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) Running State Change
>>>>> CS_HANGUP
>>>>> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:499
>>>>> (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State HANGUP
>>>>> 2010-02-19 12:26:21.255201 [DEBUG] mod_sofia.c:411 Channel
>>>>> sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 hanging up, cause:
>>>>> DESTINATION_OUT_OF_ORDER
>>>>> 2010-02-19 12:26:21.255201 [DEBUG] mod_sofia.c:454 Sending BYE to
>>>>> sofia/backend/XXXXXXXXXX at 10.10.1.110:7065
>>>>> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:46
>>>>> sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 Standard HANGUP, cause:
>>>>> DESTINATION_OUT_OF_ORDER
>>>>> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:499
>>>>> (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State HANGUP going to
>>>>> sleep
>>>>> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:333
>>>>> (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State Change CS_HANGUP ->
>>>>> CS_REPORTING
>>>>> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_session.c:1018 Send
>>>>> signal sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [BREAK]
>>>>> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:314
>>>>> (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) Running State Change
>>>>> CS_REPORTING
>>>>> 2010-02-19 12:26:21.255201 [DEBUG] switch_ivr_originate.c:3185
>>>>> Originate Resulted in Error Cause: 27 [DESTINATION_OUT_OF_ORDER]
>>>>> 2010-02-19 12:26:21.255201 [INFO] mod_dptools.c:2353 Originate Failed.
>>>>> Cause: DESTINATION_OUT_OF_ORDER
>>>>> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:590
>>>>> (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State REPORTING
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100224/4555411c/attachment-0001.html 

From intralanman at freeswitch.org  Wed Feb 24 07:06:31 2010
From: intralanman at freeswitch.org (Raymond Chandler)
Date: Wed, 24 Feb 2010 10:06:31 -0500
Subject: [Freeswitch-users] SIP provider recommendation for US
	termination
In-Reply-To: <6b65470d1002231243x4268de5di655831071c9a28ab@mail.gmail.com>
References: <F56BB50F-6584-4235-91C4-54E4C6EF4867@gmail.com>
	<6b65470d1002231243x4268de5di655831071c9a28ab@mail.gmail.com>
Message-ID: <4B854077.5070600@freeswitch.org>

You might also check the freeswitch.org front page for "friends of 
freeswitch"... These are companies that help to support the FreeSWITCH 
community, so they would probably be recommended first.

-Ray

On 2/23/10 3:43 PM, William Suffill wrote:
> There is a freeswitch-biz list too. I'm sure more people are faced 
> with this issue as well so it might be a good topic for the biz list.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>    
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100224/609ba4ef/attachment.html 

From msc at freeswitch.org  Wed Feb 24 08:11:59 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 24 Feb 2010 08:11:59 -0800
Subject: [Freeswitch-users] freeswitch minimum install
In-Reply-To: <8171AC69-88E7-4793-8C24-2594387C15E5@jerris.com>
References: <219A4D36FE4A42A3B40A594F960461B9@MOBILEE1705>
	<8171AC69-88E7-4793-8C24-2594387C15E5@jerris.com>
Message-ID: <87f2f3b91002240811j1c01ad31t9fe5a87b3d83794e@mail.gmail.com>

On Wed, Feb 24, 2010 at 12:14 AM, Michael Jerris <mike at jerris.com> wrote:

> no, other than manually creating that minimum conf
>
> Mike
>

Although if you want an example of a small configuration look at bkw's FS
softphone configuration:
http://svn.freeswitch.org/svn/configs/softphone/

-MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100224/f0183f6e/attachment.html 

From ivanov.maxim at gmail.com  Wed Feb 24 08:13:53 2010
From: ivanov.maxim at gmail.com (Max Ivanov)
Date: Wed, 24 Feb 2010 16:13:53 +0000
Subject: [Freeswitch-users] Multiple gateways dial string and user busy
Message-ID: <fbfa7b0c1002240813x58ba8541h7fef5b94f045cbda@mail.gmail.com>

Hi all!

when I do test call from fs_cli:
originate sofia/gateway/panas110/223|sofia/gateway/panas111/223
&playaback(local_stream://moh)

If firest attempt returns USER_BUSY it tries to call via second one.
Is it normal? How can I stop calling attempts after first USER_BUSY?


From msc at freeswitch.org  Wed Feb 24 08:18:00 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 24 Feb 2010 08:18:00 -0800
Subject: [Freeswitch-users] internal/external profiles
In-Reply-To: <3c5cf5261002231627s1dfb04b2m64bf9bfeb3ff171d@mail.gmail.com>
References: <3c5cf5261002231627s1dfb04b2m64bf9bfeb3ff171d@mail.gmail.com>
Message-ID: <87f2f3b91002240818q9f95269lc2cee35e8ac60498@mail.gmail.com>

On Tue, Feb 23, 2010 at 4:27 PM, Brian May
<brian at microcomaustralia.com.au>wrote:

> Hello,
>
> Why is it recommended to use separate profiles for internal and external
> SIP?
>
> This page:
> <http://wiki.freeswitch.org/wiki/External_profile>
> suggests it is because of NAT.
>
> However this page recommends using separate profiles even if NAT is
> not an issue:
> <http://wiki.freeswitch.org/wiki/Auto_NAT>: "NOTE: It is still
> recommended that you use a second profile for your SIP providers. The
> default conf/sip_profiles/external.xml is set up specifically for use
> with providers."
>
> However I am still left uncertain what this means.
>
> Not trying to criticize here, just trying to learn.
>

No problem. Separating them has a few advantages: security, scalability, and
readability. The first one on the list is definitely the most important. If
you stuff everything in the internal profile it's easier to open yourself up
to toll fraud.
-MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100224/1dbfc9ed/attachment.html 

From matt at webcontracts.co.uk  Wed Feb 24 08:24:16 2010
From: matt at webcontracts.co.uk (Matthew Law)
Date: Wed, 24 Feb 2010 16:24:16 -0000
Subject: [Freeswitch-users] Snom 300. Any good?
Message-ID: <ce2fe116d187873bee34e78b01169589.squirrel@www.webcontracts.co.uk>


I am very new to VOIP in general and after spending some time getting a
simple FS installation running on a small Xen instance, I am looking to
buy my first VOIP phone.

I don't need anything too fancy.  I have looked at quite a few and the
Snom 300 looks the most favourable so far.  I need something which has a
reasonably priced headset option and will allow me to make and answer
calls 'as' my two businesses from my home office to the FS VM which is out
on the internet.  Do people on the list have experience of this handset or
could you recommend another with similar features and headset available?

Many thanks,

Matt.



From phunk0000 at hotmail.com  Wed Feb 24 08:35:22 2010
From: phunk0000 at hotmail.com (Todd)
Date: Wed, 24 Feb 2010 11:35:22 -0500
Subject: [Freeswitch-users] mod_nibblebill
In-Reply-To: <b39273531002240606p5bc03f34vbc8e934fedfe65a3@mail.gmail.com>
References: <BLU110-W28EA8BD06A601D5D85268BA7420@phx.gbl>	<b39273531002231537k626d1698vd88549a7f59e73a@mail.gmail.com>	<BLU110-DS68941CBE9213A6DA338ACA7410@phx.gbl>
	<b39273531002240606p5bc03f34vbc8e934fedfe65a3@mail.gmail.com>
Message-ID: <BLU110-DS17A81B781D7FAF7A1D188CA7410@phx.gbl>

Yeah,  think I'm going to give it a shot from the beginning again and be
very careful about install and config.  Thanks, I will keep you posted. 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rupa
Schomaker
Sent: Wednesday, February 24, 2010 9:06 AM
To: freeswitch-users
Subject: Re: [Freeswitch-users] mod_nibblebill

 

I am only passingly familiar with MySQL.   There must be a way for it to log
all sql statements sent to it?  

 

Setting up postgresql would be the same (in broad terms) as mysql.  Install
packages, create database/user/tables, populate data, configure odbc dsn,
test.

On Wed, Feb 24, 2010 at 7:03 AM, Todd <phunk0000 at hotmail.com> wrote:

I am attempting to us MySQL.  I installed the spidermonkey mod, newest ODBC,
compiled FS with ODBC, configured xml's in FS.  not 100% sure I did it right
though..followed wiki directions as close as possible.  What is the best way
to verify the SQL is talking to MySQL. or perhaps the easiest way to switch
to postgresql?  Still kinda new to DB admin.  Thanks a ton.

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rupa
Schomaker
Sent: Tuesday, February 23, 2010 6:38 PM
To: freeswitch-users
Subject: Re: [Freeswitch-users] mod_nibblebill

 

what database backend are you using?  Have you verified the SQL is going to
the right database backend?  I use mod_nibblebill against postgresql w/out
problems.

On Tue, Feb 23, 2010 at 9:42 AM, Meg Stroodle <phunk0000 at hotmail.com> wrote:

Hello List!  I am trying to install mod_nibblebill on my FS installation.  I
get the following log entry form FS & nibblebill, but the database table I
setup remains unchanged.  Any help in this matter would be greatly
appreciated.  Following is an excerpt from the FS log:

 

2010-02-23 10:32:52.869149 [DEBUG] switch_rtp.c:2004 Correct ip/port
confirmed.

2010-02-23 10:32:53.077162 [DEBUG] sofia.c:3727 Channel
sofia/internal/3007 at 192.168.15.177 entering state [ready][200]

2010-02-23 10:32:53.149166 [DEBUG] switch_core_session.c:706 Send signal
sofia/internal/sip:3008 at 192.168.15.176:21828 [BREAK]

2010-02-23 10:32:53.149166 [DEBUG] switch_core_session.c:706 Send signal
sofia/internal/3007 at 192.168.15.177 [BREAK]

2010-02-23 10:32:53.149166 [DEBUG] switch_ivr_bridge.c:131
sofia/internal/sip:3008 at 192.168.15.176:21828 receive message [DISPLAY]

2010-02-23 10:32:53.169168 [DEBUG] switch_ivr_bridge.c:131
sofia/internal/3007 at 192.168.15.177 receive message [DISPLAY]

2010-02-23 10:33:21.388949 [DEBUG] mod_nibblebill.c:529 Received request via
SESSION_HEARTBEAT!

2010-02-23 10:33:21.388949 [DEBUG] mod_nibblebill.c:397 Attempting to bill
at $1.0 per minute to account 3007

2010-02-23 10:33:21.388949 [INFO] mod_nibblebill.c:447 Beginning new billing
on 3ec17ffc-c94e-46ff-bb0d-7ad0ef27bdab

2010-02-23 10:33:21.388949 [DEBUG] mod_nibblebill.c:455 28 seconds passed
since last bill time of 2010-02-23 10:32:52

2010-02-23 10:33:21.388949 [DEBUG] mod_nibblebill.c:461 Billing $0.475997 to
3007 (Call: 3ec17ffc-c94e-46ff-bb0d-7ad0ef27bdab / 0.000000 so far)

2010-02-23 10:33:51.508822 [DEBUG] mod_nibblebill.c:529 Received request via
SESSION_HEARTBEAT!

2010-02-23 10:33:51.508822 [DEBUG] mod_nibblebill.c:397 Attempting to bill
at $1.0 per minute to account 3007

2010-02-23 10:33:51.508822 [DEBUG] mod_nibblebill.c:449 Last successful
billing time was 

2010-02-23 10:33:51.508822 [DEBUG] mod_nibblebill.c:455 30 seconds passed
since last bill time of 2010-02-23 10:33:21

2010-02-23 10:33:51.508822 [DEBUG] mod_nibblebill.c:461 Billing $0.501998 to
3007 (Call: 3ec17ffc-c94e-46ff-bb0d-7ad0ef27bdab / 0.475997 so far)

2010-02-23 10:34:21.649708 [DEBUG] mod_nibblebill.c:529 Received request via
SESSION_HEARTBEAT!

2010-02-23 10:34:21.649708 [DEBUG] mod_nibblebill.c:397 Attempting to bill
at $1.0 per minute to account 3007

2010-02-23 10:34:21.649708 [DEBUG] mod_nibblebill.c:449 Last successful
billing time was 

2010-02-23 10:34:21.649708 [DEBUG] mod_nibblebill.c:455 30 seconds passed
since last bill time of 2010-02-23 10:33:51

2010-02-23 10:34:21.649708 [DEBUG] mod_nibblebill.c:461 Billing $0.502348 to
3007 (Call: 3ec17ffc-c94e-46ff-bb0d-7ad0ef27bdab / 0.977995 so far)

2010-02-23 10:34:26.254308 [NOTICE] sofia.c:329 Hangup
sofia/internal/3007 at 192.168.15.177 [CS_EXECUTE] [NORMAL_CLEARING]

2010-02-23 10:34:26.254308 [DEBUG] switch_channel.c:1912 Send signal
sofia/internal/3007 at 192.168.15.177 [KILL]

2010-02-23 10:34:26.254308 [DEBUG] switch_core_session.c:999 Send signal
sofia/internal/3007 at 192.168.15.177 [BREAK]

2010-02-23 10:34:26.254308 [DEBUG] switch_core_state_machine.c:459
sofia/internal/3007 at 192.168.15.177 thread mismatch skipping state handler.

2010-02-23 10:34:26.269008 [DEBUG] switch_ivr_bridge.c:470
sofia/internal/3007 at 192.168.15.177 ending bridge by request from read
function

2010-02-23 10:34:26.269008 [DEBUG] switch_ivr_bridge.c:563 BRIDGE THREAD
DONE [sofia/internal/3007 at 192.168.15.177]

2010-02-23 10:34:26.269008 [DEBUG] switch_ivr_bridge.c:565 Send signal
sofia/internal/sip:3008 at 192.168.15.176:21828 [BREAK]

2010-02-23 10:34:26.269008 [DEBUG] switch_ivr_bridge.c:520
sofia/internal/sip:3008 at 192.168.15.176:21828 receive message [UNBRIDGE]

2010-02-23 10:34:26.269008 [DEBUG] switch_core_session.c:645 Send signal
sofia/internal/sip:3008 at 192.168.15.176:21828 [BREAK]

2010-02-23 10:34:26.269008 [DEBUG] switch_ivr_bridge.c:563 BRIDGE THREAD
DONE [sofia/internal/sip:3008 at 192.168.15.176:21828]

2010-02-23 10:34:26.269008 [DEBUG] switch_ivr_bridge.c:565 Send signal
sofia/internal/3007 at 192.168.15.177 [BREAK]

2010-02-23 10:34:26.269008 [NOTICE] switch_ivr_bridge.c:617 Hangup
sofia/internal/sip:3008 at 192.168.15.176:21828 [CS_EXCHANGE_MEDIA]
[NORMAL_CLEARING]

2010-02-23 10:34:26.269008 [DEBUG] switch_channel.c:1912 Send signal
sofia/internal/sip:3008 at 192.168.15.176:21828 [KILL]

2010-02-23 10:34:26.269008 [DEBUG] switch_core_session.c:999 Send signal
sofia/internal/sip:3008 at 192.168.15.176:21828 [BREAK]

2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:488
(sofia/internal/sip:3008 at 192.168.15.176:21828) State HANGUP

2010-02-23 10:34:26.269008 [DEBUG] mod_sofia.c:358 Channel
sofia/internal/sip:3008 at 192.168.15.176:21828 hanging up, cause:
NORMAL_CLEARING

2010-02-23 10:34:26.269008 [DEBUG] mod_sofia.c:400 Sending BYE to
sofia/internal/sip:3008 at 192.168.15.176:21828

2010-02-23 10:34:26.269008 [DEBUG] mod_nibblebill.c:397 Attempting to bill
at $2.30 per minute to account 3008

2010-02-23 10:34:26.269008 [INFO] mod_nibblebill.c:447 Beginning new billing
on c1c42712-89e9-44c5-914f-11ed8f6aebb1

2010-02-23 10:34:26.269008 [DEBUG] mod_nibblebill.c:455 93 seconds passed
since last bill time of 2010-02-23 10:32:52

2010-02-23 10:34:26.269008 [DEBUG] mod_nibblebill.c:461 Billing $3.582436 to
3008 (Call: c1c42712-89e9-44c5-914f-11ed8f6aebb1 / 0.000000 so far)

2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:46
sofia/internal/sip:3008 at 192.168.15.176:21828 Standard HANGUP, cause:
NORMAL_CLEARING

2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:488
(sofia/internal/sip:3008 at 192.168.15.176:21828) State HANGUP going to sleep

2010-02-23 10:34:26.269008 [INFO] switch_core_session.c:1108
sofia/internal/sip:3008 at 192.168.15.176:21828 setting session heartbeat to 30
second(s).

2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:351
(sofia/internal/sip:3008 at 192.168.15.176:21828) State EXCHANGE_MEDIA going to
sleep

2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/sip:3008 at 192.168.15.176:21828) Running State Change
CS_HANGUP

2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:465
sofia/internal/sip:3008 at 192.168.15.176:21828 handler already called,
skipping state handler.

2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:333
(sofia/internal/sip:3008 at 192.168.15.176:21828) State Change CS_HANGUP ->
CS_REPORTING

2010-02-23 10:34:26.269008 [DEBUG] switch_core_session.c:999 Send signal
sofia/internal/sip:3008 at 192.168.15.176:21828 [BREAK]

2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/sip:3008 at 192.168.15.176:21828) Running State Change
CS_REPORTING

2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:579
(sofia/internal/sip:3008 at 192.168.15.176:21828) State REPORTING

2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:53
sofia/internal/sip:3008 at 192.168.15.176:21828 Standard REPORTING, cause:
NORMAL_CLEARING

2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:579
(sofia/internal/sip:3008 at 192.168.15.176:21828) State REPORTING going to
sleep

2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:327
(sofia/internal/sip:3008 at 192.168.15.176:21828) State Change CS_REPORTING ->
CS_DESTROY

2010-02-23 10:34:26.269008 [DEBUG] switch_core_session.c:999 Send signal
sofia/internal/sip:3008 at 192.168.15.176:21828 [BREAK]

2010-02-23 10:34:26.269008 [DEBUG] switch_core_session.c:1136 Session 6
(sofia/internal/sip:3008 at 192.168.15.176:21828) Locked, Waiting on external
entities

2010-02-23 10:34:26.272437 [DEBUG] switch_core_state_machine.c:348
(sofia/internal/3007 at 192.168.15.177) State EXECUTE going to sleep

2010-02-23 10:34:26.272437 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/3007 at 192.168.15.177) Running State Change CS_HANGUP

2010-02-23 10:34:26.272437 [DEBUG] switch_core_state_machine.c:488
(sofia/internal/3007 at 192.168.15.177) State HANGUP

2010-02-23 10:34:26.272437 [DEBUG] mod_sofia.c:358 Channel
sofia/internal/3007 at 192.168.15.177 hanging up, cause: NORMAL_CLEARING

2010-02-23 10:34:26.272437 [DEBUG] mod_nibblebill.c:397 Attempting to bill
at $1.0 per minute to account 3007

2010-02-23 10:34:26.272437 [DEBUG] mod_nibblebill.c:449 Last successful
billing time was 

2010-02-23 10:34:26.272437 [DEBUG] mod_nibblebill.c:455 4 seconds passed
since last bill time of 2010-02-23 10:34:21

2010-02-23 10:34:26.272437 [DEBUG] mod_nibblebill.c:461 Billing $0.077045 to
3007 (Call: 3ec17ffc-c94e-46ff-bb0d-7ad0ef27bdab / 1.480343 so far)

2010-02-23 10:34:26.272437 [DEBUG] switch_core_state_machine.c:46
sofia/internal/3007 at 192.168.15.177 Standard HANGUP, cause: NORMAL_CLEARING

 

Anyhelp getting nibblebill to connect to the database would be greatly
appreciated.  Thanks

 

 

  _____  

Hotmail: Trusted email with Microsoft's powerful SPAM protection. Sign up
<http://clk.atdmt.com/GBL/go/201469226/direct/01/>  now.


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




-- 
-Rupa


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




-- 
-Rupa

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100224/e99d37d3/attachment-0001.html 

From infos at madovsky.org  Wed Feb 24 09:36:39 2010
From: infos at madovsky.org (Madovsky)
Date: Wed, 24 Feb 2010 12:36:39 -0500
Subject: [Freeswitch-users] Setting username in the header To
References: <15b9404e1002240512y47beb0afha555c24e24d9cc2d@mail.gmail.com>
Message-ID: <B95749F8EE194B39A0220C8D31065444@MOBILEE1705>


  ----- Original Message ----- 
  From: mayamatakeshi 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Wednesday, February 24, 2010 8:12 AM
  Subject: [Freeswitch-users] Setting username in the header To


  Hello,
  while doing a bridge or originate,
  is it possible to send a username in the header To that is different from the one in the Request-URI?
  This is to interoperate with a GW that understands this as a request for redirection (it will send a call to the PSTN with a parameter ISUP RedirectingNumber).

  br,
  Takeshi




------------------------------------------------------------------------------


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  http://www.freeswitch.org



  Maybe with chanel variables
  on wiki
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100224/1a1cf523/attachment.html 

From infos at madovsky.org  Wed Feb 24 09:40:15 2010
From: infos at madovsky.org (Madovsky)
Date: Wed, 24 Feb 2010 12:40:15 -0500
Subject: [Freeswitch-users] FScomm
References: <157FDB96FB88476FBAEC69C1309A0A5F@MOBILEE1705>,
	<COL117-W5563292C886257AFC56548B9420@phx.gbl>,
	<D64783281177440B9094B8181E55EFA7@MOBILEE1705><COL117-W429F3BED9F781A58264238B9420@phx.gbl><11F8492DFC2546598E6FDC3E538301CD@MOBILEE1705>
	<BAA7D201-6E09-4EDE-BBA0-FDFA64E38391@jerris.com>
Message-ID: <8F829C80D359455095B38C0F848E9780@MOBILEE1705>


----- Original Message ----- 
From: "Michael Jerris" <mike at jerris.com>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Wednesday, February 24, 2010 3:40 AM
Subject: Re: [Freeswitch-users] FScomm



On Feb 23, 2010, at 6:18 PM, Madovsky wrote:

>
> ----- Original Message ----- 
> From: Jeff Lenk
> To: freeswitch-users at lists.freeswitch.org
> Sent: Tuesday, February 23, 2010 5:05 PM
> Subject: Re: [Freeswitch-users] FScomm
>
>
> http://wiki.freeswitch.org/wiki/FSComm#Linux
>
> you must run those from the FSComm directory
>

?.

>
> It's what I did,
> but from FS trunk, inside fscomm directory,
> there s only
>
> account.cpp  conf                fshost.h        mainwindow.ui
> resources.qrc
> account.h    FSComm.2008.vcproj  main.cpp        mod_qsettings
> call.cpp     FSComm.pro          mainwindow.cpp  preferences
> call.h       fshost.cpp          mainwindow.h    resources
>


Read those installation instructions again and do them step by step, you 
skipped one.
_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



OOoops, I did gmake rather than qmake... not good to become old.... ;) 



From infos at madovsky.org  Wed Feb 24 09:42:38 2010
From: infos at madovsky.org (Madovsky)
Date: Wed, 24 Feb 2010 12:42:38 -0500
Subject: [Freeswitch-users] FScomm
References: <157FDB96FB88476FBAEC69C1309A0A5F@MOBILEE1705>,
	<COL117-W5563292C886257AFC56548B9420@phx.gbl>,
	<D64783281177440B9094B8181E55EFA7@MOBILEE1705><COL117-W429F3BED9F781A58264238B9420@phx.gbl><11F8492DFC2546598E6FDC3E538301CD@MOBILEE1705>
	<BAA7D201-6E09-4EDE-BBA0-FDFA64E38391@jerris.com>
Message-ID: <D262917F3F4943D49370D0CC015A7582@MOBILEE1705>


----- Original Message ----- 
From: "Michael Jerris" <mike at jerris.com>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Wednesday, February 24, 2010 3:40 AM
Subject: Re: [Freeswitch-users] FScomm



On Feb 23, 2010, at 6:18 PM, Madovsky wrote:

>
> ----- Original Message ----- 
> From: Jeff Lenk
> To: freeswitch-users at lists.freeswitch.org
> Sent: Tuesday, February 23, 2010 5:05 PM
> Subject: Re: [Freeswitch-users] FScomm
>
>
> http://wiki.freeswitch.org/wiki/FSComm#Linux
>
> you must run those from the FSComm directory
>

?.

>
> It's what I did,
> but from FS trunk, inside fscomm directory,
> there s only
>
> account.cpp  conf                fshost.h        mainwindow.ui
> resources.qrc
> account.h    FSComm.2008.vcproj  main.cpp        mod_qsettings
> call.cpp     FSComm.pro          mainwindow.cpp  preferences
> call.h       fshost.cpp          mainwindow.h    resources
>


Read those installation instructions again and do them step by step, you 
skipped one.
_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


Ok now I have

[root at node250 fscomm]# qmake
WARNING: Found potential symbol conflict of mainwindow.cpp (mainwindow.cpp) 
in SOURCES
WARNING: Found potential symbol conflict of mainwindow.h (mainwindow.h) in 
HEADERS
WARNING: Found potential symbol conflict of prefdialog.cpp 
(preferences/prefdialog.cpp) in SOURCES
WARNING: Found potential symbol conflict of prefdialog.h 
(preferences/prefdialog.h) in HEADERS
WARNING: Found potential symbol conflict of accountdialog.cpp 
(preferences/accountdialog.cpp) in SOURCES
WARNING: Found potential symbol conflict of accountdialog.h 
(preferences/accountdialog.h) in HEADERS

[root at node250 fscomm]# qmake
WARNING: Found potential symbol conflict of mainwindow.cpp (mainwindow.cpp) 
in SOURCES
WARNING: Found potential symbol conflict of mainwindow.h (mainwindow.h) in 
HEADERS
WARNING: Found potential symbol conflict of prefdialog.cpp 
(preferences/prefdialog.cpp) in SOURCES
WARNING: Found potential symbol conflict of prefdialog.h 
(preferences/prefdialog.h) in HEADERS
WARNING: Found potential symbol conflict of accountdialog.cpp 
(preferences/accountdialog.cpp) in SOURCES
WARNING: Found potential symbol conflict of accountdialog.h 
(preferences/accountdialog.h) in HEADERS
[root at node250 fscomm]# make
Makefile:278: warning: overriding commands for target `prefdialog.o'
Makefile:215: warning: ignoring old commands for target `prefdialog.o'
Makefile:285: warning: overriding commands for target `accountdialog.o'
Makefile:234: warning: ignoring old commands for target `accountdialog.o'
Makefile:320: warning: overriding commands for target `moc_prefdialog.o'
Makefile:298: warning: ignoring old commands for target `moc_prefdialog.o'
Makefile:323: warning: overriding commands for target `moc_accountdialog.o'
Makefile:307: warning: ignoring old commands for target 
`moc_accountdialog.o'
Makefile:347: warning: overriding commands for target `moc_mainwindow.cpp'
Makefile:326: warning: ignoring old commands for target `moc_mainwindow.cpp'
Makefile:350: warning: overriding commands for target 
`preferences/moc_prefdialog.cpp'
Makefile:332: warning: ignoring old commands for target 
`preferences/moc_prefdialog.cpp'
Makefile:353: warning: overriding commands for target 
`preferences/moc_accountdialog.cpp'
Makefile:341: warning: ignoring old commands for target 
`preferences/moc_accountdialog.cpp'
g++ -c -pipe -Wall -W -O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions 
 -fstack-protector --param=ssp-buffer-size=4 -m64 -mtune=generic  -DQT_NO_DEBUG 
 -DQT_SHARED -DQT_TABLET_SUPPORT -DQT_THREAD_SUPPORT -I/usr/lib64/qt-3.3/mkspecs/default 
 -I. -I../src/include -I../libs/apr/include -I../libs/libteletone/src -I/usr/lib64/qt-3.3/include 
 -o main.o main.cpp
main.cpp:31:25: error: QSplashScreen: No such file or directory
In file included from main.cpp:32:
mainwindow.h:34:23: error: QMainWindow: No such file or directory
mainwindow.h:35:28: error: QTableWidgetItem: No such file or directory
mainwindow.h:36:25: error: QSignalMapper: No such file or directory
mainwindow.h:37:27: error: QSystemTrayIcon: No such file or directory
In file included from mainwindow.h:39,
                 from main.cpp:32:
./fshost.h:32:19: error: QThread: No such file or directory
./fshost.h:33:17: error: QHash: No such file or directory
./fshost.h:34:26: error: QSharedPointer: No such file or directory
In file included from ./fshost.h:36,
                 from mainwindow.h:39,
                 from main.cpp:32:
./call.h:32:18: error: QtCore: No such file or directory
./call.h:33:19: error: QString: No such file or directory
In file included from mainwindow.h:42,
                 from main.cpp:32:
preferences/prefdialog.h:4:19: error: QDialog: No such file or directory
preferences/prefdialog.h:5:24: error: QDomDocument: No such file or 
directory
preferences/prefdialog.h:6:21: error: QSettings: No such file or directory
In file included from ./fshost.h:37,
                 from mainwindow.h:39,
                 from main.cpp:32:
./account.h:18: error: expected constructor, destructor, or type conversion 
before ?static?
In file included from mainwindow.h:39,
                 from main.cpp:32:
./fshost.h:40: error: invalid use of incomplete type ?struct QThread?
/usr/include/QtCore/qobject.h:68: error: forward declaration of ?struct 
QThread?
./fshost.h:46: error: ISO C++ forbids declaration of ?QSharedPointer? with 
no type
./fshost.h:46: error: expected ?;? before ?<? token
./fshost.h:47: error: expected `;' before ?QSharedPointer?
./fshost.h:47: error: ISO C++ forbids declaration of ?QSharedPointer? with 
no type
./fshost.h:47: error: expected ?;? before ?<? token
./fshost.h:48: error: ?QSharedPointer? was not declared in this scope
./fshost.h:48: error: template argument 1 is invalid
./fshost.h:48: error: expected unqualified-id before ?>? token
./fshost.h:49: error: ISO C++ forbids declaration of ?QSharedPointer? with 
no type
./fshost.h:49: error: expected ?;? before ?<? token
./fshost.h:50: error: ISO C++ forbids declaration of ?QSharedPointer? with 
no type
./fshost.h:50: error: expected ?;? before ?<? token
./fshost.h:51: error: ISO C++ forbids declaration of ?QSharedPointer? with 
no type
./fshost.h:51: error: expected ?;? before ?<? token
./fshost.h:52: error: ?QSharedPointer? has not been declared
./fshost.h:52: error: expected ?,? or ?...? before ?<? token
./fshost.h:60: error: ?QSharedPointer? has not been declared
./fshost.h:60: error: expected ?,? or ?...? before ?<? token
./fshost.h:61: error: ?QSharedPointer? has not been declared
./fshost.h:61: error: expected ?,? or ?...? before ?<? token
./fshost.h:62: error: ?QSharedPointer? has not been declared
./fshost.h:62: error: expected ?,? or ?...? before ?<? token
./fshost.h:63: error: ?QSharedPointer? has not been declared
./fshost.h:63: error: expected ?,? or ?...? before ?<? token
./fshost.h:64: error: ?QSharedPointer? has not been declared
./fshost.h:64: error: expected ?,? or ?...? before ?<? token
./fshost.h:65: error: ?QSharedPointer? has not been declared
./fshost.h:65: error: expected ?,? or ?...? before ?<? token
./fshost.h:66: error: ?QSharedPointer? has not been declared
./fshost.h:66: error: expected ?,? or ?...? before ?<? token
./fshost.h:67: error: ?QSharedPointer? has not been declared
./fshost.h:67: error: expected ?,? or ?...? before ?<? token
./fshost.h:71: error: ?QSharedPointer? has not been declared
./fshost.h:71: error: expected ?,? or ?...? before ?<? token
./fshost.h:78: error: ?QSharedPointer? was not declared in this scope
./fshost.h:78: error: template argument 2 is invalid
./fshost.h:78: error: expected unqualified-id before ?>? token
./fshost.h:79: error: ?QSharedPointer? was not declared in this scope
./fshost.h:79: error: template argument 2 is invalid
./fshost.h:79: error: expected unqualified-id before ?>? token
./fshost.h:80: error: field ?_bleg_uuids? has incomplete type
In file included from mainwindow.h:42,
                 from main.cpp:32:
preferences/prefdialog.h:17: error: invalid use of incomplete type ?struct 
QDialog?
/usr/include/QtGui/qwindowdefs.h:57: error: forward declaration of ?struct 
QDialog?
preferences/prefdialog.h:31: error: ISO C++ forbids declaration of 
?QSettings? with no type
preferences/prefdialog.h:31: error: expected ?;? before ?*? token
In file included from main.cpp:32:
mainwindow.h:48: error: expected class-name before ?{? token
mainwindow.h:65: error: ?QTableWidgetItem? has not been declared
mainwindow.h:71: error: ?QSharedPointer? has not been declared
mainwindow.h:71: error: expected ?,? or ?...? before ?<? token
mainwindow.h:72: error: ?QSharedPointer? has not been declared
mainwindow.h:72: error: expected ?,? or ?...? before ?<? token
mainwindow.h:73: error: ?QSharedPointer? has not been declared
mainwindow.h:73: error: expected ?,? or ?...? before ?<? token
mainwindow.h:74: error: ?QSharedPointer? has not been declared
mainwindow.h:74: error: expected ?,? or ?...? before ?<? token
mainwindow.h:75: error: ?QSharedPointer? has not been declared
mainwindow.h:75: error: expected ?,? or ?...? before ?<? token
mainwindow.h:78: error: ?QSharedPointer? has not been declared
mainwindow.h:78: error: expected ?,? or ?...? before ?<? token
mainwindow.h:79: error: ?QSharedPointer? has not been declared
mainwindow.h:79: error: expected ?,? or ?...? before ?<? token
mainwindow.h:80: error: ?QSharedPointer? has not been declared
mainwindow.h:80: error: expected ?,? or ?...? before ?<? token
mainwindow.h:81: error: ?QSystemTrayIcon? has not been declared
mainwindow.h:81: error: expected ?,? or ?...? before ?reason?
mainwindow.h:86: error: ISO C++ forbids declaration of ?QSignalMapper? with 
no type
mainwindow.h:86: error: expected ?;? before ?*? token
mainwindow.h:88: error: ISO C++ forbids declaration of ?QSystemTrayIcon? 
with no type
mainwindow.h:88: error: expected ?;? before ?*? token
main.cpp: In function ?int main(int, char**)?:
main.cpp:41: error: variable ?QPixmap image? has initializer but incomplete 
type
main.cpp:42: error: ?QSplashScreen? was not declared in this scope
main.cpp:42: error: ?splash? was not declared in this scope
main.cpp:42: error: expected type-specifier before ?QSplashScreen?
main.cpp:42: error: expected `;' before ?QSplashScreen?
main.cpp:48: error: no matching function for call to 
?QObject::connect(FSHost*, const char [9], MainWindow*, const char [8])?
/usr/include/QtCore/qobject.h:202: note: candidates are: static bool 
QObject::connect(const QObject*, const char*, const QObject*, const char*, 
Qt::ConnectionType)
/usr/include/QtCore/qobject.h:307: note:                 bool 
QObject::connect(const QObject*, const char*, const char*, 
Qt::ConnectionType) const
main.cpp:49: error: ?class FSHost? has no member named ?start?
main.cpp:41: warning: unused variable ?image?
./fshost.h: At global scope:
./fshost.h:90: warning: ?void eventHandlerCallback(switch_event_t*)? defined 
but not used
make: *** [main.o] Error 1


Any idea ?

Thx




From jerry.richards at teotech.com  Wed Feb 24 09:56:22 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Wed, 24 Feb 2010 09:56:22 -0800
Subject: [Freeswitch-users] Two Ethernet Ports, One FS Instance?
Message-ID: <EDBC42167D634005BF79FC52D8D6CB56@greyhawk.tonecommander.com>


If my server has two ethernet ports, do I need two FS instances?  Or can a
single FS instance send/receive messages through both ports using two
different sip_profiles?

Best Regards,
Jerry



From mrene_lists at avgs.ca  Wed Feb 24 09:58:29 2010
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Wed, 24 Feb 2010 12:58:29 -0500
Subject: [Freeswitch-users] Two Ethernet Ports, One FS Instance?
In-Reply-To: <EDBC42167D634005BF79FC52D8D6CB56@greyhawk.tonecommander.com>
References: <EDBC42167D634005BF79FC52D8D6CB56@greyhawk.tonecommander.com>
Message-ID: <98EA0C25-9D5A-46A4-9DF9-FDA77BA8EFFB@avgs.ca>

You can use two different sip profiles. Each profile has its own  
binding parameters for sip and rtp.

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 24-Feb-10, at 12:56 PM, Jerry Richards wrote:

>
> If my server has two ethernet ports, do I need two FS instances?  Or  
> can a
> single FS instance send/receive messages through both ports using two
> different sip_profiles?
>
> Best Regards,
> Jerry
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From ledoktre at meanie.us  Wed Feb 24 10:02:51 2010
From: ledoktre at meanie.us (Tim Streit)
Date: Wed, 24 Feb 2010 12:02:51 -0600
Subject: [Freeswitch-users] Skypiax- how to know which card to use?
In-Reply-To: <7b197bef1002191441u455e8d8an4eb03c460254c6d3@mail.gmail.com>
References: <4B7F0302.3060303@meanie.us>
	<7b197bef1002191441u455e8d8an4eb03c460254c6d3@mail.gmail.com>
Message-ID: <4B8569CB.6070804@meanie.us>

Hello,

I was writing to inquire how this skypiax update was coming along.  I 
didn't see it in the mailing list, but also since it had been nearly 1 
week, I wanted to be sure if I didn't miss the announcement.  I am very 
anxious to try this new update of the module.. It should be awesome!

Thanks,

Tim

Giovanni Maruzzelli wrote:
> before to delve in the troubleshooting, I have to say that I'm
> modifying the audio skypiax code in svn, so maybe it's just my fault
> ;).
>
> please be patient for a little while, I hope to have done with it in a
> couple days.
>
> I'll announce to the mailing list when done.
>
> In the mean time, at least one good news for you user of SkypeIn
> service: a new feature of mod_skypiax is intended to recognize the
> DTMFs coming from SkypeIn, so the incoming calls will be able to use
> ivr, voicemail, etc.


From errotan at gmail.com  Wed Feb 24 10:35:14 2010
From: errotan at gmail.com (=?iso-8859-1?q?Pusk=E1s_Zsolt?=)
Date: Wed, 24 Feb 2010 19:35:14 +0100
Subject: [Freeswitch-users] mod_fax undefined symbol
In-Reply-To: <41F9D585-2116-4219-9AF7-E8E944D43362@ttnc.co.uk>
References: <6FCE44FF-70C1-412D-B722-4C8D359381BA@ttnc.co.uk>
	<201002191904.39081.errotan@gmail.com>
	<41F9D585-2116-4219-9AF7-E8E944D43362@ttnc.co.uk>
Message-ID: <201002241935.14921.errotan@gmail.com>

2010. febru?r 21. 22.25.05 TTNC - Technical d?tummal ezt ?rta:
> Out of interest, I downgraded my versions of libtiff and libjpeg to the
>  versions shipped with Lenny:
> 
> voipin1:/opt# dpkg -l | egrep 'libtiff|libjpeg'
> ii  libjpeg62                            6b-14                    The
>  Independent JPEG Group's JPEG runtime library ii  libjpeg62-dev           
>              6b-14                    Development files for the IJG JPEG
>  library ii  libtiff4                             3.8.2-11.2              
>  Tag Image File Format (TIFF) library ii  libtiff4-dev                     
>     3.8.2-11.2               Tag Image File Format library (TIFF),
>  development files ii  libtiffxx0c2                         3.8.2-11.2     
>           Tag Image File Format (TIFF) library -- C++ interface
> 
> Everything else stayed at the 'squeeze' version.
> 
> Still didn't make any different, **/opt/freeswitch/mod/mod_fax.so:
>  undefined symbol: TIFFDefaultStripSize**
> 
> I'm guessing that points to it being a problem outside of these packages
>  and somewhere else in Debian?
> 
> Russ
> 
> On 19 Feb 2010, at 18:04, Pusk?s Zsolt wrote:
> > 2010. febru?r 19. 18.44.32 TTNC - Technical d?tummal ezt ?rta:
> >> On 19 Feb 2010, at 17:25, Pusk?s Zsolt wrote:
> >>> Just compiled svn16700 on Debian "lenny" x86, mod_fax loads and works
> >>> perfectly. I have an ongoing compile on another machine (amd64) if It
> >>> don't works i will send a mail (in 1 hour) otherwise consider it
> >>> working.
> >>
> >> How did you compile it? Using dpkg-buildpackage or via make/make
> >> install?
> >>
> >> Do you have any debian versions of libtiff4(-dev) installed?
> >
> > Okay i can confirm that mod_fax works on Debian "lenny" amd64 but it
> > don't work on Debian "testing,squeeze" amd64.
> >
> > 2010-02-19 18:49:14.610297 [CRIT] switch_loadable_module.c:882 Error
> > Loading module /usr/local/freeswitch/mod/mod_fax.so
> > **/usr/local/freeswitch/mod/mod_fax.so: undefined symbol:
> > TIFFDefaultStripSize**
> >
> > I haven't tried to compile mod_fax on testing before so i don't know what
> > is causeing the problem :(
> >
> > # ldd mod_fax.so
> >        linux-vdso.so.1 =>  (0x00007fff106f6000)
> >        libm.so.6 => /lib/libm.so.6 (0x00007f506b711000)
> >        libfreeswitch.so.1 => /usr/local/freeswitch/lib/libfreeswitch.so.1
> > (0x00007f506b345000)
> >        libpthread.so.0 => /lib/libpthread.so.0 (0x00007f506b128000)
> >        libc.so.6 => /lib/libc.so.6 (0x00007f506add4000)
> >        libssl.so.0.9.8 => /usr/lib/libssl.so.0.9.8 (0x00007f506ab82000)
> >        libcrypto.so.0.9.8 => /usr/lib/libcrypto.so.0.9.8
> > (0x00007f506a7e2000) libncurses.so.5 => /lib/libncurses.so.5
> > (0x00007f506a59d000) libstdc++.so.6 => /usr/lib/libstdc++.so.6
> > (0x00007f506a28d000) libgcc_s.so.1 => /lib/libgcc_s.so.1
> > (0x00007f506a076000)
> >        libodbc.so.1 => /usr/lib/libodbc.so.1 (0x00007f5069e17000)
> >        /lib64/ld-linux-x86-64.so.2 (0x00007f506bc31000)
> >        libdl.so.2 => /lib/libdl.so.2 (0x00007f5069c13000)
> >        libz.so.1 => /usr/lib/libz.so.1 (0x00007f50699fb000)
> >        libltdl.so.7 => /usr/lib/libltdl.so.7 (0x00007f50697f2000)
> >
> > Recently in debian "testing" libtiff4 and libjpeg is upgraded:
> > libtiff 3.9.2-3+b1
> > libjpeg62 6b-16.1
> > libjeg8 8-2.1
> >
> > Q&A:
> > Q: How did you compile it? Using dpkg-buildpackage or via make/make
> > install? A: svn-clean ./bootsrap ./configure make etc.
> >
> > Q: Do you have any debian versions of libtiff4(-dev) installed?
> > A: Yes:3.8.2-11.2
> >
> > I open a jira for this.
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 

Could you retest if it works for you now ? It seems after the update from 
debootstrap 1.0.21 to 1.0.22 it works. As ldd shows libjpeg.so.62 was not 
linked to mod_fax. (maybe that was the problem)

If you can upgrade all your packages to the latest and report back that it 
works we can close this issue.

Current ldd for me:

# ldd /usr/local/freeswitch/mod/mod_fax.so
        linux-vdso.so.1 =>  (0x00007fff96794000)
        libm.so.6 => /lib/libm.so.6 (0x00007f57921b9000)
        libfreeswitch.so.1 => /usr/local/freeswitch/lib/libfreeswitch.so.1 
(0x00007f5791ded000)
        libjpeg.so.62 => /usr/lib/libjpeg.so.62 (0x00007f5791bc9000)
        libpthread.so.0 => /lib/libpthread.so.0 (0x00007f57919ad000)
        libc.so.6 => /lib/libc.so.6 (0x00007f5791659000)
        libssl.so.0.9.8 => /usr/lib/libssl.so.0.9.8 (0x00007f5791406000)
        libcrypto.so.0.9.8 => /usr/lib/libcrypto.so.0.9.8 (0x00007f5791067000)
        libncurses.so.5 => /lib/libncurses.so.5 (0x00007f5790e22000)
        libstdc++.so.6 => /usr/lib/libstdc++.so.6 (0x00007f5790b11000)
        libgcc_s.so.1 => /lib/libgcc_s.so.1 (0x00007f57908fb000)
        libodbc.so.1 => /usr/lib/libodbc.so.1 (0x00007f579069c000)
        /lib64/ld-linux-x86-64.so.2 (0x00007f579272a000)
        libdl.so.2 => /lib/libdl.so.2 (0x00007f5790497000)
        libz.so.1 => /usr/lib/libz.so.1 (0x00007f5790280000)
        libltdl.so.7 => /usr/lib/libltdl.so.7 (0x00007f5790077000)


From christian.loeschenkohl at xpirio.com  Wed Feb 24 10:38:30 2010
From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=)
Date: Wed, 24 Feb 2010 19:38:30 +0100
Subject: [Freeswitch-users] conferences lead to high server load
Message-ID: <4B857226.10308@xpirio.com>

hi

we do experience a unusual high server load with the latest freeswitch versions.
about 50 conference users lead to a server load of over 10 - reproducible by the way.
this wans't the case until my latest trunk update.

fs version: 16714
os: debian lenny x86_64

has something substantially changed in mod_conference recently?

br

-- 
Ing. Christian L?schenkohl
Technische Leitung, Forschung & Entwicklung VoIP

xpirio
Telekommunikation & Service GmbH
Lakeside B04
9020 Klagenfurt
Austria

T  +43 (0) 5 77 11 - 1000
F  +43 (0) 5 77 11 - 1002
E  christian.loeschenkohl at xpirio.com


From anthony.minessale at gmail.com  Wed Feb 24 10:56:19 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 24 Feb 2010 12:56:19 -0600
Subject: [Freeswitch-users] mod_fax undefined symbol
In-Reply-To: <201002241935.14921.errotan@gmail.com>
References: <6FCE44FF-70C1-412D-B722-4C8D359381BA@ttnc.co.uk>
	<201002191904.39081.errotan@gmail.com>
	<41F9D585-2116-4219-9AF7-E8E944D43362@ttnc.co.uk>
	<201002241935.14921.errotan@gmail.com>
Message-ID: <191c3a031002241056k34e2a743t5427f29aa765b526@mail.gmail.com>

This issue was already fixed yesterday afternoon.
It was a libtool2 issue.


On Wed, Feb 24, 2010 at 12:35 PM, Pusk?s Zsolt <errotan at gmail.com> wrote:

> 2010. febru?r 21. 22.25.05 TTNC - Technical d?tummal ezt ?rta:
> > Out of interest, I downgraded my versions of libtiff and libjpeg to the
> >  versions shipped with Lenny:
> >
> > voipin1:/opt# dpkg -l | egrep 'libtiff|libjpeg'
> > ii  libjpeg62                            6b-14                    The
> >  Independent JPEG Group's JPEG runtime library ii  libjpeg62-dev
> >              6b-14                    Development files for the IJG JPEG
> >  library ii  libtiff4                             3.8.2-11.2
> >  Tag Image File Format (TIFF) library ii  libtiff4-dev
> >     3.8.2-11.2               Tag Image File Format library (TIFF),
> >  development files ii  libtiffxx0c2                         3.8.2-11.2
> >           Tag Image File Format (TIFF) library -- C++ interface
> >
> > Everything else stayed at the 'squeeze' version.
> >
> > Still didn't make any different, **/opt/freeswitch/mod/mod_fax.so:
> >  undefined symbol: TIFFDefaultStripSize**
> >
> > I'm guessing that points to it being a problem outside of these packages
> >  and somewhere else in Debian?
> >
> > Russ
> >
> > On 19 Feb 2010, at 18:04, Pusk?s Zsolt wrote:
> > > 2010. febru?r 19. 18.44.32 TTNC - Technical d?tummal ezt ?rta:
> > >> On 19 Feb 2010, at 17:25, Pusk?s Zsolt wrote:
> > >>> Just compiled svn16700 on Debian "lenny" x86, mod_fax loads and works
> > >>> perfectly. I have an ongoing compile on another machine (amd64) if It
> > >>> don't works i will send a mail (in 1 hour) otherwise consider it
> > >>> working.
> > >>
> > >> How did you compile it? Using dpkg-buildpackage or via make/make
> > >> install?
> > >>
> > >> Do you have any debian versions of libtiff4(-dev) installed?
> > >
> > > Okay i can confirm that mod_fax works on Debian "lenny" amd64 but it
> > > don't work on Debian "testing,squeeze" amd64.
> > >
> > > 2010-02-19 18:49:14.610297 [CRIT] switch_loadable_module.c:882 Error
> > > Loading module /usr/local/freeswitch/mod/mod_fax.so
> > > **/usr/local/freeswitch/mod/mod_fax.so: undefined symbol:
> > > TIFFDefaultStripSize**
> > >
> > > I haven't tried to compile mod_fax on testing before so i don't know
> what
> > > is causeing the problem :(
> > >
> > > # ldd mod_fax.so
> > >        linux-vdso.so.1 =>  (0x00007fff106f6000)
> > >        libm.so.6 => /lib/libm.so.6 (0x00007f506b711000)
> > >        libfreeswitch.so.1 =>
> /usr/local/freeswitch/lib/libfreeswitch.so.1
> > > (0x00007f506b345000)
> > >        libpthread.so.0 => /lib/libpthread.so.0 (0x00007f506b128000)
> > >        libc.so.6 => /lib/libc.so.6 (0x00007f506add4000)
> > >        libssl.so.0.9.8 => /usr/lib/libssl.so.0.9.8 (0x00007f506ab82000)
> > >        libcrypto.so.0.9.8 => /usr/lib/libcrypto.so.0.9.8
> > > (0x00007f506a7e2000) libncurses.so.5 => /lib/libncurses.so.5
> > > (0x00007f506a59d000) libstdc++.so.6 => /usr/lib/libstdc++.so.6
> > > (0x00007f506a28d000) libgcc_s.so.1 => /lib/libgcc_s.so.1
> > > (0x00007f506a076000)
> > >        libodbc.so.1 => /usr/lib/libodbc.so.1 (0x00007f5069e17000)
> > >        /lib64/ld-linux-x86-64.so.2 (0x00007f506bc31000)
> > >        libdl.so.2 => /lib/libdl.so.2 (0x00007f5069c13000)
> > >        libz.so.1 => /usr/lib/libz.so.1 (0x00007f50699fb000)
> > >        libltdl.so.7 => /usr/lib/libltdl.so.7 (0x00007f50697f2000)
> > >
> > > Recently in debian "testing" libtiff4 and libjpeg is upgraded:
> > > libtiff 3.9.2-3+b1
> > > libjpeg62 6b-16.1
> > > libjeg8 8-2.1
> > >
> > > Q&A:
> > > Q: How did you compile it? Using dpkg-buildpackage or via make/make
> > > install? A: svn-clean ./bootsrap ./configure make etc.
> > >
> > > Q: Do you have any debian versions of libtiff4(-dev) installed?
> > > A: Yes:3.8.2-11.2
> > >
> > > I open a jira for this.
> > >
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> Could you retest if it works for you now ? It seems after the update from
> debootstrap 1.0.21 to 1.0.22 it works. As ldd shows libjpeg.so.62 was not
> linked to mod_fax. (maybe that was the problem)
>
> If you can upgrade all your packages to the latest and report back that it
> works we can close this issue.
>
> Current ldd for me:
>
> # ldd /usr/local/freeswitch/mod/mod_fax.so
>        linux-vdso.so.1 =>  (0x00007fff96794000)
>        libm.so.6 => /lib/libm.so.6 (0x00007f57921b9000)
>         libfreeswitch.so.1 => /usr/local/freeswitch/lib/libfreeswitch.so.1
> (0x00007f5791ded000)
>        libjpeg.so.62 => /usr/lib/libjpeg.so.62 (0x00007f5791bc9000)
>        libpthread.so.0 => /lib/libpthread.so.0 (0x00007f57919ad000)
>        libc.so.6 => /lib/libc.so.6 (0x00007f5791659000)
>        libssl.so.0.9.8 => /usr/lib/libssl.so.0.9.8 (0x00007f5791406000)
>        libcrypto.so.0.9.8 => /usr/lib/libcrypto.so.0.9.8
> (0x00007f5791067000)
>        libncurses.so.5 => /lib/libncurses.so.5 (0x00007f5790e22000)
>        libstdc++.so.6 => /usr/lib/libstdc++.so.6 (0x00007f5790b11000)
>        libgcc_s.so.1 => /lib/libgcc_s.so.1 (0x00007f57908fb000)
>        libodbc.so.1 => /usr/lib/libodbc.so.1 (0x00007f579069c000)
>        /lib64/ld-linux-x86-64.so.2 (0x00007f579272a000)
>        libdl.so.2 => /lib/libdl.so.2 (0x00007f5790497000)
>        libz.so.1 => /usr/lib/libz.so.1 (0x00007f5790280000)
>        libltdl.so.7 => /usr/lib/libltdl.so.7 (0x00007f5790077000)
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100224/e6f88f89/attachment.html 

From anthony.minessale at gmail.com  Wed Feb 24 10:58:38 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 24 Feb 2010 12:58:38 -0600
Subject: [Freeswitch-users] conferences lead to high server load
In-Reply-To: <4B857226.10308@xpirio.com>
References: <4B857226.10308@xpirio.com>
Message-ID: <191c3a031002241058y517e63fdte0d929b97a33f9ac@mail.gmail.com>

load average has no meaning with FS, you have to look at the CPU usage per
CPU and thread.
Are you experiencing any audio problems or are you just concerned about that
load number?

If you have a box that has trouble with timing it could cost more resources.
you can always run freeswitch -vm to use an alternate form of timing that
may not manifest into the load average.


2010/2/24 Christian L?schenkohl <christian.loeschenkohl at xpirio.com>

> hi
>
> we do experience a unusual high server load with the latest freeswitch
> versions.
> about 50 conference users lead to a server load of over 10 - reproducible
> by the way.
> this wans't the case until my latest trunk update.
>
> fs version: 16714
> os: debian lenny x86_64
>
> has something substantially changed in mod_conference recently?
>
> br
>
> --
> Ing. Christian L?schenkohl
> Technische Leitung, Forschung & Entwicklung VoIP
>
> xpirio
> Telekommunikation & Service GmbH
> Lakeside B04
> 9020 Klagenfurt
> Austria
>
> T  +43 (0) 5 77 11 - 1000
> F  +43 (0) 5 77 11 - 1002
> E  christian.loeschenkohl at xpirio.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100224/0d6fe45e/attachment-0001.html 

From anthony.minessale at gmail.com  Wed Feb 24 11:08:51 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 24 Feb 2010 13:08:51 -0600
Subject: [Freeswitch-users] Setting username in the header To
In-Reply-To: <B95749F8EE194B39A0220C8D31065444@MOBILEE1705>
References: <15b9404e1002240512y47beb0afha555c24e24d9cc2d@mail.gmail.com>
	<B95749F8EE194B39A0220C8D31065444@MOBILEE1705>
Message-ID: <191c3a031002241108q3e268e64m983005b60c196f8c@mail.gmail.com>

use the variable
{sip_invite_to_uri=<desired number>}
at the beginning of your dial string
you can either supply a full sup uri or just then number
alternatively, you can terminate your dial string with ^<desired number>


On Wed, Feb 24, 2010 at 11:36 AM, Madovsky <infos at madovsky.org> wrote:

>
>
> ----- Original Message -----
> *From:* mayamatakeshi <mayamatakeshi at gmail.com>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Wednesday, February 24, 2010 8:12 AM
> *Subject:* [Freeswitch-users] Setting username in the header To
>
> Hello,
> while doing a bridge or originate,
> is it possible to send a username in the header To that is different from
> the one in the Request-URI?
> This is to interoperate with a GW that understands this as a request for
> redirection (it will send a call to the PSTN with a parameter ISUP
> RedirectingNumber).
>
> br,
> Takeshi
>
>  ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> Maybe with chanel variables
> on wiki
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100224/c7c5a59a/attachment.html 

From brian at freeswitch.org  Wed Feb 24 11:10:40 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 24 Feb 2010 13:10:40 -0600
Subject: [Freeswitch-users] Setting username in the header To
In-Reply-To: <B95749F8EE194B39A0220C8D31065444@MOBILEE1705>
References: <15b9404e1002240512y47beb0afha555c24e24d9cc2d@mail.gmail.com>
	<B95749F8EE194B39A0220C8D31065444@MOBILEE1705>
Message-ID: <237546EF-35DF-4826-A330-C7F4D3190BDF@freeswitch.org>

Example when replying to the list... Please do not echo back the full headers of possible LIKE below.

To answer your question this is possible if you set the sip_invite_to_uri (needs the full URI)

/b

On Feb 24, 2010, at 11:36 AM, Madovsky wrote:

>  
> ----- Original Message -----
> From: mayamatakeshi
> To: freeswitch-users at lists.freeswitch.org
> Sent: Wednesday, February 24, 2010 8:12 AM
> Subject: [Freeswitch-users] Setting username in the header To
> 
> Hello,
> while doing a bridge or originate,
> is it possible to send a username in the header To that is different from the one in the Request-URI?
> This is to interoperate with a GW that understands this as a request for redirection (it will send a call to the PSTN with a parameter ISUP RedirectingNumber).
> 
> br,
> Takeshi
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>  
>  
> Maybe with chanel variables
> on wiki
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100224/8f0d0669/attachment.html 

From msc at freeswitch.org  Wed Feb 24 11:14:48 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 24 Feb 2010 11:14:48 -0800
Subject: [Freeswitch-users] Snom 300. Any good?
In-Reply-To: <ce2fe116d187873bee34e78b01169589.squirrel@www.webcontracts.co.uk>
References: <ce2fe116d187873bee34e78b01169589.squirrel@www.webcontracts.co.uk>
Message-ID: <87f2f3b91002241114q25ca604es6f6863fc714006d2@mail.gmail.com>

On Wed, Feb 24, 2010 at 8:24 AM, Matthew Law <matt at webcontracts.co.uk>wrote:

>
> I am very new to VOIP in general and after spending some time getting a
> simple FS installation running on a small Xen instance, I am looking to
> buy my first VOIP phone.
>
> I don't need anything too fancy.  I have looked at quite a few and the
> Snom 300 looks the most favourable so far.  I need something which has a
> reasonably priced headset option and will allow me to make and answer
> calls 'as' my two businesses from my home office to the FS VM which is out
> on the internet.  Do people on the list have experience of this handset or
> could you recommend another with similar features and headset available?
>
> Many thanks,
>
> Matt.
>
I have used this phone quite a bit. It is nothing fancy but it works. The
only complaint I've heard is that the handset volume doesn't go too high so
you might want to consider a headset.
-MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100224/ef8ca4c3/attachment.html 

From anthony.minessale at gmail.com  Wed Feb 24 11:17:13 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 24 Feb 2010 13:17:13 -0600
Subject: [Freeswitch-users] big thanks to all freeswitch developers and
	contributing users
In-Reply-To: <4B83F040.7040005@xpirio.com>
References: <4B83F040.7040005@xpirio.com>
Message-ID: <191c3a031002241117r3d630ccdo82e5d2a3cb24efa9@mail.gmail.com>

Thank you,

I should frame this email =p

2010/2/23 Christian L?schenkohl <christian.loeschenkohl at xpirio.com>

> i want to say a big THANKY YOU to all contributing freeswitch community
> members.
>
> over one year has passed since i did fall in love with this project.
> it is getting better every day, one get's help and advices if needed.
> the admins do care about nearly every problem - no matter if it's big or
> small.
> i also did manage an opensource project and i wish i had done it with that
> much
> heart and intense power that i see here.
>
> i also hope that i can contribute back enough (questions, bug reports, wiki
> enhancements).
>
> wishing you all the best
> br
>
> --
> Ing. Christian L?schenkohl
> Technische Leitung, Forschung & Entwicklung VoIP
>
> xpirio
> Telekommunikation & Service GmbH
> Lakeside B04
> 9020 Klagenfurt
> Austria
>
> T  +43 (0) 5 77 11 - 1000
> F  +43 (0) 5 77 11 - 1002
> E  christian.loeschenkohl at xpirio.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100224/9ad61d47/attachment.html 

From christian.loeschenkohl at xpirio.com  Wed Feb 24 11:25:49 2010
From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=)
Date: Wed, 24 Feb 2010 20:25:49 +0100
Subject: [Freeswitch-users] conferences lead to high server load
In-Reply-To: <191c3a031002241058y517e63fdte0d929b97a33f9ac@mail.gmail.com>
References: <4B857226.10308@xpirio.com>
	<191c3a031002241058y517e63fdte0d929b97a33f9ac@mail.gmail.com>
Message-ID: <4B857D3D.5080000@xpirio.com>

yes, only the high load number concerned me.
i tested participating in one of the conferences, there is no audio problem.

i'll try -vm and give feedback on this.

br



Anthony Minessale wrote:

> load average has no meaning with FS, you have to look at the CPU usage 
> per CPU and thread.
> Are you experiencing any audio problems or are you just concerned about 
> that load number?
> 
> If you have a box that has trouble with timing it could cost more resources.
> you can always run freeswitch -vm to use an alternate form of timing 
> that may not manifest into the load average.
> 
> 
> 2010/2/24 Christian L?schenkohl <christian.loeschenkohl at xpirio.com 
> <mailto:christian.loeschenkohl at xpirio.com>>
> 
>     hi
> 
>     we do experience a unusual high server load with the latest
>     freeswitch versions.
>     about 50 conference users lead to a server load of over 10 -
>     reproducible by the way.
>     this wans't the case until my latest trunk update.
> 
>     fs version: 16714
>     os: debian lenny x86_64
> 
>     has something substantially changed in mod_conference recently?
> 
>     br
> 
>     --
>     Ing. Christian L?schenkohl
>     Technische Leitung, Forschung & Entwicklung VoIP
> 
>     xpirio
>     Telekommunikation & Service GmbH
>     Lakeside B04
>     9020 Klagenfurt
>     Austria
> 
>     T  +43 (0) 5 77 11 - 1000
>     F  +43 (0) 5 77 11 - 1002
>     E  christian.loeschenkohl at xpirio.com
>     <mailto:christian.loeschenkohl at xpirio.com>
> 
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
> 
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com 
> <mailto:MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com 
> <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org 
> <mailto:sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888 
> <http://iax:guest at conference.freeswitch.org/888>
> googletalk:conf+888 at conference.freeswitch.org 
> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


-- 
Ing. Christian L?schenkohl
Technische Leitung, Forschung & Entwicklung VoIP

xpirio
Telekommunikation & Service GmbH
Lakeside B04
9020 Klagenfurt
Austria

T  +43 (0) 5 77 11 - 1000
F  +43 (0) 5 77 11 - 1002
E  christian.loeschenkohl at xpirio.com


From msc at freeswitch.org  Wed Feb 24 11:42:17 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 24 Feb 2010 11:42:17 -0800
Subject: [Freeswitch-users] Increasing call Volume
In-Reply-To: <c55d46951002240620s7d95789bidc564488d580ec0e@mail.gmail.com>
References: <c55d46951002240553k9bf14bak1b684096e2a09fff@mail.gmail.com>
	<COL117-W7D89C1D1F7F15CDFBF1D3B9410@phx.gbl>
	<c55d46951002240620s7d95789bidc564488d580ec0e@mail.gmail.com>
Message-ID: <87f2f3b91002241142k1fc8a9c9ge573b8d913d29e80@mail.gmail.com>

On Wed, Feb 24, 2010 at 6:20 AM, Max Bridgewater
<max.bridgewater at gmail.com>wrote:

> Thanks jeff. so what would be the recommended way for solving this problem?
>
> You need to know why there's echo. Just tinkering with the audio levels
might lessen the symptoms for a while but if you don't know the underlying
cause then your solution may just be like putting a bandage on a gunshot
wound. I'm not familiar with Portech... are these VoIP calls? Get a pcap and
analyze with Wireshark. Also, do you experience echo on all calls to/from
Portech? Is it only with them? Do you have multiple devices on your end to
test with?  See if you can narrow the scope a little as that might help you
figure it out.
-MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100224/53158c37/attachment.html 

From infos at madovsky.org  Wed Feb 24 12:05:29 2010
From: infos at madovsky.org (Madovsky)
Date: Wed, 24 Feb 2010 15:05:29 -0500
Subject: [Freeswitch-users] qt framework link broken
Message-ID: <0CD7185862B54C1ABF67D77BA55664F7@MOBILEE1705>

Just to inform that
at the link
http://wiki.freeswitch.org/wiki/FSComm#Linux

the qt framework link is broken, so as I'm new to this emailist
I don't want to correct myself on wiki.

Regards

Franck
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100224/e5678561/attachment.html 

From gmaruzz at celliax.org  Wed Feb 24 12:07:56 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Wed, 24 Feb 2010 21:07:56 +0100
Subject: [Freeswitch-users] Skypiax- how to know which card to use?
In-Reply-To: <4B8569CB.6070804@meanie.us>
References: <4B7F0302.3060303@meanie.us>
	<7b197bef1002191441u455e8d8an4eb03c460254c6d3@mail.gmail.com> 
	<4B8569CB.6070804@meanie.us>
Message-ID: <7b197bef1002241207k4da85b36l5e2e9c8e14f4fe32@mail.gmail.com>

On Wed, Feb 24, 2010 at 7:02 PM, Tim Streit <ledoktre at meanie.us> wrote:
> Hello,
>
> I was writing to inquire how this skypiax update was coming along. ?I
> didn't see it in the mailing list, but also since it had been nearly 1
> week, I wanted to be sure if I didn't miss the announcement. ?I am very
> anxious to try this new update of the module.. It should be awesome!

Hehehe, no, you've not missed the announcement.

Is taking me some more time than I was expecting.
But's arriving...
I'll post here the announcement ;)

-giovanni



>
> Thanks,
>
> Tim
>
> Giovanni Maruzzelli wrote:
>> before to delve in the troubleshooting, I have to say that I'm
>> modifying the audio skypiax code in svn, so maybe it's just my fault
>> ;).
>>
>> please be patient for a little while, I hope to have done with it in a
>> couple days.
>>
>> I'll announce to the mailing list when done.
>>
>> In the mean time, at least one good news for you user of SkypeIn
>> service: a new feature of mod_skypiax is intended to recognize the
>> DTMFs coming from SkypeIn, so the incoming calls will be able to use
>> ivr, voicemail, etc.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From feeswitch.ml at hez.ca  Wed Feb 24 12:17:11 2010
From: feeswitch.ml at hez.ca (Hez Ronningen)
Date: Wed, 24 Feb 2010 12:17:11 -0800
Subject: [Freeswitch-users] error loading module dingaling
References: <C1FC75D9-29E0-4233-8613-2E23551045B6@hez.ca>
Message-ID: <C622EDED-A96F-412F-B98B-021946A6347D@hez.ca>

Hello,

Sorry for reposting this, but I have dug in to this extensively and cannot find the problem.

I've dug in to the libraries to try and figure out what exact library they expect but cannot find the link with ldd. I've tried a couple other gnutls libraries with no success. I've searched the mailing lists, google, and hounded the irc channel with no results. This loading problem is happening with both the compiled from source ver and the deb installed version.

Begin forwarded message:
> Installed freeswitch on ubuntu and enabled the dingaling module but when it boots I get the following error
> 
> 2010-02-21 11:59:55.213568 [CRIT] switch_loadable_module.c:882 Error Loading module /opt/freeswitch/mod/mod_dingaling.so
> **/opt/freeswitch/mod/mod_dingaling.so: undefined symbol: gnutls_global_init**
> 
> I have the following libraries installed
> 
> ii  libgnutls-dev                        2.8.3-2                                                the GNU TLS library - development files
> ii  libgnutls26                          2.8.3-2                                                the GNU TLS library - runtime library
> 
> 
> Is there a library I am missing or an incompatibility?
> 
> Any help is much appreciated,
> Hez



From phunk0000 at hotmail.com  Wed Feb 24 13:19:59 2010
From: phunk0000 at hotmail.com (Todd)
Date: Wed, 24 Feb 2010 16:19:59 -0500
Subject: [Freeswitch-users] mod_nibblebill
In-Reply-To: <BLU110-DS17A81B781D7FAF7A1D188CA7410@phx.gbl>
References: <BLU110-W28EA8BD06A601D5D85268BA7420@phx.gbl>	<b39273531002231537k626d1698vd88549a7f59e73a@mail.gmail.com>	<BLU110-DS68941CBE9213A6DA338ACA7410@phx.gbl>	<b39273531002240606p5bc03f34vbc8e934fedfe65a3@mail.gmail.com>
	<BLU110-DS17A81B781D7FAF7A1D188CA7410@phx.gbl>
Message-ID: <BLU110-DS16A00D0320F590FC964F93A7410@phx.gbl>

Sweet, figured it out.

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Todd
Sent: Wednesday, February 24, 2010 11:35 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] mod_nibblebill

 

Yeah,  think I'm going to give it a shot from the beginning again and be
very careful about install and config.  Thanks, I will keep you posted. 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rupa
Schomaker
Sent: Wednesday, February 24, 2010 9:06 AM
To: freeswitch-users
Subject: Re: [Freeswitch-users] mod_nibblebill

 

I am only passingly familiar with MySQL.   There must be a way for it to log
all sql statements sent to it?  

 

Setting up postgresql would be the same (in broad terms) as mysql.  Install
packages, create database/user/tables, populate data, configure odbc dsn,
test.

On Wed, Feb 24, 2010 at 7:03 AM, Todd <phunk0000 at hotmail.com> wrote:

I am attempting to us MySQL.  I installed the spidermonkey mod, newest ODBC,
compiled FS with ODBC, configured xml's in FS.  not 100% sure I did it right
though..followed wiki directions as close as possible.  What is the best way
to verify the SQL is talking to MySQL. or perhaps the easiest way to switch
to postgresql?  Still kinda new to DB admin.  Thanks a ton.

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rupa
Schomaker
Sent: Tuesday, February 23, 2010 6:38 PM
To: freeswitch-users
Subject: Re: [Freeswitch-users] mod_nibblebill

 

what database backend are you using?  Have you verified the SQL is going to
the right database backend?  I use mod_nibblebill against postgresql w/out
problems.

On Tue, Feb 23, 2010 at 9:42 AM, Meg Stroodle <phunk0000 at hotmail.com> wrote:

Hello List!  I am trying to install mod_nibblebill on my FS installation.  I
get the following log entry form FS & nibblebill, but the database table I
setup remains unchanged.  Any help in this matter would be greatly
appreciated.  Following is an excerpt from the FS log:

 

2010-02-23 10:32:52.869149 [DEBUG] switch_rtp.c:2004 Correct ip/port
confirmed.

2010-02-23 10:32:53.077162 [DEBUG] sofia.c:3727 Channel
sofia/internal/3007 at 192.168.15.177 entering state [ready][200]

2010-02-23 10:32:53.149166 [DEBUG] switch_core_session.c:706 Send signal
sofia/internal/sip:3008 at 192.168.15.176:21828 [BREAK]

2010-02-23 10:32:53.149166 [DEBUG] switch_core_session.c:706 Send signal
sofia/internal/3007 at 192.168.15.177 [BREAK]

2010-02-23 10:32:53.149166 [DEBUG] switch_ivr_bridge.c:131
sofia/internal/sip:3008 at 192.168.15.176:21828 receive message [DISPLAY]

2010-02-23 10:32:53.169168 [DEBUG] switch_ivr_bridge.c:131
sofia/internal/3007 at 192.168.15.177 receive message [DISPLAY]

2010-02-23 10:33:21.388949 [DEBUG] mod_nibblebill.c:529 Received request via
SESSION_HEARTBEAT!

2010-02-23 10:33:21.388949 [DEBUG] mod_nibblebill.c:397 Attempting to bill
at $1.0 per minute to account 3007

2010-02-23 10:33:21.388949 [INFO] mod_nibblebill.c:447 Beginning new billing
on 3ec17ffc-c94e-46ff-bb0d-7ad0ef27bdab

2010-02-23 10:33:21.388949 [DEBUG] mod_nibblebill.c:455 28 seconds passed
since last bill time of 2010-02-23 10:32:52

2010-02-23 10:33:21.388949 [DEBUG] mod_nibblebill.c:461 Billing $0.475997 to
3007 (Call: 3ec17ffc-c94e-46ff-bb0d-7ad0ef27bdab / 0.000000 so far)

2010-02-23 10:33:51.508822 [DEBUG] mod_nibblebill.c:529 Received request via
SESSION_HEARTBEAT!

2010-02-23 10:33:51.508822 [DEBUG] mod_nibblebill.c:397 Attempting to bill
at $1.0 per minute to account 3007

2010-02-23 10:33:51.508822 [DEBUG] mod_nibblebill.c:449 Last successful
billing time was 

2010-02-23 10:33:51.508822 [DEBUG] mod_nibblebill.c:455 30 seconds passed
since last bill time of 2010-02-23 10:33:21

2010-02-23 10:33:51.508822 [DEBUG] mod_nibblebill.c:461 Billing $0.501998 to
3007 (Call: 3ec17ffc-c94e-46ff-bb0d-7ad0ef27bdab / 0.475997 so far)

2010-02-23 10:34:21.649708 [DEBUG] mod_nibblebill.c:529 Received request via
SESSION_HEARTBEAT!

2010-02-23 10:34:21.649708 [DEBUG] mod_nibblebill.c:397 Attempting to bill
at $1.0 per minute to account 3007

2010-02-23 10:34:21.649708 [DEBUG] mod_nibblebill.c:449 Last successful
billing time was 

2010-02-23 10:34:21.649708 [DEBUG] mod_nibblebill.c:455 30 seconds passed
since last bill time of 2010-02-23 10:33:51

2010-02-23 10:34:21.649708 [DEBUG] mod_nibblebill.c:461 Billing $0.502348 to
3007 (Call: 3ec17ffc-c94e-46ff-bb0d-7ad0ef27bdab / 0.977995 so far)

2010-02-23 10:34:26.254308 [NOTICE] sofia.c:329 Hangup
sofia/internal/3007 at 192.168.15.177 [CS_EXECUTE] [NORMAL_CLEARING]

2010-02-23 10:34:26.254308 [DEBUG] switch_channel.c:1912 Send signal
sofia/internal/3007 at 192.168.15.177 [KILL]

2010-02-23 10:34:26.254308 [DEBUG] switch_core_session.c:999 Send signal
sofia/internal/3007 at 192.168.15.177 [BREAK]

2010-02-23 10:34:26.254308 [DEBUG] switch_core_state_machine.c:459
sofia/internal/3007 at 192.168.15.177 thread mismatch skipping state handler.

2010-02-23 10:34:26.269008 [DEBUG] switch_ivr_bridge.c:470
sofia/internal/3007 at 192.168.15.177 ending bridge by request from read
function

2010-02-23 10:34:26.269008 [DEBUG] switch_ivr_bridge.c:563 BRIDGE THREAD
DONE [sofia/internal/3007 at 192.168.15.177]

2010-02-23 10:34:26.269008 [DEBUG] switch_ivr_bridge.c:565 Send signal
sofia/internal/sip:3008 at 192.168.15.176:21828 [BREAK]

2010-02-23 10:34:26.269008 [DEBUG] switch_ivr_bridge.c:520
sofia/internal/sip:3008 at 192.168.15.176:21828 receive message [UNBRIDGE]

2010-02-23 10:34:26.269008 [DEBUG] switch_core_session.c:645 Send signal
sofia/internal/sip:3008 at 192.168.15.176:21828 [BREAK]

2010-02-23 10:34:26.269008 [DEBUG] switch_ivr_bridge.c:563 BRIDGE THREAD
DONE [sofia/internal/sip:3008 at 192.168.15.176:21828]

2010-02-23 10:34:26.269008 [DEBUG] switch_ivr_bridge.c:565 Send signal
sofia/internal/3007 at 192.168.15.177 [BREAK]

2010-02-23 10:34:26.269008 [NOTICE] switch_ivr_bridge.c:617 Hangup
sofia/internal/sip:3008 at 192.168.15.176:21828 [CS_EXCHANGE_MEDIA]
[NORMAL_CLEARING]

2010-02-23 10:34:26.269008 [DEBUG] switch_channel.c:1912 Send signal
sofia/internal/sip:3008 at 192.168.15.176:21828 [KILL]

2010-02-23 10:34:26.269008 [DEBUG] switch_core_session.c:999 Send signal
sofia/internal/sip:3008 at 192.168.15.176:21828 [BREAK]

2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:488
(sofia/internal/sip:3008 at 192.168.15.176:21828) State HANGUP

2010-02-23 10:34:26.269008 [DEBUG] mod_sofia.c:358 Channel
sofia/internal/sip:3008 at 192.168.15.176:21828 hanging up, cause:
NORMAL_CLEARING

2010-02-23 10:34:26.269008 [DEBUG] mod_sofia.c:400 Sending BYE to
sofia/internal/sip:3008 at 192.168.15.176:21828

2010-02-23 10:34:26.269008 [DEBUG] mod_nibblebill.c:397 Attempting to bill
at $2.30 per minute to account 3008

2010-02-23 10:34:26.269008 [INFO] mod_nibblebill.c:447 Beginning new billing
on c1c42712-89e9-44c5-914f-11ed8f6aebb1

2010-02-23 10:34:26.269008 [DEBUG] mod_nibblebill.c:455 93 seconds passed
since last bill time of 2010-02-23 10:32:52

2010-02-23 10:34:26.269008 [DEBUG] mod_nibblebill.c:461 Billing $3.582436 to
3008 (Call: c1c42712-89e9-44c5-914f-11ed8f6aebb1 / 0.000000 so far)

2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:46
sofia/internal/sip:3008 at 192.168.15.176:21828 Standard HANGUP, cause:
NORMAL_CLEARING

2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:488
(sofia/internal/sip:3008 at 192.168.15.176:21828) State HANGUP going to sleep

2010-02-23 10:34:26.269008 [INFO] switch_core_session.c:1108
sofia/internal/sip:3008 at 192.168.15.176:21828 setting session heartbeat to 30
second(s).

2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:351
(sofia/internal/sip:3008 at 192.168.15.176:21828) State EXCHANGE_MEDIA going to
sleep

2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/sip:3008 at 192.168.15.176:21828) Running State Change
CS_HANGUP

2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:465
sofia/internal/sip:3008 at 192.168.15.176:21828 handler already called,
skipping state handler.

2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:333
(sofia/internal/sip:3008 at 192.168.15.176:21828) State Change CS_HANGUP ->
CS_REPORTING

2010-02-23 10:34:26.269008 [DEBUG] switch_core_session.c:999 Send signal
sofia/internal/sip:3008 at 192.168.15.176:21828 [BREAK]

2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/sip:3008 at 192.168.15.176:21828) Running State Change
CS_REPORTING

2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:579
(sofia/internal/sip:3008 at 192.168.15.176:21828) State REPORTING

2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:53
sofia/internal/sip:3008 at 192.168.15.176:21828 Standard REPORTING, cause:
NORMAL_CLEARING

2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:579
(sofia/internal/sip:3008 at 192.168.15.176:21828) State REPORTING going to
sleep

2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:327
(sofia/internal/sip:3008 at 192.168.15.176:21828) State Change CS_REPORTING ->
CS_DESTROY

2010-02-23 10:34:26.269008 [DEBUG] switch_core_session.c:999 Send signal
sofia/internal/sip:3008 at 192.168.15.176:21828 [BREAK]

2010-02-23 10:34:26.269008 [DEBUG] switch_core_session.c:1136 Session 6
(sofia/internal/sip:3008 at 192.168.15.176:21828) Locked, Waiting on external
entities

2010-02-23 10:34:26.272437 [DEBUG] switch_core_state_machine.c:348
(sofia/internal/3007 at 192.168.15.177) State EXECUTE going to sleep

2010-02-23 10:34:26.272437 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/3007 at 192.168.15.177) Running State Change CS_HANGUP

2010-02-23 10:34:26.272437 [DEBUG] switch_core_state_machine.c:488
(sofia/internal/3007 at 192.168.15.177) State HANGUP

2010-02-23 10:34:26.272437 [DEBUG] mod_sofia.c:358 Channel
sofia/internal/3007 at 192.168.15.177 hanging up, cause: NORMAL_CLEARING

2010-02-23 10:34:26.272437 [DEBUG] mod_nibblebill.c:397 Attempting to bill
at $1.0 per minute to account 3007

2010-02-23 10:34:26.272437 [DEBUG] mod_nibblebill.c:449 Last successful
billing time was 

2010-02-23 10:34:26.272437 [DEBUG] mod_nibblebill.c:455 4 seconds passed
since last bill time of 2010-02-23 10:34:21

2010-02-23 10:34:26.272437 [DEBUG] mod_nibblebill.c:461 Billing $0.077045 to
3007 (Call: 3ec17ffc-c94e-46ff-bb0d-7ad0ef27bdab / 1.480343 so far)

2010-02-23 10:34:26.272437 [DEBUG] switch_core_state_machine.c:46
sofia/internal/3007 at 192.168.15.177 Standard HANGUP, cause: NORMAL_CLEARING

 

Anyhelp getting nibblebill to connect to the database would be greatly
appreciated.  Thanks

 

 

  _____  

Hotmail: Trusted email with Microsoft's powerful SPAM protection. Sign up
<http://clk.atdmt.com/GBL/go/201469226/direct/01/>  now.


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




-- 
-Rupa


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




-- 
-Rupa

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100224/1122d508/attachment-0001.html 

From joseph.puchalski at personalcyberspace.com  Wed Feb 24 15:45:02 2010
From: joseph.puchalski at personalcyberspace.com (Joseph Puchalski)
Date: Wed, 24 Feb 2010 23:45:02 +0000
Subject: [Freeswitch-users] Retrieving voicemail without entering user ID
	(extension)
Message-ID: <093DD565390C1E4FB15D7B383E86BB05AFF6BC@Goose.personalcyberspace.net>

I'm trying to modify my dialplan so that I can press a single button on my phone, be connected to voicemail, and enter only a password to gain access.

Currently I use a programmable key to dial 4000. I am prompted for my ID, and then password.

I've poked around "mod voicemail" on the wiki and searched the mailing list and web, but haven't found enough info. I have discovered that this behavior seems to have been available in previous versions of the default dialplan.

Is it still possible? Is it advisable? Was this feature/behavior removed for security reasons?

I apologize ahead of time if the answer is somewhere in plain sight that I haven't looked yet. If so, I'd much appreciate being pointed in the right direction.

As always, thanks for any help,

Joe P.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100224/9ab0a2a9/attachment.html 

From rupa at rupa.com  Wed Feb 24 16:01:15 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Wed, 24 Feb 2010 18:01:15 -0600
Subject: [Freeswitch-users] Retrieving voicemail without entering user
	ID (extension)
In-Reply-To: <093DD565390C1E4FB15D7B383E86BB05AFF6BC@Goose.personalcyberspace.net>
References: <093DD565390C1E4FB15D7B383E86BB05AFF6BC@Goose.personalcyberspace.net>
Message-ID: <b39273531002241601m273ff333y2261d930dbcb3fa7@mail.gmail.com>

Look at the end of:

http://wiki.freeswitch.org/wiki/Mod_voicemail#Check_Voice_Mail

<http://wiki.freeswitch.org/wiki/Mod_voicemail#Check_Voice_Mail>Advisable?
 With it enabled, I can walk up to anyone's phone and retrieve their VM
w/out authentication.  It was removed on purpose due to that reason as far
as I remember.

On Wed, Feb 24, 2010 at 5:45 PM, Joseph Puchalski <
joseph.puchalski at personalcyberspace.com> wrote:

>  I?m trying to modify my dialplan so that I can press a single button on
> my phone, be connected to voicemail, and enter only a password to gain
> access.
>
>
>
> Currently I use a programmable key to dial 4000. I am prompted for my ID,
> and then password.
>
>
>
> I?ve poked around ?mod voicemail? on the wiki and searched the mailing list
> and web, but haven?t found enough info. I have discovered that this behavior
> seems to have been available in previous versions of the default dialplan.
>
>
>
> Is it still possible? Is it advisable? Was this feature/behavior removed
> for security reasons?
>
>
>
> I apologize ahead of time if the answer is somewhere in plain sight that I
> haven?t looked yet. If so, I?d much appreciate being pointed in the right
> direction.
>
>
>
> As always, thanks for any help,
>
>
>
> Joe P.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100224/42eddbfa/attachment.html 

From lists at redbonez.net  Wed Feb 24 17:10:23 2010
From: lists at redbonez.net (Adam Ford)
Date: Wed, 24 Feb 2010 18:10:23 -0700
Subject: [Freeswitch-users] Retrieving voicemail without entering
	user	ID (extension)
In-Reply-To: <b39273531002241601m273ff333y2261d930dbcb3fa7@mail.gmail.com>
References: <093DD565390C1E4FB15D7B383E86BB05AFF6BC@Goose.personalcyberspace.net>
	<b39273531002241601m273ff333y2261d930dbcb3fa7@mail.gmail.com>
Message-ID: <012c01cab5b7$5052d990$f0f88cb0$@net>

>From reading that wiki article it seems to me that the key to achieving the
functionality you are looking for would simply be a matter of adding the
desired extension to the end of the default action (where the $1 is):

 

<action application="voicemail" data="check $${voicemail_profile} $${domain}
$1"/>

 

If I am reading it correctly, this should bypass having to enter a mailbox
ID, but still require your voicemail password. Off the top of my head, you
could probably achieve this by replacing the $1 with a variable storing the
extension which called 4000.  I would have to look it up to see if there is
a system variable for that or if you would have to assign a custom one. I am
still relatively new to FreeSWITCH myself.

 

-Adam

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rupa
Schomaker
Sent: Wednesday, February 24, 2010 5:01 PM
To: freeswitch-users
Subject: Re: [Freeswitch-users] Retrieving voicemail without entering user
ID (extension)

 

Look at the end of:

 

http://wiki.freeswitch.org/wiki/Mod_voicemail#Check_Voice_Mail

 

Advisable?  With it enabled, I can walk up to anyone's phone and retrieve
their VM w/out authentication.  It was removed on purpose due to that reason
as far as I remember.

On Wed, Feb 24, 2010 at 5:45 PM, Joseph Puchalski
<joseph.puchalski at personalcyberspace.com> wrote:

I'm trying to modify my dialplan so that I can press a single button on my
phone, be connected to voicemail, and enter only a password to gain access. 

 

Currently I use a programmable key to dial 4000. I am prompted for my ID,
and then password.

 

I've poked around "mod voicemail" on the wiki and searched the mailing list
and web, but haven't found enough info. I have discovered that this behavior
seems to have been available in previous versions of the default dialplan.

 

Is it still possible? Is it advisable? Was this feature/behavior removed for
security reasons?

 

I apologize ahead of time if the answer is somewhere in plain sight that I
haven't looked yet. If so, I'd much appreciate being pointed in the right
direction.

 

As always, thanks for any help,

 

Joe P.


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




-- 
-Rupa

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100224/4fd70554/attachment.html 

From mayamatakeshi at gmail.com  Wed Feb 24 17:14:14 2010
From: mayamatakeshi at gmail.com (mayamatakeshi)
Date: Thu, 25 Feb 2010 10:14:14 +0900
Subject: [Freeswitch-users] Setting username in the header To
In-Reply-To: <191c3a031002241108q3e268e64m983005b60c196f8c@mail.gmail.com>
References: <15b9404e1002240512y47beb0afha555c24e24d9cc2d@mail.gmail.com>
	<B95749F8EE194B39A0220C8D31065444@MOBILEE1705>
	<191c3a031002241108q3e268e64m983005b60c196f8c@mail.gmail.com>
Message-ID: <15b9404e1002241714w782cd54i10d3c4c74626c0cc@mail.gmail.com>

Thanks,
I have added an entry for it in the wiki:

http://wiki.freeswitch.org/wiki/Channel_Variables#sip_invite_to_uri


On Thu, Feb 25, 2010 at 4:08 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> use the variable
> {sip_invite_to_uri=<desired number>}
> at the beginning of your dial string
> you can either supply a full sup uri or just then number
> alternatively, you can terminate your dial string with ^<desired number>
>
>
>
> On Wed, Feb 24, 2010 at 11:36 AM, Madovsky <infos at madovsky.org> wrote:
>
>>
>>
>> ----- Original Message -----
>>  *From:* mayamatakeshi <mayamatakeshi at gmail.com>
>> *To:* freeswitch-users at lists.freeswitch.org
>> *Sent:* Wednesday, February 24, 2010 8:12 AM
>> *Subject:* [Freeswitch-users] Setting username in the header To
>>
>> Hello,
>> while doing a bridge or originate,
>> is it possible to send a username in the header To that is different from
>> the one in the Request-URI?
>> This is to interoperate with a GW that understands this as a request for
>> redirection (it will send a call to the PSTN with a parameter ISUP
>> RedirectingNumber).
>>
>> br,
>> Takeshi
>>
>>  ------------------------------
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> Maybe with chanel variables
>> on wiki
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100225/f2c8e09d/attachment-0001.html 

From brian at freeswitch.org  Wed Feb 24 17:20:57 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 24 Feb 2010 19:20:57 -0600
Subject: [Freeswitch-users] Retrieving voicemail without entering
	user	ID (extension)
In-Reply-To: <012c01cab5b7$5052d990$f0f88cb0$@net>
References: <093DD565390C1E4FB15D7B383E86BB05AFF6BC@Goose.personalcyberspace.net>
	<b39273531002241601m273ff333y2261d930dbcb3fa7@mail.gmail.com>
	<012c01cab5b7$5052d990$f0f88cb0$@net>
Message-ID: <0180F807-95D2-4F1A-9F3A-679795053EA2@freeswitch.org>

You are 100% correct.

/b

On Feb 24, 2010, at 7:10 PM, Adam Ford wrote:

> From reading that wiki article it seems to me that the key to achieving the functionality you are looking for would simply be a matter of adding the desired extension to the end of the default action (where the $1 is):
>  
> <action application="voicemail" data="check $${voicemail_profile} $${domain} $1"/>
>  
> If I am reading it correctly, this should bypass having to enter a mailbox ID, but still require your voicemail password. Off the top of my head, you could probably achieve this by replacing the $1 with a variable storing the extension which called 4000.  I would have to look it up to see if there is a system variable for that or if you would have to assign a custom one. I am still relatively new to FreeSWITCH myself.
>  
> -Adam

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100224/7a7be836/attachment.html 

From larclap at yahoo.com  Wed Feb 24 19:08:45 2010
From: larclap at yahoo.com (Lars Zeb)
Date: Wed, 24 Feb 2010 19:08:45 -0800
Subject: [Freeswitch-users] Retrieving voicemail without entering
	user	ID (extension)
In-Reply-To: <b39273531002241601m273ff333y2261d930dbcb3fa7@mail.gmail.com>
References: <093DD565390C1E4FB15D7B383E86BB05AFF6BC@Goose.personalcyberspace.net>
	<b39273531002241601m273ff333y2261d930dbcb3fa7@mail.gmail.com>
Message-ID: <000f01cab5c7$d7e292f0$87a7b8d0$@com>

Joe,

 

I used the extension below, but I think that Brian said it was too insecure.
Being a total beginner, I removed the condition. 

 

    <extension name="Local_Extension_Abe">

      <condition field="destination_number" expression="^(101[0-9])$">

        <action application="set" data="dialed_ext=$1"/>

        <action application="export" data="dialed_ext=$1"/>

      </condition>

      <condition field="destination_number"
expression="^${caller_id_number}$">

        <action application="set"
data="voicemail_authorized=${sip_authorized}"/>

        <action application="answer"/>

        <action application="sleep" data="1000"/>

        <action application="voicemail" data="check default $${domain}
${dialed_ext}"/>

        <anti-action application="bind_meta_app" data="1 a a
execute_extension::dx XML features"/>

        <anti-action application="bind_meta_app" data="2 a a
record_session::$${base_dir}/recordings/${caller_id_number}.${strftime(%Y-%m
-%d-%H-%M-%S)}.wav"/>

        <anti-action application="bind_meta_app" data="3 a a
execute_extension::cf XML features"/>

        <anti-action application="set" data="transfer_ringback=${us-ring}"/>

        <anti-action application="set" data="call_timeout=15"/>

        <anti-action application="set"
data="sip_exclude_contact=${network_addr}"/>

        <anti-action application="set" data="hangup_after_bridge=true"/>

        <!--<anti-action application="set"
data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,BUSY,USER_BUSY,NO_ANSWER,TIM
EOUT,NO_ROUTE_DESTINATION"/> -->

        <anti-action application="set" data="continue_on_fail=true"/>

        <anti-action application="db"
data="insert/call_return/${dialed_ext}/${caller_id_number}"/>

        <anti-action application="db"
data="insert/last_dial_ext/${dialed_ext}/${uuid}"/>

        <anti-action application="bridge"
data="user/${dialed_ext}@$${domain}"/>

        <anti-action application="answer"/>

        <anti-action application="sleep" data="1000"/>

        <anti-action application="voicemail" data="default $${domain}
${dialed_ext}"/>

      </condition>

    </extension>

 

Lars

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rupa
Schomaker
Sent: Wednesday, February 24, 2010 4:01 PM
To: freeswitch-users
Subject: Re: [Freeswitch-users] Retrieving voicemail without entering user
ID (extension)

 

Look at the end of:

 

http://wiki.freeswitch.org/wiki/Mod_voicemail#Check_Voice_Mail

 

Advisable?  With it enabled, I can walk up to anyone's phone and retrieve
their VM w/out authentication.  It was removed on purpose due to that reason
as far as I remember.

On Wed, Feb 24, 2010 at 5:45 PM, Joseph Puchalski
<joseph.puchalski at personalcyberspace.com> wrote:

I'm trying to modify my dialplan so that I can press a single button on my
phone, be connected to voicemail, and enter only a password to gain access. 

 

Currently I use a programmable key to dial 4000. I am prompted for my ID,
and then password.

 

I've poked around "mod voicemail" on the wiki and searched the mailing list
and web, but haven't found enough info. I have discovered that this behavior
seems to have been available in previous versions of the default dialplan.

 

Is it still possible? Is it advisable? Was this feature/behavior removed for
security reasons?

 

I apologize ahead of time if the answer is somewhere in plain sight that I
haven't looked yet. If so, I'd much appreciate being pointed in the right
direction.

 

As always, thanks for any help,

 

Joe P.


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




-- 
-Rupa

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100224/a70f78ec/attachment.html 

From msc at freeswitch.org  Wed Feb 24 20:08:00 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 24 Feb 2010 20:08:00 -0800
Subject: [Freeswitch-users] Call for help - adding information to the wiki:
	SIP ALG's
Message-ID: <87f2f3b91002242008y599f86f0t86f467f8c16b4a20@mail.gmail.com>

Hi all,

I've just completed a new wiki page:

http://wiki.freeswitch.org/wiki/ALG

I would like all of you who have dealt with routers with SIP ALG's to submit
your input. I would like to see this page have a list of how-to's for all of
the popular routers. If we can make it easy for people to disable SIP ALG's
then I think we can all save ourselves time and energy answering questions
in IRC and the mailing lists. Please by all means add your knowledge here. I
started with the 2wire 3800HGV that I got for my ATT Uverse service.

If you have knowledge that you like to add to the wiki (on this subject or
any other) but are not confident in your wiki editing skills then contact me
off list and I will be happy to help you get up to speed. Editing your first
wiki page is always the hardest... :)

Thanks again for all of your help! By the way, today's community conference
call was great. Please plan on attending next week and we'll talk about more
great FreeSWITCH stuff. I will have the recording of Rupa discussing
mod_limit up on line as soon as I can.

Take care,
Michael
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100224/568aca63/attachment-0001.html 

From jason at jasonjgw.net  Wed Feb 24 20:27:30 2010
From: jason at jasonjgw.net (Jason White)
Date: Thu, 25 Feb 2010 15:27:30 +1100
Subject: [Freeswitch-users] Call for help - adding information to the
 wiki: SIP ALG's
In-Reply-To: <87f2f3b91002242008y599f86f0t86f467f8c16b4a20@mail.gmail.com>
References: <87f2f3b91002242008y599f86f0t86f467f8c16b4a20@mail.gmail.com>
Message-ID: <20100225042730.GA19249@jdc.jasonjgw.net>

Michael Collins <msc at freeswitch.org> wrote:
> I've just completed a new wiki page:
> 
> http://wiki.freeswitch.org/wiki/ALG
> 
> I would like all of you who have dealt with routers with SIP ALG's to submit
> your input. I would like to see this page have a list of how-to's for all of
> the popular routers. If we can make it easy for people to disable SIP ALG's
> then I think we can all save ourselves time and energy answering questions
> in IRC and the mailing lists. Please by all means add your knowledge here. 

For Cisco IOS, the following commands do it:
no ip nat service sip tcp port 5060
no ip nat service sip udp port 5060

Regrettably I'm not in a position to edit the wiki, but anyone is welcome to
add the above.



From mouncifbb at gmail.com  Wed Feb 24 20:48:19 2010
From: mouncifbb at gmail.com (Mouncif Benniane)
Date: Wed, 24 Feb 2010 23:48:19 -0500
Subject: [Freeswitch-users] compilation error of skypiax_protocol.c
In-Reply-To: <13529f9d0912022243y700728d4l30c7eb4e3152d1c9@mail.gmail.com>
References: <13529f9d0912022202g3254517dscc635bfb9c8b4439@mail.gmail.com>
	<9387ED56-4CDE-4685-AFF7-98E930BBA1F9@avgs.ca>
	<13529f9d0912022233i368b84c7s9bd960eb1380b53e@mail.gmail.com>
	<13529f9d0912022243y700728d4l30c7eb4e3152d1c9@mail.gmail.com>
Message-ID: <ade741d61002242048v150cb025q23c4463d38c9bbb9@mail.gmail.com>

you probably need: libjpeg-devel instead. just a thought.



On Thu, Dec 3, 2009 at 1:43 AM, Jingwei Yang <jingwei.yang at gmail.com> wrote:

> Not sure whether this error is due to the lack of libjpeg. I just double
> checked, this library had been installed.
>
> Package libjpeg-6b-37.i386 already installed and latest version
>
>
>
> On Thu, Dec 3, 2009 at 2:33 PM, Jingwei Yang <jingwei.yang at gmail.com>wrote:
>
>> Hi Mathieu, thanks for the promptly reply. The error has been fixed.
>> However, I encounter another one.
>>
>> gcc -I/usr/src/freeswitch/libs/tiff-3.8.2/libtiff -DNDEBUG -std=gnu99
>> -ffast-math -Wall -Wunused-variable -Wwrite-strings -Wstrict-prototypes
>> -Wmissing-prototypes -fvisibility=hidden -DSWITCH_API_VISIBILITY=1
>> -DHAVE_VISIBILITY=1 -g -O2 -o make_at_dictionary make_at_dictionary.o
>> -L/usr/src/freeswitch/libs/tiff-3.8.2/libtiff
>> /usr/src/freeswitch/libs/tiff-3.8.2/libtiff/.libs/libtiff.a -ljpeg -lz -lm
>> -lc
>> ./make_at_dictionary: error while loading shared libraries: libjpeg.so.7:
>> cannot open shared object file: No such file or directory
>> make[8]: *** [at_interpreter_dictionary.h] Error 127
>> make[7]: *** [all] Error 2
>> make[6]: *** [all-recursive] Error 1
>> make[5]: *** [../../../../libs/spandsp/src/libspandsp.la] Error 2
>>
>> make[4]: *** [install] Error 1
>> make[3]: *** [mod_voipcodecs-install] Error 1
>>
>> make[2]: *** [install-recursive] Error 1
>>
>> Do you have idea about this one?
>>
>> Thanks!
>>
>>
>> On Thu, Dec 3, 2009 at 2:09 PM, Mathieu Rene <mrene_lists at avgs.ca> wrote:
>>
>>> Consider it fixed.
>>> Committed revision 15765.
>>>
>>> Mathieu Rene
>>> Avant-Garde Solutions Inc
>>> Office: + 1 (514) 664-1044 x100
>>> Cell: +1 (514) 664-1044 x200
>>> mrene at avgs.ca
>>>
>>>
>>>
>>>
>>> On 3-Dec-09, at 1:02 AM, Jingwei Yang wrote:
>>>
>>> Hi Guys,
>>>
>>> I got a compilation error of skypiax_protocol.c with the latest version
>>> r15764.
>>>
>>> Compiling skypiax_protocol.c...
>>> *cc1: warnings being treated as errors*
>>> skypiax_protocol.c: In function ???X11_errors_handler???:
>>> skypiax_protocol.c:1548: warning: ISO C90 forbids mixed declarations and
>>> code
>>> skypiax_protocol.c: In function ???skypiax_send_message???:
>>> skypiax_protocol.c:1582: warning: ISO C90 forbids mixed declarations and
>>> code
>>> skypiax_protocol.c: In function ???skypiax_do_skypeapi_thread_func???:
>>> skypiax_protocol.c:1726: warning: ISO C90 forbids mixed declarations and
>>> code
>>> skypiax_protocol.c:1758: warning: ISO C90 forbids mixed declarations and
>>> code
>>> make[5]: *** [skypiax_protocol.o] Error 1
>>> make[4]: *** [install] Error 1
>>> make[3]: *** [mod_skypiax-install] Error 1
>>> make[2]: *** [install-recursive] Error 1
>>>
>>> I personally checked the file and it shouldn't be a merge problem. Does
>>> anyone encounter this as well?
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100224/deafda57/attachment.html 

From ahmedmunir007 at gmail.com  Wed Feb 24 21:31:41 2010
From: ahmedmunir007 at gmail.com (Ahmed Munir)
Date: Thu, 25 Feb 2010 10:31:41 +0500
Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 44, Issue 217
In-Reply-To: <mailman.6837.1267029336.2495.freeswitch-users@lists.freeswitch.org>
References: <mailman.6837.1267029336.2495.freeswitch-users@lists.freeswitch.org>
Message-ID: <a60fed5e1002242131w43f622e1r3dd46d541868b1b6@mail.gmail.com>

Hi Tod,

After configuring ODBC connection using MySQL database issue isql *ODBC
Connection Name *on cli. If it connects successfully you can see the
database's tables what you've mentioned in odbc.ini file i.e.

isql mysql_fs (ODBC connection name)



---------- Forwarded message ----------
> From: "Todd" <phunk0000 at hotmail.com>
> To: <freeswitch-users at lists.freeswitch.org>
> Date: Wed, 24 Feb 2010 11:35:22 -0500
> Subject: Re: [Freeswitch-users] mod_nibblebill
>
> Yeah,  think I?m going to give it a shot from the beginning again and be
> very careful about install and config.  Thanks, I will keep you posted.
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Rupa
> Schomaker
> *Sent:* Wednesday, February 24, 2010 9:06 AM
> *To:* freeswitch-users
> *Subject:* Re: [Freeswitch-users] mod_nibblebill
>
>
>
> I am only passingly familiar with MySQL.   There must be a way for it to
> log all sql statements sent to it?
>
>
>
> Setting up postgresql would be the same (in broad terms) as mysql.  Install
> packages, create database/user/tables, populate data, configure odbc dsn,
> test.
>
> On Wed, Feb 24, 2010 at 7:03 AM, Todd <phunk0000 at hotmail.com> wrote:
>
> I am attempting to us MySQL.  I installed the spidermonkey mod, newest
> ODBC, compiled FS with ODBC, configured xml?s in FS?  not 100% sure I did it
> right though..followed wiki directions as close as possible.  What is the
> best way to verify the SQL is talking to MySQL? or perhaps the easiest way
> to switch  to postgresql?  Still kinda new to DB admin.  Thanks a ton.
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Rupa
> Schomaker
> *Sent:* Tuesday, February 23, 2010 6:38 PM
> *To:* freeswitch-users
> *Subject:* Re: [Freeswitch-users] mod_nibblebill
>
>
>
> what database backend are you using?  Have you verified the SQL is going to
> the right database backend?  I use mod_nibblebill against postgresql w/out
> problems.
>
> On Tue, Feb 23, 2010 at 9:42 AM, Meg Stroodle <phunk0000 at hotmail.com>
> wrote:
>
> Hello List!  I am trying to install mod_nibblebill on my FS installation.
> I get the following log entry form FS & nibblebill, but the database table I
> setup remains unchanged.  Any help in this matter would be greatly
> appreciated.  Following is an excerpt from the FS log:
>
>
>
> 2010-02-23 10:32:52.869149 [DEBUG] switch_rtp.c:2004 Correct ip/port
> confirmed.
>
> 2010-02-23 10:32:53.077162 [DEBUG] sofia.c:3727 Channel
> sofia/internal/3007 at 192.168.15.177 entering state [ready][200]
>
> 2010-02-23 10:32:53.149166 [DEBUG] switch_core_session.c:706 Send signal
> sofia/internal/sip:3008 at 192.168.15.176:21828 [BREAK]
>
> 2010-02-23 10:32:53.149166 [DEBUG] switch_core_session.c:706 Send signal
> sofia/internal/3007 at 192.168.15.177 [BREAK]
>
> 2010-02-23 10:32:53.149166 [DEBUG] switch_ivr_bridge.c:131 sofia/internal/
> sip:3008 at 192.168.15.176:21828 receive message [DISPLAY]
>
> 2010-02-23 10:32:53.169168 [DEBUG] switch_ivr_bridge.c:131
> sofia/internal/3007 at 192.168.15.177 receive message [DISPLAY]
>
> 2010-02-23 10:33:21.388949 [DEBUG] mod_nibblebill.c:529 Received request
> via SESSION_HEARTBEAT!
>
> 2010-02-23 10:33:21.388949 [DEBUG] mod_nibblebill.c:397 Attempting to bill
> at $1.0 per minute to account 3007
>
> 2010-02-23 10:33:21.388949 [INFO] mod_nibblebill.c:447 Beginning new
> billing on 3ec17ffc-c94e-46ff-bb0d-7ad0ef27bdab
>
> 2010-02-23 10:33:21.388949 [DEBUG] mod_nibblebill.c:455 28 seconds passed
> since last bill time of 2010-02-23 10:32:52
>
> 2010-02-23 10:33:21.388949 [DEBUG] mod_nibblebill.c:461 Billing $0.475997
> to 3007 (Call: 3ec17ffc-c94e-46ff-bb0d-7ad0ef27bdab / 0.000000 so far)
>
> 2010-02-23 10:33:51.508822 [DEBUG] mod_nibblebill.c:529 Received request
> via SESSION_HEARTBEAT!
>
> 2010-02-23 10:33:51.508822 [DEBUG] mod_nibblebill.c:397 Attempting to bill
> at $1.0 per minute to account 3007
>
> 2010-02-23 10:33:51.508822 [DEBUG] mod_nibblebill.c:449 Last successful
> billing time was
>
> 2010-02-23 10:33:51.508822 [DEBUG] mod_nibblebill.c:455 30 seconds passed
> since last bill time of 2010-02-23 10:33:21
>
> 2010-02-23 10:33:51.508822 [DEBUG] mod_nibblebill.c:461 Billing $0.501998
> to 3007 (Call: 3ec17ffc-c94e-46ff-bb0d-7ad0ef27bdab / 0.475997 so far)
>
> 2010-02-23 10:34:21.649708 [DEBUG] mod_nibblebill.c:529 Received request
> via SESSION_HEARTBEAT!
>
> 2010-02-23 10:34:21.649708 [DEBUG] mod_nibblebill.c:397 Attempting to bill
> at $1.0 per minute to account 3007
>
> 2010-02-23 10:34:21.649708 [DEBUG] mod_nibblebill.c:449 Last successful
> billing time was
>
> 2010-02-23 10:34:21.649708 [DEBUG] mod_nibblebill.c:455 30 seconds passed
> since last bill time of 2010-02-23 10:33:51
>
> 2010-02-23 10:34:21.649708 [DEBUG] mod_nibblebill.c:461 Billing $0.502348
> to 3007 (Call: 3ec17ffc-c94e-46ff-bb0d-7ad0ef27bdab / 0.977995 so far)
>
> 2010-02-23 10:34:26.254308 [NOTICE] sofia.c:329 Hangup
> sofia/internal/3007 at 192.168.15.177 [CS_EXECUTE] [NORMAL_CLEARING]
>
> 2010-02-23 10:34:26.254308 [DEBUG] switch_channel.c:1912 Send signal
> sofia/internal/3007 at 192.168.15.177 [KILL]
>
> 2010-02-23 10:34:26.254308 [DEBUG] switch_core_session.c:999 Send signal
> sofia/internal/3007 at 192.168.15.177 [BREAK]
>
> 2010-02-23 10:34:26.254308 [DEBUG] switch_core_state_machine.c:459
> sofia/internal/3007 at 192.168.15.177 thread mismatch skipping state handler.
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_ivr_bridge.c:470
> sofia/internal/3007 at 192.168.15.177 ending bridge by request from read
> function
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_ivr_bridge.c:563 BRIDGE THREAD
> DONE [sofia/internal/3007 at 192.168.15.177]
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_ivr_bridge.c:565 Send signal
> sofia/internal/sip:3008 at 192.168.15.176:21828 [BREAK]
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_ivr_bridge.c:520 sofia/internal/
> sip:3008 at 192.168.15.176:21828 receive message [UNBRIDGE]
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_session.c:645 Send signal
> sofia/internal/sip:3008 at 192.168.15.176:21828 [BREAK]
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_ivr_bridge.c:563 BRIDGE THREAD
> DONE [sofia/internal/sip:3008 at 192.168.15.176:21828]
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_ivr_bridge.c:565 Send signal
> sofia/internal/3007 at 192.168.15.177 [BREAK]
>
> 2010-02-23 10:34:26.269008 [NOTICE] switch_ivr_bridge.c:617 Hangup
> sofia/internal/sip:3008 at 192.168.15.176:21828 [CS_EXCHANGE_MEDIA]
> [NORMAL_CLEARING]
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_channel.c:1912 Send signal
> sofia/internal/sip:3008 at 192.168.15.176:21828 [KILL]
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_session.c:999 Send signal
> sofia/internal/sip:3008 at 192.168.15.176:21828 [BREAK]
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:488
> (sofia/internal/sip:3008 at 192.168.15.176:21828) State HANGUP
>
> 2010-02-23 10:34:26.269008 [DEBUG] mod_sofia.c:358 Channel sofia/internal/
> sip:3008 at 192.168.15.176:21828 hanging up, cause: NORMAL_CLEARING
>
> 2010-02-23 10:34:26.269008 [DEBUG] mod_sofia.c:400 Sending BYE to
> sofia/internal/sip:3008 at 192.168.15.176:21828
>
> 2010-02-23 10:34:26.269008 [DEBUG] mod_nibblebill.c:397 Attempting to bill
> at $2.30 per minute to account 3008
>
> 2010-02-23 10:34:26.269008 [INFO] mod_nibblebill.c:447 Beginning new
> billing on c1c42712-89e9-44c5-914f-11ed8f6aebb1
>
> 2010-02-23 10:34:26.269008 [DEBUG] mod_nibblebill.c:455 93 seconds passed
> since last bill time of 2010-02-23 10:32:52
>
> 2010-02-23 10:34:26.269008 [DEBUG] mod_nibblebill.c:461 Billing $3.582436
> to 3008 (Call: c1c42712-89e9-44c5-914f-11ed8f6aebb1 / 0.000000 so far)
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:46
> sofia/internal/sip:3008 at 192.168.15.176:21828 Standard HANGUP, cause:
> NORMAL_CLEARING
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:488
> (sofia/internal/sip:3008 at 192.168.15.176:21828) State HANGUP going to sleep
>
> 2010-02-23 10:34:26.269008 [INFO] switch_core_session.c:1108
> sofia/internal/sip:3008 at 192.168.15.176:21828 setting session heartbeat to
> 30 second(s).
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:351
> (sofia/internal/sip:3008 at 192.168.15.176:21828) State EXCHANGE_MEDIA going
> to sleep
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:314
> (sofia/internal/sip:3008 at 192.168.15.176:21828) Running State Change
> CS_HANGUP
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:465
> sofia/internal/sip:3008 at 192.168.15.176:21828 handler already called,
> skipping state handler.
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:333
> (sofia/internal/sip:3008 at 192.168.15.176:21828) State Change CS_HANGUP ->
> CS_REPORTING
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_session.c:999 Send signal
> sofia/internal/sip:3008 at 192.168.15.176:21828 [BREAK]
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:314
> (sofia/internal/sip:3008 at 192.168.15.176:21828) Running State Change
> CS_REPORTING
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:579
> (sofia/internal/sip:3008 at 192.168.15.176:21828) State REPORTING
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:53
> sofia/internal/sip:3008 at 192.168.15.176:21828 Standard REPORTING, cause:
> NORMAL_CLEARING
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:579
> (sofia/internal/sip:3008 at 192.168.15.176:21828) State REPORTING going to
> sleep
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:327
> (sofia/internal/sip:3008 at 192.168.15.176:21828) State Change CS_REPORTING
> -> CS_DESTROY
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_session.c:999 Send signal
> sofia/internal/sip:3008 at 192.168.15.176:21828 [BREAK]
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_session.c:1136 Session 6
> (sofia/internal/sip:3008 at 192.168.15.176:21828) Locked, Waiting on external
> entities
>
> 2010-02-23 10:34:26.272437 [DEBUG] switch_core_state_machine.c:348 (
> sofia/internal/3007 at 192.168.15.177) State EXECUTE going to sleep
>
> 2010-02-23 10:34:26.272437 [DEBUG] switch_core_state_machine.c:314 (
> sofia/internal/3007 at 192.168.15.177) Running State Change CS_HANGUP
>
> 2010-02-23 10:34:26.272437 [DEBUG] switch_core_state_machine.c:488 (
> sofia/internal/3007 at 192.168.15.177) State HANGUP
>
> 2010-02-23 10:34:26.272437 [DEBUG] mod_sofia.c:358 Channel
> sofia/internal/3007 at 192.168.15.177 hanging up, cause: NORMAL_CLEARING
>
> 2010-02-23 10:34:26.272437 [DEBUG] mod_nibblebill.c:397 Attempting to bill
> at $1.0 per minute to account 3007
>
> 2010-02-23 10:34:26.272437 [DEBUG] mod_nibblebill.c:449 Last successful
> billing time was
>
> 2010-02-23 10:34:26.272437 [DEBUG] mod_nibblebill.c:455 4 seconds passed
> since last bill time of 2010-02-23 10:34:21
>
> 2010-02-23 10:34:26.272437 [DEBUG] mod_nibblebill.c:461 Billing $0.077045
> to 3007 (Call: 3ec17ffc-c94e-46ff-bb0d-7ad0ef27bdab / 1.480343 so far)
>
> 2010-02-23 10:34:26.272437 [DEBUG] switch_core_state_machine.c:46
> sofia/internal/3007 at 192.168.15.177 Standard HANGUP, cause: NORMAL_CLEARING
>
>
>
> Anyhelp getting nibblebill to connect to the database would be greatly
> appreciated.  Thanks
>
>
>
>
>  ------------------------------
>
> Hotmail: Trusted email with Microsoft?s powerful SPAM protection. Sign up
> now. <http://clk.atdmt.com/GBL/go/201469226/direct/01/>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> --
> -Rupa
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Regards,

Ahmed Munir
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100225/29cb30e4/attachment-0001.html 

From moizchinoy at gmail.com  Wed Feb 24 22:04:46 2010
From: moizchinoy at gmail.com (Moiz Chinoy)
Date: Thu, 25 Feb 2010 10:04:46 +0400
Subject: [Freeswitch-users] Dingaling - external rtp supported?
In-Reply-To: <191c3a031002231131x19c9791cgd986a6c1be18e09a@mail.gmail.com>
References: <29b888f81002160835y6e290319ub0d04c088caa1b79@mail.gmail.com>
	<DEA39AD1-0EE9-473A-B78C-CDDF3F100578@freeswitch.org>
	<29b888f81002170318p63622f3fhc8d26ad0e85602d2@mail.gmail.com>
	<191c3a031002170538r3b029515i8a000812a70ad558@mail.gmail.com>
	<29b888f81002170741y3d13cb57ycb4436a372c4eded@mail.gmail.com>
	<191c3a031002171136q7424e9bai34379abb542daf93@mail.gmail.com>
	<29b888f81002210800v617bca9dk4bf1abf1bd791ee0@mail.gmail.com>
	<29b888f81002222359w6ae4cd8ej944d453a5b099af4@mail.gmail.com>
	<191c3a031002231131x19c9791cgd986a6c1be18e09a@mail.gmail.com>
Message-ID: <29b888f81002242204x171f4ab3la34d4ace604cd50a@mail.gmail.com>

I was using GuntTls-2.7.3 for windows. Now I am using GuntTls-2.9.9. I have
modified only gnutls.h, added following line:

typedef long ssize_t;

because otherwise it was giving errors...

What is the recommended version of the TLS lib for windows?

After upgrading the the GnuTls and freeswitch to rev 16806, I ran the
freeswitch with mod_dingalilg enabled. Once started, I issued just the
'shutdown' command on the console, exception happened.

......................
2010-02-25 09:45:29.795285 [CONSOLE] switch_loadable_module.c:1277 Stopping:
CORE_SOFTTIMER_MODULE
2010-02-25 09:45:29.810910 [CONSOLE] switch_time.c:780 Soft timer thread
exiting.
2010-02-25 09:45:29.810910 [NOTICE] switch_loadable_module.c:98 Thread ended
for CORE_SOFTTIMER_MODULE
2010-02-25 09:45:29.826535 [DEBUG] switch_loadable_module.c:456 Write lock
interface 'dingaling' to wait for existing references.
2010-02-25 09:45:29.826535 [NOTICE] switch_loadable_module.c:464 Deleting
Endpoint 'dingaling'
2010-02-25 09:45:29.826535 [NOTICE] switch_loadable_module.c:572 Deleting
API Function 'dl_debug'
2010-02-25 09:45:29.826535 [DEBUG] switch_loadable_module.c:574 Write lock
interface 'dl_debug' to wait for existing references.
2010-02-25 09:45:29.826535 [NOTICE] switch_loadable_module.c:572 Deleting
API Function 'dl_pres'
2010-02-25 09:45:29.826535 [DEBUG] switch_loadable_module.c:574 Write lock
interface 'dl_pres' to wait for existing references.
2010-02-25 09:45:29.826535 [NOTICE] switch_loadable_module.c:572 Deleting
API Function 'dl_logout'
2010-02-25 09:45:29.826535 [DEBUG] switch_loadable_module.c:574 Write lock
interface 'dl_logout' to wait for existing references.
2010-02-25 09:45:29.826535 [NOTICE] switch_loadable_module.c:572 Deleting
API Function 'dl_login'
2010-02-25 09:45:29.826535 [DEBUG] switch_loadable_module.c:574 Write lock
interface 'dl_login' to wait for existing references.
2010-02-25 09:45:29.826535 [NOTICE] switch_loadable_module.c:572 Deleting
API Function 'dingaling'
2010-02-25 09:45:29.826535 [DEBUG] switch_loadable_module.c:574 Write lock
interface 'dingaling' to wait for existing references.
2010-02-25 09:45:29.826535 [DEBUG] switch_loadable_module.c:710 Write lock
interface 'jingle' to wait for existing references.
2010-02-25 09:45:29.826535 [NOTICE] switch_loadable_module.c:719 Deleting
Chat interface 'jingle'
2010-02-25 09:45:29.826535 [CONSOLE] switch_loadable_module.c:1277 Stopping:
mod_dingaling
2010-02-25 09:45:31.185910 [DEBUG] libdingaling.c:1546 io error 2 7 retry in
1 second(s)
........................

And the code went in the stream.c...

int
iks_fd (iksparser *prs)
{
struct stream_data *data;

if (prs) {
data = iks_user_data (prs);
if (data) {
return (int) data->sock;
}
}
return -1;
}



On Tue, Feb 23, 2010 at 11:31 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:
> If you are modifying your build to add libgcrypt / libgnutls to win32, you
> have chosen an incompatible version of one of these libs.  We do not
support
> manually adding this modification to the code, you will need to find
someone
> else who has done it successfully to help you.
>
>
>
>
> On Tue, Feb 23, 2010 at 1:59 AM, Moiz Chinoy <moizchinoy at gmail.com> wrote:
>>
>> Moreover, if I gtalk client is on the same machine as FS and i have
>> following settings, FS crashes with the same mutex error.
>>
>> External Sip Profile has following lines:
>> ---------------------------------------------------------
>>   <param name="rtp-ip" value="$${local_ip_v4}"/>
>>   <param name="sip-ip" value="$${local_ip_v4}"/>
>>   <param name="ext-rtp-ip" value="auto-nat"/>
>>   <param name="ext-sip-ip" value="auto-nat"/>
>>
>> Jingle Client.xml has following lines:
>> -----------------------------------------------------
>>   <param name="rtp-ip" value="$${bind_server_ip}"/>
>>   <!--<param name="ext-rtp-ip" value="$${external_rtp_ip}"/>-->
>>   <!--<param name="disable-rtp-auto-adjust" value="true"/>-->
>>   <!--<param name="avatar" value="/path/to/tiny.jpg"/>-->
>>   <!--<param name="candidate-acl" value="wan.auto"/>-->
>>   <!--<param name="local-network-acl" value="localnet.auto"/>-->
>>
>> If I uncomment the following line in client.xml (Jingle profile)
>>   <!--<param name="local-network-acl" value="localnet.auto"/>-->
>> then exception does not happen.
>>
>> Is this a known issue or do I need to post it in JIRA?
>>
>> Tell me if more logs are needed...
>>
>>
>> On Sun, Feb 21, 2010 at 8:00 PM, Moiz Chinoy <moizchinoy at gmail.com>
wrote:
>> > Guys,
>> >
>> > To make things simple gtalk client is entirely on different network.
>> >
>> > Call comes from outside through external Sip profile.
>> >
>> > If gtalk answers the call after 3-4 rings both parties can hear each
>> > other.
>> > If gtalk answers the call after 2 rings both parties no one can hear
>> > each other.
>> > If gtalk answers the call immediately FS crashes.
>> >
>> > Attached is the screen shot of the error...
>> >
>> > Here is the FS log...
>> > --------------------------------
>> > http://pastebin.freeswitch.org/12197
>> >
>> > External Sip Profile has following lines:
>> > ---------------------------------------------------------
>> >    <param name="rtp-ip" value="$${local_ip_v4}"/>
>> >    <param name="sip-ip" value="$${local_ip_v4}"/>
>> >    <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
>> >    <param name="ext-sip-ip" value="$${external_sip_ip}"/>
>> >
>> > Jingle Client.xml has following lines:
>> > -----------------------------------------------------
>> >    <param name="rtp-ip" value="$${bind_server_ip}"/>
>> >    <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
>> >    <param name="disable-rtp-auto-adjust" value="true"/>
>> >    <!--<param name="avatar" value="/path/to/tiny.jpg"/>-->
>> >    <!--<param name="candidate-acl" value="wan.auto"/>-->
>> >    <!--<param name="local-network-acl" value="localnet.auto"/>-->
>> >
>> > Vars.xml has following lines:
>> > -------------------------------------------
>> > <X-PRE-PROCESS cmd="set"
>> > data="external_rtp_ip=stun:stun.freeswitch.org"/>
>> > <X-PRE-PROCESS cmd="set"
>> > data="external_sip_ip=stun:stun.freeswitch.org"/>
>> >
>> >
>> > Please advise me how can I provide more of the required data.
>> >
>> > On Wed, Feb 17, 2010 at 11:36 PM, Anthony Minessale
>> > <anthony.minessale at gmail.com> wrote:
>> >> you cant combine stun and gtalk and boxes in the same lan very easily
>> >> if you
>> >> do need to do that you will need to mess with
>> >>
>> >> <param name="candidate-acl" value="wan.auto"/>
>> >> <param name="local-network-acl" value="localnet.auto"/>
>> >>
>> >>
>> >>
>> >>
>> >>
>> >> On Wed, Feb 17, 2010 at 9:41 AM, Moiz Chinoy <moizchinoy at gmail.com>
>> >> wrote:
>> >>>
>> >>> Guys I am unable to produce the crash but now both parties cannot
hear
>> >>> each other!
>> >>>
>> >>> Vars.xml has following lines:
>> >>>  <X-PRE-PROCESS cmd="set"
>> >>> data="external_rtp_ip=stun:stun.freeswitch.org"/>
>> >>>  <X-PRE-PROCESS cmd="set"
>> >>> data="external_sip_ip=stun:stun.freeswitch.org"/>
>> >>>
>> >>> Jingle Client.xml has following lines:
>> >>>    <param name="rtp-ip" value="$${bind_server_ip}"/>
>> >>>    <!-- <param name="ext-rtp-ip" value="auto-nat"/> -->
>> >>>    <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
>> >>>    <param name="disable-rtp-auto-adjust" value="true"/>
>> >>>    <param name="local-network-acl" value="localnet.auto"/>
>> >>>
>> >>> On Wed, Feb 17, 2010 at 5:38 PM, Anthony Minessale
>> >>> <anthony.minessale at gmail.com> wrote:
>> >>> > Obtain a stack trace from the crash.
>> >>> >
>> >>> > On Feb 17, 2010 5:26 AM, "Moiz Chinoy" <moizchinoy at gmail.com>
wrote:
>> >>> >
>> >>> > Hi,
>> >>> >
>> >>> > FS rev: 16673
>> >>> > Platform: Windows
>> >>> >
>> >>> > More details:
>> >>> >
>> >>> > FS is behind NAT and machine is running a VPN connection.
>> >>> >
>> >>> > FS and GTalk client on the same machine:
>> >>> >
>> >>> >
>> >>> >
--------------------------------------------------------------------------------------------------
>> >>> > jingle profile client.xml has following line:
>> >>> > <param name="rtp-ip" value="$${bind_server_ip}"/>
>> >>> >
>> >>> > External SIP call is successfully bridged to GTalk client.
>> >>> >
>> >>> >
>> >>> > FS and GTalk client on the different machine:
>> >>> >
>> >>> >
>> >>> >
--------------------------------------------------------------------------------------------------
>> >>> > jingle profile client.xml has following lines:
>> >>> > <param name="rtp-ip" value="$${bind_server_ip}"/>
>> >>> >
>> >>> > <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
>> >>> >
>> >>> > As soon as external SIP call land and I try to bridge the call to
>> >>> > GTalk client, FS crashes.
>> >>> >
>> >>> >
>> >>> > NAT Details:
>> >>> > ---------------------------
>> >>> > I think my NAT does not support UpNP or PMP. The reason I say it
>> >>> > because when FS starts following message is displayed:
>> >>> >
>> >>> > 2010-02-17 09:15:42.809674 [INFO] switch_nat.c:409 Scanning for NAT
>> >>> > 2010-02-17 09:15:42.809674 [ERR] switch_nat.c:197 Error checking
for
>> >>> > PMP [init failed]
>> >>> > 2010-02-17 09:15:42.809674 [DEBUG] switch_nat.c:414 Checking for
>> >>> > UPnP
>> >>> > 2010-02-17 09:15:56.825388 [DEBUG] switch_nat.c:114 No
>> >>> > InternetGatewayDevice, using first entry as default
>> >>> > (http://192.168.16.17:50144/).
>> >>> > 2010-02-17 09:15:56.887889 [INFO] switch_nat.c:429 No PMP or UPnP
>> >>> > NAT
>> >>> > devices detected!
>> >>> >
>> >>> >
>> >>> >
>> >>> > On Tue, Feb 16, 2010 at 8:41 PM, Brian West <brian at freeswitch.org>
>> >>> > wrote:
>> >>> >> can you please update...
>> >>> >
>> >>> > _______________________________________________
>> >>> > FreeSWITCH-users mailing list
>> >>> > FreeSWITCH-users at lists.freeswitch.org
>> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> >
>> >>> > UNSUBSCRIBE:
http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> > http://www.freeswitch.org
>> >>> >
>> >>> >
>> >>>
>> >>>
>> >>>
>> >>> --
>> >>> Regards,
>> >>> Moiz Chinoy.
>> >>>
>> >>> _______________________________________________
>> >>> FreeSWITCH-users mailing list
>> >>> FreeSWITCH-users at lists.freeswitch.org
>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>
>> >>> UNSUBSCRIBE:
http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> http://www.freeswitch.org
>> >>
>> >>
>> >>
>> >> --
>> >> Anthony Minessale II
>> >>
>> >> FreeSWITCH http://www.freeswitch.org/
>> >> ClueCon http://www.cluecon.com/
>> >> Twitter: http://twitter.com/FreeSWITCH_wire
>> >>
>> >> AIM: anthm
>> >> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
>> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> >> IRC: irc.freenode.net #freeswitch
>> >>
>> >> FreeSWITCH Developer Conference
>> >> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>> >> iax:guest at conference.freeswitch.org/888
>> >> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> >> pstn:+19193869900
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:
http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >>
>> >
>> >
>> >
>> > --
>> > Regards,
>> > Moiz Chinoy.
>> >
>>
>>
>>
>> --
>> Regards,
>> Moiz Chinoy.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Regards,
Moiz Chinoy.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100225/892e5636/attachment-0001.html 

From gkuri at ieee.org  Wed Feb 24 22:48:53 2010
From: gkuri at ieee.org (Gabriel Kuri)
Date: Wed, 24 Feb 2010 22:48:53 -0800
Subject: [Freeswitch-users] [Freeswitch-dev] Call for help - adding
	information to the wiki: SIP ALG's
In-Reply-To: <87f2f3b91002242008y599f86f0t86f467f8c16b4a20@mail.gmail.com>
References: <87f2f3b91002242008y599f86f0t86f467f8c16b4a20@mail.gmail.com>
Message-ID: <8b1c9cda1002242248u5e39a4ecvda6a3644527f59bf@mail.gmail.com>

I can think of several devices that have serious SIP ALG issues, I'll
spend the time and add some of the devices I know about.

Do you want to limit the page to SIP ALG problems or anything entirely
stupid that routers seem to do that could possibly break VoIP? For
example, we've found several routers with broken DNS
resolvers/forwarders that don't know how to deal with SRV records, in
particular routers that run VxWorks internally. People relying on
routers running VxWorks to resolve SRV records could bang their head
on the wall trying to figure out why nothing is working, unless they
manually configure their devices to use other DNS servers (Stinksys'
newer G and N routers running VxWorks comes to mind).

Cheers,
Gabe

On Wed, Feb 24, 2010 at 8:08 PM, Michael Collins <msc at freeswitch.org> wrote:
> Hi all,
>
> I've just completed a new wiki page:
>
> http://wiki.freeswitch.org/wiki/ALG
>
> I would like all of you who have dealt with routers with SIP ALG's to submit
> your input. I would like to see this page have a list of how-to's for all of
> the popular routers. If we can make it easy for people to disable SIP ALG's
> then I think we can all save ourselves time and energy answering questions
> in IRC and the mailing lists. Please by all means add your knowledge here. I
> started with the 2wire 3800HGV that I got for my ATT Uverse service.
>
> If you have knowledge that you like to add to the wiki (on this subject or
> any other) but are not confident in your wiki editing skills then contact me
> off list and I will be happy to help you get up to speed. Editing your first
> wiki page is always the hardest... :)
>
> Thanks again for all of your help! By the way, today's community conference
> call was great. Please plan on attending next week and we'll talk about more
> great FreeSWITCH stuff. I will have the recording of Rupa discussing
> mod_limit up on line as soon as I can.
>
> Take care,
> Michael
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>


From jingwei.yang at gmail.com  Wed Feb 24 23:14:18 2010
From: jingwei.yang at gmail.com (Jingwei Yang)
Date: Thu, 25 Feb 2010 15:14:18 +0800
Subject: [Freeswitch-users] compilation error of skypiax_protocol.c
In-Reply-To: <ade741d61002242048v150cb025q23c4463d38c9bbb9@mail.gmail.com>
References: <13529f9d0912022202g3254517dscc635bfb9c8b4439@mail.gmail.com>
	<9387ED56-4CDE-4685-AFF7-98E930BBA1F9@avgs.ca>
	<13529f9d0912022233i368b84c7s9bd960eb1380b53e@mail.gmail.com>
	<13529f9d0912022243y700728d4l30c7eb4e3152d1c9@mail.gmail.com>
	<ade741d61002242048v150cb025q23c4463d38c9bbb9@mail.gmail.com>
Message-ID: <13529f9d1002242314h4e36badbgeba7cf5500253235@mail.gmail.com>

Thanks Mouncif, the os was reinstalled and the problem disappeared.

On Thu, Feb 25, 2010 at 12:48 PM, Mouncif Benniane <mouncifbb at gmail.com>wrote:

> you probably need: libjpeg-devel instead. just a thought.
>
>
>
>
> On Thu, Dec 3, 2009 at 1:43 AM, Jingwei Yang <jingwei.yang at gmail.com>wrote:
>
>> Not sure whether this error is due to the lack of libjpeg. I just double
>> checked, this library had been installed.
>>
>> Package libjpeg-6b-37.i386 already installed and latest version
>>
>>
>>
>> On Thu, Dec 3, 2009 at 2:33 PM, Jingwei Yang <jingwei.yang at gmail.com>wrote:
>>
>>> Hi Mathieu, thanks for the promptly reply. The error has been fixed.
>>> However, I encounter another one.
>>>
>>> gcc -I/usr/src/freeswitch/libs/tiff-3.8.2/libtiff -DNDEBUG -std=gnu99
>>> -ffast-math -Wall -Wunused-variable -Wwrite-strings -Wstrict-prototypes
>>> -Wmissing-prototypes -fvisibility=hidden -DSWITCH_API_VISIBILITY=1
>>> -DHAVE_VISIBILITY=1 -g -O2 -o make_at_dictionary make_at_dictionary.o
>>> -L/usr/src/freeswitch/libs/tiff-3.8.2/libtiff
>>> /usr/src/freeswitch/libs/tiff-3.8.2/libtiff/.libs/libtiff.a -ljpeg -lz -lm
>>> -lc
>>> ./make_at_dictionary: error while loading shared libraries: libjpeg.so.7:
>>> cannot open shared object file: No such file or directory
>>> make[8]: *** [at_interpreter_dictionary.h] Error 127
>>> make[7]: *** [all] Error 2
>>> make[6]: *** [all-recursive] Error 1
>>> make[5]: *** [../../../../libs/spandsp/src/libspandsp.la] Error 2
>>>
>>> make[4]: *** [install] Error 1
>>> make[3]: *** [mod_voipcodecs-install] Error 1
>>>
>>> make[2]: *** [install-recursive] Error 1
>>>
>>> Do you have idea about this one?
>>>
>>> Thanks!
>>>
>>>
>>> On Thu, Dec 3, 2009 at 2:09 PM, Mathieu Rene <mrene_lists at avgs.ca>wrote:
>>>
>>>> Consider it fixed.
>>>> Committed revision 15765.
>>>>
>>>> Mathieu Rene
>>>> Avant-Garde Solutions Inc
>>>> Office: + 1 (514) 664-1044 x100
>>>> Cell: +1 (514) 664-1044 x200
>>>> mrene at avgs.ca
>>>>
>>>>
>>>>
>>>>
>>>> On 3-Dec-09, at 1:02 AM, Jingwei Yang wrote:
>>>>
>>>> Hi Guys,
>>>>
>>>> I got a compilation error of skypiax_protocol.c with the latest version
>>>> r15764.
>>>>
>>>> Compiling skypiax_protocol.c...
>>>> *cc1: warnings being treated as errors*
>>>> skypiax_protocol.c: In function ???X11_errors_handler???:
>>>> skypiax_protocol.c:1548: warning: ISO C90 forbids mixed declarations and
>>>> code
>>>> skypiax_protocol.c: In function ???skypiax_send_message???:
>>>> skypiax_protocol.c:1582: warning: ISO C90 forbids mixed declarations and
>>>> code
>>>> skypiax_protocol.c: In function ???skypiax_do_skypeapi_thread_func???:
>>>> skypiax_protocol.c:1726: warning: ISO C90 forbids mixed declarations and
>>>> code
>>>> skypiax_protocol.c:1758: warning: ISO C90 forbids mixed declarations and
>>>> code
>>>> make[5]: *** [skypiax_protocol.o] Error 1
>>>> make[4]: *** [install] Error 1
>>>> make[3]: *** [mod_skypiax-install] Error 1
>>>> make[2]: *** [install-recursive] Error 1
>>>>
>>>> I personally checked the file and it shouldn't be a merge problem. Does
>>>> anyone encounter this as well?
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100225/66831f31/attachment.html 

From rm at callrica.co.za  Thu Feb 25 00:33:08 2010
From: rm at callrica.co.za (Roly Maz)
Date: Thu, 25 Feb 2010 10:33:08 +0200
Subject: [Freeswitch-users] SIP Trunk with Private Static IP?
Message-ID: <008401cab5f5$4795b910$d6c12b30$@co.za>

Hi Community 

 

 

My Provider provides the following info when they supply a SIP trunk:

 

.         A direct connection into their network. i.e. they provide private
IPs:

.         An IP address I must use for my FS box e.g. IP: 10. 42.0.66 MASK:
255.255.255.248 GW: 10. 42.0.68

.         An IP address for their SIP server 10.42.0.1

 

I have setup a dual homed FS box (Windows Server 2008, latest FS version)

 

NIC 1 - Internal LAN IP: 10.0.2.185 MASK: 255.255.255.0 GW: 10.0.2.253

NIC 2 - SIP Provider (WAN) IP: 10. 42.0.66 MASK: 255.255.255.248 GW: 10.
42.0.68

 

Windows complains about multiple gateways - which I ignore? I can ping
internal addresses  and the SIP Server 

 

When I fire up FS, I can register Xlite phones on my LAN. I can dial and
hear the test IVR (5000)

 

This means my Internal SIP Profile is ok. 

 

Now, how do i route a call out to the 10.42.01 SIP Server?

 

 Creating a gateway doesn't make sense, because I am not supplied a
username/password? 

 

Any pointers would be most appreciated, I am sure I am missing something
really simple.

 

Roland

 

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100225/d18b57dc/attachment.html 

From mcampbellsmith at gmail.com  Thu Feb 25 00:43:14 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Thu, 25 Feb 2010 19:43:14 +1100
Subject: [Freeswitch-users] [Freeswitch-dev] Call for help - adding
	information to the wiki: SIP ALG's
In-Reply-To: <8b1c9cda1002242248u5e39a4ecvda6a3644527f59bf@mail.gmail.com>
References: <87f2f3b91002242008y599f86f0t86f467f8c16b4a20@mail.gmail.com>
	<8b1c9cda1002242248u5e39a4ecvda6a3644527f59bf@mail.gmail.com>
Message-ID: <33c87fa31002250043l1edbb87dg2e7a58e879656f06@mail.gmail.com>

I had an issue with a Thomson SpeedTouch 530 router that was causing
authentication to fail... its the thread titled 'Forbidden using UDP,
works with TCP/TLS' that everyone said was caused by a broken ATA ...
lucky no one chipped in so I could buy a new one (as suggested by
Anthony) - it wouldn't have helped because the router was broken and
modifying the authentication parameters!

I'll update the wiki with the info

On Thu, Feb 25, 2010 at 5:48 PM, Gabriel Kuri <gkuri at ieee.org> wrote:
> I can think of several devices that have serious SIP ALG issues, I'll
> spend the time and add some of the devices I know about.
>
> Do you want to limit the page to SIP ALG problems or anything entirely
> stupid that routers seem to do that could possibly break VoIP? For
> example, we've found several routers with broken DNS
> resolvers/forwarders that don't know how to deal with SRV records, in
> particular routers that run VxWorks internally. People relying on
> routers running VxWorks to resolve SRV records could bang their head
> on the wall trying to figure out why nothing is working, unless they
> manually configure their devices to use other DNS servers (Stinksys'
> newer G and N routers running VxWorks comes to mind).
>
> Cheers,
> Gabe
>
> On Wed, Feb 24, 2010 at 8:08 PM, Michael Collins <msc at freeswitch.org> wrote:
>> Hi all,
>>
>> I've just completed a new wiki page:
>>
>> http://wiki.freeswitch.org/wiki/ALG
>>
>> I would like all of you who have dealt with routers with SIP ALG's to submit
>> your input. I would like to see this page have a list of how-to's for all of
>> the popular routers. If we can make it easy for people to disable SIP ALG's
>> then I think we can all save ourselves time and energy answering questions
>> in IRC and the mailing lists. Please by all means add your knowledge here. I
>> started with the 2wire 3800HGV that I got for my ATT Uverse service.
>>
>> If you have knowledge that you like to add to the wiki (on this subject or
>> any other) but are not confident in your wiki editing skills then contact me
>> off list and I will be happy to help you get up to speed. Editing your first
>> wiki page is always the hardest... :)
>>
>> Thanks again for all of your help! By the way, today's community conference
>> call was great. Please plan on attending next week and we'll talk about more
>> great FreeSWITCH stuff. I will have the recording of Rupa discussing
>> mod_limit up on line as soon as I can.
>>
>> Take care,
>> Michael
>>
>> _______________________________________________
>> FreeSWITCH-dev mailing list
>> FreeSWITCH-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From steveayre at gmail.com  Thu Feb 25 00:57:39 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 25 Feb 2010 08:57:39 +0000
Subject: [Freeswitch-users] SIP Trunk with Private Static IP?
In-Reply-To: <008401cab5f5$4795b910$d6c12b30$@co.za>
References: <008401cab5f5$4795b910$d6c12b30$@co.za>
Message-ID: <eafa772f1002250057l3de5b2abu54da8725cbb77281@mail.gmail.com>

Create two SIP profiles, each bound to one of your local IPs.

You may create a gateway on the profile for the SIP trunk IP for the
10.42.0.1 server, but this is optional.

You can then bridge calls via the SIP server using one of:
<action application="bridge" data="sofia/gateway/gatewayname"/>
<action application="bridge" data="sofia/profilename/number at 10.42.0.1"/>

The advantages of using a gateway are:
- supports authentication
- will monitor the gateway to detect if it goes down (so calls fail
instantly rather than after a timeout)

As for the default gateway, it is the IP you send via to reach IPs
that are not on a network you are connected directly to - you should
probably only have one set, and it should be the one you go via to
reach the Internet.

-Steve


On 25 February 2010 08:33, Roly Maz <rm at callrica.co.za> wrote:
> Hi Community
>
>
>
>
>
> My Provider provides the following info when they supply a SIP trunk:
>
>
>
> ????????? A direct connection into their network. i.e. they provide private
> IPs:
>
> ????????? An IP address I must use for my FS box e.g. IP: 10. 42.0.66 MASK:
> 255.255.255.248 GW: 10. 42.0.68
>
> ????????? An IP address for their SIP server 10.42.0.1
>
>
>
> I have setup a dual homed FS box (Windows Server 2008, latest FS version)
>
>
>
> NIC 1 ? Internal LAN IP: 10.0.2.185 MASK: 255.255.255.0 GW: 10.0.2.253
>
> NIC 2 ? SIP Provider (WAN) IP: 10. 42.0.66 MASK: 255.255.255.248 GW: 10.
> 42.0.68
>
>
>
> Windows complains about multiple gateways ? which I ignore? I can ping
> internal addresses ?and the SIP Server
>
>
>
> When I fire up FS, I can register Xlite phones on my LAN. I can dial and
> hear the test IVR (5000)
>
>
>
> This means my Internal SIP Profile is ok.
>
>
>
> Now, how do i route a call out to the 10.42.01 SIP Server?
>
>
>
> ?Creating a gateway doesn?t make sense, because I am not supplied a
> username/password?
>
>
>
> Any pointers would be most appreciated, I am sure I am missing something
> really simple.
>
>
>
> Roland
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From steveayre at gmail.com  Thu Feb 25 00:58:59 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 25 Feb 2010 08:58:59 +0000
Subject: [Freeswitch-users] SIP Trunk with Private Static IP?
In-Reply-To: <eafa772f1002250057l3de5b2abu54da8725cbb77281@mail.gmail.com>
References: <008401cab5f5$4795b910$d6c12b30$@co.za>
	<eafa772f1002250057l3de5b2abu54da8725cbb77281@mail.gmail.com>
Message-ID: <eafa772f1002250058i2cd7b30ai42f44d0ea1d6eb88@mail.gmail.com>

Gateways do not require usernames and passwords. You are required to
set the parameter, but if no authentication is needed they are ignored
so you can put anything in the field, so that is not a reason to avoid
them.

-Steve


On 25 February 2010 08:57, Steven Ayre <steveayre at gmail.com> wrote:
> Create two SIP profiles, each bound to one of your local IPs.
>
> You may create a gateway on the profile for the SIP trunk IP for the
> 10.42.0.1 server, but this is optional.
>
> You can then bridge calls via the SIP server using one of:
> <action application="bridge" data="sofia/gateway/gatewayname"/>
> <action application="bridge" data="sofia/profilename/number at 10.42.0.1"/>
>
> The advantages of using a gateway are:
> - supports authentication
> - will monitor the gateway to detect if it goes down (so calls fail
> instantly rather than after a timeout)
>
> As for the default gateway, it is the IP you send via to reach IPs
> that are not on a network you are connected directly to - you should
> probably only have one set, and it should be the one you go via to
> reach the Internet.
>
> -Steve
>
>
> On 25 February 2010 08:33, Roly Maz <rm at callrica.co.za> wrote:
>> Hi Community
>>
>>
>>
>>
>>
>> My Provider provides the following info when they supply a SIP trunk:
>>
>>
>>
>> ????????? A direct connection into their network. i.e. they provide private
>> IPs:
>>
>> ????????? An IP address I must use for my FS box e.g. IP: 10. 42.0.66 MASK:
>> 255.255.255.248 GW: 10. 42.0.68
>>
>> ????????? An IP address for their SIP server 10.42.0.1
>>
>>
>>
>> I have setup a dual homed FS box (Windows Server 2008, latest FS version)
>>
>>
>>
>> NIC 1 ? Internal LAN IP: 10.0.2.185 MASK: 255.255.255.0 GW: 10.0.2.253
>>
>> NIC 2 ? SIP Provider (WAN) IP: 10. 42.0.66 MASK: 255.255.255.248 GW: 10.
>> 42.0.68
>>
>>
>>
>> Windows complains about multiple gateways ? which I ignore? I can ping
>> internal addresses ?and the SIP Server
>>
>>
>>
>> When I fire up FS, I can register Xlite phones on my LAN. I can dial and
>> hear the test IVR (5000)
>>
>>
>>
>> This means my Internal SIP Profile is ok.
>>
>>
>>
>> Now, how do i route a call out to the 10.42.01 SIP Server?
>>
>>
>>
>> ?Creating a gateway doesn?t make sense, because I am not supplied a
>> username/password?
>>
>>
>>
>> Any pointers would be most appreciated, I am sure I am missing something
>> really simple.
>>
>>
>>
>> Roland
>>
>>
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>


From brian at microcomaustralia.com.au  Thu Feb 25 01:07:27 2010
From: brian at microcomaustralia.com.au (Brian May)
Date: Thu, 25 Feb 2010 20:07:27 +1100
Subject: [Freeswitch-users] Retrieving voicemail without entering user
	ID (extension)
In-Reply-To: <000f01cab5c7$d7e292f0$87a7b8d0$@com>
References: <093DD565390C1E4FB15D7B383E86BB05AFF6BC@Goose.personalcyberspace.net>
	<b39273531002241601m273ff333y2261d930dbcb3fa7@mail.gmail.com>
	<000f01cab5c7$d7e292f0$87a7b8d0$@com>
Message-ID: <3c5cf5261002250107l3a4ff2fan8803c622ce59021e@mail.gmail.com>

On 25 February 2010 14:08, Lars Zeb <larclap at yahoo.com> wrote:
> I used the extension below, but I think that Brian said it was too insecure.
> Being a total beginner, I removed the condition.

Too insecure for what? I think it really depends on the installation,
what the phone are for, where the phones are positioned, who has
access, etc.

I could imagine scenarios where being able to "walk up to anyone's
phone and retrieve their VM w/out authentication" might be considered
a feature. e.g. home office. At my home, under my asterisk setup, it
always seems to be up to me to delete the messages, because others
consider it too complicated to log in. Generally people are use to
being able to walk up to an answering machine, push a button, and
retrieve messages without any authentication.

Then again, making this the default configuration would be a bad idea.
People need to understand the consequences first.

Oh, just a minor nitpick, or possibly an opportunity for me to learn
<grin>. I see near the top that there is an export immediately after
the set. Is the set really needed? I thought the export would override
this? Why is export needed?
-- 
Brian May <brian at microcomaustralia.com.au>


From srinivas.ksvreddy at gmail.com  Thu Feb 25 01:24:01 2010
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Thu, 25 Feb 2010 14:54:01 +0530
Subject: [Freeswitch-users] freeswitch to gateway
Message-ID: <f8af5741002250124s5f3a200fp178d83a90945d73b@mail.gmail.com>

Hi Good afternoon everybody,

my freeswitch domain name is gw.proxy.com, i have registered 1000 to
freeswitch,  i have configured a gateway(gateway.com) to my freeswitch, i
want to make a call to gateway from 1000 to 1003 registered in gateway.com ,
how can i make call.

Thanks--
Srinivasula Reddy K
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100225/27a1fc0f/attachment.html 

From technical at ttnc.co.uk  Thu Feb 25 03:35:14 2010
From: technical at ttnc.co.uk (TTNC - Technical)
Date: Thu, 25 Feb 2010 11:35:14 +0000
Subject: [Freeswitch-users] When using bridge_answer_timeout,
	hangup_after_bridge isn't respected.
Message-ID: <88A1EC04-4060-442E-8DC3-9CD214D48C18@ttnc.co.uk>

I'm pretty sure this is a bug, I've already opened a jira: http://jira.freeswitch.org/browse/FSCORE-561

But I thought after I'd done it it'd probably be an idea to ask here first incase I'm missing something obvious... When using bridge_answer_timeout, hangup_after_bridge isn't respected. In as much as the aleg hangs up the call and it will go back and continue executing the dialplan, in my case - the lua script. 

Anyone got any ideas how else to force freeswitch to end the call after a hangup other than using hangup_after_bridge?

Thanks




From m.krivushin at imarto.net  Thu Feb 25 04:59:11 2010
From: m.krivushin at imarto.net (Mikhail Krivushin)
Date: Thu, 25 Feb 2010 18:59:11 +0600
Subject: [Freeswitch-users] Video pass problem
Message-ID: <5be734a51002250459wb974018ue1ffd7d7a88ace59@mail.gmail.com>

Hello! We have problem with pass video over FreeSWITCH. I tshark traf, and
see that we have 1280 video packets input, and only 560 passed to B leg.
Anyone can point me to right direction? I can send pcap file by request.

Most time I have black screen, and then one frame can appear, and I see
frozen picture pair minutes, and then other frozen picture. We have not
network issues, we have good video, when bypass FS. We have not perfomance
troubles to.

We have ubuntu 9.10 x64, and powerfull server board. uname:
Linux fs 2.6.31-17-generic #54-Ubuntu SMP Thu Dec 10 17:01:44 UTC 2009
x86_64 GNU/Linux

config:
  <X-PRE-PROCESS cmd="set" data="global_codec_prefs=BV32,BV16,speex at 8000h
@20i,speex at 16000h@20i,speex at 32000h
@20i,PCMU,PCMA,GSM,H264,H263,H263-1998,H263-2000,H261"/>
  <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=BV32,BV16,speex at 8000h
@20i,speex at 16000h@20i,speex at 32000h
@20i,PCMU,PCMA,GSM,H264,H263,H263-1998,H263-2000,H261"/>


-- 
? ?????????, ???????? ??????
?. ????? ???. +7 913 865 78 66
icq: 218 744 127
xmpp: KrivushinME at jabber.ru
skype: mkrivushin
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100225/f549fc79/attachment.html 

From brian at freeswitch.org  Thu Feb 25 05:33:17 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 25 Feb 2010 07:33:17 -0600
Subject: [Freeswitch-users] [Freeswitch-dev] Call for help - adding
	information to the wiki: SIP ALG's
In-Reply-To: <33c87fa31002250043l1edbb87dg2e7a58e879656f06@mail.gmail.com>
References: <87f2f3b91002242008y599f86f0t86f467f8c16b4a20@mail.gmail.com>
	<8b1c9cda1002242248u5e39a4ecvda6a3644527f59bf@mail.gmail.com>
	<33c87fa31002250043l1edbb87dg2e7a58e879656f06@mail.gmail.com>
Message-ID: <EC776797-0320-4A5B-B959-2D3A3BF77314@freeswitch.org>

I believe all SIP ALG's are broken.  :P

/b

On Feb 25, 2010, at 2:43 AM, Mark Campbell-Smith wrote:

> I had an issue with a Thomson SpeedTouch 530 router that was causing
> authentication to fail... its the thread titled 'Forbidden using UDP,
> works with TCP/TLS' that everyone said was caused by a broken ATA ...
> lucky no one chipped in so I could buy a new one (as suggested by
> Anthony) - it wouldn't have helped because the router was broken and
> modifying the authentication parameters!
> 
> I'll update the wiki with the info



From rm at callrica.co.za  Thu Feb 25 07:02:56 2010
From: rm at callrica.co.za (Roly Maz)
Date: Thu, 25 Feb 2010 17:02:56 +0200
Subject: [Freeswitch-users] SIP Trunk with Private Static IP?
In-Reply-To: <eafa772f1002250058i2cd7b30ai42f44d0ea1d6eb88@mail.gmail.com>
References: <008401cab5f5$4795b910$d6c12b30$@co.za>	<eafa772f1002250057l3de5b2abu54da8725cbb77281@mail.gmail.com>
	<eafa772f1002250058i2cd7b30ai42f44d0ea1d6eb88@mail.gmail.com>
Message-ID: <009b01cab62b$b8889c10$2999d430$@co.za>

Many thanks for your prompt reply and the help 

I removed the LAN GW and kept the WAN GW. 

I have modified the standard internal and external sip profiles accordingly

What is odd is that if i run a ping from the windows command line, I get a
reply from the SIP Server. However, if I setup a ping within FS, it fails. 

I am investigating...




-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Steven
Ayre
Sent: 25 February 2010 10:59 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] SIP Trunk with Private Static IP?

Gateways do not require usernames and passwords. You are required to
set the parameter, but if no authentication is needed they are ignored
so you can put anything in the field, so that is not a reason to avoid
them.

-Steve


On 25 February 2010 08:57, Steven Ayre <steveayre at gmail.com> wrote:
> Create two SIP profiles, each bound to one of your local IPs.
>
> You may create a gateway on the profile for the SIP trunk IP for the
> 10.42.0.1 server, but this is optional.
>
> You can then bridge calls via the SIP server using one of:
> <action application="bridge" data="sofia/gateway/gatewayname"/>
> <action application="bridge" data="sofia/profilename/number at 10.42.0.1"/>
>
> The advantages of using a gateway are:
> - supports authentication
> - will monitor the gateway to detect if it goes down (so calls fail
> instantly rather than after a timeout)
>
> As for the default gateway, it is the IP you send via to reach IPs
> that are not on a network you are connected directly to - you should
> probably only have one set, and it should be the one you go via to
> reach the Internet.
>
> -Steve
>
>
> On 25 February 2010 08:33, Roly Maz <rm at callrica.co.za> wrote:
>> Hi Community
>>
>>
>>
>>
>>
>> My Provider provides the following info when they supply a SIP trunk:
>>
>>
>>
>> ????????? A direct connection into their network. i.e. they provide
private
>> IPs:
>>
>> ????????? An IP address I must use for my FS box e.g. IP: 10. 42.0.66
MASK:
>> 255.255.255.248 GW: 10. 42.0.68
>>
>> ????????? An IP address for their SIP server 10.42.0.1
>>
>>
>>
>> I have setup a dual homed FS box (Windows Server 2008, latest FS version)
>>
>>
>>
>> NIC 1 ? Internal LAN IP: 10.0.2.185 MASK: 255.255.255.0 GW: 10.0.2.253
>>
>> NIC 2 ? SIP Provider (WAN) IP: 10. 42.0.66 MASK: 255.255.255.248 GW: 10.
>> 42.0.68
>>
>>
>>
>> Windows complains about multiple gateways ? which I ignore? I can ping
>> internal addresses ?and the SIP Server
>>
>>
>>
>> When I fire up FS, I can register Xlite phones on my LAN. I can dial and
>> hear the test IVR (5000)
>>
>>
>>
>> This means my Internal SIP Profile is ok.
>>
>>
>>
>> Now, how do i route a call out to the 10.42.01 SIP Server?
>>
>>
>>
>> ?Creating a gateway doesn?t make sense, because I am not supplied a
>> username/password?
>>
>>
>>
>> Any pointers would be most appreciated, I am sure I am missing something
>> really simple.
>>
>>
>>
>> Roland
>>
>>
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



From peter.olsson at visionutveckling.se  Thu Feb 25 07:08:18 2010
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Thu, 25 Feb 2010 16:08:18 +0100
Subject: [Freeswitch-users] "hold" tone when dialing into FS
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C557729ECBB@cooper>

I guess this is a really stupid question, but I can't find anything about it in my config files...

When dialing into the dialplan, and I just execute "answer" and then "park", I get a ring tone played for me. But I just can't find where this ring tone can be specified, it also seems to play if I execute answer and then sleep. I want to replace this ringtone with another sound, so that's why I'm asking...

Regards,

Peter Olsson

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100225/59f5c362/attachment-0001.html 

From phunk0000 at hotmail.com  Thu Feb 25 07:54:05 2010
From: phunk0000 at hotmail.com (Todd)
Date: Thu, 25 Feb 2010 10:54:05 -0500
Subject: [Freeswitch-users] mod_shout
Message-ID: <BLU110-DS76D3E972959D2ADAB8540A7400@phx.gbl>

Hey everybody.. I followed all the directions on the wiki for mod_shout, but
I still get the following in the load up log of FS..  Any ideas on how to
get this module to work?  Thanks..

 

2010-02-25 10:39:02.473513 [CRIT] switch_loadable_module.c:872 Error Loading
module /usr/local/freeswitch/mod/mod_shout.so

**/usr/local/freeswitch/mod/mod_shout.so: cannot open shared object file: No
such file or directory**

 

.

 

2010-02-25 10:39:03.471572 [ERR] switch_core_file.c:122 Invalid file format
[mp3] for [/usr/local/freeswitch/sounds/music/8000/06 - Free The Robots -
Session Two.mp3]!

2010-02-25 10:39:03.471572 [ERR] mod_local_stream.c:210 Can't open
/usr/local/freeswitch/sounds/music/8000/06 - Free The Robots - Session
Two.mp3

2010-02-25 10:39:04.473632 [ERR] switch_core_file.c:122 Invalid file format
[mp3] for [/usr/local/freeswitch/sounds/music/8000/01 - Free The Robots -
Listen To The Future.mp3]!

2010-02-25 10:39:04.473632 [ERR] mod_local_stream.c:210 Can't open
/usr/local/freeswitch/sounds/music/8000/01 - Free The Robots - Listen To The
Future.mp3

2010-02-25 10:39:04.473632 [ERR] mod_local_stream.c:210 Can't open
/usr/local/freeswitch/sounds/music/8000/01 - Free The Robots - Listen To The
Future.mp3

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100225/c147a66c/attachment.html 

From brian at freeswitch.org  Thu Feb 25 08:01:22 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 25 Feb 2010 10:01:22 -0600
Subject: [Freeswitch-users] mod_shout
In-Reply-To: <BLU110-DS76D3E972959D2ADAB8540A7400@phx.gbl>
References: <BLU110-DS76D3E972959D2ADAB8540A7400@phx.gbl>
Message-ID: <6A0E1E89-69A2-4E6F-959E-82A377D39107@freeswitch.org>

you need to build and load mod_shout... and you also need to stop hijacking threads... When you compose a message to the list Click NEW then input the list address then the subject and then type your body.  By all means DO NOT click reply, change the subject and delete the body.  That is HOW you hijack a thread... even the archives will list it as hijacked.

edit modules.conf in the src tree and uncomment the one about mod_shout... then make mod_shout-install

/b

On Feb 25, 2010, at 9:54 AM, Todd wrote:

> Hey everybody?. I followed all the directions on the wiki for mod_shout, but I still get the following in the load up log of FS?.  Any ideas on how to get this module to work?  Thanks..
>  
> 2010-02-25 10:39:02.473513 [CRIT] switch_loadable_module.c:872 Error Loading module /usr/local/freeswitch/mod/mod_shout.so
> **/usr/local/freeswitch/mod/mod_shout.so: cannot open shared object file: No such file or directory**
>  
> ?
>  
> 2010-02-25 10:39:03.471572 [ERR] switch_core_file.c:122 Invalid file format [mp3] for [/usr/local/freeswitch/sounds/music/8000/06 - Free The Robots - Session Two.mp3]!
> 2010-02-25 10:39:03.471572 [ERR] mod_local_stream.c:210 Can't open /usr/local/freeswitch/sounds/music/8000/06 - Free The Robots - Session Two.mp3
> 2010-02-25 10:39:04.473632 [ERR] switch_core_file.c:122 Invalid file format [mp3] for [/usr/local/freeswitch/sounds/music/8000/01 - Free The Robots - Listen To The Future.mp3]!
> 2010-02-25 10:39:04.473632 [ERR] mod_local_stream.c:210 Can't open /usr/local/freeswitch/sounds/music/8000/01 - Free The Robots - Listen To The Future.mp3
> 2010-02-25 10:39:04.473632 [ERR] mod_local_stream.c:210 Can't open /usr/local/freeswitch/sounds/music/8000/01 - Free The Robots - Listen To The Future.mp3
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100225/ab19953c/attachment.html 

From m.krivushin at imarto.net  Thu Feb 25 08:03:35 2010
From: m.krivushin at imarto.net (Mikhail Krivushin)
Date: Thu, 25 Feb 2010 22:03:35 +0600
Subject: [Freeswitch-users] mod_shout
In-Reply-To: <BLU110-DS76D3E972959D2ADAB8540A7400@phx.gbl>
References: <BLU110-DS76D3E972959D2ADAB8540A7400@phx.gbl>
Message-ID: <5be734a51002250803h7e3863bg428c205cf300cc8a@mail.gmail.com>

check permissions on /usr/local/freeswitch/mod/mod_shout.so
It must be redeable by FS user.

2010/2/25 Todd <phunk0000 at hotmail.com>

>  Hey everybody?. I followed all the directions on the wiki for mod_shout,
> but I still get the following in the load up log of FS?.  Any ideas on how
> to get this module to work?  Thanks..
>
>
>
> 2010-02-25 10:39:02.473513 [CRIT] switch_loadable_module.c:872 Error
> Loading module /usr/local/freeswitch/mod/mod_shout.so
>
> **/usr/local/freeswitch/mod/mod_shout.so: cannot open shared object file:
> No such file or directory**
>
>
>
> ?
>
>
>
> 2010-02-25 10:39:03.471572 [ERR] switch_core_file.c:122 Invalid file format
> [mp3] for [/usr/local/freeswitch/sounds/music/8000/06 - Free The Robots -
> Session Two.mp3]!
>
> 2010-02-25 10:39:03.471572 [ERR] mod_local_stream.c:210 Can't open
> /usr/local/freeswitch/sounds/music/8000/06 - Free The Robots - Session
> Two.mp3
>
> 2010-02-25 10:39:04.473632 [ERR] switch_core_file.c:122 Invalid file format
> [mp3] for [/usr/local/freeswitch/sounds/music/8000/01 - Free The Robots -
> Listen To The Future.mp3]!
>
> 2010-02-25 10:39:04.473632 [ERR] mod_local_stream.c:210 Can't open
> /usr/local/freeswitch/sounds/music/8000/01 - Free The Robots - Listen To The
> Future.mp3
>
> 2010-02-25 10:39:04.473632 [ERR] mod_local_stream.c:210 Can't open
> /usr/local/freeswitch/sounds/music/8000/01 - Free The Robots - Listen To The
> Future.mp3
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
? ?????????, ???????? ??????
?. ????? ???. +7 913 865 78 66
icq: 218 744 127
xmpp: KrivushinME at jabber.ru
skype: mkrivushin
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100225/1ab2215d/attachment.html 

From brian at freeswitch.org  Thu Feb 25 08:07:13 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 25 Feb 2010 10:07:13 -0600
Subject: [Freeswitch-users] mod_shout
In-Reply-To: <5be734a51002250803h7e3863bg428c205cf300cc8a@mail.gmail.com>
References: <BLU110-DS76D3E972959D2ADAB8540A7400@phx.gbl>
	<5be734a51002250803h7e3863bg428c205cf300cc8a@mail.gmail.com>
Message-ID: <E7216619-05EB-4285-AA8D-51326770F6D0@freeswitch.org>

I think it would have said something besides "No such file or directory" if that were the case.  Its not in the default compile.

/b

On Feb 25, 2010, at 10:03 AM, Mikhail Krivushin wrote:

> check permissions on /usr/local/freeswitch/mod/mod_shout.so
> It must be redeable by FS user.



From chris.chen2004 at gmail.com  Thu Feb 25 08:08:37 2010
From: chris.chen2004 at gmail.com (Chris Chen)
Date: Thu, 25 Feb 2010 11:08:37 -0500
Subject: [Freeswitch-users] mod_shout
In-Reply-To: <BLU110-DS76D3E972959D2ADAB8540A7400@phx.gbl>
References: <BLU110-DS76D3E972959D2ADAB8540A7400@phx.gbl>
Message-ID: <507898381002250808s75aca323yb43544513dcaa9f5@mail.gmail.com>

Your mod_shout is not loaded,
make sure you compiled the shout correctly, and load the shout in the
/usr/local/freeswitch/conf/auto_configs/modules.conf.xml

Chris


On Thu, Feb 25, 2010 at 10:54 AM, Todd <phunk0000 at hotmail.com> wrote:

>  Hey everybody?. I followed all the directions on the wiki for mod_shout,
> but I still get the following in the load up log of FS?.  Any ideas on how
> to get this module to work?  Thanks..
>
>
>
> 2010-02-25 10:39:02.473513 [CRIT] switch_loadable_module.c:872 Error
> Loading module /usr/local/freeswitch/mod/mod_shout.so
>
> **/usr/local/freeswitch/mod/mod_shout.so: cannot open shared object file:
> No such file or directory**
>
>
>
> ?
>
>
>
> 2010-02-25 10:39:03.471572 [ERR] switch_core_file.c:122 Invalid file format
> [mp3] for [/usr/local/freeswitch/sounds/music/8000/06 - Free The Robots -
> Session Two.mp3]!
>
> 2010-02-25 10:39:03.471572 [ERR] mod_local_stream.c:210 Can't open
> /usr/local/freeswitch/sounds/music/8000/06 - Free The Robots - Session
> Two.mp3
>
> 2010-02-25 10:39:04.473632 [ERR] switch_core_file.c:122 Invalid file format
> [mp3] for [/usr/local/freeswitch/sounds/music/8000/01 - Free The Robots -
> Listen To The Future.mp3]!
>
> 2010-02-25 10:39:04.473632 [ERR] mod_local_stream.c:210 Can't open
> /usr/local/freeswitch/sounds/music/8000/01 - Free The Robots - Listen To The
> Future.mp3
>
> 2010-02-25 10:39:04.473632 [ERR] mod_local_stream.c:210 Can't open
> /usr/local/freeswitch/sounds/music/8000/01 - Free The Robots - Listen To The
> Future.mp3
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100225/02d4013d/attachment-0001.html 

From phunk0000 at hotmail.com  Thu Feb 25 08:15:21 2010
From: phunk0000 at hotmail.com (Todd)
Date: Thu, 25 Feb 2010 11:15:21 -0500
Subject: [Freeswitch-users] mod_shout
In-Reply-To: <6A0E1E89-69A2-4E6F-959E-82A377D39107@freeswitch.org>
References: <BLU110-DS76D3E972959D2ADAB8540A7400@phx.gbl>
	<6A0E1E89-69A2-4E6F-959E-82A377D39107@freeswitch.org>
Message-ID: <BLU110-DS192C502EC1E6527CEC7657A7400@phx.gbl>

?  That's what I did thanks.  brand new email to
freeswitch-users at lists.freeswitch.org   didn't reply to anything.  I
followed all the directions in the mod_shout wiki as far as running
./configure and make; make install.. Are you talking about something other
than that?

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: Thursday, February 25, 2010 11:01 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] mod_shout

 

you need to build and load mod_shout... and you also need to stop hijacking
threads... When you compose a message to the list Click NEW then input the
list address then the subject and then type your body.  By all means DO NOT
click reply, change the subject and delete the body.  That is HOW you hijack
a thread... even the archives will list it as hijacked.

 

edit modules.conf in the src tree and uncomment the one about mod_shout...
then make mod_shout-install

 

/b

 

On Feb 25, 2010, at 9:54 AM, Todd wrote:





Hey everybody.. I followed all the directions on the wiki for mod_shout, but
I still get the following in the load up log of FS..  Any ideas on how to
get this module to work?  Thanks..

 

2010-02-25 10:39:02.473513 [CRIT] switch_loadable_module.c:872 Error Loading
module /usr/local/freeswitch/mod/mod_shout.so

**/usr/local/freeswitch/mod/mod_shout.so: cannot open shared object file: No
such file or directory**

 

.

 

2010-02-25 10:39:03.471572 [ERR] switch_core_file.c:122 Invalid file format
[mp3] for [/usr/local/freeswitch/sounds/music/8000/06 - Free The Robots -
Session Two.mp3]!

2010-02-25 10:39:03.471572 [ERR] mod_local_stream.c:210 Can't open
/usr/local/freeswitch/sounds/music/8000/06 - Free The Robots - Session
Two.mp3

2010-02-25 10:39:04.473632 [ERR] switch_core_file.c:122 Invalid file format
[mp3] for [/usr/local/freeswitch/sounds/music/8000/01 - Free The Robots -
Listen To The Future.mp3]!

2010-02-25 10:39:04.473632 [ERR] mod_local_stream.c:210 Can't open
/usr/local/freeswitch/sounds/music/8000/01 - Free The Robots - Listen To The
Future.mp3

2010-02-25 10:39:04.473632 [ERR] mod_local_stream.c:210 Can't open
/usr/local/freeswitch/sounds/music/8000/01 - Free The Robots - Listen To The
Future.mp3

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org <http://www.freeswitch.org/> 

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100225/5b94ab0d/attachment.html 

From brian at freeswitch.org  Thu Feb 25 08:19:54 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 25 Feb 2010 10:19:54 -0600
Subject: [Freeswitch-users] mod_shout
In-Reply-To: <BLU110-DS192C502EC1E6527CEC7657A7400@phx.gbl>
References: <BLU110-DS76D3E972959D2ADAB8540A7400@phx.gbl>
	<6A0E1E89-69A2-4E6F-959E-82A377D39107@freeswitch.org>
	<BLU110-DS192C502EC1E6527CEC7657A7400@phx.gbl>
Message-ID: <A3BF8C8C-82CB-4099-B008-9E049566C236@freeswitch.org>

Seems you got mixed in with another naughty thread hijacker :)  Its ok... Sorry for pinning it on you... ;)

/b

On Feb 25, 2010, at 10:15 AM, Todd wrote:

> ?  That?s what I did thanks?  brand new email to    freeswitch-users at lists.freeswitch.org   didn?t reply to anything.  I followed all the directions in the mod_shout wiki as far as running ./configure and make; make install?. Are you talking about something other than that?
>  

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100225/564b2243/attachment.html 

From phunk0000 at hotmail.com  Thu Feb 25 08:24:11 2010
From: phunk0000 at hotmail.com (Todd)
Date: Thu, 25 Feb 2010 11:24:11 -0500
Subject: [Freeswitch-users] mod_shout
In-Reply-To: <507898381002250808s75aca323yb43544513dcaa9f5@mail.gmail.com>
References: <BLU110-DS76D3E972959D2ADAB8540A7400@phx.gbl>
	<507898381002250808s75aca323yb43544513dcaa9f5@mail.gmail.com>
Message-ID: <BLU110-DS3D63E1802B946A5B42D11A7400@phx.gbl>

Followed the directions in the wiki to compile, already loaded in
auto_configs/modules.conf.xml and in freeswitch/modules.conf...   is there
some other way/directions  to compile?

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Chris
Chen
Sent: Thursday, February 25, 2010 11:09 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] mod_shout

 

Your mod_shout is not loaded,
make sure you compiled the shout correctly, and load the shout in the
/usr/local/freeswitch/conf/auto_configs/modules.conf.xml

Chris



On Thu, Feb 25, 2010 at 10:54 AM, Todd <phunk0000 at hotmail.com> wrote:

Hey everybody.. I followed all the directions on the wiki for mod_shout, but
I still get the following in the load up log of FS..  Any ideas on how to
get this module to work?  Thanks..

 

2010-02-25 10:39:02.473513 [CRIT] switch_loadable_module.c:872 Error Loading
module /usr/local/freeswitch/mod/mod_shout.so

**/usr/local/freeswitch/mod/mod_shout.so: cannot open shared object file: No
such file or directory**

 

.

 

2010-02-25 10:39:03.471572 [ERR] switch_core_file.c:122 Invalid file format
[mp3] for [/usr/local/freeswitch/sounds/music/8000/06 - Free The Robots -
Session Two.mp3]!

2010-02-25 10:39:03.471572 [ERR] mod_local_stream.c:210 Can't open
/usr/local/freeswitch/sounds/music/8000/06 - Free The Robots - Session
Two.mp3

2010-02-25 10:39:04.473632 [ERR] switch_core_file.c:122 Invalid file format
[mp3] for [/usr/local/freeswitch/sounds/music/8000/01 - Free The Robots -
Listen To The Future.mp3]!

2010-02-25 10:39:04.473632 [ERR] mod_local_stream.c:210 Can't open
/usr/local/freeswitch/sounds/music/8000/01 - Free The Robots - Listen To The
Future.mp3

2010-02-25 10:39:04.473632 [ERR] mod_local_stream.c:210 Can't open
/usr/local/freeswitch/sounds/music/8000/01 - Free The Robots - Listen To The
Future.mp3


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100225/93ee5b86/attachment.html 

From brian at freeswitch.org  Thu Feb 25 08:31:28 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 25 Feb 2010 10:31:28 -0600
Subject: [Freeswitch-users] mod_shout
In-Reply-To: <BLU110-DS3D63E1802B946A5B42D11A7400@phx.gbl>
References: <BLU110-DS76D3E972959D2ADAB8540A7400@phx.gbl>
	<507898381002250808s75aca323yb43544513dcaa9f5@mail.gmail.com>
	<BLU110-DS3D63E1802B946A5B42D11A7400@phx.gbl>
Message-ID: <A81F3F31-7D03-4F4E-83CF-16F6238AA3D4@freeswitch.org>

While you're going down this road i'm going to highly recommend you turn around and go back to wav files.... MP3 is overly CPU hungry and if by chance you get some MP3 data thats invalid you have a chance of crashing the decoder...

/b

On Feb 25, 2010, at 10:24 AM, Todd wrote:

> Followed the directions in the wiki to compile, already loaded in auto_configs/modules.conf.xml and in freeswitch/modules.conf?..   is there some other way/directions  to compile?
>  

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100225/05d82da1/attachment-0001.html 

From jonas.gauffin at gmail.com  Thu Feb 25 08:36:00 2010
From: jonas.gauffin at gmail.com (Jonas Gauffin)
Date: Thu, 25 Feb 2010 17:36:00 +0100
Subject: [Freeswitch-users] bind_meta_app
Message-ID: <d021d3be1002250836p7010371epf943f5d68e77d5bf@mail.gmail.com>

Hello,

I'm trying to use bind_meta_app together with variables defined by me in the
original dial plan.
The problem is that they doesn't seem to follow the channel into the dial
plan when the meta application is running.

i.e.

Dialing from an external number:
<document type="freeswitch/xml">
        <section name="dialplan" description="ModCurlDialplan">
                <context name="public">
                        <extension name="dynamic">
                                <condition field="destination_number"
expression="0500650662">
                                        <action application="set"
data="hangup_after_bridge=true"/>
                                        <action application="set"
data="continue_on_fail=true"/>
                                        <action application="set"
data="gate_caller_site_id=3"/>
                                        <action application="bind_meta_app"
data="1 b s execute_extension::dx XML default"/>
                                        <action application="bind_meta_app"
data="3 b s execute_extension::cf XML default"/>
                                        <action application="set"
data="gate_bill_extension=95" />
                                        <action application="set"
data="gate_destination_user=6" />
                                        <action application="ring_ready"/>
                                        <action application="bridge"
data="[leg_timeout=5]sofia/internal/u1000006" />
                                        <action application="set"
data="gate_ivr=voicemail" />
                                        <action application="javascript"
data="voicemail.js customer 3 6" />
                                <action application="hangup"
data="NO_ANSWER"/>

                        </condition>
                </extension>
        </context>
</section>
</document>

Destination user pressed *1

Freeswitch sends second request through mod_curl to my server. My
variable_gate_XXXXX variables is not defined.

Regards,
  Jonas
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100225/8b21de39/attachment.html 

From phunk0000 at hotmail.com  Thu Feb 25 08:40:13 2010
From: phunk0000 at hotmail.com (Todd)
Date: Thu, 25 Feb 2010 11:40:13 -0500
Subject: [Freeswitch-users] mod_shout
In-Reply-To: <A81F3F31-7D03-4F4E-83CF-16F6238AA3D4@freeswitch.org>
References: <BLU110-DS76D3E972959D2ADAB8540A7400@phx.gbl>	<507898381002250808s75aca323yb43544513dcaa9f5@mail.gmail.com>	<BLU110-DS3D63E1802B946A5B42D11A7400@phx.gbl>
	<A81F3F31-7D03-4F4E-83CF-16F6238AA3D4@freeswitch.org>
Message-ID: <BLU110-DS15B7839071CA1439C6ADF7A7400@phx.gbl>

Good to know.  I was really just trying to make it work.. Don't need it for
anything

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: Thursday, February 25, 2010 11:31 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] mod_shout

 

While you're going down this road i'm going to highly recommend you turn
around and go back to wav files.... MP3 is overly CPU hungry and if by
chance you get some MP3 data thats invalid you have a chance of crashing the
decoder...

 

/b

 

On Feb 25, 2010, at 10:24 AM, Todd wrote:





Followed the directions in the wiki to compile, already loaded in
auto_configs/modules.conf.xml and in freeswitch/modules.conf...   is there
some other way/directions  to compile?

 

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100225/525b67b3/attachment.html 

From anthony.minessale at gmail.com  Thu Feb 25 08:43:38 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 25 Feb 2010 10:43:38 -0600
Subject: [Freeswitch-users] bind_meta_app
In-Reply-To: <d021d3be1002250836p7010371epf943f5d68e77d5bf@mail.gmail.com>
References: <d021d3be1002250836p7010371epf943f5d68e77d5bf@mail.gmail.com>
Message-ID: <191c3a031002250843l70761215q2cc98b4b2e2d3660@mail.gmail.com>

because you are executing the app on the B leg.
you need to set the vars on *that* channel if you want to see them.


On Thu, Feb 25, 2010 at 10:36 AM, Jonas Gauffin <jonas.gauffin at gmail.com>wrote:

> Hello,
>
> I'm trying to use bind_meta_app together with variables defined by me in
> the original dial plan.
> The problem is that they doesn't seem to follow the channel into the dial
> plan when the meta application is running.
>
> i.e.
>
> Dialing from an external number:
> <document type="freeswitch/xml">
>         <section name="dialplan" description="ModCurlDialplan">
>                 <context name="public">
>                         <extension name="dynamic">
>                                 <condition field="destination_number"
> expression="0500650662">
>                                         <action application="set"
> data="hangup_after_bridge=true"/>
>                                         <action application="set"
> data="continue_on_fail=true"/>
>                                         <action application="set"
> data="gate_caller_site_id=3"/>
>                                         <action application="bind_meta_app"
> data="1 b s execute_extension::dx XML default"/>
>                                         <action application="bind_meta_app"
> data="3 b s execute_extension::cf XML default"/>
>                                         <action application="set"
> data="gate_bill_extension=95" />
>                                         <action application="set"
> data="gate_destination_user=6" />
>                                         <action application="ring_ready"/>
>                                         <action application="bridge"
> data="[leg_timeout=5]sofia/internal/u1000006" />
>                                         <action application="set"
> data="gate_ivr=voicemail" />
>                                         <action application="javascript"
> data="voicemail.js customer 3 6" />
>                                 <action application="hangup"
> data="NO_ANSWER"/>
>
>                         </condition>
>                 </extension>
>         </context>
> </section>
> </document>
>
> Destination user pressed *1
>
> Freeswitch sends second request through mod_curl to my server. My
> variable_gate_XXXXX variables is not defined.
>
> Regards,
>   Jonas
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100225/fe4d1f87/attachment.html 

From anthony.minessale at gmail.com  Thu Feb 25 08:44:21 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 25 Feb 2010 10:44:21 -0600
Subject: [Freeswitch-users] bind_meta_app
In-Reply-To: <191c3a031002250843l70761215q2cc98b4b2e2d3660@mail.gmail.com>
References: <d021d3be1002250836p7010371epf943f5d68e77d5bf@mail.gmail.com>
	<191c3a031002250843l70761215q2cc98b4b2e2d3660@mail.gmail.com>
Message-ID: <191c3a031002250844k328a8b1fh42237da31e53d466@mail.gmail.com>

btw,

you can set the variable export_vars=<comma sep list of var names>
to copy all the variables to any B leg that may be spawned.


On Thu, Feb 25, 2010 at 10:43 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> because you are executing the app on the B leg.
> you need to set the vars on *that* channel if you want to see them.
>
>
> On Thu, Feb 25, 2010 at 10:36 AM, Jonas Gauffin <jonas.gauffin at gmail.com>wrote:
>
>> Hello,
>>
>> I'm trying to use bind_meta_app together with variables defined by me in
>> the original dial plan.
>> The problem is that they doesn't seem to follow the channel into the dial
>> plan when the meta application is running.
>>
>> i.e.
>>
>> Dialing from an external number:
>> <document type="freeswitch/xml">
>>         <section name="dialplan" description="ModCurlDialplan">
>>                 <context name="public">
>>                         <extension name="dynamic">
>>                                 <condition field="destination_number"
>> expression="0500650662">
>>                                         <action application="set"
>> data="hangup_after_bridge=true"/>
>>                                         <action application="set"
>> data="continue_on_fail=true"/>
>>                                         <action application="set"
>> data="gate_caller_site_id=3"/>
>>                                         <action
>> application="bind_meta_app" data="1 b s execute_extension::dx XML default"/>
>>                                         <action
>> application="bind_meta_app" data="3 b s execute_extension::cf XML default"/>
>>                                         <action application="set"
>> data="gate_bill_extension=95" />
>>                                         <action application="set"
>> data="gate_destination_user=6" />
>>                                         <action application="ring_ready"/>
>>                                         <action application="bridge"
>> data="[leg_timeout=5]sofia/internal/u1000006" />
>>                                         <action application="set"
>> data="gate_ivr=voicemail" />
>>                                         <action application="javascript"
>> data="voicemail.js customer 3 6" />
>>                                 <action application="hangup"
>> data="NO_ANSWER"/>
>>
>>                         </condition>
>>                 </extension>
>>         </context>
>> </section>
>> </document>
>>
>> Destination user pressed *1
>>
>> Freeswitch sends second request through mod_curl to my server. My
>> variable_gate_XXXXX variables is not defined.
>>
>> Regards,
>>   Jonas
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100225/a2a3eddd/attachment-0001.html 

From jonas.gauffin at gmail.com  Thu Feb 25 08:56:05 2010
From: jonas.gauffin at gmail.com (Jonas Gauffin)
Date: Thu, 25 Feb 2010 17:56:05 +0100
Subject: [Freeswitch-users] bind_meta_app
In-Reply-To: <191c3a031002250844k328a8b1fh42237da31e53d466@mail.gmail.com>
References: <d021d3be1002250836p7010371epf943f5d68e77d5bf@mail.gmail.com>
	<191c3a031002250843l70761215q2cc98b4b2e2d3660@mail.gmail.com>
	<191c3a031002250844k328a8b1fh42237da31e53d466@mail.gmail.com>
Message-ID: <d021d3be1002250856t56e2df49j63733b4b8f3bf6e6@mail.gmail.com>

Thanks for the quick answer =)

On Thu, Feb 25, 2010 at 5:44 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> btw,
>
> you can set the variable export_vars=<comma sep list of var names>
> to copy all the variables to any B leg that may be spawned.
>
>
>
> On Thu, Feb 25, 2010 at 10:43 AM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> because you are executing the app on the B leg.
>> you need to set the vars on *that* channel if you want to see them.
>>
>>
>> On Thu, Feb 25, 2010 at 10:36 AM, Jonas Gauffin <jonas.gauffin at gmail.com>wrote:
>>
>>> Hello,
>>>
>>> I'm trying to use bind_meta_app together with variables defined by me in
>>> the original dial plan.
>>> The problem is that they doesn't seem to follow the channel into the dial
>>> plan when the meta application is running.
>>>
>>> i.e.
>>>
>>> Dialing from an external number:
>>> <document type="freeswitch/xml">
>>>         <section name="dialplan" description="ModCurlDialplan">
>>>                 <context name="public">
>>>                         <extension name="dynamic">
>>>                                 <condition field="destination_number"
>>> expression="0500650662">
>>>                                         <action application="set"
>>> data="hangup_after_bridge=true"/>
>>>                                         <action application="set"
>>> data="continue_on_fail=true"/>
>>>                                         <action application="set"
>>> data="gate_caller_site_id=3"/>
>>>                                         <action
>>> application="bind_meta_app" data="1 b s execute_extension::dx XML default"/>
>>>                                         <action
>>> application="bind_meta_app" data="3 b s execute_extension::cf XML default"/>
>>>                                         <action application="set"
>>> data="gate_bill_extension=95" />
>>>                                         <action application="set"
>>> data="gate_destination_user=6" />
>>>                                         <action
>>> application="ring_ready"/>
>>>                                         <action application="bridge"
>>> data="[leg_timeout=5]sofia/internal/u1000006" />
>>>                                         <action application="set"
>>> data="gate_ivr=voicemail" />
>>>                                         <action application="javascript"
>>> data="voicemail.js customer 3 6" />
>>>                                 <action application="hangup"
>>> data="NO_ANSWER"/>
>>>
>>>                         </condition>
>>>                 </extension>
>>>         </context>
>>> </section>
>>> </document>
>>>
>>> Destination user pressed *1
>>>
>>> Freeswitch sends second request through mod_curl to my server. My
>>> variable_gate_XXXXX variables is not defined.
>>>
>>> Regards,
>>>   Jonas
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100225/61619343/attachment.html 

From frank at carmickle.com  Thu Feb 25 08:58:28 2010
From: frank at carmickle.com (Frank Carmickle)
Date: Thu, 25 Feb 2010 11:58:28 -0500
Subject: [Freeswitch-users] "hold" tone when dialing into FS
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C557729ECBB@cooper>
References: <549CFEF87AEDE841A38E9D15EAB4C04C557729ECBB@cooper>
Message-ID: <20100225165828.GF9832@base.carmickle.com>

On Thu, Feb 25, Peter Olsson wrote:
> I guess this is a really stupid question, but I can't find anything about it in my config files...
> 
> When dialing into the dialplan, and I just execute "answer" and then "park", I get a ring tone played for me. But I just can't find where this ring tone can be specified, it also seems to play if I execute answer and then sleep. I want to replace this ringtone with another sound, so that's why I'm asking...

<action application="set" data="transfer_ringback=$${hold_music}"/>

--FC


From frank at carmickle.com  Thu Feb 25 09:09:08 2010
From: frank at carmickle.com (Frank Carmickle)
Date: Thu, 25 Feb 2010 12:09:08 -0500
Subject: [Freeswitch-users] freeswitch to gateway
In-Reply-To: <f8af5741002250124s5f3a200fp178d83a90945d73b@mail.gmail.com>
References: <f8af5741002250124s5f3a200fp178d83a90945d73b@mail.gmail.com>
Message-ID: <20100225170907.GG9832@base.carmickle.com>


On Thu, Feb 25, srinivasula reddy wrote:
> Hi Good afternoon everybody,
> 
> my freeswitch domain name is gw.proxy.com, i have registered 1000 to
> freeswitch,  i have configured a gateway(gateway.com) to my freeswitch, i
> want to make a call to gateway from 1000 to 1003 registered in gateway.com ,
> how can i make call.

Add an extension that matches on 1003 and bridge to it
<action application="bridge" data="sofia/gateway/gatewayname/$1"/>

--FC


From peter.olsson at visionutveckling.se  Thu Feb 25 09:10:16 2010
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Thu, 25 Feb 2010 18:10:16 +0100
Subject: [Freeswitch-users] "hold" tone when dialing into FS
In-Reply-To: <20100225165828.GF9832@base.carmickle.com>
References: <549CFEF87AEDE841A38E9D15EAB4C04C557729ECBB@cooper>
	<20100225165828.GF9832@base.carmickle.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C557729ED17@cooper>

Thanks,

However, I figured out that it's probably caused by an Asterisk server that I dial through, when dialing this number. When dialing directly to FS from a SIP phone it's silent (as I expected it to be), but when dialing through a PRI connection on Asterisk, using SIP to FS, the Asterisk seems to generate this sound on the PRI card (while not receiving "real" RTP data from FS).

Sorry for bothering the wrong people :)

/Peter


-----Ursprungligt meddelande-----
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Frank Carmickle
Skickat: den 25 februari 2010 17:58
Till: freeswitch-users at lists.freeswitch.org
?mne: Re: [Freeswitch-users] "hold" tone when dialing into FS

On Thu, Feb 25, Peter Olsson wrote:
> I guess this is a really stupid question, but I can't find anything about it in my config files...
> 
> When dialing into the dialplan, and I just execute "answer" and then "park", I get a ring tone played for me. But I just can't find where this ring tone can be specified, it also seems to play if I execute answer and then sleep. I want to replace this ringtone with another sound, so that's why I'm asking...

<action application="set" data="transfer_ringback=$${hold_music}"/>

--FC

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

!DSPAM:4b86ae0132931189169766!



From lists at redbonez.net  Thu Feb 25 09:18:23 2010
From: lists at redbonez.net (Adam Ford)
Date: Thu, 25 Feb 2010 10:18:23 -0700
Subject: [Freeswitch-users] Retrieving voicemail without entering
	user	ID (extension)
In-Reply-To: <3c5cf5261002250107l3a4ff2fan8803c622ce59021e@mail.gmail.com>
References: <093DD565390C1E4FB15D7B383E86BB05AFF6BC@Goose.personalcyberspace.net>	<b39273531002241601m273ff333y2261d930dbcb3fa7@mail.gmail.com>	<000f01cab5c7$d7e292f0$87a7b8d0$@com>
	<3c5cf5261002250107l3a4ff2fan8803c622ce59021e@mail.gmail.com>
Message-ID: <016c01cab63e$8a4f97a0$9eeec6e0$@net>

I think many of you are missing the fact that he said he still wanted to
have to enter the mailbox password. He just didn't want people to have to
enter the mailbox ID AND the mailbox password.  Still plenty secure as long
as you don't have a default voicemail password for all extensions.

-Adam

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
May
Sent: Thursday, February 25, 2010 2:07 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Retrieving voicemail without entering user
ID (extension)

On 25 February 2010 14:08, Lars Zeb <larclap at yahoo.com> wrote:
> I used the extension below, but I think that Brian said it was too
insecure.
> Being a total beginner, I removed the condition.

Too insecure for what? I think it really depends on the installation,
what the phone are for, where the phones are positioned, who has
access, etc.

I could imagine scenarios where being able to "walk up to anyone's
phone and retrieve their VM w/out authentication" might be considered
a feature. e.g. home office. At my home, under my asterisk setup, it
always seems to be up to me to delete the messages, because others
consider it too complicated to log in. Generally people are use to
being able to walk up to an answering machine, push a button, and
retrieve messages without any authentication.

Then again, making this the default configuration would be a bad idea.
People need to understand the consequences first.

Oh, just a minor nitpick, or possibly an opportunity for me to learn
<grin>. I see near the top that there is an export immediately after
the set. Is the set really needed? I thought the export would override
this? Why is export needed?
-- 
Brian May <brian at microcomaustralia.com.au>

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



From m.sobkow at marketelsystems.com  Thu Feb 25 09:22:07 2010
From: m.sobkow at marketelsystems.com (Mark Sobkow)
Date: Thu, 25 Feb 2010 11:22:07 -0600
Subject: [Freeswitch-users] mod_erlang_event
In-Reply-To: <20100223224902.GB1751@hijacked.us>
References: <4B8441FE.80506@marketelsystems.com>
	<20100223224902.GB1751@hijacked.us>
Message-ID: <4B86B1BF.50809@marketelsystems.com>

An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100225/bce9e99e/attachment-0001.html 

From lists at redbonez.net  Thu Feb 25 09:26:18 2010
From: lists at redbonez.net (Adam Ford)
Date: Thu, 25 Feb 2010 10:26:18 -0700
Subject: [Freeswitch-users] [Freeswitch-dev] Call for help -
	adding	information to the wiki: SIP ALG's
In-Reply-To: <EC776797-0320-4A5B-B959-2D3A3BF77314@freeswitch.org>
References: <87f2f3b91002242008y599f86f0t86f467f8c16b4a20@mail.gmail.com>	<8b1c9cda1002242248u5e39a4ecvda6a3644527f59bf@mail.gmail.com>	<33c87fa31002250043l1edbb87dg2e7a58e879656f06@mail.gmail.com>
	<EC776797-0320-4A5B-B959-2D3A3BF77314@freeswitch.org>
Message-ID: <017001cab63f$a51397c0$ef3ac740$@net>

I wish you would have told me that back when I was trying to solve my
authentication issue that I thought was being caused by using rport. :P

You just told me to upgrade to 1.0.5, when a $%$#! Sonicwall SIP ALG would
have been more helpful, haha. As it turned out it was the Sonicwall SIP
transformations, and switching to TCP SIP resolved it. 

I have added my Sonicwall experience to the wiki page.

-Adam


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: Thursday, February 25, 2010 6:33 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] [Freeswitch-dev] Call for help - adding
information to the wiki: SIP ALG's

I believe all SIP ALG's are broken.  :P

/b

On Feb 25, 2010, at 2:43 AM, Mark Campbell-Smith wrote:

> I had an issue with a Thomson SpeedTouch 530 router that was causing
> authentication to fail... its the thread titled 'Forbidden using UDP,
> works with TCP/TLS' that everyone said was caused by a broken ATA ...
> lucky no one chipped in so I could buy a new one (as suggested by
> Anthony) - it wouldn't have helped because the router was broken and
> modifying the authentication parameters!
> 
> I'll update the wiki with the info


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



From infos at madovsky.org  Thu Feb 25 10:24:54 2010
From: infos at madovsky.org (Madovsky)
Date: Thu, 25 Feb 2010 13:24:54 -0500
Subject: [Freeswitch-users] freeswitch to gateway
References: <f8af5741002250124s5f3a200fp178d83a90945d73b@mail.gmail.com>
	<20100225170907.GG9832@base.carmickle.com>
Message-ID: <74713287E44D44D8BC02EB42F686076F@MOBILEE1705>


----- Original Message ----- 
From: "Frank Carmickle" <frank at carmickle.com>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Thursday, February 25, 2010 12:09 PM
Subject: Re: [Freeswitch-users] freeswitch to gateway


>
> On Thu, Feb 25, srinivasula reddy wrote:
>> Hi Good afternoon everybody,
>>
>> my freeswitch domain name is gw.proxy.com, i have registered 1000 to
>> freeswitch,  i have configured a gateway(gateway.com) to my freeswitch, i
>> want to make a call to gateway from 1000 to 1003 registered in 
>> gateway.com ,
>> how can i make call.
>
> Add an extension that matches on 1003 and bridge to it
> <action application="bridge" data="sofia/gateway/gatewayname/$1"/>
>
> --FC
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

Also look at the "Local_Extension" extension in dialplan/default.xml

F 



From msc at freeswitch.org  Thu Feb 25 11:49:03 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 25 Feb 2010 11:49:03 -0800
Subject: [Freeswitch-users] Multiple gateways dial string and user busy
In-Reply-To: <fbfa7b0c1002240813x58ba8541h7fef5b94f045cbda@mail.gmail.com>
References: <fbfa7b0c1002240813x58ba8541h7fef5b94f045cbda@mail.gmail.com>
Message-ID: <87f2f3b91002251149m13c0d961k722dd3375f46b6e8@mail.gmail.com>

On Wed, Feb 24, 2010 at 8:13 AM, Max Ivanov <ivanov.maxim at gmail.com> wrote:

> Hi all!
>
> when I do test call from fs_cli:
> originate sofia/gateway/panas110/223|sofia/gateway/panas111/223
> &playaback(local_stream://moh)
>
> If firest attempt returns USER_BUSY it tries to call via second one.
> Is it normal? How can I stop calling attempts after first USER_BUSY?
>

This is normal for the syntax you're using. You can try setting
ignore_early_media=true if you don't need call progress tones like ringing
and busy. It might help to know what the application is before answering
your question further. What solution are you building?
-MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100225/39d16a77/attachment.html 

From msc at freeswitch.org  Thu Feb 25 11:54:42 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 25 Feb 2010 11:54:42 -0800
Subject: [Freeswitch-users] qt framework link broken
In-Reply-To: <0CD7185862B54C1ABF67D77BA55664F7@MOBILEE1705>
References: <0CD7185862B54C1ABF67D77BA55664F7@MOBILEE1705>
Message-ID: <87f2f3b91002251154t79f6f9e8q183f0281d4dd8126@mail.gmail.com>

On Wed, Feb 24, 2010 at 12:05 PM, Madovsky <infos at madovsky.org> wrote:

>  Just to inform that
> at the link
> http://wiki.freeswitch.org/wiki/FSComm#Linux
>
> the qt framework link is broken, so as I'm new to this emailist
> I don't want to correct myself on wiki.
>
Okay I fixed it.
-MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100225/e408cbb7/attachment.html 

From msc at freeswitch.org  Thu Feb 25 11:58:14 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 25 Feb 2010 11:58:14 -0800
Subject: [Freeswitch-users] [Freeswitch-dev] Call for help - adding
	information to the wiki: SIP ALG's
In-Reply-To: <017001cab63f$a51397c0$ef3ac740$@net>
References: <87f2f3b91002242008y599f86f0t86f467f8c16b4a20@mail.gmail.com>
	<8b1c9cda1002242248u5e39a4ecvda6a3644527f59bf@mail.gmail.com>
	<33c87fa31002250043l1edbb87dg2e7a58e879656f06@mail.gmail.com>
	<EC776797-0320-4A5B-B959-2D3A3BF77314@freeswitch.org>
	<017001cab63f$a51397c0$ef3ac740$@net>
Message-ID: <87f2f3b91002251158o70b5ab25gbfbb960e31ccad5f@mail.gmail.com>

Gents,
Thanks for all of your input on this. I see a few more entries on the ALG
wiki page and that is exactly what I had hoped to see. Please keep up the
good work.

Thanks,
MC

On Thu, Feb 25, 2010 at 9:26 AM, Adam Ford <lists at redbonez.net> wrote:

> I wish you would have told me that back when I was trying to solve my
> authentication issue that I thought was being caused by using rport. :P
>
> You just told me to upgrade to 1.0.5, when a $%$#! Sonicwall SIP ALG would
> have been more helpful, haha. As it turned out it was the Sonicwall SIP
> transformations, and switching to TCP SIP resolved it.
>
> I have added my Sonicwall experience to the wiki page.
>
> -Adam
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
> West
> Sent: Thursday, February 25, 2010 6:33 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] [Freeswitch-dev] Call for help - adding
> information to the wiki: SIP ALG's
>
> I believe all SIP ALG's are broken.  :P
>
> /b
>
> On Feb 25, 2010, at 2:43 AM, Mark Campbell-Smith wrote:
>
> > I had an issue with a Thomson SpeedTouch 530 router that was causing
> > authentication to fail... its the thread titled 'Forbidden using UDP,
> > works with TCP/TLS' that everyone said was caused by a broken ATA ...
> > lucky no one chipped in so I could buy a new one (as suggested by
> > Anthony) - it wouldn't have helped because the router was broken and
> > modifying the authentication parameters!
> >
> > I'll update the wiki with the info
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100225/8b5be986/attachment.html 

From msc at freeswitch.org  Thu Feb 25 13:10:50 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 25 Feb 2010 13:10:50 -0800
Subject: [Freeswitch-users] freeswitch to gateway
In-Reply-To: <74713287E44D44D8BC02EB42F686076F@MOBILEE1705>
References: <f8af5741002250124s5f3a200fp178d83a90945d73b@mail.gmail.com>
	<20100225170907.GG9832@base.carmickle.com>
	<74713287E44D44D8BC02EB42F686076F@MOBILEE1705>
Message-ID: <87f2f3b91002251310j10ade421m5473944256564afa@mail.gmail.com>

Keep in mind that "1003" is in the default "local extension" range, so if
you want to route 1003 differently than 1000 to 1019 then be sure to put
your specific extension in the dialplan before Local_Extension, otherwise
when you dial "1003" it will go out the Local_Extension and not your custom
extension... :)

-MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100225/e1b07945/attachment.html 

From tculjaga at gmail.com  Thu Feb 25 14:34:55 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Thu, 25 Feb 2010 23:34:55 +0100
Subject: [Freeswitch-users] [Freeswitch-dev] Call for help - adding
	information to the wiki: SIP ALG's
In-Reply-To: <87f2f3b91002251158o70b5ab25gbfbb960e31ccad5f@mail.gmail.com>
References: <87f2f3b91002242008y599f86f0t86f467f8c16b4a20@mail.gmail.com>
	<8b1c9cda1002242248u5e39a4ecvda6a3644527f59bf@mail.gmail.com>
	<33c87fa31002250043l1edbb87dg2e7a58e879656f06@mail.gmail.com>
	<EC776797-0320-4A5B-B959-2D3A3BF77314@freeswitch.org>
	<017001cab63f$a51397c0$ef3ac740$@net>
	<87f2f3b91002251158o70b5ab25gbfbb960e31ccad5f@mail.gmail.com>
Message-ID: <65d96fc81002251434t5987cfcdmb7cf72a45948597e@mail.gmail.com>

well .. thats a know thing.. SIP ALG is to be disabled on every router as
this is not the way NAT traversal is donoe for SIP...

This is something a border element (SBC) on carrier side has to deal with.
If some "intelligent" router thinks it can "fix" the translation it actually
screws it from the border element perspective and you will end up with
either failed call establishment or without audio.


I had a device (zxyel) that was using SIP ALG (hiden command that was
available via telnet only) and everything was fine for a while ... but after
some time my single device behnd NAT was not able to register to the
proviser because of a broken branch string.


so, just disable ALG for SIP on ALL devices you have on customer side .. it
really doesn't have any sense to keep it on.

T.





On Thu, Feb 25, 2010 at 8:58 PM, Michael Collins <msc at freeswitch.org> wrote:

> Gents,
> Thanks for all of your input on this. I see a few more entries on the ALG
> wiki page and that is exactly what I had hoped to see. Please keep up the
> good work.
>
> Thanks,
> MC
>
>
> On Thu, Feb 25, 2010 at 9:26 AM, Adam Ford <lists at redbonez.net> wrote:
>
>> I wish you would have told me that back when I was trying to solve my
>> authentication issue that I thought was being caused by using rport. :P
>>
>> You just told me to upgrade to 1.0.5, when a $%$#! Sonicwall SIP ALG would
>> have been more helpful, haha. As it turned out it was the Sonicwall SIP
>> transformations, and switching to TCP SIP resolved it.
>>
>> I have added my Sonicwall experience to the wiki page.
>>
>> -Adam
>>
>>
>> -----Original Message-----
>> From: freeswitch-users-bounces at lists.freeswitch.org
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
>> West
>> Sent: Thursday, February 25, 2010 6:33 AM
>> To: freeswitch-users at lists.freeswitch.org
>> Subject: Re: [Freeswitch-users] [Freeswitch-dev] Call for help - adding
>> information to the wiki: SIP ALG's
>>
>> I believe all SIP ALG's are broken.  :P
>>
>> /b
>>
>> On Feb 25, 2010, at 2:43 AM, Mark Campbell-Smith wrote:
>>
>> > I had an issue with a Thomson SpeedTouch 530 router that was causing
>> > authentication to fail... its the thread titled 'Forbidden using UDP,
>> > works with TCP/TLS' that everyone said was caused by a broken ATA ...
>> > lucky no one chipped in so I could buy a new one (as suggested by
>> > Anthony) - it wouldn't have helped because the router was broken and
>> > modifying the authentication parameters!
>> >
>> > I'll update the wiki with the info
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100225/9bb90360/attachment-0001.html 

From joseph.puchalski at personalcyberspace.com  Thu Feb 25 15:29:17 2010
From: joseph.puchalski at personalcyberspace.com (Joseph Puchalski)
Date: Thu, 25 Feb 2010 23:29:17 +0000
Subject: [Freeswitch-users] Retrieving voicemail without
	entering	user	ID (extension)
In-Reply-To: <012c01cab5b7$5052d990$f0f88cb0$@net>
References: <093DD565390C1E4FB15D7B383E86BB05AFF6BC@Goose.personalcyberspace.net>
	<b39273531002241601m273ff333y2261d930dbcb3fa7@mail.gmail.com>
	<012c01cab5b7$5052d990$f0f88cb0$@net>
Message-ID: <093DD565390C1E4FB15D7B383E86BB05AFF90C@Goose.personalcyberspace.net>

Adam,

Thanks! I'll give this a try. I'm FreeSWITCH newbie myself, having a fun time figuring out everything in this amazingly rich (and challenging) environment :)

Joe

From: Adam Ford [mailto:lists at redbonez.net]
Sent: Wednesday, February 24, 2010 8:10 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Retrieving voicemail without entering user ID (extension)

>From reading that wiki article it seems to me that the key to achieving the functionality you are looking for would simply be a matter of adding the desired extension to the end of the default action (where the $1 is):

<action application="voicemail" data="check $${voicemail_profile} $${domain} $1"/>

If I am reading it correctly, this should bypass having to enter a mailbox ID, but still require your voicemail password. Off the top of my head, you could probably achieve this by replacing the $1 with a variable storing the extension which called 4000.  I would have to look it up to see if there is a system variable for that or if you would have to assign a custom one. I am still relatively new to FreeSWITCH myself.

-Adam


From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rupa Schomaker
Sent: Wednesday, February 24, 2010 5:01 PM
To: freeswitch-users
Subject: Re: [Freeswitch-users] Retrieving voicemail without entering user ID (extension)

Look at the end of:

http://wiki.freeswitch.org/wiki/Mod_voicemail#Check_Voice_Mail

Advisable?  With it enabled, I can walk up to anyone's phone and retrieve their VM w/out authentication.  It was removed on purpose due to that reason as far as I remember.
On Wed, Feb 24, 2010 at 5:45 PM, Joseph Puchalski <joseph.puchalski at personalcyberspace.com<mailto:joseph.puchalski at personalcyberspace.com>> wrote:
I'm trying to modify my dialplan so that I can press a single button on my phone, be connected to voicemail, and enter only a password to gain access.

Currently I use a programmable key to dial 4000. I am prompted for my ID, and then password.

I've poked around "mod voicemail" on the wiki and searched the mailing list and web, but haven't found enough info. I have discovered that this behavior seems to have been available in previous versions of the default dialplan.

Is it still possible? Is it advisable? Was this feature/behavior removed for security reasons?

I apologize ahead of time if the answer is somewhere in plain sight that I haven't looked yet. If so, I'd much appreciate being pointed in the right direction.

As always, thanks for any help,

Joe P.

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



--
-Rupa
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100225/2568fe08/attachment.html 

From joseph.puchalski at personalcyberspace.com  Thu Feb 25 15:30:14 2010
From: joseph.puchalski at personalcyberspace.com (Joseph Puchalski)
Date: Thu, 25 Feb 2010 23:30:14 +0000
Subject: [Freeswitch-users] Retrieving voicemail without entering
	user	ID (extension)
In-Reply-To: <b39273531002241601m273ff333y2261d930dbcb3fa7@mail.gmail.com>
References: <093DD565390C1E4FB15D7B383E86BB05AFF6BC@Goose.personalcyberspace.net>
	<b39273531002241601m273ff333y2261d930dbcb3fa7@mail.gmail.com>
Message-ID: <093DD565390C1E4FB15D7B383E86BB05AFF919@Goose.personalcyberspace.net>

Thanks for the reply.

I'm not trying to remove the requirement for a password, just the need to enter an extension from a user's "home" phone. The current methods I'm familiar with require that the user enter both a user ID and a password. I'm hoping there's a way that the user ID can be defaulted to be the calling extension. This way I can set up a "voicemail" key on a user's desktop phone and allow them to access vmail by entering just a password.

Thanks again,

Joe

From: Rupa Schomaker [mailto:rupa at rupa.com]
Sent: Wednesday, February 24, 2010 7:01 PM
To: freeswitch-users
Subject: Re: [Freeswitch-users] Retrieving voicemail without entering user ID (extension)

Look at the end of:

http://wiki.freeswitch.org/wiki/Mod_voicemail#Check_Voice_Mail

Advisable?  With it enabled, I can walk up to anyone's phone and retrieve their VM w/out authentication.  It was removed on purpose due to that reason as far as I remember.
On Wed, Feb 24, 2010 at 5:45 PM, Joseph Puchalski <joseph.puchalski at personalcyberspace.com<mailto:joseph.puchalski at personalcyberspace.com>> wrote:
I'm trying to modify my dialplan so that I can press a single button on my phone, be connected to voicemail, and enter only a password to gain access.

Currently I use a programmable key to dial 4000. I am prompted for my ID, and then password.

I've poked around "mod voicemail" on the wiki and searched the mailing list and web, but haven't found enough info. I have discovered that this behavior seems to have been available in previous versions of the default dialplan.

Is it still possible? Is it advisable? Was this feature/behavior removed for security reasons?

I apologize ahead of time if the answer is somewhere in plain sight that I haven't looked yet. If so, I'd much appreciate being pointed in the right direction.

As always, thanks for any help,

Joe P.

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



--
-Rupa
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100225/90563d0e/attachment-0001.html 

From joseph.puchalski at personalcyberspace.com  Thu Feb 25 15:31:18 2010
From: joseph.puchalski at personalcyberspace.com (Joseph Puchalski)
Date: Thu, 25 Feb 2010 23:31:18 +0000
Subject: [Freeswitch-users] Retrieving voicemail without
	entering	user	ID (extension)
In-Reply-To: <000f01cab5c7$d7e292f0$87a7b8d0$@com>
References: <093DD565390C1E4FB15D7B383E86BB05AFF6BC@Goose.personalcyberspace.net>
	<b39273531002241601m273ff333y2261d930dbcb3fa7@mail.gmail.com>
	<000f01cab5c7$d7e292f0$87a7b8d0$@com>
Message-ID: <093DD565390C1E4FB15D7B383E86BB05AFF928@Goose.personalcyberspace.net>

Lars,

Thanks!

Joe

From: Lars Zeb [mailto:larclap at yahoo.com]
Sent: Wednesday, February 24, 2010 10:09 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Retrieving voicemail without entering user ID (extension)

Joe,

I used the extension below, but I think that Brian said it was too insecure. Being a total beginner, I removed the condition.

    <extension name="Local_Extension_Abe">
      <condition field="destination_number" expression="^(101[0-9])$">
        <action application="set" data="dialed_ext=$1"/>
        <action application="export" data="dialed_ext=$1"/>
      </condition>
      <condition field="destination_number" expression="^${caller_id_number}$">
        <action application="set" data="voicemail_authorized=${sip_authorized}"/>
        <action application="answer"/>
        <action application="sleep" data="1000"/>
        <action application="voicemail" data="check default $${domain} ${dialed_ext}"/>
        <anti-action application="bind_meta_app" data="1 a a execute_extension::dx XML features"/>
        <anti-action application="bind_meta_app" data="2 a a record_session::$${base_dir}/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
        <anti-action application="bind_meta_app" data="3 a a execute_extension::cf XML features"/>
        <anti-action application="set" data="transfer_ringback=${us-ring}"/>
        <anti-action application="set" data="call_timeout=15"/>
        <anti-action application="set" data="sip_exclude_contact=${network_addr}"/>
        <anti-action application="set" data="hangup_after_bridge=true"/>
        <!--<anti-action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,BUSY,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION"/> -->
        <anti-action application="set" data="continue_on_fail=true"/>
        <anti-action application="db" data="insert/call_return/${dialed_ext}/${caller_id_number}"/>
        <anti-action application="db" data="insert/last_dial_ext/${dialed_ext}/${uuid}"/>
        <anti-action application="bridge" data="user/${dialed_ext}@$${domain}"/>
        <anti-action application="answer"/>
        <anti-action application="sleep" data="1000"/>
        <anti-action application="voicemail" data="default $${domain} ${dialed_ext}"/>
      </condition>
    </extension>

Lars
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rupa Schomaker
Sent: Wednesday, February 24, 2010 4:01 PM
To: freeswitch-users
Subject: Re: [Freeswitch-users] Retrieving voicemail without entering user ID (extension)

Look at the end of:

http://wiki.freeswitch.org/wiki/Mod_voicemail#Check_Voice_Mail

Advisable?  With it enabled, I can walk up to anyone's phone and retrieve their VM w/out authentication.  It was removed on purpose due to that reason as far as I remember.
On Wed, Feb 24, 2010 at 5:45 PM, Joseph Puchalski <joseph.puchalski at personalcyberspace.com<mailto:joseph.puchalski at personalcyberspace.com>> wrote:
I'm trying to modify my dialplan so that I can press a single button on my phone, be connected to voicemail, and enter only a password to gain access.

Currently I use a programmable key to dial 4000. I am prompted for my ID, and then password.

I've poked around "mod voicemail" on the wiki and searched the mailing list and web, but haven't found enough info. I have discovered that this behavior seems to have been available in previous versions of the default dialplan.

Is it still possible? Is it advisable? Was this feature/behavior removed for security reasons?

I apologize ahead of time if the answer is somewhere in plain sight that I haven't looked yet. If so, I'd much appreciate being pointed in the right direction.

As always, thanks for any help,

Joe P.

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



--
-Rupa
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100225/2bfa1243/attachment.html 

From robert.hadley at teotech.com  Thu Feb 25 16:13:10 2010
From: robert.hadley at teotech.com (Robert Hadley)
Date: Thu, 25 Feb 2010 16:13:10 -0800
Subject: [Freeswitch-users] Freeswitch/Openzap dials out on disconnected FXO
	line
Message-ID: <4DF42CB92831454193CEC0E375E06725@greyhawk.tonecommander.com>

When dialing out, Freeswitch/Openzap is not detecting that an analog FXO
channel is disconnected and tries dialing out on the channel anyway.  No
error is reported.  The call doesn't timeout until a minute later.
Shouldn't Freeswitch/Openzap skip over a disconnected channel to the next
connected channel?

 

I have configured a Sangoma A200 FXO card as a FXO span.

 

[span wanpipe FXO]

name => PSTN Line 1

number => 4253491059

fxo-channel => 2:3

name => PSTN Line 2

number => 4253491058

fxo-channel => 2:4

 

 

The wanpipe driver does detect and report when a CO line is connected or
disconnected (in /var/log/messages), and Freeswitch/Openzap gets an event as
reported in the log.

 

/var/log/messages: Feb 25 15:23:10 roberth-c53 kernel: wanpipe2: Module 3:
FXO Line is disconnected!

FS_CLI: 2010-02-25 15:23:10.711604 [DEBUG] ozmod_analog.c:788 EVENT
[ALARM_TRAP][3:1] STATE [DOWN]

 

/var/log/messages: Feb 25 15:23:44 roberth-c53 kernel: wanpipe2: Module 4:
FXO Line is connected!

FS_CLI: 2010-02-25 15:23:44.901979 [DEBUG] ozmod_analog.c:788 EVENT
[ALARM_CLEAR][3:2] STATE [DOWN]

 

 

I have the dialplan configured to use the next available port in the FXO
span (there will be more than 2 channels later).

 

<extension name="OutgoingFXO">

  <condition field="destination_number" expression="^7(\d+)$">

    <action application="bridge" data="openzap/FXO/a/$1"/>

  </condition>

</extension>

 

 

Here is a portion of the log when that shows dialing out on a disconnected
analog FXO channel.

 

EXECUTE sofia/internal/5410 at 192.168.72.45:5060
bridge(openzap/FXO/a/93491045)

2010-02-25 15:26:17.891443 [DEBUG] mod_openzap.c:366 Set codec PCMU 20ms

2010-02-25 15:26:17.891443 [DEBUG] mod_openzap.c:1257 Connect outbound
channel OpenZAP/3:1/93491045

2010-02-25 15:26:17.891443 [NOTICE] switch_channel.c:642 New Channel
OpenZAP/3:1/93491045 [3c8f46f5-77a8-498f-a51c-015837746cb7]

2010-02-25 15:26:17.891443 [DEBUG] mod_openzap.c:1269 (OpenZAP/3:1/93491045)
State Change CS_NEW -> CS_INIT

2010-02-25 15:26:17.891443 [DEBUG] switch_core_session.c:1019 Send signal
OpenZAP/3:1/93491045 [BREAK]

2010-02-25 15:26:17.891443 [DEBUG] ozmod_analog.c:59 Changing state on 3:1
from DOWN to DIALING

2010-02-25 15:26:17.891443 [WARNING] switch_core_session.c:486
OpenZAP/3:1/93491045 does not support the proxy feature, disabling.

2010-02-25 15:26:17.891443 [DEBUG] ozmod_analog.c:279 ANALOG CHANNEL thread
starting.

2010-02-25 15:26:17.891443 [DEBUG] ozmod_analog.c:450 Executing state
handler on 3:1 for DIALING

2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:314
(OpenZAP/3:1/93491045) Running State Change CS_INIT

2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:338
(OpenZAP/3:1/93491045) State INIT

2010-02-25 15:26:17.891443 [DEBUG] mod_openzap.c:394 (OpenZAP/3:1/93491045)
State Change CS_INIT -> CS_ROUTING

2010-02-25 15:26:17.891443 [DEBUG] switch_core_session.c:1019 Send signal
OpenZAP/3:1/93491045 [BREAK]

2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:338
(OpenZAP/3:1/93491045) State INIT going to sleep

2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:314
(OpenZAP/3:1/93491045) Running State Change CS_ROUTING

2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:341
(OpenZAP/3:1/93491045) State ROUTING

2010-02-25 15:26:17.891443 [DEBUG] mod_openzap.c:417 OpenZAP/3:1/93491045
CHANNEL ROUTING

2010-02-25 15:26:17.891443 [DEBUG] switch_ivr_originate.c:66
(OpenZAP/3:1/93491045) State Change CS_ROUTING -> CS_CONSUME_MEDIA

2010-02-25 15:26:17.891443 [DEBUG] switch_core_session.c:1019 Send signal
OpenZAP/3:1/93491045 [BREAK]

2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:341
(OpenZAP/3:1/93491045) State ROUTING going to sleep

2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:314
(OpenZAP/3:1/93491045) Running State Change CS_CONSUME_MEDIA

2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:360
(OpenZAP/3:1/93491045) State CONSUME_MEDIA

2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:360
(OpenZAP/3:1/93491045) State CONSUME_MEDIA going to sleep

 

 

Thanks,

Robert

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100225/b238b94e/attachment-0001.html 

From robert.hadley at teotech.com  Thu Feb 25 17:16:07 2010
From: robert.hadley at teotech.com (Robert Hadley)
Date: Thu, 25 Feb 2010 17:16:07 -0800
Subject: [Freeswitch-users] Cannot make calls through PRI via wanpipe
	running as daemon
Message-ID: <6FCED74CF54F466AA25CBDA7DFF42198@greyhawk.tonecommander.com>

When running Freeswitch as service called teoswitch as user teoswitch I
cannot make calls through the Sangoma PRI or analog cards using wanpipe
driver.  I have added a file called 30-wanpipe.rules to /etc/udev/rules.d
and rebooted the server.

 

cat 30-wanpipe.rules

# /etc/udev/rules.d/30-wanpipe.rules

SUBSYSTEM=="wptdm", OWNER="teoswitch", GROUP="teoswitch", MODE="0660"

SUBSYSTEM=="wanpipe", OWNER="teoswitch", GROUP="teoswitch", MODE="0660"

 

 

Freeswitch log:

Dialplan: sofia/internal/5410 at 192.168.72.45:5060 parsing
[default->SangomaPRI] continue=false

Dialplan: sofia/internal/5410 at 192.168.72.45:5060 Regex (PASS) [SangomaPRI]
destination_number(93491045) =~ /^9(\d+)$/ break=on-false

Dialplan: sofia/internal/5410 at 192.168.72.45:5060 Action
set(effective_caller_id_number=425740${caller_id_number})

Dialplan: sofia/internal/5410 at 192.168.72.45:5060 Action
bridge(openzap/smg_prid/a/3491045 at g1)

2010-02-25 16:51:11.328635 [DEBUG] switch_core_state_machine.c:122
(sofia/internal/5410 at 192.168.72.45:5060) State Change CS_ROUTING ->
CS_EXECUTE

2010-02-25 16:51:11.328635 [DEBUG] switch_core_session.c:1019 Send signal
sofia/internal/5410 at 192.168.72.45:5060 [BREAK]

2010-02-25 16:51:11.328635 [DEBUG] switch_core_state_machine.c:341
(sofia/internal/5410 at 192.168.72.45:5060) State ROUTING going to sleep

2010-02-25 16:51:11.328635 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/5410 at 192.168.72.45:5060) Running State Change CS_EXECUTE

2010-02-25 16:51:11.328635 [DEBUG] switch_core_state_machine.c:348
(sofia/internal/5410 at 192.168.72.45:5060) State EXECUTE

2010-02-25 16:51:11.328635 [DEBUG] mod_sofia.c:181
sofia/internal/5410 at 192.168.72.45:5060 SOFIA EXECUTE

2010-02-25 16:51:11.328635 [DEBUG] switch_core_state_machine.c:159
sofia/internal/5410 at 192.168.72.45:5060 Standard EXECUTE

EXECUTE sofia/internal/5410 at 192.168.72.45:5060
set(effective_caller_id_number=4257405410)

2010-02-25 16:51:11.328635 [DEBUG] mod_dptools.c:811
sofia/internal/5410 at 192.168.72.45:5060 SET
[effective_caller_id_number]=[4257405410]

EXECUTE sofia/internal/5410 at 192.168.72.45:5060
bridge(openzap/smg_prid/a/3491045 at g1)

2010-02-25 16:51:11.339637 [ERR] switch_core_session.c:357 Could not locate
channel type openzap

2010-02-25 16:51:11.339637 [ERR] switch_ivr_originate.c:2411 Cannot create
outgoing channel of type [openzap] cause: [CHAN_NOT_IMPLEMENTED]

2010-02-25 16:51:11.339637 [DEBUG] switch_ivr_originate.c:3209 Originate
Resulted in Error Cause: 66 [CHAN_NOT_IMPLEMENTED]

2010-02-25 16:51:11.339637 [INFO] mod_dptools.c:2346 Originate Failed.
Cause: CHAN_NOT_IMPLEMENTED

2010-02-25 16:51:11.339637 [NOTICE] mod_dptools.c:2409 Hangup
sofia/internal/5410 at 192.168.72.45:5060 [CS_EXECUTE] [CHAN_NOT_IMPLEMENTED]

2010-02-25 16:51:11.339637 [DEBUG] switch_channel.c:1976 Send signal
sofia/internal/5410 at 192.168.72.45:5060 [KILL]

2010-02-25 16:51:11.339637 [DEBUG] switch_core_session.c:1019 Send signal
sofia/internal/5410 at 192.168.72.45:5060 [BREAK]

 

 

Thanks,

Robert

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100225/bd95f30b/attachment.html 

From brian at freeswitch.org  Thu Feb 25 17:22:51 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 25 Feb 2010 19:22:51 -0600
Subject: [Freeswitch-users] ASR Apps
Message-ID: <AC3A71D6-4EAB-4261-B14A-2869C0C05C91@freeswitch.org>

I'm looking for some enterprising community members to create some interesting voice apps using ASR.  Please email me off list and we'll get you what you need to do this.

Thanks,
Brian



From edpimentl at gmail.com  Thu Feb 25 18:55:12 2010
From: edpimentl at gmail.com (EdPimentl)
Date: Thu, 25 Feb 2010 21:55:12 -0500
Subject: [Freeswitch-users] ASR Apps
In-Reply-To: <AC3A71D6-4EAB-4261-B14A-2869C0C05C91@freeswitch.org>
References: <AC3A71D6-4EAB-4261-B14A-2869C0C05C91@freeswitch.org>
Message-ID: <9dc4a1671002251855y586218f6h2c281b4a5acb664e@mail.gmail.com>

Hello Bryon,

We looking to create a Twilio like service using FreeSwitch.

Sincerely,
-E
http://vCardCloud.com
GV:  678.685.9858
EdPimentl: Skype



On Thu, Feb 25, 2010 at 8:22 PM, Brian West <brian at freeswitch.org> wrote:

> I'm looking for some enterprising community members to create some
> interesting voice apps using ASR.  Please email me off list and we'll get
> you what you need to do this.
>
> Thanks,
> Brian
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100225/c7098d9b/attachment.html 

From brian at freeswitch.org  Thu Feb 25 19:02:14 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 25 Feb 2010 21:02:14 -0600
Subject: [Freeswitch-users] ASR Apps
In-Reply-To: <9dc4a1671002251855y586218f6h2c281b4a5acb664e@mail.gmail.com>
References: <AC3A71D6-4EAB-4261-B14A-2869C0C05C91@freeswitch.org>
	<9dc4a1671002251855y586218f6h2c281b4a5acb664e@mail.gmail.com>
Message-ID: <16DBCD58-A962-4121-9899-F2BB56F13554@freeswitch.org>

I'm looking for someone to build some really nice apps like dial by name speech apps or other such apps or frameworks using ASR and possibly lua or js.  Anyone wanna do something.

/b

On Feb 25, 2010, at 8:55 PM, EdPimentl wrote:

> Hello Bryon,
> 
> We looking to create a Twilio like service using FreeSwitch.
> 
> Sincerely,
> -E
> http://vCardCloud.com
> GV:  678.685.9858
> EdPimentl: Skype

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100225/52881261/attachment.html 

From srinivas.ksvreddy at gmail.com  Thu Feb 25 20:42:46 2010
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Fri, 26 Feb 2010 10:12:46 +0530
Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 44, Issue 234
In-Reply-To: <mailman.6938.1267143241.2495.freeswitch-users@lists.freeswitch.org>
References: <mailman.6938.1267143241.2495.freeswitch-users@lists.freeswitch.org>
Message-ID: <f8af5741002252042r51671ec9ia0ca7a2fbcd44e62@mail.gmail.com>

Hi,

thank you very munch for reply,
this is working fine, when we configure <condition
destination_number="^1003"
<action application="bridge" data="sofia/gateway/gatewayname/$1"/>

but in my scenario, i dont want to route call based on extensions( eg, here
1003) routing, i just want to route the calls when the destination domain is
defferan from local domain,

example: INVITE packet from registered extension to sipserver like this.

   From: 1000 at gw.proxy.com:5060
   To :  1003 at gateway.com:5060

here from uri and to uri is different. any help


Srinivas

On Fri, Feb 26, 2010 at 5:44 AM, <
freeswitch-users-request at lists.freeswitch.org> wrote:

> Send FreeSWITCH-users mailing list submissions to
>        freeswitch-users at lists.freeswitch.org
>
> To subscribe or unsubscribe via the World Wide Web, visit
>        http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> or, via email, send a message with subject or body 'help' to
>        freeswitch-users-request at lists.freeswitch.org
>
> You can reach the person managing the list at
>        freeswitch-users-owner at lists.freeswitch.org
>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of FreeSWITCH-users digest..."
>
> Today's Topics:
>
>   1. Re: Retrieving voicemail without  entering        user    ID
> (extension)
>      (Joseph Puchalski)
>   2. Freeswitch/Openzap dials out on disconnected FXO  line
>      (Robert Hadley)
>
>
> ---------- Forwarded message ----------
> From: Joseph Puchalski <joseph.puchalski at personalcyberspace.com>
> To: "freeswitch-users at lists.freeswitch.org" <
> freeswitch-users at lists.freeswitch.org>
> Date: Thu, 25 Feb 2010 23:31:18 +0000
> Subject: Re: [Freeswitch-users] Retrieving voicemail without entering user
> ID (extension)
>
> Lars,
>
>
>
> Thanks!
>
>
>
> Joe
>
>
>
> *From:* Lars Zeb [mailto:larclap at yahoo.com]
> *Sent:* Wednesday, February 24, 2010 10:09 PM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Retrieving voicemail without entering
> user ID (extension)
>
>
>
> Joe,
>
>
>
> I used the extension below, but I think that Brian said it was too
> insecure. Being a total beginner, I removed the condition.
>
>
>
>     <extension name="Local_Extension_Abe">
>
>       <condition field="destination_number" expression="^(101[0-9])$">
>
>         <action application="set" data="dialed_ext=$1"/>
>
>         <action application="export" data="dialed_ext=$1"/>
>
>       </condition>
>
>       <condition field="destination_number"
> expression="^${caller_id_number}$">
>
>         <action application="set"
> data="voicemail_authorized=${sip_authorized}"/>
>
>         <action application="answer"/>
>
>         <action application="sleep" data="1000"/>
>
>         <action application="voicemail" data="check default $${domain}
> ${dialed_ext}"/>
>
>         <anti-action application="bind_meta_app" data="1 a a
> execute_extension::dx XML features"/>
>
>         <anti-action application="bind_meta_app" data="2 a a
> record_session::$${base_dir}/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
>
>         <anti-action application="bind_meta_app" data="3 a a
> execute_extension::cf XML features"/>
>
>         <anti-action application="set"
> data="transfer_ringback=${us-ring}"/>
>
>         <anti-action application="set" data="call_timeout=15"/>
>
>         <anti-action application="set"
> data="sip_exclude_contact=${network_addr}"/>
>
>         <anti-action application="set" data="hangup_after_bridge=true"/>
>
>         <!--<anti-action application="set"
> data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,BUSY,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION"/>
> -->
>
>         <anti-action application="set" data="continue_on_fail=true"/>
>
>         <anti-action application="db"
> data="insert/call_return/${dialed_ext}/${caller_id_number}"/>
>
>         <anti-action application="db"
> data="insert/last_dial_ext/${dialed_ext}/${uuid}"/>
>
>         <anti-action application="bridge" data="user/${dialed_ext}@
> $${domain}"/>
>
>         <anti-action application="answer"/>
>
>         <anti-action application="sleep" data="1000"/>
>
>         <anti-action application="voicemail" data="default $${domain}
> ${dialed_ext}"/>
>
>       </condition>
>
>     </extension>
>
>
>
> Lars
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Rupa
> Schomaker
> *Sent:* Wednesday, February 24, 2010 4:01 PM
> *To:* freeswitch-users
> *Subject:* Re: [Freeswitch-users] Retrieving voicemail without entering
> user ID (extension)
>
>
>
> Look at the end of:
>
>
>
> http://wiki.freeswitch.org/wiki/Mod_voicemail#Check_Voice_Mail
>
>
>
> Advisable?  With it enabled, I can walk up to anyone's phone and retrieve
> their VM w/out authentication.  It was removed on purpose due to that reason
> as far as I remember.
>
> On Wed, Feb 24, 2010 at 5:45 PM, Joseph Puchalski <
> joseph.puchalski at personalcyberspace.com> wrote:
>
> I?m trying to modify my dialplan so that I can press a single button on my
> phone, be connected to voicemail, and enter only a password to gain access.
>
>
>
> Currently I use a programmable key to dial 4000. I am prompted for my ID,
> and then password.
>
>
>
> I?ve poked around ?mod voicemail? on the wiki and searched the mailing list
> and web, but haven?t found enough info. I have discovered that this behavior
> seems to have been available in previous versions of the default dialplan.
>
>
>
> Is it still possible? Is it advisable? Was this feature/behavior removed
> for security reasons?
>
>
>
> I apologize ahead of time if the answer is somewhere in plain sight that I
> haven?t looked yet. If so, I?d much appreciate being pointed in the right
> direction.
>
>
>
> As always, thanks for any help,
>
>
>
> Joe P.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> --
> -Rupa
>
>
> ---------- Forwarded message ----------
> From: "Robert Hadley" <robert.hadley at teotech.com>
> To: <freeswitch-users at lists.freeswitch.org>
> Date: Thu, 25 Feb 2010 16:13:10 -0800
> Subject: [Freeswitch-users] Freeswitch/Openzap dials out on disconnected
> FXO line
>
> When dialing out, Freeswitch/Openzap is not detecting that an analog FXO
> channel is disconnected and tries dialing out on the channel anyway.  No
> error is reported.  The call doesn?t timeout until a minute later.
> Shouldn?t Freeswitch/Openzap skip over a disconnected channel to the next
> connected channel?
>
>
>
> I have configured a Sangoma A200 FXO card as a FXO span.
>
>
>
> [span wanpipe FXO]
>
> name => PSTN Line 1
>
> number => 4253491059
>
> fxo-channel => 2:3
>
> name => PSTN Line 2
>
> number => 4253491058
>
> fxo-channel => 2:4
>
>
>
>
>
> The wanpipe driver does detect and report when a CO line is connected or
> disconnected (in /var/log/messages), and Freeswitch/Openzap gets an event as
> reported in the log.
>
>
>
> /var/log/messages: Feb 25 15:23:10 roberth-c53 kernel: wanpipe2: Module 3:
> FXO Line is disconnected!
>
> FS_CLI: 2010-02-25 15:23:10.711604 [DEBUG] ozmod_analog.c:788 EVENT
> [ALARM_TRAP][3:1] STATE [DOWN]
>
>
>
> /var/log/messages: Feb 25 15:23:44 roberth-c53 kernel: wanpipe2: Module 4:
> FXO Line is connected!
>
> FS_CLI: 2010-02-25 15:23:44.901979 [DEBUG] ozmod_analog.c:788 EVENT
> [ALARM_CLEAR][3:2] STATE [DOWN]
>
>
>
>
>
> I have the dialplan configured to use the next available port in the FXO
> span (there will be more than 2 channels later).
>
>
>
> <extension name="OutgoingFXO">
>
>   <condition field="destination_number" expression="^7(\d+)$">
>
>     <action application="bridge" data="openzap/FXO/a/$1"/>
>
>   </condition>
>
> </extension>
>
>
>
>
>
> Here is a portion of the log when that shows dialing out on a disconnected
> analog FXO channel.
>
>
>
> EXECUTE sofia/internal/5410 at 192.168.72.45:5060bridge(openzap/FXO/a/93491045)
>
> 2010-02-25 15:26:17.891443 [DEBUG] mod_openzap.c:366 Set codec PCMU 20ms
>
> 2010-02-25 15:26:17.891443 [DEBUG] mod_openzap.c:1257 Connect outbound
> channel OpenZAP/3:1/93491045
>
> 2010-02-25 15:26:17.891443 [NOTICE] switch_channel.c:642 New Channel
> OpenZAP/3:1/93491045 [3c8f46f5-77a8-498f-a51c-015837746cb7]
>
> 2010-02-25 15:26:17.891443 [DEBUG] mod_openzap.c:1269
> (OpenZAP/3:1/93491045) State Change CS_NEW -> CS_INIT
>
> 2010-02-25 15:26:17.891443 [DEBUG] switch_core_session.c:1019 Send signal
> OpenZAP/3:1/93491045 [BREAK]
>
> 2010-02-25 15:26:17.891443 [DEBUG] ozmod_analog.c:59 Changing state on 3:1
> from DOWN to DIALING
>
> 2010-02-25 15:26:17.891443 [WARNING] switch_core_session.c:486
> OpenZAP/3:1/93491045 does not support the proxy feature, disabling.
>
> 2010-02-25 15:26:17.891443 [DEBUG] ozmod_analog.c:279 ANALOG CHANNEL thread
> starting.
>
> 2010-02-25 15:26:17.891443 [DEBUG] ozmod_analog.c:450 Executing state
> handler on 3:1 for DIALING
>
> 2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:314
> (OpenZAP/3:1/93491045) Running State Change CS_INIT
>
> 2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:338
> (OpenZAP/3:1/93491045) State INIT
>
> 2010-02-25 15:26:17.891443 [DEBUG] mod_openzap.c:394 (OpenZAP/3:1/93491045)
> State Change CS_INIT -> CS_ROUTING
>
> 2010-02-25 15:26:17.891443 [DEBUG] switch_core_session.c:1019 Send signal
> OpenZAP/3:1/93491045 [BREAK]
>
> 2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:338
> (OpenZAP/3:1/93491045) State INIT going to sleep
>
> 2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:314
> (OpenZAP/3:1/93491045) Running State Change CS_ROUTING
>
> 2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:341
> (OpenZAP/3:1/93491045) State ROUTING
>
> 2010-02-25 15:26:17.891443 [DEBUG] mod_openzap.c:417 OpenZAP/3:1/93491045
> CHANNEL ROUTING
>
> 2010-02-25 15:26:17.891443 [DEBUG] switch_ivr_originate.c:66
> (OpenZAP/3:1/93491045) State Change CS_ROUTING -> CS_CONSUME_MEDIA
>
> 2010-02-25 15:26:17.891443 [DEBUG] switch_core_session.c:1019 Send signal
> OpenZAP/3:1/93491045 [BREAK]
>
> 2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:341
> (OpenZAP/3:1/93491045) State ROUTING going to sleep
>
> 2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:314
> (OpenZAP/3:1/93491045) Running State Change CS_CONSUME_MEDIA
>
> 2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:360
> (OpenZAP/3:1/93491045) State CONSUME_MEDIA
>
> 2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:360
> (OpenZAP/3:1/93491045) State CONSUME_MEDIA going to sleep
>
>
>
>
>
> Thanks,
>
> Robert
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Srinivasula Reddy K
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100226/637c285b/attachment-0001.html 

From srinivas.ksvreddy at gmail.com  Thu Feb 25 20:51:21 2010
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Fri, 26 Feb 2010 10:21:21 +0530
Subject: [Freeswitch-users] freeswitch to gateway
Message-ID: <f8af5741002252051id2e247al1138ea08f8a507ec@mail.gmail.com>

Hi,

thank you very munch for reply,
this is working fine, when we configure <condition
destination_number="^1003"
<action application="bridge" data="sofia/gateway/gatewayname/$1"/>

but in my scenario, i dont want to route call based on extensions( eg, here
1003) routing, i just want to route the calls when the destination host uri
is defferan from local domain,

example: INVITE packet from registered extension to sipserver like this.

   From: 1000 at gw.proxy.com:5060
   To :  1003 at gateway.com:5060

here from uri and to uri is different. any help


Srinivas
-- 
Srinivasula Reddy K
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100226/b701f674/attachment.html 

From nagalenoj at gmail.com  Thu Feb 25 21:33:28 2010
From: nagalenoj at gmail.com (Nagalenoj H.)
Date: Fri, 26 Feb 2010 11:03:28 +0530
Subject: [Freeswitch-users] Error loading
	/usr/local/freeswitch/mod/ozmod_sangoma_boost.so
In-Reply-To: <D62532C0-53A1-4327-9BAA-DB2FB0C4BEF8@jerris.com>
References: <fe67a3341002230202y1d45a8f3t3a2cda07de2240a2@mail.gmail.com>
	<87f2f3b91002231357t6c3e40cdld6a0a9861a9ec5a5@mail.gmail.com>
	<fe67a3341002232043w23e5c04bh2598bdd9641d9c23@mail.gmail.com>
	<D62532C0-53A1-4327-9BAA-DB2FB0C4BEF8@jerris.com>
Message-ID: <fe67a3341002252133k6e2793a7t102a580ad1c879f5@mail.gmail.com>

I was missing lksctp-tools and libsctp-dev packages.
After installing these two, I started installing from first. The issue got
solved.!

On Wed, Feb 24, 2010 at 1:53 PM, Michael Jerris <mike at jerris.com> wrote:

> you missed the second 1/2 of step 3 of *Wan**pipe TDM Installation*
> *
> *
> *
> *
> On Feb 23, 2010, at 11:43 PM, Nagalenoj H. wrote:
>
> ozmod_sangoma_boost.so doesn't exist anywhere. It may not be a old version,
> since I've checked out the source yesterday.
>
> I've a doubt in the installation steps given. It is given to edit the
> modules.conf after executing ./configure. Is it right? Do I need to edit the
> modules.conf before ./configure??
>
> On Wed, Feb 24, 2010 at 3:27 AM, Michael Collins <msc at freeswitch.org>wrote:
>
>> Does the file ozmod_sangoma_boost.so exist somewhere in your openzap build
>> tree? If not then something went wrong while compiling or you have an old
>> version. If it does exist, do a quick test: cp the file into
>> /usr/local/freeswitch/mod and restart FreeSWITCH and see if it finds the
>> file and loads OpenZAP properly. Let us know the results so we can determine
>> if it's a bug in the build system or not.
>>
>> -MC
>>
>> On Tue, Feb 23, 2010 at 2:02 AM, Nagalenoj H. <nagalenoj at gmail.com>wrote:
>>
>>> Dear friends,
>>>     I've installed freeswitch trunk - 16729 and tried to configure with
>>> wanpipe for sangoma A102 pri card.
>>>
>>> Followed the steps given in
>>> http://wiki.sangoma.com/wanpipe-freeswitch-install
>>>
>>> When loading the freeswitch, I've got the following error.
>>>
>>> 2010-02-23 14:49:58.545726 [INFO] ozmod_wanpipe.c:335 configuring device
>>> s2c15 as OpenZAP device 1:30 fd:57 DTMF: software
>>> 2010-02-23 14:49:58.546004 [ERR] ozmod_wanpipe.c:436 No valid wanpipe
>>> span and channel was specified
>>> 2010-02-23 14:49:58.546275 [INFO] zap_io.c:2500 Configured 30 channel(s)
>>> 2010-02-23 14:49:58.560724 [ERR] zap_io.c:2560 Error loading
>>> /usr/local/freeswitch/mod/ozmod_sangoma_boost.so
>>> [/usr/local/freeswitch/mod/ozmod_sangoma_boost.so: cannot open shared object
>>> file: No such file or directory]
>>> 2010-02-23 14:49:58.561048 [ERR] zap_io.c:2720 can't find 'sangoma_boost'
>>> 2010-02-23 14:49:58.561266 [ERR] mod_openzap.c:2458 Error starting
>>> OpenZAP span 1 error:
>>> 2010-02-23 14:49:58.561712 [NOTICE] switch_loadable_module.c:144 Adding
>>> Endpoint 'openzap'
>>>
>>> Configuration and log files are pasted to pastebin. Kindly someone help
>>> me to solve this issue.
>>>
>>> openzap.conf and openzap.conf.xml
>>>    http://pastebin.freeswitch.org/12214
>>>
>>> freeswitch log
>>>    http://pastebin.freeswitch.org/12216
>>>
>>> smg_pri.conf
>>>    http://pastebin.freeswitch.org/12217
>>>
>>> --
>>> Regards,
>>> Nagalenoj H.
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Regards,
> Nagalenoj H.
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Regards,
Nagalenoj H.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100226/dec9909b/attachment.html 

From lakindia89 at gmail.com  Thu Feb 25 21:55:29 2010
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Fri, 26 Feb 2010 11:25:29 +0530
Subject: [Freeswitch-users] smg_prid not bridging the call
Message-ID: <7d79b3931002252155t47c86968q9e451482926d93ac@mail.gmail.com>

Dear all,
I'm having a A102 Sangoma hardware. I configured it with freeswitch.
wanrouter status, says both the port as connected.
My smg_prid version is

Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: ================System
restart=============
Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: = Sangoma PRI Protocol Stack
Daemon       =
Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: = Version:
1.54                           =
Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: = Date: Feb 15
2010                       =
Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: = Wanpipe Release:
wanpipe-3.5.8.6       =
Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: = Revision:Revision:
15288                =
Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid:
===========================================

My freeswitch version is 16729.
I started freeswitch.

oz list
+OK
span: 1 (smg_prid)
type: Sangoma (boost)
chan_count: 60
dialplan: XML
context: default
dial_regex:
fail_dial_regex:
hold_music:
analog_options none

I originated a call as
originate openzap/smg_prid/a/9952248266 &park(), which hits my mobile.

But when I issued the following command:
originate openzap/smg_prid/a/9952248266
&bridge(openzap/smg_prid/a/8122133885)
It rings my mobile (9952248266) first, but after that the following error
was displayed

2010-02-26 16:20:51.736080 [ERR] switch_ivr_originate.c:2387 Cannot create
outgoing channel of type [openzap] cause: [NORMAL_CIRCUIT_CONGESTION]
The call got ended in my mobile.

Freeswitch log and smg_pri.conf
          http://pastebin.freeswitch.org/12248
openzap.conf:
[span wanpipe smg_prid]
name => smg_prid
trunk_type =>e1
b-channel => 1:1-15
b-channel => 1:17-31
trunk_type =>e1
b-channel => 2:1-15
b-channel => 2:17-31

openzap.conf.xml:
<configuration name="openzap.conf" description="OpenZAP Configuration">
  <settings>
    <param name="debug" value="0"/>
  </settings>
  <boost_spans>
        <span id="smg_prid">
        <param name="dialplan" value="XML"/>
        <param name="context" value="default"/>
        </span>
  </boost_spans>
</configuration>

Please guide me to setup this one!!.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100226/c62fd7ab/attachment.html 

From david.varnes at gmail.com  Fri Feb 26 02:18:34 2010
From: david.varnes at gmail.com (david varnes)
Date: Fri, 26 Feb 2010 21:18:34 +1100
Subject: [Freeswitch-users] ASR Apps
In-Reply-To: <AC3A71D6-4EAB-4261-B14A-2869C0C05C91@freeswitch.org>
References: <AC3A71D6-4EAB-4261-B14A-2869C0C05C91@freeswitch.org>
Message-ID: <74a861001002260218i38bfbf72s5637ed20f684a40b@mail.gmail.com>

Hi Brian,

I have just started porting an ASR based framework from
a VXML engine to use FS.

It is java based, which I know is not a big focus for the
project ...

Do you have some ASR ports we could use for testing ?
I am very interested ..

davidv

On 26 February 2010 12:22, Brian West <brian at freeswitch.org> wrote:
> I'm looking for some enterprising community members to create some interesting voice apps using ASR. ?Please email me off list and we'll get you what you need to do this.
>
> Thanks,
> Brian
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
david varnes

e: david.varnes at gmail.com
p: +61 404 925 633


From infos at madovsky.org  Fri Feb 26 02:39:37 2010
From: infos at madovsky.org (Madovsky)
Date: Fri, 26 Feb 2010 05:39:37 -0500
Subject: [Freeswitch-users] about pizza demo
Message-ID: <1871231B5C344CB08333565ED2EC2260@MOBILEE1705>

Hi,

I'm trying to change the language of pizza demo script.
is it need to change only words inside addItemAlias() ?

Many thanks

Franck
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100226/b776d532/attachment.html 

From steveayre at gmail.com  Fri Feb 26 04:52:36 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 26 Feb 2010 12:52:36 +0000
Subject: [Freeswitch-users] SIP Trunk with Private Static IP?
In-Reply-To: <009b01cab62b$b8889c10$2999d430$@co.za>
References: <008401cab5f5$4795b910$d6c12b30$@co.za>
	<eafa772f1002250057l3de5b2abu54da8725cbb77281@mail.gmail.com>
	<eafa772f1002250058i2cd7b30ai42f44d0ea1d6eb88@mail.gmail.com>
	<009b01cab62b$b8889c10$2999d430$@co.za>
Message-ID: <eafa772f1002260452h14dd6baavd2a5ac5fa012abd5@mail.gmail.com>

A SIP 'ping' is not a ICMP ping...

It works by sending a OPTIONS SIP request to the gateway, which then
responds with 200 OK. It has the advantage of working even if ICMP is
filtered by a firewall and testing whether the SIP server software is
running, not just whether the server is online.

Best Regards,
-Steve


On 25 February 2010 15:02, Roly Maz <rm at callrica.co.za> wrote:
> Many thanks for your prompt reply and the help
>
> I removed the LAN GW and kept the WAN GW.
>
> I have modified the standard internal and external sip profiles accordingly
>
> What is odd is that if i run a ping from the windows command line, I get a
> reply from the SIP Server. However, if I setup a ping within FS, it fails.
>
> I am investigating...
>
>
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Steven
> Ayre
> Sent: 25 February 2010 10:59 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] SIP Trunk with Private Static IP?
>
> Gateways do not require usernames and passwords. You are required to
> set the parameter, but if no authentication is needed they are ignored
> so you can put anything in the field, so that is not a reason to avoid
> them.
>
> -Steve
>
>
> On 25 February 2010 08:57, Steven Ayre <steveayre at gmail.com> wrote:
>> Create two SIP profiles, each bound to one of your local IPs.
>>
>> You may create a gateway on the profile for the SIP trunk IP for the
>> 10.42.0.1 server, but this is optional.
>>
>> You can then bridge calls via the SIP server using one of:
>> <action application="bridge" data="sofia/gateway/gatewayname"/>
>> <action application="bridge" data="sofia/profilename/number at 10.42.0.1"/>
>>
>> The advantages of using a gateway are:
>> - supports authentication
>> - will monitor the gateway to detect if it goes down (so calls fail
>> instantly rather than after a timeout)
>>
>> As for the default gateway, it is the IP you send via to reach IPs
>> that are not on a network you are connected directly to - you should
>> probably only have one set, and it should be the one you go via to
>> reach the Internet.
>>
>> -Steve
>>
>>
>> On 25 February 2010 08:33, Roly Maz <rm at callrica.co.za> wrote:
>>> Hi Community
>>>
>>>
>>>
>>>
>>>
>>> My Provider provides the following info when they supply a SIP trunk:
>>>
>>>
>>>
>>> ????????? A direct connection into their network. i.e. they provide
> private
>>> IPs:
>>>
>>> ????????? An IP address I must use for my FS box e.g. IP: 10. 42.0.66
> MASK:
>>> 255.255.255.248 GW: 10. 42.0.68
>>>
>>> ????????? An IP address for their SIP server 10.42.0.1
>>>
>>>
>>>
>>> I have setup a dual homed FS box (Windows Server 2008, latest FS version)
>>>
>>>
>>>
>>> NIC 1 ? Internal LAN IP: 10.0.2.185 MASK: 255.255.255.0 GW: 10.0.2.253
>>>
>>> NIC 2 ? SIP Provider (WAN) IP: 10. 42.0.66 MASK: 255.255.255.248 GW: 10.
>>> 42.0.68
>>>
>>>
>>>
>>> Windows complains about multiple gateways ? which I ignore? I can ping
>>> internal addresses ?and the SIP Server
>>>
>>>
>>>
>>> When I fire up FS, I can register Xlite phones on my LAN. I can dial and
>>> hear the test IVR (5000)
>>>
>>>
>>>
>>> This means my Internal SIP Profile is ok.
>>>
>>>
>>>
>>> Now, how do i route a call out to the 10.42.01 SIP Server?
>>>
>>>
>>>
>>> ?Creating a gateway doesn?t make sense, because I am not supplied a
>>> username/password?
>>>
>>>
>>>
>>> Any pointers would be most appreciated, I am sure I am missing something
>>> really simple.
>>>
>>>
>>>
>>> Roland
>>>
>>>
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From rob4manhere at gmail.com  Fri Feb 26 05:10:49 2010
From: rob4manhere at gmail.com (Rob Forman)
Date: Fri, 26 Feb 2010 07:10:49 -0600
Subject: [Freeswitch-users] SIP Trunk with Private Static IP?
In-Reply-To: <eafa772f1002260452h14dd6baavd2a5ac5fa012abd5@mail.gmail.com>
References: <008401cab5f5$4795b910$d6c12b30$@co.za>
	<eafa772f1002250057l3de5b2abu54da8725cbb77281@mail.gmail.com>
	<eafa772f1002250058i2cd7b30ai42f44d0ea1d6eb88@mail.gmail.com>
	<009b01cab62b$b8889c10$2999d430$@co.za>
	<eafa772f1002260452h14dd6baavd2a5ac5fa012abd5@mail.gmail.com>
Message-ID: <AC723E3A-93AD-4FE5-A79D-C59C48197808@gmail.com>

Here's a simple sip ping script (sip_ping.pl) I found and like:

http://pastebin.freeswitch.org/12250

Good luck,
Rob

On Feb 26, 2010, at 6:52 AM, Steven Ayre wrote:

> A SIP 'ping' is not a ICMP ping...
>
> It works by sending a OPTIONS SIP request to the gateway, which then
> responds with 200 OK. It has the advantage of working even if ICMP is
> filtered by a firewall and testing whether the SIP server software is
> running, not just whether the server is online.
>
> Best Regards,
> -Steve
>
>
> On 25 February 2010 15:02, Roly Maz <rm at callrica.co.za> wrote:
>> Many thanks for your prompt reply and the help
>>
>> I removed the LAN GW and kept the WAN GW.
>>
>> I have modified the standard internal and external sip profiles  
>> accordingly
>>
>> What is odd is that if i run a ping from the windows command line,  
>> I get a
>> reply from the SIP Server. However, if I setup a ping within FS, it  
>> fails.
>>
>> I am investigating...
>>
>>
>>
>>
>> -----Original Message-----
>> From: freeswitch-users-bounces at lists.freeswitch.org
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of  
>> Steven
>> Ayre
>> Sent: 25 February 2010 10:59 AM
>> To: freeswitch-users at lists.freeswitch.org
>> Subject: Re: [Freeswitch-users] SIP Trunk with Private Static IP?
>>
>> Gateways do not require usernames and passwords. You are required to
>> set the parameter, but if no authentication is needed they are  
>> ignored
>> so you can put anything in the field, so that is not a reason to  
>> avoid
>> them.
>>
>> -Steve
>>
>>
>> On 25 February 2010 08:57, Steven Ayre <steveayre at gmail.com> wrote:
>>> Create two SIP profiles, each bound to one of your local IPs.
>>>
>>> You may create a gateway on the profile for the SIP trunk IP for the
>>> 10.42.0.1 server, but this is optional.
>>>
>>> You can then bridge calls via the SIP server using one of:
>>> <action application="bridge" data="sofia/gateway/gatewayname"/>
>>> <action application="bridge" data="sofia/profilename/number at 10.42.0.1 
>>> "/>
>>>
>>> The advantages of using a gateway are:
>>> - supports authentication
>>> - will monitor the gateway to detect if it goes down (so calls fail
>>> instantly rather than after a timeout)
>>>
>>> As for the default gateway, it is the IP you send via to reach IPs
>>> that are not on a network you are connected directly to - you should
>>> probably only have one set, and it should be the one you go via to
>>> reach the Internet.
>>>
>>> -Steve
>>>
>>>
>>> On 25 February 2010 08:33, Roly Maz <rm at callrica.co.za> wrote:
>>>> Hi Community
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> My Provider provides the following info when they supply a SIP  
>>>> trunk:
>>>>
>>>>
>>>>
>>>> ?         A direct connection into their network. i.e. they provide
>> private
>>>> IPs:
>>>>
>>>> ?         An IP address I must use for my FS box e.g. IP: 10.  
>>>> 42.0.66
>> MASK:
>>>> 255.255.255.248 GW: 10. 42.0.68
>>>>
>>>> ?         An IP address for their SIP server 10.42.0.1
>>>>
>>>>
>>>>
>>>> I have setup a dual homed FS box (Windows Server 2008, latest FS  
>>>> version)
>>>>
>>>>
>>>>
>>>> NIC 1 ? Internal LAN IP: 10.0.2.185 MASK: 255.255.255.0 GW:  
>>>> 10.0.2.253
>>>>
>>>> NIC 2 ? SIP Provider (WAN) IP: 10. 42.0.66 MASK: 255.255.255.248  
>>>> GW: 10.
>>>> 42.0.68
>>>>
>>>>
>>>>
>>>> Windows complains about multiple gateways ? which I ignore? I can  
>>>> ping
>>>> internal addresses  and the SIP Server
>>>>
>>>>
>>>>
>>>> When I fire up FS, I can register Xlite phones on my LAN. I can  
>>>> dial and
>>>> hear the test IVR (5000)
>>>>
>>>>
>>>>
>>>> This means my Internal SIP Profile is ok.
>>>>
>>>>
>>>>
>>>> Now, how do i route a call out to the 10.42.01 SIP Server?
>>>>
>>>>
>>>>
>>>>  Creating a gateway doesn?t make sense, because I am not supplied a
>>>> username/password?
>>>>
>>>>
>>>>
>>>> Any pointers would be most appreciated, I am sure I am missing  
>>>> something
>>>> really simple.
>>>>
>>>>
>>>>
>>>> Roland
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From javieraristizabal at gmail.com  Fri Feb 26 05:48:45 2010
From: javieraristizabal at gmail.com (=?ISO-8859-1?Q?Javier_Aristiz=E1bal?=)
Date: Fri, 26 Feb 2010 08:48:45 -0500
Subject: [Freeswitch-users] ASR Apps
In-Reply-To: <74a861001002260218i38bfbf72s5637ed20f684a40b@mail.gmail.com>
References: <AC3A71D6-4EAB-4261-B14A-2869C0C05C91@freeswitch.org>
	<74a861001002260218i38bfbf72s5637ed20f684a40b@mail.gmail.com>
Message-ID: <b6f11f301002260548r740ea3fdpd8102d4f2ce1f53@mail.gmail.com>

Hi Brain.. i Want!!! :D

/Javier

On Fri, Feb 26, 2010 at 5:18 AM, david varnes <david.varnes at gmail.com>wrote:

> Hi Brian,
>
> I have just started porting an ASR based framework from
> a VXML engine to use FS.
>
> It is java based, which I know is not a big focus for the
> project ...
>
> Do you have some ASR ports we could use for testing ?
> I am very interested ..
>
> davidv
>
> On 26 February 2010 12:22, Brian West <brian at freeswitch.org> wrote:
> > I'm looking for some enterprising community members to create some
> interesting voice apps using ASR.  Please email me off list and we'll get
> you what you need to do this.
> >
> > Thanks,
> > Brian
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
>
> --
> david varnes
>
> e: david.varnes at gmail.com
> p: +61 404 925 633
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100226/5bae7124/attachment.html 

From moises.silva at gmail.com  Fri Feb 26 07:31:34 2010
From: moises.silva at gmail.com (Moises Silva)
Date: Fri, 26 Feb 2010 10:31:34 -0500
Subject: [Freeswitch-users] smg_prid not bridging the call
In-Reply-To: <7d79b3931002252155t47c86968q9e451482926d93ac@mail.gmail.com>
References: <7d79b3931002252155t47c86968q9e451482926d93ac@mail.gmail.com>
Message-ID: <c4d05cbe1002260731v29dbc428i1e72d24a220c62eb@mail.gmail.com>

Hello lakshmanan,

Please enable the dchan_log=q931 in /etc/wanpipe/smg_pri.conf and then
restart it (smg_ctrl restart), then pastebin the logs

 /var/log/sangoma_pri/dchan_<span-no>.log
/var/log/sangoma_mgd.log

That will contain the Q931 details (if any). Also pastebin your
smg_pri.conf.

Also enable the FreeSWITCH debug logging (see the FreeSWITCH wiki for
details about that) and paste them too.

-- 
Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com

On Fri, Feb 26, 2010 at 12:55 AM, lakshmanan ganapathy <lakindia89 at gmail.com
> wrote:

> Dear all,
> I'm having a A102 Sangoma hardware. I configured it with freeswitch.
> wanrouter status, says both the port as connected.
> My smg_prid version is
>
> Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: ================System
> restart=============
> Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: = Sangoma PRI Protocol Stack
> Daemon       =
> Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: = Version:
> 1.54                           =
> Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: = Date: Feb 15
> 2010                       =
> Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: = Wanpipe Release:
> wanpipe-3.5.8.6       =
> Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: = Revision:Revision:
> 15288                =
> Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid:
> ===========================================
>
> My freeswitch version is 16729.
> I started freeswitch.
>
> oz list
> +OK
> span: 1 (smg_prid)
> type: Sangoma (boost)
> chan_count: 60
> dialplan: XML
> context: default
> dial_regex:
> fail_dial_regex:
> hold_music:
> analog_options none
>
> I originated a call as
> originate openzap/smg_prid/a/9952248266 &park(), which hits my mobile.
>
> But when I issued the following command:
> originate openzap/smg_prid/a/9952248266
> &bridge(openzap/smg_prid/a/8122133885)
> It rings my mobile (9952248266) first, but after that the following error
> was displayed
>
> 2010-02-26 16:20:51.736080 [ERR] switch_ivr_originate.c:2387 Cannot create
> outgoing channel of type [openzap] cause: [NORMAL_CIRCUIT_CONGESTION]
> The call got ended in my mobile.
>
> Freeswitch log and smg_pri.conf
>           http://pastebin.freeswitch.org/12248
> openzap.conf:
> [span wanpipe smg_prid]
> name => smg_prid
> trunk_type =>e1
> b-channel => 1:1-15
> b-channel => 1:17-31
> trunk_type =>e1
> b-channel => 2:1-15
> b-channel => 2:17-31
>
> openzap.conf.xml:
> <configuration name="openzap.conf" description="OpenZAP Configuration">
>   <settings>
>     <param name="debug" value="0"/>
>   </settings>
>   <boost_spans>
>         <span id="smg_prid">
>         <param name="dialplan" value="XML"/>
>         <param name="context" value="default"/>
>         </span>
>   </boost_spans>
> </configuration>
>
> Please guide me to setup this one!!.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100226/0d5cda53/attachment.html 

From christian.loeschenkohl at xpirio.com  Fri Feb 26 08:02:59 2010
From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=)
Date: Fri, 26 Feb 2010 17:02:59 +0100
Subject: [Freeswitch-users] conferences lead to high server load
In-Reply-To: <191c3a031002241058y517e63fdte0d929b97a33f9ac@mail.gmail.com>
References: <4B857226.10308@xpirio.com>
	<191c3a031002241058y517e63fdte0d929b97a33f9ac@mail.gmail.com>
Message-ID: <4B87F0B3.2090606@xpirio.com>

hello

yesterday we did experience high audio delays (2-3 sec) and drops every few seconds.
we had about 70 users in 4 conference rooms, the server had a load of about 40
and used all 4 cpu's (we had a load of 10 with 50 users)

i didn't have the chance to try out -vm so far, the next chance will be this evening
or tomorrow - but i think some change has hit the performance of conferencing very badly.

br

On 2010-02-24 19:58, Anthony Minessale wrote:

> load average has no meaning with FS, you have to look at the CPU usage
> per CPU and thread.
> Are you experiencing any audio problems or are you just concerned about
> that load number?
>
> If you have a box that has trouble with timing it could cost more resources.
> you can always run freeswitch -vm to use an alternate form of timing
> that may not manifest into the load average.
>
>
> 2010/2/24 Christian L?schenkohl <christian.loeschenkohl at xpirio.com
> <mailto:christian.loeschenkohl at xpirio.com>>
>
>     hi
>
>     we do experience a unusual high server load with the latest
>     freeswitch versions.
>     about 50 conference users lead to a server load of over 10 -
>     reproducible by the way.
>     this wans't the case until my latest trunk update.
>
>     fs version: 16714
>     os: debian lenny x86_64
>
>     has something substantially changed in mod_conference recently?
>
>     br
>
>     --
>     Ing. Christian L?schenkohl
>     Technische Leitung, Forschung & Entwicklung VoIP
>
>     xpirio
>     Telekommunikation & Service GmbH
>     Lakeside B04
>     9020 Klagenfurt
>     Austria
>
>     T  +43 (0) 5 77 11 - 1000
>     F  +43 (0) 5 77 11 - 1002
>     E christian.loeschenkohl at xpirio.com
>     <mailto:christian.loeschenkohl at xpirio.com>
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> <mailto:MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> <mailto:sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> <http://iax:guest at conference.freeswitch.org/888>
> googletalk:conf+888 at conference.freeswitch.org
> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


-- 
Ing. Christian L?schenkohl
Technische Leitung, Forschung & Entwicklung VoIP

xpirio
Telekommunikation & Service GmbH
Lakeside B04
9020 Klagenfurt
Austria

T  +43 (0) 5 77 11 - 1000
F  +43 (0) 5 77 11 - 1002
E  christian.loeschenkohl at xpirio.com


From ivanov.maxim at gmail.com  Fri Feb 26 08:23:52 2010
From: ivanov.maxim at gmail.com (Max Ivanov)
Date: Fri, 26 Feb 2010 16:23:52 +0000
Subject: [Freeswitch-users] Multiple gateways dial string and user busy
In-Reply-To: <87f2f3b91002251149m13c0d961k722dd3375f46b6e8@mail.gmail.com>
References: <fbfa7b0c1002240813x58ba8541h7fef5b94f045cbda@mail.gmail.com> 
	<87f2f3b91002251149m13c0d961k722dd3375f46b6e8@mail.gmail.com>
Message-ID: <fbfa7b0c1002260823l11b8d609lf4162eeb61ffc036@mail.gmail.com>

> This is normal for the syntax you're using. You can try setting
> ignore_early_media=true if you don't need call progress tones like ringing
> and busy. It might help to know what the application is before answering
> your question further. What solution are you building?

I use Panasonic station with multiple SIP extensions, each of them is
different gateway in FS. To call panasonics user I use dialstring
sofia/gateway/panas110/223|sofia/gateway/panas111/223|sofia/gateway/panasNNN/223
where 223 is panasonics extension number. Panasonics allows only 1
call per SIP extension, that's why I have to try all of them for each
call, to find first aviable. If SIP extension (but not destination
extension) is occupied by another call it Panasonic return
NO_USER_RESPONSE, if destination extension is busy it returns
USER_BUSY.

Also I use custom calls logging software and each attempt to call to
user appears in logs as tens of call attempts even if user was busy.
What I want to is to stop whole call attempt on USER_BUSY and try next
gateway on NO_USER_RESPONSE. Is it possible? How dialstring have to
look like to achieve that?


From kristian.kielhofner at gmail.com  Fri Feb 26 08:38:15 2010
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Fri, 26 Feb 2010 11:38:15 -0500
Subject: [Freeswitch-users] Multiple gateways dial string and user busy
In-Reply-To: <fbfa7b0c1002260823l11b8d609lf4162eeb61ffc036@mail.gmail.com>
References: <fbfa7b0c1002240813x58ba8541h7fef5b94f045cbda@mail.gmail.com>
	<87f2f3b91002251149m13c0d961k722dd3375f46b6e8@mail.gmail.com>
	<fbfa7b0c1002260823l11b8d609lf4162eeb61ffc036@mail.gmail.com>
Message-ID: <4d15ff861002260838v6aa624a1t398eb50bddc75ab1@mail.gmail.com>

http://wiki.freeswitch.org/wiki/Channel_Variables#failure_causes

On Fri, Feb 26, 2010 at 11:23 AM, Max Ivanov <ivanov.maxim at gmail.com> wrote:
>> This is normal for the syntax you're using. You can try setting
>> ignore_early_media=true if you don't need call progress tones like ringing
>> and busy. It might help to know what the application is before answering
>> your question further. What solution are you building?
>
> I use Panasonic station with multiple SIP extensions, each of them is
> different gateway in FS. To call panasonics user I use dialstring
> sofia/gateway/panas110/223|sofia/gateway/panas111/223|sofia/gateway/panasNNN/223
> where 223 is panasonics extension number. Panasonics allows only 1
> call per SIP extension, that's why I have to try all of them for each
> call, to find first aviable. If SIP extension (but not destination
> extension) is occupied by another call it Panasonic return
> NO_USER_RESPONSE, if destination extension is busy it returns
> USER_BUSY.
>
> Also I use custom calls logging software and each attempt to call to
> user appears in logs as tens of call attempts even if user was busy.
> What I want to is to stop whole call attempt on USER_BUSY and try next
> gateway on NO_USER_RESPONSE. Is it possible? How dialstring have to
> look like to achieve that?
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com


From kristian.kielhofner at gmail.com  Fri Feb 26 08:41:45 2010
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Fri, 26 Feb 2010 11:41:45 -0500
Subject: [Freeswitch-users] SIP Trunk with Private Static IP?
In-Reply-To: <eafa772f1002260452h14dd6baavd2a5ac5fa012abd5@mail.gmail.com>
References: <008401cab5f5$4795b910$d6c12b30$@co.za>
	<eafa772f1002250057l3de5b2abu54da8725cbb77281@mail.gmail.com>
	<eafa772f1002250058i2cd7b30ai42f44d0ea1d6eb88@mail.gmail.com>
	<009b01cab62b$b8889c10$2999d430$@co.za>
	<eafa772f1002260452h14dd6baavd2a5ac5fa012abd5@mail.gmail.com>
Message-ID: <4d15ff861002260841s3ae98464l64244285523bc1ab@mail.gmail.com>

Keep in mind the remote side might not always send a 200 OK.  Some
send 404, 501, 503, etc.  When pinging a gateway (I believe) FS treats
all of them the same.  As long as *something* comes back the gw is
marked "up".  On some software (OpenSIPS with various gw modules, etc)
this behavior is configurable.  Not sure about FS.

On Fri, Feb 26, 2010 at 7:52 AM, Steven Ayre <steveayre at gmail.com> wrote:
> A SIP 'ping' is not a ICMP ping...
>
> It works by sending a OPTIONS SIP request to the gateway, which then
> responds with 200 OK. It has the advantage of working even if ICMP is
> filtered by a firewall and testing whether the SIP server software is
> running, not just whether the server is online.
>
> Best Regards,
> -Steve


-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com


From mike at jerris.com  Fri Feb 26 08:45:22 2010
From: mike at jerris.com (Michael Jerris)
Date: Fri, 26 Feb 2010 11:45:22 -0500
Subject: [Freeswitch-users] Dingaling - external rtp supported?
In-Reply-To: <29b888f81002242204x171f4ab3la34d4ace604cd50a@mail.gmail.com>
References: <29b888f81002160835y6e290319ub0d04c088caa1b79@mail.gmail.com>
	<DEA39AD1-0EE9-473A-B78C-CDDF3F100578@freeswitch.org>
	<29b888f81002170318p63622f3fhc8d26ad0e85602d2@mail.gmail.com>
	<191c3a031002170538r3b029515i8a000812a70ad558@mail.gmail.com>
	<29b888f81002170741y3d13cb57ycb4436a372c4eded@mail.gmail.com>
	<191c3a031002171136q7424e9bai34379abb542daf93@mail.gmail.com>
	<29b888f81002210800v617bca9dk4bf1abf1bd791ee0@mail.gmail.com>
	<29b888f81002222359w6ae4cd8ej944d453a5b099af4@mail.gmail.com>
	<191c3a031002231131x19c9791cgd986a6c1be18e09a@mail.gmail.com>
	<29b888f81002242204x171f4ab3la34d4ace604cd50a@mail.gmail.com>
Message-ID: <67ABA2B8-8335-486A-A43F-0025ECD13D7E@jerris.com>

There is no recommendation because no one has ever contributed a working build for windows, if there was, it would just build and work.


On Feb 25, 2010, at 1:04 AM, Moiz Chinoy wrote:

>  
> I was using GuntTls-2.7.3 for windows. Now I am using GuntTls-2.9.9. I have modified only gnutls.h, added following line:
>  
> typedef long ssize_t;
>  
> because otherwise it was giving errors...
>  
> What is the recommended version of the TLS lib for windows?
>  
> After upgrading the the GnuTls and freeswitch to rev 16806, I ran the freeswitch with mod_dingalilg enabled. Once started, I issued just the 'shutdown' command on the console, exception happened.
>  
> ......................
> 2010-02-25 09:45:29.795285 [CONSOLE] switch_loadable_module.c:1277 Stopping: CORE_SOFTTIMER_MODULE
> 2010-02-25 09:45:29.810910 [CONSOLE] switch_time.c:780 Soft timer thread exiting.
> 2010-02-25 09:45:29.810910 [NOTICE] switch_loadable_module.c:98 Thread ended for CORE_SOFTTIMER_MODULE
> 2010-02-25 09:45:29.826535 [DEBUG] switch_loadable_module.c:456 Write lock interface 'dingaling' to wait for existing references.
> 2010-02-25 09:45:29.826535 [NOTICE] switch_loadable_module.c:464 Deleting Endpoint 'dingaling'
> 2010-02-25 09:45:29.826535 [NOTICE] switch_loadable_module.c:572 Deleting API Function 'dl_debug'
> 2010-02-25 09:45:29.826535 [DEBUG] switch_loadable_module.c:574 Write lock interface 'dl_debug' to wait for existing references.
> 2010-02-25 09:45:29.826535 [NOTICE] switch_loadable_module.c:572 Deleting API Function 'dl_pres'
> 2010-02-25 09:45:29.826535 [DEBUG] switch_loadable_module.c:574 Write lock interface 'dl_pres' to wait for existing references.
> 2010-02-25 09:45:29.826535 [NOTICE] switch_loadable_module.c:572 Deleting API Function 'dl_logout'
> 2010-02-25 09:45:29.826535 [DEBUG] switch_loadable_module.c:574 Write lock interface 'dl_logout' to wait for existing references.
> 2010-02-25 09:45:29.826535 [NOTICE] switch_loadable_module.c:572 Deleting API Function 'dl_login'
> 2010-02-25 09:45:29.826535 [DEBUG] switch_loadable_module.c:574 Write lock interface 'dl_login' to wait for existing references.
> 2010-02-25 09:45:29.826535 [NOTICE] switch_loadable_module.c:572 Deleting API Function 'dingaling'
> 2010-02-25 09:45:29.826535 [DEBUG] switch_loadable_module.c:574 Write lock interface 'dingaling' to wait for existing references.
> 2010-02-25 09:45:29.826535 [DEBUG] switch_loadable_module.c:710 Write lock interface 'jingle' to wait for existing references.
> 2010-02-25 09:45:29.826535 [NOTICE] switch_loadable_module.c:719 Deleting Chat interface 'jingle'
> 2010-02-25 09:45:29.826535 [CONSOLE] switch_loadable_module.c:1277 Stopping: mod_dingaling
> 2010-02-25 09:45:31.185910 [DEBUG] libdingaling.c:1546 io error 2 7 retry in 1 second(s)
> ........................
>  
> And the code went in the stream.c...
>  
> int
> iks_fd (iksparser *prs)
> {
> struct stream_data *data;
>  
> if (prs) {
> data = iks_user_data (prs);
> if (data) {
> return (int) data->sock;
> }
> }
> return -1;
> }
>  
>  
>  
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100226/45e519e0/attachment.html 

From anthony.minessale at gmail.com  Fri Feb 26 08:54:08 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 26 Feb 2010 10:54:08 -0600
Subject: [Freeswitch-users] Dingaling - external rtp supported?
In-Reply-To: <67ABA2B8-8335-486A-A43F-0025ECD13D7E@jerris.com>
References: <29b888f81002160835y6e290319ub0d04c088caa1b79@mail.gmail.com>
	<29b888f81002170318p63622f3fhc8d26ad0e85602d2@mail.gmail.com>
	<191c3a031002170538r3b029515i8a000812a70ad558@mail.gmail.com>
	<29b888f81002170741y3d13cb57ycb4436a372c4eded@mail.gmail.com>
	<191c3a031002171136q7424e9bai34379abb542daf93@mail.gmail.com>
	<29b888f81002210800v617bca9dk4bf1abf1bd791ee0@mail.gmail.com>
	<29b888f81002222359w6ae4cd8ej944d453a5b099af4@mail.gmail.com>
	<191c3a031002231131x19c9791cgd986a6c1be18e09a@mail.gmail.com>
	<29b888f81002242204x171f4ab3la34d4ace604cd50a@mail.gmail.com>
	<67ABA2B8-8335-486A-A43F-0025ECD13D7E@jerris.com>
Message-ID: <191c3a031002260854q550618bbj8c0e54a6bb5ded54@mail.gmail.com>

its probably not an upgrade you need, more likely a downgrade (1 or 2 years
ago version), and we have no idea, as mike said, nobody has contributed it
back in working order so maybe you can ask whoever showed you how to add
your own gnutls how they did it.


On Fri, Feb 26, 2010 at 10:45 AM, Michael Jerris <mike at jerris.com> wrote:

> There is no recommendation because no one has ever contributed a working
> build for windows, if there was, it would just build and work.
>
>
> On Feb 25, 2010, at 1:04 AM, Moiz Chinoy wrote:
>
>
> I was using GuntTls-2.7.3 for windows. Now I am using GuntTls-2.9.9. I have
> modified only gnutls.h, added following line:
>
> typedef long ssize_t;
>
> because otherwise it was giving errors...
>
> What is the recommended version of the TLS lib for windows?
>
> After upgrading the the GnuTls and freeswitch to rev 16806, I ran the
> freeswitch with mod_dingalilg enabled. Once started, I issued just the
> 'shutdown' command on the console, exception happened.
>
> ......................
> 2010-02-25 09:45:29.795285 [CONSOLE] switch_loadable_module.c:1277
> Stopping: CORE_SOFTTIMER_MODULE
> 2010-02-25 09:45:29.810910 [CONSOLE] switch_time.c:780 Soft timer thread
> exiting.
> 2010-02-25 09:45:29.810910 [NOTICE] switch_loadable_module.c:98 Thread
> ended for CORE_SOFTTIMER_MODULE
> 2010-02-25 09:45:29.826535 [DEBUG] switch_loadable_module.c:456 Write lock
> interface 'dingaling' to wait for existing references.
> 2010-02-25 09:45:29.826535 [NOTICE] switch_loadable_module.c:464 Deleting
> Endpoint 'dingaling'
> 2010-02-25 09:45:29.826535 [NOTICE] switch_loadable_module.c:572 Deleting
> API Function 'dl_debug'
> 2010-02-25 09:45:29.826535 [DEBUG] switch_loadable_module.c:574 Write lock
> interface 'dl_debug' to wait for existing references.
> 2010-02-25 09:45:29.826535 [NOTICE] switch_loadable_module.c:572 Deleting
> API Function 'dl_pres'
> 2010-02-25 09:45:29.826535 [DEBUG] switch_loadable_module.c:574 Write lock
> interface 'dl_pres' to wait for existing references.
> 2010-02-25 09:45:29.826535 [NOTICE] switch_loadable_module.c:572 Deleting
> API Function 'dl_logout'
> 2010-02-25 09:45:29.826535 [DEBUG] switch_loadable_module.c:574 Write lock
> interface 'dl_logout' to wait for existing references.
> 2010-02-25 09:45:29.826535 [NOTICE] switch_loadable_module.c:572 Deleting
> API Function 'dl_login'
> 2010-02-25 09:45:29.826535 [DEBUG] switch_loadable_module.c:574 Write lock
> interface 'dl_login' to wait for existing references.
> 2010-02-25 09:45:29.826535 [NOTICE] switch_loadable_module.c:572 Deleting
> API Function 'dingaling'
> 2010-02-25 09:45:29.826535 [DEBUG] switch_loadable_module.c:574 Write lock
> interface 'dingaling' to wait for existing references.
> 2010-02-25 09:45:29.826535 [DEBUG] switch_loadable_module.c:710 Write lock
> interface 'jingle' to wait for existing references.
> 2010-02-25 09:45:29.826535 [NOTICE] switch_loadable_module.c:719 Deleting
> Chat interface 'jingle'
> 2010-02-25 09:45:29.826535 [CONSOLE] switch_loadable_module.c:1277
> Stopping: mod_dingaling
> 2010-02-25 09:45:31.185910 [DEBUG] libdingaling.c:1546 io error 2 7 retry
> in 1 second(s)
> ........................
>
> And the code went in the stream.c...
>
> int
> iks_fd (iksparser *prs)
> {
> struct stream_data *data;
>
> if (prs) {
> data = iks_user_data (prs);
> if (data) {
> return (int) data->sock;
> }
> }
> return -1;
> }
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100226/32e8a5a2/attachment.html 

From Suneel.Papineni at mettoni.com  Fri Feb 26 08:59:36 2010
From: Suneel.Papineni at mettoni.com (Suneel Papineni)
Date: Fri, 26 Feb 2010 16:59:36 -0000
Subject: [Freeswitch-users] FSComm basic issue
Message-ID: <3181A30B8C35AB4AA8577B78DDF4613806886903@nickel.mettonigroup.com>

Hi,

 

I am trying to use FSComm with Freeswitch and facing following issues. 

 

1.       Using pre-build binary (windows), when the application is
started FSComm is getting Registered properly. When I tried to make a
call, UI displays Dialing... but unable to see any SIP (INVITE) messages
in wireshark traces. After sometime UI displays with message "Call with
(destination number) failed with reason DESTINATION_OUT_OF_ORDER though
destination number is registered with another FSComm"

2.       Also I am unable to see any logs generated in the log folder.

 

Downloaded the latest source code (Freeswitch 1.0.5 latest updated as on
26/02/10 at 4am) and tried to build FSComm. Build was succeeded.
Application (FSComm) also started and displayed with UI. When I try to
change the preferences, it has thrown Porta Audio Error saying "Error
Querying Audio Devices" even though proper audio devices are present.
Also it doesn't create folders like "conf", "mod". Even after copying
all the required dll's and mod files (as specified in FSComm wiki
pages), application is throwing the same error.

 

I am using Windows XP machine. Built a Debug & Release version with
32-bit option.

If someone has built FSComm for windows environment and is working fine,
could you please let me know if there are any additional things I need
to do to make it work.

 

Thanks & Regards

Suneel

 

 


*************************************************************************
Please consider the environment before printing this e-mail
*************************************************************************
This email and any files transmitted with it are confidential and
intended solely for the use of the individual or entity to whom they
are addressed. If you have received this email in error please notify
the system manager.  http://www.mettoni.com

Mettoni Ltd
Registered in England and Wales: 4485956
9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN
*************************************************************************

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100226/f188c6ac/attachment-0001.html 

From mbsip at gazeta.pl  Fri Feb 26 09:10:30 2010
From: mbsip at gazeta.pl (Maciej Bylica)
Date: Fri, 26 Feb 2010 18:10:30 +0100
Subject: [Freeswitch-users] Phrases - Can't find macro
Message-ID: <28f27f5d1002260910h7b3ce773l28294e7354078b05@mail.gmail.com>

Hello,

I am playing around with Phrases to use them with conference application.
But i've encountered rudimentary problem of how to use newly added macro.

What I already did is (according to wiki Speech Phrase Management)
http://wiki.freeswitch.org/wiki/Speech_Phrase_Management
- confirmed that there is "mod_say_en" loaded (<load module="mod_say_en"/>)
- confirmed that there are proper .wav files
- modified onf/lang/en/en.xml file:
 <section name="phrases" description="Speech Phrase Management">
   <macros>
     <language name="en" sound_path="$${sounds_dir}/en/us/callie"
tts_engine="cepstral" tts_voice="callie">
       <macro name="confwelcome">
         <input pattern="(.*)">
           <match>
              <action function="play-file" data="vm-youhave.wav"/>
<!-- just to test the phrase-->
           </match>
     </language>
   </macros>
 </section>
- modified a part of dialplan:
    <extension name="conference_48112223344">
      <condition field="destination_number" expression="^(\d{11})$">
         <action application="phrase" data="confwelcome"/>
      </condition>
     </extension>

I have following outcome:
2010-02-26 19:57:09.487245 [DEBUG] switch_core_session.c:638 Send
signal sofia/internal/1000 at 217.153.192.36 [BREAK]
EXECUTE sofia/internal/1000 at 217.153.192.36 phrase(confwelcome)
2010-02-26 19:57:09.487245 [DEBUG] mod_dptools.c:1850 Execute confwelcome() lang
2010-02-26 19:57:09.487245 [DEBUG] switch_ivr_play_say.c:118 No
language specified - Using [en]
2010-02-26 19:57:09.496322 [ERR] switch_ivr_play_say.c:202 Can't find
macro confwelcome.
2010-02-26 19:57:09.496322 [WARNING] switch_ivr_play_say.c:368 Macro
[confwelcome] did not match any patterns



Strange is that if i use a wiki example, it works. To be more precise:
- conf/lang/en/en.xml file was overwritten with an example macros
(directly from aforementioned wiki).
- dialplan was modified:
    <extension name="conference_48112223344">
      <condition field="destination_number" expression="^(\d{11})$">
         <action application="phrase" data="msgcount"/>
      </condition>
     </extension>


Am i doing something wrong?

Thx,
Maciej.


From lawwton at gmail.com  Fri Feb 26 10:19:00 2010
From: lawwton at gmail.com (Alfredo Quiroga-Villamil)
Date: Fri, 26 Feb 2010 13:19:00 -0500
Subject: [Freeswitch-users] Conference - Originate Question
Message-ID: <5fe6fa8f1002261019p3812ce5bq297e91f7ddeda1ab@mail.gmail.com>

All:

I am currently using the following cmd to dynamically create a conference:

originate sofia/gateway/url/1XXXXXXXXXX &conference(myconf)

I have noticed that when I send that cmd even if I specify:

originate sofia/gateway/url/1XXXXXXXXXX &conference(myconf) public

I am not hitting the dialplan. Is there a way to send the command and
force it to hit the dialplan?

Thanks in advance,

Alfredo


From rupa at rupa.com  Fri Feb 26 10:50:48 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Fri, 26 Feb 2010 12:50:48 -0600
Subject: [Freeswitch-users] Conference - Originate Question
In-Reply-To: <5fe6fa8f1002261019p3812ce5bq297e91f7ddeda1ab@mail.gmail.com>
References: <5fe6fa8f1002261019p3812ce5bq297e91f7ddeda1ab@mail.gmail.com>
Message-ID: <b39273531002261050m327590bfs637bd9fff1f40391@mail.gmail.com>

Use the loopback endpoint to have it go back through the dialplan.

http://wiki.freeswitch.org/wiki/Loopback

On Fri, Feb 26, 2010 at 12:19 PM, Alfredo Quiroga-Villamil <
lawwton at gmail.com> wrote:

> All:
>
> I am currently using the following cmd to dynamically create a conference:
>
> originate sofia/gateway/url/1XXXXXXXXXX &conference(myconf)
>
> I have noticed that when I send that cmd even if I specify:
>
> originate sofia/gateway/url/1XXXXXXXXXX &conference(myconf) public
>
> I am not hitting the dialplan. Is there a way to send the command and
> force it to hit the dialplan?
>
> Thanks in advance,
>
> Alfredo
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
-Rupa
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100226/3c68ca16/attachment.html 

From phunk0000 at hotmail.com  Fri Feb 26 10:53:25 2010
From: phunk0000 at hotmail.com (Todd)
Date: Fri, 26 Feb 2010 13:53:25 -0500
Subject: [Freeswitch-users] actition after a set time during call
Message-ID: <BLU110-DS12F13B3418ABF2053C86AAA73F0@phx.gbl>

Hey List-

 

I want to have nibblebill pause after a certain time during a call.  I was
wondering what the best way to put this into the dialplan is?  Still kind of
new to this.. 
 
 <action application="nibblebill" data="pause"/>
 
Is the action I want to implement 2 minutes into a call. what is the best
way to do this?  I have the nibblerate set in the individual extension XMLs
and the nibblerate heartbeat set in the nibble.conf.xml  Thanks

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100226/3e630b68/attachment.html 

From jerry.richards at teotech.com  Fri Feb 26 10:57:06 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Fri, 26 Feb 2010 10:57:06 -0800
Subject: [Freeswitch-users] 415 Unsupported Media Handling
Message-ID: <AE66B63BAF424BBCBE585E244E0C42B4@greyhawk.tonecommander.com>


I have two types of devices, one supports text/html MESSAGE content and one
that only supports text/plain MESSAGE content.  When I send an IM from the
first to the second, the second replies with 415 Unsupported Media Type (as
shown below).  Section 8.1.3.5 of RFC 3261 says the sender should retry
using the media type acceptable to the receiver (in this case: plain/text).

The problem I have is that Freeswitch doesn't pass the error back to the
sender (nor does it retry itself using plain/text).  So the IM is lost.
Does anyone see the reason why the error is not being handled correctly?

   ------------------------------------------------------------------------
send 668 bytes to udp/[192.168.72.141]:5062 at 18:43:24.720446:
   ------------------------------------------------------------------------
   MESSAGE sip:5398 at 192.168.72.141:5062 SIP/2.0
   Via: SIP/2.0/UDP 192.168.72.141;rport;branch=z9hG4bKXg6SNUcQa5Kre
   Max-Forwards: 70
   From: "5382 on 141" <sip:5382 at 192.168.72.141>;tag=66661130
   To: "5398" <sip:5398 at 192.168.72.141:5062>
   Call-ID: a9891a15-9da9-122d-b3ad-003048d7e9f0
   CSeq: 127444135 MESSAGE
   Contact: <sip:mod_sofia at 192.168.72.141:5060>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-32M
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: timer, precondition, path, replaces
   Content-Type: text/html
   Content-Length: 63

   <font size="2" face="Arial">hello this is Jerry from Teo</font>
   ------------------------------------------------------------------------
recv 459 bytes from udp/[192.168.72.141]:5062 at 18:43:24.728390:
   ------------------------------------------------------------------------
   SIP/2.0 415 Unsupported media type
   Via: SIP/2.0/UDP
192.168.72.141;rport=5060;branch=z9hG4bKXg6SNUcQa5Kre;received=192.168.72.14
1
   From: "5382 on 141" <sip:5382 at 192.168.72.141>;tag=66661130
   To: "5398" <sip:5398 at 192.168.72.141:5062>
   Call-ID: a9891a15-9da9-122d-b3ad-003048d7e9f0
   CSeq: 127444135 MESSAGE
   Date: Fri, 26 Feb 2010 18:43:24 GMT
   User-Agent: MobilityGateway-2.0.34078
   Server: MobilityGateway-2.0.34078
   Accept: text/plain
   Content-Length: 0

Best Regards,
Jerry



From brian at freeswitch.org  Fri Feb 26 11:01:04 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 26 Feb 2010 13:01:04 -0600
Subject: [Freeswitch-users] 415 Unsupported Media Handling
In-Reply-To: <AE66B63BAF424BBCBE585E244E0C42B4@greyhawk.tonecommander.com>
References: <AE66B63BAF424BBCBE585E244E0C42B4@greyhawk.tonecommander.com>
Message-ID: <835A36B2-1C04-42B0-86C1-36528E93589D@freeswitch.org>

Its really clear here you'll need to say text/plain in the content type their accept header says they only take text/plain.

/b

On Feb 26, 2010, at 12:57 PM, Jerry Richards wrote:

> 
> I have two types of devices, one supports text/html MESSAGE content and one
> that only supports text/plain MESSAGE content.  When I send an IM from the
> first to the second, the second replies with 415 Unsupported Media Type (as
> shown below).  Section 8.1.3.5 of RFC 3261 says the sender should retry
> using the media type acceptable to the receiver (in this case: plain/text).
> 
> The problem I have is that Freeswitch doesn't pass the error back to the
> sender (nor does it retry itself using plain/text).  So the IM is lost.
> Does anyone see the reason why the error is not being handled correctly?
> 
>   ------------------------------------------------------------------------
> send 668 bytes to udp/[192.168.72.141]:5062 at 18:43:24.720446:
>   ------------------------------------------------------------------------
>   MESSAGE sip:5398 at 192.168.72.141:5062 SIP/2.0
>   Via: SIP/2.0/UDP 192.168.72.141;rport;branch=z9hG4bKXg6SNUcQa5Kre
>   Max-Forwards: 70
>   From: "5382 on 141" <sip:5382 at 192.168.72.141>;tag=66661130
>   To: "5398" <sip:5398 at 192.168.72.141:5062>
>   Call-ID: a9891a15-9da9-122d-b3ad-003048d7e9f0
>   CSeq: 127444135 MESSAGE
>   Contact: <sip:mod_sofia at 192.168.72.141:5060>
>   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-32M
>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>   Supported: timer, precondition, path, replaces
>   Content-Type: text/html
>   Content-Length: 63
> 
>   <font size="2" face="Arial">hello this is Jerry from Teo</font>
>   ------------------------------------------------------------------------
> recv 459 bytes from udp/[192.168.72.141]:5062 at 18:43:24.728390:
>   ------------------------------------------------------------------------
>   SIP/2.0 415 Unsupported media type
>   Via: SIP/2.0/UDP
> 192.168.72.141;rport=5060;branch=z9hG4bKXg6SNUcQa5Kre;received=192.168.72.14
> 1
>   From: "5382 on 141" <sip:5382 at 192.168.72.141>;tag=66661130
>   To: "5398" <sip:5398 at 192.168.72.141:5062>
>   Call-ID: a9891a15-9da9-122d-b3ad-003048d7e9f0
>   CSeq: 127444135 MESSAGE
>   Date: Fri, 26 Feb 2010 18:43:24 GMT
>   User-Agent: MobilityGateway-2.0.34078
>   Server: MobilityGateway-2.0.34078
>   Accept: text/plain
>   Content-Length: 0
> 
> Best Regards,
> Jerry
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From lawwton at gmail.com  Fri Feb 26 11:24:03 2010
From: lawwton at gmail.com (Alfredo Quiroga-Villamil)
Date: Fri, 26 Feb 2010 14:24:03 -0500
Subject: [Freeswitch-users] Conference - Originate Question
In-Reply-To: <b39273531002261050m327590bfs637bd9fff1f40391@mail.gmail.com>
References: <5fe6fa8f1002261019p3812ce5bq297e91f7ddeda1ab@mail.gmail.com>
	<b39273531002261050m327590bfs637bd9fff1f40391@mail.gmail.com>
Message-ID: <5fe6fa8f1002261124o27b5921fl7ca64ba5b32e9c92@mail.gmail.com>

Thanks Rupa.

On Fri, Feb 26, 2010 at 1:50 PM, Rupa Schomaker <rupa at rupa.com> wrote:
> Use the loopback endpoint to have it go back through the dialplan.
> http://wiki.freeswitch.org/wiki/Loopback
>
> On Fri, Feb 26, 2010 at 12:19 PM, Alfredo Quiroga-Villamil
> <lawwton at gmail.com> wrote:
>>
>> All:
>>
>> I am currently using the following cmd to dynamically create a conference:
>>
>> originate sofia/gateway/url/1XXXXXXXXXX &conference(myconf)
>>
>> I have noticed that when I send that cmd even if I specify:
>>
>> originate sofia/gateway/url/1XXXXXXXXXX &conference(myconf) public
>>
>> I am not hitting the dialplan. Is there a way to send the command and
>> force it to hit the dialplan?
>>
>> Thanks in advance,
>>
>> Alfredo
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From msc at freeswitch.org  Fri Feb 26 11:28:28 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 26 Feb 2010 11:28:28 -0800
Subject: [Freeswitch-users] Cannot make calls through PRI via wanpipe
	running as daemon
In-Reply-To: <6FCED74CF54F466AA25CBDA7DFF42198@greyhawk.tonecommander.com>
References: <6FCED74CF54F466AA25CBDA7DFF42198@greyhawk.tonecommander.com>
Message-ID: <87f2f3b91002261128n17ea78acx160fcfdb04532004@mail.gmail.com>

On Thu, Feb 25, 2010 at 5:16 PM, Robert Hadley <robert.hadley at teotech.com>wrote:

>  When running Freeswitch as service called teoswitch as user teoswitch I
> cannot make calls through the Sangoma PRI or analog cards using wanpipe
> driver.  I have added a file called 30-wanpipe.rules to /etc/udev/rules.d
> and rebooted the server.
>
>
>
> cat 30-wanpipe.rules
>
> # /etc/udev/rules.d/30-wanpipe.rules
>
> SUBSYSTEM=="wptdm", OWNER="teoswitch", GROUP="teoswitch", MODE="0660"
>
> SUBSYSTEM=="wanpipe", OWNER="teoswitch", GROUP="teoswitch", MODE="0660"
>
>
>
>
>
> Freeswitch log:
>
> Dialplan: sofia/internal/5410 at 192.168.72.45:5060 parsing
> [default->SangomaPRI] continue=false
>
> Dialplan: sofia/internal/5410 at 192.168.72.45:5060 Regex (PASS) [SangomaPRI]
> destination_number(93491045) =~ /^9(\d+)$/ break=on-false
>
> Dialplan: sofia/internal/5410 at 192.168.72.45:5060 Action
> set(effective_caller_id_number=425740${caller_id_number})
>
> Dialplan: sofia/internal/5410 at 192.168.72.45:5060 Action
> bridge(openzap/smg_prid/a/3491045 at g1)
>
> 2010-02-25 16:51:11.328635 [DEBUG] switch_core_state_machine.c:122
> (sofia/internal/5410 at 192.168.72.45:5060) State Change CS_ROUTING ->
> CS_EXECUTE
>
> 2010-02-25 16:51:11.328635 [DEBUG] switch_core_session.c:1019 Send signal
> sofia/internal/5410 at 192.168.72.45:5060 [BREAK]
>
> 2010-02-25 16:51:11.328635 [DEBUG] switch_core_state_machine.c:341
> (sofia/internal/5410 at 192.168.72.45:5060) State ROUTING going to sleep
>
> 2010-02-25 16:51:11.328635 [DEBUG] switch_core_state_machine.c:314
> (sofia/internal/5410 at 192.168.72.45:5060) Running State Change CS_EXECUTE
>
> 2010-02-25 16:51:11.328635 [DEBUG] switch_core_state_machine.c:348
> (sofia/internal/5410 at 192.168.72.45:5060) State EXECUTE
>
> 2010-02-25 16:51:11.328635 [DEBUG] mod_sofia.c:181 sofia/internal/
> 5410 at 192.168.72.45:5060 SOFIA EXECUTE
>
> 2010-02-25 16:51:11.328635 [DEBUG] switch_core_state_machine.c:159
> sofia/internal/5410 at 192.168.72.45:5060 Standard EXECUTE
>
> EXECUTE sofia/internal/5410 at 192.168.72.45:5060set(effective_caller_id_number=4257405410)
>
> 2010-02-25 16:51:11.328635 [DEBUG] mod_dptools.c:811 sofia/internal/
> 5410 at 192.168.72.45:5060 SET [effective_caller_id_number]=[4257405410]
>
> EXECUTE sofia/internal/5410 at 192.168.72.45:5060bridge(openzap/smg_prid/a/3491045 at g1
> )
>
> 2010-02-25 16:51:11.339637 [ERR] switch_core_session.c:357 Could not locate
> channel type openzap
>
> 2010-02-25 16:51:11.339637 [ERR] switch_ivr_originate.c:2411 Cannot create
> outgoing channel of type [openzap] cause: [CHAN_NOT_IMPLEMENTED]
>
> 2010-02-25 16:51:11.339637 [DEBUG] switch_ivr_originate.c:3209 Originate
> Resulted in Error Cause: 66 [CHAN_NOT_IMPLEMENTED]
>
> 2010-02-25 16:51:11.339637 [INFO] mod_dptools.c:2346 Originate Failed.
> Cause: CHAN_NOT_IMPLEMENTED
>

CHAN_NOT_IMPLEMENTED implies that OpenZAP did not load. Capture the output
of "load mod_openzap" and look for the reason that it is failing to load.
-MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100226/c722137c/attachment-0001.html 

From jerry.richards at teotech.com  Fri Feb 26 11:43:48 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Fri, 26 Feb 2010 11:43:48 -0800
Subject: [Freeswitch-users] 415 Unsupported Media Handling
In-Reply-To: <835A36B2-1C04-42B0-86C1-36528E93589D@freeswitch.org>
References: <AE66B63BAF424BBCBE585E244E0C42B4@greyhawk.tonecommander.com>
	<835A36B2-1C04-42B0-86C1-36528E93589D@freeswitch.org>
Message-ID: <ADCCADE853D6427CBC7CD978E7E0A0ED@greyhawk.tonecommander.com>

I'm not sure I follow your comment.  The first device prefers text/html so
that's what it normally sets in the initial MESSAGE.  Devices that support
text/html will not generate this 415 error reply.  It's only devices that
don't support it that would send the 415 reply, so the issue is that the 415
is not getting passed back to the originator.

Best Regards,
Jerry


-----Original Message-----
From: Brian West [mailto:brian at freeswitch.org] 
Sent: Friday, February 26, 2010 11:01 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] 415 Unsupported Media Handling

Its really clear here you'll need to say text/plain in the content type
their accept header says they only take text/plain.

/b

On Feb 26, 2010, at 12:57 PM, Jerry Richards wrote:

> 
> I have two types of devices, one supports text/html MESSAGE content 
> and one that only supports text/plain MESSAGE content.  When I send an 
> IM from the first to the second, the second replies with 415 
> Unsupported Media Type (as shown below).  Section 8.1.3.5 of RFC 3261 
> says the sender should retry using the media type acceptable to the
receiver (in this case: plain/text).
> 
> The problem I have is that Freeswitch doesn't pass the error back to 
> the sender (nor does it retry itself using plain/text).  So the IM is
lost.
> Does anyone see the reason why the error is not being handled correctly?
> 
>   
> ----------------------------------------------------------------------
> -- send 668 bytes to udp/[192.168.72.141]:5062 at 18:43:24.720446:
>   ------------------------------------------------------------------------
>   MESSAGE sip:5398 at 192.168.72.141:5062 SIP/2.0
>   Via: SIP/2.0/UDP 192.168.72.141;rport;branch=z9hG4bKXg6SNUcQa5Kre
>   Max-Forwards: 70
>   From: "5382 on 141" <sip:5382 at 192.168.72.141>;tag=66661130
>   To: "5398" <sip:5398 at 192.168.72.141:5062>
>   Call-ID: a9891a15-9da9-122d-b3ad-003048d7e9f0
>   CSeq: 127444135 MESSAGE
>   Contact: <sip:mod_sofia at 192.168.72.141:5060>
>   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-32M
>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, 
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>   Supported: timer, precondition, path, replaces
>   Content-Type: text/html
>   Content-Length: 63
> 
>   <font size="2" face="Arial">hello this is Jerry from Teo</font>
>   
> ----------------------------------------------------------------------
> -- recv 459 bytes from udp/[192.168.72.141]:5062 at 18:43:24.728390:
>   ------------------------------------------------------------------------
>   SIP/2.0 415 Unsupported media type
>   Via: SIP/2.0/UDP
> 192.168.72.141;rport=5060;branch=z9hG4bKXg6SNUcQa5Kre;received=192.168
> .72.14
> 1
>   From: "5382 on 141" <sip:5382 at 192.168.72.141>;tag=66661130
>   To: "5398" <sip:5398 at 192.168.72.141:5062>
>   Call-ID: a9891a15-9da9-122d-b3ad-003048d7e9f0
>   CSeq: 127444135 MESSAGE
>   Date: Fri, 26 Feb 2010 18:43:24 GMT
>   User-Agent: MobilityGateway-2.0.34078
>   Server: MobilityGateway-2.0.34078
>   Accept: text/plain
>   Content-Length: 0
> 
> Best Regards,
> Jerry
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> rs
> http://www.freeswitch.org






From ben at langfeld.co.uk  Fri Feb 26 10:41:14 2010
From: ben at langfeld.co.uk (Ben Langfeld)
Date: Fri, 26 Feb 2010 18:41:14 +0000
Subject: [Freeswitch-users] Freeswitch SPA3000 HUP Not Sent
Message-ID: <fb4ab1e41002261041v381dd63ejafac36712f33cf96@mail.gmail.com>

Hey,

I have a small 7 seat mostly softphone based PBX installation in place using
freeswitch and a couple of SPA3000s for PSTN termination. Currently, aside
from PSTN HUP detection, outbound calls are fine setup to directly dial the
SPAs. When the internal VoIP side hangs up, freeswitch sends NORMAL_CLEARING
to the ATA and the ATA drops the call. Lovely.

Unfortunately inbound calls aren't so successful. Calls are sent using the
SPA dialplan to an internal freeswitch extension, and the calls connect
fine. This time, when the VoIP side hangs up, no NORMAL_CLEARING is sent to
the ATA. Is there a reason for this?

Can anyone give me any idea how I can get freeswitch to instruct the SPA to
drop the line?

Regards,
Ben Langfeld

Wave > Email
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100226/4ca31ca4/attachment.html 

From anthony.minessale at gmail.com  Fri Feb 26 12:01:23 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 26 Feb 2010 14:01:23 -0600
Subject: [Freeswitch-users] Freeswitch SPA3000 HUP Not Sent
In-Reply-To: <fb4ab1e41002261041v381dd63ejafac36712f33cf96@mail.gmail.com>
References: <fb4ab1e41002261041v381dd63ejafac36712f33cf96@mail.gmail.com>
Message-ID: <191c3a031002261201s40a7a53cyfbe30529b77918eb@mail.gmail.com>

you would have to be more specific.
We would need to have you test this on latest SVN trunk with a full debug
log.

enter the following at your cli and reproduce the call saving all the
output.

console loglevel debug
sofia profile internal siptrace on



On Fri, Feb 26, 2010 at 12:41 PM, Ben Langfeld <ben at langfeld.co.uk> wrote:

> Hey,
>
> I have a small 7 seat mostly softphone based PBX installation in place
> using freeswitch and a couple of SPA3000s for PSTN termination. Currently,
> aside from PSTN HUP detection, outbound calls are fine setup to directly
> dial the SPAs. When the internal VoIP side hangs up, freeswitch sends
> NORMAL_CLEARING to the ATA and the ATA drops the call. Lovely.
>
> Unfortunately inbound calls aren't so successful. Calls are sent using the
> SPA dialplan to an internal freeswitch extension, and the calls connect
> fine. This time, when the VoIP side hangs up, no NORMAL_CLEARING is sent to
> the ATA. Is there a reason for this?
>
> Can anyone give me any idea how I can get freeswitch to instruct the SPA to
> drop the line?
>
> Regards,
> Ben Langfeld
>
> Wave > Email
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100226/a5da23c2/attachment.html 

From robert.hadley at teotech.com  Fri Feb 26 12:36:47 2010
From: robert.hadley at teotech.com (Robert Hadley)
Date: Fri, 26 Feb 2010 12:36:47 -0800
Subject: [Freeswitch-users] Cannot make calls through PRI via
	wanpiperunning as daemon
In-Reply-To: <87f2f3b91002261128n17ea78acx160fcfdb04532004@mail.gmail.com>
References: <6FCED74CF54F466AA25CBDA7DFF42198@greyhawk.tonecommander.com>
	<87f2f3b91002261128n17ea78acx160fcfdb04532004@mail.gmail.com>
Message-ID: <D98570990911498788224B48CF455501@greyhawk.tonecommander.com>

Hi Mike,

 

Thanks for your help.

 

Manually loading mod_openzap fails.  It looks like there is something wrong
with my udev rules not changing permission of the /dev/wanpipe files.

 

freeswitch at internal> load mod_openzap

-ERR [module load file routine returned an error]

 

2010-02-26 12:24:26.807413 [DEBUG] zap_config.c:56 Configuration file is
/opt/teoswitch/conf/modules.conf.

freeswitch at internal> 2010-02-26 12:24:26.807413 [NOTICE] zap_io.c:2778
Modules configured: 1

2010-02-26 12:24:26.807413 [DEBUG] zap_config.c:56 Configuration file is
/opt/teoswitch/conf/openzap.conf.

2010-02-26 12:24:26.807413 [DEBUG] zap_io.c:2362 found config for span

2010-02-26 12:24:26.807413 [INFO] zap_io.c:2579 Loading IO from
/opt/teoswitch/mod/ozmod_wanpipe.so [wanpipe]

2010-02-26 12:24:26.807413 [DEBUG] zap_config.c:56 Configuration file is
/opt/teoswitch/conf/wanpipe.conf.

2010-02-26 12:24:26.807413 [INFO] zap_io.c:2379 auto-loaded 'wanpipe'

2010-02-26 12:24:26.807413 [DEBUG] zap_io.c:2400 created span 1 (smg_prid)
of type wanpipe

2010-02-26 12:24:26.807413 [DEBUG] zap_io.c:2413 span 1 [name]=[smg_prid]

2010-02-26 12:24:26.807413 [DEBUG] zap_io.c:2413 span 1 [trunk_type]=[t1]

2010-02-26 12:24:26.817864 [DEBUG] zap_io.c:2417 setting trunk type to 'T1'

2010-02-26 12:24:26.817864 [DEBUG] zap_io.c:2413 span 1 [b-channel]=[1:1-23]

2010-02-26 12:24:26.817864 [ERR] ozmod_wanpipe.c:233 Failed to open wanpipe
device span 1 channel 1

2010-02-26 12:24:26.817864 [ERR] ozmod_wanpipe.c:233 Failed to open wanpipe
device span 1 channel 2

2010-02-26 12:24:26.817864 [ERR] ozmod_wanpipe.c:233 Failed to open wanpipe
device span 1 channel 3

2010-02-26 12:24:26.817864 [ERR] ozmod_wanpipe.c:233 Failed to open wanpipe
device span 1 channel 4

2010-02-26 12:24:26.817864 [ERR] ozmod_wanpipe.c:233 Failed to open wanpipe
device span 1 channel 5

2010-02-26 12:24:26.817864 [ERR] ozmod_wanpipe.c:233 Failed to open wanpipe
device span 1 channel 6

2010-02-26 12:24:26.817864 [ERR] ozmod_wanpipe.c:233 Failed to open wanpipe
device span 1 channel 7

 

 

Here is what /dev/wan* looks like after udev changes:

[root at roberth-c53 bin]# ls -l /dev/wan*

crw------- 1 root root 242,    0 Feb 26 11:24 /dev/wanec

crw------- 1 root root 241, 2080 Feb 26 11:24 /dev/wanpipe

crw------- 1 root root 241,    1 Feb 26 11:24 /dev/wanpipe1_if1

crw------- 1 root root 241,   10 Feb 26 11:24 /dev/wanpipe1_if10

crw------- 1 root root 241,   11 Feb 26 11:24 /dev/wanpipe1_if11

crw------- 1 root root 241,   12 Feb 26 11:24 /dev/wanpipe1_if12

 

CUT A FEW LINES

crw------- 1 root root 241,   34 Feb 26 11:24 /dev/wanpipe2_if2

crw------- 1 root root 241,   35 Feb 26 11:24 /dev/wanpipe2_if3

crw------- 1 root root 241,   36 Feb 26 11:24 /dev/wanpipe2_if4

crw------- 1 root root 241, 2112 Feb 26 11:24 /dev/wanpipe_ctrl

crw------- 1 root root 241, 2144 Feb 26 11:24 /dev/wanpipe_logger

crw------- 1 root root 241, 2368 Feb 26 11:24 /dev/wanpipe_timer0

 

 

I made and installed the wanpipe driver as root, is that part of the
problem?

 

Thanks again,

Robert

 

 

  _____  

From: Michael Collins [mailto:msc at freeswitch.org] 
Sent: Friday, February 26, 2010 11:28 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Cannot make calls through PRI via
wanpiperunning as daemon

 

 

On Thu, Feb 25, 2010 at 5:16 PM, Robert Hadley <robert.hadley at teotech.com>
wrote:

When running Freeswitch as service called teoswitch as user teoswitch I
cannot make calls through the Sangoma PRI or analog cards using wanpipe
driver.  I have added a file called 30-wanpipe.rules to /etc/udev/rules.d
and rebooted the server.

 

cat 30-wanpipe.rules

# /etc/udev/rules.d/30-wanpipe.rules

SUBSYSTEM=="wptdm", OWNER="teoswitch", GROUP="teoswitch", MODE="0660"

SUBSYSTEM=="wanpipe", OWNER="teoswitch", GROUP="teoswitch", MODE="0660"

I also tried adding:

SUBSYSTEM=="wanec", OWNER="teoswitch", GROUP="teoswitch", MODE="0660"

 

Freeswitch log:

Dialplan: sofia/internal/5410 at 192.168.72.45:5060 parsing
[default->SangomaPRI] continue=false

Dialplan: sofia/internal/5410 at 192.168.72.45:5060 Regex (PASS) [SangomaPRI]
destination_number(93491045) =~ /^9(\d+)$/ break=on-false

Dialplan: sofia/internal/5410 at 192.168.72.45:5060 Action
set(effective_caller_id_number=425740${caller_id_number})

Dialplan: sofia/internal/5410 at 192.168.72.45:5060 Action
bridge(openzap/smg_prid/a/3491045 at g1)

2010-02-25 16:51:11.328635 [DEBUG] switch_core_state_machine.c:122
(sofia/internal/5410 at 192.168.72.45:5060) State Change CS_ROUTING ->
CS_EXECUTE

2010-02-25 16:51:11.328635 [DEBUG] switch_core_session.c:1019 Send signal
sofia/internal/5410 at 192.168.72.45:5060 [BREAK]

2010-02-25 16:51:11.328635 [DEBUG] switch_core_state_machine.c:341
(sofia/internal/5410 at 192.168.72.45:5060) State ROUTING going to sleep

2010-02-25 16:51:11.328635 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/5410 at 192.168.72.45:5060) Running State Change CS_EXECUTE

2010-02-25 16:51:11.328635 [DEBUG] switch_core_state_machine.c:348
(sofia/internal/5410 at 192.168.72.45:5060) State EXECUTE

2010-02-25 16:51:11.328635 [DEBUG] mod_sofia.c:181
sofia/internal/5410 at 192.168.72.45:5060 SOFIA EXECUTE

2010-02-25 16:51:11.328635 [DEBUG] switch_core_state_machine.c:159
sofia/internal/5410 at 192.168.72.45:5060 Standard EXECUTE

EXECUTE sofia/internal/5410 at 192.168.72.45:5060
set(effective_caller_id_number=4257405410)

2010-02-25 16:51:11.328635 [DEBUG] mod_dptools.c:811
sofia/internal/5410 at 192.168.72.45:5060 SET
[effective_caller_id_number]=[4257405410]

EXECUTE sofia/internal/5410 at 192.168.72.45:5060
bridge(openzap/smg_prid/a/3491045 at g1)

2010-02-25 16:51:11.339637 [ERR] switch_core_session.c:357 Could not locate
channel type openzap

2010-02-25 16:51:11.339637 [ERR] switch_ivr_originate.c:2411 Cannot create
outgoing channel of type [openzap] cause: [CHAN_NOT_IMPLEMENTED]

2010-02-25 16:51:11.339637 [DEBUG] switch_ivr_originate.c:3209 Originate
Resulted in Error Cause: 66 [CHAN_NOT_IMPLEMENTED]

2010-02-25 16:51:11.339637 [INFO] mod_dptools.c:2346 Originate Failed.
Cause: CHAN_NOT_IMPLEMENTED


CHAN_NOT_IMPLEMENTED implies that OpenZAP did not load. Capture the output
of "load mod_openzap" and look for the reason that it is failing to load.
-MC

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100226/aba32892/attachment-0001.html 

From msc at freeswitch.org  Fri Feb 26 12:45:22 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 26 Feb 2010 12:45:22 -0800
Subject: [Freeswitch-users] Phrases - Can't find macro
In-Reply-To: <28f27f5d1002260910h7b3ce773l28294e7354078b05@mail.gmail.com>
References: <28f27f5d1002260910h7b3ce773l28294e7354078b05@mail.gmail.com>
Message-ID: <87f2f3b91002261245l176c49f6j769bd673be0324fe@mail.gmail.com>

You might try this suggestion:
Create a new file for your custom macros:
/conf/lang/en/demo/custom-phrases.xml
<include>
  <macro name="confwelcome">
    <input pattern="(.*)">
      <action function="play-file" data="vm/vm-youhave.wav"/>
    </input>
  </macro>
</include>

Now you have a single place to put all of your custom macros. Be sure to
reloadxml!
-MC

On Fri, Feb 26, 2010 at 9:10 AM, Maciej Bylica <mbsip at gazeta.pl> wrote:

> Hello,
>
> I am playing around with Phrases to use them with conference application.
> But i've encountered rudimentary problem of how to use newly added macro.
>
> What I already did is (according to wiki Speech Phrase Management)
> http://wiki.freeswitch.org/wiki/Speech_Phrase_Management
> - confirmed that there is "mod_say_en" loaded (<load module="mod_say_en"/>)
> - confirmed that there are proper .wav files
> - modified onf/lang/en/en.xml file:
>  <section name="phrases" description="Speech Phrase Management">
>   <macros>
>     <language name="en" sound_path="$${sounds_dir}/en/us/callie"
> tts_engine="cepstral" tts_voice="callie">
>       <macro name="confwelcome">
>         <input pattern="(.*)">
>           <match>
>              <action function="play-file" data="vm-youhave.wav"/>
> <!-- just to test the phrase-->
>           </match>
>     </language>
>   </macros>
>  </section>
> - modified a part of dialplan:
>    <extension name="conference_48112223344">
>      <condition field="destination_number" expression="^(\d{11})$">
>         <action application="phrase" data="confwelcome"/>
>      </condition>
>     </extension>
>
> I have following outcome:
> 2010-02-26 19:57:09.487245 [DEBUG] switch_core_session.c:638 Send
> signal sofia/internal/1000 at 217.153.192.36 [BREAK]
> EXECUTE sofia/internal/1000 at 217.153.192.36 phrase(confwelcome)
> 2010-02-26 19:57:09.487245 [DEBUG] mod_dptools.c:1850 Execute confwelcome()
> lang
> 2010-02-26 19:57:09.487245 [DEBUG] switch_ivr_play_say.c:118 No
> language specified - Using [en]
> 2010-02-26 19:57:09.496322 [ERR] switch_ivr_play_say.c:202 Can't find
> macro confwelcome.
> 2010-02-26 19:57:09.496322 [WARNING] switch_ivr_play_say.c:368 Macro
> [confwelcome] did not match any patterns
>
>
>
> Strange is that if i use a wiki example, it works. To be more precise:
> - conf/lang/en/en.xml file was overwritten with an example macros
> (directly from aforementioned wiki).
> - dialplan was modified:
>    <extension name="conference_48112223344">
>      <condition field="destination_number" expression="^(\d{11})$">
>         <action application="phrase" data="msgcount"/>
>      </condition>
>     </extension>
>
>
> Am i doing something wrong?
>
> Thx,
> Maciej.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100226/0143ba00/attachment.html 

From anthony.minessale at gmail.com  Fri Feb 26 12:55:32 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 26 Feb 2010 14:55:32 -0600
Subject: [Freeswitch-users] 415 Unsupported Media Handling
In-Reply-To: <ADCCADE853D6427CBC7CD978E7E0A0ED@greyhawk.tonecommander.com>
References: <AE66B63BAF424BBCBE585E244E0C42B4@greyhawk.tonecommander.com>
	<835A36B2-1C04-42B0-86C1-36528E93589D@freeswitch.org>
	<ADCCADE853D6427CBC7CD978E7E0A0ED@greyhawk.tonecommander.com>
Message-ID: <191c3a031002261255x7e037e6ftc119eab0ed21b69b@mail.gmail.com>

It says SHOULD, not MUST right?

The message passing in FS is abstracted and protocol agnostic and we are a
b2bua not a proxy in terms of SIP.
You are sending a message to 1 UA on FS who is accepting the message and
delivering it to the core who is happy to receive messages in any format.
Then it's routed back out another sip dialog where it's rejected.  It's too
late to go tell the sender that is unacceptable because we already happily
accepted it (messages are not always passed out to other phones they can
easily be directed at other internal resources).  We don't know what the
content-type means as we are a neutral party in the whole thing so there is
not much else we can do but violate this scope issue and break out of our
role as a neutral party and translate it to plain text and try again which
is not very elegant.

If FS was a proxy software, like openser and friends, we would be passing
the data between UA in the way you expect but we are not a proxy.


Based on the frequency and specific nature of your constant inquiries, and
the likelihood that you are offering commercial VoIP services to customers.
I suggest you contact us at consulting at freeswitch.org to investigate
commercial support options for FreeSWITCH.  Even then, I am not sure I could
help you besides maybe a param to convert all text/html messages to plain
text or some other sad hack.




On Fri, Feb 26, 2010 at 1:43 PM, Jerry Richards
<jerry.richards at teotech.com>wrote:

> I'm not sure I follow your comment.  The first device prefers text/html so
> that's what it normally sets in the initial MESSAGE.  Devices that support
> text/html will not generate this 415 error reply.  It's only devices that
> don't support it that would send the 415 reply, so the issue is that the
> 415
> is not getting passed back to the originator.
>
> Best Regards,
> Jerry
>
>
> -----Original Message-----
> From: Brian West [mailto:brian at freeswitch.org]
> Sent: Friday, February 26, 2010 11:01 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] 415 Unsupported Media Handling
>
> Its really clear here you'll need to say text/plain in the content type
> their accept header says they only take text/plain.
>
> /b
>
> On Feb 26, 2010, at 12:57 PM, Jerry Richards wrote:
>
> >
> > I have two types of devices, one supports text/html MESSAGE content
> > and one that only supports text/plain MESSAGE content.  When I send an
> > IM from the first to the second, the second replies with 415
> > Unsupported Media Type (as shown below).  Section 8.1.3.5 of RFC 3261
> > says the sender should retry using the media type acceptable to the
> receiver (in this case: plain/text).
> >
> > The problem I have is that Freeswitch doesn't pass the error back to
> > the sender (nor does it retry itself using plain/text).  So the IM is
> lost.
> > Does anyone see the reason why the error is not being handled correctly?
> >
> >
> > ----------------------------------------------------------------------
> > -- send 668 bytes to udp/[192.168.72.141]:5062 at 18:43:24.720446:
> >
> ------------------------------------------------------------------------
> >   MESSAGE sip:5398 at 192.168.72.141:5062 SIP/2.0
> >   Via: SIP/2.0/UDP 192.168.72.141;rport;branch=z9hG4bKXg6SNUcQa5Kre
> >   Max-Forwards: 70
> >   From: "5382 on 141" <sip:5382 at 192.168.72.141<sip%3A5382 at 192.168.72.141>
> >;tag=66661130
> >   To: "5398" <sip:5398 at 192.168.72.141:5062>
> >   Call-ID: a9891a15-9da9-122d-b3ad-003048d7e9f0
> >   CSeq: 127444135 MESSAGE
> >   Contact: <sip:mod_sofia at 192.168.72.141:5060>
> >   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-32M
> >   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
> >   Supported: timer, precondition, path, replaces
> >   Content-Type: text/html
> >   Content-Length: 63
> >
> >   <font size="2" face="Arial">hello this is Jerry from Teo</font>
> >
> > ----------------------------------------------------------------------
> > -- recv 459 bytes from udp/[192.168.72.141]:5062 at 18:43:24.728390:
> >
> ------------------------------------------------------------------------
> >   SIP/2.0 415 Unsupported media type
> >   Via: SIP/2.0/UDP
> > 192.168.72.141;rport=5060;branch=z9hG4bKXg6SNUcQa5Kre;received=192.168
> > .72.14
> > 1
> >   From: "5382 on 141" <sip:5382 at 192.168.72.141<sip%3A5382 at 192.168.72.141>
> >;tag=66661130
> >   To: "5398" <sip:5398 at 192.168.72.141:5062>
> >   Call-ID: a9891a15-9da9-122d-b3ad-003048d7e9f0
> >   CSeq: 127444135 MESSAGE
> >   Date: Fri, 26 Feb 2010 18:43:24 GMT
> >   User-Agent: MobilityGateway-2.0.34078
> >   Server: MobilityGateway-2.0.34078
> >   Accept: text/plain
> >   Content-Length: 0
> >
> > Best Regards,
> > Jerry
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> > rs
> > http://www.freeswitch.org
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100226/9ae01931/attachment.html 

From anthony.minessale at gmail.com  Fri Feb 26 13:01:20 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 26 Feb 2010 15:01:20 -0600
Subject: [Freeswitch-users] Cannot make calls through PRI via
	wanpiperunning as daemon
In-Reply-To: <D98570990911498788224B48CF455501@greyhawk.tonecommander.com>
References: <6FCED74CF54F466AA25CBDA7DFF42198@greyhawk.tonecommander.com>
	<87f2f3b91002261128n17ea78acx160fcfdb04532004@mail.gmail.com>
	<D98570990911498788224B48CF455501@greyhawk.tonecommander.com>
Message-ID: <191c3a031002261301m53b47e84hbcf87dbfb29a0be9@mail.gmail.com>

yes you will need to give your user access to a group who can read and write
/dev


On Fri, Feb 26, 2010 at 2:36 PM, Robert Hadley <robert.hadley at teotech.com>wrote:

>  Hi Mike,
>
>
>
> Thanks for your help.
>
>
>
> Manually loading mod_openzap fails.  It looks like there is something wrong
> with my udev rules not changing permission of the /dev/wanpipe files.
>
>
>
> freeswitch at internal> load mod_openzap
>
> -ERR [module load file routine returned an error]
>
>
>
> 2010-02-26 12:24:26.807413 [DEBUG] zap_config.c:56 Configuration file is
> /opt/teoswitch/conf/modules.conf.
>
> freeswitch at internal> 2010-02-26 12:24:26.807413 [NOTICE] zap_io.c:2778
> Modules configured: 1
>
> 2010-02-26 12:24:26.807413 [DEBUG] zap_config.c:56 Configuration file is
> /opt/teoswitch/conf/openzap.conf.
>
> 2010-02-26 12:24:26.807413 [DEBUG] zap_io.c:2362 found config for span
>
> 2010-02-26 12:24:26.807413 [INFO] zap_io.c:2579 Loading IO from
> /opt/teoswitch/mod/ozmod_wanpipe.so [wanpipe]
>
> 2010-02-26 12:24:26.807413 [DEBUG] zap_config.c:56 Configuration file is
> /opt/teoswitch/conf/wanpipe.conf.
>
> 2010-02-26 12:24:26.807413 [INFO] zap_io.c:2379 auto-loaded 'wanpipe'
>
> 2010-02-26 12:24:26.807413 [DEBUG] zap_io.c:2400 created span 1 (smg_prid)
> of type wanpipe
>
> 2010-02-26 12:24:26.807413 [DEBUG] zap_io.c:2413 span 1 [name]=[smg_prid]
>
> 2010-02-26 12:24:26.807413 [DEBUG] zap_io.c:2413 span 1 [trunk_type]=[t1]
>
> 2010-02-26 12:24:26.817864 [DEBUG] zap_io.c:2417 setting trunk type to 'T1'
>
> 2010-02-26 12:24:26.817864 [DEBUG] zap_io.c:2413 span 1
> [b-channel]=[1:1-23]
>
> 2010-02-26 12:24:26.817864 [ERR] ozmod_wanpipe.c:233 Failed to open wanpipe
> device span 1 channel 1
>
> 2010-02-26 12:24:26.817864 [ERR] ozmod_wanpipe.c:233 Failed to open wanpipe
> device span 1 channel 2
>
> 2010-02-26 12:24:26.817864 [ERR] ozmod_wanpipe.c:233 Failed to open wanpipe
> device span 1 channel 3
>
> 2010-02-26 12:24:26.817864 [ERR] ozmod_wanpipe.c:233 Failed to open wanpipe
> device span 1 channel 4
>
> 2010-02-26 12:24:26.817864 [ERR] ozmod_wanpipe.c:233 Failed to open wanpipe
> device span 1 channel 5
>
> 2010-02-26 12:24:26.817864 [ERR] ozmod_wanpipe.c:233 Failed to open wanpipe
> device span 1 channel 6
>
> 2010-02-26 12:24:26.817864 [ERR] ozmod_wanpipe.c:233 Failed to open wanpipe
> device span 1 channel 7
>
>
>
>
>
> Here is what /dev/wan* looks like after udev changes:
>
> [root at roberth-c53 bin]# ls -l /dev/wan*
>
> crw------- 1 root root 242,    0 Feb 26 11:24 /dev/wanec
>
> crw------- 1 root root 241, 2080 Feb 26 11:24 /dev/wanpipe
>
> crw------- 1 root root 241,    1 Feb 26 11:24 /dev/wanpipe1_if1
>
> crw------- 1 root root 241,   10 Feb 26 11:24 /dev/wanpipe1_if10
>
> crw------- 1 root root 241,   11 Feb 26 11:24 /dev/wanpipe1_if11
>
> crw------- 1 root root 241,   12 Feb 26 11:24 /dev/wanpipe1_if12
>
>
>
> CUT A FEW LINES
>
> crw------- 1 root root 241,   34 Feb 26 11:24 /dev/wanpipe2_if2
>
> crw------- 1 root root 241,   35 Feb 26 11:24 /dev/wanpipe2_if3
>
> crw------- 1 root root 241,   36 Feb 26 11:24 /dev/wanpipe2_if4
>
> crw------- 1 root root 241, 2112 Feb 26 11:24 /dev/wanpipe_ctrl
>
> crw------- 1 root root 241, 2144 Feb 26 11:24 /dev/wanpipe_logger
>
> crw------- 1 root root 241, 2368 Feb 26 11:24 /dev/wanpipe_timer0
>
>
>
>
>
> I made and installed the wanpipe driver as root, is that part of the
> problem?
>
>
>
> Thanks again,
>
> Robert
>
>
>
>
>  ------------------------------
>
> *From:* Michael Collins [mailto:msc at freeswitch.org]
> *Sent:* Friday, February 26, 2010 11:28 AM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Cannot make calls through PRI via
> wanpiperunning as daemon
>
>
>
>
>
> On Thu, Feb 25, 2010 at 5:16 PM, Robert Hadley <robert.hadley at teotech.com>
> wrote:
>
> When running Freeswitch as service called teoswitch as user teoswitch I
> cannot make calls through the Sangoma PRI or analog cards using wanpipe
> driver.  I have added a file called 30-wanpipe.rules to /etc/udev/rules.d
> and rebooted the server.
>
>
>
> cat 30-wanpipe.rules
>
> # /etc/udev/rules.d/30-wanpipe.rules
>
> SUBSYSTEM=="wptdm", OWNER="teoswitch", GROUP="teoswitch", MODE="0660"
>
> SUBSYSTEM=="wanpipe", OWNER="teoswitch", GROUP="teoswitch", MODE="0660"
>
> I also tried adding:
>
> SUBSYSTEM=="wanec", OWNER="teoswitch", GROUP="teoswitch", MODE="0660"
>
>
>
> Freeswitch log:
>
> Dialplan: sofia/internal/5410 at 192.168.72.45:5060 parsing
> [default->SangomaPRI] continue=false
>
> Dialplan: sofia/internal/5410 at 192.168.72.45:5060 Regex (PASS) [SangomaPRI]
> destination_number(93491045) =~ /^9(\d+)$/ break=on-false
>
> Dialplan: sofia/internal/5410 at 192.168.72.45:5060 Action
> set(effective_caller_id_number=425740${caller_id_number})
>
> Dialplan: sofia/internal/5410 at 192.168.72.45:5060 Action
> bridge(openzap/smg_prid/a/3491045 at g1)
>
> 2010-02-25 16:51:11.328635 [DEBUG] switch_core_state_machine.c:122
> (sofia/internal/5410 at 192.168.72.45:5060) State Change CS_ROUTING ->
> CS_EXECUTE
>
> 2010-02-25 16:51:11.328635 [DEBUG] switch_core_session.c:1019 Send signal
> sofia/internal/5410 at 192.168.72.45:5060 [BREAK]
>
> 2010-02-25 16:51:11.328635 [DEBUG] switch_core_state_machine.c:341
> (sofia/internal/5410 at 192.168.72.45:5060) State ROUTING going to sleep
>
> 2010-02-25 16:51:11.328635 [DEBUG] switch_core_state_machine.c:314
> (sofia/internal/5410 at 192.168.72.45:5060) Running State Change CS_EXECUTE
>
> 2010-02-25 16:51:11.328635 [DEBUG] switch_core_state_machine.c:348
> (sofia/internal/5410 at 192.168.72.45:5060) State EXECUTE
>
> 2010-02-25 16:51:11.328635 [DEBUG] mod_sofia.c:181 sofia/internal/
> 5410 at 192.168.72.45:5060 SOFIA EXECUTE
>
> 2010-02-25 16:51:11.328635 [DEBUG] switch_core_state_machine.c:159
> sofia/internal/5410 at 192.168.72.45:5060 Standard EXECUTE
>
> EXECUTE sofia/internal/5410 at 192.168.72.45:5060set(effective_caller_id_number=4257405410)
>
> 2010-02-25 16:51:11.328635 [DEBUG] mod_dptools.c:811 sofia/internal/
> 5410 at 192.168.72.45:5060 SET [effective_caller_id_number]=[4257405410]
>
> EXECUTE sofia/internal/5410 at 192.168.72.45:5060bridge(openzap/smg_prid/a/3491045 at g1
> )
>
> 2010-02-25 16:51:11.339637 [ERR] switch_core_session.c:357 Could not locate
> channel type openzap
>
> 2010-02-25 16:51:11.339637 [ERR] switch_ivr_originate.c:2411 Cannot create
> outgoing channel of type [openzap] cause: [CHAN_NOT_IMPLEMENTED]
>
> 2010-02-25 16:51:11.339637 [DEBUG] switch_ivr_originate.c:3209 Originate
> Resulted in Error Cause: 66 [CHAN_NOT_IMPLEMENTED]
>
> 2010-02-25 16:51:11.339637 [INFO] mod_dptools.c:2346 Originate Failed.
> Cause: CHAN_NOT_IMPLEMENTED
>
>
> CHAN_NOT_IMPLEMENTED implies that OpenZAP did not load. Capture the output
> of "load mod_openzap" and look for the reason that it is failing to load.
> -MC
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100226/8a286def/attachment-0001.html 

From dave at 3c.co.uk  Fri Feb 26 13:59:12 2010
From: dave at 3c.co.uk (David Knell)
Date: Fri, 26 Feb 2010 14:59:12 -0700
Subject: [Freeswitch-users] ASR Apps
References: <AC3A71D6-4EAB-4261-B14A-2869C0C05C91@freeswitch.org><9dc4a1671002251855y586218f6h2c281b4a5acb664e@mail.gmail.com>
	<16DBCD58-A962-4121-9899-F2BB56F13554@freeswitch.org>
Message-ID: <8AE0BC1F0D764475ADF09B938F15E432@DELL9>

Hi Brian,

Here's a starting point for someone wanting to do voice dial from a Google address book:
http://www.softivr.com/wiki/index.php/Voice_dial

Cheers --

Dave
  ----- Original Message ----- 
  From: Brian West 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Thursday, February 25, 2010 8:02 PM
  Subject: Re: [Freeswitch-users] ASR Apps


  I'm looking for someone to build some really nice apps like dial by name speech apps or other such apps or frameworks using ASR and possibly lua or js.  Anyone wanna do something.


  /b


  On Feb 25, 2010, at 8:55 PM, EdPimentl wrote:


    Hello Bryon,

    We looking to create a Twilio like service using FreeSwitch.

    Sincerely,
    -E
    http://vCardCloud.com
    GV:  678.685.9858
    EdPimentl: Skype




------------------------------------------------------------------------------


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100226/74eb7da5/attachment.html 

From brian at microcomaustralia.com.au  Fri Feb 26 16:23:05 2010
From: brian at microcomaustralia.com.au (Brian May)
Date: Sat, 27 Feb 2010 11:23:05 +1100
Subject: [Freeswitch-users] end call detect on FXO port
Message-ID: <3c5cf5261002261623tca4e78em6800859844465d35@mail.gmail.com>

Hello,

I need to get Freeswitch to detect when the caller has hang up, so it
will hang up the ilne also. Especially important for when the caller
has left a message, although ideally it should work for all calls.

With Asterisk this required the use of automatically detecting the
busy signal, at the driver level. How can I do something similar with
Freeswitch?

Some web pages suggest I use this:

<action application="tone_detect" data="busy 425 r 0 hangup
'normal_clearing' 3"/>

Unfortunately this has the side affect that it answers the call, I
don't want to change the answer behaviour,
only the hang up behaviour.

2010-02-27 14:27:52.655081 [DEBUG] switch_core_session.c:1728
Application tone_detect Requires media! pre_answering channel
OpenZAP/3:1/0397551926

I also see this message, but so far no solution:

http://lists.freeswitch.org/pipermail/freeswitch-users/2008-July/004567.html

How do I do this?

Thanks
-- 
Brian May <brian at microcomaustralia.com.au>


From brian at freeswitch.org  Fri Feb 26 16:30:53 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 26 Feb 2010 18:30:53 -0600
Subject: [Freeswitch-users] end call detect on FXO port
In-Reply-To: <3c5cf5261002261623tca4e78em6800859844465d35@mail.gmail.com>
References: <3c5cf5261002261623tca4e78em6800859844465d35@mail.gmail.com>
Message-ID: <11E1DA2C-0A14-4421-8E6A-994346BE941D@freeswitch.org>

What are you using for your PSTN interface?

/b

On Feb 26, 2010, at 6:23 PM, Brian May wrote:

> Hello,
> 
> I need to get Freeswitch to detect when the caller has hang up, so it
> will hang up the ilne also. Especially important for when the caller
> has left a message, although ideally it should work for all calls.
> 
> With Asterisk this required the use of automatically detecting the
> busy signal, at the driver level. How can I do something similar with
> Freeswitch?
> 
> Some web pages suggest I use this:
> 
> <action application="tone_detect" data="busy 425 r 0 hangup
> 'normal_clearing' 3"/>
> 
> Unfortunately this has the side affect that it answers the call, I
> don't want to change the answer behaviour,
> only the hang up behaviour.
> 
> 2010-02-27 14:27:52.655081 [DEBUG] switch_core_session.c:1728
> Application tone_detect Requires media! pre_answering channel
> OpenZAP/3:1/0397551926
> 
> I also see this message, but so far no solution:
> 
> http://lists.freeswitch.org/pipermail/freeswitch-users/2008-July/004567.html
> 
> How do I do this?
> 
> Thanks
> -- 
> Brian May <brian at microcomaustralia.com.au>
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From brian at microcomaustralia.com.au  Fri Feb 26 18:00:43 2010
From: brian at microcomaustralia.com.au (Brian May)
Date: Sat, 27 Feb 2010 13:00:43 +1100
Subject: [Freeswitch-users] end call detect on FXO port
In-Reply-To: <11E1DA2C-0A14-4421-8E6A-994346BE941D@freeswitch.org>
References: <3c5cf5261002261623tca4e78em6800859844465d35@mail.gmail.com>
	<11E1DA2C-0A14-4421-8E6A-994346BE941D@freeswitch.org>
Message-ID: <3c5cf5261002261800j21248353u46e5279a12ed7b65@mail.gmail.com>

On 27 February 2010 11:30, Brian West <brian at freeswitch.org> wrote:
> What are you using for your PSTN interface?

Not quite sure if this is what you are asking; however I am using a TDM400 card.
-- 
Brian May <brian at microcomaustralia.com.au>


From brian at freeswitch.org  Fri Feb 26 18:09:42 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 26 Feb 2010 20:09:42 -0600
Subject: [Freeswitch-users] end call detect on FXO port
In-Reply-To: <3c5cf5261002261800j21248353u46e5279a12ed7b65@mail.gmail.com>
References: <3c5cf5261002261623tca4e78em6800859844465d35@mail.gmail.com>
	<11E1DA2C-0A14-4421-8E6A-994346BE941D@freeswitch.org>
	<3c5cf5261002261800j21248353u46e5279a12ed7b65@mail.gmail.com>
Message-ID: <DA13207F-7196-4177-8C6E-D2CECA5D3E83@freeswitch.org>

I think the hangup detection should work exactly the same then.

/b

On Feb 26, 2010, at 8:00 PM, Brian May wrote:

> Not quite sure if this is what you are asking; however I am using a TDM400 card.



From brian at microcomaustralia.com.au  Fri Feb 26 18:24:24 2010
From: brian at microcomaustralia.com.au (Brian May)
Date: Sat, 27 Feb 2010 13:24:24 +1100
Subject: [Freeswitch-users] end call detect on FXO port
In-Reply-To: <DA13207F-7196-4177-8C6E-D2CECA5D3E83@freeswitch.org>
References: <3c5cf5261002261623tca4e78em6800859844465d35@mail.gmail.com>
	<11E1DA2C-0A14-4421-8E6A-994346BE941D@freeswitch.org>
	<3c5cf5261002261800j21248353u46e5279a12ed7b65@mail.gmail.com>
	<DA13207F-7196-4177-8C6E-D2CECA5D3E83@freeswitch.org>
Message-ID: <3c5cf5261002261824j68b9f74ckc7f5b5c9b7ae12bb@mail.gmail.com>

On 27 February 2010 13:09, Brian West <brian at freeswitch.org> wrote:
> I think the hangup detection should work exactly the same then.

Exactly the same as what?
-- 
Brian May <brian at microcomaustralia.com.au>


From lakindia89 at gmail.com  Fri Feb 26 20:57:07 2010
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Sat, 27 Feb 2010 10:27:07 +0530
Subject: [Freeswitch-users] smg_prid not bridging the call
In-Reply-To: <c4d05cbe1002260731v29dbc428i1e72d24a220c62eb@mail.gmail.com>
References: <7d79b3931002252155t47c86968q9e451482926d93ac@mail.gmail.com>
	<c4d05cbe1002260731v29dbc428i1e72d24a220c62eb@mail.gmail.com>
Message-ID: <7d79b3931002262057g3b325bctcdbcfb833d4aaf55@mail.gmail.com>

Dear Moy,
Here are the details:

FreeSwitch Log:
http://pastebin.freeswitch.org/12256

 /var/log/sangoma_pri/dchan_<span-no>.log:
 http://pastebin.freeswitch.org/12257

/var/log/sangoma_mgd.log:
http://pastebin.freeswitch.org/12258

smg_pri.conf
http://pastebin.freeswitch.org/12259


On Fri, Feb 26, 2010 at 9:01 PM, Moises Silva <moises.silva at gmail.com>wrote:

> Hello lakshmanan,
>
> Please enable the dchan_log=q931 in /etc/wanpipe/smg_pri.conf and then
> restart it (smg_ctrl restart), then pastebin the logs
>
>  /var/log/sangoma_pri/dchan_<span-no>.log
> /var/log/sangoma_mgd.log
>
> That will contain the Q931 details (if any). Also pastebin your
> smg_pri.conf.
>
> Also enable the FreeSWITCH debug logging (see the FreeSWITCH wiki for
> details about that) and paste them too.
>
> --
> Moises Silva
> Senior Software Engineer
> Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R
> 9T3 Canada
> t. 1 905 474 1990 x 128 | e. moy at sangoma.com
>
> On Fri, Feb 26, 2010 at 12:55 AM, lakshmanan ganapathy <
> lakindia89 at gmail.com> wrote:
>
>> Dear all,
>> I'm having a A102 Sangoma hardware. I configured it with freeswitch.
>> wanrouter status, says both the port as connected.
>> My smg_prid version is
>>
>> Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: ================System
>> restart=============
>> Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: = Sangoma PRI Protocol Stack
>> Daemon       =
>> Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: = Version:
>> 1.54                           =
>> Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: = Date: Feb 15
>> 2010                       =
>> Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: = Wanpipe Release:
>> wanpipe-3.5.8.6       =
>> Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: = Revision:Revision:
>> 15288                =
>> Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid:
>> ===========================================
>>
>> My freeswitch version is 16729.
>> I started freeswitch.
>>
>> oz list
>> +OK
>> span: 1 (smg_prid)
>> type: Sangoma (boost)
>> chan_count: 60
>> dialplan: XML
>> context: default
>> dial_regex:
>> fail_dial_regex:
>> hold_music:
>> analog_options none
>>
>> I originated a call as
>> originate openzap/smg_prid/a/9952248266 &park(), which hits my mobile.
>>
>> But when I issued the following command:
>> originate openzap/smg_prid/a/9952248266
>> &bridge(openzap/smg_prid/a/8122133885)
>> It rings my mobile (9952248266) first, but after that the following error
>> was displayed
>>
>> 2010-02-26 16:20:51.736080 [ERR] switch_ivr_originate.c:2387 Cannot create
>> outgoing channel of type [openzap] cause: [NORMAL_CIRCUIT_CONGESTION]
>> The call got ended in my mobile.
>>
>> Freeswitch log and smg_pri.conf
>>           http://pastebin.freeswitch.org/12248
>> openzap.conf:
>> [span wanpipe smg_prid]
>> name => smg_prid
>> trunk_type =>e1
>> b-channel => 1:1-15
>> b-channel => 1:17-31
>> trunk_type =>e1
>> b-channel => 2:1-15
>> b-channel => 2:17-31
>>
>> openzap.conf.xml:
>> <configuration name="openzap.conf" description="OpenZAP Configuration">
>>   <settings>
>>     <param name="debug" value="0"/>
>>   </settings>
>>   <boost_spans>
>>         <span id="smg_prid">
>>         <param name="dialplan" value="XML"/>
>>         <param name="context" value="default"/>
>>         </span>
>>   </boost_spans>
>> </configuration>
>>
>> Please guide me to setup this one!!.
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100227/9b76b747/attachment-0001.html 

From lakindia89 at gmail.com  Fri Feb 26 21:02:08 2010
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Sat, 27 Feb 2010 10:32:08 +0530
Subject: [Freeswitch-users] smg_prid not bridging the call
In-Reply-To: <7d79b3931002262057g3b325bctcdbcfb833d4aaf55@mail.gmail.com>
References: <7d79b3931002252155t47c86968q9e451482926d93ac@mail.gmail.com>
	<c4d05cbe1002260731v29dbc428i1e72d24a220c62eb@mail.gmail.com>
	<7d79b3931002262057g3b325bctcdbcfb833d4aaf55@mail.gmail.com>
Message-ID: <7d79b3931002262102w79e16ae1nd786600b44dfb758@mail.gmail.com>

In the Dchan log it is saying Invalid Information Elements. That might be a
problem??? But I even don't know why it is saying Invalid Information
Element??
Please guide me!!!


On Sat, Feb 27, 2010 at 10:27 AM, lakshmanan ganapathy <lakindia89 at gmail.com
> wrote:

> Dear Moy,
> Here are the details:
>
> FreeSwitch Log:
> http://pastebin.freeswitch.org/12256
>
>  /var/log/sangoma_pri/dchan_<span-no>.log:
>  http://pastebin.freeswitch.org/12257
>
> /var/log/sangoma_mgd.log:
> http://pastebin.freeswitch.org/12258
>
> smg_pri.conf
> http://pastebin.freeswitch.org/12259
>
>
>
> On Fri, Feb 26, 2010 at 9:01 PM, Moises Silva <moises.silva at gmail.com>wrote:
>
>> Hello lakshmanan,
>>
>> Please enable the dchan_log=q931 in /etc/wanpipe/smg_pri.conf and then
>> restart it (smg_ctrl restart), then pastebin the logs
>>
>>  /var/log/sangoma_pri/dchan_<span-no>.log
>> /var/log/sangoma_mgd.log
>>
>> That will contain the Q931 details (if any). Also pastebin your
>> smg_pri.conf.
>>
>> Also enable the FreeSWITCH debug logging (see the FreeSWITCH wiki for
>> details about that) and paste them too.
>>
>> --
>> Moises Silva
>> Senior Software Engineer
>> Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R
>> 9T3 Canada
>> t. 1 905 474 1990 x 128 | e. moy at sangoma.com
>>
>>  On Fri, Feb 26, 2010 at 12:55 AM, lakshmanan ganapathy <
>> lakindia89 at gmail.com> wrote:
>>
>>>  Dear all,
>>> I'm having a A102 Sangoma hardware. I configured it with freeswitch.
>>> wanrouter status, says both the port as connected.
>>> My smg_prid version is
>>>
>>> Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: ================System
>>> restart=============
>>> Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: = Sangoma PRI Protocol Stack
>>> Daemon       =
>>> Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: = Version:
>>> 1.54                           =
>>> Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: = Date: Feb 15
>>> 2010                       =
>>> Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: = Wanpipe Release:
>>> wanpipe-3.5.8.6       =
>>> Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: = Revision:Revision:
>>> 15288                =
>>> Feb 26 16:08:14 FMS-FreeSwitch
>>> sangoma_prid:
>>> ===========================================
>>>
>>> My freeswitch version is 16729.
>>> I started freeswitch.
>>>
>>> oz list
>>> +OK
>>> span: 1 (smg_prid)
>>> type: Sangoma (boost)
>>> chan_count: 60
>>> dialplan: XML
>>> context: default
>>> dial_regex:
>>> fail_dial_regex:
>>> hold_music:
>>> analog_options none
>>>
>>> I originated a call as
>>> originate openzap/smg_prid/a/9952248266 &park(), which hits my mobile.
>>>
>>> But when I issued the following command:
>>> originate openzap/smg_prid/a/9952248266
>>> &bridge(openzap/smg_prid/a/8122133885)
>>> It rings my mobile (9952248266) first, but after that the following error
>>> was displayed
>>>
>>> 2010-02-26 16:20:51.736080 [ERR] switch_ivr_originate.c:2387 Cannot
>>> create outgoing channel of type [openzap] cause: [NORMAL_CIRCUIT_CONGESTION]
>>> The call got ended in my mobile.
>>>
>>> Freeswitch log and smg_pri.conf
>>>           http://pastebin.freeswitch.org/12248
>>> openzap.conf:
>>> [span wanpipe smg_prid]
>>> name => smg_prid
>>> trunk_type =>e1
>>> b-channel => 1:1-15
>>> b-channel => 1:17-31
>>> trunk_type =>e1
>>> b-channel => 2:1-15
>>> b-channel => 2:17-31
>>>
>>> openzap.conf.xml:
>>> <configuration name="openzap.conf" description="OpenZAP Configuration">
>>>   <settings>
>>>     <param name="debug" value="0"/>
>>>   </settings>
>>>   <boost_spans>
>>>         <span id="smg_prid">
>>>         <param name="dialplan" value="XML"/>
>>>         <param name="context" value="default"/>
>>>         </span>
>>>   </boost_spans>
>>> </configuration>
>>>
>>> Please guide me to setup this one!!.
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100227/03b7567c/attachment.html 

From lakindia89 at gmail.com  Fri Feb 26 21:13:45 2010
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Sat, 27 Feb 2010 10:43:45 +0530
Subject: [Freeswitch-users] smg_prid not bridging the call
In-Reply-To: <7d79b3931002262102w79e16ae1nd786600b44dfb758@mail.gmail.com>
References: <7d79b3931002252155t47c86968q9e451482926d93ac@mail.gmail.com>
	<c4d05cbe1002260731v29dbc428i1e72d24a220c62eb@mail.gmail.com>
	<7d79b3931002262057g3b325bctcdbcfb833d4aaf55@mail.gmail.com>
	<7d79b3931002262102w79e16ae1nd786600b44dfb758@mail.gmail.com>
Message-ID: <7d79b3931002262113s49495e7cga59715f41fdbb35d@mail.gmail.com>

I think it says Invalid Information Element for the DISPLAY
smg_prid/a/8122133885!!!
correct?? If so, can you please help me to solve this?

On Sat, Feb 27, 2010 at 10:32 AM, lakshmanan ganapathy <lakindia89 at gmail.com
> wrote:

> In the Dchan log it is saying Invalid Information Elements. That might be a
> problem??? But I even don't know why it is saying Invalid Information
> Element??
> Please guide me!!!
>
>
>
> On Sat, Feb 27, 2010 at 10:27 AM, lakshmanan ganapathy <
> lakindia89 at gmail.com> wrote:
>
>> Dear Moy,
>> Here are the details:
>>
>> FreeSwitch Log:
>> http://pastebin.freeswitch.org/12256
>>
>>  /var/log/sangoma_pri/dchan_<span-no>.log:
>>  http://pastebin.freeswitch.org/12257
>>
>> /var/log/sangoma_mgd.log:
>> http://pastebin.freeswitch.org/12258
>>
>> smg_pri.conf
>> http://pastebin.freeswitch.org/12259
>>
>>
>>
>> On Fri, Feb 26, 2010 at 9:01 PM, Moises Silva <moises.silva at gmail.com>wrote:
>>
>>> Hello lakshmanan,
>>>
>>> Please enable the dchan_log=q931 in /etc/wanpipe/smg_pri.conf and then
>>> restart it (smg_ctrl restart), then pastebin the logs
>>>
>>>  /var/log/sangoma_pri/dchan_<span-no>.log
>>> /var/log/sangoma_mgd.log
>>>
>>> That will contain the Q931 details (if any). Also pastebin your
>>> smg_pri.conf.
>>>
>>> Also enable the FreeSWITCH debug logging (see the FreeSWITCH wiki for
>>> details about that) and paste them too.
>>>
>>> --
>>> Moises Silva
>>> Senior Software Engineer
>>> Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R
>>> 9T3 Canada
>>> t. 1 905 474 1990 x 128 | e. moy at sangoma.com
>>>
>>>  On Fri, Feb 26, 2010 at 12:55 AM, lakshmanan ganapathy <
>>> lakindia89 at gmail.com> wrote:
>>>
>>>>  Dear all,
>>>> I'm having a A102 Sangoma hardware. I configured it with freeswitch.
>>>> wanrouter status, says both the port as connected.
>>>> My smg_prid version is
>>>>
>>>> Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: ================System
>>>> restart=============
>>>> Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: = Sangoma PRI Protocol
>>>> Stack Daemon       =
>>>> Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: = Version:
>>>> 1.54                           =
>>>> Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: = Date: Feb 15
>>>> 2010                       =
>>>> Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: = Wanpipe Release:
>>>> wanpipe-3.5.8.6       =
>>>> Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: = Revision:Revision:
>>>> 15288                =
>>>> Feb 26 16:08:14 FMS-FreeSwitch
>>>> sangoma_prid:
>>>> ===========================================
>>>>
>>>> My freeswitch version is 16729.
>>>> I started freeswitch.
>>>>
>>>> oz list
>>>> +OK
>>>> span: 1 (smg_prid)
>>>> type: Sangoma (boost)
>>>> chan_count: 60
>>>> dialplan: XML
>>>> context: default
>>>> dial_regex:
>>>> fail_dial_regex:
>>>> hold_music:
>>>> analog_options none
>>>>
>>>> I originated a call as
>>>> originate openzap/smg_prid/a/9952248266 &park(), which hits my mobile.
>>>>
>>>> But when I issued the following command:
>>>> originate openzap/smg_prid/a/9952248266
>>>> &bridge(openzap/smg_prid/a/8122133885)
>>>> It rings my mobile (9952248266) first, but after that the following
>>>> error was displayed
>>>>
>>>> 2010-02-26 16:20:51.736080 [ERR] switch_ivr_originate.c:2387 Cannot
>>>> create outgoing channel of type [openzap] cause: [NORMAL_CIRCUIT_CONGESTION]
>>>> The call got ended in my mobile.
>>>>
>>>> Freeswitch log and smg_pri.conf
>>>>           http://pastebin.freeswitch.org/12248
>>>> openzap.conf:
>>>> [span wanpipe smg_prid]
>>>> name => smg_prid
>>>> trunk_type =>e1
>>>> b-channel => 1:1-15
>>>> b-channel => 1:17-31
>>>> trunk_type =>e1
>>>> b-channel => 2:1-15
>>>> b-channel => 2:17-31
>>>>
>>>> openzap.conf.xml:
>>>> <configuration name="openzap.conf" description="OpenZAP Configuration">
>>>>   <settings>
>>>>     <param name="debug" value="0"/>
>>>>   </settings>
>>>>   <boost_spans>
>>>>         <span id="smg_prid">
>>>>         <param name="dialplan" value="XML"/>
>>>>         <param name="context" value="default"/>
>>>>         </span>
>>>>   </boost_spans>
>>>> </configuration>
>>>>
>>>> Please guide me to setup this one!!.
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100227/e7b3482b/attachment.html 

From kond at nstel.ru  Fri Feb 26 23:15:22 2010
From: kond at nstel.ru (Nikolay Kondratyev)
Date: Sat, 27 Feb 2010 10:15:22 +0300
Subject: [Freeswitch-users] How to tie context to a gateway?
In-Reply-To: <20100216141634.75B3511FC6@mail.nstel.ru>
Message-ID: <20100227071523.10E2A1226D@mail.nstel.ru>

Hi all,

Raising my question again (see below).

I have some idea how to do it, but I'd like to know what experienced FS
users think.

Thanks in advance,

Nikolay.

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Nikolay
Kondratyev
Sent: Tuesday, February 16, 2010 5:17 PM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] How to tie context to a gateway?

 

Hi all,

I have several gateways in the external profile.

Let's say GW1 and GW2. I'd like to process calls from the GW1 in the context
C1 and calls from GW2 in the context C2.

Parameter "context", as far as I understand works for the whole profile, not
for individual gateways in the profile.

How do send calls from GW1 into context C1?

What will be a good practice to do that?

Thanks in advance,

Nikolay.

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100227/8f550720/attachment-0001.html 

From moises.silva at gmail.com  Sat Feb 27 10:35:33 2010
From: moises.silva at gmail.com (Moises Silva)
Date: Sat, 27 Feb 2010 13:35:33 -0500
Subject: [Freeswitch-users] smg_prid not bridging the call
In-Reply-To: <7d79b3931002262113s49495e7cga59715f41fdbb35d@mail.gmail.com>
References: <7d79b3931002252155t47c86968q9e451482926d93ac@mail.gmail.com>
	<c4d05cbe1002260731v29dbc428i1e72d24a220c62eb@mail.gmail.com>
	<7d79b3931002262057g3b325bctcdbcfb833d4aaf55@mail.gmail.com>
	<7d79b3931002262102w79e16ae1nd786600b44dfb758@mail.gmail.com>
	<7d79b3931002262113s49495e7cga59715f41fdbb35d@mail.gmail.com>
Message-ID: <c4d05cbe1002271035g6112b277hdf305b61233b94ad@mail.gmail.com>

I believe the problem FreeSWITCH is setting that as a default callerid name,
which your telco does not like.

Try setting the caller id name and number by yourself as explained in the
"originate" section here http://wiki.freeswitch.org/wiki/Mod_commands
<http://wiki.freeswitch.org/wiki/Mod_commands>

On Sat, Feb 27, 2010 at 12:13 AM, lakshmanan ganapathy <lakindia89 at gmail.com
> wrote:

> I think it says Invalid Information Element for the DISPLAY
> smg_prid/a/8122133885!!!
> correct?? If so, can you please help me to solve this?
>
>
> On Sat, Feb 27, 2010 at 10:32 AM, lakshmanan ganapathy <
> lakindia89 at gmail.com> wrote:
>
>> In the Dchan log it is saying Invalid Information Elements. That might be
>> a problem??? But I even don't know why it is saying Invalid Information
>> Element??
>> Please guide me!!!
>>
>>
>>
>> On Sat, Feb 27, 2010 at 10:27 AM, lakshmanan ganapathy <
>> lakindia89 at gmail.com> wrote:
>>
>>> Dear Moy,
>>> Here are the details:
>>>
>>> FreeSwitch Log:
>>> http://pastebin.freeswitch.org/12256
>>>
>>>  /var/log/sangoma_pri/dchan_<span-no>.log:
>>>  http://pastebin.freeswitch.org/12257
>>>
>>> /var/log/sangoma_mgd.log:
>>> http://pastebin.freeswitch.org/12258
>>>
>>> smg_pri.conf
>>> http://pastebin.freeswitch.org/12259
>>>
>>>
>>>
>>> On Fri, Feb 26, 2010 at 9:01 PM, Moises Silva <moises.silva at gmail.com>wrote:
>>>
>>>> Hello lakshmanan,
>>>>
>>>> Please enable the dchan_log=q931 in /etc/wanpipe/smg_pri.conf and then
>>>> restart it (smg_ctrl restart), then pastebin the logs
>>>>
>>>>  /var/log/sangoma_pri/dchan_<span-no>.log
>>>> /var/log/sangoma_mgd.log
>>>>
>>>> That will contain the Q931 details (if any). Also pastebin your
>>>> smg_pri.conf.
>>>>
>>>> Also enable the FreeSWITCH debug logging (see the FreeSWITCH wiki for
>>>> details about that) and paste them too.
>>>>
>>>> --
>>>> Moises Silva
>>>> Senior Software Engineer
>>>> Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R
>>>> 9T3 Canada
>>>> t. 1 905 474 1990 x 128 | e. moy at sangoma.com
>>>>
>>>>  On Fri, Feb 26, 2010 at 12:55 AM, lakshmanan ganapathy <
>>>> lakindia89 at gmail.com> wrote:
>>>>
>>>>>  Dear all,
>>>>> I'm having a A102 Sangoma hardware. I configured it with freeswitch.
>>>>> wanrouter status, says both the port as connected.
>>>>> My smg_prid version is
>>>>>
>>>>> Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: ================System
>>>>> restart=============
>>>>> Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: = Sangoma PRI Protocol
>>>>> Stack Daemon       =
>>>>> Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: = Version:
>>>>> 1.54                           =
>>>>> Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: = Date: Feb 15
>>>>> 2010                       =
>>>>> Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: = Wanpipe Release:
>>>>> wanpipe-3.5.8.6       =
>>>>> Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: = Revision:Revision:
>>>>> 15288                =
>>>>> Feb 26 16:08:14 FMS-FreeSwitch
>>>>> sangoma_prid:
>>>>> ===========================================
>>>>>
>>>>> My freeswitch version is 16729.
>>>>> I started freeswitch.
>>>>>
>>>>> oz list
>>>>> +OK
>>>>> span: 1 (smg_prid)
>>>>> type: Sangoma (boost)
>>>>> chan_count: 60
>>>>> dialplan: XML
>>>>> context: default
>>>>> dial_regex:
>>>>> fail_dial_regex:
>>>>> hold_music:
>>>>> analog_options none
>>>>>
>>>>> I originated a call as
>>>>> originate openzap/smg_prid/a/9952248266 &park(), which hits my mobile.
>>>>>
>>>>> But when I issued the following command:
>>>>> originate openzap/smg_prid/a/9952248266
>>>>> &bridge(openzap/smg_prid/a/8122133885)
>>>>> It rings my mobile (9952248266) first, but after that the following
>>>>> error was displayed
>>>>>
>>>>> 2010-02-26 16:20:51.736080 [ERR] switch_ivr_originate.c:2387 Cannot
>>>>> create outgoing channel of type [openzap] cause: [NORMAL_CIRCUIT_CONGESTION]
>>>>> The call got ended in my mobile.
>>>>>
>>>>> Freeswitch log and smg_pri.conf
>>>>>           http://pastebin.freeswitch.org/12248
>>>>> openzap.conf:
>>>>> [span wanpipe smg_prid]
>>>>> name => smg_prid
>>>>> trunk_type =>e1
>>>>> b-channel => 1:1-15
>>>>> b-channel => 1:17-31
>>>>> trunk_type =>e1
>>>>> b-channel => 2:1-15
>>>>> b-channel => 2:17-31
>>>>>
>>>>> openzap.conf.xml:
>>>>> <configuration name="openzap.conf" description="OpenZAP Configuration">
>>>>>   <settings>
>>>>>     <param name="debug" value="0"/>
>>>>>   </settings>
>>>>>   <boost_spans>
>>>>>         <span id="smg_prid">
>>>>>         <param name="dialplan" value="XML"/>
>>>>>         <param name="context" value="default"/>
>>>>>         </span>
>>>>>   </boost_spans>
>>>>> </configuration>
>>>>>
>>>>> Please guide me to setup this one!!.
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100227/4d266a48/attachment.html 

From michal.kalinowski at interia.pl  Sat Feb 27 13:10:16 2010
From: michal.kalinowski at interia.pl (michal kalinowski)
Date: Sat, 27 Feb 2010 22:10:16 +0100
Subject: [Freeswitch-users] ivr from mysql
In-Reply-To: <E9EBF591-5DEC-4FF3-B2D8-20DDA54B4FF5@jerris.com>
References: <7c74f5761002150824i41a2ed5did008f36cbb4499a4@mail.gmail.com>
	<87f2f3b91002151002v6bbc3e7fgb67dd6d3e72a915@mail.gmail.com>
	<7c74f5761002151425g512fa685n13c58d8f4b894740@mail.gmail.com>
	<0FD30AAA-C2E2-4228-A09B-DB09F4E26811@avgs.ca>
	<7c74f5761002151525p7cc690ces632d46c6ab5e236c@mail.gmail.com>
	<E9EBF591-5DEC-4FF3-B2D8-20DDA54B4FF5@jerris.com>
Message-ID: <7c74f5761002271310h7e925f7cy25dd69ba60b896ef@mail.gmail.com>

Coming back to this case I create in lua some script with XML ivr.

#!/usr/local/bin/lua

mydialplan = [[
<?xml version="1.0" encoding="UTF-8" standalone="no"?>
<document type="freeswitch/xml">
<include>
  <!-- demo IVR setup -->
  <!-- demo IVR, Main Menu -->
  <menu name="demo_ivr"
      greet-long="phrase:demo_ivr_main_menu"
      greet-short="phrase:demo_ivr_main_menu_short"
      invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"
      exit-sound="voicemail/vm-goodbye.wav"
      confirm-macro=""
      confirm-key=""
      tts-engine="flite"
      tts-voice="rms"
      confirm-attempts="3"
      timeout="10000"
      inter-digit-timeout="2000"
      max-failures="3"
      max-timeouts="3"
      digit-len="4">

    <!-- The following are the definitions for the digits the user dials -->
    <!-- Digit 1 transfer caller to the public FreeSWITCH conference -->
    <entry action="menu-exec-app" digits="1" param="bridge
sofia/$${domain}/888 at conference.freeswitch.org"/>
    <entry action="menu-exec-app" digits="2" param="transfer 9996 XML
default"/>    <!-- FS echo -->
    <entry action="menu-exec-app" digits="3" param="transfer 9999 XML
default"/>    <!-- MOH -->
    <entry action="menu-exec-app" digits="4" param="transfer 9991 XML
default"/>    <!-- ClueCon -->
    <entry action="menu-exec-app" digits="5" param="transfer 1234*256
enum"/>       <!-- Screaming monkeys -->
    <entry action="menu-sub" digits="6" param="demo_ivr_submenu"/>
             <!-- demo sub menu -->
    <!-- Using a regex in the digits tag lets you define a dial
pattern for the caller
         You may define multiple regexes if you need a different
pattern for some reason -->
    <entry action="menu-exec-app" digits="/^(10[01][0-9])$/"
param="transfer $1 XML features"/>
    <entry action="menu-top" digits="9"/>          <!-- Repeat this menu -->
  </menu>

</include>
]]
XML_STRING = mydialplan

in dialplan I have context with this ivr

  <context name="ivr">
<extension name="ivr">
     <condition field="caller_id_number" expression="^[0-9]"/>
     <condition field="destination_number" expression="^(\d{11})$">
        <action application="answer"/>
        <action application="ivr" data="demo_ivr"/>
     </condition>
</extension>
  </context>

in ivr.conf i have this

<configuration name="ivr.conf" description="IVR menus">
  <menus>
        <X-PRE-PROCESS cmd="set" data="application=lua lua.ivr"/>
  </menus>
</configuration>

but for some reasons Freeswitch say "2010-02-27 22:27:48.380342 [ERR]
mod_dptools.c:1247 Unable to find menu"
what I do wrong ?


BR,
Micha?

2010/2/18 Michael Jerris <mike at jerris.com>:
> an example is available here : ? http://svn.freeswitch.org/svn/freeswitch/trunk/conf/ivr_menus/demo_ivr.xml
>
> Mike
>
> On Feb 15, 2010, at 6:25 PM, michal kalinowski wrote:
>> Could you insert several examples here?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From brian at freeswitch.org  Sat Feb 27 13:48:04 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 27 Feb 2010 15:48:04 -0600
Subject: [Freeswitch-users] ivr from mysql
In-Reply-To: <7c74f5761002271310h7e925f7cy25dd69ba60b896ef@mail.gmail.com>
References: <7c74f5761002150824i41a2ed5did008f36cbb4499a4@mail.gmail.com>
	<87f2f3b91002151002v6bbc3e7fgb67dd6d3e72a915@mail.gmail.com>
	<7c74f5761002151425g512fa685n13c58d8f4b894740@mail.gmail.com>
	<0FD30AAA-C2E2-4228-A09B-DB09F4E26811@avgs.ca>
	<7c74f5761002151525p7cc690ces632d46c6ab5e236c@mail.gmail.com>
	<E9EBF591-5DEC-4FF3-B2D8-20DDA54B4FF5@jerris.com>
	<7c74f5761002271310h7e925f7cy25dd69ba60b896ef@mail.gmail.com>
Message-ID: <23CC9D8A-65D5-438F-B117-00FEC087418D@freeswitch.org>

can you point out on the wiki that indicated you are able to dot his?  You're mixing two concepts incorrectly here.

For example you can't do "set" and exec the file to get the contents.. you can however on linux use "exec" instead of set.  But your script needs to print it to stdout.  Your example in your lua script is for the config engine in in lua thats like XML curl... in which case you're not building the full document like you should.  Read thru the XML CURL docs if you want to do "XML_STRING = mydialplan".  

/b


On Feb 27, 2010, at 3:10 PM, michal kalinowski wrote:

> Coming back to this case I create in lua some script with XML ivr.
> 
> #!/usr/local/bin/lua
> 
> mydialplan = [[
> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
> <document type="freeswitch/xml">
> <include>
>  <!-- demo IVR setup -->
>  <!-- demo IVR, Main Menu -->
>  <menu name="demo_ivr"
>      greet-long="phrase:demo_ivr_main_menu"
>      greet-short="phrase:demo_ivr_main_menu_short"
>      invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"
>      exit-sound="voicemail/vm-goodbye.wav"
>      confirm-macro=""
>      confirm-key=""
>      tts-engine="flite"
>      tts-voice="rms"
>      confirm-attempts="3"
>      timeout="10000"
>      inter-digit-timeout="2000"
>      max-failures="3"
>      max-timeouts="3"
>      digit-len="4">
> 
>    <!-- The following are the definitions for the digits the user dials -->
>    <!-- Digit 1 transfer caller to the public FreeSWITCH conference -->
>    <entry action="menu-exec-app" digits="1" param="bridge
> sofia/$${domain}/888 at conference.freeswitch.org"/>
>    <entry action="menu-exec-app" digits="2" param="transfer 9996 XML
> default"/>    <!-- FS echo -->
>    <entry action="menu-exec-app" digits="3" param="transfer 9999 XML
> default"/>    <!-- MOH -->
>    <entry action="menu-exec-app" digits="4" param="transfer 9991 XML
> default"/>    <!-- ClueCon -->
>    <entry action="menu-exec-app" digits="5" param="transfer 1234*256
> enum"/>       <!-- Screaming monkeys -->
>    <entry action="menu-sub" digits="6" param="demo_ivr_submenu"/>
>             <!-- demo sub menu -->
>    <!-- Using a regex in the digits tag lets you define a dial
> pattern for the caller
>         You may define multiple regexes if you need a different
> pattern for some reason -->
>    <entry action="menu-exec-app" digits="/^(10[01][0-9])$/"
> param="transfer $1 XML features"/>
>    <entry action="menu-top" digits="9"/>          <!-- Repeat this menu -->
>  </menu>
> 
> </include>
> ]]
> XML_STRING = mydialplan
> 
> in dialplan I have context with this ivr
> 
>  <context name="ivr">
> <extension name="ivr">
>     <condition field="caller_id_number" expression="^[0-9]"/>
>     <condition field="destination_number" expression="^(\d{11})$">
>        <action application="answer"/>
>        <action application="ivr" data="demo_ivr"/>
>     </condition>
> </extension>
>  </context>
> 
> in ivr.conf i have this
> 
> <configuration name="ivr.conf" description="IVR menus">
>  <menus>
>        <X-PRE-PROCESS cmd="set" data="application=lua lua.ivr"/>
>  </menus>
> </configuration>
> 
> but for some reasons Freeswitch say "2010-02-27 22:27:48.380342 [ERR]
> mod_dptools.c:1247 Unable to find menu"
> what I do wrong ?
> 
> 
> BR,
> Micha?



From mbsip at gazeta.pl  Sat Feb 27 14:35:40 2010
From: mbsip at gazeta.pl (Maciej Bylica)
Date: Sat, 27 Feb 2010 23:35:40 +0100
Subject: [Freeswitch-users] Phrases - Can't find macro
In-Reply-To: <87f2f3b91002261245l176c49f6j769bd673be0324fe@mail.gmail.com>
References: <28f27f5d1002260910h7b3ce773l28294e7354078b05@mail.gmail.com>
	<87f2f3b91002261245l176c49f6j769bd673be0324fe@mail.gmail.com>
Message-ID: <28f27f5d1002271435x20e068b1w6112c5f19ef6b146@mail.gmail.com>

Thx Michael for Your prompt answer.
I did exactly what you had said -- in addition there was a need to use
<match>, but result is possitive.

Thank You,
Maciej

> You might try this suggestion:
> Create a new file for your custom macros:
> /conf/lang/en/demo/custom-phrases.xml
> <include>
> ? <macro name="confwelcome">
> ??? <input pattern="(.*)">
> ????? <action function="play-file" data="vm/vm-youhave.wav"/>
> ??? </input>
> ? </macro>
> </include>
>
> Now you have a single place to put all of your custom macros. Be sure to
> reloadxml!
> -MC


From msc at freeswitch.org  Sat Feb 27 16:45:15 2010
From: msc at freeswitch.org (Michael Collins)
Date: Sat, 27 Feb 2010 16:45:15 -0800
Subject: [Freeswitch-users] Phrases - Can't find macro
In-Reply-To: <28f27f5d1002271435x20e068b1w6112c5f19ef6b146@mail.gmail.com>
References: <28f27f5d1002260910h7b3ce773l28294e7354078b05@mail.gmail.com>
	<87f2f3b91002261245l176c49f6j769bd673be0324fe@mail.gmail.com>
	<28f27f5d1002271435x20e068b1w6112c5f19ef6b146@mail.gmail.com>
Message-ID: <87f2f3b91002271645x54e2dd5eqc3a2169b9eca2e39@mail.gmail.com>

On Sat, Feb 27, 2010 at 2:35 PM, Maciej Bylica <mbsip at gazeta.pl> wrote:

> Thx Michael for Your prompt answer.
> I did exactly what you had said -- in addition there was a need to use
> <match>, but result is possitive.
>

You are quite correct - I left out the <match> and optional <nomatch> nodes.
-MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100227/04bf978f/attachment.html 

From brian at microcomaustralia.com.au  Sat Feb 27 17:11:33 2010
From: brian at microcomaustralia.com.au (Brian May)
Date: Sun, 28 Feb 2010 12:11:33 +1100
Subject: [Freeswitch-users] Possible typo on Openzap.conf wiki page
	specifying fxs/fxo-channels
In-Reply-To: <a689f57fe216d671d7d1ff132c6b636e@thom.fr.eu.org>
References: <2C33D9D6C1444AAC80950CDF8D38A428@greyhawk.tonecommander.com>
	<3c5cf5261002231524n4e9af307y80b3accb1ee4ff26@mail.gmail.com>
	<a689f57fe216d671d7d1ff132c6b636e@thom.fr.eu.org>
Message-ID: <3c5cf5261002271711t59294b4fm9aebfdf5b48a6b31@mail.gmail.com>

On 24 February 2010 20:25, Fran?ois Legal <devel at thom.fr.eu.org> wrote:
> I use sangoma card and the openzap file is generated by the Setup script
> from sangoma driver.
> It seems that the terminology used by zaptel is not used in wanpipe
> configuration.

Yes, that is correct.

> I have an A400 card with an FXO module (providing ports 11 and 12) and an
> FXS module (providing ports 9 and 10)
>
> My openzap.conf is like this :
>
> [span wanpipe FXS]
> name => Analog phone 1
> number => 9000
> fxs-channel => 1:9
> name => Analog phone 2
> number => 9001
> fxs-channel => 1:10
>
> [span wanpipe FXO]
> name => POTS line 1
> number => 1234567890
> fxo-channel => 1:11
> name => POTS line 2
> number => 0987654321
> fxo-channel => 1:12

So ports 9 and 10 are actually FXO ports - extension ports; ports 11
and 12 are FXS ports, or telephone lines. This is what I have been
saying.

Oh, wait, no it isn't. Looks like I was confused. :-(

It matches my config however.

Hopefully this fixed the problems with the wiki:

http://wiki.freeswitch.org/index.php?title=Openzap.conf_Examples&diff=18693&oldid=18491

My guess is that this change is needed also (not absolutely sure here):

http://wiki.freeswitch.org/index.php?title=Openzap.conf_Examples&diff=18694&oldid=18693
-- 
Brian May <brian at microcomaustralia.com.au>


From christian.loeschenkohl at xpirio.com  Sun Feb 28 00:57:48 2010
From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=)
Date: Sun, 28 Feb 2010 09:57:48 +0100
Subject: [Freeswitch-users] conferences lead to high server load
In-Reply-To: <191c3a031002241058y517e63fdte0d929b97a33f9ac@mail.gmail.com>
References: <4B857226.10308@xpirio.com>
	<191c3a031002241058y517e63fdte0d929b97a33f9ac@mail.gmail.com>
Message-ID: <4B8A300C.4060805@xpirio.com>

hello

problem solved with -vm
with this option we now have the usual low load for 50-70 conference users

i think it would be good to explain this in the wiki, how can i get more information
on this to put it in there - "use possibly more vm-friendly timing code" wouldn't be enough :-)
i could create an article about the startup flags in general

br


Anthony Minessale wrote:

> load average has no meaning with FS, you have to look at the CPU usage 
> per CPU and thread.
> Are you experiencing any audio problems or are you just concerned about 
> that load number?
> 
> If you have a box that has trouble with timing it could cost more resources.
> you can always run freeswitch -vm to use an alternate form of timing 
> that may not manifest into the load average.
> 
> 
> 2010/2/24 Christian L?schenkohl <christian.loeschenkohl at xpirio.com 
> <mailto:christian.loeschenkohl at xpirio.com>>
> 
>     hi
> 
>     we do experience a unusual high server load with the latest
>     freeswitch versions.
>     about 50 conference users lead to a server load of over 10 -
>     reproducible by the way.
>     this wans't the case until my latest trunk update.
> 
>     fs version: 16714
>     os: debian lenny x86_64
> 
>     has something substantially changed in mod_conference recently?
> 
>     br
> 
>     --
>     Ing. Christian L?schenkohl
>     Technische Leitung, Forschung & Entwicklung VoIP
> 
>     xpirio
>     Telekommunikation & Service GmbH
>     Lakeside B04
>     9020 Klagenfurt
>     Austria
> 
>     T  +43 (0) 5 77 11 - 1000
>     F  +43 (0) 5 77 11 - 1002
>     E  christian.loeschenkohl at xpirio.com
>     <mailto:christian.loeschenkohl at xpirio.com>
> 
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
> 
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com 
> <mailto:MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com 
> <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org 
> <mailto:sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888 
> <http://iax:guest at conference.freeswitch.org/888>
> googletalk:conf+888 at conference.freeswitch.org 
> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


-- 
Ing. Christian L?schenkohl
Technische Leitung, Forschung & Entwicklung VoIP

xpirio
Telekommunikation & Service GmbH
Lakeside B04
9020 Klagenfurt
Austria

T  +43 (0) 5 77 11 - 1000
F  +43 (0) 5 77 11 - 1002
E  christian.loeschenkohl at xpirio.com


From yehavi.bourvine at gmail.com  Sun Feb 28 01:49:16 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Sun, 28 Feb 2010 11:49:16 +0200
Subject: [Freeswitch-users] Announcement: FreeSWITCH Add Broadsoft SCA
	Support
In-Reply-To: <191c3a031002190908r3922efccqc6d223a5fabd458b@mail.gmail.com>
References: <87f2f3b91001111742k691966cfs82515db77c13d4d4@mail.gmail.com>
	<191c3a031002170557g5d8c9df9xf40d1b74b9a6f19d@mail.gmail.com>
	<d361ca7b1002170700x15630ab2tabc6f3d91ebc3303@mail.gmail.com>
	<ADC8B813-4161-481C-877E-93FC012858B4@freeswitch.org>
	<d361ca7b1002182101p60b4e8f8hf4dc95e1d2251e30@mail.gmail.com>
	<8b1c9cda1002182144m65ccf979w71384a1e7a55e2ea@mail.gmail.com>
	<d361ca7b1002182154o52352c62h8c0ec02fa66e8444@mail.gmail.com>
	<2F87B95C-EA23-4072-B845-3F7AE0617BE3@jerris.com>
	<d361ca7b1002182346k71fd602fh6e6725834a6de528@mail.gmail.com>
	<191c3a031002190908r3922efccqc6d223a5fabd458b@mail.gmail.com>
Message-ID: <d361ca7b1002280149i18e66cbdo2031e7f385f23075@mail.gmail.com>

Hello all,

The problem is solved, at least for my Polycoms.

The solution is to have "domain" variable (in vars.xml) set to the IP
address of the profile that is used for phones registration (the default is
ok if you have only one interface). Define the phone's registration and
proxy servers to that address. You must use IP addresses and not DNS names.

                  Thanks to Anthony for his helpfull tips!  __Yehavi:

 2010/2/19 Anthony Minessale <anthony.minessale at gmail.com>

> go see my comments on that bug note.
> be prepared to give us ssh access and call or irc so we can can see you
> reproducing it.
>
> If you are not on the latest firmware on all the phones, we will not
> continue with this process.
>
>
>
>
> On Fri, Feb 19, 2010 at 1:46 AM, Yehavi Bourvine <
> yehavi.bourvine at gmail.com> wrote:
>
>>  A jira issue has been created: *MODSOFIA-61*<http://jira.freeswitch.org/browse/MODSOFIA-61>
>> .
>>
>>                   Thanks, __Yehavi:
>>
>>  2010/2/19 Michael Jerris <mike at jerris.com>
>>
>>>  If this issue is not already on jira could you please make sure it gets
>>> added?
>>>
>>> Mike
>>>
>>>
>>> On Feb 19, 2010, at 12:54 AM, Yehavi Bourvine <yehavi.bourvine at gmail.com>
>>> wrote:
>>>
>>>     Hello Gabe,
>>>
>>>   As you can see - Brian is actively investigating it, so you can expect
>>> for some fix soon...
>>>
>>>                        Regards, __Yehavi:
>>>
>>> 2010/2/19 Gabriel Kuri < <gkuri at ieee.org>gkuri at ieee.org>
>>>
>>>> >  When a call arrives, both ring; the one that did not answer gets only
>>>> a
>>>> > cancel mesage without any further notification that the extension is
>>>> in use
>>>> > by the other phone.
>>>>
>>>> These are the same exact symptoms I posted about earlier this week,
>>>> with the Cisco SPA-5xx series phones. I still have yet to figure out
>>>> why this is happening, if you find out what's going on, please post
>>>> back the solution, I'd like to know the resolution.
>>>>
>>>> Thanks,
>>>> Gabe
>>>>
>>>>
>>>>
>>>> On Thu, Feb 18, 2010 at 9:01 PM, Yehavi Bourvine
>>>> < <yehavi.bourvine at gmail.com>yehavi.bourvine at gmail.com> wrote:
>>>> > Thanks Brian. It now works better, but not fully (using 16659M).
>>>> >
>>>> > What happens is:
>>>> >
>>>> > When one of the Polycoms seize the line it is ok - the other phone
>>>> gets
>>>> > notification and the extension status is "in use".
>>>> > When one of the Polycom phones initiates a call - all is ok:
>>>> >
>>>> > The other side sees that the extension is in use.
>>>> > When it is put to hold all phones who share this extension see it and
>>>> can
>>>> > pick the call.
>>>> >
>>>>
>>>> >
>>>>  >                          Thanks! __Yehavi:
>>>> >
>>>> > 2010/2/17 Brian West < <brian at freeswitch.org>brian at freeswitch.org>
>>>> >>
>>>> >> Step 1. Enable manage-shared-appearance=true
>>>> >>
>>>> >> Step 2. Now in the phone's config Configure the phone as usually, set
>>>> the
>>>> >> line shared and DO NOT set the third party name.
>>>> >>
>>>> >> Step 3. Reboot
>>>> >>
>>>> >> It should work.
>>>> >>
>>>> >> I wish someone that has this working would write some wiki docs these
>>>> >> threads about it not working are getting rather old when I know for a
>>>> fact
>>>> >> they work fine.
>>>> >>
>>>> >> The gateway info missing is a gateway you have configured getting a
>>>> >> notify.  It has nothing to do with SCA.
>>>> >>
>>>> >> /b
>>>> >>
>>>> >> On Feb 17, 2010, at 9:00 AM, Yehavi Bourvine wrote:
>>>> >>
>>>> >> > .
>>>> >>
>>>> >>
>>>> >> _______________________________________________
>>>> >> FreeSWITCH-users mailing list
>>>> >> <FreeSWITCH-users at lists.freeswitch.org>
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> >> <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> >> <http://www.freeswitch.org/>http://www.freeswitch.org
>>>> >
>>>> > _______________________________________________
>>>> > FreeSWITCH-users mailing list
>>>> > <FreeSWITCH-users at lists.freeswitch.org>
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> > <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> > UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> > <http://www.freeswitch.org/>http://www.freeswitch.org
>>>> >
>>>> >
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> <FreeSWITCH-users at lists.freeswitch.org>
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> <http://www.freeswitch.org/>http://www.freeswitch.org
>>>>
>>>  _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100228/31dbad40/attachment-0001.html 

From mattdfong at gmail.com  Sun Feb 28 03:36:18 2010
From: mattdfong at gmail.com (Matthew Fong)
Date: Sun, 28 Feb 2010 18:36:18 +0700
Subject: [Freeswitch-users] Detecting Energy Levels on a Channel with mod_lua
Message-ID: <4256bf831002280336vb7268e2oe48b0f74bac335f8@mail.gmail.com>

Is there anyway to detect energy levels on a channel that is being
controlled by mod_lua? Please point me in the right direction if there is.
Thanks.

--matt
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100228/4e5c6183/attachment.html 

From yehavi.bourvine at gmail.com  Sun Feb 28 05:14:44 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Sun, 28 Feb 2010 15:14:44 +0200
Subject: [Freeswitch-users] SQL ERR: library routine called out of
	sequence
In-Reply-To: <191c3a031002161230h2b6e6f9wf3f65ddb71f08297@mail.gmail.com>
References: <d361ca7b1002160656w7d616f6dn5e80087ebde94dbe@mail.gmail.com>
	<191c3a031002160705w4050c3cdk2bf2922e9aff5ba3@mail.gmail.com>
	<d361ca7b1002160727ucb6e0c8id62e803757333213@mail.gmail.com>
	<191c3a031002160755q569fe2ednfefb01b6d9cf7f0e@mail.gmail.com>
	<d361ca7b1002161059w1ef9484dr87e69a33aa2b62fc@mail.gmail.com>
	<191c3a031002161110s637a48a2r717ae94557fe9e9a@mail.gmail.com>
	<d361ca7b1002161130k3eb2d88by3100b30a9d29bd51@mail.gmail.com>
	<191c3a031002161230h2b6e6f9wf3f65ddb71f08297@mail.gmail.com>
Message-ID: <d361ca7b1002280514p71003588t8a0daebce5ba9e4d@mail.gmail.com>

Hello,

  I've installed a machine with CentOS-5.4 and the latest FreeSwitch
(16841M). The problem still happens, even if I set sql-in-transactions to
false. It happens also on a very light load.

I would like to remind that we query the SQLite core DB from a LUA script
which is called from the dial plan; might this be the cause of it?

I am willing to test whatever you like, and can do it now as this machine is
not in production yet.

            Thanks! __Yehavi:

 2010/2/16 Anthony Minessale <anthony.minessale at gmail.com>

> Strange, even on abusive testing we have not seen this problem.
>
> please update to latest trunk.
> There was only one change I can think of that may cause your issue and I
> added a patch for it.
> If it persists try setting the sql-in-transactions profile param to false.
>
>
>
>
>
> On Tue, Feb 16, 2010 at 1:30 PM, Yehavi Bourvine <
> yehavi.bourvine at gmail.com> wrote:
>
>>  Most of the queries are ok, only some fail, thus it doesn't look like
>> permission problem. Furthermore, under 1.0.5pre10 it works for months.
>>
>> Might it be thread unsafe function calls? I've found the following while
>> searching the WEB:
>>
>> *According to the MSDN docs, System.Timers.Timer operates in a thread
>> pool.  If that's the case, your code is breaking the "connections cannot be
>> shared across threads" rule for SQLit*
>>
>> Although it quotes MSDN, it might be related to Linux as well.
>>
>>                     Thanks, __Yehavi:
>>
>>  2010/2/16 Anthony Minessale <anthony.minessale at gmail.com>
>>
>>> That sounds about right.
>>>
>>> That error usually has something to do with using db calls on a closed
>>> file or something along those lines.
>>> Maybe you have a permission problem on the directory where the db files
>>> are?
>>>
>>>
>>>
>>> On Tue, Feb 16, 2010 at 12:59 PM, Yehavi Bourvine <
>>> yehavi.bourvine at gmail.com> wrote:
>>>
>>>>  The OS is Fedora-10 (soon to be upgraded to 12).
>>>>
>>>> What I do when I want to test a new version:
>>>>
>>>>    - Download the latest one into a fresh directory
>>>>    - bootstrap.sh, configure and make
>>>>    - stop Freeswitch, delete everything in lib, mod, bin ,db
>>>>    - make install and run it.
>>>>
>>>>
>>>> Is there additional place to clean?
>>>>
>>>>                         Thanks! __Yehavi:
>>>>
>>>>  2010/2/16 Anthony Minessale <anthony.minessale at gmail.com>
>>>>
>>>>> you may want to do a clean wipe of all files related to FS then.
>>>>> you clearly have some problem with legacy something or other because we
>>>>> don't see that on dozens of dev boxes.
>>>>>
>>>>> What os is it?
>>>>>
>>>>>
>>>>>
>>>>> On Tue, Feb 16, 2010 at 9:27 AM, Yehavi Bourvine <
>>>>> yehavi.bourvine at gmail.com> wrote:
>>>>>
>>>>>>  Tried this, but it didn't help. I delete these DB files before any
>>>>>> upgrade just to be sure.
>>>>>>
>>>>>>                   Thanks! __Yehavi:
>>>>>>
>>>>>>  2010/2/16 Anthony Minessale <anthony.minessale at gmail.com>
>>>>>>
>>>>>>> try removing all the .db files from /usr/local/freeswitch/db
>>>>>>>
>>>>>>>
>>>>>>>   On Tue, Feb 16, 2010 at 8:56 AM, Yehavi Bourvine <
>>>>>>> yehavi.bourvine at gmail.com> wrote:
>>>>>>>
>>>>>>>>   Hello,
>>>>>>>>
>>>>>>>>   After upgrading from 1.0.5pre10 to 1.0.5-20100216-0400 we started
>>>>>>>> getting the above errors (I append bellow two samples). It seems Freeswitch
>>>>>>>> fails to read a database using Sqlite.
>>>>>>>> Anyone have seen this?
>>>>>>>>
>>>>>>>> Other details: Fedora 10, SQlite 3.5.9.
>>>>>>>> We also do SQLite quesries during call setup via LUA from CoreDB. Is
>>>>>>>> it an SQLite problem?
>>>>>>>>
>>>>>>>>                          Thanks! __Yehavi:
>>>>>>>>
>>>>>>>> The samples:
>>>>>>>> 2010-02-16 16:22:13.762679 [ERR] switch_core_sqldb.c:404 SQL ERR
>>>>>>>> [library routin
>>>>>>>> e called out of sequence]
>>>>>>>> delete from sip_dialogs where call_id='
>>>>>>>> 8656841832142-120172129116107 at 10.64.1.2'
>>>>>>>>
>>>>>>>> 2010-02-16 16:42:00.802442 [ERR] switch_core_sqldb.c:722 SQL ERR:
>>>>>>>> [select call_i
>>>>>>>>
>>>>>>>> d,sip_user,sip_host,contact,status,rpid,expires,user_agent,server_user,server_ho
>>>>>>>>
>>>>>>>> st,profile_name,hostname,network_ip,network_port,sip_username,sip_realm,mwi_user
>>>>>>>> ,mwi_host from sip_registrations where profile_name='phones' and
>>>>>>>> contact like '%
>>>>>>>> 80635%'] library routine called out of sequence
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> --
>>>>>>> Anthony Minessale II
>>>>>>>
>>>>>>> FreeSWITCH http://www.freeswitch.org/
>>>>>>> ClueCon http://www.cluecon.com/
>>>>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>>>>
>>>>>>> AIM: anthm
>>>>>>> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
>>>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>>>>> IRC: irc.freenode.net #freeswitch
>>>>>>>
>>>>>>> FreeSWITCH Developer Conference
>>>>>>> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>>>>>>> iax:guest at conference.freeswitch.org/888
>>>>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>>>>> pstn:+19193869900
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> Anthony Minessale II
>>>>>
>>>>> FreeSWITCH http://www.freeswitch.org/
>>>>> ClueCon http://www.cluecon.com/
>>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>>
>>>>> AIM: anthm
>>>>> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>>> IRC: irc.freenode.net #freeswitch
>>>>>
>>>>> FreeSWITCH Developer Conference
>>>>> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>>>>> iax:guest at conference.freeswitch.org/888
>>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>>> pstn:+19193869900
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>> iax:guest at conference.freeswitch.org/888
>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> pstn:+19193869900
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100228/956a418b/attachment-0001.html 

From yehavi.bourvine at gmail.com  Sun Feb 28 06:47:56 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Sun, 28 Feb 2010 16:47:56 +0200
Subject: [Freeswitch-users] SQL ERR: library routine called out of
	sequence
In-Reply-To: <d361ca7b1002280514p71003588t8a0daebce5ba9e4d@mail.gmail.com>
References: <d361ca7b1002160656w7d616f6dn5e80087ebde94dbe@mail.gmail.com>
	<191c3a031002160705w4050c3cdk2bf2922e9aff5ba3@mail.gmail.com>
	<d361ca7b1002160727ucb6e0c8id62e803757333213@mail.gmail.com>
	<191c3a031002160755q569fe2ednfefb01b6d9cf7f0e@mail.gmail.com>
	<d361ca7b1002161059w1ef9484dr87e69a33aa2b62fc@mail.gmail.com>
	<191c3a031002161110s637a48a2r717ae94557fe9e9a@mail.gmail.com>
	<d361ca7b1002161130k3eb2d88by3100b30a9d29bd51@mail.gmail.com>
	<191c3a031002161230h2b6e6f9wf3f65ddb71f08297@mail.gmail.com>
	<d361ca7b1002280514p71003588t8a0daebce5ba9e4d@mail.gmail.com>
Message-ID: <d361ca7b1002280647l700206cfv6e130351d8ac6d5@mail.gmail.com>

After a few more tests I *think* it is related to SLA. Since it cannot be
reproduced consistenty I say "think".

What I have is:
- 80635 which is an SLA extension between two Polycoms.
- 80636 which is a private extension on Polycom
- 80632 which is a Cisco extension.
- 86111 which is connected behind a Cisco SIP<->PSTN gateway. Since this
gateway doesn't hav any login information it is not defined as a gateway but
accepted via ACL.

The tests that passed ok:
- From/to 80636/80632 to 80635 (i.e. - all internal).
- From/to 80635 to 80636/80632 (i.e. - all internal)
- From/to 80636/80632 to 86111 and vice versa (i.e. via the gateway, single
extensions).

The one that fails after a few attempts:
- From 86111 to 80635 (i.e. from the gateway to a shared extension).

I hope that this gives some more clues.

                          Thanks, __Yehavi:

2010/2/28 Yehavi Bourvine <yehavi.bourvine at gmail.com>

>  Hello,
>
>   I've installed a machine with CentOS-5.4 and the latest FreeSwitch
> (16841M). The problem still happens, even if I set sql-in-transactions to
> false. It happens also on a very light load.
>
> I would like to remind that we query the SQLite core DB from a LUA script
> which is called from the dial plan; might this be the cause of it?
>
> I am willing to test whatever you like, and can do it now as this machine
> is not in production yet.
>
>             Thanks! __Yehavi:
>
>  2010/2/16 Anthony Minessale <anthony.minessale at gmail.com>
>
>> Strange, even on abusive testing we have not seen this problem.
>>
>> please update to latest trunk.
>> There was only one change I can think of that may cause your issue and I
>> added a patch for it.
>> If it persists try setting the sql-in-transactions profile param to false.
>>
>>
>>
>>
>>
>>
>> On Tue, Feb 16, 2010 at 1:30 PM, Yehavi Bourvine <
>> yehavi.bourvine at gmail.com> wrote:
>>
>>>  Most of the queries are ok, only some fail, thus it doesn't look like
>>> permission problem. Furthermore, under 1.0.5pre10 it works for months.
>>>
>>> Might it be thread unsafe function calls? I've found the following while
>>> searching the WEB:
>>>
>>> *According to the MSDN docs, System.Timers.Timer operates in a thread
>>> pool.  If that's the case, your code is breaking the "connections cannot be
>>> shared across threads" rule for SQLit*
>>>
>>> Although it quotes MSDN, it might be related to Linux as well.
>>>
>>>                     Thanks, __Yehavi:
>>>
>>>  2010/2/16 Anthony Minessale <anthony.minessale at gmail.com>
>>>
>>>> That sounds about right.
>>>>
>>>> That error usually has something to do with using db calls on a closed
>>>> file or something along those lines.
>>>> Maybe you have a permission problem on the directory where the db files
>>>> are?
>>>>
>>>>
>>>>
>>>> On Tue, Feb 16, 2010 at 12:59 PM, Yehavi Bourvine <
>>>> yehavi.bourvine at gmail.com> wrote:
>>>>
>>>>>  The OS is Fedora-10 (soon to be upgraded to 12).
>>>>>
>>>>> What I do when I want to test a new version:
>>>>>
>>>>>    - Download the latest one into a fresh directory
>>>>>    - bootstrap.sh, configure and make
>>>>>    - stop Freeswitch, delete everything in lib, mod, bin ,db
>>>>>    - make install and run it.
>>>>>
>>>>>
>>>>> Is there additional place to clean?
>>>>>
>>>>>                         Thanks! __Yehavi:
>>>>>
>>>>>  2010/2/16 Anthony Minessale <anthony.minessale at gmail.com>
>>>>>
>>>>>> you may want to do a clean wipe of all files related to FS then.
>>>>>> you clearly have some problem with legacy something or other because
>>>>>> we don't see that on dozens of dev boxes.
>>>>>>
>>>>>> What os is it?
>>>>>>
>>>>>>
>>>>>>
>>>>>> On Tue, Feb 16, 2010 at 9:27 AM, Yehavi Bourvine <
>>>>>> yehavi.bourvine at gmail.com> wrote:
>>>>>>
>>>>>>>  Tried this, but it didn't help. I delete these DB files before any
>>>>>>> upgrade just to be sure.
>>>>>>>
>>>>>>>                   Thanks! __Yehavi:
>>>>>>>
>>>>>>>  2010/2/16 Anthony Minessale <anthony.minessale at gmail.com>
>>>>>>>
>>>>>>>> try removing all the .db files from /usr/local/freeswitch/db
>>>>>>>>
>>>>>>>>
>>>>>>>>   On Tue, Feb 16, 2010 at 8:56 AM, Yehavi Bourvine <
>>>>>>>> yehavi.bourvine at gmail.com> wrote:
>>>>>>>>
>>>>>>>>>   Hello,
>>>>>>>>>
>>>>>>>>>   After upgrading from 1.0.5pre10 to 1.0.5-20100216-0400 we started
>>>>>>>>> getting the above errors (I append bellow two samples). It seems Freeswitch
>>>>>>>>> fails to read a database using Sqlite.
>>>>>>>>> Anyone have seen this?
>>>>>>>>>
>>>>>>>>> Other details: Fedora 10, SQlite 3.5.9.
>>>>>>>>> We also do SQLite quesries during call setup via LUA from CoreDB.
>>>>>>>>> Is it an SQLite problem?
>>>>>>>>>
>>>>>>>>>                          Thanks! __Yehavi:
>>>>>>>>>
>>>>>>>>> The samples:
>>>>>>>>> 2010-02-16 16:22:13.762679 [ERR] switch_core_sqldb.c:404 SQL ERR
>>>>>>>>> [library routin
>>>>>>>>> e called out of sequence]
>>>>>>>>> delete from sip_dialogs where call_id='
>>>>>>>>> 8656841832142-120172129116107 at 10.64.1.2'
>>>>>>>>>
>>>>>>>>> 2010-02-16 16:42:00.802442 [ERR] switch_core_sqldb.c:722 SQL ERR:
>>>>>>>>> [select call_i
>>>>>>>>>
>>>>>>>>> d,sip_user,sip_host,contact,status,rpid,expires,user_agent,server_user,server_ho
>>>>>>>>>
>>>>>>>>> st,profile_name,hostname,network_ip,network_port,sip_username,sip_realm,mwi_user
>>>>>>>>> ,mwi_host from sip_registrations where profile_name='phones' and
>>>>>>>>> contact like '%
>>>>>>>>> 80635%'] library routine called out of sequence
>>>>>>>>>
>>>>>>>>> _______________________________________________
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>
>>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> --
>>>>>>>> Anthony Minessale II
>>>>>>>>
>>>>>>>> FreeSWITCH http://www.freeswitch.org/
>>>>>>>> ClueCon http://www.cluecon.com/
>>>>>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>>>>>
>>>>>>>> AIM: anthm
>>>>>>>> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
>>>>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>>>>>> IRC: irc.freenode.net #freeswitch
>>>>>>>>
>>>>>>>> FreeSWITCH Developer Conference
>>>>>>>> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>>>>>>>> iax:guest at conference.freeswitch.org/888
>>>>>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>>>>>> pstn:+19193869900
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>> Anthony Minessale II
>>>>>>
>>>>>> FreeSWITCH http://www.freeswitch.org/
>>>>>> ClueCon http://www.cluecon.com/
>>>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>>>
>>>>>> AIM: anthm
>>>>>> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
>>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>>>> IRC: irc.freenode.net #freeswitch
>>>>>>
>>>>>> FreeSWITCH Developer Conference
>>>>>> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>>>>>> iax:guest at conference.freeswitch.org/888
>>>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>>>> pstn:+19193869900
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>>
>>>> --
>>>> Anthony Minessale II
>>>>
>>>> FreeSWITCH http://www.freeswitch.org/
>>>> ClueCon http://www.cluecon.com/
>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>
>>>> AIM: anthm
>>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>> IRC: irc.freenode.net #freeswitch
>>>>
>>>> FreeSWITCH Developer Conference
>>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>>> iax:guest at conference.freeswitch.org/888
>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>> pstn:+19193869900
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100228/170a6557/attachment-0001.html 

From jbrucehopkins at gmail.com  Sun Feb 28 07:37:51 2010
From: jbrucehopkins at gmail.com (Bruce Hopkins)
Date: Sun, 28 Feb 2010 15:37:51 +0000
Subject: [Freeswitch-users] Kernel timer warning with CentOS 5.4
Message-ID: <d03a10e01002280737l61e1db84xa8388a73e932d460@mail.gmail.com>

Hi,

I wonder if anyone would be able to advise please:

When I build FreeSWITCH on yum updated CentOS 5.4, I get a warning when I
start FreeSWITCH that

"Abnormally large timer gap detected"
"Do you have your kernel timer set to greater than 1kHz?  You may experience
audio problems".

I get no such warning if I build on CentOS 5.3, and the test timings it
measures on starting FreeSWITCH do look lower.  All I was doing to upgrade
to Centos5.4 was a yum update on the 5.3 build.

I guess the warning comes from here:
http://fisheye.freeswitch.org/browse/~raw,r=16409/FreeSWITCH/src/switch_time.c

This is all on pretty low spec hardware - a couple of different Dell
optiplex p4's I use for testing.

Does anyone happen to know if I should just stick to Cent)S 5.3, or use 5.4
and not worry about the warnings, or if there is something I can do to fix
the problem it is warning about.  Perhaps it is just that I shouldn't use
such crummy hardware?!

Many thanks in advance
Bruce
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100228/aee29df4/attachment.html 

From gkuri at ieee.org  Sun Feb 28 09:47:14 2010
From: gkuri at ieee.org (Gabriel Kuri)
Date: Sun, 28 Feb 2010 09:47:14 -0800
Subject: [Freeswitch-users] Announcement: FreeSWITCH Add Broadsoft SCA
	Support
In-Reply-To: <d361ca7b1002280149i18e66cbdo2031e7f385f23075@mail.gmail.com>
References: <87f2f3b91001111742k691966cfs82515db77c13d4d4@mail.gmail.com>
	<d361ca7b1002170700x15630ab2tabc6f3d91ebc3303@mail.gmail.com>
	<ADC8B813-4161-481C-877E-93FC012858B4@freeswitch.org>
	<d361ca7b1002182101p60b4e8f8hf4dc95e1d2251e30@mail.gmail.com>
	<8b1c9cda1002182144m65ccf979w71384a1e7a55e2ea@mail.gmail.com>
	<d361ca7b1002182154o52352c62h8c0ec02fa66e8444@mail.gmail.com>
	<2F87B95C-EA23-4072-B845-3F7AE0617BE3@jerris.com>
	<d361ca7b1002182346k71fd602fh6e6725834a6de528@mail.gmail.com>
	<191c3a031002190908r3922efccqc6d223a5fabd458b@mail.gmail.com>
	<d361ca7b1002280149i18e66cbdo2031e7f385f23075@mail.gmail.com>
Message-ID: <8b1c9cda1002280947qcb5ec5anfeef645b447b1866@mail.gmail.com>

That's great it's working, but doesn't that seems more like a
workaround than an actual solution? What if you want to avail of DNS
and use SRV records, do you really want to be hardcoding all your
phones with IPs?

Cheers,
Gabe

On Sun, Feb 28, 2010 at 1:49 AM, Yehavi Bourvine
<yehavi.bourvine at gmail.com> wrote:
> Hello all,
>
> The problem is solved, at least for my Polycoms.
>
> The solution is to have "domain" variable (in vars.xml) set to the IP
> address of the profile that is used for phones registration (the default is
> ok if you have only one interface). Define the phone's registration and
> proxy servers to that address. You must use IP addresses and not DNS names.
>
> ????????????????? Thanks to Anthony for his helpfull tips!? __Yehavi:
>
> 2010/2/19 Anthony Minessale <anthony.minessale at gmail.com>
>>
>> go see my comments on that bug note.
>> be prepared to give us ssh access and call or irc so we can can see you
>> reproducing it.
>>
>> If you are not on the latest firmware on all the phones, we will not
>> continue with this process.
>>
>>
>>
>> On Fri, Feb 19, 2010 at 1:46 AM, Yehavi Bourvine
>> <yehavi.bourvine at gmail.com> wrote:
>>>
>>> A jira issue has been created: MODSOFIA-61.
>>>
>>> ????????????????? Thanks, __Yehavi:
>>>
>>> 2010/2/19 Michael Jerris <mike at jerris.com>
>>>>
>>>> If this issue is not already on jira could you please make sure it gets
>>>> added?
>>>> Mike
>>>>
>>>> On Feb 19, 2010, at 12:54 AM, Yehavi Bourvine
>>>> <yehavi.bourvine at gmail.com> wrote:
>>>>
>>>> Hello Gabe,
>>>>
>>>> ? As you can see - Brian is actively investigating it, so?you can?expect
>>>> for some fix soon...
>>>>
>>>> ?????????????????????? Regards, __Yehavi:
>>>>
>>>> 2010/2/19 Gabriel Kuri <gkuri at ieee.org>
>>>>>
>>>>> > ?When a call arrives, both ring; the one that did not answer gets
>>>>> > only a
>>>>> > cancel mesage without any further notification that the extension is
>>>>> > in use
>>>>> > by the other phone.
>>>>>
>>>>> These are the same exact symptoms I posted about earlier this week,
>>>>> with the Cisco SPA-5xx series phones. I still have yet to figure out
>>>>> why this is happening, if you find out what's going on, please post
>>>>> back the solution, I'd like to know the resolution.
>>>>>
>>>>> Thanks,
>>>>> Gabe
>>>>>
>>>>>
>>>>>
>>>>> On Thu, Feb 18, 2010 at 9:01 PM, Yehavi Bourvine
>>>>> <yehavi.bourvine at gmail.com> wrote:
>>>>> > Thanks Brian. It now works better, but not fully (using 16659M).
>>>>> >
>>>>> > What happens is:
>>>>> >
>>>>> > When one of the Polycoms seize the line it is ok?- the other phone
>>>>> > gets
>>>>> > notification and the extension status is "in use".
>>>>> > When?one of the Polycom phones initiates a call - all is ok:
>>>>> >
>>>>> > The other side sees that the extension is in use.
>>>>> > When it is put to hold all phones?who share this extension see it and
>>>>> > can
>>>>> > pick the call.
>>>>> >
>>>>>
>>>>> >
>>>>> > ???????????????????????? Thanks! __Yehavi:
>>>>> >
>>>>> > 2010/2/17 Brian West <brian at freeswitch.org>
>>>>> >>
>>>>> >> Step 1. Enable manage-shared-appearance=true
>>>>> >>
>>>>> >> Step 2. Now in the phone's config Configure the phone as usually,
>>>>> >> set the
>>>>> >> line shared and DO NOT set the third party name.
>>>>> >>
>>>>> >> Step 3. Reboot
>>>>> >>
>>>>> >> It should work.
>>>>> >>
>>>>> >> I wish someone that has this working would write some wiki docs
>>>>> >> these
>>>>> >> threads about it not working are getting rather old when I know for
>>>>> >> a fact
>>>>> >> they work fine.
>>>>> >>
>>>>> >> The gateway info missing is a gateway you have configured getting a
>>>>> >> notify. ?It has nothing to do with SCA.
>>>>> >>
>>>>> >> /b
>>>>> >>
>>>>> >> On Feb 17, 2010, at 9:00 AM, Yehavi Bourvine wrote:
>>>>> >>
>>>>> >> > .
>>>>> >>
>>>>> >>
>>>>> >> _______________________________________________
>>>>> >> FreeSWITCH-users mailing list
>>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> >>
>>>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> >> http://www.freeswitch.org
>>>>> >
>>>>> > _______________________________________________
>>>>> > FreeSWITCH-users mailing list
>>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> >
>>>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> > http://www.freeswitch.org
>>>>> >
>>>>> >
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From yehavi.bourvine at gmail.com  Sun Feb 28 09:56:21 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Sun, 28 Feb 2010 19:56:21 +0200
Subject: [Freeswitch-users] Announcement: FreeSWITCH Add Broadsoft SCA
	Support
In-Reply-To: <8b1c9cda1002280947qcb5ec5anfeef645b447b1866@mail.gmail.com>
References: <87f2f3b91001111742k691966cfs82515db77c13d4d4@mail.gmail.com>
	<ADC8B813-4161-481C-877E-93FC012858B4@freeswitch.org>
	<d361ca7b1002182101p60b4e8f8hf4dc95e1d2251e30@mail.gmail.com>
	<8b1c9cda1002182144m65ccf979w71384a1e7a55e2ea@mail.gmail.com>
	<d361ca7b1002182154o52352c62h8c0ec02fa66e8444@mail.gmail.com>
	<2F87B95C-EA23-4072-B845-3F7AE0617BE3@jerris.com>
	<d361ca7b1002182346k71fd602fh6e6725834a6de528@mail.gmail.com>
	<191c3a031002190908r3922efccqc6d223a5fabd458b@mail.gmail.com>
	<d361ca7b1002280149i18e66cbdo2031e7f385f23075@mail.gmail.com>
	<8b1c9cda1002280947qcb5ec5anfeef645b447b1866@mail.gmail.com>
Message-ID: <d361ca7b1002280956g8ef520bw5f4fb891cfc2d473@mail.gmail.com>

Hello Gabe,

  I agree that this is somewhat limiting, but with Polycom's central
provisioning (via XML files) I don't see this as a major drawback.

                         __Yehavi:

2010/2/28 Gabriel Kuri <gkuri at ieee.org>

> That's great it's working, but doesn't that seems more like a
> workaround than an actual solution? What if you want to avail of DNS
> and use SRV records, do you really want to be hardcoding all your
> phones with IPs?
>
> Cheers,
> Gabe
>
> On Sun, Feb 28, 2010 at 1:49 AM, Yehavi Bourvine
>  <yehavi.bourvine at gmail.com> wrote:
> > Hello all,
> >
> > The problem is solved, at least for my Polycoms.
> >
> > The solution is to have "domain" variable (in vars.xml) set to the IP
> > address of the profile that is used for phones registration (the default
> is
> > ok if you have only one interface). Define the phone's registration and
> > proxy servers to that address. You must use IP addresses and not DNS
> names.
> >
> >                   Thanks to Anthony for his helpfull tips!  __Yehavi:
> >
> > 2010/2/19 Anthony Minessale <anthony.minessale at gmail.com>
> >>
> >> go see my comments on that bug note.
> >> be prepared to give us ssh access and call or irc so we can can see you
> >> reproducing it.
> >>
> >> If you are not on the latest firmware on all the phones, we will not
> >> continue with this process.
> >>
> >>
> >>
> >> On Fri, Feb 19, 2010 at 1:46 AM, Yehavi Bourvine
> >> <yehavi.bourvine at gmail.com> wrote:
> >>>
> >>> A jira issue has been created: MODSOFIA-61.
> >>>
> >>>                   Thanks, __Yehavi:
> >>>
> >>> 2010/2/19 Michael Jerris <mike at jerris.com>
> >>>>
> >>>> If this issue is not already on jira could you please make sure it
> gets
> >>>> added?
> >>>> Mike
> >>>>
> >>>> On Feb 19, 2010, at 12:54 AM, Yehavi Bourvine
> >>>> <yehavi.bourvine at gmail.com> wrote:
> >>>>
> >>>> Hello Gabe,
> >>>>
> >>>>   As you can see - Brian is actively investigating it, so you
> can expect
> >>>> for some fix soon...
> >>>>
> >>>>                        Regards, __Yehavi:
> >>>>
> >>>> 2010/2/19 Gabriel Kuri <gkuri at ieee.org>
> >>>>>
> >>>>> >  When a call arrives, both ring; the one that did not answer gets
> >>>>> > only a
> >>>>> > cancel mesage without any further notification that the extension
> is
> >>>>> > in use
> >>>>> > by the other phone.
> >>>>>
> >>>>> These are the same exact symptoms I posted about earlier this week,
> >>>>> with the Cisco SPA-5xx series phones. I still have yet to figure out
> >>>>> why this is happening, if you find out what's going on, please post
> >>>>> back the solution, I'd like to know the resolution.
> >>>>>
> >>>>> Thanks,
> >>>>> Gabe
> >>>>>
> >>>>>
> >>>>>
> >>>>> On Thu, Feb 18, 2010 at 9:01 PM, Yehavi Bourvine
> >>>>> <yehavi.bourvine at gmail.com> wrote:
> >>>>> > Thanks Brian. It now works better, but not fully (using 16659M).
> >>>>> >
> >>>>> > What happens is:
> >>>>> >
> >>>>> > When one of the Polycoms seize the line it is ok - the other phone
> >>>>> > gets
> >>>>> > notification and the extension status is "in use".
> >>>>> > When one of the Polycom phones initiates a call - all is ok:
> >>>>> >
> >>>>> > The other side sees that the extension is in use.
> >>>>> > When it is put to hold all phones who share this extension see it
> and
> >>>>> > can
> >>>>> > pick the call.
> >>>>> >
> >>>>>
> >>>>> >
> >>>>> >                          Thanks! __Yehavi:
> >>>>> >
> >>>>> > 2010/2/17 Brian West <brian at freeswitch.org>
> >>>>> >>
> >>>>> >> Step 1. Enable manage-shared-appearance=true
> >>>>> >>
> >>>>> >> Step 2. Now in the phone's config Configure the phone as usually,
> >>>>> >> set the
> >>>>> >> line shared and DO NOT set the third party name.
> >>>>> >>
> >>>>> >> Step 3. Reboot
> >>>>> >>
> >>>>> >> It should work.
> >>>>> >>
> >>>>> >> I wish someone that has this working would write some wiki docs
> >>>>> >> these
> >>>>> >> threads about it not working are getting rather old when I know
> for
> >>>>> >> a fact
> >>>>> >> they work fine.
> >>>>> >>
> >>>>> >> The gateway info missing is a gateway you have configured getting
> a
> >>>>> >> notify.  It has nothing to do with SCA.
> >>>>> >>
> >>>>> >> /b
> >>>>> >>
> >>>>> >> On Feb 17, 2010, at 9:00 AM, Yehavi Bourvine wrote:
> >>>>> >>
> >>>>> >> > .
> >>>>> >>
> >>>>> >>
> >>>>> >> _______________________________________________
> >>>>> >> FreeSWITCH-users mailing list
> >>>>> >> FreeSWITCH-users at lists.freeswitch.org
> >>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>> >>
> >>>>> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>> >> http://www.freeswitch.org
> >>>>> >
> >>>>> > _______________________________________________
> >>>>> > FreeSWITCH-users mailing list
> >>>>> > FreeSWITCH-users at lists.freeswitch.org
> >>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>> >
> >>>>> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>> > http://www.freeswitch.org
> >>>>> >
> >>>>> >
> >>>>>
> >>>>> _______________________________________________
> >>>>> FreeSWITCH-users mailing list
> >>>>> FreeSWITCH-users at lists.freeswitch.org
> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>>
> >>>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>> http://www.freeswitch.org
> >>>>
> >>>> _______________________________________________
> >>>> FreeSWITCH-users mailing list
> >>>> FreeSWITCH-users at lists.freeswitch.org
> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>> http://www.freeswitch.org
> >>>>
> >>>> _______________________________________________
> >>>> FreeSWITCH-users mailing list
> >>>> FreeSWITCH-users at lists.freeswitch.org
> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>> http://www.freeswitch.org
> >>>>
> >>>
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>>
> >>
> >>
> >>
> >> --
> >> Anthony Minessale II
> >>
> >> FreeSWITCH http://www.freeswitch.org/
> >> ClueCon http://www.cluecon.com/
> >> Twitter: http://twitter.com/FreeSWITCH_wire
> >>
> >> AIM: anthm
> >> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >> IRC: irc.freenode.net #freeswitch
> >>
> >> FreeSWITCH Developer Conference
> >> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> >> iax:guest at conference.freeswitch.org/888
> >> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >> pstn:+19193869900
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100228/763fa274/attachment-0001.html 

From jbrucehopkins at gmail.com  Sun Feb 28 10:03:03 2010
From: jbrucehopkins at gmail.com (Bruce Hopkins)
Date: Sun, 28 Feb 2010 18:03:03 +0000
Subject: [Freeswitch-users] Kernel timer warning with CentOS 5.4
In-Reply-To: <d03a10e01002280737l61e1db84xa8388a73e932d460@mail.gmail.com>
References: <d03a10e01002280737l61e1db84xa8388a73e932d460@mail.gmail.com>
Message-ID: <d03a10e01002281003j669b6e75t31db4dc351f38ece@mail.gmail.com>

OK - I've realised I do get the same warning with CentOS 5.3, it just goes
past more quickly so I didn't see it.  Maybe it is just the hardware ....

On 28 February 2010 15:37, Bruce Hopkins <jbrucehopkins at gmail.com> wrote:

> Hi,
>
> I wonder if anyone would be able to advise please:
>
> When I build FreeSWITCH on yum updated CentOS 5.4, I get a warning when I
> start FreeSWITCH that
>
> "Abnormally large timer gap detected"
> "Do you have your kernel timer set to greater than 1kHz?  You may
> experience audio problems".
>
> I get no such warning if I build on CentOS 5.3, and the test timings it
> measures on starting FreeSWITCH do look lower.  All I was doing to upgrade
> to Centos5.4 was a yum update on the 5.3 build.
>
> I guess the warning comes from here:
> http://fisheye.freeswitch.org/browse/~raw,r=16409/FreeSWITCH/src/switch_time.c<http://fisheye.freeswitch.org/browse/%7Eraw,r=16409/FreeSWITCH/src/switch_time.c>
>
> This is all on pretty low spec hardware - a couple of different Dell
> optiplex p4's I use for testing.
>
> Does anyone happen to know if I should just stick to Cent)S 5.3, or use 5.4
> and not worry about the warnings, or if there is something I can do to fix
> the problem it is warning about.  Perhaps it is just that I shouldn't use
> such crummy hardware?!
>
> Many thanks in advance
> Bruce
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100228/c4424531/attachment.html 

From brian at freeswitch.org  Sun Feb 28 10:04:12 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 28 Feb 2010 12:04:12 -0600
Subject: [Freeswitch-users] Announcement: FreeSWITCH Add Broadsoft SCA
	Support
In-Reply-To: <d361ca7b1002280956g8ef520bw5f4fb891cfc2d473@mail.gmail.com>
References: <87f2f3b91001111742k691966cfs82515db77c13d4d4@mail.gmail.com>
	<ADC8B813-4161-481C-877E-93FC012858B4@freeswitch.org>
	<d361ca7b1002182101p60b4e8f8hf4dc95e1d2251e30@mail.gmail.com>
	<8b1c9cda1002182144m65ccf979w71384a1e7a55e2ea@mail.gmail.com>
	<d361ca7b1002182154o52352c62h8c0ec02fa66e8444@mail.gmail.com>
	<2F87B95C-EA23-4072-B845-3F7AE0617BE3@jerris.com>
	<d361ca7b1002182346k71fd602fh6e6725834a6de528@mail.gmail.com>
	<191c3a031002190908r3922efccqc6d223a5fabd458b@mail.gmail.com>
	<d361ca7b1002280149i18e66cbdo2031e7f385f23075@mail.gmail.com>
	<8b1c9cda1002280947qcb5ec5anfeef645b447b1866@mail.gmail.com>
	<d361ca7b1002280956g8ef520bw5f4fb891cfc2d473@mail.gmail.com>
Message-ID: <987F87B1-FA49-47B4-9823-5A7140459AFD@freeswitch.org>

Really?  Come on guys... the feature is something you can't find elsewhere without paying and you're all still not totally pleased with it?  

/me shakes his head.

We have already started talking about how to make the feature more robust. 

/b

On Feb 28, 2010, at 11:56 AM, Yehavi Bourvine wrote:

> Hello Gabe,
>  
>   I agree that this is somewhat limiting, but with Polycom's central provisioning (via XML files) I don't see this as a major drawback.
>  
>                          __Yehavi:



From yehavi.bourvine at gmail.com  Sun Feb 28 10:23:53 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Sun, 28 Feb 2010 20:23:53 +0200
Subject: [Freeswitch-users] Announcement: FreeSWITCH Add Broadsoft SCA
	Support
In-Reply-To: <987F87B1-FA49-47B4-9823-5A7140459AFD@freeswitch.org>
References: <87f2f3b91001111742k691966cfs82515db77c13d4d4@mail.gmail.com>
	<8b1c9cda1002182144m65ccf979w71384a1e7a55e2ea@mail.gmail.com>
	<d361ca7b1002182154o52352c62h8c0ec02fa66e8444@mail.gmail.com>
	<2F87B95C-EA23-4072-B845-3F7AE0617BE3@jerris.com>
	<d361ca7b1002182346k71fd602fh6e6725834a6de528@mail.gmail.com>
	<191c3a031002190908r3922efccqc6d223a5fabd458b@mail.gmail.com>
	<d361ca7b1002280149i18e66cbdo2031e7f385f23075@mail.gmail.com>
	<8b1c9cda1002280947qcb5ec5anfeef645b447b1866@mail.gmail.com>
	<d361ca7b1002280956g8ef520bw5f4fb891cfc2d473@mail.gmail.com>
	<987F87B1-FA49-47B4-9823-5A7140459AFD@freeswitch.org>
Message-ID: <d361ca7b1002281023s38d81815t48d985aa830a87c9@mail.gmail.com>

This is one of the most important feature my users want. They don't care how
I do it, they are just happy it works.

                        Thanks! __Yehavi:

2010/2/28 Brian West <brian at freeswitch.org>

> Really?  Come on guys... the feature is something you can't find elsewhere
> without paying and you're all still not totally pleased with it?
>
> /me shakes his head.
>
> We have already started talking about how to make the feature more robust.
>
> /b
>
> On Feb 28, 2010, at 11:56 AM, Yehavi Bourvine wrote:
>
> > Hello Gabe,
> >
> >   I agree that this is somewhat limiting, but with Polycom's central
> provisioning (via XML files) I don't see this as a major drawback.
> >
> >                          __Yehavi:
>
>
>  _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100228/0dcb42a2/attachment.html 

From anthony.minessale at gmail.com  Sun Feb 28 21:06:38 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sun, 28 Feb 2010 23:06:38 -0600
Subject: [Freeswitch-users] SQL ERR: library routine called out of
	sequence
In-Reply-To: <191c3a031002282105p4ed80135jc115a007ec7e0d4a@mail.gmail.com>
References: <d361ca7b1002160656w7d616f6dn5e80087ebde94dbe@mail.gmail.com>
	<d361ca7b1002160727ucb6e0c8id62e803757333213@mail.gmail.com>
	<191c3a031002160755q569fe2ednfefb01b6d9cf7f0e@mail.gmail.com>
	<d361ca7b1002161059w1ef9484dr87e69a33aa2b62fc@mail.gmail.com>
	<191c3a031002161110s637a48a2r717ae94557fe9e9a@mail.gmail.com>
	<d361ca7b1002161130k3eb2d88by3100b30a9d29bd51@mail.gmail.com>
	<191c3a031002161230h2b6e6f9wf3f65ddb71f08297@mail.gmail.com>
	<d361ca7b1002280514p71003588t8a0daebce5ba9e4d@mail.gmail.com>
	<d361ca7b1002280647l700206cfv6e130351d8ac6d5@mail.gmail.com>
	<191c3a031002282105p4ed80135jc115a007ec7e0d4a@mail.gmail.com>
Message-ID: <191c3a031002282106t59801ddctb12c4fab160e9a07@mail.gmail.com>

You probably had it right in the last email.
Polling the core's db from lua is not recommended.
We use sla extensively and never once see your issue.

On Feb 28, 2010 8:55 AM, "Yehavi Bourvine" <yehavi.bourvine at gmail.com>
wrote:

After a few more tests I *think* it is related to SLA. Since it cannot be
reproduced consistenty I say "think".

What I have is:
- 80635 which is an SLA extension between two Polycoms.
- 80636 which is a private extension on Polycom
- 80632 which is a Cisco extension.
- 86111 which is connected behind a Cisco SIP<->PSTN gateway. Since this
gateway doesn't hav any login information it is not defined as a gateway but
accepted via ACL.

The tests that passed ok:
- From/to 80636/80632 to 80635 (i.e. - all internal).
- From/to 80635 to 80636/80632 (i.e. - all internal)
- From/to 80636/80632 to 86111 and vice versa (i.e. via the gateway, single
extensions).

The one that fails after a few attempts:
- From 86111 to 80635 (i.e. from the gateway to a shared extension).

I hope that this gives some more clues.

                          Thanks, __Yehavi:

2010/2/28 Yehavi Bourvine <yehavi.bourvine at gmail.com>


>
> Hello,
>
>   I've installed a machine with CentOS-5.4 and the latest FreeSwitch
(16841M). The...

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100228/bd2346ea/attachment.html 

From lakindia89 at gmail.com  Sun Feb 28 22:16:20 2010
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Mon, 1 Mar 2010 11:46:20 +0530
Subject: [Freeswitch-users] smg_prid not bridging the call
In-Reply-To: <c4d05cbe1002271035g6112b277hdf305b61233b94ad@mail.gmail.com>
References: <7d79b3931002252155t47c86968q9e451482926d93ac@mail.gmail.com>
	<c4d05cbe1002260731v29dbc428i1e72d24a220c62eb@mail.gmail.com>
	<7d79b3931002262057g3b325bctcdbcfb833d4aaf55@mail.gmail.com>
	<7d79b3931002262102w79e16ae1nd786600b44dfb758@mail.gmail.com>
	<7d79b3931002262113s49495e7cga59715f41fdbb35d@mail.gmail.com>
	<c4d05cbe1002271035g6112b277hdf305b61233b94ad@mail.gmail.com>
Message-ID: <7d79b3931002282216w2e2ec844q36d28b3f50423ec4@mail.gmail.com>

Dear Moy,
That's didn't seem to solve the problem.
I gave the following command.
originate
{origination_caller_id_number=04439114600}openzap/smg_prid/a/9952248266
&bridge({origination_caller_id_number=04439114600}openzap/smg_prid/a/9976975781)

The D-Chan Log is
http://pastebin.freeswitch.org/12268

Kindly refer the attached pcap file that I captured with wanpipemon utility.
I think it might help.


On Sun, Feb 28, 2010 at 12:05 AM, Moises Silva <moises.silva at gmail.com>wrote:

> I believe the problem FreeSWITCH is setting that as a default callerid
> name, which your telco does not like.
>
> Try setting the caller id name and number by yourself as explained in the
> "originate" section here http://wiki.freeswitch.org/wiki/Mod_commands
>  <http://wiki.freeswitch.org/wiki/Mod_commands>
>
> On Sat, Feb 27, 2010 at 12:13 AM, lakshmanan ganapathy <
> lakindia89 at gmail.com> wrote:
>
>> I think it says Invalid Information Element for the DISPLAY
>> smg_prid/a/8122133885!!!
>> correct?? If so, can you please help me to solve this?
>>
>>
>> On Sat, Feb 27, 2010 at 10:32 AM, lakshmanan ganapathy <
>> lakindia89 at gmail.com> wrote:
>>
>>> In the Dchan log it is saying Invalid Information Elements. That might be
>>> a problem??? But I even don't know why it is saying Invalid Information
>>> Element??
>>> Please guide me!!!
>>>
>>>
>>>
>>> On Sat, Feb 27, 2010 at 10:27 AM, lakshmanan ganapathy <
>>> lakindia89 at gmail.com> wrote:
>>>
>>>> Dear Moy,
>>>> Here are the details:
>>>>
>>>> FreeSwitch Log:
>>>> http://pastebin.freeswitch.org/12256
>>>>
>>>>  /var/log/sangoma_pri/dchan_<span-no>.log:
>>>>  http://pastebin.freeswitch.org/12257
>>>>
>>>> /var/log/sangoma_mgd.log:
>>>> http://pastebin.freeswitch.org/12258
>>>>
>>>> smg_pri.conf
>>>> http://pastebin.freeswitch.org/12259
>>>>
>>>>
>>>>
>>>> On Fri, Feb 26, 2010 at 9:01 PM, Moises Silva <moises.silva at gmail.com>wrote:
>>>>
>>>>> Hello lakshmanan,
>>>>>
>>>>> Please enable the dchan_log=q931 in /etc/wanpipe/smg_pri.conf and then
>>>>> restart it (smg_ctrl restart), then pastebin the logs
>>>>>
>>>>>  /var/log/sangoma_pri/dchan_<span-no>.log
>>>>> /var/log/sangoma_mgd.log
>>>>>
>>>>> That will contain the Q931 details (if any). Also pastebin your
>>>>> smg_pri.conf.
>>>>>
>>>>> Also enable the FreeSWITCH debug logging (see the FreeSWITCH wiki for
>>>>> details about that) and paste them too.
>>>>>
>>>>> --
>>>>> Moises Silva
>>>>> Senior Software Engineer
>>>>> Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON
>>>>> L3R 9T3 Canada
>>>>> t. 1 905 474 1990 x 128 | e. moy at sangoma.com
>>>>>
>>>>>  On Fri, Feb 26, 2010 at 12:55 AM, lakshmanan ganapathy <
>>>>> lakindia89 at gmail.com> wrote:
>>>>>
>>>>>>  Dear all,
>>>>>> I'm having a A102 Sangoma hardware. I configured it with freeswitch.
>>>>>> wanrouter status, says both the port as connected.
>>>>>> My smg_prid version is
>>>>>>
>>>>>> Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: ================System
>>>>>> restart=============
>>>>>> Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: = Sangoma PRI Protocol
>>>>>> Stack Daemon       =
>>>>>> Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: = Version:
>>>>>> 1.54                           =
>>>>>> Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: = Date: Feb 15
>>>>>> 2010                       =
>>>>>> Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: = Wanpipe Release:
>>>>>> wanpipe-3.5.8.6       =
>>>>>> Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: = Revision:Revision:
>>>>>> 15288                =
>>>>>> Feb 26 16:08:14 FMS-FreeSwitch
>>>>>> sangoma_prid:
>>>>>> ===========================================
>>>>>>
>>>>>> My freeswitch version is 16729.
>>>>>> I started freeswitch.
>>>>>>
>>>>>> oz list
>>>>>> +OK
>>>>>> span: 1 (smg_prid)
>>>>>> type: Sangoma (boost)
>>>>>> chan_count: 60
>>>>>> dialplan: XML
>>>>>> context: default
>>>>>> dial_regex:
>>>>>> fail_dial_regex:
>>>>>> hold_music:
>>>>>> analog_options none
>>>>>>
>>>>>> I originated a call as
>>>>>> originate openzap/smg_prid/a/9952248266 &park(), which hits my mobile.
>>>>>>
>>>>>> But when I issued the following command:
>>>>>> originate openzap/smg_prid/a/9952248266
>>>>>> &bridge(openzap/smg_prid/a/8122133885)
>>>>>> It rings my mobile (9952248266) first, but after that the following
>>>>>> error was displayed
>>>>>>
>>>>>> 2010-02-26 16:20:51.736080 [ERR] switch_ivr_originate.c:2387 Cannot
>>>>>> create outgoing channel of type [openzap] cause: [NORMAL_CIRCUIT_CONGESTION]
>>>>>> The call got ended in my mobile.
>>>>>>
>>>>>> Freeswitch log and smg_pri.conf
>>>>>>           http://pastebin.freeswitch.org/12248
>>>>>> openzap.conf:
>>>>>> [span wanpipe smg_prid]
>>>>>> name => smg_prid
>>>>>> trunk_type =>e1
>>>>>> b-channel => 1:1-15
>>>>>> b-channel => 1:17-31
>>>>>> trunk_type =>e1
>>>>>> b-channel => 2:1-15
>>>>>> b-channel => 2:17-31
>>>>>>
>>>>>> openzap.conf.xml:
>>>>>> <configuration name="openzap.conf" description="OpenZAP
>>>>>> Configuration">
>>>>>>   <settings>
>>>>>>     <param name="debug" value="0"/>
>>>>>>   </settings>
>>>>>>   <boost_spans>
>>>>>>         <span id="smg_prid">
>>>>>>         <param name="dialplan" value="XML"/>
>>>>>>         <param name="context" value="default"/>
>>>>>>         </span>
>>>>>>   </boost_spans>
>>>>>> </configuration>
>>>>>>
>>>>>> Please guide me to setup this one!!.
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Moises Silva
> Senior Software Engineer
> Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R
> 9T3 Canada
> t. 1 905 474 1990 x 128 | e. moy at sangoma.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100301/5d790312/attachment-0001.html 
-------------- next part --------------
A non-text attachment was scrubbed...
Name: isdn.pcap
Type: application/cap
Size: 1231 bytes
Desc: not available
Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100301/5d790312/attachment-0001.bin 

From srinivas.ksvreddy at gmail.com  Sun Feb 28 23:39:56 2010
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Mon, 1 Mar 2010 13:09:56 +0530
Subject: [Freeswitch-users] call routing from freeswitch based on INVITE
Message-ID: <f8af5741002282339l52fe9ad7q44a7e4c65f82cd95@mail.gmail.com>

I have two sipservers like server1 and server2,  if sever1 receives invite
packet like

INVITE From:   1000 at server1.domain.com
           To:   1002 at server2.railvoice.com.

how can i route the invite packet to server2 from server1,

Thanks & Regards
Srinivasula Reddy K
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100301/59f89744/attachment.html 

From tomek.augustyn at gmail.com  Sat Feb 27 00:33:44 2010
From: tomek.augustyn at gmail.com (Tomasz Augustyn)
Date: Sat, 27 Feb 2010 09:33:44 +0100
Subject: [Freeswitch-users] smg_prid not bridging the call
Message-ID: <6d15d07f1002270033n71d8ac85u1895b05f75540e63@mail.gmail.com>

Hello,

I had similar problem and I think it is more a problem between Sangoma card
and your E1 provider than with freeswitch.

In my case it was necessary to set "origination_caller_id_number" to one of
the telephone numbers linked to my E1 line. In other case the calls were
rejected with "invalid information element" error.

You can try Sangoma's support they are very helpful.

Tomasz Augustyn


---------- Forwarded message ----------
From: lakshmanan ganapathy <lakindia89 at gmail.com>
To: freeswitch-users at lists.freeswitch.org
Date: Sat, 27 Feb 2010 10:32:08 +0530
Subject: Re: [Freeswitch-users] smg_prid not bridging the call
In the Dchan log it is saying Invalid Information Elements. That might be a
problem??? But I even don't know why it is saying Invalid Information
Element??
Please guide me!!!
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100227/88e0b91e/attachment.html 

From mmg at transtelco.net  Sun Feb 28 21:20:28 2010
From: mmg at transtelco.net (=?iso-8859-1?Q?Manuel_Mar=EDn?=)
Date: Mon, 1 Mar 2010 00:20:28 -0500
Subject: [Freeswitch-users] High CPU usage 1.0.5
Message-ID: <4502F03F8260234AB94179D6E1BDD0CF3F53555A73@VMBX113.ihostexchange.net>

Dear freeswitch group

I Just upgraded a system from 1.0.4 to 1.0.5 and we are seeing high CPU usage even if there are only a few calls on the system or no calls at all. We are running Debian with kernel 2.6.26-2-686

Anyone experimenting a similar issue?


Thanks in advance


freeswitch at internal> version
FreeSWITCH Version 1.0.5-20100225-0400 (16810M)






Manuel Mar?n
Transtelco
US 1.915.2172232
MX 52.656.6921109
FAX 1.915.2311214

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100301/22dc4991/attachment.html 

From mailinglist at fribert.dk  Mon Feb  1 00:28:51 2010
From: mailinglist at fribert.dk (mailinglist)
Date: Mon, 01 Feb 2010 09:28:51 +0100
Subject: [Freeswitch-users] Svar: Re: Somebody help me understand the	
	'features' set up please :-)
In-Reply-To: <b39273531001311645u618ea87am8a3d84ee6b1bcfcb@mail.gmail.com>
References: <4B66226C020000E10000043C@mail.fribert.dk>
	<b39273531001311645u618ea87am8a3d84ee6b1bcfcb@mail.gmail.com>
Message-ID: <4B669ED3020000E100000447@mail.fribert.dk>

Hi Rupa
 
Ahh, I see.
Looking at example 2, and http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bind_meta_app
I still need to get where I enter this.
Is it in the call handling in the dialplan?
So I've gotten my current entry to handle incoming call from the outside:
 
<extension name="PrivateIn">
   <condition field="destination_number" expression="^(8202)$">
       <action application="set" data="call_timeout=45"/>
       <action application="set" data="domain_name=10.11.12.25"/>
       <action application="set" data="continue_on_fail=true"/>
       <action application="set" data="hangup_after_bridge=true"/>
       <action application="set" data="transfer_ringback=$${hold_music}"/>
       <action application="bridge" data="group/private@${domain_name}"/ ( 'mailto:group/private@$domain_name"/' )>
       <action application="answer" data=""/>
       <action application="sleep" data="1000"/>
       <action application="voicemail" data="default 10.11.12.25 45"/>
   </condition>
</extension>
Do I stick in there, or do I enter it in features.xml, or?
 
 
Best regards
Fribse
 
/Running Freeswitch on pfSense running on a VMWare ESXi 4.0


>>> 01-02-2010 kl. 01:45 skrev Rupa Schomaker <rupa at rupa.com> i meddelelsen <b39273531001311645u618ea87am8a3d84ee6b1bcfcb at mail.gmail.com>:

Look at bind_meta_app in the default dialplan.  It binds the dtmf to
the features context.

On Sun, Jan 31, 2010 at 5:38 PM, mailinglist <mailinglist at fribert.dk> wrote:
> Ok, I've gotten the Freeswitch to register to my VoIP provider.
> I've gotten my phones to register to Freeswitch, and I can receive and make
> calls, all very nice.
>
> I've gotten a Sipura SPA901 and a Siemens Gigaset S68IP registered to the
> Freeswitch.
>
> When I receive a call, I would like to be able to transfer the call to
> another phone, or change the call to a conference call with two local
> phones.
>
> So I've been looking at the examples in the wiki, and I can't make them
> work, not as I understand them anyways. Especially the att_xfer seems to be
> able to do what I need.
>
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_att_xfer
>
> As I understand Example1, I should answer the call, and then press *3 during
> the call, and either transfer it or change it to a threeway call.
>
> I get the first part, create an extension in the dialplan called att_xfer.
> But what is meant by the second par 'then bind this feature to DTMF 3', how
> do I enter that, and where?
>
> I hope somebody can help me with this (again)?
>
>
>
> Best regards
> Fribse
>
> /Running Freeswitch on pfSense running on a VMWare ESXi 4.0
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
-Rupa

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100201/fbf23b78/attachment-0002.html 

From codecomplete at free.fr  Mon Feb  1 01:22:28 2010
From: codecomplete at free.fr (Fred-145)
Date: Mon, 01 Feb 2010 10:22:28 +0100
Subject: [Freeswitch-users] Equivalent to Asterisk's"directrtpsetup=yes"?
References: <20100129164359.9B3C22B10D@cuneorg-email.cune.pri><j4e8m5tqfl3fo8ug7d3sdgdiu7gofidbj8@4ax.com><j4e8m5tqfl3fo8ug7d3sdgdiu7gofidbj8-e09XROE/p8c@public.gmane.org><4468a6771001310247n4bf5d1a4rf4aea9f2d6c35a1f@mail.gmail.com>
	<qiqbm55lueuqup1m6r6d0mff36b4l4fb9l@4ax.com>
	<15D48404014D48D19F85CFFFC4BBC76F@cune.pri>
	<15D48404014D48D19F85CFFFC4BBC76F-u4Jt3PGDs+M@public.gmane.org>
	<7FA9CDA6-EAB3-4C1D-9CD3-8D4B06B0F335@freeswitch.org>
Message-ID: <b87dm5dn1qg1pkk8480064qjvc9j62jvj1@4ax.com>

On Sun, 31 Jan 2010 22:36:25 -0600, Brian West
<brian at freeswitch.org> wrote:
>Going back no_media after hold isn't supported yet.. Anthony said he would add it if someone really really wanted it and posted a bounty of $500 to cover his time to implement it.

Thanks all for the info. The bottom line seems to be that it's really
not a good idea to remove Freeswitch from the media path, and it's a
better idea to look elsewhere if scalability is an issue.




From mailinglist at fribert.dk  Mon Feb  1 00:57:45 2010
From: mailinglist at fribert.dk (mailinglist)
Date: Mon, 01 Feb 2010 09:57:45 +0100
Subject: [Freeswitch-users] Svar: Re: Somebody help me understand the		
	'features' set up please :-)
In-Reply-To: <4B669ED3020000E100000447@mail.fribert.dk>
References: <4B66226C020000E10000043C@mail.fribert.dk>
	<b39273531001311645u618ea87am8a3d84ee6b1bcfcb@mail.gmail.com>
	<4B669ED3020000E100000447@mail.fribert.dk>
Message-ID: <4B66A599020000E10000044C@mail.fribert.dk>

Sorry, I think I'm being unclear here.
 
I should add 
    <extension name="att_xfer">
       <condition field="destination_number" expression="^att_xfer$">
           <action application="read" data="3 4 sounds/getdigits.wav attxfer_callthis 30000 #"/>
           <action application="att_xfer" data="sofia/default/${attxfer_callthis}"/>
       </condition>
    </extension>
 
to the features.xml
 
Then add 
 
<action application="bind_meta_app" data="3 a a execute_extension::att_xfer XML features"/>
 
To the handling of the incoming call, right?
 
Something like
 
<extension name="PrivateIn">
   <condition field="destination_number" expression="^(8202)$">
       <action application="set" data="call_timeout=45"/>
       <action application="set" data="domain_name=10.11.12.25"/>
       <action application="set" data="continue_on_fail=true"/>
       <action application="set" data="hangup_after_bridge=true"/>
       <action application="set" data="transfer_ringback=$${hold_music}"/>
       <action application="bind_meta_app" data="3 a a execute_extension::att_xfer XML features"/>
       <action application="bridge" data="group/private@${domain_name}"/ ( mailto:group/private@$%7Bdomain_name%7D%22/ )>
       <action application="answer" data=""/>
       <action application="sleep" data="1000"/>
       <action application="voicemail" data="default 10.11.12.25 45"/>
   </condition>
</extension>
 
or?
 
 
 
Best regards
Fribse
 
/Running Freeswitch on pfSense running on a VMWare ESXi 4.0

>>> 01-02-2010 kl. 09:28 skrev "mailinglist" <mailinglist at fribert.dk> i meddelelsen <4B669ED3020000E100000447 at mail.fribert.dk>:

Hi Rupa
 
Ahh, I see.
Looking at example 2, and http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bind_meta_app
I still need to get where I enter this.
Is it in the call handling in the dialplan?
So I've gotten my current entry to handle incoming call from the outside:
 
<extension name="PrivateIn">
   <condition field="destination_number" expression="^(8202)$">
       <action application="set" data="call_timeout=45"/>
       <action application="set" data="domain_name=10.11.12.25"/>
       <action application="set" data="continue_on_fail=true"/>
       <action application="set" data="hangup_after_bridge=true"/>
       <action application="set" data="transfer_ringback=$${hold_music}"/>
       <action application="bridge" data="group/private@${domain_name}"/ ( 'mailto:group/private@$domain_name"/' )>
       <action application="answer" data=""/>
       <action application="sleep" data="1000"/>
       <action application="voicemail" data="default 10.11.12.25 45"/>
   </condition>
</extension>
Do I stick in there, or do I enter it in features.xml, or?
 
 
Best regards
Fribse
 
/Running Freeswitch on pfSense running on a VMWare ESXi 4.0


>>> 01-02-2010 kl. 01:45 skrev Rupa Schomaker <rupa at rupa.com> i meddelelsen <b39273531001311645u618ea87am8a3d84ee6b1bcfcb at mail.gmail.com>:

Look at bind_meta_app in the default dialplan.  It binds the dtmf to
the features context.

On Sun, Jan 31, 2010 at 5:38 PM, mailinglist <mailinglist at fribert.dk> wrote:
> Ok, I've gotten the Freeswitch to register to my VoIP provider.
> I've gotten my phones to register to Freeswitch, and I can receive and make
> calls, all very nice.
>
> I've gotten a Sipura SPA901 and a Siemens Gigaset S68IP registered to the
> Freeswitch.
>
> When I receive a call, I would like to be able to transfer the call to
> another phone, or change the call to a conference call with two local
> phones.
>
> So I've been looking at the examples in the wiki, and I can't make them
> work, not as I understand them anyways. Especially the att_xfer seems to be
> able to do what I need.
>
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_att_xfer
>
> As I understand Example1, I should answer the call, and then press *3 during
> the call, and either transfer it or change it to a threeway call.
>
> I get the first part, create an extension in the dialplan called att_xfer.
> But what is meant by the second par 'then bind this feature to DTMF 3', how
> do I enter that, and where?
>
> I hope somebody can help me with this (again)?
>
>
>
> Best regards
> Fribse
>
> /Running Freeswitch on pfSense running on a VMWare ESXi 4.0
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
-Rupa

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100201/58efb0ae/attachment-0002.html 

From jason at jasonjgw.net  Mon Feb  1 02:43:37 2010
From: jason at jasonjgw.net (Jason White)
Date: Mon, 1 Feb 2010 21:43:37 +1100
Subject: [Freeswitch-users] Equivalent to Asterisk's"directrtpsetup=yes"?
In-Reply-To: <b87dm5dn1qg1pkk8480064qjvc9j62jvj1@4ax.com>
References: <20100129164359.9B3C22B10D@cuneorg-email.cune.pri>
	<j4e8m5tqfl3fo8ug7d3sdgdiu7gofidbj8@4ax.com>
	<j4e8m5tqfl3fo8ug7d3sdgdiu7gofidbj8-e09XROE/p8c@public.gmane.org>
	<4468a6771001310247n4bf5d1a4rf4aea9f2d6c35a1f@mail.gmail.com>
	<qiqbm55lueuqup1m6r6d0mff36b4l4fb9l@4ax.com>
	<15D48404014D48D19F85CFFFC4BBC76F@cune.pri>
	<15D48404014D48D19F85CFFFC4BBC76F-u4Jt3PGDs+M@public.gmane.org>
	<7FA9CDA6-EAB3-4C1D-9CD3-8D4B06B0F335@freeswitch.org>
	<b87dm5dn1qg1pkk8480064qjvc9j62jvj1@4ax.com>
Message-ID: <20100201104337.GA985@jdc.jasonjgw.net>

Fred-145 <codecomplete at free.fr> wrote:
> On Sun, 31 Jan 2010 22:36:25 -0600, Brian West
> <brian at freeswitch.org> wrote:
> >Going back no_media after hold isn't supported yet.. Anthony said he would add it if someone really really wanted it and posted a bounty of $500 to cover his time to implement it.
> 
> Thanks all for the info. The bottom line seems to be that it's really
> not a good idea to remove Freeswitch from the media path, and it's a
> better idea to look elsewhere if scalability is an issue.

I don't think that's a good summary. What contributors to this thread are
saying is that there are trade-offs: you can keep FreeSWITCH in the media
path, with its consequent load on the server, or you can configure it for
bypass media or proxy media modes to reduce the load. In the latter case, nat
traversal issues may have to be dealt with, if relevant to your situation, and
if anyone wants to return to bypass media after hold, they're welcome to pay
the bounty to have this functionality implemented.





From lakindia89 at gmail.com  Mon Feb  1 02:57:25 2010
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Mon, 1 Feb 2010 16:27:25 +0530
Subject: [Freeswitch-users] nixevent behavior
In-Reply-To: <191c3a031001300848h65d65c9cg9b355cd07e922@mail.gmail.com>
References: <7d79b3931001300648j6aa55258yfc496d9cea5c4b8b@mail.gmail.com>
	<191c3a031001300848h65d65c9cg9b355cd07e922@mail.gmail.com>
Message-ID: <7d79b3931002010257hc84a997k26bdc172f27315bf@mail.gmail.com>

Thanks antony.. That work's well. I also understood the functionality of
send and sendRecv. Thanks..



On Sat, Jan 30, 2010 at 10:18 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> use $e = $con->sendRecv("command");
>
> every time
> for each send you do you must do a recv so this does both.
>
>
> On Sat, Jan 30, 2010 at 8:48 AM, lakshmanan ganapathy <
> lakindia89 at gmail.com> wrote:
>
>> Dear all
>>
>> I've done the following sample script to experiment the nixevent. I found
>> some difference in behavior because of nixevent. Let me explain my question
>> down the script.
>>
>> require ESL;
>> use IO::Socket::INET;
>>
>> my $ip = "192.168.1.222";
>> my $sock = new IO::Socket::INET ( LocalHost => $ip,  LocalPort => '8447',
>> Proto => 'tcp',  Listen => 1,  Reuse => 1 );
>> die "Could not create socket: $!\n" unless $sock;
>> for(;;) {
>>     my $new_sock = $sock->accept();
>>     next if (not defined ($new_sock));
>>     my $pid = fork();
>>     if ($pid) {
>>         close($new_sock);
>>         next;
>>     }
>>     print "CHILD PID: $$\n";
>>     my $host = $new_sock->sockhost();
>>     my $fd = fileno($new_sock);
>>
>>     my $con = new ESL::ESLconnection($fd);
>>     my $info = $con->getInfo();
>>
>>     my $uuid = $info->getHeader("unique-id");
>>
>>     printf "Connected call %s, from %s\n", $uuid,
>> $info->getHeader("caller-caller-id-number");
>>     my $r=$con->execute("answer");
>>     $con->events("plain","all");
>> ##########################
>>     $con->send("nixevent DTMF");
>>     my $val=$con->api("create_uuid");
>>     $val = $val->getBody();            # LINE 1
>>     chomp($val);
>>     print "UUID is $val\n";
>>     my $e = $con->recvEvent();
>>     $val = $e->getBody();               # LINE 2
>>     chomp($val);
>>     print "UUID is $val\n";
>>     close($new_sock);
>> }
>>
>> # If the line ($con->send("nixevent DTMF");) is commented, then the result
>> of create_uuid is obtained in LINE 1.
>> # else, the result isn't obtained in the LINE 1 and it has nothing. The
>> result is obtained only when I do a recvEvent,
>> # followed by a getBody (LINE 2)
>>
>> Just want to know why the behavior differs when nixevent is present???
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100201/13ce2f90/attachment-0002.html 

From rupa at rupa.com  Mon Feb  1 03:03:51 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Mon, 1 Feb 2010 05:03:51 -0600
Subject: [Freeswitch-users] Svar: Re: Somebody help me understand the
	'features' set up please :-)
In-Reply-To: <4B669ED3020000E100000447@mail.fribert.dk>
References: <4B66226C020000E10000043C@mail.fribert.dk>
	<b39273531001311645u618ea87am8a3d84ee6b1bcfcb@mail.gmail.com>
	<4B669ED3020000E100000447@mail.fribert.dk>
Message-ID: <b39273531002010303h6e66baf7tf77be8d67449122b@mail.gmail.com>

Look at line 758 in the default dialplan.  It shows how it is used.

On Mon, Feb 1, 2010 at 2:28 AM, mailinglist <mailinglist at fribert.dk> wrote:

>  Hi Rupa
>
> Ahh, I see.
> Looking at example 2, and
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bind_meta_app
> I still need to get where I enter this.
> Is it in the call handling in the dialplan?
> So I've gotten my current entry to handle incoming call from the outside:
>
> <extension name="PrivateIn">
>    <condition field="destination_number" expression="^(8202)$">
>        <action application="set" data="call_timeout=45"/>
>        <action application="set" data="domain_name=10.11.12.25"/>
>        <action application="set" data="continue_on_fail=true"/>
>        <action application="set" data="hangup_after_bridge=true"/>
>        <action application="set" data="transfer_ringback=$${hold_music}"/>
>        <action application="bridge" data="group/private@${domain_name}"/<group/private@$%7Bdomain_name%7D%22/>
> >
>        <action application="answer" data=""/>
>        <action application="sleep" data="1000"/>
>        <action application="voicemail" data="default 10.11.12.25 45"/>
>    </condition>
> </extension>
> Do I stick in there, or do I enter it in features.xml, or?
>
>
>  Best regards
> Fribse
>
> /Running Freeswitch on pfSense running on a VMWare ESXi 4.0
>
>
> >>> 01-02-2010 kl. 01:45 skrev Rupa Schomaker <rupa at rupa.com> i
> meddelelsen <b39273531001311645u618ea87am8a3d84ee6b1bcfcb at mail.gmail.com>:
>   Look at bind_meta_app in the default dialplan.  It binds the dtmf to
> the features context.
>
> On Sun, Jan 31, 2010 at 5:38 PM, mailinglist <mailinglist at fribert.dk>
> wrote:
> > Ok, I've gotten the Freeswitch to register to my VoIP provider.
> > I've gotten my phones to register to Freeswitch, and I can receive and
> make
> > calls, all very nice.
> >
> > I've gotten a Sipura SPA901 and a Siemens Gigaset S68IP registered to the
> > Freeswitch.
> >
> > When I receive a call, I would like to be able to transfer the call to
> > another phone, or change the call to a conference call with two local
> > phones.
> >
> > So I've been looking at the examples in the wiki, and I can't make them
> > work, not as I understand them anyways. Especially the att_xfer seems to
> be
> > able to do what I need.
> >
> > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_att_xfer
> >
> > As I understand Example1, I should answer the call, and then press *3
> during
> > the call, and either transfer it or change it to a threeway call.
> >
> > I get the first part, create an extension in the dialplan called
> att_xfer.
> > But what is meant by the second par 'then bind this feature to DTMF 3',
> how
> > do I enter that, and where?
> >
> > I hope somebody can help me with this (again)?
> >
> >
> >
> > Best regards
> > Fribse
> >
> > /Running Freeswitch on pfSense running on a VMWare ESXi 4.0
>
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> -Rupa
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100201/6030a10f/attachment-0002.html 

From moizchinoy at gmail.com  Mon Feb  1 03:08:29 2010
From: moizchinoy at gmail.com (Moiz Chinoy)
Date: Mon, 1 Feb 2010 15:08:29 +0400
Subject: [Freeswitch-users] Dialogic Board (JCT & DM3) Supported?
Message-ID: <29b888f81002010308uf875b53h838453b6e77e9a9e@mail.gmail.com>

Dear All,

Can anyone please advise that whether Dialogic boards (JCT and DM3)
are supported by FS.

-- 
Regards,
Moiz Chinoy.



From david.villasmil.work at gmail.com  Mon Feb  1 03:41:14 2010
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Mon, 1 Feb 2010 12:41:14 +0100
Subject: [Freeswitch-users] error loading module 'luasql.mysql'
In-Reply-To: <191c3a031001310621s20264d79u27afbf8ff0ba1a64@mail.gmail.com>
References: <eb7797d71001271615n2b86c197wc943d533fc0ffbe1@mail.gmail.com>
	<9853f4ff1001271709w60445c0ar7d3f4cb5fee36d4@mail.gmail.com>
	<eb7797d71001272047l48614934v3a35567f9fd0aa25@mail.gmail.com>
	<536A30FA-3494-40C2-9B9D-D8F63CA6BCC6@jerris.com>
	<4B657FDC.5080109@puzzled.xs4all.nl>
	<191c3a031001310621s20264d79u27afbf8ff0ba1a64@mail.gmail.com>
Message-ID: <9853f4ff1002010341w1aad5980h2cef90b541c21770@mail.gmail.com>

Hello Anthony,

Is this in planning?

David

On Sun, Jan 31, 2010 at 3:21 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> Be careful with lua and sql
> I have heard countless reports of the luasql leaking memory like a fire
> hydrant.....
>
> We may need to make our own odbc obj so every embedded lang can share it.
> But it takes time and resources.
>
> On Jan 31, 2010 7:11 AM, "Patrick" <freeswitch-list at puzzled.xs4all.nl>
> wrote:
>
> To fix a similar error message this is what I had in an old spec file:
> /sbin/restorecon -v /usr/lib64/somelib.so
>
> Iirc this is not the proper way to fix this and one should use the chcon
> command (chcon -t ...) or create an selinux policy. man chcon and google
> has more info.
>
> Regards,
> Patrick
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at l...
>
> On 01/31/2010 06:58 AM, Michael Jerris wrote:
>> http://www.google.com/search?q=cannot+restore+segmen...
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From sos at sokhapkin.dyndns.org  Mon Feb  1 03:58:21 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Mon, 1 Feb 2010 06:58:21 -0500
Subject: [Freeswitch-users] Equivalent to Asterisk's"directrtpsetup=yes"?
In-Reply-To: <b87dm5dn1qg1pkk8480064qjvc9j62jvj1@4ax.com>
References: <20100129164359.9B3C22B10D@cuneorg-email.cune.pri>
	<7FA9CDA6-EAB3-4C1D-9CD3-8D4B06B0F335@freeswitch.org>
	<b87dm5dn1qg1pkk8480064qjvc9j62jvj1@4ax.com>
Message-ID: <201002010658.21644.sos@sokhapkin.dyndns.org>

I'd say, you need bypass_media if provide SIP wholesale kind of service, and 
stay in audio path if you provide PBX-like kind of service.

On Monday 01 February 2010, Fred-145 wrote:
> On Sun, 31 Jan 2010 22:36:25 -0600, Brian West
>
> <brian at freeswitch.org> wrote:
> >Going back no_media after hold isn't supported yet.. Anthony said he would
> > add it if someone really really wanted it and posted a bounty of $500 to
> > cover his time to implement it.
>
> Thanks all for the info. The bottom line seems to be that it's really
> not a good idea to remove Freeswitch from the media path, and it's a
> better idea to look elsewhere if scalability is an issue.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org





From codecomplete at free.fr  Mon Feb  1 04:07:03 2010
From: codecomplete at free.fr (Fred-145)
Date: Mon, 01 Feb 2010 13:07:03 +0100
Subject: [Freeswitch-users] Equivalent to Asterisk's"directrtpsetup=yes"?
References: <20100129164359.9B3C22B10D@cuneorg-email.cune.pri>
	<7FA9CDA6-EAB3-4C1D-9CD3-8D4B06B0F335@freeswitch.org>
	<b87dm5dn1qg1pkk8480064qjvc9j62jvj1@4ax.com>
	<b87dm5dn1qg1pkk8480064qjvc9j62jvj1-e09XROE/p8c@public.gmane.org>
	<201002010658.21644.sos@sokhapkin.dyndns.org>
Message-ID: <ntgdm5lpgjal4psaq7dg5p9f6udqqsleg4@4ax.com>

On Mon, 1 Feb 2010 06:58:21 -0500, Sergey Okhapkin
<sos at sokhapkin.dyndns.org> wrote:
>I'd say, you need bypass_media if provide SIP wholesale kind of service, and 
>stay in audio path if you provide PBX-like kind of service.

Yup, that's a better summary. Thanks all for the information. Things
make a lot more sense now :-)




From stevendt at primrosebank.net  Mon Feb  1 05:27:13 2010
From: stevendt at primrosebank.net (Dave Stevenson)
Date: Mon, 1 Feb 2010 13:27:13 -0000
Subject: [Freeswitch-users] Trunk Version Number
References: <2360B060EDF44D368707DD12064BD416@bp1.ad.bp.com>
	<8976E2C4-CF8C-433A-AC79-B58F65105D08@jerris.com>
Message-ID: <12301A128C654024832FC6AA8CE43F31@bp1.ad.bp.com>

Hi Michael,

thanks for the reply. Yes, I have built from an SVN checkout (using Tortoise SVN).

I did not quite understand what is required to fix things though ?

regards
Dave
  ----- Original Message ----- 
  From: Michael Jerris 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Monday, February 01, 2010 6:11 AM
  Subject: Re: [Freeswitch-users] Trunk Version Number


  it should.  This can happen if you build from an svn checkout and the svn client your using is newer than our static linked svnversion.exe.  If anyone can make me a newer stripped down version like that I would appreciate it I have not had the time.


  On Jan 31, 2010, at 9:30 AM, Dave Stevenson wrote:


    Hi,

    Running the latest SVN (16453) under Windows, the console "Version" command displays :-

    "FreeSWITCH Version 1.0.trunk (UNKNOWN)"

    Should the version number not include a meaningful build version in the brackets ?

    regards
    Dave
    _______________________________________________
    FreeSWITCH-users mailing list
    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    http://www.freeswitch.org





------------------------------------------------------------------------------


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100201/02c440d3/attachment-0002.html 

From mailinglist at fribert.dk  Mon Feb  1 05:37:40 2010
From: mailinglist at fribert.dk (mailinglist)
Date: Mon, 01 Feb 2010 14:37:40 +0100
Subject: [Freeswitch-users] Svar: Re: Somebody help me understand	 the
	'features' set up please :-)
In-Reply-To: <b39273531002010303h6e66baf7tf77be8d67449122b@mail.gmail.com>
References: <4B66226C020000E10000043C@mail.fribert.dk>
	<b39273531001311645u618ea87am8a3d84ee6b1bcfcb@mail.gmail.com>
	<4B669ED3020000E100000447@mail.fribert.dk>
	<b39273531002010303h6e66baf7tf77be8d67449122b@mail.gmail.com>
Message-ID: <4B66E734020000E100000451@mail.fribert.dk>

Hmm, I've just downloaded the default.xml under conf/dialplan from the SVN just to be on the safe side.
Line 758 is the last </include>, but I did find some examples on line 249-251.
 
So I've changed my dialplan entry handling calls from the outside to this:

<extension name="PrivateIn">
   <condition field="destination_number" expression="^(8202)$">
       <action application="bind_meta_app" data="1 b s execute_extension::att_xfer XML features"/>
       <action application="bind_meta_app" data="2 b s record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
       <action application="set" data="call_timeout=45"/>
       <action application="set" data="domain_name=10.11.12.25"/>
       <action application="set" data="continue_on_fail=true"/>
       <action application="set" data="hangup_after_bridge=true"/>
       <action application="set" data="transfer_ringback=$${hold_music}"/>
       <action application="bridge" data="group/private@${domain_name}"/ ( 'mailto:group/private@$domain_name"/' )>
       <action application="answer" data=""/>
       <action application="sleep" data="1000"/>
       <action application="voicemail" data="default 10.11.12.25 45"/>
   </condition>
</extension>
As I understand the bind_meta_app it listens for *1 and then it runs the att_xfer, *2 to record the call.
I've included the att_xfer in the XML features.
 
Question is, will it work at all when I bridge to a group?
 
Nothing happens when I press *1 and an extension.
 
 
 
 
Best regards
Fribse
 
/Running Freeswitch on pfSense running on a VMWare ESXi 4.0

>>> 01-02-2010 kl. 12:03 skrev Rupa Schomaker <rupa at rupa.com> i meddelelsen <b39273531002010303h6e66baf7tf77be8d67449122b at mail.gmail.com>:

Look at line 758 in the default dialplan. It shows how it is used.

On Mon, Feb 1, 2010 at 2:28 AM, mailinglist <mailinglist at fribert.dk> wrote:


Hi Rupa
Ahh, I see.
Looking at example 2, and http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bind_meta_app
I still need to get where I enter this.
Is it in the call handling in the dialplan?
So I've gotten my current entry to handle incoming call from the outside:
<extension name="PrivateIn">
<condition field="destination_number" expression="^(8202)$">
<action application="set" data="call_timeout=45"/>
<action application="set" data="domain_name=10.11.12.25"/>
<action application="set" data="continue_on_fail=true"/>
<action application="set" data="hangup_after_bridge=true"/>
<action application="set" data="transfer_ringback=$${hold_music}"/>
<action application="bridge" data="group/private@${domain_name}"/ ( mailto:group/private@$%7Bdomain_name%7D%22/ )>
<action application="answer" data=""/>
<action application="sleep" data="1000"/>
<action application="voicemail" data="default 10.11.12.25 45"/>
</condition>
</extension>
Do I stick in there, or do I enter it in features.xml, or?
Best regards
Fribse
/Running Freeswitch on pfSense running on a VMWare ESXi 4.0


>>> 01-02-2010 kl. 01:45 skrev Rupa Schomaker <rupa at rupa.com> i meddelelsen <b39273531001311645u618ea87am8a3d84ee6b1bcfcb at mail.gmail.com>:

Look at bind_meta_app in the default dialplan. It binds the dtmf to
the features context.

On Sun, Jan 31, 2010 at 5:38 PM, mailinglist <mailinglist at fribert.dk> wrote:
> Ok, I've gotten the Freeswitch to register to my VoIP provider.
> I've gotten my phones to register to Freeswitch, and I can receive and make
> calls, all very nice.
>
> I've gotten a Sipura SPA901 and a Siemens Gigaset S68IP registered to the
> Freeswitch.
>
> When I receive a call, I would like to be able to transfer the call to
> another phone, or change the call to a conference call with two local
> phones.
>
> So I've been looking at the examples in the wiki, and I can't make them
> work, not as I understand them anyways. Especially the att_xfer seems to be
> able to do what I need.
>
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_att_xfer
>
> As I understand Example1, I should answer the call, and then press *3 during
> the call, and either transfer it or change it to a threeway call.
>
> I get the first part, create an extension in the dialplan called att_xfer.
> But what is meant by the second par 'then bind this feature to DTMF 3', how
> do I enter that, and where?
>
> I hope somebody can help me with this (again)?
>
>
>
> Best regards
> Fribse
>
> /Running Freeswitch on pfSense running on a VMWare ESXi 4.0
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
-Rupa

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org





-- 
-Rupa
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100201/d8c1bc6a/attachment-0002.html 

From frank at carmickle.com  Mon Feb  1 07:29:16 2010
From: frank at carmickle.com (Frank Carmickle)
Date: Mon, 1 Feb 2010 10:29:16 -0500
Subject: [Freeswitch-users] latency growing with conference
Message-ID: <20100201152916.GD27405@base.carmickle.com>

Hello

I am noticing that when short phrases are spoken with cepstral in a conference it pauses all the audio before speaking it.  It seems as though the pauses are of different lengths and over time this adds up to a great amount of latency for the conference members to each other.  Rejoining the conference fixes this.  I already have 

      <param name="member-flags" value="waste"/>

I also notice that one member, who has lots of dropped packets, becomes very latent also.  I have

        <action application="set" data="rtp_autoflush=true"/>

Any thing else I can do about the two similar yet different issues?  I am on 16431 at the moment.

Thanks
--FC




From brian at freeswitch.org  Mon Feb  1 07:30:32 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 1 Feb 2010 09:30:32 -0600
Subject: [Freeswitch-users] Wrong RTP port submitted?
In-Reply-To: <4B6011EF.6090706@gmx.net>
References: <4B6011EF.6090706@gmx.net>
Message-ID: <2EBC2DDA-1799-44B5-9D1B-EE6EC1618482@freeswitch.org>

You have proxy media on don't you? From what this looks like you have an inbound invite from a Snom and we have an outbound invite.  If you're doing something such as proxy media it will try to pass it thru as is...  Giving you little or NO control over the port.  Unless you fix your endpoint to also use the same restrictive port range.  Because by default we will not use anything in the 48000 range .. the giveaway here is the fact I see P-Key-Flags: header which is indication you're using a snom with proxy media.  Am I correct?

/b

On Jan 27, 2010, at 4:14 AM, Peter P GMX wrote:

> I have defined the rtp port range for 12000-12100 in switch.conf.xml.
> However Freeswitch is offering a port 48320 in the invite message. The
> result is, that the incoming RTP stream is blocked by the firewall (I
> can see a reject for UDP 48320).
> Any hint how to solve this?
> 
> Best regards Peter
> 
> See config and invite message:
>    <!--RTP port range -->
>    <param name="rtp-start-port" value="12000"/>-->
>    <param name="rtp-end-port" value="12100"/>-->
> 
> Invite:
> INVITE sip:027xxxxxxxx at sip.itsp.de SIP/2.0.
> Via: SIP/2.0/UDP 217.24.xx.xxx:5080;rport;branch=z9hG4bKjD923NvctXaFm.
> Max-Forwards: 69.
> From: "0608xxxxxxx"
> <sip:4918xxxxxxxxxxxx at sip.itsp.de;transport=udp>;tag=0Kp4tvU44UmXp.
> To: <sip:027xxxxxxxx at sip.itsp.de>.
> Call-ID: 30c86b94-85ca-122d-f88e-080027e51f59.
> CSeq: 126174137 INVITE.
> Contact: <sip:gw+tgnet3 at 217.24.xx.xxx:5080;transport=udp;gw=tgnet3>.
> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16032.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY.
> Supported: timer, precondition, path, replaces.
> Allow-Events: talk, refer.
> Content-Type: application/sdp.
> Content-Disposition: session.
> Content-Length: 320.
> P-Key-Flags: keys="3".
> X-FS-Support: update_display.
> Remote-Party-ID: "0608xxxxxxx"
> <sip:0608xxxxxxx at sip.itsp.de>;party=calling;screen=yes;privacy=off.
> .
> v=0.
> o=FreeSWITCH 1264536651 1264536652 IN IP4 217.24.xx.xxx.
> s=FreeSWITCH.
> c=IN IP4 217.24.xx.xxx.
> t=0 0.
> m=audio 48320 RTP/AVP 8 0 98 3 101 13.
> a=rtpmap:8 PCMA/8000.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:98 SPEEX/8000.
> a=rtpmap:3 GSM/8000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-16.
> a=rtpmap:13 CN/8000.
> a=ptime:20.
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From pjintheusa at gmail.com  Mon Feb  1 07:37:52 2010
From: pjintheusa at gmail.com (Phillip Jones)
Date: Mon, 1 Feb 2010 10:37:52 -0500
Subject: [Freeswitch-users] Establishing a Call from .Net based
	application
In-Reply-To: <df97b9c51001310545n54195cadr9faa3db46917ca9e@mail.gmail.com>
References: <874941.17255.qm@web33502.mail.mud.yahoo.com>
	<df97b9c51001310545n54195cadr9faa3db46917ca9e@mail.gmail.com>
Message-ID: <367751821002010737o4ca83476j62c96cb60c8a456a@mail.gmail.com>

As mentioned http://wiki.freeswitch.org/wiki/Mod_managed should give you
every thing you need to get mod_managed set up.

Then in the source take a look at demo.csx and particularity AppDemo class.

That should get you started.


On Sun, Jan 31, 2010 at 8:45 AM, Scott Fernandez <scottferri09 at gmail.com>wrote:

> Hi,
>
> Thx for the information. Can I have some detailed steps to configure
> mod_managed class call control and how do we write the API commands in .Net
> applications?
>
> In addition, how do we get the current STATE of the call when I use
> webapi?. Because it is required for me to route the call to the user upon it
> is answered or disconnect  it.
>
> Thanks,
> Scott
>
> On Wed, Jan 20, 2010 at 8:47 PM, Diego Toro <dftoro at yahoo.com> wrote:
>
>> Hi, the answer is yes, you can to use mod_managed wich offer C# managed
>> class to call control http://wiki.freeswitch.org/wiki/Mod_managed. Or
>> using managed ESL (libs/esl/managed) which offer C# managed class to receive
>> and send events and commands to FreeSwitch.
>>
>> Diego Toro
>> http://lacarretade.blogspot.com/
>>
>>
>> --- On Wed, 1/20/10, Scott Fernandez <scottferri09 at gmail.com> wrote:
>>
>> > From: Scott Fernandez <scottferri09 at gmail.com>
>> > Subject: Re: [Freeswitch-users] Establishing a Call from .Net based
>> application
>> > To: freeswitch-users at lists.freeswitch.org
>> > Date: Wednesday, January 20, 2010, 2:17 AM
>>  > Thanks Dome. Will try it out and get back to
>> > you if I come across any issues.
>> >
>> > Regards,
>> > Scott.
>> >
>> > On Wed, Jan 20, 2010 at 11:02 AM,
>> > Dome Charoenyost <dome at tel.co.th>
>> > wrote:
>> >
>> > Please try http://wiki.freeswitch.org/wiki/Webapi
>> >
>> >
>> > you can create class and map to webapi.
>> >
>> >
>> >
>> > Dome C.
>> >
>> >
>> >
>> > 2010/1/19 Scott Fernandez <scottferri09 at gmail.com>:
>> >
>> > > Hi,
>> >
>> > >
>> >
>> > > Is there any API modules available for me to initiate
>> > a call from .Net based
>> >
>> > > application?.
>> >
>> > >
>> >
>> > > The idea is to include the API modules if any with the
>> > .NET base classes so
>> >
>> > > that the API commands will be made available on it. I
>> > know it is doable when
>> >
>> > > I use socket programming in .NET in which Telnet
>> > session is created.
>> >
>> > > However, this would potentially hamper the performance
>> > of the application
>> >
>> > > because of multiple sessions that will be created for
>> > each call.
>> >
>> > >
>> >
>> > > Other than that, Is there any Freeswitch API modules
>> > (like plug-ins)
>> >
>> > > available in order to include it into the .Net classes
>> > and start building
>> >
>> > > the customized application?
>> >
>> > >
>> >
>> > > Any help from any one is highly appreciated.
>> >
>> > >
>> >
>> > > Thanks,
>> >
>> > > Scott
>> >
>> > >
>> >
>> > >
>> > _______________________________________________
>> >
>> > > FreeSWITCH-users mailing list
>> >
>> > > FreeSWITCH-users at lists.freeswitch.org
>> >
>> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >
>> > > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >
>> > > http://www.freeswitch.org
>> >
>> > >
>> >
>> > >
>> >
>> >
>> >
>> > _______________________________________________
>> >
>> > FreeSWITCH-users mailing list
>> >
>> > FreeSWITCH-users at lists.freeswitch.org
>> >
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >
>> > http://www.freeswitch.org
>> >
>> >
>> >
>> >
>> > -----Inline Attachment Follows-----
>>  >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>>
>>
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100201/fba1c5e5/attachment-0002.html 

From frank at carmickle.com  Mon Feb  1 07:44:22 2010
From: frank at carmickle.com (Frank Carmickle)
Date: Mon, 1 Feb 2010 10:44:22 -0500
Subject: [Freeswitch-users] conference and xmpp
Message-ID: <20100201154421.GE27405@base.carmickle.com>

Hello

I am trying to interact with conferences over xmpp.  I want to be able to see status changes and mute/unmute etc...  I can only list the members if I ask for a list.  I am having trouble understanding the wiki entree on this topic.  Profile names are confusing because all the examples show default as a name so it is hard to see what each profile name corresponds to.  My config looks like this currently.

   <profile name="ourchatperms">
    <!-- both of these users have a functionally equivalent capability set -->
    <user name="shawn at djernes.net" commands="all"/>
    <!-- individually specified commands must be book-end'ed with the | charachter -->
    <user name="frankiec at djernes.net" commands="|deaf|dial|energy|kick|list|lock|mute|norecord|play|record|relate|say|saymember|stop|transfer|undeaf|unlock|unmute|volume_in|volume_out|"/>
   </profile>
  </chat-permissions>
 
    <profile name="cdquality">
<param name="caller-controls" value="custom-ewr"/>
      <param name="chat-permissions" value="ourchatperms"/>
      <param name="domain" value="$${domain}"/>

Also we can only talk to the conference by conf+confnum-fqdn at fqdn.  Is there a way to make it conf+confnum at fqdn?

Any help would be greatly appreciated.  Thank you.
--FC



From mouncifbb at gmail.com  Mon Feb  1 07:47:22 2010
From: mouncifbb at gmail.com (Mouncif Benniane)
Date: Mon, 1 Feb 2010 10:47:22 -0500
Subject: [Freeswitch-users] mod_lcr problem
In-Reply-To: <b39273531001311507hb1bfe02m8d370443b882ef81@mail.gmail.com>
References: <ade741d61001291443l6134a3achf28be64fc7746c8e@mail.gmail.com>
	<b39273531001300702r1a81f094pac0bd20aeb5ceed6@mail.gmail.com>
	<ade741d61001301338x8014e86t9c9094403f3361f@mail.gmail.com>
	<b39273531001301559h60668746kd5c63ada240513f1@mail.gmail.com>
	<ade741d61001301623w70d7e449i44a9f1e9fb7beae4@mail.gmail.com>
	<b39273531001301645w10446308o3578479dcbf2da2@mail.gmail.com>
	<ade741d61001302057w6ce72355iee6e8b4a2872fc81@mail.gmail.com>
	<b39273531001310532y1de53609rf474755cec04f218@mail.gmail.com>
	<9AB8B620-5A69-4F08-B62C-FFEB03FC6762@gmail.com>
	<b39273531001311507hb1bfe02m8d370443b882ef81@mail.gmail.com>
Message-ID: <ade741d61002010747v6da3d2e1x554b46aa26c27b18@mail.gmail.com>

any example on how to use: set_profile_var?

thanks


On Sun, Jan 31, 2010 at 6:07 PM, Rupa Schomaker <rupa at rupa.com> wrote:

> Yes, you need to normalize the values passed to lcr.  Otherwise, how could
> it work?
>
> You can normalize the CID by matching and adding a 1 for 10 digit #s, or
> removing the leading + or other things you might need then setting it back
> to the profile using the set_profile_var app (
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_set_profile_var).
>  (mod_cidlookup will set it after doing a #->name/area lookup - but for now
> you can set it yourself)
>
> You can normalize the DID by doing similar matching rules as above and then
> transfering to that normalized DID for the rest of your call plan
> processing.
>
> I'm pretty sure mod_cidlookup has an example of normalizing... yeah:
> http://wiki.freeswitch.org/wiki/Mod_cidlookup#Dialplan_Application
>
> On Sun, Jan 31, 2010 at 4:18 PM, Mouncifbb <mouncifbb at gmail.com> wrote:
>
>> So the CID must have 1 at front also? Usually people
>> Send only npa and nxx ex 6176427788 7817612233
>> Do I need to alter it?
>>
>> Sent from my iPhone
>>
>> On Jan 31, 2010, at 8:32 AM, Rupa Schomaker <rupa at rupa.com> wrote:
>>
>>
>>
>> On Sat, Jan 30, 2010 at 10:57 PM, Mouncif Benniane <<mouncifbb at gmail.com>
>> mouncifbb at gmail.com> wrote:
>>
>>> OK going back to use default profile to keep things simple below 2
>>> results
>>>
>>> Using:
>>>
>>> lcr 16179470890  default  19785223241  ( this one consult
>>> npa_nxx_company_ocn)
>>>
>>> lcr 6179470890  default  9785223241 ( this one don't!! )
>>>
>>>
>>>
>> Oh, right!  mod_lcr really expects you to normalize your prefix to e164
>> format.  I thought there was discussion about this in the wiki, but maybe
>> not.  For simple prefix matching it doesn't matter, but for things that make
>> decisions based on the # (like the lata/state stuff) it does.
>>
>> npanxx lookup only makes sense for NANPA numbers.  NANPA numbers have a
>> country code of "1" and a total length of 11 (including the 1).
>>
>> This is the only rational way to do it when you have a rate table with
>> both domestic (NANPA) and international prefixes.
>>
>>
>>> freeswitch> lcr 16179470890  default  19785223241
>>> 2010-01-30 23:53:34.681842 [DEBUG] mod_lcr.c:1329 data passed to lcr is
>>> [16179470890  default  19785223241]
>>> 2010-01-30 23:53:34.681842 [DEBUG] mod_lcr.c:1365 Set Caller ID to
>>> [19785223241]
>>> 2010-01-30 23:53:34.681842 [DEBUG] mod_lcr.c:736 SQL: SELECT 'state',
>>> count(DISTINCT state) FROM npa_nxx_company_ocn WHERE (npa=617 AND nxx=947)
>>> OR (npa=978 AND nxx=522) UNION SELECT 'lata', count(DISTINCT lata) FROM
>>> npa_nxx_company_ocn WHERE (npa=617 AND nxx=947) OR (npa=978 AND nxx=522)
>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:696 Type: state, Count: 1
>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:696 Type: lata, Count: 1
>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:786 intra routing [state:1
>>> lata:1] so rate field is [intralata_rate]
>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:802 we have an event
>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:826 SQL: SELECT l.digits,
>>> c.carrier_name, l.intralata_rate, cg.prefix AS gw_prefix, cg.suffix AS
>>> gw_suffix, l.lead_strip, l.trail_strip, l.prefix, l.suffix , cg.codec ,
>>> l.cid FROM lcr l JOIN carriers c ON l.carrier_id=c.id JOIN
>>> carrier_gateway cg ON c.id=cg.carrier_id WHERE c.enabled = '1' AND
>>> cg.enabled = '1' AND l.enabled = '1' AND digits IN (16179470890, 1617947089,
>>> 161794708, 16179470, 1617947, 161794, 16179, 1617, 161, 16, 1) AND
>>> CURRENT_TIMESTAMP BETWEEN date_start AND date_end ORDER BY digits DESC,
>>> intralata_rate, rand();
>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:307 Returning Dialstring
>>> [lcr_carrier=carrier1,lcr_rate=0.00000]sofia/gateway/carrier1/16179470890
>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:615 Adding carrier1 to head
>>> of list
>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:307 Returning Dialstring
>>> [lcr_carrier=carrier2,lcr_rate=0.00000]sofia/external/
>>> 06179470890 at proxy.carrier2.net:5060
>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:667 adding carrier2 to end
>>> of list after carrier1
>>>
>>>
>>>  | Digit Match | Carrier  | Rate     | Codec | CID Regexp |
>>> Dialstring
>>> |
>>>  | 1           | carrier1 | 0.00000  |       |            |
>>> [lcr_carrier=carrier1,lcr_rate=0.00000]sofia/gateway/carrier1/16179470890
>>> |
>>>  | 1           | carrier2 | 0.00000  |       |            |
>>> [lcr_carrier=carrier2,lcr_rate=0.00000]sofia/external/
>>> 06179470890 at proxy.carrier2.net:5060 |
>>>
>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:307 Returning Dialstring
>>> [lcr_carrier=carrier1,lcr_rate=0.00000]sofia/gateway/carrier1/16179470890
>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:307 Returning Dialstring
>>> [lcr_carrier=carrier2,lcr_rate=0.00000]sofia/external/
>>> 06179470890 at proxy.carrier2.net:5060
>>>
>>>
>>>
>>>
>>>
>>> freeswitch> lcr 6179470890  default  9785223241
>>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:1329 data passed to lcr is
>>> [6179470890  default  9785223241]
>>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:1365 Set Caller ID to
>>> [9785223241]
>>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:786 intra routing [state:0
>>> lata:0] so rate field is [rate]
>>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:802 we have an event
>>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:826 SQL: SELECT l.digits,
>>> c.carrier_name, l.rate, cg.prefix AS gw_prefix, cg.suffix AS gw_suffix,
>>> l.lead_strip, l.trail_strip, l.prefix, l.suffix , cg.codec , l.cid FROM lcr
>>> l JOIN carriers c ON l.carrier_id=c.id JOIN carrier_gateway cg ON c.id=cg.carrier_id
>>> WHERE c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1' AND digits IN
>>> (6179470890, 617947089, 61794708, 6179470, 617947, 61794, 6179, 617, 61, 6)
>>> AND CURRENT_TIMESTAMP BETWEEN date_start AND date_end ORDER BY digits DESC,
>>> rate, rand();
>>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:307 Returning Dialstring
>>> [lcr_carrier=carrier1,lcr_rate=0.09000]sofia/gateway/carrier1/16179470890
>>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:615 Adding carrier1 to head
>>> of list
>>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:307 Returning Dialstring
>>> [lcr_carrier=carrier1,lcr_rate=0.09000]sofia/gateway/carrier1/16179470890
>>>
>>>
>>>  | Digit Match | Carrier  | Rate     | Codec | CID Regexp |
>>> Dialstring                                                                |
>>>   | 617947      | carrier1 | 0.09000  |       |            |
>>> [lcr_carrier=carrier1,lcr_rate=0.09000]sofia/gateway/carrier1/16179470890 |
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>> On Sat, Jan 30, 2010 at 7:45 PM, Rupa Schomaker < <rupa at rupa.com>
>>> rupa at rupa.com> wrote:
>>>
>>>> turn up logging to debug again, and then reload mod_lcr.  It'll spit out
>>>> a bunch of crap when it tests out each profile you have defined.  Give me
>>>> the full log (here or in <http://pastebin.freeswitch.org>
>>>> pastebin.freeswitch.org).  That may show more useful info as to why
>>>> things are mucked up?
>>>>
>>>>
>>>> On Sat, Jan 30, 2010 at 6:23 PM, Mouncif Benniane <<mouncifbb at gmail.com>
>>>> mouncifbb at gmail.com> wrote:
>>>>
>>>>> yes I use NANPA_STD profile instead of default cause I thought the
>>>>> custom profile was causing issues, but looks like it's returning same
>>>>> results.
>>>>>
>>>>> There is this line in thw wiki:
>>>>> intra lata/state selection is done manually by setting the channel
>>>>> variables *intrastate* or *intralata* to the value *true*.
>>>>>
>>>>> do I have to set these ? if yes how?
>>>>>
>>>>> Thanks
>>>>>
>>>>>
>>>>> On Sat, Jan 30, 2010 at 6:59 PM, Rupa Schomaker < <rupa at rupa.com>
>>>>> rupa at rupa.com> wrote:
>>>>>
>>>>>> Stuff inline.
>>>>>>
>>>>>> On Sat, Jan 30, 2010 at 3:38 PM, Mouncif Benniane <<mouncifbb at gmail.com>
>>>>>> mouncifbb at gmail.com> wrote:
>>>>>>
>>>>>>>  NPANXX,"LATA","OCN","NTER","INTRA"
>>>>>>> 201007,"224","7229","0.0059","0.0127"
>>>>>>> 201040,"224","9206","0.0036","0.0036"
>>>>>>>
>>>>>>
>>>>>> Looks like they give you the LATA and OCN values with the prefix.  We
>>>>>> (should) look that up ourselves.
>>>>>>
>>>>>>
>>>>>>> FreeSWITCH Version 1.0.trunk (16540)
>>>>>>>
>>>>>>>
>>>>>>> Also I noticed the *npa_nxx_ocn* table never get consulted.
>>>>>>>
>>>>>>> I also see this now when making a real call instead of running
>>>>>>> thorugh CLI
>>>>>>>
>>>>>>> EXECUTE sofia/external/6179472456 at 174.x.x.x lcr(16179470890
>>>>>>> NANPA_STD)
>>>>>>> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:1230 *intrastate
>>>>>>> channel var is [undef]*
>>>>>>
>>>>>>
>>>>>> This is fine.  it is a leftover from when you would tell mod_lcr via a
>>>>>> channel var that it should do intrastate.  I later had mod_lcr do the lookup
>>>>>> itself, but we still honor the old var.  There are no channel vars
>>>>>> associated with the cli, so you wouldn't see that msg.
>>>>>>
>>>>>>
>>>>>>>
>>>>>>> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:1233 Select routes based
>>>>>>> on interstate rates
>>>>>>> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:1252 LCR Lookup on
>>>>>>> 16179470893 using profile NANPA_STD
>>>>>>> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:786 intra routing
>>>>>>> [state:0 lata:0] so rate field is [rate]
>>>>>>>
>>>>>>> called number 6179470890  caller ID: 6179472456
>>>>>>>
>>>>>>> any ideas??
>>>>>>>
>>>>>>>
>>>>>> Only thing that jumps out at me.
>>>>>>
>>>>>> The output from lcr_admin show profiles showed only the default one.
>>>>>>  On the dialplan you use the NANPA_STD profile.  Can you check lcr_admin
>>>>>> list and see if that profile is defined and if so if it says it is using the
>>>>>> npanxx table?
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> On Sat, Jan 30, 2010 at 10:02 AM, Rupa Schomaker < <rupa at rupa.com>
>>>>>>> rupa at rupa.com> wrote:
>>>>>>>
>>>>>>>> Something is still missing from the logs. Note the query of the
>>>>>>>> npanxx
>>>>>>>> table, the flags being set, and the rate field being chosen.  Umm..
>>>>>>>> oh, what version of fs are you running?
>>>>>>>>
>>>>>>>> Yes, the npa_nxx_ocn table needs to be loaded up as described in:
>>>>>>>> <http://wiki.freeswitch.org/wiki/Mod_cidlookup#Falling_back_to_.22City_State.22_in_the_absense_of_a_name>
>>>>>>>> http://wiki.freeswitch.org/wiki/Mod_cidlookup#Falling_back_to_.22City_State.22_in_the_absense_of_a_name
>>>>>>>> (there is a link to that from mod_lcr's wiki page).
>>>>>>>>
>>>>>>>> An example from my own setup:
>>>>>>>>
>>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:1384 data passed to lcr
>>>>>>>> is [12148267711 default 12148267712]
>>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:1420 Set Caller ID to
>>>>>>>> [12148267712]
>>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:759 SQL: SELECT
>>>>>>>> 'state',
>>>>>>>> count(DISTINCT state) FROM npa_nxx_company_ocn WHERE (npa=214 AND
>>>>>>>> nxx=826) OR (npa=214 AND nxx=826) UNION SELECT 'lata',
>>>>>>>> count(DISTINCT
>>>>>>>> lata) FROM npa_nxx_company_ocn WHERE (npa=214 AND nxx=826) OR
>>>>>>>> (npa=214
>>>>>>>> AND nxx=826)
>>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:719 Type: lata, Count:
>>>>>>>> 1
>>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:719 Type: state, Count:
>>>>>>>> 1
>>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:809 intra routing
>>>>>>>> [state:1 lata:1] so rate field is [intralata_rate]
>>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:825 we have an event
>>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:849 SQL:  SELECT
>>>>>>>> l.digits
>>>>>>>> AS lcr_digits,         c.carrier_name AS lcr_carrier_name,
>>>>>>>> l.intralata_rate as lcr_rate_field,        cg.prefix AS
>>>>>>>> lcr_gw_prefix,
>>>>>>>> cg.suffix AS lcr_gw_suffix,        l.lead_strip AS lcr_lead_strip,
>>>>>>>> l.trail_strip AS lcr_trail_strip,        l.prefix AS lcr_prefix,
>>>>>>>> l.suffix AS lcr_suffix,        cg.codec AS lcr_codec, l.cid AS
>>>>>>>> lcr_cid,        'carriers' AS lcr_limit_realm, c.carrier_name AS
>>>>>>>> lcr_limit_id, 5 AS lcr_limit_max   FROM lcr l      JOIN carriers c
>>>>>>>> ON
>>>>>>>> l.carrier_id=c.id      JOIN carrier_gateway cg ON c.id
>>>>>>>> =cg.carrier_id
>>>>>>>>  WHERE c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1'
>>>>>>>> AND digits_prefix @> '12148267711'      AND CURRENT_TIMESTAMP
>>>>>>>> BETWEEN
>>>>>>>> date_start AND date_end   ORDER BY digits DESC, intralata_rate,
>>>>>>>> random();
>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning
>>>>>>>> Dialstring
>>>>>>>>
>>>>>>>> [lcr_carrier=grnvoip_us,lcr_rate=0.00591]sofia/gateway/grnvoip/XXXX12148267711
>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:638 Adding grnvoip_us
>>>>>>>> to
>>>>>>>> head of list
>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning
>>>>>>>> Dialstring
>>>>>>>>
>>>>>>>> [lcr_carrier=teliax,lcr_rate=0.01000]sofia/gateway/teliax_atl/12148267711
>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:660 Adding teliax to
>>>>>>>> end of list
>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning
>>>>>>>> Dialstring
>>>>>>>>
>>>>>>>> [lcr_carrier=teliax,lcr_rate=0.01000]sofia/gateway/teliax/12148267711
>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:660 Adding teliax to
>>>>>>>> end of list
>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning
>>>>>>>> Dialstring
>>>>>>>>
>>>>>>>> [lcr_carrier=vitelity,lcr_rate=0.01440]sofia/gateway/vitelity/12148267711
>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:660 Adding vitelity to
>>>>>>>> end of list
>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning
>>>>>>>> Dialstring
>>>>>>>>
>>>>>>>> [lcr_carrier=grnvoip_std,lcr_rate=0.01500]sofia/gateway/grnvoip/YYYY12148267711
>>>>>>>> [...]
>>>>>>>>
>>>>>>>> On Fri, Jan 29, 2010 at 10:42 PM, Mouncif Benniane <<mouncifbb at gmail.com>
>>>>>>>> mouncifbb at gmail.com> wrote:
>>>>>>>> > Also the Provider has presented the rates in this format?
>>>>>>>> > NPANXXLATA OCN INTER INTRA
>>>>>>>> >
>>>>>>>> >
>>>>>>>> > On Fri, Jan 29, 2010 at 11:30 PM, Mouncif Benniane <<mouncifbb at gmail.com>
>>>>>>>> mouncifbb at gmail.com>
>>>>>>>> > wrote:
>>>>>>>> >>
>>>>>>>> >> Tried it and it's not giving me intralata  instead I get
>>>>>>>> interstate, does
>>>>>>>> >> the npa_nxx_company_ocn table needs to be used in this case?,
>>>>>>>> also do I have
>>>>>>>> >> to have the rate field in lcr table?
>>>>>>>> >>
>>>>>>>> >> lcr 617642  default 6176421212
>>>>>>>> >>  | Digit Match | Carrier  | Rate     | Codec | CID Regexp |
>>>>>>>> Dialstring
>>>>>>>> >>                                                        |
>>>>>>>> >>  | 617642      | carrier1 | 0.00500  |       |            |
>>>>>>>> >>
>>>>>>>> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642 |
>>>>>>>> >>
>>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:1329 data passed to
>>>>>>>> lcr is
>>>>>>>> >> [617642  default 6176421212]
>>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:1365 Set Caller ID
>>>>>>>> to
>>>>>>>> >> [6176421212]
>>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:786 intra routing
>>>>>>>> [state:0
>>>>>>>> >> lata:0] so rate field is [rate]
>>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:802 we have an event
>>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:826 SQL: SELECT
>>>>>>>> l.digits,
>>>>>>>> >> c.carrier_name, l.rate, cg.prefix AS gw_prefix, cg.suffix AS
>>>>>>>> gw_suffix,
>>>>>>>> >> l.lead_strip, l.trail_strip, l.prefix, l.suffix , cg.codec ,
>>>>>>>> l.cid FROM lcr
>>>>>>>> >> l JOIN carriers c ON l.carrier_id=c.id JOIN carrier_gateway cg
>>>>>>>> ON
>>>>>>>> >> c.id=cg.carrier_id WHERE c.enabled = '1' AND cg.enabled = '1'
>>>>>>>> AND l.enabled
>>>>>>>> >> = '1' AND digits IN (617642, 61764, 6176, 617, 61, 6) AND
>>>>>>>> CURRENT_TIMESTAMP
>>>>>>>> >> BETWEEN date_start AND date_end ORDER BY digits DESC,  rate,
>>>>>>>> rand();
>>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:307 Returning
>>>>>>>> Dialstring
>>>>>>>> >>
>>>>>>>> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642
>>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:615 Adding carrier1
>>>>>>>> to head
>>>>>>>> >> of list
>>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:307 Returning
>>>>>>>> Dialstring
>>>>>>>> >>
>>>>>>>> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642
>>>>>>>> >>
>>>>>>>> >> Thank you Rupa!
>>>>>>>> >>
>>>>>>>> >> On Fri, Jan 29, 2010 at 7:37 PM, Rupa Schomaker <<rupa at rupa.com>
>>>>>>>> rupa at rupa.com> wrote:
>>>>>>>> >>>
>>>>>>>> >>> turn console logging up to debug and redo the lcr lookup.  The
>>>>>>>> sql
>>>>>>>> >>> statements along with status info will show up.  This should
>>>>>>>> give
>>>>>>>> >>> enough information to debug what is happening.
>>>>>>>> >>>
>>>>>>>> >>> I'm assuming the npanxx table is actually populated and not just
>>>>>>>> >>> existing?
>>>>>>>> >>>
>>>>>>>> >>> When doing the lookup from the cli you have to tell lcr what CID
>>>>>>>> to
>>>>>>>> >>> use (remember, it is relative to the src/dest number).  I'm
>>>>>>>> pretty
>>>>>>>> >>> sure you get something on the console log when you don't specify
>>>>>>>> a CID
>>>>>>>> >>> when using the commandline.  Anyway:
>>>>>>>> >>>
>>>>>>>> >>> lcr 617642  default 6176421212
>>>>>>>> >>>
>>>>>>>> >>> should give you intralata.
>>>>>>>> >>>
>>>>>>>> >>> Note that the definition of intralata doesn't mean "local" for
>>>>>>>> some
>>>>>>>> >>> providers.  Some providers define local to "same ratecenter"
>>>>>>>> which is
>>>>>>>> >>> even more restrictive.
>>>>>>>> >>>
>>>>>>>> >>> On Fri, Jan 29, 2010 at 4:43 PM, Mouncif Benniane <<mouncifbb at gmail.com>
>>>>>>>> mouncifbb at gmail.com>
>>>>>>>> >>> wrote:
>>>>>>>> >>> > i can't make use of mod_lcr using Intra/Interstate rating, I
>>>>>>>> am using
>>>>>>>> >>> > svn: FreeSWITCH Version 1.0.trunk (16517)
>>>>>>>> >>> >
>>>>>>>> >>> > lcr mysql table structure:
>>>>>>>> >>> > CREATE TABLE `lcr` (
>>>>>>>> >>> >   `id` INT(11) NOT NULL AUTO_INCREMENT,
>>>>>>>> >>> >   `digits` VARCHAR(15) DEFAULT NULL,
>>>>>>>> >>> >   `rate` FLOAT(11,5) DEFAULT NULL,
>>>>>>>> >>> >   `intrastate_rate` FLOAT(11,5) DEFAULT NULL,
>>>>>>>> >>> >   `intralata_rate` FLOAT(11,5) DEFAULT NULL,
>>>>>>>> >>> >   `carrier_id` INT(11) NOT NULL,
>>>>>>>> >>> >   `lead_strip` INT(11) NOT NULL,
>>>>>>>> >>> >   `trail_strip` INT(11) NOT NULL,
>>>>>>>> >>> >   `prefix` VARCHAR(16) NOT NULL,
>>>>>>>> >>> >   `suffix` VARCHAR(16) NOT NULL,
>>>>>>>> >>> >   `lcr_profile` VARCHAR(32) DEFAULT NULL,
>>>>>>>> >>> >   `date_start` DATETIME NOT NULL DEFAULT '1970-01-01
>>>>>>>> 00:00:00',
>>>>>>>> >>> >   `date_end` DATETIME NOT NULL DEFAULT '2030-12-31 00:00:00',
>>>>>>>> >>> >   `quality` FLOAT(10,6) NOT NULL,
>>>>>>>> >>> >   `reliability` FLOAT(10,6) NOT NULL,
>>>>>>>> >>> >   `cid` VARCHAR(32) NOT NULL DEFAULT '',
>>>>>>>> >>> >   `enabled` TINYINT(1) NOT NULL DEFAULT '1',
>>>>>>>> >>> >   PRIMARY KEY  (`id`),
>>>>>>>> >>> >   KEY `carrier_id` (`carrier_id`),
>>>>>>>> >>> >   KEY `digits` (`digits`),
>>>>>>>> >>> >   KEY `lcr_profile` (`lcr_profile`),
>>>>>>>> >>> >   KEY `digits_profile_cid_rate` USING BTREE (`digits`),
>>>>>>>> >>> >   CONSTRAINT `carrier_id` FOREIGN KEY (`carrier_id`)
>>>>>>>> REFERENCES
>>>>>>>> >>> > `carriers`
>>>>>>>> >>> > (`id`) ON DELETE CASCADE ON UPDATE CASCADE
>>>>>>>> >>> > ) ENGINE=INNODB AUTO_INCREMENT=6 DEFAULT CHARSET=latin1
>>>>>>>> >>> >
>>>>>>>> >>> >
>>>>>>>> >>> > lcr_admin show profiles
>>>>>>>> >>> > Name:           default
>>>>>>>> >>> >  custom sql:    SELECT l.digits, c.carrier_name,
>>>>>>>> l.${lcr_rate_field},
>>>>>>>> >>> > cg.prefix AS gw_prefix, cg.suffix AS gw_suffix, l.lead_strip,
>>>>>>>> >>> > l.trail_strip,
>>>>>>>> >>> > l.prefix, l.suffix , cg.codec , l.cid FROM lcr l JOIN carriers
>>>>>>>> c ON
>>>>>>>> >>> > l.carrier_id=c.id JOIN carrier_gateway cg ON c.id=cg.carrier_id
>>>>>>>> WHERE
>>>>>>>> >>> > c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1' AND
>>>>>>>> digits IN
>>>>>>>> >>> > (${lcr_query_expanded_digits}) AND CURRENT_TIMESTAMP BETWEEN
>>>>>>>> date_start
>>>>>>>> >>> > AND
>>>>>>>> >>> > date_end ORDER BY digits DESC,  ${lcr_rate_field},  quality
>>>>>>>> DESC,
>>>>>>>> >>> >  reliability DESC, rand();
>>>>>>>> >>> >  has %:         false
>>>>>>>> >>> >  has vars:      true
>>>>>>>> >>> >  has intrastate:        true
>>>>>>>> >>> >  has intralata: true
>>>>>>>> >>> >  has npanxx:    true
>>>>>>>> >>> >  Reorder rate:  enabled
>>>>>>>> >>> >  Info in headers:       disabled
>>>>>>>> >>> >  Quote IN() List:       disabled
>>>>>>>> >>> >
>>>>>>>> >>> >
>>>>>>>> >>> >
>>>>>>>> >>> > lcr 617642  default  returns rate from the rate field table
>>>>>>>> and not
>>>>>>>> >>> > intra/inter state fields rates.
>>>>>>>> >>> >
>>>>>>>> >>> > Any ideas? thanks!
>>>>>>>> >>> >
>>>>>>>> >>> >
>>>>>>>> >>> >
>>>>>>>> >>> >
>>>>>>>> >>> > _______________________________________________
>>>>>>>> >>> > FreeSWITCH-users mailing list
>>>>>>>> >>> > <FreeSWITCH-users at lists.freeswitch.org>
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> >>> >
>>>>>>>> <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> >>> >
>>>>>>>> >>> > UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> >>> > <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>>> >>> >
>>>>>>>> >>> >
>>>>>>>> >>>
>>>>>>>> >>>
>>>>>>>> >>>
>>>>>>>> >>> --
>>>>>>>> >>> -Rupa
>>>>>>>> >>>
>>>>>>>> >>> _______________________________________________
>>>>>>>> >>> FreeSWITCH-users mailing list
>>>>>>>> >>> <FreeSWITCH-users at lists.freeswitch.org>
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> >>> <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> >>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> >>> <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>>> >>
>>>>>>>> >
>>>>>>>> >
>>>>>>>> > _______________________________________________
>>>>>>>> > FreeSWITCH-users mailing list
>>>>>>>> > <FreeSWITCH-users at lists.freeswitch.org>
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> > <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> > UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> > <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>>> >
>>>>>>>> >
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> --
>>>>>>>> -Rupa
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>> -Rupa
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>>
>>>> --
>>>> -Rupa
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>> FreeSWITCH-users at lists.freeswitch.org
>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> -Rupa
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100201/342b0520/attachment-0002.html 

From mouncifbb at gmail.com  Mon Feb  1 08:16:29 2010
From: mouncifbb at gmail.com (Mouncif Benniane)
Date: Mon, 1 Feb 2010 11:16:29 -0500
Subject: [Freeswitch-users] mod_lcr problem
In-Reply-To: <ade741d61002010747v6da3d2e1x554b46aa26c27b18@mail.gmail.com>
References: <ade741d61001291443l6134a3achf28be64fc7746c8e@mail.gmail.com>
	<ade741d61001301338x8014e86t9c9094403f3361f@mail.gmail.com>
	<b39273531001301559h60668746kd5c63ada240513f1@mail.gmail.com>
	<ade741d61001301623w70d7e449i44a9f1e9fb7beae4@mail.gmail.com>
	<b39273531001301645w10446308o3578479dcbf2da2@mail.gmail.com>
	<ade741d61001302057w6ce72355iee6e8b4a2872fc81@mail.gmail.com>
	<b39273531001310532y1de53609rf474755cec04f218@mail.gmail.com>
	<9AB8B620-5A69-4F08-B62C-FFEB03FC6762@gmail.com>
	<b39273531001311507hb1bfe02m8d370443b882ef81@mail.gmail.com>
	<ade741d61002010747v6da3d2e1x554b46aa26c27b18@mail.gmail.com>
Message-ID: <ade741d61002010816x4e42ab0eqbfdb00089883654a@mail.gmail.com>

I got it! nevermind.

session.execute("set_profile_var","caller_id_number=1617947XXXX"); ( since I
am using js)

Thanks

On Mon, Feb 1, 2010 at 10:47 AM, Mouncif Benniane <mouncifbb at gmail.com>wrote:

> any example on how to use: set_profile_var?
>
> thanks
>
>
>
> On Sun, Jan 31, 2010 at 6:07 PM, Rupa Schomaker <rupa at rupa.com> wrote:
>
>> Yes, you need to normalize the values passed to lcr.  Otherwise, how could
>> it work?
>>
>> You can normalize the CID by matching and adding a 1 for 10 digit #s, or
>> removing the leading + or other things you might need then setting it back
>> to the profile using the set_profile_var app (
>> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_set_profile_var).
>>  (mod_cidlookup will set it after doing a #->name/area lookup - but for now
>> you can set it yourself)
>>
>> You can normalize the DID by doing similar matching rules as above and
>> then transfering to that normalized DID for the rest of your call plan
>> processing.
>>
>> I'm pretty sure mod_cidlookup has an example of normalizing... yeah:
>> http://wiki.freeswitch.org/wiki/Mod_cidlookup#Dialplan_Application
>>
>> On Sun, Jan 31, 2010 at 4:18 PM, Mouncifbb <mouncifbb at gmail.com> wrote:
>>
>>> So the CID must have 1 at front also? Usually people
>>> Send only npa and nxx ex 6176427788 7817612233
>>> Do I need to alter it?
>>>
>>> Sent from my iPhone
>>>
>>> On Jan 31, 2010, at 8:32 AM, Rupa Schomaker <rupa at rupa.com> wrote:
>>>
>>>
>>>
>>> On Sat, Jan 30, 2010 at 10:57 PM, Mouncif Benniane <<mouncifbb at gmail.com>
>>> mouncifbb at gmail.com> wrote:
>>>
>>>> OK going back to use default profile to keep things simple below 2
>>>> results
>>>>
>>>> Using:
>>>>
>>>> lcr 16179470890  default  19785223241  ( this one consult
>>>> npa_nxx_company_ocn)
>>>>
>>>> lcr 6179470890  default  9785223241 ( this one don't!! )
>>>>
>>>>
>>>>
>>> Oh, right!  mod_lcr really expects you to normalize your prefix to e164
>>> format.  I thought there was discussion about this in the wiki, but maybe
>>> not.  For simple prefix matching it doesn't matter, but for things that make
>>> decisions based on the # (like the lata/state stuff) it does.
>>>
>>> npanxx lookup only makes sense for NANPA numbers.  NANPA numbers have a
>>> country code of "1" and a total length of 11 (including the 1).
>>>
>>> This is the only rational way to do it when you have a rate table with
>>> both domestic (NANPA) and international prefixes.
>>>
>>>
>>>> freeswitch> lcr 16179470890  default  19785223241
>>>> 2010-01-30 23:53:34.681842 [DEBUG] mod_lcr.c:1329 data passed to lcr is
>>>> [16179470890  default  19785223241]
>>>> 2010-01-30 23:53:34.681842 [DEBUG] mod_lcr.c:1365 Set Caller ID to
>>>> [19785223241]
>>>> 2010-01-30 23:53:34.681842 [DEBUG] mod_lcr.c:736 SQL: SELECT 'state',
>>>> count(DISTINCT state) FROM npa_nxx_company_ocn WHERE (npa=617 AND nxx=947)
>>>> OR (npa=978 AND nxx=522) UNION SELECT 'lata', count(DISTINCT lata) FROM
>>>> npa_nxx_company_ocn WHERE (npa=617 AND nxx=947) OR (npa=978 AND nxx=522)
>>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:696 Type: state, Count: 1
>>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:696 Type: lata, Count: 1
>>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:786 intra routing [state:1
>>>> lata:1] so rate field is [intralata_rate]
>>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:802 we have an event
>>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:826 SQL: SELECT l.digits,
>>>> c.carrier_name, l.intralata_rate, cg.prefix AS gw_prefix, cg.suffix AS
>>>> gw_suffix, l.lead_strip, l.trail_strip, l.prefix, l.suffix , cg.codec ,
>>>> l.cid FROM lcr l JOIN carriers c ON l.carrier_id=c.id JOIN
>>>> carrier_gateway cg ON c.id=cg.carrier_id WHERE c.enabled = '1' AND
>>>> cg.enabled = '1' AND l.enabled = '1' AND digits IN (16179470890, 1617947089,
>>>> 161794708, 16179470, 1617947, 161794, 16179, 1617, 161, 16, 1) AND
>>>> CURRENT_TIMESTAMP BETWEEN date_start AND date_end ORDER BY digits DESC,
>>>> intralata_rate, rand();
>>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:307 Returning Dialstring
>>>> [lcr_carrier=carrier1,lcr_rate=0.00000]sofia/gateway/carrier1/16179470890
>>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:615 Adding carrier1 to head
>>>> of list
>>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:307 Returning Dialstring
>>>> [lcr_carrier=carrier2,lcr_rate=0.00000]sofia/external/
>>>> 06179470890 at proxy.carrier2.net:5060
>>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:667 adding carrier2 to end
>>>> of list after carrier1
>>>>
>>>>
>>>>  | Digit Match | Carrier  | Rate     | Codec | CID Regexp |
>>>> Dialstring
>>>> |
>>>>  | 1           | carrier1 | 0.00000  |       |            |
>>>> [lcr_carrier=carrier1,lcr_rate=0.00000]sofia/gateway/carrier1/16179470890
>>>> |
>>>>  | 1           | carrier2 | 0.00000  |       |            |
>>>> [lcr_carrier=carrier2,lcr_rate=0.00000]sofia/external/
>>>> 06179470890 at proxy.carrier2.net:5060 |
>>>>
>>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:307 Returning Dialstring
>>>> [lcr_carrier=carrier1,lcr_rate=0.00000]sofia/gateway/carrier1/16179470890
>>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:307 Returning Dialstring
>>>> [lcr_carrier=carrier2,lcr_rate=0.00000]sofia/external/
>>>> 06179470890 at proxy.carrier2.net:5060
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> freeswitch> lcr 6179470890  default  9785223241
>>>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:1329 data passed to lcr is
>>>> [6179470890  default  9785223241]
>>>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:1365 Set Caller ID to
>>>> [9785223241]
>>>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:786 intra routing [state:0
>>>> lata:0] so rate field is [rate]
>>>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:802 we have an event
>>>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:826 SQL: SELECT l.digits,
>>>> c.carrier_name, l.rate, cg.prefix AS gw_prefix, cg.suffix AS gw_suffix,
>>>> l.lead_strip, l.trail_strip, l.prefix, l.suffix , cg.codec , l.cid FROM lcr
>>>> l JOIN carriers c ON l.carrier_id=c.id JOIN carrier_gateway cg ON c.id=cg.carrier_id
>>>> WHERE c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1' AND digits IN
>>>> (6179470890, 617947089, 61794708, 6179470, 617947, 61794, 6179, 617, 61, 6)
>>>> AND CURRENT_TIMESTAMP BETWEEN date_start AND date_end ORDER BY digits DESC,
>>>> rate, rand();
>>>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:307 Returning Dialstring
>>>> [lcr_carrier=carrier1,lcr_rate=0.09000]sofia/gateway/carrier1/16179470890
>>>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:615 Adding carrier1 to head
>>>> of list
>>>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:307 Returning Dialstring
>>>> [lcr_carrier=carrier1,lcr_rate=0.09000]sofia/gateway/carrier1/16179470890
>>>>
>>>>
>>>>  | Digit Match | Carrier  | Rate     | Codec | CID Regexp |
>>>> Dialstring                                                                |
>>>>   | 617947      | carrier1 | 0.09000  |       |            |
>>>> [lcr_carrier=carrier1,lcr_rate=0.09000]sofia/gateway/carrier1/16179470890 |
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> On Sat, Jan 30, 2010 at 7:45 PM, Rupa Schomaker < <rupa at rupa.com>
>>>> rupa at rupa.com> wrote:
>>>>
>>>>> turn up logging to debug again, and then reload mod_lcr.  It'll spit
>>>>> out a bunch of crap when it tests out each profile you have defined.  Give
>>>>> me the full log (here or in <http://pastebin.freeswitch.org>
>>>>> pastebin.freeswitch.org).  That may show more useful info as to why
>>>>> things are mucked up?
>>>>>
>>>>>
>>>>> On Sat, Jan 30, 2010 at 6:23 PM, Mouncif Benniane <<mouncifbb at gmail.com>
>>>>> mouncifbb at gmail.com> wrote:
>>>>>
>>>>>> yes I use NANPA_STD profile instead of default cause I thought the
>>>>>> custom profile was causing issues, but looks like it's returning same
>>>>>> results.
>>>>>>
>>>>>> There is this line in thw wiki:
>>>>>> intra lata/state selection is done manually by setting the channel
>>>>>> variables *intrastate* or *intralata* to the value *true*.
>>>>>>
>>>>>> do I have to set these ? if yes how?
>>>>>>
>>>>>> Thanks
>>>>>>
>>>>>>
>>>>>> On Sat, Jan 30, 2010 at 6:59 PM, Rupa Schomaker < <rupa at rupa.com>
>>>>>> rupa at rupa.com> wrote:
>>>>>>
>>>>>>> Stuff inline.
>>>>>>>
>>>>>>> On Sat, Jan 30, 2010 at 3:38 PM, Mouncif Benniane <<mouncifbb at gmail.com>
>>>>>>> mouncifbb at gmail.com> wrote:
>>>>>>>
>>>>>>>>  NPANXX,"LATA","OCN","NTER","INTRA"
>>>>>>>> 201007,"224","7229","0.0059","0.0127"
>>>>>>>> 201040,"224","9206","0.0036","0.0036"
>>>>>>>>
>>>>>>>
>>>>>>> Looks like they give you the LATA and OCN values with the prefix.  We
>>>>>>> (should) look that up ourselves.
>>>>>>>
>>>>>>>
>>>>>>>> FreeSWITCH Version 1.0.trunk (16540)
>>>>>>>>
>>>>>>>>
>>>>>>>> Also I noticed the *npa_nxx_ocn* table never get consulted.
>>>>>>>>
>>>>>>>> I also see this now when making a real call instead of running
>>>>>>>> thorugh CLI
>>>>>>>>
>>>>>>>> EXECUTE sofia/external/6179472456 at 174.x.x.x lcr(16179470890
>>>>>>>> NANPA_STD)
>>>>>>>> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:1230 *intrastate
>>>>>>>> channel var is [undef]*
>>>>>>>
>>>>>>>
>>>>>>> This is fine.  it is a leftover from when you would tell mod_lcr via
>>>>>>> a channel var that it should do intrastate.  I later had mod_lcr do the
>>>>>>> lookup itself, but we still honor the old var.  There are no channel vars
>>>>>>> associated with the cli, so you wouldn't see that msg.
>>>>>>>
>>>>>>>
>>>>>>>>
>>>>>>>> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:1233 Select routes
>>>>>>>> based on interstate rates
>>>>>>>> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:1252 LCR Lookup on
>>>>>>>> 16179470893 using profile NANPA_STD
>>>>>>>> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:786 intra routing
>>>>>>>> [state:0 lata:0] so rate field is [rate]
>>>>>>>>
>>>>>>>> called number 6179470890  caller ID: 6179472456
>>>>>>>>
>>>>>>>> any ideas??
>>>>>>>>
>>>>>>>>
>>>>>>> Only thing that jumps out at me.
>>>>>>>
>>>>>>> The output from lcr_admin show profiles showed only the default one.
>>>>>>>  On the dialplan you use the NANPA_STD profile.  Can you check lcr_admin
>>>>>>> list and see if that profile is defined and if so if it says it is using the
>>>>>>> npanxx table?
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> On Sat, Jan 30, 2010 at 10:02 AM, Rupa Schomaker < <rupa at rupa.com>
>>>>>>>> rupa at rupa.com> wrote:
>>>>>>>>
>>>>>>>>> Something is still missing from the logs. Note the query of the
>>>>>>>>> npanxx
>>>>>>>>> table, the flags being set, and the rate field being chosen.  Umm..
>>>>>>>>> oh, what version of fs are you running?
>>>>>>>>>
>>>>>>>>> Yes, the npa_nxx_ocn table needs to be loaded up as described in:
>>>>>>>>> <http://wiki.freeswitch.org/wiki/Mod_cidlookup#Falling_back_to_.22City_State.22_in_the_absense_of_a_name>
>>>>>>>>> http://wiki.freeswitch.org/wiki/Mod_cidlookup#Falling_back_to_.22City_State.22_in_the_absense_of_a_name
>>>>>>>>> (there is a link to that from mod_lcr's wiki page).
>>>>>>>>>
>>>>>>>>> An example from my own setup:
>>>>>>>>>
>>>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:1384 data passed to
>>>>>>>>> lcr
>>>>>>>>> is [12148267711 default 12148267712]
>>>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:1420 Set Caller ID to
>>>>>>>>> [12148267712]
>>>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:759 SQL: SELECT
>>>>>>>>> 'state',
>>>>>>>>> count(DISTINCT state) FROM npa_nxx_company_ocn WHERE (npa=214 AND
>>>>>>>>> nxx=826) OR (npa=214 AND nxx=826) UNION SELECT 'lata',
>>>>>>>>> count(DISTINCT
>>>>>>>>> lata) FROM npa_nxx_company_ocn WHERE (npa=214 AND nxx=826) OR
>>>>>>>>> (npa=214
>>>>>>>>> AND nxx=826)
>>>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:719 Type: lata, Count:
>>>>>>>>> 1
>>>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:719 Type: state,
>>>>>>>>> Count: 1
>>>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:809 intra routing
>>>>>>>>> [state:1 lata:1] so rate field is [intralata_rate]
>>>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:825 we have an event
>>>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:849 SQL:  SELECT
>>>>>>>>> l.digits
>>>>>>>>> AS lcr_digits,         c.carrier_name AS lcr_carrier_name,
>>>>>>>>> l.intralata_rate as lcr_rate_field,        cg.prefix AS
>>>>>>>>> lcr_gw_prefix,
>>>>>>>>> cg.suffix AS lcr_gw_suffix,        l.lead_strip AS lcr_lead_strip,
>>>>>>>>> l.trail_strip AS lcr_trail_strip,        l.prefix AS lcr_prefix,
>>>>>>>>> l.suffix AS lcr_suffix,        cg.codec AS lcr_codec, l.cid AS
>>>>>>>>> lcr_cid,        'carriers' AS lcr_limit_realm, c.carrier_name AS
>>>>>>>>> lcr_limit_id, 5 AS lcr_limit_max   FROM lcr l      JOIN carriers c
>>>>>>>>> ON
>>>>>>>>> l.carrier_id=c.id      JOIN carrier_gateway cg ON c.id
>>>>>>>>> =cg.carrier_id
>>>>>>>>>  WHERE c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1'
>>>>>>>>> AND digits_prefix @> '12148267711'      AND CURRENT_TIMESTAMP
>>>>>>>>> BETWEEN
>>>>>>>>> date_start AND date_end   ORDER BY digits DESC, intralata_rate,
>>>>>>>>> random();
>>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning
>>>>>>>>> Dialstring
>>>>>>>>>
>>>>>>>>> [lcr_carrier=grnvoip_us,lcr_rate=0.00591]sofia/gateway/grnvoip/XXXX12148267711
>>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:638 Adding grnvoip_us
>>>>>>>>> to
>>>>>>>>> head of list
>>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning
>>>>>>>>> Dialstring
>>>>>>>>>
>>>>>>>>> [lcr_carrier=teliax,lcr_rate=0.01000]sofia/gateway/teliax_atl/12148267711
>>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:660 Adding teliax to
>>>>>>>>> end of list
>>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning
>>>>>>>>> Dialstring
>>>>>>>>>
>>>>>>>>> [lcr_carrier=teliax,lcr_rate=0.01000]sofia/gateway/teliax/12148267711
>>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:660 Adding teliax to
>>>>>>>>> end of list
>>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning
>>>>>>>>> Dialstring
>>>>>>>>>
>>>>>>>>> [lcr_carrier=vitelity,lcr_rate=0.01440]sofia/gateway/vitelity/12148267711
>>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:660 Adding vitelity to
>>>>>>>>> end of list
>>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning
>>>>>>>>> Dialstring
>>>>>>>>>
>>>>>>>>> [lcr_carrier=grnvoip_std,lcr_rate=0.01500]sofia/gateway/grnvoip/YYYY12148267711
>>>>>>>>> [...]
>>>>>>>>>
>>>>>>>>> On Fri, Jan 29, 2010 at 10:42 PM, Mouncif Benniane <<mouncifbb at gmail.com>
>>>>>>>>> mouncifbb at gmail.com> wrote:
>>>>>>>>> > Also the Provider has presented the rates in this format?
>>>>>>>>> > NPANXXLATA OCN INTER INTRA
>>>>>>>>> >
>>>>>>>>> >
>>>>>>>>> > On Fri, Jan 29, 2010 at 11:30 PM, Mouncif Benniane <<mouncifbb at gmail.com>
>>>>>>>>> mouncifbb at gmail.com>
>>>>>>>>> > wrote:
>>>>>>>>> >>
>>>>>>>>> >> Tried it and it's not giving me intralata  instead I get
>>>>>>>>> interstate, does
>>>>>>>>> >> the npa_nxx_company_ocn table needs to be used in this case?,
>>>>>>>>> also do I have
>>>>>>>>> >> to have the rate field in lcr table?
>>>>>>>>> >>
>>>>>>>>> >> lcr 617642  default 6176421212
>>>>>>>>> >>  | Digit Match | Carrier  | Rate     | Codec | CID Regexp |
>>>>>>>>> Dialstring
>>>>>>>>> >>                                                        |
>>>>>>>>> >>  | 617642      | carrier1 | 0.00500  |       |            |
>>>>>>>>> >>
>>>>>>>>> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642 |
>>>>>>>>> >>
>>>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:1329 data passed to
>>>>>>>>> lcr is
>>>>>>>>> >> [617642  default 6176421212]
>>>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:1365 Set Caller ID
>>>>>>>>> to
>>>>>>>>> >> [6176421212]
>>>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:786 intra routing
>>>>>>>>> [state:0
>>>>>>>>> >> lata:0] so rate field is [rate]
>>>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:802 we have an
>>>>>>>>> event
>>>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:826 SQL: SELECT
>>>>>>>>> l.digits,
>>>>>>>>> >> c.carrier_name, l.rate, cg.prefix AS gw_prefix, cg.suffix AS
>>>>>>>>> gw_suffix,
>>>>>>>>> >> l.lead_strip, l.trail_strip, l.prefix, l.suffix , cg.codec ,
>>>>>>>>> l.cid FROM lcr
>>>>>>>>> >> l JOIN carriers c ON l.carrier_id=c.id JOIN carrier_gateway cg
>>>>>>>>> ON
>>>>>>>>> >> c.id=cg.carrier_id WHERE c.enabled = '1' AND cg.enabled = '1'
>>>>>>>>> AND l.enabled
>>>>>>>>> >> = '1' AND digits IN (617642, 61764, 6176, 617, 61, 6) AND
>>>>>>>>> CURRENT_TIMESTAMP
>>>>>>>>> >> BETWEEN date_start AND date_end ORDER BY digits DESC,  rate,
>>>>>>>>> rand();
>>>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:307 Returning
>>>>>>>>> Dialstring
>>>>>>>>> >>
>>>>>>>>> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642
>>>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:615 Adding carrier1
>>>>>>>>> to head
>>>>>>>>> >> of list
>>>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:307 Returning
>>>>>>>>> Dialstring
>>>>>>>>> >>
>>>>>>>>> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642
>>>>>>>>> >>
>>>>>>>>> >> Thank you Rupa!
>>>>>>>>> >>
>>>>>>>>> >> On Fri, Jan 29, 2010 at 7:37 PM, Rupa Schomaker <<rupa at rupa.com>
>>>>>>>>> rupa at rupa.com> wrote:
>>>>>>>>> >>>
>>>>>>>>> >>> turn console logging up to debug and redo the lcr lookup.  The
>>>>>>>>> sql
>>>>>>>>> >>> statements along with status info will show up.  This should
>>>>>>>>> give
>>>>>>>>> >>> enough information to debug what is happening.
>>>>>>>>> >>>
>>>>>>>>> >>> I'm assuming the npanxx table is actually populated and not
>>>>>>>>> just
>>>>>>>>> >>> existing?
>>>>>>>>> >>>
>>>>>>>>> >>> When doing the lookup from the cli you have to tell lcr what
>>>>>>>>> CID to
>>>>>>>>> >>> use (remember, it is relative to the src/dest number).  I'm
>>>>>>>>> pretty
>>>>>>>>> >>> sure you get something on the console log when you don't
>>>>>>>>> specify a CID
>>>>>>>>> >>> when using the commandline.  Anyway:
>>>>>>>>> >>>
>>>>>>>>> >>> lcr 617642  default 6176421212
>>>>>>>>> >>>
>>>>>>>>> >>> should give you intralata.
>>>>>>>>> >>>
>>>>>>>>> >>> Note that the definition of intralata doesn't mean "local" for
>>>>>>>>> some
>>>>>>>>> >>> providers.  Some providers define local to "same ratecenter"
>>>>>>>>> which is
>>>>>>>>> >>> even more restrictive.
>>>>>>>>> >>>
>>>>>>>>> >>> On Fri, Jan 29, 2010 at 4:43 PM, Mouncif Benniane <<mouncifbb at gmail.com>
>>>>>>>>> mouncifbb at gmail.com>
>>>>>>>>> >>> wrote:
>>>>>>>>> >>> > i can't make use of mod_lcr using Intra/Interstate rating, I
>>>>>>>>> am using
>>>>>>>>> >>> > svn: FreeSWITCH Version 1.0.trunk (16517)
>>>>>>>>> >>> >
>>>>>>>>> >>> > lcr mysql table structure:
>>>>>>>>> >>> > CREATE TABLE `lcr` (
>>>>>>>>> >>> >   `id` INT(11) NOT NULL AUTO_INCREMENT,
>>>>>>>>> >>> >   `digits` VARCHAR(15) DEFAULT NULL,
>>>>>>>>> >>> >   `rate` FLOAT(11,5) DEFAULT NULL,
>>>>>>>>> >>> >   `intrastate_rate` FLOAT(11,5) DEFAULT NULL,
>>>>>>>>> >>> >   `intralata_rate` FLOAT(11,5) DEFAULT NULL,
>>>>>>>>> >>> >   `carrier_id` INT(11) NOT NULL,
>>>>>>>>> >>> >   `lead_strip` INT(11) NOT NULL,
>>>>>>>>> >>> >   `trail_strip` INT(11) NOT NULL,
>>>>>>>>> >>> >   `prefix` VARCHAR(16) NOT NULL,
>>>>>>>>> >>> >   `suffix` VARCHAR(16) NOT NULL,
>>>>>>>>> >>> >   `lcr_profile` VARCHAR(32) DEFAULT NULL,
>>>>>>>>> >>> >   `date_start` DATETIME NOT NULL DEFAULT '1970-01-01
>>>>>>>>> 00:00:00',
>>>>>>>>> >>> >   `date_end` DATETIME NOT NULL DEFAULT '2030-12-31 00:00:00',
>>>>>>>>> >>> >   `quality` FLOAT(10,6) NOT NULL,
>>>>>>>>> >>> >   `reliability` FLOAT(10,6) NOT NULL,
>>>>>>>>> >>> >   `cid` VARCHAR(32) NOT NULL DEFAULT '',
>>>>>>>>> >>> >   `enabled` TINYINT(1) NOT NULL DEFAULT '1',
>>>>>>>>> >>> >   PRIMARY KEY  (`id`),
>>>>>>>>> >>> >   KEY `carrier_id` (`carrier_id`),
>>>>>>>>> >>> >   KEY `digits` (`digits`),
>>>>>>>>> >>> >   KEY `lcr_profile` (`lcr_profile`),
>>>>>>>>> >>> >   KEY `digits_profile_cid_rate` USING BTREE (`digits`),
>>>>>>>>> >>> >   CONSTRAINT `carrier_id` FOREIGN KEY (`carrier_id`)
>>>>>>>>> REFERENCES
>>>>>>>>> >>> > `carriers`
>>>>>>>>> >>> > (`id`) ON DELETE CASCADE ON UPDATE CASCADE
>>>>>>>>> >>> > ) ENGINE=INNODB AUTO_INCREMENT=6 DEFAULT CHARSET=latin1
>>>>>>>>> >>> >
>>>>>>>>> >>> >
>>>>>>>>> >>> > lcr_admin show profiles
>>>>>>>>> >>> > Name:           default
>>>>>>>>> >>> >  custom sql:    SELECT l.digits, c.carrier_name,
>>>>>>>>> l.${lcr_rate_field},
>>>>>>>>> >>> > cg.prefix AS gw_prefix, cg.suffix AS gw_suffix, l.lead_strip,
>>>>>>>>> >>> > l.trail_strip,
>>>>>>>>> >>> > l.prefix, l.suffix , cg.codec , l.cid FROM lcr l JOIN
>>>>>>>>> carriers c ON
>>>>>>>>> >>> > l.carrier_id=c.id JOIN carrier_gateway cg ON c.id=cg.carrier_id
>>>>>>>>> WHERE
>>>>>>>>> >>> > c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1' AND
>>>>>>>>> digits IN
>>>>>>>>> >>> > (${lcr_query_expanded_digits}) AND CURRENT_TIMESTAMP BETWEEN
>>>>>>>>> date_start
>>>>>>>>> >>> > AND
>>>>>>>>> >>> > date_end ORDER BY digits DESC,  ${lcr_rate_field},  quality
>>>>>>>>> DESC,
>>>>>>>>> >>> >  reliability DESC, rand();
>>>>>>>>> >>> >  has %:         false
>>>>>>>>> >>> >  has vars:      true
>>>>>>>>> >>> >  has intrastate:        true
>>>>>>>>> >>> >  has intralata: true
>>>>>>>>> >>> >  has npanxx:    true
>>>>>>>>> >>> >  Reorder rate:  enabled
>>>>>>>>> >>> >  Info in headers:       disabled
>>>>>>>>> >>> >  Quote IN() List:       disabled
>>>>>>>>> >>> >
>>>>>>>>> >>> >
>>>>>>>>> >>> >
>>>>>>>>> >>> > lcr 617642  default  returns rate from the rate field table
>>>>>>>>> and not
>>>>>>>>> >>> > intra/inter state fields rates.
>>>>>>>>> >>> >
>>>>>>>>> >>> > Any ideas? thanks!
>>>>>>>>> >>> >
>>>>>>>>> >>> >
>>>>>>>>> >>> >
>>>>>>>>> >>> >
>>>>>>>>> >>> > _______________________________________________
>>>>>>>>> >>> > FreeSWITCH-users mailing list
>>>>>>>>> >>> > <FreeSWITCH-users at lists.freeswitch.org>
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> >>> >
>>>>>>>>> <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> >>> >
>>>>>>>>> >>> > UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> >>> > <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>>>> >>> >
>>>>>>>>> >>> >
>>>>>>>>> >>>
>>>>>>>>> >>>
>>>>>>>>> >>>
>>>>>>>>> >>> --
>>>>>>>>> >>> -Rupa
>>>>>>>>> >>>
>>>>>>>>> >>> _______________________________________________
>>>>>>>>> >>> FreeSWITCH-users mailing list
>>>>>>>>> >>> <FreeSWITCH-users at lists.freeswitch.org>
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> >>>
>>>>>>>>> <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> >>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> >>> <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>>>> >>
>>>>>>>>> >
>>>>>>>>> >
>>>>>>>>> > _______________________________________________
>>>>>>>>> > FreeSWITCH-users mailing list
>>>>>>>>> > <FreeSWITCH-users at lists.freeswitch.org>
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> > <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> > UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> > <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>>>> >
>>>>>>>>> >
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> --
>>>>>>>>> -Rupa
>>>>>>>>>
>>>>>>>>> _______________________________________________
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> --
>>>>>>> -Rupa
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> -Rupa
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> -Rupa
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> -Rupa
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100201/eb884ed3/attachment-0002.html 

From shouldbeq931 at googlemail.com  Mon Feb  1 08:54:33 2010
From: shouldbeq931 at googlemail.com (shouldbe q931)
Date: Mon, 1 Feb 2010 16:54:33 +0000
Subject: [Freeswitch-users] PRI Simulator
In-Reply-To: <1AF7532D-2B17-4D6E-97A0-421C595FF674@freeswitch.org>
References: <mailman.5734.1264787871.2495.freeswitch-users@lists.freeswitch.org>
	<AF84ED7F907F493DAC0A29C8FBC54E44@greyhawk.tonecommander.com>
	<1AF7532D-2B17-4D6E-97A0-421C595FF674@freeswitch.org>
Message-ID: <649eaa471002010854s2052d14dr32d5679b97f65c86@mail.gmail.com>

I use a 2nd user Avaya Prologix (v8, so it has teh trace functions),
its not exactly a simulator, but it suffices for most things :-)

Cheers

On Sat, Jan 30, 2010 at 12:21 AM, Michael S  Collins <msc at freeswitch.org> wrote:
> What's your budget?
>
> Sent from my iPhone
>
> On Jan 29, 2010, at 1:14 PM, "Jerry Richards" <jerry.richards at teotech.com
> ?> wrote:
>
>>
>> Can anyone recommend a good PRI simulator? ?Sorry this is off topic
>> a bit.
>>
>> Thanks,
>> Jerry
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>> users
>> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From mbsip at gazeta.pl  Mon Feb  1 09:30:36 2010
From: mbsip at gazeta.pl (mbsip)
Date: Mon, 1 Feb 2010 18:30:36 +0100
Subject: [Freeswitch-users] vm-disk-quota
In-Reply-To: <7DFBD163-A72A-4186-BB89-6D468FD9ABA2@jerris.com>
References: <28f27f5d1001311246h17b426a4x39e0d48d3d305342@mail.gmail.com>
	<7DFBD163-A72A-4186-BB89-6D468FD9ABA2@jerris.com>
Message-ID: <28f27f5d1002010930x61bcef3v54e22621761c8bbd@mail.gmail.com>

Probably You are right Mike.
I am about to do some tests and give you feadback here.

Thx,
Maciej.

2010/2/1 Michael Jerris <mike at jerris.com>:
> If I read the code right (mod_voicemail.c:3051) it looks like it is measuring in seconds of vocicemail, but the wiki indicates number of voicemails, neither seems to match the name of the param.
>
> Tamas- ?Can you comment on how this was intended to work?
>
> Mike
>
> On Jan 31, 2010, at 3:46 PM, mbsip wrote:
>
>> Hi ALL,
>>
>> Maybe this question will be piece of cake for most of you, but it
>> makes me think.
>>
>> I would like to configure "vm-disk-quota" for all users i have.
>> I followed the wiki page and provided:
>> <param name="vm-disk-quota" value="10"/>
>> to /conf/directory/default/1000.xml
>>
>> After reloadxml, incoming call give me "mod_voicemail.c:3057 Voicemail
>> disk quota is exceeded" feedback
>> No surprise for me because i had more less 10 voice mails already
>> recorded (before the vm-disk-quota was set up).
>> Strange is that increasing value even to 100 does not change anything.
>> The same thing with deleting recordings from user directory.
>> The only wayout is to set it to default value=0 (even FS shutdown
>> doesn't change anything)
>>
>> I am wondering why vm-disk-quota produces "Voicemail disk quota is
>> exceeded" all the time
>> Where the module is looking for stored voicemail recordings.
>>
>> Below is part of my configuration.
>> 1) /conf/autoload_configs/voicemail.conf.xml
>> <param name="storage-dir" value="/usr/local/freeswitch/vm"/>
>> 2) /conf/directory/default/1000.xml
>> <param name="vm-disk-quota" value="100"/>
>> 3) /vm/FS_ip_address/1000 is empty
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From jerry.richards at teotech.com  Mon Feb  1 09:30:57 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Mon, 1 Feb 2010 09:30:57 -0800
Subject: [Freeswitch-users] Presence Change Distribution
In-Reply-To: <191c3a031001271308l5c0c4eedw925e7660fbc2069d@mail.gmail.com>
References: <96888A19920E403880AAA9F6EE061BB6@greyhawk.tonecommander.com><2160023e0912290002q3d0f3fden5adee6d87d4bde25@mail.gmail.com><26B8578C14BA4BE18F2D1278B0C9561B@greyhawk.tonecommander.com><191c3a031001251104p55ba7009g9381841f7de56d65@mail.gmail.com><BE6F5600B8E54EE297963A8B08F70E5E@greyhawk.tonecommander.com><191c3a031001261321v2e8ea21cm7da19e01a11b59f9@mail.gmail.com><591B9C113F064880993543272B16ADF3@greyhawk.tonecommander.com>
	<191c3a031001271308l5c0c4eedw925e7660fbc2069d@mail.gmail.com>
Message-ID: <0616A915D48F4D439FA49390E881B2B6@greyhawk.tonecommander.com>

Cool.  It appears to be working now.  I see instantaneous changes in
presence between two Bria softphones.
 
Thanks and Best Regards,
Jerry


  _____  

From: Anthony Minessale [mailto:anthony.minessale at gmail.com] 
Sent: Wednesday, January 27, 2010 1:09 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Presence Change Distribution


Try latest trunk.
I tried forcing the db update in real-time to avoid a race on the event.



 


On Wed, Jan 27, 2010 at 1:56 PM, Jerry Richards <jerry.richards at teotech.com>
wrote:


There are two places in the XML body that are diffierent:
 
FS Rcvd PUBLISH has:  <rpid:away/> and <dm:note>Away</dm:note>
FS Sent NOTIFY   has:  <rpid:busy/> and <dm:note>Busy</dm:note>
 
This behavior (above) is why I'm not seeing the published presence at the
subscribing softphone.  FS should be sending the new Away status in the
NOTIFY message.
 
I did notice there is an "[ERR] sofia_presnece.c:674" FS log between the
PUBLISH and the NOTIFY (please see Line 89 of
http://pastebin.freeswitch.org/11953).  Line 674 is in the
sofia_presence_event_thread_run() function where it calls
switch_mutex_unlock(mod_sofia_globals.mutex).  Do you think this [ERR] is
related to why FS sends the previous status and not updated status?
 
Thanks And Best Regards,
Jerry
 

  _____  


From: Anthony Minessale [mailto:anthony.minessale at gmail.com] 

Sent: Tuesday, January 26, 2010 1:22 PM 

To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Presence Change Distribution


its sending a notify to them right away (line 174 of your PB)
the xml in the notify we send looks the same as what they sent except one
thing 

They send: <rpid:away/>
We send: <rpid:busy/>

everybody who implements this seems to have their own idea of what to say
here.

This crazy xml presence crap is pure garbage so maybe that's it.

 


On Tue, Jan 26, 2010 at 3:02 PM, Jerry Richards <jerry.richards at teotech.com>
wrote:


Okay, I setup my FS to force SUBSCRIBE Expires to 3600 seconds.  Then I
captured a FS console trace of a Bria softphone changing it's presence state
from 'Busy' to 'Away' (see http://pastebin.freeswitch.org/11953) and
observed that the subscribing Bria softphone did not update to 'Away'.  At
the same time, I executed the sqlite3 app and pasted each of the 3 SQL
select statements I saw in the FS console log, and pasted them below.  I'm
new to sqlite3.  Do you see what my issue is?
 
sqlite>
sqlite> select
sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_hos
t,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscripti
ons.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscripti
ons.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subsc
riptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,
'Away','away','192.168.72.79',sip_presence.status,sip_presence.rpid from
sip_subscriptions left join sip_presence on
(sip_subscriptions.sub_to_user=sip_presence.sip_user and
sip_subscriptions.sub_to_host=sip_presence.sip_host and
sip_subscriptions.profile_name=sip_presence.profile_name) where
sip_subscriptions.expires > -1 and (event='presence' or event='presence')
and sub_to_user='5401' and (sub_to_host='192.168.72.79' or presence_hosts
like '%192.168.72.79%') and (sip_subscriptions.profile_name = 'internal' or
sip_subscriptions.presence_hosts != sip_subscriptions.sub_to_host);
sip|5382|192.168.72.79|5401|192.168.72.79|presence|"5382 on 79"
<sip:5382 at 192.168.72.150:34672>|ZTQ2ZWQwZGRlZjRiNTdkYTJjNGM5NTgzOWIyNmIwZmU.
|"5382 on 79" <sip:5382 at 192.168.72.79 <mailto:sip%3A5382 at 192.168.72.79>
>;tag=68bb4eb6|SIP/2.0/UDP
192.168.72.150:34672;branch=z9hG4bK-d8754z-eafc60166305eaef-1---d8754z-;rpor
t=34672|1264546204|Teo Softphone release 2.5.4 stamp
55958||internal|Away|away|192.168.72.79|Away|away
sqlite>
sqlite>
sqlite>
sqlite>
sqlite>
sqlite>
sqlite>
sqlite>
sqlite> select
sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_hos
t,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscripti
ons.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscripti
ons.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subsc
riptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,
'Away','away','192.168.72.79',sip_presence.status,sip_presence.rpid from
sip_subscriptions left join sip_presence on
(sip_subscriptions.sub_to_user=sip_presence.sip_user and
sip_subscriptions.sub_to_host=sip_presence.sip_host and
sip_subscriptions.profile_name=sip_presence.profile_name) where
sip_subscriptions.expires > -1 and (event='presence' or event='presence')
and sub_to_user='5401' and (sub_to_host='192.168.72.79' or presence_hosts
like '%192.168.72.79%') and (sip_subscriptions.profile_name =
'internal-ipv6' or sip_subscriptions.presence_hosts !=
sip_subscriptions.sub_to_host);
sqlite>
sqlite>
sqlite>
sqlite>
sqlite>
sqlite>
sqlite>
sqlite> select
sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_hos
t,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscripti
ons.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscripti
ons.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subsc
riptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,
'Away','away','192.168.72.79',sip_presence.status,sip_presence.rpid from
sip_subscriptions left join sip_presence on
(sip_subscriptions.sub_to_user=sip_presence.sip_user and
sip_subscriptions.sub_to_host=sip_presence.sip_host and
sip_subscriptions.profile_name=sip_presence.profile_name) where
sip_subscriptions.expires > -1 and (event='presence' or event='presence')
and sub_to_user='5401' and (sub_to_host='192.168.72.79' or presence_hosts
like '%192.168.72.79%') and (sip_subscriptions.profile_name = 'external' or
sip_subscriptions.presence_hosts != sip_subscriptions.sub_to_host);
sqlite>

Thanks and Best Regards,
Jerry


  _____  

From: Anthony Minessale [mailto:anthony.minessale at gmail.com] 
Sent: Monday, January 25, 2010 11:05 AM 

To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Presence Change Distribution


the notify will be instant after the publish
the notify you see are not triggered by the publish or they would be
instant.

Same drill, turn on presence debugging in sofia.conf.xml 
and look at the sql stmts and see why



On Mon, Jan 25, 2010 at 12:30 PM, Jerry Richards
<jerry.richards at teotech.com> wrote:


Okay, I notice that if I reduce the Presence SUBSCRIBE Expires duration
(from 3600 seconds to 60 seconds), then the delay between PUBLISH's and
NOTIFY's is reduced, but FS still waits nearly 45 seconds to send the
NOTIFY's after it receives a PUBLISH.
 
Can a change be made in FS so that NOTIFYs are sent as a direct result of
receipt of the PUBLISH message?  And not tied to the SUBSCRIBE expiration?
I really don't want to configure all my phones to re-subscribe every 30 or
15 seconds.
 
Thanks and Best Regards,
Jerry
 


  _____  

From: RobertT [mailto:siniypin at gmail.com] 
Sent: Tuesday, December 29, 2009 12:02 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Presence Change Distribution


You can try to reduce your registration time. 
I for one made my client apps send PUBLISH message every minute in addition
to reduced registration time.

Regards, Robert.


2009/12/28 Jerry Richards <jerry.richards at teotech.com>


Is there a setting to control how fast FS distributes presence changes to
subscribers?  Currently, it appears to take several minutes before I see
presence changes.  I would like to see them almost instantaneously, if
possible.

Thanks and Best Regards,
Jerry


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org






-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
<mailto:MSN%3Aanthony_minessale at hotmail.com> 
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
<mailto:PAYPAL%3Aanthony.minessale at gmail.com> 
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:+19193869900



_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org






-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
<mailto:MSN%3Aanthony_minessale at hotmail.com> 
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
<mailto:PAYPAL%3Aanthony.minessale at gmail.com> 
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:+19193869900



_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org






-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
<mailto:MSN%3Aanthony_minessale at hotmail.com> 
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
<mailto:PAYPAL%3Aanthony.minessale at gmail.com> 
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:+19193869900


-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100201/04eb7edc/attachment-0002.html 

From rupa at rupa.com  Mon Feb  1 09:32:06 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Mon, 1 Feb 2010 11:32:06 -0600
Subject: [Freeswitch-users] mod_lcr problem
In-Reply-To: <ade741d61002010816x4e42ab0eqbfdb00089883654a@mail.gmail.com>
References: <ade741d61001291443l6134a3achf28be64fc7746c8e@mail.gmail.com>
	<b39273531001301559h60668746kd5c63ada240513f1@mail.gmail.com>
	<ade741d61001301623w70d7e449i44a9f1e9fb7beae4@mail.gmail.com>
	<b39273531001301645w10446308o3578479dcbf2da2@mail.gmail.com>
	<ade741d61001302057w6ce72355iee6e8b4a2872fc81@mail.gmail.com>
	<b39273531001310532y1de53609rf474755cec04f218@mail.gmail.com>
	<9AB8B620-5A69-4F08-B62C-FFEB03FC6762@gmail.com>
	<b39273531001311507hb1bfe02m8d370443b882ef81@mail.gmail.com>
	<ade741d61002010747v6da3d2e1x554b46aa26c27b18@mail.gmail.com>
	<ade741d61002010816x4e42ab0eqbfdb00089883654a@mail.gmail.com>
Message-ID: <b39273531002010932h387914dq8836aa26fbb2cda9@mail.gmail.com>

Thanks.  It is a bit of a foot-gun.  This example might be worth adding to
the wiki for that app, would you mind doing it?

On Mon, Feb 1, 2010 at 10:16 AM, Mouncif Benniane <mouncifbb at gmail.com>wrote:

> I got it! nevermind.
>
> session.execute("set_profile_var","caller_id_number=1617947XXXX"); ( since
> I am using js)
>
> Thanks
>
>
> On Mon, Feb 1, 2010 at 10:47 AM, Mouncif Benniane <mouncifbb at gmail.com>wrote:
>
>> any example on how to use: set_profile_var?
>>
>> thanks
>>
>>
>>
>> On Sun, Jan 31, 2010 at 6:07 PM, Rupa Schomaker <rupa at rupa.com> wrote:
>>
>>> Yes, you need to normalize the values passed to lcr.  Otherwise, how
>>> could it work?
>>>
>>> You can normalize the CID by matching and adding a 1 for 10 digit #s, or
>>> removing the leading + or other things you might need then setting it back
>>> to the profile using the set_profile_var app (
>>> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_set_profile_var).
>>>  (mod_cidlookup will set it after doing a #->name/area lookup - but for now
>>> you can set it yourself)
>>>
>>> You can normalize the DID by doing similar matching rules as above and
>>> then transfering to that normalized DID for the rest of your call plan
>>> processing.
>>>
>>> I'm pretty sure mod_cidlookup has an example of normalizing... yeah:
>>> http://wiki.freeswitch.org/wiki/Mod_cidlookup#Dialplan_Application
>>>
>>> On Sun, Jan 31, 2010 at 4:18 PM, Mouncifbb <mouncifbb at gmail.com> wrote:
>>>
>>>> So the CID must have 1 at front also? Usually people
>>>> Send only npa and nxx ex 6176427788 7817612233
>>>> Do I need to alter it?
>>>>
>>>> Sent from my iPhone
>>>>
>>>> On Jan 31, 2010, at 8:32 AM, Rupa Schomaker <rupa at rupa.com> wrote:
>>>>
>>>>
>>>>
>>>> On Sat, Jan 30, 2010 at 10:57 PM, Mouncif Benniane <<mouncifbb at gmail.com>
>>>> mouncifbb at gmail.com> wrote:
>>>>
>>>>> OK going back to use default profile to keep things simple below 2
>>>>> results
>>>>>
>>>>> Using:
>>>>>
>>>>> lcr 16179470890  default  19785223241  ( this one consult
>>>>> npa_nxx_company_ocn)
>>>>>
>>>>> lcr 6179470890  default  9785223241 ( this one don't!! )
>>>>>
>>>>>
>>>>>
>>>> Oh, right!  mod_lcr really expects you to normalize your prefix to e164
>>>> format.  I thought there was discussion about this in the wiki, but maybe
>>>> not.  For simple prefix matching it doesn't matter, but for things that make
>>>> decisions based on the # (like the lata/state stuff) it does.
>>>>
>>>> npanxx lookup only makes sense for NANPA numbers.  NANPA numbers have a
>>>> country code of "1" and a total length of 11 (including the 1).
>>>>
>>>> This is the only rational way to do it when you have a rate table with
>>>> both domestic (NANPA) and international prefixes.
>>>>
>>>>
>>>>> freeswitch> lcr 16179470890  default  19785223241
>>>>> 2010-01-30 23:53:34.681842 [DEBUG] mod_lcr.c:1329 data passed to lcr is
>>>>> [16179470890  default  19785223241]
>>>>> 2010-01-30 23:53:34.681842 [DEBUG] mod_lcr.c:1365 Set Caller ID to
>>>>> [19785223241]
>>>>> 2010-01-30 23:53:34.681842 [DEBUG] mod_lcr.c:736 SQL: SELECT 'state',
>>>>> count(DISTINCT state) FROM npa_nxx_company_ocn WHERE (npa=617 AND nxx=947)
>>>>> OR (npa=978 AND nxx=522) UNION SELECT 'lata', count(DISTINCT lata) FROM
>>>>> npa_nxx_company_ocn WHERE (npa=617 AND nxx=947) OR (npa=978 AND nxx=522)
>>>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:696 Type: state, Count: 1
>>>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:696 Type: lata, Count: 1
>>>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:786 intra routing [state:1
>>>>> lata:1] so rate field is [intralata_rate]
>>>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:802 we have an event
>>>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:826 SQL: SELECT l.digits,
>>>>> c.carrier_name, l.intralata_rate, cg.prefix AS gw_prefix, cg.suffix AS
>>>>> gw_suffix, l.lead_strip, l.trail_strip, l.prefix, l.suffix , cg.codec ,
>>>>> l.cid FROM lcr l JOIN carriers c ON l.carrier_id=c.id JOIN
>>>>> carrier_gateway cg ON c.id=cg.carrier_id WHERE c.enabled = '1' AND
>>>>> cg.enabled = '1' AND l.enabled = '1' AND digits IN (16179470890, 1617947089,
>>>>> 161794708, 16179470, 1617947, 161794, 16179, 1617, 161, 16, 1) AND
>>>>> CURRENT_TIMESTAMP BETWEEN date_start AND date_end ORDER BY digits DESC,
>>>>> intralata_rate, rand();
>>>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:307 Returning Dialstring
>>>>> [lcr_carrier=carrier1,lcr_rate=0.00000]sofia/gateway/carrier1/16179470890
>>>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:615 Adding carrier1 to
>>>>> head of list
>>>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:307 Returning Dialstring
>>>>> [lcr_carrier=carrier2,lcr_rate=0.00000]sofia/external/
>>>>> 06179470890 at proxy.carrier2.net:5060
>>>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:667 adding carrier2 to end
>>>>> of list after carrier1
>>>>>
>>>>>
>>>>>  | Digit Match | Carrier  | Rate     | Codec | CID Regexp |
>>>>> Dialstring
>>>>> |
>>>>>  | 1           | carrier1 | 0.00000  |       |            |
>>>>> [lcr_carrier=carrier1,lcr_rate=0.00000]sofia/gateway/carrier1/16179470890
>>>>> |
>>>>>  | 1           | carrier2 | 0.00000  |       |            |
>>>>> [lcr_carrier=carrier2,lcr_rate=0.00000]sofia/external/
>>>>> 06179470890 at proxy.carrier2.net:5060 |
>>>>>
>>>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:307 Returning Dialstring
>>>>> [lcr_carrier=carrier1,lcr_rate=0.00000]sofia/gateway/carrier1/16179470890
>>>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:307 Returning Dialstring
>>>>> [lcr_carrier=carrier2,lcr_rate=0.00000]sofia/external/
>>>>> 06179470890 at proxy.carrier2.net:5060
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> freeswitch> lcr 6179470890  default  9785223241
>>>>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:1329 data passed to lcr is
>>>>> [6179470890  default  9785223241]
>>>>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:1365 Set Caller ID to
>>>>> [9785223241]
>>>>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:786 intra routing [state:0
>>>>> lata:0] so rate field is [rate]
>>>>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:802 we have an event
>>>>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:826 SQL: SELECT l.digits,
>>>>> c.carrier_name, l.rate, cg.prefix AS gw_prefix, cg.suffix AS gw_suffix,
>>>>> l.lead_strip, l.trail_strip, l.prefix, l.suffix , cg.codec , l.cid FROM lcr
>>>>> l JOIN carriers c ON l.carrier_id=c.id JOIN carrier_gateway cg ON c.id=cg.carrier_id
>>>>> WHERE c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1' AND digits IN
>>>>> (6179470890, 617947089, 61794708, 6179470, 617947, 61794, 6179, 617, 61, 6)
>>>>> AND CURRENT_TIMESTAMP BETWEEN date_start AND date_end ORDER BY digits DESC,
>>>>> rate, rand();
>>>>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:307 Returning Dialstring
>>>>> [lcr_carrier=carrier1,lcr_rate=0.09000]sofia/gateway/carrier1/16179470890
>>>>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:615 Adding carrier1 to
>>>>> head of list
>>>>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:307 Returning Dialstring
>>>>> [lcr_carrier=carrier1,lcr_rate=0.09000]sofia/gateway/carrier1/16179470890
>>>>>
>>>>>
>>>>>  | Digit Match | Carrier  | Rate     | Codec | CID Regexp |
>>>>> Dialstring                                                                |
>>>>>   | 617947      | carrier1 | 0.09000  |       |            |
>>>>> [lcr_carrier=carrier1,lcr_rate=0.09000]sofia/gateway/carrier1/16179470890 |
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> On Sat, Jan 30, 2010 at 7:45 PM, Rupa Schomaker < <rupa at rupa.com>
>>>>> rupa at rupa.com> wrote:
>>>>>
>>>>>> turn up logging to debug again, and then reload mod_lcr.  It'll spit
>>>>>> out a bunch of crap when it tests out each profile you have defined.  Give
>>>>>> me the full log (here or in <http://pastebin.freeswitch.org>
>>>>>> pastebin.freeswitch.org).  That may show more useful info as to why
>>>>>> things are mucked up?
>>>>>>
>>>>>>
>>>>>> On Sat, Jan 30, 2010 at 6:23 PM, Mouncif Benniane <<mouncifbb at gmail.com>
>>>>>> mouncifbb at gmail.com> wrote:
>>>>>>
>>>>>>> yes I use NANPA_STD profile instead of default cause I thought the
>>>>>>> custom profile was causing issues, but looks like it's returning same
>>>>>>> results.
>>>>>>>
>>>>>>> There is this line in thw wiki:
>>>>>>> intra lata/state selection is done manually by setting the channel
>>>>>>> variables *intrastate* or *intralata* to the value *true*.
>>>>>>>
>>>>>>> do I have to set these ? if yes how?
>>>>>>>
>>>>>>> Thanks
>>>>>>>
>>>>>>>
>>>>>>> On Sat, Jan 30, 2010 at 6:59 PM, Rupa Schomaker < <rupa at rupa.com>
>>>>>>> rupa at rupa.com> wrote:
>>>>>>>
>>>>>>>> Stuff inline.
>>>>>>>>
>>>>>>>> On Sat, Jan 30, 2010 at 3:38 PM, Mouncif Benniane <<mouncifbb at gmail.com>
>>>>>>>> mouncifbb at gmail.com> wrote:
>>>>>>>>
>>>>>>>>>  NPANXX,"LATA","OCN","NTER","INTRA"
>>>>>>>>> 201007,"224","7229","0.0059","0.0127"
>>>>>>>>> 201040,"224","9206","0.0036","0.0036"
>>>>>>>>>
>>>>>>>>
>>>>>>>> Looks like they give you the LATA and OCN values with the prefix.
>>>>>>>>  We (should) look that up ourselves.
>>>>>>>>
>>>>>>>>
>>>>>>>>> FreeSWITCH Version 1.0.trunk (16540)
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> Also I noticed the *npa_nxx_ocn* table never get consulted.
>>>>>>>>>
>>>>>>>>> I also see this now when making a real call instead of running
>>>>>>>>> thorugh CLI
>>>>>>>>>
>>>>>>>>> EXECUTE sofia/external/6179472456 at 174.x.x.x lcr(16179470890
>>>>>>>>> NANPA_STD)
>>>>>>>>> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:1230 *intrastate
>>>>>>>>> channel var is [undef]*
>>>>>>>>
>>>>>>>>
>>>>>>>> This is fine.  it is a leftover from when you would tell mod_lcr via
>>>>>>>> a channel var that it should do intrastate.  I later had mod_lcr do the
>>>>>>>> lookup itself, but we still honor the old var.  There are no channel vars
>>>>>>>> associated with the cli, so you wouldn't see that msg.
>>>>>>>>
>>>>>>>>
>>>>>>>>>
>>>>>>>>> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:1233 Select routes
>>>>>>>>> based on interstate rates
>>>>>>>>> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:1252 LCR Lookup on
>>>>>>>>> 16179470893 using profile NANPA_STD
>>>>>>>>> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:786 intra routing
>>>>>>>>> [state:0 lata:0] so rate field is [rate]
>>>>>>>>>
>>>>>>>>> called number 6179470890  caller ID: 6179472456
>>>>>>>>>
>>>>>>>>> any ideas??
>>>>>>>>>
>>>>>>>>>
>>>>>>>> Only thing that jumps out at me.
>>>>>>>>
>>>>>>>> The output from lcr_admin show profiles showed only the default one.
>>>>>>>>  On the dialplan you use the NANPA_STD profile.  Can you check lcr_admin
>>>>>>>> list and see if that profile is defined and if so if it says it is using the
>>>>>>>> npanxx table?
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> On Sat, Jan 30, 2010 at 10:02 AM, Rupa Schomaker < <rupa at rupa.com>
>>>>>>>>> rupa at rupa.com> wrote:
>>>>>>>>>
>>>>>>>>>> Something is still missing from the logs. Note the query of the
>>>>>>>>>> npanxx
>>>>>>>>>> table, the flags being set, and the rate field being chosen.
>>>>>>>>>>  Umm..
>>>>>>>>>> oh, what version of fs are you running?
>>>>>>>>>>
>>>>>>>>>> Yes, the npa_nxx_ocn table needs to be loaded up as described in:
>>>>>>>>>> <http://wiki.freeswitch.org/wiki/Mod_cidlookup#Falling_back_to_.22City_State.22_in_the_absense_of_a_name>
>>>>>>>>>> http://wiki.freeswitch.org/wiki/Mod_cidlookup#Falling_back_to_.22City_State.22_in_the_absense_of_a_name
>>>>>>>>>> (there is a link to that from mod_lcr's wiki page).
>>>>>>>>>>
>>>>>>>>>> An example from my own setup:
>>>>>>>>>>
>>>>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:1384 data passed to
>>>>>>>>>> lcr
>>>>>>>>>> is [12148267711 default 12148267712]
>>>>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:1420 Set Caller ID to
>>>>>>>>>> [12148267712]
>>>>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:759 SQL: SELECT
>>>>>>>>>> 'state',
>>>>>>>>>> count(DISTINCT state) FROM npa_nxx_company_ocn WHERE (npa=214 AND
>>>>>>>>>> nxx=826) OR (npa=214 AND nxx=826) UNION SELECT 'lata',
>>>>>>>>>> count(DISTINCT
>>>>>>>>>> lata) FROM npa_nxx_company_ocn WHERE (npa=214 AND nxx=826) OR
>>>>>>>>>> (npa=214
>>>>>>>>>> AND nxx=826)
>>>>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:719 Type: lata,
>>>>>>>>>> Count: 1
>>>>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:719 Type: state,
>>>>>>>>>> Count: 1
>>>>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:809 intra routing
>>>>>>>>>> [state:1 lata:1] so rate field is [intralata_rate]
>>>>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:825 we have an event
>>>>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:849 SQL:  SELECT
>>>>>>>>>> l.digits
>>>>>>>>>> AS lcr_digits,         c.carrier_name AS lcr_carrier_name,
>>>>>>>>>> l.intralata_rate as lcr_rate_field,        cg.prefix AS
>>>>>>>>>> lcr_gw_prefix,
>>>>>>>>>> cg.suffix AS lcr_gw_suffix,        l.lead_strip AS lcr_lead_strip,
>>>>>>>>>> l.trail_strip AS lcr_trail_strip,        l.prefix AS lcr_prefix,
>>>>>>>>>> l.suffix AS lcr_suffix,        cg.codec AS lcr_codec, l.cid AS
>>>>>>>>>> lcr_cid,        'carriers' AS lcr_limit_realm, c.carrier_name AS
>>>>>>>>>> lcr_limit_id, 5 AS lcr_limit_max   FROM lcr l      JOIN carriers c
>>>>>>>>>> ON
>>>>>>>>>> l.carrier_id=c.id      JOIN carrier_gateway cg ON c.id
>>>>>>>>>> =cg.carrier_id
>>>>>>>>>>  WHERE c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1'
>>>>>>>>>> AND digits_prefix @> '12148267711'      AND CURRENT_TIMESTAMP
>>>>>>>>>> BETWEEN
>>>>>>>>>> date_start AND date_end   ORDER BY digits DESC, intralata_rate,
>>>>>>>>>> random();
>>>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning
>>>>>>>>>> Dialstring
>>>>>>>>>>
>>>>>>>>>> [lcr_carrier=grnvoip_us,lcr_rate=0.00591]sofia/gateway/grnvoip/XXXX12148267711
>>>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:638 Adding grnvoip_us
>>>>>>>>>> to
>>>>>>>>>> head of list
>>>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning
>>>>>>>>>> Dialstring
>>>>>>>>>>
>>>>>>>>>> [lcr_carrier=teliax,lcr_rate=0.01000]sofia/gateway/teliax_atl/12148267711
>>>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:660 Adding teliax to
>>>>>>>>>> end of list
>>>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning
>>>>>>>>>> Dialstring
>>>>>>>>>>
>>>>>>>>>> [lcr_carrier=teliax,lcr_rate=0.01000]sofia/gateway/teliax/12148267711
>>>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:660 Adding teliax to
>>>>>>>>>> end of list
>>>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning
>>>>>>>>>> Dialstring
>>>>>>>>>>
>>>>>>>>>> [lcr_carrier=vitelity,lcr_rate=0.01440]sofia/gateway/vitelity/12148267711
>>>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:660 Adding vitelity
>>>>>>>>>> to end of list
>>>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning
>>>>>>>>>> Dialstring
>>>>>>>>>>
>>>>>>>>>> [lcr_carrier=grnvoip_std,lcr_rate=0.01500]sofia/gateway/grnvoip/YYYY12148267711
>>>>>>>>>> [...]
>>>>>>>>>>
>>>>>>>>>> On Fri, Jan 29, 2010 at 10:42 PM, Mouncif Benniane <<mouncifbb at gmail.com>
>>>>>>>>>> mouncifbb at gmail.com> wrote:
>>>>>>>>>> > Also the Provider has presented the rates in this format?
>>>>>>>>>> > NPANXXLATA OCN INTER INTRA
>>>>>>>>>> >
>>>>>>>>>> >
>>>>>>>>>> > On Fri, Jan 29, 2010 at 11:30 PM, Mouncif Benniane <<mouncifbb at gmail.com>
>>>>>>>>>> mouncifbb at gmail.com>
>>>>>>>>>> > wrote:
>>>>>>>>>> >>
>>>>>>>>>> >> Tried it and it's not giving me intralata  instead I get
>>>>>>>>>> interstate, does
>>>>>>>>>> >> the npa_nxx_company_ocn table needs to be used in this case?,
>>>>>>>>>> also do I have
>>>>>>>>>> >> to have the rate field in lcr table?
>>>>>>>>>> >>
>>>>>>>>>> >> lcr 617642  default 6176421212
>>>>>>>>>> >>  | Digit Match | Carrier  | Rate     | Codec | CID Regexp |
>>>>>>>>>> Dialstring
>>>>>>>>>> >>                                                        |
>>>>>>>>>> >>  | 617642      | carrier1 | 0.00500  |       |            |
>>>>>>>>>> >>
>>>>>>>>>> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642 |
>>>>>>>>>> >>
>>>>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:1329 data passed
>>>>>>>>>> to lcr is
>>>>>>>>>> >> [617642  default 6176421212]
>>>>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:1365 Set Caller ID
>>>>>>>>>> to
>>>>>>>>>> >> [6176421212]
>>>>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:786 intra routing
>>>>>>>>>> [state:0
>>>>>>>>>> >> lata:0] so rate field is [rate]
>>>>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:802 we have an
>>>>>>>>>> event
>>>>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:826 SQL: SELECT
>>>>>>>>>> l.digits,
>>>>>>>>>> >> c.carrier_name, l.rate, cg.prefix AS gw_prefix, cg.suffix AS
>>>>>>>>>> gw_suffix,
>>>>>>>>>> >> l.lead_strip, l.trail_strip, l.prefix, l.suffix , cg.codec ,
>>>>>>>>>> l.cid FROM lcr
>>>>>>>>>> >> l JOIN carriers c ON l.carrier_id=c.id JOIN carrier_gateway cg
>>>>>>>>>> ON
>>>>>>>>>> >> c.id=cg.carrier_id WHERE c.enabled = '1' AND cg.enabled = '1'
>>>>>>>>>> AND l.enabled
>>>>>>>>>> >> = '1' AND digits IN (617642, 61764, 6176, 617, 61, 6) AND
>>>>>>>>>> CURRENT_TIMESTAMP
>>>>>>>>>> >> BETWEEN date_start AND date_end ORDER BY digits DESC,  rate,
>>>>>>>>>> rand();
>>>>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:307 Returning
>>>>>>>>>> Dialstring
>>>>>>>>>> >>
>>>>>>>>>> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642
>>>>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:615 Adding
>>>>>>>>>> carrier1 to head
>>>>>>>>>> >> of list
>>>>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:307 Returning
>>>>>>>>>> Dialstring
>>>>>>>>>> >>
>>>>>>>>>> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642
>>>>>>>>>> >>
>>>>>>>>>> >> Thank you Rupa!
>>>>>>>>>> >>
>>>>>>>>>> >> On Fri, Jan 29, 2010 at 7:37 PM, Rupa Schomaker <<rupa at rupa.com>
>>>>>>>>>> rupa at rupa.com> wrote:
>>>>>>>>>> >>>
>>>>>>>>>> >>> turn console logging up to debug and redo the lcr lookup.  The
>>>>>>>>>> sql
>>>>>>>>>> >>> statements along with status info will show up.  This should
>>>>>>>>>> give
>>>>>>>>>> >>> enough information to debug what is happening.
>>>>>>>>>> >>>
>>>>>>>>>> >>> I'm assuming the npanxx table is actually populated and not
>>>>>>>>>> just
>>>>>>>>>> >>> existing?
>>>>>>>>>> >>>
>>>>>>>>>> >>> When doing the lookup from the cli you have to tell lcr what
>>>>>>>>>> CID to
>>>>>>>>>> >>> use (remember, it is relative to the src/dest number).  I'm
>>>>>>>>>> pretty
>>>>>>>>>> >>> sure you get something on the console log when you don't
>>>>>>>>>> specify a CID
>>>>>>>>>> >>> when using the commandline.  Anyway:
>>>>>>>>>> >>>
>>>>>>>>>> >>> lcr 617642  default 6176421212
>>>>>>>>>> >>>
>>>>>>>>>> >>> should give you intralata.
>>>>>>>>>> >>>
>>>>>>>>>> >>> Note that the definition of intralata doesn't mean "local" for
>>>>>>>>>> some
>>>>>>>>>> >>> providers.  Some providers define local to "same ratecenter"
>>>>>>>>>> which is
>>>>>>>>>> >>> even more restrictive.
>>>>>>>>>> >>>
>>>>>>>>>> >>> On Fri, Jan 29, 2010 at 4:43 PM, Mouncif Benniane <<mouncifbb at gmail.com>
>>>>>>>>>> mouncifbb at gmail.com>
>>>>>>>>>> >>> wrote:
>>>>>>>>>> >>> > i can't make use of mod_lcr using Intra/Interstate rating, I
>>>>>>>>>> am using
>>>>>>>>>> >>> > svn: FreeSWITCH Version 1.0.trunk (16517)
>>>>>>>>>> >>> >
>>>>>>>>>> >>> > lcr mysql table structure:
>>>>>>>>>> >>> > CREATE TABLE `lcr` (
>>>>>>>>>> >>> >   `id` INT(11) NOT NULL AUTO_INCREMENT,
>>>>>>>>>> >>> >   `digits` VARCHAR(15) DEFAULT NULL,
>>>>>>>>>> >>> >   `rate` FLOAT(11,5) DEFAULT NULL,
>>>>>>>>>> >>> >   `intrastate_rate` FLOAT(11,5) DEFAULT NULL,
>>>>>>>>>> >>> >   `intralata_rate` FLOAT(11,5) DEFAULT NULL,
>>>>>>>>>> >>> >   `carrier_id` INT(11) NOT NULL,
>>>>>>>>>> >>> >   `lead_strip` INT(11) NOT NULL,
>>>>>>>>>> >>> >   `trail_strip` INT(11) NOT NULL,
>>>>>>>>>> >>> >   `prefix` VARCHAR(16) NOT NULL,
>>>>>>>>>> >>> >   `suffix` VARCHAR(16) NOT NULL,
>>>>>>>>>> >>> >   `lcr_profile` VARCHAR(32) DEFAULT NULL,
>>>>>>>>>> >>> >   `date_start` DATETIME NOT NULL DEFAULT '1970-01-01
>>>>>>>>>> 00:00:00',
>>>>>>>>>> >>> >   `date_end` DATETIME NOT NULL DEFAULT '2030-12-31
>>>>>>>>>> 00:00:00',
>>>>>>>>>> >>> >   `quality` FLOAT(10,6) NOT NULL,
>>>>>>>>>> >>> >   `reliability` FLOAT(10,6) NOT NULL,
>>>>>>>>>> >>> >   `cid` VARCHAR(32) NOT NULL DEFAULT '',
>>>>>>>>>> >>> >   `enabled` TINYINT(1) NOT NULL DEFAULT '1',
>>>>>>>>>> >>> >   PRIMARY KEY  (`id`),
>>>>>>>>>> >>> >   KEY `carrier_id` (`carrier_id`),
>>>>>>>>>> >>> >   KEY `digits` (`digits`),
>>>>>>>>>> >>> >   KEY `lcr_profile` (`lcr_profile`),
>>>>>>>>>> >>> >   KEY `digits_profile_cid_rate` USING BTREE (`digits`),
>>>>>>>>>> >>> >   CONSTRAINT `carrier_id` FOREIGN KEY (`carrier_id`)
>>>>>>>>>> REFERENCES
>>>>>>>>>> >>> > `carriers`
>>>>>>>>>> >>> > (`id`) ON DELETE CASCADE ON UPDATE CASCADE
>>>>>>>>>> >>> > ) ENGINE=INNODB AUTO_INCREMENT=6 DEFAULT CHARSET=latin1
>>>>>>>>>> >>> >
>>>>>>>>>> >>> >
>>>>>>>>>> >>> > lcr_admin show profiles
>>>>>>>>>> >>> > Name:           default
>>>>>>>>>> >>> >  custom sql:    SELECT l.digits, c.carrier_name,
>>>>>>>>>> l.${lcr_rate_field},
>>>>>>>>>> >>> > cg.prefix AS gw_prefix, cg.suffix AS gw_suffix,
>>>>>>>>>> l.lead_strip,
>>>>>>>>>> >>> > l.trail_strip,
>>>>>>>>>> >>> > l.prefix, l.suffix , cg.codec , l.cid FROM lcr l JOIN
>>>>>>>>>> carriers c ON
>>>>>>>>>> >>> > l.carrier_id=c.id JOIN carrier_gateway cg ON c.id=cg.carrier_id
>>>>>>>>>> WHERE
>>>>>>>>>> >>> > c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1' AND
>>>>>>>>>> digits IN
>>>>>>>>>> >>> > (${lcr_query_expanded_digits}) AND CURRENT_TIMESTAMP BETWEEN
>>>>>>>>>> date_start
>>>>>>>>>> >>> > AND
>>>>>>>>>> >>> > date_end ORDER BY digits DESC,  ${lcr_rate_field},  quality
>>>>>>>>>> DESC,
>>>>>>>>>> >>> >  reliability DESC, rand();
>>>>>>>>>> >>> >  has %:         false
>>>>>>>>>> >>> >  has vars:      true
>>>>>>>>>> >>> >  has intrastate:        true
>>>>>>>>>> >>> >  has intralata: true
>>>>>>>>>> >>> >  has npanxx:    true
>>>>>>>>>> >>> >  Reorder rate:  enabled
>>>>>>>>>> >>> >  Info in headers:       disabled
>>>>>>>>>> >>> >  Quote IN() List:       disabled
>>>>>>>>>> >>> >
>>>>>>>>>> >>> >
>>>>>>>>>> >>> >
>>>>>>>>>> >>> > lcr 617642  default  returns rate from the rate field table
>>>>>>>>>> and not
>>>>>>>>>> >>> > intra/inter state fields rates.
>>>>>>>>>> >>> >
>>>>>>>>>> >>> > Any ideas? thanks!
>>>>>>>>>> >>> >
>>>>>>>>>> >>> >
>>>>>>>>>> >>> >
>>>>>>>>>> >>> >
>>>>>>>>>> >>> > _______________________________________________
>>>>>>>>>> >>> > FreeSWITCH-users mailing list
>>>>>>>>>> >>> > <FreeSWITCH-users at lists.freeswitch.org>
>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>> >>> >
>>>>>>>>>> <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>> >>> >
>>>>>>>>>> >>> > UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>> >>> > <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>>>>> >>> >
>>>>>>>>>> >>> >
>>>>>>>>>> >>>
>>>>>>>>>> >>>
>>>>>>>>>> >>>
>>>>>>>>>> >>> --
>>>>>>>>>> >>> -Rupa
>>>>>>>>>> >>>
>>>>>>>>>> >>> _______________________________________________
>>>>>>>>>> >>> FreeSWITCH-users mailing list
>>>>>>>>>> >>> <FreeSWITCH-users at lists.freeswitch.org>
>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>> >>>
>>>>>>>>>> <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>> >>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>> >>> <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>>>>> >>
>>>>>>>>>> >
>>>>>>>>>> >
>>>>>>>>>> > _______________________________________________
>>>>>>>>>> > FreeSWITCH-users mailing list
>>>>>>>>>> > <FreeSWITCH-users at lists.freeswitch.org>
>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>> > <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>> > UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>> > <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>>>>> >
>>>>>>>>>> >
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> --
>>>>>>>>>> -Rupa
>>>>>>>>>>
>>>>>>>>>> _______________________________________________
>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> _______________________________________________
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>>>>
>>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> --
>>>>>>>> -Rupa
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>> -Rupa
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>>
>>>> --
>>>> -Rupa
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> -Rupa
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100201/72f5cf14/attachment-0002.html 

From ranjtech at gmail.com  Mon Feb  1 09:51:31 2010
From: ranjtech at gmail.com (RR)
Date: Mon, 1 Feb 2010 12:51:31 -0500
Subject: [Freeswitch-users] Outbound Call (No Registration)
In-Reply-To: <02dd01caa127$428aa760$c79ff620$@com>
References: <020c01ca9fe9$1d5952f0$580bf8d0$@com>
	<1254B3C5-D7D3-413D-BA82-54FE4789B360@freeswitch.org>
	<022701caa04a$44f60b80$cee22280$@com>
	<697C7F93-3737-4ABD-8934-6A48DC09C088@freeswitch.org>
	<025701caa0a7$e1ca6200$a55f2600$@com>
	<02b001caa10c$913b9100$b3b2b300$@com>
	<F03237B7-89EE-4CE4-A67F-FCB8E8D26700@freeswitch.org>
	<02c501caa113$105532b0$30ff9810$@com>
	<87f2f3b91001291110l226ce6d8u6e806d0a1b782c5f@mail.gmail.com>
	<02dd01caa127$428aa760$c79ff620$@com>
Message-ID: <032301caa367$307c6c60$91754520$@com>

Hi Folks,

 

As I continue to learn configuring FS, I am trying to use FS as a peering
switch sending a call to our SBC on the other side of the pond (across the
pacific) where the IP address of the FS (in the US) is configured to receive
traffic from. In this case no registration is required by the sending
gateway to make calls through the system overseas as its IP address is in
the "known" gateways (ACL) list. 

 

Two questions:

a)      Where must the configuration of this overseas gateway be? Currently
I have it in $FREESWITCH_HOME/conf/sip_profiles/external. Is that the right
location for it?

b)      Looking at the gateway configuration that was present in that
directory for the sample gateway, I didn't see any of the params that were
relevant other than the "realm" and maybe the "proxy". Hence my
configuration is extremely simple (2 Lines). As I mentioned, I don't need to
register with the overseas SBC. The problem with this now is that this
gateway is not being picked up by FS when I do a reloadxml OR restart FS. On
typing "sofia status gateway MyGW", I get "Invalid Gateway". Following from
this then, when the dialplan is configured to send the call using this
gateway, I see the messages like

 

2010-02-01 12:42:50.138444 [ERR] mod_sofia.c:3108 Invalid Gateway

2010-02-01 12:42:50.138444 [ERR] switch_ivr_originate.c:2411 Cannot create
outgoing channel of type [sofia] cause: [INVALID_NUMBER_FORMAT]

2010-02-01 12:42:50.138444 [DEBUG] switch_ivr_originate.c:3209 Originate
Resulted in Error Cause: 28 [INVALID_NUMBER_FORMAT]

2010-02-01 12:42:50.138444 [INFO] mod_dptools.c:2346 Originate Failed.
Cause: INVALID_NUMBER_FORMAT

2010-02-01 12:42:50.138444 [NOTICE] mod_dptools.c:2409 Hangup
sofia/internal/1000 at 10.1.2.110 [CS_EXECUTE] [INVALID_NUMBER_FORMAT]

 

c)       Please note that the call has NOT even left FS so this message is
not coming from the remote gateway/SBC as I have a trace running there as
well and it never sees the call. Also, when I start FS, I see the error
"username is a required param". Upon entering a bogus username and password
field, I am able to load the gateway and the prfile status looks good.
However, when I try and make a call I get the following:

d)       

2010-02-01 12:49:03.118479 [DEBUG] sofia.c:4011 Channel
sofia/external/0061434144942 entering state [terminated][403]

2010-02-01 12:49:03.118479 [NOTICE] sofia.c:4655 Hangup
sofia/external/0061434144942 [CS_CONSUME_MEDIA] [CALL_REJECTED]

2010-02-01 12:49:03.118479 [DEBUG] switch_channel.c:1947 Send signal
sofia/external/0061434144942 [KILL]

2010-02-01 12:49:03.118479 [DEBUG] switch_core_session.c:1019 Send signal
sofia/external/0061434144942 [BREAK]

2010-02-01 12:49:03.118479 [DEBUG] switch_ivr_originate.c:3209 Originate
Resulted in Error Cause: 21 [CALL_REJECTED]

2010-02-01 12:49:03.118479 [INFO] mod_dptools.c:2346 Originate Failed.
Cause: CALL_REJECTED

2010-02-01 12:49:03.118479 [NOTICE] mod_dptools.c:2409 Hangup
sofia/internal/1000 at 10.1.2.110 [CS_EXECUTE] [CALL_REJECTED]

 

Any ideas?

 

Thanks

\R

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100201/0138de52/attachment-0002.html 

From msc at freeswitch.org  Mon Feb  1 09:58:10 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 1 Feb 2010 09:58:10 -0800
Subject: [Freeswitch-users] PRI Simulator
In-Reply-To: <649eaa471002010854s2052d14dr32d5679b97f65c86@mail.gmail.com>
References: <mailman.5734.1264787871.2495.freeswitch-users@lists.freeswitch.org>
	<AF84ED7F907F493DAC0A29C8FBC54E44@greyhawk.tonecommander.com>
	<1AF7532D-2B17-4D6E-97A0-421C595FF674@freeswitch.org>
	<649eaa471002010854s2052d14dr32d5679b97f65c86@mail.gmail.com>
Message-ID: <87f2f3b91002010958t1e0d9188v659a237bade6e7d5@mail.gmail.com>

On Mon, Feb 1, 2010 at 8:54 AM, shouldbe q931
<shouldbeq931 at googlemail.com>wrote:

> I use a 2nd user Avaya Prologix (v8, so it has teh trace functions),
> its not exactly a simulator, but it suffices for most things :-)
>
> Cheers
>

Heck, in a pinch I've used an Asterisk box. Sure it's unpredictable, crashes
all the time and you never know if it's doing what it's supposed to be
doing. Hmm, on second thought, I guess that makes it a great PRI simulator
because it's just like all the lame telcos...

-MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100201/84a23366/attachment-0002.html 

From shouldbeq931 at googlemail.com  Mon Feb  1 10:06:40 2010
From: shouldbeq931 at googlemail.com (shouldbe q931)
Date: Mon, 1 Feb 2010 18:06:40 +0000
Subject: [Freeswitch-users] PRI Simulator
In-Reply-To: <87f2f3b91002010958t1e0d9188v659a237bade6e7d5@mail.gmail.com>
References: <mailman.5734.1264787871.2495.freeswitch-users@lists.freeswitch.org>
	<AF84ED7F907F493DAC0A29C8FBC54E44@greyhawk.tonecommander.com>
	<1AF7532D-2B17-4D6E-97A0-421C595FF674@freeswitch.org>
	<649eaa471002010854s2052d14dr32d5679b97f65c86@mail.gmail.com>
	<87f2f3b91002010958t1e0d9188v659a237bade6e7d5@mail.gmail.com>
Message-ID: <649eaa471002011006o397574f0nc7e239acfbba229d@mail.gmail.com>

The Prologix might be a little more stable :-)

Eicon (now Dialogic) Diva Server cards running under Windows or Linux
are quite good as well...

On Mon, Feb 1, 2010 at 5:58 PM, Michael Collins <msc at freeswitch.org> wrote:
>
>
> On Mon, Feb 1, 2010 at 8:54 AM, shouldbe q931 <shouldbeq931 at googlemail.com>
> wrote:
>>
>> I use a 2nd user Avaya Prologix (v8, so it has teh trace functions),
>> its not exactly a simulator, but it suffices for most things :-)
>>
>> Cheers
>
> Heck, in a pinch I've used an Asterisk box. Sure it's unpredictable, crashes
> all the time and you never know if it's doing what it's supposed to be
> doing. Hmm, on second thought, I guess that makes it a great PRI simulator
> because it's just like all the lame telcos...
>
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From paul.gore.j at gmail.com  Mon Feb  1 11:18:52 2010
From: paul.gore.j at gmail.com (paul gore)
Date: Mon, 1 Feb 2010 14:18:52 -0500
Subject: [Freeswitch-users] Logging question
In-Reply-To: <8BEB200D-32AC-4D80-B59D-07C8228D7380@jerris.com>
References: <da4ab26e1001292020o280f330ft62a278b93c7fb1fd@mail.gmail.com>
	<8BEB200D-32AC-4D80-B59D-07C8228D7380@jerris.com>
Message-ID: <da4ab26e1002011118i2a5dc8a0yc836b799662adfab@mail.gmail.com>

I did "sofia profile internal siptrace on" - and that gave me the trace on
the console, but still nothing in log files.
This command seem to have same effect as changing the parameter in the
profile xml file and then do profile rescan.
So there is no way to get the trace in logs?

On Mon, Feb 1, 2010 at 1:04 AM, Michael Jerris <mike at jerris.com> wrote:

> sofia profile <profile_name> siptrace on
>
> There is also a config param, it should be documented int he current
> default configs.
>
> Mike
>
>
> On Jan 29, 2010, at 11:20 PM, paul gore wrote:
>
> > Hi there,
> > I am running FS 1.0.trunk (14501) (I know it's old but we serve a small
> community and don't have time to upgrade/test the latest/greatest). I am
> having troubles understanding how to switch SIP trace in log files, I tried
> >
> > fsctl loglevel debug
> > sofia tracelevel debug
> >
> > but it seem to have no effect, I only get sofia debug messages but no
> detailed SIP info.
> > What also puzzling me is if I do
> >
> > console loglevel 0
> >
> > I still get debug information on console.
> > What am I doing wrong?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100201/b5e769a5/attachment-0002.html 

From scottferri09 at gmail.com  Mon Feb  1 11:22:52 2010
From: scottferri09 at gmail.com (Scott Fernandez)
Date: Tue, 2 Feb 2010 00:52:52 +0530
Subject: [Freeswitch-users] Limit the extension creation
Message-ID: <df97b9c51002011122t29c505adu60c4ccb73746432c@mail.gmail.com>

Hi,

Is there a way to restrict the number of extension that FS supports/serves?.
The idea is to limit the concurrent usage of the system for which we need to
restrict the FS to support upto a predefined no. of users/extensions.

Can anyone assist please?

Thanks,
Scott.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100202/2b0c01de/attachment-0002.html 

From frank at carmickle.com  Mon Feb  1 11:34:59 2010
From: frank at carmickle.com (Frank Carmickle)
Date: Mon, 1 Feb 2010 14:34:59 -0500
Subject: [Freeswitch-users] Outbound Call (No Registration)
In-Reply-To: <032301caa367$307c6c60$91754520$@com>
References: <1254B3C5-D7D3-413D-BA82-54FE4789B360@freeswitch.org>
	<022701caa04a$44f60b80$cee22280$@com>
	<697C7F93-3737-4ABD-8934-6A48DC09C088@freeswitch.org>
	<025701caa0a7$e1ca6200$a55f2600$@com>
	<02b001caa10c$913b9100$b3b2b300$@com>
	<F03237B7-89EE-4CE4-A67F-FCB8E8D26700@freeswitch.org>
	<02c501caa113$105532b0$30ff9810$@com>
	<87f2f3b91001291110l226ce6d8u6e806d0a1b782c5f@mail.gmail.com>
	<02dd01caa127$428aa760$c79ff620$@com>
	<032301caa367$307c6c60$91754520$@com>
Message-ID: <20100201193459.GI27405@base.carmickle.com>

On Mon, Feb 01, RR wrote:
> Hi Folks,
> 
>  
> 
> As I continue to learn configuring FS, I am trying to use FS as a peering
> switch sending a call to our SBC on the other side of the pond (across the
> pacific) where the IP address of the FS (in the US) is configured to receive
> traffic from. In this case no registration is required by the sending
> gateway to make calls through the system overseas as its IP address is in
> the "known" gateways (ACL) list. 
> 

I allowed calls in to my pbx from a specific ip in the dialplan with out an acl or a gateway by putting a statement like

  <extension name="fjc-pbx-inbound">
  <condition field="network_addr" expression="^2001\:470\:XXXX\:XXXX\:X:X\:X\:X$"/>
    <condition field="destination_number" expression="^(.*)$">
      <action application="transfer" data="$1 xml default"/>
    </condition>
  </extension>

in the public section of the dialplan.  In this example you see that it is a ipv6 address.  This could easily be replaced with a ipv4 address like the level3 address 4.3.2.1.

There are other ways to do this with a gateway if you so choose.

--FC



From ranjtech at gmail.com  Mon Feb  1 12:24:17 2010
From: ranjtech at gmail.com (RR)
Date: Mon, 1 Feb 2010 15:24:17 -0500
Subject: [Freeswitch-users] Outbound Call (No Registration)
In-Reply-To: <20100201193459.GI27405@base.carmickle.com>
References: <1254B3C5-D7D3-413D-BA82-54FE4789B360@freeswitch.org>
	<022701caa04a$44f60b80$cee22280$@com>
	<697C7F93-3737-4ABD-8934-6A48DC09C088@freeswitch.org>
	<025701caa0a7$e1ca6200$a55f2600$@com>
	<02b001caa10c$913b9100$b3b2b300$@com>
	<F03237B7-89EE-4CE4-A67F-FCB8E8D26700@freeswitch.org>
	<02c501caa113$105532b0$30ff9810$@com>
	<87f2f3b91001291110l226ce6d8u6e806d0a1b782c5f@mail.gmail.com>
	<02dd01caa127$428aa760$c79ff620$@com>
	<032301caa367$307c6c60$91754520$@com>
	<20100201193459.GI27405@base.carmickle.com>
Message-ID: <033e01caa37c$87cd2080$97676180$@com>

Hi Frank,

Thanks for the response. The remote gateway is not running FreeSWITCH. But
that's ok, I figured out the problem. I was capturing packets from the wrong
IP address so after adding the username/password, the call did manage to get
to the other end (after I realized the source IP was wrong) and realized
that my dialplan at the other side wasn't setup correctly. Should be able to
fix that. 

BTW, before I go hunting in the Wiki, can you off the top of your head tell
me how I can manipulate numbers in the dialplan?

If my destination number = 00614xxxxxxxx but I want to send 120#614xxxxxxx
instead, how would I do that?

Thanks
\R

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Frank
Carmickle
Sent: Monday, February 01, 2010 2:35 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Outbound Call (No Registration)

On Mon, Feb 01, RR wrote:
> Hi Folks,
> 
>  
> 
> As I continue to learn configuring FS, I am trying to use FS as a peering
> switch sending a call to our SBC on the other side of the pond (across the
> pacific) where the IP address of the FS (in the US) is configured to
receive
> traffic from. In this case no registration is required by the sending
> gateway to make calls through the system overseas as its IP address is in
> the "known" gateways (ACL) list. 
> 

I allowed calls in to my pbx from a specific ip in the dialplan with out an
acl or a gateway by putting a statement like

  <extension name="fjc-pbx-inbound">
  <condition field="network_addr"
expression="^2001\:470\:XXXX\:XXXX\:X:X\:X\:X$"/>
    <condition field="destination_number" expression="^(.*)$">
      <action application="transfer" data="$1 xml default"/>
    </condition>
  </extension>

in the public section of the dialplan.  In this example you see that it is a
ipv6 address.  This could easily be replaced with a ipv4 address like the
level3 address 4.3.2.1.

There are other ways to do this with a gateway if you so choose.

--FC

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
 

__________ Information from ESET NOD32 Antivirus, version of virus signature
database 4824 (20100201) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com
 
 

__________ Information from ESET NOD32 Antivirus, version of virus signature
database 4825 (20100201) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com
 




From mouncifbb at gmail.com  Mon Feb  1 12:25:57 2010
From: mouncifbb at gmail.com (Mouncif Benniane)
Date: Mon, 1 Feb 2010 15:25:57 -0500
Subject: [Freeswitch-users] mod_lcr problem
In-Reply-To: <b39273531001311507hb1bfe02m8d370443b882ef81@mail.gmail.com>
References: <ade741d61001291443l6134a3achf28be64fc7746c8e@mail.gmail.com>
	<b39273531001300702r1a81f094pac0bd20aeb5ceed6@mail.gmail.com>
	<ade741d61001301338x8014e86t9c9094403f3361f@mail.gmail.com>
	<b39273531001301559h60668746kd5c63ada240513f1@mail.gmail.com>
	<ade741d61001301623w70d7e449i44a9f1e9fb7beae4@mail.gmail.com>
	<b39273531001301645w10446308o3578479dcbf2da2@mail.gmail.com>
	<ade741d61001302057w6ce72355iee6e8b4a2872fc81@mail.gmail.com>
	<b39273531001310532y1de53609rf474755cec04f218@mail.gmail.com>
	<9AB8B620-5A69-4F08-B62C-FFEB03FC6762@gmail.com>
	<b39273531001311507hb1bfe02m8d370443b882ef81@mail.gmail.com>
Message-ID: <ade741d61002011225t15d66333v69f31b0572880de9@mail.gmail.com>

So I have to alter my LCR table to look like:

NPANXX,"LATA","OCN","NTER","INTRA"  1201007,"224","7229","0.0059","0.0127"
1201040,"224","9206","0.0036","0.0036"


instead of:


NPANXX,"LATA","OCN","NTER","INTRA"  201007,"224","7229","0.0059","0.0127"
201040,"224","9206","0.0036","0.0036"


On Sun, Jan 31, 2010 at 6:07 PM, Rupa Schomaker <rupa at rupa.com> wrote:

> Yes, you need to normalize the values passed to lcr.  Otherwise, how could
> it work?
>
> You can normalize the CID by matching and adding a 1 for 10 digit #s, or
> removing the leading + or other things you might need then setting it back
> to the profile using the set_profile_var app (
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_set_profile_var).
>  (mod_cidlookup will set it after doing a #->name/area lookup - but for now
> you can set it yourself)
>
> You can normalize the DID by doing similar matching rules as above and then
> transfering to that normalized DID for the rest of your call plan
> processing.
>
> I'm pretty sure mod_cidlookup has an example of normalizing... yeah:
> http://wiki.freeswitch.org/wiki/Mod_cidlookup#Dialplan_Application
>
> On Sun, Jan 31, 2010 at 4:18 PM, Mouncifbb <mouncifbb at gmail.com> wrote:
>
>> So the CID must have 1 at front also? Usually people
>> Send only npa and nxx ex 6176427788 7817612233
>> Do I need to alter it?
>>
>> Sent from my iPhone
>>
>> On Jan 31, 2010, at 8:32 AM, Rupa Schomaker <rupa at rupa.com> wrote:
>>
>>
>>
>> On Sat, Jan 30, 2010 at 10:57 PM, Mouncif Benniane <<mouncifbb at gmail.com>
>> mouncifbb at gmail.com> wrote:
>>
>>> OK going back to use default profile to keep things simple below 2
>>> results
>>>
>>> Using:
>>>
>>> lcr 16179470890  default  19785223241  ( this one consult
>>> npa_nxx_company_ocn)
>>>
>>> lcr 6179470890  default  9785223241 ( this one don't!! )
>>>
>>>
>>>
>> Oh, right!  mod_lcr really expects you to normalize your prefix to e164
>> format.  I thought there was discussion about this in the wiki, but maybe
>> not.  For simple prefix matching it doesn't matter, but for things that make
>> decisions based on the # (like the lata/state stuff) it does.
>>
>> npanxx lookup only makes sense for NANPA numbers.  NANPA numbers have a
>> country code of "1" and a total length of 11 (including the 1).
>>
>> This is the only rational way to do it when you have a rate table with
>> both domestic (NANPA) and international prefixes.
>>
>>
>>> freeswitch> lcr 16179470890  default  19785223241
>>> 2010-01-30 23:53:34.681842 [DEBUG] mod_lcr.c:1329 data passed to lcr is
>>> [16179470890  default  19785223241]
>>> 2010-01-30 23:53:34.681842 [DEBUG] mod_lcr.c:1365 Set Caller ID to
>>> [19785223241]
>>> 2010-01-30 23:53:34.681842 [DEBUG] mod_lcr.c:736 SQL: SELECT 'state',
>>> count(DISTINCT state) FROM npa_nxx_company_ocn WHERE (npa=617 AND nxx=947)
>>> OR (npa=978 AND nxx=522) UNION SELECT 'lata', count(DISTINCT lata) FROM
>>> npa_nxx_company_ocn WHERE (npa=617 AND nxx=947) OR (npa=978 AND nxx=522)
>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:696 Type: state, Count: 1
>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:696 Type: lata, Count: 1
>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:786 intra routing [state:1
>>> lata:1] so rate field is [intralata_rate]
>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:802 we have an event
>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:826 SQL: SELECT l.digits,
>>> c.carrier_name, l.intralata_rate, cg.prefix AS gw_prefix, cg.suffix AS
>>> gw_suffix, l.lead_strip, l.trail_strip, l.prefix, l.suffix , cg.codec ,
>>> l.cid FROM lcr l JOIN carriers c ON l.carrier_id=c.id JOIN
>>> carrier_gateway cg ON c.id=cg.carrier_id WHERE c.enabled = '1' AND
>>> cg.enabled = '1' AND l.enabled = '1' AND digits IN (16179470890, 1617947089,
>>> 161794708, 16179470, 1617947, 161794, 16179, 1617, 161, 16, 1) AND
>>> CURRENT_TIMESTAMP BETWEEN date_start AND date_end ORDER BY digits DESC,
>>> intralata_rate, rand();
>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:307 Returning Dialstring
>>> [lcr_carrier=carrier1,lcr_rate=0.00000]sofia/gateway/carrier1/16179470890
>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:615 Adding carrier1 to head
>>> of list
>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:307 Returning Dialstring
>>> [lcr_carrier=carrier2,lcr_rate=0.00000]sofia/external/
>>> 06179470890 at proxy.carrier2.net:5060
>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:667 adding carrier2 to end
>>> of list after carrier1
>>>
>>>
>>>  | Digit Match | Carrier  | Rate     | Codec | CID Regexp |
>>> Dialstring
>>> |
>>>  | 1           | carrier1 | 0.00000  |       |            |
>>> [lcr_carrier=carrier1,lcr_rate=0.00000]sofia/gateway/carrier1/16179470890
>>> |
>>>  | 1           | carrier2 | 0.00000  |       |            |
>>> [lcr_carrier=carrier2,lcr_rate=0.00000]sofia/external/
>>> 06179470890 at proxy.carrier2.net:5060 |
>>>
>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:307 Returning Dialstring
>>> [lcr_carrier=carrier1,lcr_rate=0.00000]sofia/gateway/carrier1/16179470890
>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:307 Returning Dialstring
>>> [lcr_carrier=carrier2,lcr_rate=0.00000]sofia/external/
>>> 06179470890 at proxy.carrier2.net:5060
>>>
>>>
>>>
>>>
>>>
>>> freeswitch> lcr 6179470890  default  9785223241
>>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:1329 data passed to lcr is
>>> [6179470890  default  9785223241]
>>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:1365 Set Caller ID to
>>> [9785223241]
>>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:786 intra routing [state:0
>>> lata:0] so rate field is [rate]
>>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:802 we have an event
>>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:826 SQL: SELECT l.digits,
>>> c.carrier_name, l.rate, cg.prefix AS gw_prefix, cg.suffix AS gw_suffix,
>>> l.lead_strip, l.trail_strip, l.prefix, l.suffix , cg.codec , l.cid FROM lcr
>>> l JOIN carriers c ON l.carrier_id=c.id JOIN carrier_gateway cg ON c.id=cg.carrier_id
>>> WHERE c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1' AND digits IN
>>> (6179470890, 617947089, 61794708, 6179470, 617947, 61794, 6179, 617, 61, 6)
>>> AND CURRENT_TIMESTAMP BETWEEN date_start AND date_end ORDER BY digits DESC,
>>> rate, rand();
>>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:307 Returning Dialstring
>>> [lcr_carrier=carrier1,lcr_rate=0.09000]sofia/gateway/carrier1/16179470890
>>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:615 Adding carrier1 to head
>>> of list
>>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:307 Returning Dialstring
>>> [lcr_carrier=carrier1,lcr_rate=0.09000]sofia/gateway/carrier1/16179470890
>>>
>>>
>>>  | Digit Match | Carrier  | Rate     | Codec | CID Regexp |
>>> Dialstring                                                                |
>>>   | 617947      | carrier1 | 0.09000  |       |            |
>>> [lcr_carrier=carrier1,lcr_rate=0.09000]sofia/gateway/carrier1/16179470890 |
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>> On Sat, Jan 30, 2010 at 7:45 PM, Rupa Schomaker < <rupa at rupa.com>
>>> rupa at rupa.com> wrote:
>>>
>>>> turn up logging to debug again, and then reload mod_lcr.  It'll spit out
>>>> a bunch of crap when it tests out each profile you have defined.  Give me
>>>> the full log (here or in <http://pastebin.freeswitch.org>
>>>> pastebin.freeswitch.org).  That may show more useful info as to why
>>>> things are mucked up?
>>>>
>>>>
>>>> On Sat, Jan 30, 2010 at 6:23 PM, Mouncif Benniane <<mouncifbb at gmail.com>
>>>> mouncifbb at gmail.com> wrote:
>>>>
>>>>> yes I use NANPA_STD profile instead of default cause I thought the
>>>>> custom profile was causing issues, but looks like it's returning same
>>>>> results.
>>>>>
>>>>> There is this line in thw wiki:
>>>>> intra lata/state selection is done manually by setting the channel
>>>>> variables *intrastate* or *intralata* to the value *true*.
>>>>>
>>>>> do I have to set these ? if yes how?
>>>>>
>>>>> Thanks
>>>>>
>>>>>
>>>>> On Sat, Jan 30, 2010 at 6:59 PM, Rupa Schomaker < <rupa at rupa.com>
>>>>> rupa at rupa.com> wrote:
>>>>>
>>>>>> Stuff inline.
>>>>>>
>>>>>> On Sat, Jan 30, 2010 at 3:38 PM, Mouncif Benniane <<mouncifbb at gmail.com>
>>>>>> mouncifbb at gmail.com> wrote:
>>>>>>
>>>>>>>  NPANXX,"LATA","OCN","NTER","INTRA"
>>>>>>> 201007,"224","7229","0.0059","0.0127"
>>>>>>> 201040,"224","9206","0.0036","0.0036"
>>>>>>>
>>>>>>
>>>>>> Looks like they give you the LATA and OCN values with the prefix.  We
>>>>>> (should) look that up ourselves.
>>>>>>
>>>>>>
>>>>>>> FreeSWITCH Version 1.0.trunk (16540)
>>>>>>>
>>>>>>>
>>>>>>> Also I noticed the *npa_nxx_ocn* table never get consulted.
>>>>>>>
>>>>>>> I also see this now when making a real call instead of running
>>>>>>> thorugh CLI
>>>>>>>
>>>>>>> EXECUTE sofia/external/6179472456 at 174.x.x.x lcr(16179470890
>>>>>>> NANPA_STD)
>>>>>>> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:1230 *intrastate
>>>>>>> channel var is [undef]*
>>>>>>
>>>>>>
>>>>>> This is fine.  it is a leftover from when you would tell mod_lcr via a
>>>>>> channel var that it should do intrastate.  I later had mod_lcr do the lookup
>>>>>> itself, but we still honor the old var.  There are no channel vars
>>>>>> associated with the cli, so you wouldn't see that msg.
>>>>>>
>>>>>>
>>>>>>>
>>>>>>> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:1233 Select routes based
>>>>>>> on interstate rates
>>>>>>> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:1252 LCR Lookup on
>>>>>>> 16179470893 using profile NANPA_STD
>>>>>>> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:786 intra routing
>>>>>>> [state:0 lata:0] so rate field is [rate]
>>>>>>>
>>>>>>> called number 6179470890  caller ID: 6179472456
>>>>>>>
>>>>>>> any ideas??
>>>>>>>
>>>>>>>
>>>>>> Only thing that jumps out at me.
>>>>>>
>>>>>> The output from lcr_admin show profiles showed only the default one.
>>>>>>  On the dialplan you use the NANPA_STD profile.  Can you check lcr_admin
>>>>>> list and see if that profile is defined and if so if it says it is using the
>>>>>> npanxx table?
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> On Sat, Jan 30, 2010 at 10:02 AM, Rupa Schomaker < <rupa at rupa.com>
>>>>>>> rupa at rupa.com> wrote:
>>>>>>>
>>>>>>>> Something is still missing from the logs. Note the query of the
>>>>>>>> npanxx
>>>>>>>> table, the flags being set, and the rate field being chosen.  Umm..
>>>>>>>> oh, what version of fs are you running?
>>>>>>>>
>>>>>>>> Yes, the npa_nxx_ocn table needs to be loaded up as described in:
>>>>>>>> <http://wiki.freeswitch.org/wiki/Mod_cidlookup#Falling_back_to_.22City_State.22_in_the_absense_of_a_name>
>>>>>>>> http://wiki.freeswitch.org/wiki/Mod_cidlookup#Falling_back_to_.22City_State.22_in_the_absense_of_a_name
>>>>>>>> (there is a link to that from mod_lcr's wiki page).
>>>>>>>>
>>>>>>>> An example from my own setup:
>>>>>>>>
>>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:1384 data passed to lcr
>>>>>>>> is [12148267711 default 12148267712]
>>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:1420 Set Caller ID to
>>>>>>>> [12148267712]
>>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:759 SQL: SELECT
>>>>>>>> 'state',
>>>>>>>> count(DISTINCT state) FROM npa_nxx_company_ocn WHERE (npa=214 AND
>>>>>>>> nxx=826) OR (npa=214 AND nxx=826) UNION SELECT 'lata',
>>>>>>>> count(DISTINCT
>>>>>>>> lata) FROM npa_nxx_company_ocn WHERE (npa=214 AND nxx=826) OR
>>>>>>>> (npa=214
>>>>>>>> AND nxx=826)
>>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:719 Type: lata, Count:
>>>>>>>> 1
>>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:719 Type: state, Count:
>>>>>>>> 1
>>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:809 intra routing
>>>>>>>> [state:1 lata:1] so rate field is [intralata_rate]
>>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:825 we have an event
>>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:849 SQL:  SELECT
>>>>>>>> l.digits
>>>>>>>> AS lcr_digits,         c.carrier_name AS lcr_carrier_name,
>>>>>>>> l.intralata_rate as lcr_rate_field,        cg.prefix AS
>>>>>>>> lcr_gw_prefix,
>>>>>>>> cg.suffix AS lcr_gw_suffix,        l.lead_strip AS lcr_lead_strip,
>>>>>>>> l.trail_strip AS lcr_trail_strip,        l.prefix AS lcr_prefix,
>>>>>>>> l.suffix AS lcr_suffix,        cg.codec AS lcr_codec, l.cid AS
>>>>>>>> lcr_cid,        'carriers' AS lcr_limit_realm, c.carrier_name AS
>>>>>>>> lcr_limit_id, 5 AS lcr_limit_max   FROM lcr l      JOIN carriers c
>>>>>>>> ON
>>>>>>>> l.carrier_id=c.id      JOIN carrier_gateway cg ON c.id
>>>>>>>> =cg.carrier_id
>>>>>>>>  WHERE c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1'
>>>>>>>> AND digits_prefix @> '12148267711'      AND CURRENT_TIMESTAMP
>>>>>>>> BETWEEN
>>>>>>>> date_start AND date_end   ORDER BY digits DESC, intralata_rate,
>>>>>>>> random();
>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning
>>>>>>>> Dialstring
>>>>>>>>
>>>>>>>> [lcr_carrier=grnvoip_us,lcr_rate=0.00591]sofia/gateway/grnvoip/XXXX12148267711
>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:638 Adding grnvoip_us
>>>>>>>> to
>>>>>>>> head of list
>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning
>>>>>>>> Dialstring
>>>>>>>>
>>>>>>>> [lcr_carrier=teliax,lcr_rate=0.01000]sofia/gateway/teliax_atl/12148267711
>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:660 Adding teliax to
>>>>>>>> end of list
>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning
>>>>>>>> Dialstring
>>>>>>>>
>>>>>>>> [lcr_carrier=teliax,lcr_rate=0.01000]sofia/gateway/teliax/12148267711
>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:660 Adding teliax to
>>>>>>>> end of list
>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning
>>>>>>>> Dialstring
>>>>>>>>
>>>>>>>> [lcr_carrier=vitelity,lcr_rate=0.01440]sofia/gateway/vitelity/12148267711
>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:660 Adding vitelity to
>>>>>>>> end of list
>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning
>>>>>>>> Dialstring
>>>>>>>>
>>>>>>>> [lcr_carrier=grnvoip_std,lcr_rate=0.01500]sofia/gateway/grnvoip/YYYY12148267711
>>>>>>>> [...]
>>>>>>>>
>>>>>>>> On Fri, Jan 29, 2010 at 10:42 PM, Mouncif Benniane <<mouncifbb at gmail.com>
>>>>>>>> mouncifbb at gmail.com> wrote:
>>>>>>>> > Also the Provider has presented the rates in this format?
>>>>>>>> > NPANXXLATA OCN INTER INTRA
>>>>>>>> >
>>>>>>>> >
>>>>>>>> > On Fri, Jan 29, 2010 at 11:30 PM, Mouncif Benniane <<mouncifbb at gmail.com>
>>>>>>>> mouncifbb at gmail.com>
>>>>>>>> > wrote:
>>>>>>>> >>
>>>>>>>> >> Tried it and it's not giving me intralata  instead I get
>>>>>>>> interstate, does
>>>>>>>> >> the npa_nxx_company_ocn table needs to be used in this case?,
>>>>>>>> also do I have
>>>>>>>> >> to have the rate field in lcr table?
>>>>>>>> >>
>>>>>>>> >> lcr 617642  default 6176421212
>>>>>>>> >>  | Digit Match | Carrier  | Rate     | Codec | CID Regexp |
>>>>>>>> Dialstring
>>>>>>>> >>                                                        |
>>>>>>>> >>  | 617642      | carrier1 | 0.00500  |       |            |
>>>>>>>> >>
>>>>>>>> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642 |
>>>>>>>> >>
>>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:1329 data passed to
>>>>>>>> lcr is
>>>>>>>> >> [617642  default 6176421212]
>>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:1365 Set Caller ID
>>>>>>>> to
>>>>>>>> >> [6176421212]
>>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:786 intra routing
>>>>>>>> [state:0
>>>>>>>> >> lata:0] so rate field is [rate]
>>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:802 we have an event
>>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:826 SQL: SELECT
>>>>>>>> l.digits,
>>>>>>>> >> c.carrier_name, l.rate, cg.prefix AS gw_prefix, cg.suffix AS
>>>>>>>> gw_suffix,
>>>>>>>> >> l.lead_strip, l.trail_strip, l.prefix, l.suffix , cg.codec ,
>>>>>>>> l.cid FROM lcr
>>>>>>>> >> l JOIN carriers c ON l.carrier_id=c.id JOIN carrier_gateway cg
>>>>>>>> ON
>>>>>>>> >> c.id=cg.carrier_id WHERE c.enabled = '1' AND cg.enabled = '1'
>>>>>>>> AND l.enabled
>>>>>>>> >> = '1' AND digits IN (617642, 61764, 6176, 617, 61, 6) AND
>>>>>>>> CURRENT_TIMESTAMP
>>>>>>>> >> BETWEEN date_start AND date_end ORDER BY digits DESC,  rate,
>>>>>>>> rand();
>>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:307 Returning
>>>>>>>> Dialstring
>>>>>>>> >>
>>>>>>>> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642
>>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:615 Adding carrier1
>>>>>>>> to head
>>>>>>>> >> of list
>>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:307 Returning
>>>>>>>> Dialstring
>>>>>>>> >>
>>>>>>>> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642
>>>>>>>> >>
>>>>>>>> >> Thank you Rupa!
>>>>>>>> >>
>>>>>>>> >> On Fri, Jan 29, 2010 at 7:37 PM, Rupa Schomaker <<rupa at rupa.com>
>>>>>>>> rupa at rupa.com> wrote:
>>>>>>>> >>>
>>>>>>>> >>> turn console logging up to debug and redo the lcr lookup.  The
>>>>>>>> sql
>>>>>>>> >>> statements along with status info will show up.  This should
>>>>>>>> give
>>>>>>>> >>> enough information to debug what is happening.
>>>>>>>> >>>
>>>>>>>> >>> I'm assuming the npanxx table is actually populated and not just
>>>>>>>> >>> existing?
>>>>>>>> >>>
>>>>>>>> >>> When doing the lookup from the cli you have to tell lcr what CID
>>>>>>>> to
>>>>>>>> >>> use (remember, it is relative to the src/dest number).  I'm
>>>>>>>> pretty
>>>>>>>> >>> sure you get something on the console log when you don't specify
>>>>>>>> a CID
>>>>>>>> >>> when using the commandline.  Anyway:
>>>>>>>> >>>
>>>>>>>> >>> lcr 617642  default 6176421212
>>>>>>>> >>>
>>>>>>>> >>> should give you intralata.
>>>>>>>> >>>
>>>>>>>> >>> Note that the definition of intralata doesn't mean "local" for
>>>>>>>> some
>>>>>>>> >>> providers.  Some providers define local to "same ratecenter"
>>>>>>>> which is
>>>>>>>> >>> even more restrictive.
>>>>>>>> >>>
>>>>>>>> >>> On Fri, Jan 29, 2010 at 4:43 PM, Mouncif Benniane <<mouncifbb at gmail.com>
>>>>>>>> mouncifbb at gmail.com>
>>>>>>>> >>> wrote:
>>>>>>>> >>> > i can't make use of mod_lcr using Intra/Interstate rating, I
>>>>>>>> am using
>>>>>>>> >>> > svn: FreeSWITCH Version 1.0.trunk (16517)
>>>>>>>> >>> >
>>>>>>>> >>> > lcr mysql table structure:
>>>>>>>> >>> > CREATE TABLE `lcr` (
>>>>>>>> >>> >   `id` INT(11) NOT NULL AUTO_INCREMENT,
>>>>>>>> >>> >   `digits` VARCHAR(15) DEFAULT NULL,
>>>>>>>> >>> >   `rate` FLOAT(11,5) DEFAULT NULL,
>>>>>>>> >>> >   `intrastate_rate` FLOAT(11,5) DEFAULT NULL,
>>>>>>>> >>> >   `intralata_rate` FLOAT(11,5) DEFAULT NULL,
>>>>>>>> >>> >   `carrier_id` INT(11) NOT NULL,
>>>>>>>> >>> >   `lead_strip` INT(11) NOT NULL,
>>>>>>>> >>> >   `trail_strip` INT(11) NOT NULL,
>>>>>>>> >>> >   `prefix` VARCHAR(16) NOT NULL,
>>>>>>>> >>> >   `suffix` VARCHAR(16) NOT NULL,
>>>>>>>> >>> >   `lcr_profile` VARCHAR(32) DEFAULT NULL,
>>>>>>>> >>> >   `date_start` DATETIME NOT NULL DEFAULT '1970-01-01
>>>>>>>> 00:00:00',
>>>>>>>> >>> >   `date_end` DATETIME NOT NULL DEFAULT '2030-12-31 00:00:00',
>>>>>>>> >>> >   `quality` FLOAT(10,6) NOT NULL,
>>>>>>>> >>> >   `reliability` FLOAT(10,6) NOT NULL,
>>>>>>>> >>> >   `cid` VARCHAR(32) NOT NULL DEFAULT '',
>>>>>>>> >>> >   `enabled` TINYINT(1) NOT NULL DEFAULT '1',
>>>>>>>> >>> >   PRIMARY KEY  (`id`),
>>>>>>>> >>> >   KEY `carrier_id` (`carrier_id`),
>>>>>>>> >>> >   KEY `digits` (`digits`),
>>>>>>>> >>> >   KEY `lcr_profile` (`lcr_profile`),
>>>>>>>> >>> >   KEY `digits_profile_cid_rate` USING BTREE (`digits`),
>>>>>>>> >>> >   CONSTRAINT `carrier_id` FOREIGN KEY (`carrier_id`)
>>>>>>>> REFERENCES
>>>>>>>> >>> > `carriers`
>>>>>>>> >>> > (`id`) ON DELETE CASCADE ON UPDATE CASCADE
>>>>>>>> >>> > ) ENGINE=INNODB AUTO_INCREMENT=6 DEFAULT CHARSET=latin1
>>>>>>>> >>> >
>>>>>>>> >>> >
>>>>>>>> >>> > lcr_admin show profiles
>>>>>>>> >>> > Name:           default
>>>>>>>> >>> >  custom sql:    SELECT l.digits, c.carrier_name,
>>>>>>>> l.${lcr_rate_field},
>>>>>>>> >>> > cg.prefix AS gw_prefix, cg.suffix AS gw_suffix, l.lead_strip,
>>>>>>>> >>> > l.trail_strip,
>>>>>>>> >>> > l.prefix, l.suffix , cg.codec , l.cid FROM lcr l JOIN carriers
>>>>>>>> c ON
>>>>>>>> >>> > l.carrier_id=c.id JOIN carrier_gateway cg ON c.id=cg.carrier_id
>>>>>>>> WHERE
>>>>>>>> >>> > c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1' AND
>>>>>>>> digits IN
>>>>>>>> >>> > (${lcr_query_expanded_digits}) AND CURRENT_TIMESTAMP BETWEEN
>>>>>>>> date_start
>>>>>>>> >>> > AND
>>>>>>>> >>> > date_end ORDER BY digits DESC,  ${lcr_rate_field},  quality
>>>>>>>> DESC,
>>>>>>>> >>> >  reliability DESC, rand();
>>>>>>>> >>> >  has %:         false
>>>>>>>> >>> >  has vars:      true
>>>>>>>> >>> >  has intrastate:        true
>>>>>>>> >>> >  has intralata: true
>>>>>>>> >>> >  has npanxx:    true
>>>>>>>> >>> >  Reorder rate:  enabled
>>>>>>>> >>> >  Info in headers:       disabled
>>>>>>>> >>> >  Quote IN() List:       disabled
>>>>>>>> >>> >
>>>>>>>> >>> >
>>>>>>>> >>> >
>>>>>>>> >>> > lcr 617642  default  returns rate from the rate field table
>>>>>>>> and not
>>>>>>>> >>> > intra/inter state fields rates.
>>>>>>>> >>> >
>>>>>>>> >>> > Any ideas? thanks!
>>>>>>>> >>> >
>>>>>>>> >>> >
>>>>>>>> >>> >
>>>>>>>> >>> >
>>>>>>>> >>> > _______________________________________________
>>>>>>>> >>> > FreeSWITCH-users mailing list
>>>>>>>> >>> > <FreeSWITCH-users at lists.freeswitch.org>
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> >>> >
>>>>>>>> <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> >>> >
>>>>>>>> >>> > UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> >>> > <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>>> >>> >
>>>>>>>> >>> >
>>>>>>>> >>>
>>>>>>>> >>>
>>>>>>>> >>>
>>>>>>>> >>> --
>>>>>>>> >>> -Rupa
>>>>>>>> >>>
>>>>>>>> >>> _______________________________________________
>>>>>>>> >>> FreeSWITCH-users mailing list
>>>>>>>> >>> <FreeSWITCH-users at lists.freeswitch.org>
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> >>> <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> >>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> >>> <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>>> >>
>>>>>>>> >
>>>>>>>> >
>>>>>>>> > _______________________________________________
>>>>>>>> > FreeSWITCH-users mailing list
>>>>>>>> > <FreeSWITCH-users at lists.freeswitch.org>
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> > <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> > UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> > <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>>> >
>>>>>>>> >
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> --
>>>>>>>> -Rupa
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>> -Rupa
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>>
>>>> --
>>>> -Rupa
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>> FreeSWITCH-users at lists.freeswitch.org
>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> -Rupa
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100201/773e798f/attachment-0002.html 

From paul.gore.j at gmail.com  Mon Feb  1 13:07:26 2010
From: paul.gore.j at gmail.com (paul gore)
Date: Mon, 1 Feb 2010 16:07:26 -0500
Subject: [Freeswitch-users] Logging question
Message-ID: <da4ab26e1002011307n6b10e97tea3e281cbff90e13@mail.gmail.com>

I tried "sofia profile internal siptrace on" from console and got SIP trace
on the console,
but nothing in logs still.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100201/47e22d6f/attachment-0002.html 

From rupa at rupa.com  Mon Feb  1 14:39:03 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Mon, 1 Feb 2010 16:39:03 -0600
Subject: [Freeswitch-users] mod_lcr problem
In-Reply-To: <ade741d61002011225t15d66333v69f31b0572880de9@mail.gmail.com>
References: <ade741d61001291443l6134a3achf28be64fc7746c8e@mail.gmail.com>
	<ade741d61001301338x8014e86t9c9094403f3361f@mail.gmail.com>
	<b39273531001301559h60668746kd5c63ada240513f1@mail.gmail.com>
	<ade741d61001301623w70d7e449i44a9f1e9fb7beae4@mail.gmail.com>
	<b39273531001301645w10446308o3578479dcbf2da2@mail.gmail.com>
	<ade741d61001302057w6ce72355iee6e8b4a2872fc81@mail.gmail.com>
	<b39273531001310532y1de53609rf474755cec04f218@mail.gmail.com>
	<9AB8B620-5A69-4F08-B62C-FFEB03FC6762@gmail.com>
	<b39273531001311507hb1bfe02m8d370443b882ef81@mail.gmail.com>
	<ade741d61002011225t15d66333v69f31b0572880de9@mail.gmail.com>
Message-ID: <b39273531002011439h2a2ce74ahca048432579129c1@mail.gmail.com>

yes, otherwise you'll have issues when you load your international rates in
the same table.

On Mon, Feb 1, 2010 at 2:25 PM, Mouncif Benniane <mouncifbb at gmail.com>wrote:

> So I have to alter my LCR table to look like:
>
> NPANXX,"LATA","OCN","NTER","INTRA"  1201007,"224","7229","0.0059","0.0127"
> 1201040,"224","9206","0.0036","0.0036"
>
>
> instead of:
>
>
>
> NPANXX,"LATA","OCN","NTER","INTRA"  201007,"224","7229","0.0059","0.0127"
> 201040,"224","9206","0.0036","0.0036"
>
>
> On Sun, Jan 31, 2010 at 6:07 PM, Rupa Schomaker <rupa at rupa.com> wrote:
>
>> Yes, you need to normalize the values passed to lcr.  Otherwise, how could
>> it work?
>>
>> You can normalize the CID by matching and adding a 1 for 10 digit #s, or
>> removing the leading + or other things you might need then setting it back
>> to the profile using the set_profile_var app (
>> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_set_profile_var).
>>  (mod_cidlookup will set it after doing a #->name/area lookup - but for now
>> you can set it yourself)
>>
>> You can normalize the DID by doing similar matching rules as above and
>> then transfering to that normalized DID for the rest of your call plan
>> processing.
>>
>> I'm pretty sure mod_cidlookup has an example of normalizing... yeah:
>> http://wiki.freeswitch.org/wiki/Mod_cidlookup#Dialplan_Application
>>
>> On Sun, Jan 31, 2010 at 4:18 PM, Mouncifbb <mouncifbb at gmail.com> wrote:
>>
>>> So the CID must have 1 at front also? Usually people
>>> Send only npa and nxx ex 6176427788 7817612233
>>> Do I need to alter it?
>>>
>>> Sent from my iPhone
>>>
>>> On Jan 31, 2010, at 8:32 AM, Rupa Schomaker <rupa at rupa.com> wrote:
>>>
>>>
>>>
>>> On Sat, Jan 30, 2010 at 10:57 PM, Mouncif Benniane <<mouncifbb at gmail.com>
>>> mouncifbb at gmail.com> wrote:
>>>
>>>> OK going back to use default profile to keep things simple below 2
>>>> results
>>>>
>>>> Using:
>>>>
>>>> lcr 16179470890  default  19785223241  ( this one consult
>>>> npa_nxx_company_ocn)
>>>>
>>>> lcr 6179470890  default  9785223241 ( this one don't!! )
>>>>
>>>>
>>>>
>>> Oh, right!  mod_lcr really expects you to normalize your prefix to e164
>>> format.  I thought there was discussion about this in the wiki, but maybe
>>> not.  For simple prefix matching it doesn't matter, but for things that make
>>> decisions based on the # (like the lata/state stuff) it does.
>>>
>>> npanxx lookup only makes sense for NANPA numbers.  NANPA numbers have a
>>> country code of "1" and a total length of 11 (including the 1).
>>>
>>> This is the only rational way to do it when you have a rate table with
>>> both domestic (NANPA) and international prefixes.
>>>
>>>
>>>> freeswitch> lcr 16179470890  default  19785223241
>>>> 2010-01-30 23:53:34.681842 [DEBUG] mod_lcr.c:1329 data passed to lcr is
>>>> [16179470890  default  19785223241]
>>>> 2010-01-30 23:53:34.681842 [DEBUG] mod_lcr.c:1365 Set Caller ID to
>>>> [19785223241]
>>>> 2010-01-30 23:53:34.681842 [DEBUG] mod_lcr.c:736 SQL: SELECT 'state',
>>>> count(DISTINCT state) FROM npa_nxx_company_ocn WHERE (npa=617 AND nxx=947)
>>>> OR (npa=978 AND nxx=522) UNION SELECT 'lata', count(DISTINCT lata) FROM
>>>> npa_nxx_company_ocn WHERE (npa=617 AND nxx=947) OR (npa=978 AND nxx=522)
>>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:696 Type: state, Count: 1
>>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:696 Type: lata, Count: 1
>>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:786 intra routing [state:1
>>>> lata:1] so rate field is [intralata_rate]
>>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:802 we have an event
>>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:826 SQL: SELECT l.digits,
>>>> c.carrier_name, l.intralata_rate, cg.prefix AS gw_prefix, cg.suffix AS
>>>> gw_suffix, l.lead_strip, l.trail_strip, l.prefix, l.suffix , cg.codec ,
>>>> l.cid FROM lcr l JOIN carriers c ON l.carrier_id=c.id JOIN
>>>> carrier_gateway cg ON c.id=cg.carrier_id WHERE c.enabled = '1' AND
>>>> cg.enabled = '1' AND l.enabled = '1' AND digits IN (16179470890, 1617947089,
>>>> 161794708, 16179470, 1617947, 161794, 16179, 1617, 161, 16, 1) AND
>>>> CURRENT_TIMESTAMP BETWEEN date_start AND date_end ORDER BY digits DESC,
>>>> intralata_rate, rand();
>>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:307 Returning Dialstring
>>>> [lcr_carrier=carrier1,lcr_rate=0.00000]sofia/gateway/carrier1/16179470890
>>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:615 Adding carrier1 to head
>>>> of list
>>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:307 Returning Dialstring
>>>> [lcr_carrier=carrier2,lcr_rate=0.00000]sofia/external/
>>>> 06179470890 at proxy.carrier2.net:5060
>>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:667 adding carrier2 to end
>>>> of list after carrier1
>>>>
>>>>
>>>>  | Digit Match | Carrier  | Rate     | Codec | CID Regexp |
>>>> Dialstring
>>>> |
>>>>  | 1           | carrier1 | 0.00000  |       |            |
>>>> [lcr_carrier=carrier1,lcr_rate=0.00000]sofia/gateway/carrier1/16179470890
>>>> |
>>>>  | 1           | carrier2 | 0.00000  |       |            |
>>>> [lcr_carrier=carrier2,lcr_rate=0.00000]sofia/external/
>>>> 06179470890 at proxy.carrier2.net:5060 |
>>>>
>>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:307 Returning Dialstring
>>>> [lcr_carrier=carrier1,lcr_rate=0.00000]sofia/gateway/carrier1/16179470890
>>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:307 Returning Dialstring
>>>> [lcr_carrier=carrier2,lcr_rate=0.00000]sofia/external/
>>>> 06179470890 at proxy.carrier2.net:5060
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> freeswitch> lcr 6179470890  default  9785223241
>>>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:1329 data passed to lcr is
>>>> [6179470890  default  9785223241]
>>>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:1365 Set Caller ID to
>>>> [9785223241]
>>>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:786 intra routing [state:0
>>>> lata:0] so rate field is [rate]
>>>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:802 we have an event
>>>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:826 SQL: SELECT l.digits,
>>>> c.carrier_name, l.rate, cg.prefix AS gw_prefix, cg.suffix AS gw_suffix,
>>>> l.lead_strip, l.trail_strip, l.prefix, l.suffix , cg.codec , l.cid FROM lcr
>>>> l JOIN carriers c ON l.carrier_id=c.id JOIN carrier_gateway cg ON c.id=cg.carrier_id
>>>> WHERE c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1' AND digits IN
>>>> (6179470890, 617947089, 61794708, 6179470, 617947, 61794, 6179, 617, 61, 6)
>>>> AND CURRENT_TIMESTAMP BETWEEN date_start AND date_end ORDER BY digits DESC,
>>>> rate, rand();
>>>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:307 Returning Dialstring
>>>> [lcr_carrier=carrier1,lcr_rate=0.09000]sofia/gateway/carrier1/16179470890
>>>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:615 Adding carrier1 to head
>>>> of list
>>>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:307 Returning Dialstring
>>>> [lcr_carrier=carrier1,lcr_rate=0.09000]sofia/gateway/carrier1/16179470890
>>>>
>>>>
>>>>  | Digit Match | Carrier  | Rate     | Codec | CID Regexp |
>>>> Dialstring                                                                |
>>>>   | 617947      | carrier1 | 0.09000  |       |            |
>>>> [lcr_carrier=carrier1,lcr_rate=0.09000]sofia/gateway/carrier1/16179470890 |
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> On Sat, Jan 30, 2010 at 7:45 PM, Rupa Schomaker < <rupa at rupa.com>
>>>> rupa at rupa.com> wrote:
>>>>
>>>>> turn up logging to debug again, and then reload mod_lcr.  It'll spit
>>>>> out a bunch of crap when it tests out each profile you have defined.  Give
>>>>> me the full log (here or in <http://pastebin.freeswitch.org>
>>>>> pastebin.freeswitch.org).  That may show more useful info as to why
>>>>> things are mucked up?
>>>>>
>>>>>
>>>>> On Sat, Jan 30, 2010 at 6:23 PM, Mouncif Benniane <<mouncifbb at gmail.com>
>>>>> mouncifbb at gmail.com> wrote:
>>>>>
>>>>>> yes I use NANPA_STD profile instead of default cause I thought the
>>>>>> custom profile was causing issues, but looks like it's returning same
>>>>>> results.
>>>>>>
>>>>>> There is this line in thw wiki:
>>>>>> intra lata/state selection is done manually by setting the channel
>>>>>> variables *intrastate* or *intralata* to the value *true*.
>>>>>>
>>>>>> do I have to set these ? if yes how?
>>>>>>
>>>>>> Thanks
>>>>>>
>>>>>>
>>>>>> On Sat, Jan 30, 2010 at 6:59 PM, Rupa Schomaker < <rupa at rupa.com>
>>>>>> rupa at rupa.com> wrote:
>>>>>>
>>>>>>> Stuff inline.
>>>>>>>
>>>>>>> On Sat, Jan 30, 2010 at 3:38 PM, Mouncif Benniane <<mouncifbb at gmail.com>
>>>>>>> mouncifbb at gmail.com> wrote:
>>>>>>>
>>>>>>>>  NPANXX,"LATA","OCN","NTER","INTRA"
>>>>>>>> 201007,"224","7229","0.0059","0.0127"
>>>>>>>> 201040,"224","9206","0.0036","0.0036"
>>>>>>>>
>>>>>>>
>>>>>>> Looks like they give you the LATA and OCN values with the prefix.  We
>>>>>>> (should) look that up ourselves.
>>>>>>>
>>>>>>>
>>>>>>>> FreeSWITCH Version 1.0.trunk (16540)
>>>>>>>>
>>>>>>>>
>>>>>>>> Also I noticed the *npa_nxx_ocn* table never get consulted.
>>>>>>>>
>>>>>>>> I also see this now when making a real call instead of running
>>>>>>>> thorugh CLI
>>>>>>>>
>>>>>>>> EXECUTE sofia/external/6179472456 at 174.x.x.x lcr(16179470890
>>>>>>>> NANPA_STD)
>>>>>>>> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:1230 *intrastate
>>>>>>>> channel var is [undef]*
>>>>>>>
>>>>>>>
>>>>>>> This is fine.  it is a leftover from when you would tell mod_lcr via
>>>>>>> a channel var that it should do intrastate.  I later had mod_lcr do the
>>>>>>> lookup itself, but we still honor the old var.  There are no channel vars
>>>>>>> associated with the cli, so you wouldn't see that msg.
>>>>>>>
>>>>>>>
>>>>>>>>
>>>>>>>> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:1233 Select routes
>>>>>>>> based on interstate rates
>>>>>>>> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:1252 LCR Lookup on
>>>>>>>> 16179470893 using profile NANPA_STD
>>>>>>>> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:786 intra routing
>>>>>>>> [state:0 lata:0] so rate field is [rate]
>>>>>>>>
>>>>>>>> called number 6179470890  caller ID: 6179472456
>>>>>>>>
>>>>>>>> any ideas??
>>>>>>>>
>>>>>>>>
>>>>>>> Only thing that jumps out at me.
>>>>>>>
>>>>>>> The output from lcr_admin show profiles showed only the default one.
>>>>>>>  On the dialplan you use the NANPA_STD profile.  Can you check lcr_admin
>>>>>>> list and see if that profile is defined and if so if it says it is using the
>>>>>>> npanxx table?
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> On Sat, Jan 30, 2010 at 10:02 AM, Rupa Schomaker < <rupa at rupa.com>
>>>>>>>> rupa at rupa.com> wrote:
>>>>>>>>
>>>>>>>>> Something is still missing from the logs. Note the query of the
>>>>>>>>> npanxx
>>>>>>>>> table, the flags being set, and the rate field being chosen.  Umm..
>>>>>>>>> oh, what version of fs are you running?
>>>>>>>>>
>>>>>>>>> Yes, the npa_nxx_ocn table needs to be loaded up as described in:
>>>>>>>>> <http://wiki.freeswitch.org/wiki/Mod_cidlookup#Falling_back_to_.22City_State.22_in_the_absense_of_a_name>
>>>>>>>>> http://wiki.freeswitch.org/wiki/Mod_cidlookup#Falling_back_to_.22City_State.22_in_the_absense_of_a_name
>>>>>>>>> (there is a link to that from mod_lcr's wiki page).
>>>>>>>>>
>>>>>>>>> An example from my own setup:
>>>>>>>>>
>>>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:1384 data passed to
>>>>>>>>> lcr
>>>>>>>>> is [12148267711 default 12148267712]
>>>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:1420 Set Caller ID to
>>>>>>>>> [12148267712]
>>>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:759 SQL: SELECT
>>>>>>>>> 'state',
>>>>>>>>> count(DISTINCT state) FROM npa_nxx_company_ocn WHERE (npa=214 AND
>>>>>>>>> nxx=826) OR (npa=214 AND nxx=826) UNION SELECT 'lata',
>>>>>>>>> count(DISTINCT
>>>>>>>>> lata) FROM npa_nxx_company_ocn WHERE (npa=214 AND nxx=826) OR
>>>>>>>>> (npa=214
>>>>>>>>> AND nxx=826)
>>>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:719 Type: lata, Count:
>>>>>>>>> 1
>>>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:719 Type: state,
>>>>>>>>> Count: 1
>>>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:809 intra routing
>>>>>>>>> [state:1 lata:1] so rate field is [intralata_rate]
>>>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:825 we have an event
>>>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:849 SQL:  SELECT
>>>>>>>>> l.digits
>>>>>>>>> AS lcr_digits,         c.carrier_name AS lcr_carrier_name,
>>>>>>>>> l.intralata_rate as lcr_rate_field,        cg.prefix AS
>>>>>>>>> lcr_gw_prefix,
>>>>>>>>> cg.suffix AS lcr_gw_suffix,        l.lead_strip AS lcr_lead_strip,
>>>>>>>>> l.trail_strip AS lcr_trail_strip,        l.prefix AS lcr_prefix,
>>>>>>>>> l.suffix AS lcr_suffix,        cg.codec AS lcr_codec, l.cid AS
>>>>>>>>> lcr_cid,        'carriers' AS lcr_limit_realm, c.carrier_name AS
>>>>>>>>> lcr_limit_id, 5 AS lcr_limit_max   FROM lcr l      JOIN carriers c
>>>>>>>>> ON
>>>>>>>>> l.carrier_id=c.id      JOIN carrier_gateway cg ON c.id
>>>>>>>>> =cg.carrier_id
>>>>>>>>>  WHERE c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1'
>>>>>>>>> AND digits_prefix @> '12148267711'      AND CURRENT_TIMESTAMP
>>>>>>>>> BETWEEN
>>>>>>>>> date_start AND date_end   ORDER BY digits DESC, intralata_rate,
>>>>>>>>> random();
>>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning
>>>>>>>>> Dialstring
>>>>>>>>>
>>>>>>>>> [lcr_carrier=grnvoip_us,lcr_rate=0.00591]sofia/gateway/grnvoip/XXXX12148267711
>>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:638 Adding grnvoip_us
>>>>>>>>> to
>>>>>>>>> head of list
>>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning
>>>>>>>>> Dialstring
>>>>>>>>>
>>>>>>>>> [lcr_carrier=teliax,lcr_rate=0.01000]sofia/gateway/teliax_atl/12148267711
>>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:660 Adding teliax to
>>>>>>>>> end of list
>>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning
>>>>>>>>> Dialstring
>>>>>>>>>
>>>>>>>>> [lcr_carrier=teliax,lcr_rate=0.01000]sofia/gateway/teliax/12148267711
>>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:660 Adding teliax to
>>>>>>>>> end of list
>>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning
>>>>>>>>> Dialstring
>>>>>>>>>
>>>>>>>>> [lcr_carrier=vitelity,lcr_rate=0.01440]sofia/gateway/vitelity/12148267711
>>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:660 Adding vitelity to
>>>>>>>>> end of list
>>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning
>>>>>>>>> Dialstring
>>>>>>>>>
>>>>>>>>> [lcr_carrier=grnvoip_std,lcr_rate=0.01500]sofia/gateway/grnvoip/YYYY12148267711
>>>>>>>>> [...]
>>>>>>>>>
>>>>>>>>> On Fri, Jan 29, 2010 at 10:42 PM, Mouncif Benniane <<mouncifbb at gmail.com>
>>>>>>>>> mouncifbb at gmail.com> wrote:
>>>>>>>>> > Also the Provider has presented the rates in this format?
>>>>>>>>> > NPANXXLATA OCN INTER INTRA
>>>>>>>>> >
>>>>>>>>> >
>>>>>>>>> > On Fri, Jan 29, 2010 at 11:30 PM, Mouncif Benniane <<mouncifbb at gmail.com>
>>>>>>>>> mouncifbb at gmail.com>
>>>>>>>>> > wrote:
>>>>>>>>> >>
>>>>>>>>> >> Tried it and it's not giving me intralata  instead I get
>>>>>>>>> interstate, does
>>>>>>>>> >> the npa_nxx_company_ocn table needs to be used in this case?,
>>>>>>>>> also do I have
>>>>>>>>> >> to have the rate field in lcr table?
>>>>>>>>> >>
>>>>>>>>> >> lcr 617642  default 6176421212
>>>>>>>>> >>  | Digit Match | Carrier  | Rate     | Codec | CID Regexp |
>>>>>>>>> Dialstring
>>>>>>>>> >>                                                        |
>>>>>>>>> >>  | 617642      | carrier1 | 0.00500  |       |            |
>>>>>>>>> >>
>>>>>>>>> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642 |
>>>>>>>>> >>
>>>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:1329 data passed to
>>>>>>>>> lcr is
>>>>>>>>> >> [617642  default 6176421212]
>>>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:1365 Set Caller ID
>>>>>>>>> to
>>>>>>>>> >> [6176421212]
>>>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:786 intra routing
>>>>>>>>> [state:0
>>>>>>>>> >> lata:0] so rate field is [rate]
>>>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:802 we have an
>>>>>>>>> event
>>>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:826 SQL: SELECT
>>>>>>>>> l.digits,
>>>>>>>>> >> c.carrier_name, l.rate, cg.prefix AS gw_prefix, cg.suffix AS
>>>>>>>>> gw_suffix,
>>>>>>>>> >> l.lead_strip, l.trail_strip, l.prefix, l.suffix , cg.codec ,
>>>>>>>>> l.cid FROM lcr
>>>>>>>>> >> l JOIN carriers c ON l.carrier_id=c.id JOIN carrier_gateway cg
>>>>>>>>> ON
>>>>>>>>> >> c.id=cg.carrier_id WHERE c.enabled = '1' AND cg.enabled = '1'
>>>>>>>>> AND l.enabled
>>>>>>>>> >> = '1' AND digits IN (617642, 61764, 6176, 617, 61, 6) AND
>>>>>>>>> CURRENT_TIMESTAMP
>>>>>>>>> >> BETWEEN date_start AND date_end ORDER BY digits DESC,  rate,
>>>>>>>>> rand();
>>>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:307 Returning
>>>>>>>>> Dialstring
>>>>>>>>> >>
>>>>>>>>> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642
>>>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:615 Adding carrier1
>>>>>>>>> to head
>>>>>>>>> >> of list
>>>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:307 Returning
>>>>>>>>> Dialstring
>>>>>>>>> >>
>>>>>>>>> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642
>>>>>>>>> >>
>>>>>>>>> >> Thank you Rupa!
>>>>>>>>> >>
>>>>>>>>> >> On Fri, Jan 29, 2010 at 7:37 PM, Rupa Schomaker <<rupa at rupa.com>
>>>>>>>>> rupa at rupa.com> wrote:
>>>>>>>>> >>>
>>>>>>>>> >>> turn console logging up to debug and redo the lcr lookup.  The
>>>>>>>>> sql
>>>>>>>>> >>> statements along with status info will show up.  This should
>>>>>>>>> give
>>>>>>>>> >>> enough information to debug what is happening.
>>>>>>>>> >>>
>>>>>>>>> >>> I'm assuming the npanxx table is actually populated and not
>>>>>>>>> just
>>>>>>>>> >>> existing?
>>>>>>>>> >>>
>>>>>>>>> >>> When doing the lookup from the cli you have to tell lcr what
>>>>>>>>> CID to
>>>>>>>>> >>> use (remember, it is relative to the src/dest number).  I'm
>>>>>>>>> pretty
>>>>>>>>> >>> sure you get something on the console log when you don't
>>>>>>>>> specify a CID
>>>>>>>>> >>> when using the commandline.  Anyway:
>>>>>>>>> >>>
>>>>>>>>> >>> lcr 617642  default 6176421212
>>>>>>>>> >>>
>>>>>>>>> >>> should give you intralata.
>>>>>>>>> >>>
>>>>>>>>> >>> Note that the definition of intralata doesn't mean "local" for
>>>>>>>>> some
>>>>>>>>> >>> providers.  Some providers define local to "same ratecenter"
>>>>>>>>> which is
>>>>>>>>> >>> even more restrictive.
>>>>>>>>> >>>
>>>>>>>>> >>> On Fri, Jan 29, 2010 at 4:43 PM, Mouncif Benniane <<mouncifbb at gmail.com>
>>>>>>>>> mouncifbb at gmail.com>
>>>>>>>>> >>> wrote:
>>>>>>>>> >>> > i can't make use of mod_lcr using Intra/Interstate rating, I
>>>>>>>>> am using
>>>>>>>>> >>> > svn: FreeSWITCH Version 1.0.trunk (16517)
>>>>>>>>> >>> >
>>>>>>>>> >>> > lcr mysql table structure:
>>>>>>>>> >>> > CREATE TABLE `lcr` (
>>>>>>>>> >>> >   `id` INT(11) NOT NULL AUTO_INCREMENT,
>>>>>>>>> >>> >   `digits` VARCHAR(15) DEFAULT NULL,
>>>>>>>>> >>> >   `rate` FLOAT(11,5) DEFAULT NULL,
>>>>>>>>> >>> >   `intrastate_rate` FLOAT(11,5) DEFAULT NULL,
>>>>>>>>> >>> >   `intralata_rate` FLOAT(11,5) DEFAULT NULL,
>>>>>>>>> >>> >   `carrier_id` INT(11) NOT NULL,
>>>>>>>>> >>> >   `lead_strip` INT(11) NOT NULL,
>>>>>>>>> >>> >   `trail_strip` INT(11) NOT NULL,
>>>>>>>>> >>> >   `prefix` VARCHAR(16) NOT NULL,
>>>>>>>>> >>> >   `suffix` VARCHAR(16) NOT NULL,
>>>>>>>>> >>> >   `lcr_profile` VARCHAR(32) DEFAULT NULL,
>>>>>>>>> >>> >   `date_start` DATETIME NOT NULL DEFAULT '1970-01-01
>>>>>>>>> 00:00:00',
>>>>>>>>> >>> >   `date_end` DATETIME NOT NULL DEFAULT '2030-12-31 00:00:00',
>>>>>>>>> >>> >   `quality` FLOAT(10,6) NOT NULL,
>>>>>>>>> >>> >   `reliability` FLOAT(10,6) NOT NULL,
>>>>>>>>> >>> >   `cid` VARCHAR(32) NOT NULL DEFAULT '',
>>>>>>>>> >>> >   `enabled` TINYINT(1) NOT NULL DEFAULT '1',
>>>>>>>>> >>> >   PRIMARY KEY  (`id`),
>>>>>>>>> >>> >   KEY `carrier_id` (`carrier_id`),
>>>>>>>>> >>> >   KEY `digits` (`digits`),
>>>>>>>>> >>> >   KEY `lcr_profile` (`lcr_profile`),
>>>>>>>>> >>> >   KEY `digits_profile_cid_rate` USING BTREE (`digits`),
>>>>>>>>> >>> >   CONSTRAINT `carrier_id` FOREIGN KEY (`carrier_id`)
>>>>>>>>> REFERENCES
>>>>>>>>> >>> > `carriers`
>>>>>>>>> >>> > (`id`) ON DELETE CASCADE ON UPDATE CASCADE
>>>>>>>>> >>> > ) ENGINE=INNODB AUTO_INCREMENT=6 DEFAULT CHARSET=latin1
>>>>>>>>> >>> >
>>>>>>>>> >>> >
>>>>>>>>> >>> > lcr_admin show profiles
>>>>>>>>> >>> > Name:           default
>>>>>>>>> >>> >  custom sql:    SELECT l.digits, c.carrier_name,
>>>>>>>>> l.${lcr_rate_field},
>>>>>>>>> >>> > cg.prefix AS gw_prefix, cg.suffix AS gw_suffix, l.lead_strip,
>>>>>>>>> >>> > l.trail_strip,
>>>>>>>>> >>> > l.prefix, l.suffix , cg.codec , l.cid FROM lcr l JOIN
>>>>>>>>> carriers c ON
>>>>>>>>> >>> > l.carrier_id=c.id JOIN carrier_gateway cg ON c.id=cg.carrier_id
>>>>>>>>> WHERE
>>>>>>>>> >>> > c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1' AND
>>>>>>>>> digits IN
>>>>>>>>> >>> > (${lcr_query_expanded_digits}) AND CURRENT_TIMESTAMP BETWEEN
>>>>>>>>> date_start
>>>>>>>>> >>> > AND
>>>>>>>>> >>> > date_end ORDER BY digits DESC,  ${lcr_rate_field},  quality
>>>>>>>>> DESC,
>>>>>>>>> >>> >  reliability DESC, rand();
>>>>>>>>> >>> >  has %:         false
>>>>>>>>> >>> >  has vars:      true
>>>>>>>>> >>> >  has intrastate:        true
>>>>>>>>> >>> >  has intralata: true
>>>>>>>>> >>> >  has npanxx:    true
>>>>>>>>> >>> >  Reorder rate:  enabled
>>>>>>>>> >>> >  Info in headers:       disabled
>>>>>>>>> >>> >  Quote IN() List:       disabled
>>>>>>>>> >>> >
>>>>>>>>> >>> >
>>>>>>>>> >>> >
>>>>>>>>> >>> > lcr 617642  default  returns rate from the rate field table
>>>>>>>>> and not
>>>>>>>>> >>> > intra/inter state fields rates.
>>>>>>>>> >>> >
>>>>>>>>> >>> > Any ideas? thanks!
>>>>>>>>> >>> >
>>>>>>>>> >>> >
>>>>>>>>> >>> >
>>>>>>>>> >>> >
>>>>>>>>> >>> > _______________________________________________
>>>>>>>>> >>> > FreeSWITCH-users mailing list
>>>>>>>>> >>> > <FreeSWITCH-users at lists.freeswitch.org>
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> >>> >
>>>>>>>>> <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> >>> >
>>>>>>>>> >>> > UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> >>> > <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>>>> >>> >
>>>>>>>>> >>> >
>>>>>>>>> >>>
>>>>>>>>> >>>
>>>>>>>>> >>>
>>>>>>>>> >>> --
>>>>>>>>> >>> -Rupa
>>>>>>>>> >>>
>>>>>>>>> >>> _______________________________________________
>>>>>>>>> >>> FreeSWITCH-users mailing list
>>>>>>>>> >>> <FreeSWITCH-users at lists.freeswitch.org>
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> >>>
>>>>>>>>> <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> >>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> >>> <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>>>> >>
>>>>>>>>> >
>>>>>>>>> >
>>>>>>>>> > _______________________________________________
>>>>>>>>> > FreeSWITCH-users mailing list
>>>>>>>>> > <FreeSWITCH-users at lists.freeswitch.org>
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> > <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> > UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> > <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>>>> >
>>>>>>>>> >
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> --
>>>>>>>>> -Rupa
>>>>>>>>>
>>>>>>>>> _______________________________________________
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> --
>>>>>>> -Rupa
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> -Rupa
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> -Rupa
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> -Rupa
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100201/6ef88a40/attachment-0002.html 

From mbsip at gazeta.pl  Mon Feb  1 15:15:28 2010
From: mbsip at gazeta.pl (mbsip)
Date: Tue, 2 Feb 2010 00:15:28 +0100
Subject: [Freeswitch-users] vm-disk-quota
In-Reply-To: <28f27f5d1002010930x61bcef3v54e22621761c8bbd@mail.gmail.com>
References: <28f27f5d1001311246h17b426a4x39e0d48d3d305342@mail.gmail.com>
	<7DFBD163-A72A-4186-BB89-6D468FD9ABA2@jerris.com>
	<28f27f5d1002010930x61bcef3v54e22621761c8bbd@mail.gmail.com>
Message-ID: <28f27f5d1002011515u2c270c3cp4b38450e56a0bbf8@mail.gmail.com>

>> If I read the code right (mod_voicemail.c:3051) it looks like it is measuring in seconds of vocicemail, but the wiki indicates number of voicemails, neither seems to match the name of the param.

I double checked - as Mike stated "vm-disk-quota" limits seconds of
voicemail messages.

As an example let's provide FS with <param name="vm-disk-quota" value="10"/>.
It is then possible to store let say 4sek + 5sek + 6sek of vm messages.

Thx for clearing this up.
Maciej.



From mbsip at gazeta.pl  Mon Feb  1 15:22:38 2010
From: mbsip at gazeta.pl (mbsip)
Date: Tue, 2 Feb 2010 00:22:38 +0100
Subject: [Freeswitch-users] voicemail_greeting_number - question
In-Reply-To: <28f27f5d1001310905r41b16ca7r5ef1f236f76a070c@mail.gmail.com>
References: <28f27f5d1001310905r41b16ca7r5ef1f236f76a070c@mail.gmail.com>
Message-ID: <28f27f5d1002011522h31e03c0aoa71f26dabfa1d174@mail.gmail.com>

> Hi ALL,
>
> I am playing around with VM and want to play user recorded greeting
> instead of default one.
> I've scaned wiki Mod_Voicemail and found proper parameter
> "voicemail_greeting_number".
> Unfortunately there is a lack of example hence i dont know if it is
> already working.
>
> Aforementioned param was placed in /conf/directory/default/1000.xml
> file (param name="voicemail_greeting_number", i tried many values)
> The effect is that the default greeting is played.
>
> Is this param embeeded into FS right now?
> How to use it?
> Is there any other place I should do the changes?
>
> I am running ?FreeSWITCH Version 1.0.trunk (16456).
>
> Thx in advance.
> Maciej
>

Anyone knows how to use this param?

Of course i may provide voicemail_default.db with proper greeting_path
manually but i am not sure if "voicemail_greeting_number" works the
same way and is somehow correlated?

Thx,
Maciej.



From msc at freeswitch.org  Mon Feb  1 16:47:47 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 1 Feb 2010 16:47:47 -0800
Subject: [Freeswitch-users] Svar: Re: Somebody help me understand the
	'features' set up please :-)
In-Reply-To: <4B66E734020000E100000451@mail.fribert.dk>
References: <4B66226C020000E10000043C@mail.fribert.dk>
	<b39273531001311645u618ea87am8a3d84ee6b1bcfcb@mail.gmail.com>
	<4B669ED3020000E100000447@mail.fribert.dk>
	<b39273531002010303h6e66baf7tf77be8d67449122b@mail.gmail.com>
	<4B66E734020000E100000451@mail.fribert.dk>
Message-ID: <87f2f3b91002011647k792d2fdbyf4eaa7c2a41d9ab5@mail.gmail.com>

On Mon, Feb 1, 2010 at 5:37 AM, mailinglist <mailinglist at fribert.dk> wrote:

>  Hmm, I've just downloaded the default.xml under conf/dialplan from the
> SVN just to be on the safe side.
> Line 758 is the last </include>, but I did find some examples on line
> 249-251.
>
> So I've changed my dialplan entry handling calls from the outside to this:
>
> <extension name="PrivateIn">
>    <condition field="destination_number" expression="^(8202)$">
>        <action application="bind_meta_app" data="1 b s
> execute_extension::att_xfer XML features"/>
>        <action application="bind_meta_app" data="2 b s
> record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
>
>        <action application="set" data="call_timeout=45"/>
>        <action application="set" data="domain_name=10.11.12.25"/>
>        <action application="set" data="continue_on_fail=true"/>
>        <action application="set" data="hangup_after_bridge=true"/>
>        <action application="set" data="transfer_ringback=$${hold_music}"/>
>        <action application="bridge" data="group/private@${domain_name}"/<group/private@$%7Bdomain_name%7D%22/>
> >
>        <action application="answer" data=""/>
>        <action application="sleep" data="1000"/>
>        <action application="voicemail" data="default 10.11.12.25 45"/>
>    </condition>
> </extension>
> As I understand the bind_meta_app it listens for *1 and then it runs the
> att_xfer, *2 to record the call.
> I've included the att_xfer in the XML features.
>
> Question is, will it work at all when I bridge to a group?
>
> Nothing happens when I press *1 and an extension.
>

Fribse,

I think your confusion might be from the purpose of the bind_meta_app
application. The *1 or *2, etc. must be dialed after the call has been
established. In other words, bind_meta_app sits there on an existing call,
listening for *1 or *2 (etc.) and if the person on the appropriate call leg
dials it then the application in bind_meta_app gets executed. Example:

<action application="bind_meta_app" data="2 b s
record_session::$${recordings_
dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>

The "2 b s" part of that means: Listen for *2 on the b leg and execute the
app on the s leg. So, the b leg could dial *2 and it would initiate call
recording.

Let's take a step back... are you sure you need bind_meta_app? What exactly
is your use case here? In general terms, what are you trying to accomplish
with your dialplan extension?

-MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100201/2cb0cded/attachment-0002.html 

From msc at freeswitch.org  Mon Feb  1 16:49:07 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 1 Feb 2010 16:49:07 -0800
Subject: [Freeswitch-users] voicemail_greeting_number - question
In-Reply-To: <28f27f5d1002011522h31e03c0aoa71f26dabfa1d174@mail.gmail.com>
References: <28f27f5d1001310905r41b16ca7r5ef1f236f76a070c@mail.gmail.com>
	<28f27f5d1002011522h31e03c0aoa71f26dabfa1d174@mail.gmail.com>
Message-ID: <87f2f3b91002011649p62ddff50o3f47bbb2b0be538a@mail.gmail.com>

On Mon, Feb 1, 2010 at 3:22 PM, mbsip <mbsip at gazeta.pl> wrote:

> > Hi ALL,
> >
> > I am playing around with VM and want to play user recorded greeting
> > instead of default one.
> > I've scaned wiki Mod_Voicemail and found proper parameter
> > "voicemail_greeting_number".
> > Unfortunately there is a lack of example hence i dont know if it is
> > already working.
> >
> > Aforementioned param was placed in /conf/directory/default/1000.xml
> > file (param name="voicemail_greeting_number", i tried many values)
> > The effect is that the default greeting is played.
> >
> > Is this param embeeded into FS right now?
> > How to use it?
> > Is there any other place I should do the changes?
> >
> > I am running  FreeSWITCH Version 1.0.trunk (16456).
> >
> > Thx in advance.
> > Maciej
> >
>
> Anyone knows how to use this param?
>
> Of course i may provide voicemail_default.db with proper greeting_path
> manually but i am not sure if "voicemail_greeting_number" works the
> same way and is somehow correlated?
>
> Have you recorded vm greeting one, vm greeting two, etc. before changing
the param?
-MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100201/d69c62b7/attachment-0002.html 

From msc at freeswitch.org  Mon Feb  1 16:55:47 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 1 Feb 2010 16:55:47 -0800
Subject: [Freeswitch-users] Limit the extension creation
In-Reply-To: <df97b9c51002011122t29c505adu60c4ccb73746432c@mail.gmail.com>
References: <df97b9c51002011122t29c505adu60c4ccb73746432c@mail.gmail.com>
Message-ID: <87f2f3b91002011655q186d29eif5a7124d256dcb54@mail.gmail.com>

On Mon, Feb 1, 2010 at 11:22 AM, Scott Fernandez <scottferri09 at gmail.com>wrote:

> Hi,
>
> Is there a way to restrict the number of extension that FS
> supports/serves?. The idea is to limit the concurrent usage of the system
> for which we need to restrict the FS to support upto a predefined no. of
> users/extensions.
>
> Can anyone assist please?
>
>
Do you mean limiting the number of extensions defined in the XML configs, or
do you mean the number of concurrent calls? If the former you'll probably
need to use mod_xml_curl and have your backend db & logic enforce the max
number of users/extensions. If the latter then you'll want to read up on
mod_limit: http://wiki.freeswitch.org/wiki/Mod_limit

-MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100201/5dddbc9d/attachment-0002.html 

From msc at freeswitch.org  Mon Feb  1 16:59:54 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 1 Feb 2010 16:59:54 -0800
Subject: [Freeswitch-users] Logging question
In-Reply-To: <da4ab26e1002011307n6b10e97tea3e281cbff90e13@mail.gmail.com>
References: <da4ab26e1002011307n6b10e97tea3e281cbff90e13@mail.gmail.com>
Message-ID: <87f2f3b91002011659i25f9c3f5la3f2f6706fd259d8@mail.gmail.com>

On Mon, Feb 1, 2010 at 1:07 PM, paul gore <paul.gore.j at gmail.com> wrote:

> I tried "sofia profile internal siptrace on" from console and got SIP
> trace on the console,
> but nothing in logs still.
>
> AFAIK the SIP logs will only go to the console. Is there something in
particular that you are trying to capture? SIP is quite the chatty protocol
and will happily fill up your entire disk with requests and responses. In
many cases the FreeSWITCH log has the basic information from the SIP
message, like why a call failed (i.e. which 4xx message was received). Are
you tracking a particular problem?

-MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100201/45256d86/attachment-0002.html 

From sos at sokhapkin.dyndns.org  Mon Feb  1 17:08:21 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Mon, 1 Feb 2010 20:08:21 -0500
Subject: [Freeswitch-users] Logging question
In-Reply-To: <87f2f3b91002011659i25f9c3f5la3f2f6706fd259d8@mail.gmail.com>
References: <da4ab26e1002011307n6b10e97tea3e281cbff90e13@mail.gmail.com>
	<87f2f3b91002011659i25f9c3f5la3f2f6706fd259d8@mail.gmail.com>
Message-ID: <201002012008.21189.sos@sokhapkin.dyndns.org>

"sofia tracelevel info" sends log to log file.

On Monday 01 February 2010, Michael Collins wrote:
> On Mon, Feb 1, 2010 at 1:07 PM, paul gore <paul.gore.j at gmail.com> wrote:
> > I tried "sofia profile internal siptrace on" from console and got SIP
> > trace on the console,
> > but nothing in logs still.
> >
> > AFAIK the SIP logs will only go to the console. Is there something in
>
> particular that you are trying to capture? SIP is quite the chatty protocol
> and will happily fill up your entire disk with requests and responses. In
> many cases the FreeSWITCH log has the basic information from the SIP
> message, like why a call failed (i.e. which 4xx message was received). Are
> you tracking a particular problem?
>
> -MC





From paul.gore.j at gmail.com  Mon Feb  1 17:31:18 2010
From: paul.gore.j at gmail.com (paul gore)
Date: Mon, 1 Feb 2010 20:31:18 -0500
Subject: [Freeswitch-users] Logging question
In-Reply-To: <87f2f3b91002011659i25f9c3f5la3f2f6706fd259d8@mail.gmail.com>
References: <da4ab26e1002011307n6b10e97tea3e281cbff90e13@mail.gmail.com>
	<87f2f3b91002011659i25f9c3f5la3f2f6706fd259d8@mail.gmail.com>
Message-ID: <da4ab26e1002011731s4b5f8906se23de33a713e0396@mail.gmail.com>

Yes, I am trying to understand why Grandstream GXP2000 times out when
connected to FS voice mail at exactly 60 sec. Can't seem to figure it out
from regular debug log, it just shows "normal clearing". I already used
"record_waste.." param but it does not seem to have any effect.
I guess I have to do tcp dump.


On Mon, Feb 1, 2010 at 7:59 PM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Mon, Feb 1, 2010 at 1:07 PM, paul gore <paul.gore.j at gmail.com> wrote:
>
>> I tried "sofia profile internal siptrace on" from console and got SIP
>> trace on the console,
>> but nothing in logs still.
>>
>> AFAIK the SIP logs will only go to the console. Is there something in
> particular that you are trying to capture? SIP is quite the chatty protocol
> and will happily fill up your entire disk with requests and responses. In
> many cases the FreeSWITCH log has the basic information from the SIP
> message, like why a call failed (i.e. which 4xx message was received). Are
> you tracking a particular problem?
>
> -MC
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100201/c4feb2ad/attachment-0002.html 

From brian at freeswitch.org  Mon Feb  1 17:36:33 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 1 Feb 2010 19:36:33 -0600
Subject: [Freeswitch-users] Logging question
In-Reply-To: <da4ab26e1002011731s4b5f8906se23de33a713e0396@mail.gmail.com>
References: <da4ab26e1002011307n6b10e97tea3e281cbff90e13@mail.gmail.com>
	<87f2f3b91002011659i25f9c3f5la3f2f6706fd259d8@mail.gmail.com>
	<da4ab26e1002011731s4b5f8906se23de33a713e0396@mail.gmail.com>
Message-ID: <50A3745B-369A-40B9-9484-40E3FD41AFF9@freeswitch.org>

Well if you do "sofia profile internal siptrace on" and then press F8, make the call I'm going to guess its the session timers and its freaking out and not answering the reinvite.

/b

On Feb 1, 2010, at 7:31 PM, paul gore wrote:

> Yes, I am trying to understand why Grandstream GXP2000 times out when connected to FS voice mail at exactly 60 sec. Can't seem to figure it out from regular debug log, it just shows "normal clearing". I already used "record_waste.." param but it does not seem to have any effect. 
> I guess I have to do tcp dump.
> 




From emptysands at gmail.com  Mon Feb  1 19:52:29 2010
From: emptysands at gmail.com (Nicholas Lee)
Date: Tue, 2 Feb 2010 16:52:29 +1300
Subject: [Freeswitch-users] Hybrid Encryption?
In-Reply-To: <B2BE2748-ADA4-4D1C-8004-99AD72FB7806@jerris.com>
References: <2b6116b31001272154l3c0bbe80y8bf3db94961e8e1d@mail.gmail.com>
	<8A9EDC4E-C49B-488D-9DBF-169A185462AB@freeswitch.org>
	<2b6116b31001281631u7ada7876wc419bb7afadd7ef7@mail.gmail.com>
	<46C06209-9515-4B1B-B449-F55A51FF548B@freeswitch.org>
	<2b6116b31001281808x1a004cd2ne7e8dcb9f16fec3e@mail.gmail.com>
	<B2BE2748-ADA4-4D1C-8004-99AD72FB7806@jerris.com>
Message-ID: <2b6116b31002011952w71108bf7sfd10c9e29ad2af7c@mail.gmail.com>

Ok, so when [1] talks about Freeswitch acting as intermediary in order to
encrypt further communications between an end node SIP device and the main
PBX, it means that the FS intermediary node is actually a full SIP node. The
phones will need to auth to this and any calls will be routed to trunks via
the dial plan.


[1] http://wiki.freeswitch.org/wiki/SIP_TLS#Hybrid_Encryption


On Sat, Jan 30, 2010 at 8:50 PM, Michael Jerris <mike at jerris.com> wrote:

> Freeswitch isn't a proxy, and no, we don't provide support for passthrough
> auth like this.  A proxy would, but not sure of any proxy based solution
> that would do the srtp work for you.
>
> Mike
>
> On Jan 28, 2010, at 9:08 PM, Nicholas Lee wrote:
>
> Is there a way to do it transparently?  The FS proxies will past though the
> extension creds.
>
> On Fri, Jan 29, 2010 at 1:52 PM, Brian West <brian at freeswitch.org> wrote:
>
>> Then yes you could use FreeSWITCH to augment your Asterisk install and
>> enable encryption from site to site.
>>
>> /b
>>
>>
>> > Unfortunately it's not going to cover every situation.
>> >
>> >
>> > Nicholas
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100202/f6472db4/attachment-0002.html 

From mike at jerris.com  Mon Feb  1 22:57:26 2010
From: mike at jerris.com (Michael Jerris)
Date: Tue, 2 Feb 2010 01:57:26 -0500
Subject: [Freeswitch-users] vm-disk-quota
In-Reply-To: <28f27f5d1002011515u2c270c3cp4b38450e56a0bbf8@mail.gmail.com>
References: <28f27f5d1001311246h17b426a4x39e0d48d3d305342@mail.gmail.com>
	<7DFBD163-A72A-4186-BB89-6D468FD9ABA2@jerris.com>
	<28f27f5d1002010930x61bcef3v54e22621761c8bbd@mail.gmail.com>
	<28f27f5d1002011515u2c270c3cp4b38450e56a0bbf8@mail.gmail.com>
Message-ID: <1BBB9F7F-47B0-4280-B3B3-315282D150F0@jerris.com>

Don't depend on this behavior, really, open a bug on jira for us to figure this one out.  We either need to rename this or change behavior

Mike

On Feb 1, 2010, at 6:15 PM, mbsip wrote:

>>> If I read the code right (mod_voicemail.c:3051) it looks like it is measuring in seconds of vocicemail, but the wiki indicates number of voicemails, neither seems to match the name of the param.
> 
> I double checked - as Mike stated "vm-disk-quota" limits seconds of
> voicemail messages.
> 
> As an example let's provide FS with <param name="vm-disk-quota" value="10"/>.
> It is then possible to store let say 4sek + 5sek + 6sek of vm messages.




From mike at jerris.com  Tue Feb  2 00:14:35 2010
From: mike at jerris.com (Michael Jerris)
Date: Tue, 2 Feb 2010 03:14:35 -0500
Subject: [Freeswitch-users] Dialogic Board (JCT & DM3) Supported?
In-Reply-To: <29b888f81002010308uf875b53h838453b6e77e9a9e@mail.gmail.com>
References: <29b888f81002010308uf875b53h838453b6e77e9a9e@mail.gmail.com>
Message-ID: <D7BC4112-CA0B-4825-8ADA-A8F228C2F201@jerris.com>

Nope

On Feb 1, 2010, at 6:08 AM, Moiz Chinoy wrote:
> Can anyone please advise that whether Dialogic boards (JCT and DM3)
> are supported by FS.




From mike at jerris.com  Tue Feb  2 00:19:00 2010
From: mike at jerris.com (Michael Jerris)
Date: Tue, 2 Feb 2010 03:19:00 -0500
Subject: [Freeswitch-users] Outbound Call (No Registration)
In-Reply-To: <033e01caa37c$87cd2080$97676180$@com>
References: <1254B3C5-D7D3-413D-BA82-54FE4789B360@freeswitch.org>
	<022701caa04a$44f60b80$cee22280$@com>
	<697C7F93-3737-4ABD-8934-6A48DC09C088@freeswitch.org>
	<025701caa0a7$e1ca6200$a55f2600$@com>
	<02b001caa10c$913b9100$b3b2b300$@com>
	<F03237B7-89EE-4CE4-A67F-FCB8E8D26700@freeswitch.org>
	<02c501caa113$105532b0$30ff9810$@com>
	<87f2f3b91001291110l226ce6d8u6e806d0a1b782c5f@mail.gmail.com>
	<02dd01caa127$428aa760$c79ff620$@com>
	<032301caa367$307c6c60$91754520$@com>
	<20100201193459.GI27405@base.carmickle.com>
	<033e01caa37c$87cd2080$97676180$@com>
Message-ID: <21BCBF5F-F609-474C-A295-0B4FFAE75144@jerris.com>

you use normal regex replacement


^00614(\d{8})

in your dialing rule you would use 120#614$1

On Feb 1, 2010, at 3:24 PM, RR wrote:

> Hi Frank,
> 
> Thanks for the response. The remote gateway is not running FreeSWITCH. But
> that's ok, I figured out the problem. I was capturing packets from the wrong
> IP address so after adding the username/password, the call did manage to get
> to the other end (after I realized the source IP was wrong) and realized
> that my dialplan at the other side wasn't setup correctly. Should be able to
> fix that. 
> 
> BTW, before I go hunting in the Wiki, can you off the top of your head tell
> me how I can manipulate numbers in the dialplan?
> 
> If my destination number = 00614xxxxxxxx but I want to send 120#614xxxxxxx
> instead, how would I do that?
> 




From mailinglist at fribert.dk  Tue Feb  2 00:12:11 2010
From: mailinglist at fribert.dk (mailinglist)
Date: Tue, 02 Feb 2010 09:12:11 +0100
Subject: [Freeswitch-users] Svar: Re: Somebody help me understand	 the
	'features' set up please :-)
In-Reply-To: <87f2f3b91002011647k792d2fdbyf4eaa7c2a41d9ab5@mail.gmail.com>
References: <4B66226C020000E10000043C@mail.fribert.dk>
	<b39273531001311645u618ea87am8a3d84ee6b1bcfcb@mail.gmail.com>
	<4B669ED3020000E100000447@mail.fribert.dk>
	<b39273531002010303h6e66baf7tf77be8d67449122b@mail.gmail.com>
	<4B66E734020000E100000451@mail.fribert.dk>
	<87f2f3b91002011647k792d2fdbyf4eaa7c2a41d9ab5@mail.gmail.com>
Message-ID: <4B67EC6B020000E100000456@mail.fribert.dk>

Hi Michael
 
I'm trying to get the possibility of transfering an incoming call from one extension to another, and give the possibility of turning it into a conference.
I don't have a 'transfer' button.
I do have an 'R' button on the Siemens handsets, and a 'Flash' button on the Sipura. The 'Flash' button gives me a new dialtone, gives the caller MOH, and then I can dial the new extension, and transfer the call, but not create a conference.
But the Siemens handset does not have a 'flash', and pressing the R doesn't do anything.
 
It might be two different features 'transfer' and 'conference'...

But I thought that using the bind_meta_app would accomplish both.
 
It's on an incoming call from the outside.
So the situation:
The Public folder has an entry that matches the dialed number, and does a transfer to 8202.
Then the dialplan matches the 8202 with a group, and the phone rings.
Somebody picks it up, finds out that it needs to be transferred to another extension, or transferred to a conference with a second extension.
 
 
Best regards
Fribse
 
/Running Freeswitch on pfSense running on a VMWare ESXi 4.0

>>> 02-02-2010 kl. 01:47 skrev Michael Collins <msc at freeswitch.org> i meddelelsen <87f2f3b91002011647k792d2fdbyf4eaa7c2a41d9ab5 at mail.gmail.com>:



On Mon, Feb 1, 2010 at 5:37 AM, mailinglist <mailinglist at fribert.dk> wrote:


Hmm, I've just downloaded the default.xml under conf/dialplan from the SVN just to be on the safe side.
Line 758 is the last </include>, but I did find some examples on line 249-251.
So I've changed my dialplan entry handling calls from the outside to this:

<extension name="PrivateIn">
<condition field="destination_number" expression="^(8202)$">
<action application="bind_meta_app" data="1 b s execute_extension::att_xfer XML features"/>
<action application="bind_meta_app" data="2 b s record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
<action application="set" data="call_timeout=45"/>
<action application="set" data="domain_name=10.11.12.25"/>
<action application="set" data="continue_on_fail=true"/>
<action application="set" data="hangup_after_bridge=true"/>
<action application="set" data="transfer_ringback=$${hold_music}"/>
<action application="bridge" data="group/private@${domain_name}"/ ( mailto:group/private@$%7Bdomain_name%7D%22/ )>
<action application="answer" data=""/>
<action application="sleep" data="1000"/>
<action application="voicemail" data="default 10.11.12.25 45"/>
</condition>
</extension>
As I understand the bind_meta_app it listens for *1 and then it runs the att_xfer, *2 to record the call.
I've included the att_xfer in the XML features.
Question is, will it work at all when I bridge to a group?
Nothing happens when I press *1 and an extension.


Fribse,

I think your confusion might be from the purpose of the bind_meta_app application. The *1 or *2, etc. must be dialed after the call has been established. In other words, bind_meta_app sits there on an existing call, listening for *1 or *2 (etc.) and if the person on the appropriate call leg dials it then the application in bind_meta_app gets executed. Example:

<action application="bind_meta_app" data="2 b s record_session::$${recordings_
dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>

The "2 b s" part of that means: Listen for *2 on the b leg and execute the app on the s leg. So, the b leg could dial *2 and it would initiate call recording. 

Let's take a step back... are you sure you need bind_meta_app? What exactly is your use case here? In general terms, what are you trying to accomplish with your dialplan extension?

-MC


-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100202/22f4bb1a/attachment-0002.html 

From d at d-man.org  Tue Feb  2 07:26:03 2010
From: d at d-man.org (Darren Schreiber)
Date: Tue, 2 Feb 2010 07:26:03 -0800
Subject: [Freeswitch-users] Announcing... the FreeSWITCH Users Group! COME
	JOIN US!
Message-ID: <8A034A3098ED3C4990F7D9DE40F5585F177DEE2F7C@EXVMBX020-3.exch020.serverdata.net>

Hey everyone!
    I'm writing to invite you all to sign up for a new set of events - FreeSWITCH Users Group Meetups. These in-person gatherings across the country exist to encourage you in creating awesome telephony and general communications software, hardware and services. Anyone interested in, working on or otherwise wanting to learn about FreeSWITCH and general telecommunications services should attend our meetings. Note that topics are not necessarily limited to just FreeSWITCH!

     Meetings will involve formal trainings, informal install fests, lots of Q&A time and general time to just share ideas and meet people.

     Each meeting has a formal topic to help kick things off, so don't worry if you are not sure what you can offer to the group - just showing up and asking questions is a great help! If you do have a specific topic you'd like to learn more about,  email the group's organizer and we'll see what we can do.

We'll kick off with three group locations across the country. You can sign-up to learn more about the scheduled meet-ups by using the links below:
    * San Francisco, California: http://www.meetup.com/fsusers/
    * Manhattan, New York: http://www.meetup.com/fsusers-ny/
    * Orlando, Florida: http://www.meetup.com/fsusers-orlando/

PLEASE SIGN-UP ONLINE SO WE CAN GAUGE INTEREST - You are just showing you might come - not committing to a date!

     Dates for each meet-up will be announced on the relevant websites in the coming months. We'll also have a VoIP dial-in for locations that will allow it.

---> If you'd like to start a meet-up in your area, please let me know. I'll help you arrange the meet-up spot, topics and advertising of the event.

     These events are currently free. The events are run by an independent FreeSWITCH enthusiast and are not associated with any corporation or formal group.

     Enjoy and we look forward to seeing you at our first meet-up!

- Darren Schreiber
  the FreeSWITCH Users Group



From lawwton at gmail.com  Tue Feb  2 09:05:36 2010
From: lawwton at gmail.com (Alfredo Quiroga-Villamil)
Date: Tue, 2 Feb 2010 12:05:36 -0500
Subject: [Freeswitch-users] Announcing... the FreeSWITCH Users Group!
	COME JOIN US!
In-Reply-To: <8A034A3098ED3C4990F7D9DE40F5585F177DEE2F7C@EXVMBX020-3.exch020.serverdata.net>
References: <8A034A3098ED3C4990F7D9DE40F5585F177DEE2F7C@EXVMBX020-3.exch020.serverdata.net>
Message-ID: <5fe6fa8f1002020905s4ebba5d3hc6c59cfa4fa2b27d@mail.gmail.com>

Any freeswitch users in the RTP area in NC?

On Tue, Feb 2, 2010 at 10:26 AM, Darren Schreiber <d at d-man.org> wrote:
> Hey everyone!
> ? ?I'm writing to invite you all to sign up for a new set of events - FreeSWITCH Users Group Meetups. These in-person gatherings across the country exist to encourage you in creating awesome telephony and general communications software, hardware and services. Anyone interested in, working on or otherwise wanting to learn about FreeSWITCH and general telecommunications services should attend our meetings. Note that topics are not necessarily limited to just FreeSWITCH!
>
> ? ? Meetings will involve formal trainings, informal install fests, lots of Q&A time and general time to just share ideas and meet people.
>
> ? ? Each meeting has a formal topic to help kick things off, so don't worry if you are not sure what you can offer to the group - just showing up and asking questions is a great help! If you do have a specific topic you'd like to learn more about, ?email the group's organizer and we'll see what we can do.
>
> We'll kick off with three group locations across the country. You can sign-up to learn more about the scheduled meet-ups by using the links below:
> ? ?* San Francisco, California: http://www.meetup.com/fsusers/
> ? ?* Manhattan, New York: http://www.meetup.com/fsusers-ny/
> ? ?* Orlando, Florida: http://www.meetup.com/fsusers-orlando/
>
> PLEASE SIGN-UP ONLINE SO WE CAN GAUGE INTEREST - You are just showing you might come - not committing to a date!
>
> ? ? Dates for each meet-up will be announced on the relevant websites in the coming months. We'll also have a VoIP dial-in for locations that will allow it.
>
> ---> If you'd like to start a meet-up in your area, please let me know. I'll help you arrange the meet-up spot, topics and advertising of the event.
>
> ? ? These events are currently free. The events are run by an independent FreeSWITCH enthusiast and are not associated with any corporation or formal group.
>
> ? ? Enjoy and we look forward to seeing you at our first meet-up!
>
> - Darren Schreiber
> ?the FreeSWITCH Users Group
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From max at ramax.it  Tue Feb  2 07:03:05 2010
From: max at ramax.it (Massimiliano Ravelli)
Date: Tue, 2 Feb 2010 16:03:05 +0100
Subject: [Freeswitch-users] Fifo: ring agents without answering to caller
Message-ID: <302375DF-6EC3-492D-A82A-7FC8344B01D6@ramax.it>

Hi everybody.

I am using asterisk for a call center pbx and evaluating if I can replace it with FS.
With asterisk queue I can ring agents without actually answer to the caller so he won't pay till the agent pickup.
Can I do the same with FS ?

Moreover I'd really like to fullfill this scenario: I managed to solve it in asterisk only with an ugly workaround.
Customer calls the queue and he doesn't get answered for 20 seconds and then he hears welcome message, music and whatever is configured in the fifo.
Obviously the agents should ring even in the starting 20 seconds.

Any hint or pointer ?

Thanks in advance,
Massimiliano




From stevendt at primrosebank.net  Tue Feb  2 09:29:34 2010
From: stevendt at primrosebank.net (Dave Stevenson)
Date: Tue, 2 Feb 2010 17:29:34 -0000
Subject: [Freeswitch-users] Phones losing registration
Message-ID: <0C95C94A5E2B42449ECF9244C9760D51@bp1.ad.bp.com>

Hi,

I have a mixture of phones and am having a problem with some of them periodically losing registration with FreeSwitch.

All the phones start off working after provisioning and have registration expiry times configured of 3600 seconds.

The Cisco and Thomson phones are OK, staying registered without problems but other phones, including SwissVoice IP10Ss and a cheap wireless VOIP phone lose registration after a random time, sometimes hours, sometimes days.

Is this a common issue with known work-arounds or can someone point me in the right direction for how to see if any errors are being generated or if I need to turn on any additional FreeSWITCH logging options to trap the errors please ?

regards
Dave
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100202/ddd7d241/attachment-0002.html 

From msc at freeswitch.org  Tue Feb  2 09:31:10 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 2 Feb 2010 09:31:10 -0800
Subject: [Freeswitch-users] Fifo: ring agents without answering to caller
In-Reply-To: <302375DF-6EC3-492D-A82A-7FC8344B01D6@ramax.it>
References: <302375DF-6EC3-492D-A82A-7FC8344B01D6@ramax.it>
Message-ID: <87f2f3b91002020931r424629fcqee1eae8a2875be7c@mail.gmail.com>

On Tue, Feb 2, 2010 at 7:03 AM, Massimiliano Ravelli <max at ramax.it> wrote:

> Hi everybody.
>
> I am using asterisk for a call center pbx and evaluating if I can replace
> it with FS.
> With asterisk queue I can ring agents without actually answer to the caller
> so he won't pay till the agent pickup.
> Can I do the same with FS ?
>
> Yes. However you will need to investigate how to handle your billing and
accounting. FreeSWITCH has many hooks for this and there are some 3rd party
projects that use these. You can even use mod_nibblebill for pre-paid
billing.


> Moreover I'd really like to fullfill this scenario: I managed to solve it
> in asterisk only with an ugly workaround.
> Customer calls the queue and he doesn't get answered for 20 seconds and
> then he hears welcome message, music and whatever is configured in the fifo.
> Obviously the agents should ring even in the starting 20 seconds.
>
>
What happens in the first 20 seconds? Do any phones ring? Just curious
what's happening in that limbo period. In any case, I highly recommend that
you grab a spare Linux box and load up FreeSWITCH and play around. The wiki
has tons of information and you can get good real-time help in #freeswitch
in irc.freenode.net.

-MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100202/f41cd995/attachment-0002.html 

From edpimentl at gmail.com  Tue Feb  2 09:33:52 2010
From: edpimentl at gmail.com (EdPimentl)
Date: Tue, 2 Feb 2010 12:33:52 -0500
Subject: [Freeswitch-users] Announcing... the FreeSWITCH Users Group!
	COME JOIN US!
In-Reply-To: <5fe6fa8f1002020905s4ebba5d3hc6c59cfa4fa2b27d@mail.gmail.com>
References: <8A034A3098ED3C4990F7D9DE40F5585F177DEE2F7C@EXVMBX020-3.exch020.serverdata.net>
	<5fe6fa8f1002020905s4ebba5d3hc6c59cfa4fa2b27d@mail.gmail.com>
Message-ID: <9dc4a1671002020933p1c05b00at188274ab564e7aa6@mail.gmail.com>

Any FS users  in ATlanta?

-E
http://vCardCloud.com
http://JustGoogl.Me
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100202/05894bb5/attachment-0002.html 

From Suneel.Papineni at mettoni.com  Tue Feb  2 10:09:50 2010
From: Suneel.Papineni at mettoni.com (Suneel Papineni)
Date: Tue, 2 Feb 2010 18:09:50 -0000
Subject: [Freeswitch-users] Attendant call transfer
Message-ID: <3181A30B8C35AB4AA8577B78DDF461380668B5D3@nickel.mettonigroup.com>

Hi,

 

I am trying to establish attendant call transfer using event sockets. 

1.       A call has come into Freeswitch from an external Gateway  and this call is parked (it is configured to park all calls coming to freeswitch) {Caller A ? FS}

2.       Once the call is parked, I am sending a command to originate a call to another number connected to external gateway. {FS ? Caller B}. Call is established between FS and caller B. ("api originate sofia/external/<another number>@<gateway IP address> 9999")

3.       On receiving event message as "Application: Answer", I am sending another command to bridge call between A & B. ("api uuid_bridge <callerA UUID> <caller UUID>")

4.       With this call is established between A & B, but there is a huge delay (appox 30 secs).

 

I believe that FS is still in the call and might be this is creating  delay (not sure).

 

Could you please tell me if I am doing something wrong or process to achieve this scenario working.

 

I tried in to transfer the call instead of bridging  using the command ("uuid_transfer <callerA UUID> intercept: <caller UUID> inline"), but the response is same as above with huge delay.

 

Thanks & Regards

Suneel

 


*************************************************************************
Please consider the environment before printing this e-mail
*************************************************************************
This email and any files transmitted with it are confidential and
intended solely for the use of the individual or entity to whom they
are addressed. If you have received this email in error please notify
the system manager.  http://www.mettoni.com

Mettoni Ltd
Registered in England and Wales: 4485956
9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN
*************************************************************************

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100202/de05f44a/attachment-0002.html 

From brian at freeswitch.org  Tue Feb  2 10:17:14 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 2 Feb 2010 12:17:14 -0600
Subject: [Freeswitch-users] Attendant call transfer
In-Reply-To: <3181A30B8C35AB4AA8577B78DDF461380668B5D3@nickel.mettonigroup.com>
References: <3181A30B8C35AB4AA8577B78DDF461380668B5D3@nickel.mettonigroup.com>
Message-ID: <07247F72-DE53-4028-AFBF-BB5EB23FEA7B@freeswitch.org>

Suneel,
	After printing 100 copies of this email It dawned on me that you failed to include any details about what SVN revision you're using.  If you can reply with that info I can promptly print out 100 more copies and see if we can find your problem.

Thanks,
Brian
PS: just kidding about the printing part, but the svn rev would be helpful.

On Feb 2, 2010, at 12:09 PM, Suneel Papineni wrote:

> Hi,
>  
> I am trying to establish attendant call transfer using event sockets.
> 1.       A call has come into Freeswitch from an external Gateway  and this call is parked (it is configured to park all calls coming to freeswitch) {Caller A ? FS}
> 2.       Once the call is parked, I am sending a command to originate a call to another number connected to external gateway. {FS ? Caller B}. Call is established between FS and caller B. (?api originate sofia/external/<another number>@<gateway IP address> 9999?)
> 3.       On receiving event message as ?Application: Answer?, I am sending another command to bridge call between A & B. (?api uuid_bridge <callerA UUID> <caller UUID>?)
> 4.       With this call is established between A & B, but there is a huge delay (appox 30 secs).
>  
> I believe that FS is still in the call and might be this is creating  delay (not sure).
>  
> Could you please tell me if I am doing something wrong or process to achieve this scenario working.
>  
> I tried in to transfer the call instead of bridging  using the command (?uuid_transfer <callerA UUID> intercept: <caller UUID> inline?), but the response is same as above with huge delay.
>  
> Thanks & Regards
> Suneel
>  
> *************************************************************************
> Please consider the environment before printing this e-mail
> *************************************************************************
> This email and any files transmitted with it are confidential and
> intended solely for the use of the individual or entity to whom they
> are addressed. If you have received this email in error please notify
> the system manager.  http://www.mettoni.com
> 
> Mettoni Ltd
> Registered in England and Wales: 4485956
> 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN
> *************************************************************************
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100202/6083a1ae/attachment-0002.html 

From tim at communicatefreely.net  Tue Feb  2 10:32:06 2010
From: tim at communicatefreely.net (Tim St. Pierre)
Date: Tue, 02 Feb 2010 13:32:06 -0500
Subject: [Freeswitch-users] Subdomain directory
Message-ID: <4B686FA6.7070400@communicatefreely.net>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hi folks,

I'm building a multi-tenant system, and I need to separate customers from each other, but only in
certain situations.

I want to apply a subdomain to directory entries, so that a search for a user within a subdomain
will only return a user in that domain, but a search in the parent domain will match all users in
that domain AND all subdomains.

Is there some sort of domain aliasing mechanism, or should I do this in the external scripting?

I am doing everything with xml_curl.  I could write the script so that it will look at the domain
requested and decide if a user should be returned or not.

Does anyone see any problems with this setup, or know of an easier way?

The goal is so that things like SIP subscription and registrations are all done in the parent domain
(only one domain system wide), but voice mail, directory application (dial by name), and some other
features can be restricted to the same customer.

All our user ID's (extension numbers for us) are unique.

Thanks!

- -Tim
- --
Tim St. Pierre
IP Voice technician
Communicate Freely
1-877-291-8647 x5101
sip:5101 at communicatefreely.net
tim at communicatefreely.net
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v2.0.4 (FreeBSD)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org

iQCVAwUBS2hvpoqVcvNCnHOrAQK9sgP/eiEtcEx1+OnMZPw0ZQ4gQUG+v/ddewa7
M8c5At4MoqjqNum5ruHtVA22UQt9U3gCG4gsXwaIXmDEDiMm6CTnjYYAgv5q4DvQ
0CmgHo/e1tgTly65GWIys2kwwUqyFsktZ0AeC7FFTyHLfr8l3uE4FJEY5lMRRdrC
0Eceg7MVoY8=
=e45O
-----END PGP SIGNATURE-----



From jerry.richards at teotech.com  Tue Feb  2 12:30:06 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Tue, 2 Feb 2010 12:30:06 -0800
Subject: [Freeswitch-users] Sangoma_boost_client.c:356 Invalid Boost Version
	100 Expecting 101
Message-ID: <0713E04C73F84E47842F6AE17699609C@greyhawk.tonecommander.com>


I upgraded my FS version to 20100201 (i.e. Feb 1st 2010) and the console is
endlessly logging the following errors:

2010-02-02 12:18:10.125121 [CRIT] sangoma_boost_client.c:356 Invalid Boost
Version 100  Expecting 101
2010-02-02 12:18:10.125121 [CRIT] sangoma_boost_client.c:370 Critical Error:
PQ Invalid Event lenght from boost rxlen=23 evsz=1031
2010-02-02 12:18:12.134803 [CRIT] sangoma_boost_client.c:356 Invalid Boost
Version 100  Expecting 101
2010-02-02 12:18:12.134803 [CRIT] sangoma_boost_client.c:370 Critical Error:
PQ Invalid Event lenght from boost rxlen=23 evsz=1031
2010-02-02 12:18:14.134440 [CRIT] sangoma_boost_client.c:356 Invalid Boost
Version 100  Expecting 101
2010-02-02 12:18:14.134440 [CRIT] sangoma_boost_client.c:370 Critical Error:
PQ Invalid Event lenght from boost rxlen=23 evsz=1031

Does anyone know what is causing this?  I am using Wanpipe Driver
wanpipe-3.5.8.6.  I reinstalled the driver and rebuilt FS a couple of times.

Thanks and Best Regards,
Jerry




From moises.silva at gmail.com  Tue Feb  2 13:11:09 2010
From: moises.silva at gmail.com (Moises Silva)
Date: Tue, 2 Feb 2010 16:11:09 -0500
Subject: [Freeswitch-users] Sangoma_boost_client.c:356 Invalid Boost
	Version 100 Expecting 101
In-Reply-To: <0713E04C73F84E47842F6AE17699609C@greyhawk.tonecommander.com>
References: <0713E04C73F84E47842F6AE17699609C@greyhawk.tonecommander.com>
Message-ID: <c4d05cbe1002021311y3d015c3avcab6bcbe4e851c15@mail.gmail.com>

Hello Jerry,

Please download the wanpipe driver version at
ftp://ftp.sangoma.com/linux/custom/DavidYS/wanpipe-3.5.10.smg_pri.1.tgz

As per instructions found at
http://wiki.sangoma.com/wanpipe-SmgPriInstallation

The problem is that at some point FreeSWITCH started requiring a very recent
Sangoma boost version, which has not been released formally yet.

If you run into any other issue let me know,

-- 
Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com

On Tue, Feb 2, 2010 at 3:30 PM, Jerry Richards
<jerry.richards at teotech.com>wrote:

>
> I upgraded my FS version to 20100201 (i.e. Feb 1st 2010) and the console is
> endlessly logging the following errors:
>
> 2010-02-02 12:18:10.125121 [CRIT] sangoma_boost_client.c:356 Invalid Boost
> Version 100  Expecting 101
> 2010-02-02 12:18:10.125121 [CRIT] sangoma_boost_client.c:370 Critical
> Error:
> PQ Invalid Event lenght from boost rxlen=23 evsz=1031
> 2010-02-02 12:18:12.134803 [CRIT] sangoma_boost_client.c:356 Invalid Boost
> Version 100  Expecting 101
> 2010-02-02 12:18:12.134803 [CRIT] sangoma_boost_client.c:370 Critical
> Error:
> PQ Invalid Event lenght from boost rxlen=23 evsz=1031
> 2010-02-02 12:18:14.134440 [CRIT] sangoma_boost_client.c:356 Invalid Boost
> Version 100  Expecting 101
> 2010-02-02 12:18:14.134440 [CRIT] sangoma_boost_client.c:370 Critical
> Error:
> PQ Invalid Event lenght from boost rxlen=23 evsz=1031
>
> Does anyone know what is causing this?  I am using Wanpipe Driver
> wanpipe-3.5.8.6.  I reinstalled the driver and rebuilt FS a couple of
> times.
>
> Thanks and Best Regards,
> Jerry
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100202/0e4fa933/attachment-0002.html 

From jerry.richards at teotech.com  Tue Feb  2 13:13:21 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Tue, 2 Feb 2010 13:13:21 -0800
Subject: [Freeswitch-users] Sangoma_boost_client.c:356 Invalid Boost
	Version 100 Expecting 101
In-Reply-To: <4B688D0D.4080407@sangoma.com>
References: <0713E04C73F84E47842F6AE17699609C@greyhawk.tonecommander.com>
	<4B688D0D.4080407@sangoma.com>
Message-ID: <5C5862A8AFBB4A6785D9EA001E2F21DE@greyhawk.tonecommander.com>

Yannick,
 
That procedure had no effect on the problem.  Any other ideas?
 
Thanks and Best Regards,
Jerry


  _____  

From: Yannick Lam [mailto:yannick at sangoma.com] 
Sent: Tuesday, February 02, 2010 12:38 PM
To: Jerry Richards
Subject: Re: Sangoma_boost_client.c:356 Invalid Boost Version 100 Expecting
101


Hi Jerry,

Can you please stop freeswitch, smg and the wanpipe driver.
Then shutdown your computer and then re insert the card in the systema nd
then boot again and see if you are still getting the issue.



Thank-you, 



Yannick Lam Hang, B.Eng, Tech Support Engineer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
e. yannick at sangoma.com  |   <skype:yannick.sangoma?add> Skype  |
<mailto:yannick_sangoma at hotmail.com> msn 

 
<mailbox://C%7C/Documents%20and%20Settings/Marc/Application%20Data/Thunderbi
rd/Profiles/6u5xr3yx.default/Mail/Local%20Folders/www.sangoma.com>
www.sangoma.com  |  <http://wiki.sangoma.com/> wiki.sangoma.com


Lifetime Warranty. Because it must work!







 





Jerry Richards wrote: 

I upgraded my FS version to 20100201 (i.e. Feb 1st 2010) and the console is

endlessly logging the following errors:



2010-02-02 12:18:10.125121 [CRIT] sangoma_boost_client.c:356 Invalid Boost

Version 100  Expecting 101

2010-02-02 12:18:10.125121 [CRIT] sangoma_boost_client.c:370 Critical Error:

PQ Invalid Event lenght from boost rxlen=23 evsz=1031

2010-02-02 12:18:12.134803 [CRIT] sangoma_boost_client.c:356 Invalid Boost

Version 100  Expecting 101

2010-02-02 12:18:12.134803 [CRIT] sangoma_boost_client.c:370 Critical Error:

PQ Invalid Event lenght from boost rxlen=23 evsz=1031

2010-02-02 12:18:14.134440 [CRIT] sangoma_boost_client.c:356 Invalid Boost

Version 100  Expecting 101

2010-02-02 12:18:14.134440 [CRIT] sangoma_boost_client.c:370 Critical Error:

PQ Invalid Event lenght from boost rxlen=23 evsz=1031



Does anyone know what is causing this?  I am using Wanpipe Driver

wanpipe-3.5.8.6.  I reinstalled the driver and rebuilt FS a couple of times.



Thanks and Best Regards,

Jerry





  

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100202/86746748/attachment-0002.html 

From robert.hadley at teotech.com  Tue Feb  2 14:54:27 2010
From: robert.hadley at teotech.com (Robert Hadley)
Date: Tue, 2 Feb 2010 14:54:27 -0800
Subject: [Freeswitch-users] Sangoma A200 FXS callwaiting and dialplan
	statements
Message-ID: <56F86E3E1D2F45ACB4AEE1FE448E68E9@greyhawk.tonecommander.com>

I have a server that has Freeswitch (1.0.5.pre9) a Sangoma A101 (E1/T1) and
Analog A200 cards using OpenZap.  The cards are connected.  The wanrouter
version is 3.5.8.6 as the newer version didn't work for us yet.  Also trying
to test with the FS trunk but having other wanpipe driver issues.

 

I have a question about using the A200 card.  The intent is to hook up FAX
machines to the two FXS ports.  The desired behavior is the first call goes
to the first FXS port one, and while port one is busy a second call goes to
the second port.

 

In the Freeswitch dialplan there is a statement to bridge to FXS port 1 and
if that fails then bridge to FXS port 2.  However, the first bridge never
fails even when it's busy.  I notice the A200 card supports call waiting.
The FS debug messages show that is what is happening.

 

Question: How do I disable call waiting on the A200 card when used with
OpenZap?

 

I am not sure about whether the hangup_after_bridge=true and
continue_on_fail=true are necessary or in the correct spots.  I've tried
calling with these statements present or not and moved after the first
bridge but doesn't appear to affect the status of the first bridge
statement.

 

Question: Are the hangup_after_bridge=true and continue_on_fail=true
statements necessary and in the right spot?

 

Dialplan statements:

    <extension name="fxsfax">

      <!-- This routes destinations 1010 and 1011 calls to FXS card -->

      <condition field="destination_number" expression="^(101[0-1])$">

        <action application="answer"/>

        <action application="set" data="hangup_after_bridge=true"/>

        <action application="set" data="continue_on_fail=true"/>

        <action application="bridge" data="openzap/FXS/1"/>

        <action application="bridge" data="openzap/FXS/2"/>

 

Debug statements for second call to FXS/1 extension:

(Notice that several of statements mention CALLWAITING.)

2010-02-02 14:00:55.193948 [DEBUG] switch_core_session.c:639 Send signal
sofia/internal/1045 at 192.168.72.141:5060 [BREAK]

2010-02-02 14:00:55.193948 [NOTICE] mod_dptools.c:658 Channel
[sofia/internal/1045 at 192.168.72.141:5060] has been answered

2010-02-02 14:00:55.193948 [DEBUG] switch_channel.c:182
sofia/internal/1045 at 192.168.72.141:5060 receive message [AUDIO_SYNC]

EXECUTE sofia/internal/1045 at 192.168.72.141:5060 set(continue_on_fail=true)

2010-02-02 14:00:55.193948 [DEBUG] sofia.c:3787 Channel
sofia/internal/1045 at 192.168.72.141:5060 entering state [completed][200]

2010-02-02 14:00:55.193948 [DEBUG] mod_dptools.c:768
sofia/internal/1045 at 192.168.72.141:5060 SET [continue_on_fail]=[true]

EXECUTE sofia/internal/1045 at 192.168.72.141:5060 bridge(openzap/FXS/1)

2010-02-02 14:00:55.194948 [DEBUG] mod_openzap.c:366 Set codec PCMU 20ms

2010-02-02 14:00:55.194948 [DEBUG] mod_openzap.c:1191 Connect outbound
channel OpenZAP/2:1/

2010-02-02 14:00:55.194948 [NOTICE] switch_channel.c:613 New Channel
OpenZAP/2:1/ [bc08ce84-eff2-491c-b601-14b7329c71ea]

2010-02-02 14:00:55.194948 [DEBUG] mod_openzap.c:1203 (OpenZAP/2:1/) State
Change CS_NEW -> CS_INIT

2010-02-02 14:00:55.194948 [DEBUG] switch_core_session.c:1018 Send signal
OpenZAP/2:1/ [BREAK]

2010-02-02 14:00:55.194948 [DEBUG] ozmod_analog.c:78 Changing state on 2:1
from UP to CALLWAITING

2010-02-02 14:00:55.194948 [DEBUG] switch_core_state_machine.c:314
(OpenZAP/2:1/) Running State Change CS_INIT

2010-02-02 14:00:55.194948 [DEBUG] switch_core_state_machine.c:338
(OpenZAP/2:1/) State INIT

2010-02-02 14:00:55.194948 [DEBUG] mod_openzap.c:390 (OpenZAP/2:1/) State
Change CS_INIT -> CS_ROUTING

2010-02-02 14:00:55.194948 [DEBUG] switch_core_session.c:1018 Send signal
OpenZAP/2:1/ [BREAK]

2010-02-02 14:00:55.194948 [DEBUG] switch_core_state_machine.c:338
(OpenZAP/2:1/) State INIT going to sleep

2010-02-02 14:00:55.194948 [DEBUG] switch_core_state_machine.c:314
(OpenZAP/2:1/) Running State Change CS_ROUTING

2010-02-02 14:00:55.194948 [DEBUG] switch_core_state_machine.c:341
(OpenZAP/2:1/) State ROUTING

2010-02-02 14:00:55.194948 [DEBUG] mod_openzap.c:413 OpenZAP/2:1/ CHANNEL
ROUTING

2010-02-02 14:00:55.194948 [DEBUG] switch_ivr_originate.c:66 (OpenZAP/2:1/)
State Change CS_ROUTING -> CS_CONSUME_MEDIA

2010-02-02 14:00:55.194948 [DEBUG] switch_core_session.c:1018 Send signal
OpenZAP/2:1/ [BREAK]

2010-02-02 14:00:55.194948 [DEBUG] switch_core_state_machine.c:341
(OpenZAP/2:1/) State ROUTING going to sleep

2010-02-02 14:00:55.194948 [DEBUG] switch_core_state_machine.c:314
(OpenZAP/2:1/) Running State Change CS_CONSUME_MEDIA

2010-02-02 14:00:55.194948 [DEBUG] switch_core_state_machine.c:360
(OpenZAP/2:1/) State CONSUME_MEDIA

2010-02-02 14:00:55.194948 [DEBUG] switch_core_state_machine.c:360
(OpenZAP/2:1/) State CONSUME_MEDIA going to sleep

2010-02-02 14:00:55.210047 [DEBUG] sofia.c:3787 Channel
sofia/internal/1045 at 192.168.72.141:5060 entering state [ready][200]

2010-02-02 14:00:55.213006 [DEBUG] ozmod_analog.c:450 Executing state
handler on 2:1 for CALLWAITING

2010-02-02 14:00:55.657988 [DEBUG] switch_rtp.c:2004 Correct ip/port
confirmed.

2010-02-02 14:01:05.723334 [DEBUG] ozmod_analog.c:422 Changing state on 2:1
from CALLWAITING to UP

 

 

Thanks,

Robert

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100202/cc56a777/attachment-0002.html 

From mbsip at gazeta.pl  Tue Feb  2 14:57:05 2010
From: mbsip at gazeta.pl (mbsip)
Date: Tue, 2 Feb 2010 23:57:05 +0100
Subject: [Freeswitch-users] vm-disk-quota
In-Reply-To: <1BBB9F7F-47B0-4280-B3B3-315282D150F0@jerris.com>
References: <28f27f5d1001311246h17b426a4x39e0d48d3d305342@mail.gmail.com>
	<7DFBD163-A72A-4186-BB89-6D468FD9ABA2@jerris.com>
	<28f27f5d1002010930x61bcef3v54e22621761c8bbd@mail.gmail.com>
	<28f27f5d1002011515u2c270c3cp4b38450e56a0bbf8@mail.gmail.com>
	<1BBB9F7F-47B0-4280-B3B3-315282D150F0@jerris.com>
Message-ID: <28f27f5d1002021457i39e03944j230ee3e9fda6e622@mail.gmail.com>

Done, MODAPP-173.

Thx,
Maciej.

2010/2/2 Michael Jerris <mike at jerris.com>:
> Don't depend on this behavior, really, open a bug on jira for us to figure this one out. ?We either need to rename this or change behavior
>



From robert.hadley at teotech.com  Tue Feb  2 15:02:22 2010
From: robert.hadley at teotech.com (Robert Hadley)
Date: Tue, 2 Feb 2010 15:02:22 -0800
Subject: [Freeswitch-users] Announcing... the FreeSWITCH Users Group!
	COMEJOIN US!
In-Reply-To: <8A034A3098ED3C4990F7D9DE40F5585F177DEE2F7C@EXVMBX020-3.exch020.serverdata.net>
References: <8A034A3098ED3C4990F7D9DE40F5585F177DEE2F7C@EXVMBX020-3.exch020.serverdata.net>
Message-ID: <2974D37F499B455FBC036B02A4DC6107@greyhawk.tonecommander.com>


Any interested FreeSWITCH users in the Seattle area?


-----Original Message-----
From: Darren Schreiber [mailto:d at d-man.org] 
Sent: Tuesday, February 02, 2010 7:26 AM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Announcing... the FreeSWITCH Users Group!
COMEJOIN US!

Hey everyone!
    I'm writing to invite you all to sign up for a new set of events -
FreeSWITCH Users Group Meetups. These in-person gatherings across the
country exist to encourage you in creating awesome telephony and general
communications software, hardware and services. Anyone interested in,
working on or otherwise wanting to learn about FreeSWITCH and general
telecommunications services should attend our meetings. Note that topics are
not necessarily limited to just FreeSWITCH!

     Meetings will involve formal trainings, informal install fests, lots of
Q&A time and general time to just share ideas and meet people.

     Each meeting has a formal topic to help kick things off, so don't worry
if you are not sure what you can offer to the group - just showing up and
asking questions is a great help! If you do have a specific topic you'd like
to learn more about,  email the group's organizer and we'll see what we can
do.

We'll kick off with three group locations across the country. You can
sign-up to learn more about the scheduled meet-ups by using the links below:
    * San Francisco, California: http://www.meetup.com/fsusers/
    * Manhattan, New York: http://www.meetup.com/fsusers-ny/
    * Orlando, Florida: http://www.meetup.com/fsusers-orlando/

PLEASE SIGN-UP ONLINE SO WE CAN GAUGE INTEREST - You are just showing you
might come - not committing to a date!

     Dates for each meet-up will be announced on the relevant websites in
the coming months. We'll also have a VoIP dial-in for locations that will
allow it.

---> If you'd like to start a meet-up in your area, please let me know. I'll
help you arrange the meet-up spot, topics and advertising of the event.

     These events are currently free. The events are run by an independent
FreeSWITCH enthusiast and are not associated with any corporation or formal
group.

     Enjoy and we look forward to seeing you at our first meet-up!

- Darren Schreiber
  the FreeSWITCH Users Group






From jerry.richards at teotech.com  Tue Feb  2 15:38:30 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Tue, 2 Feb 2010 15:38:30 -0800
Subject: [Freeswitch-users] Sangoma_boost_client.c:356 Invalid
	BoostVersion 100 Expecting 101
In-Reply-To: <c4d05cbe1002021311y3d015c3avcab6bcbe4e851c15@mail.gmail.com>
References: <0713E04C73F84E47842F6AE17699609C@greyhawk.tonecommander.com>
	<c4d05cbe1002021311y3d015c3avcab6bcbe4e851c15@mail.gmail.com>
Message-ID: <CC572C08D8F342EBA856DDACAF354C19@greyhawk.tonecommander.com>

Thank You Moises.  The
<ftp://ftp.sangoma.com/linux/custom/DavidYS/wanpipe-3.5.10.smg_pri.1.tgz>
ftp://ftp.sangoma.com/linux/custom/DavidYS/wanpipe-3.5.10.smg_pri.1.tgz
driver works.
 
Best Regards,
Jerry
 


  _____  

From: Moises Silva [mailto:moises.silva at gmail.com] 
Sent: Tuesday, February 02, 2010 1:11 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Sangoma_boost_client.c:356 Invalid
BoostVersion 100 Expecting 101


Hello Jerry,

Please download the wanpipe driver version at
ftp://ftp.sangoma.com/linux/custom/DavidYS/wanpipe-3.5.10.smg_pri.1.tgz

As per instructions found at
http://wiki.sangoma.com/wanpipe-SmgPriInstallation

The problem is that at some point FreeSWITCH started requiring a very recent
Sangoma boost version, which has not been released formally yet.

If you run into any other issue let me know,

-- 
Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com


On Tue, Feb 2, 2010 at 3:30 PM, Jerry Richards <jerry.richards at teotech.com>
wrote:



I upgraded my FS version to 20100201 (i.e. Feb 1st 2010) and the console is
endlessly logging the following errors:

2010-02-02 12:18:10.125121 [CRIT] sangoma_boost_client.c:356 Invalid Boost
Version 100  Expecting 101
2010-02-02 12:18:10.125121 [CRIT] sangoma_boost_client.c:370 Critical Error:
PQ Invalid Event lenght from boost rxlen=23 evsz=1031
2010-02-02 12:18:12.134803 [CRIT] sangoma_boost_client.c:356 Invalid Boost
Version 100  Expecting 101
2010-02-02 12:18:12.134803 [CRIT] sangoma_boost_client.c:370 Critical Error:
PQ Invalid Event lenght from boost rxlen=23 evsz=1031
2010-02-02 12:18:14.134440 [CRIT] sangoma_boost_client.c:356 Invalid Boost
Version 100  Expecting 101
2010-02-02 12:18:14.134440 [CRIT] sangoma_boost_client.c:370 Critical Error:
PQ Invalid Event lenght from boost rxlen=23 evsz=1031

Does anyone know what is causing this?  I am using Wanpipe Driver
wanpipe-3.5.8.6.  I reinstalled the driver and rebuilt FS a couple of times.

Thanks and Best Regards,
Jerry


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100202/ec0787c5/attachment-0002.html 

From anthony.minessale at gmail.com  Tue Feb  2 15:38:41 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 2 Feb 2010 17:38:41 -0600
Subject: [Freeswitch-users] Attendant call transfer
In-Reply-To: <07247F72-DE53-4028-AFBF-BB5EB23FEA7B@freeswitch.org>
References: <3181A30B8C35AB4AA8577B78DDF461380668B5D3@nickel.mettonigroup.com>
	<07247F72-DE53-4028-AFBF-BB5EB23FEA7B@freeswitch.org>
Message-ID: <191c3a031002021538j3d7ab405w3cf727bf98c04b03@mail.gmail.com>

better still,
just download the latest code because we can only help you with the very
latest release.


On Tue, Feb 2, 2010 at 12:17 PM, Brian West <brian at freeswitch.org> wrote:

> Suneel,
> After printing 100 copies of this email It dawned on me that you failed to
> include any details about what SVN revision you're using.  If you can reply
> with that info I can promptly print out 100 more copies and see if we can
> find your problem.
>
> Thanks,
> Brian
> PS: just kidding about the printing part, but the svn rev would be helpful.
>
> On Feb 2, 2010, at 12:09 PM, Suneel Papineni wrote:
>
> Hi,
>
> I am trying to establish attendant call transfer using event sockets.
> 1.       A call has come into Freeswitch from an external Gateway  and
> this call is parked (it is configured to park all calls coming to
> freeswitch) {Caller A ? FS}
> 2.       Once the call is parked, I am sending a command to originate a
> call to another number connected to external gateway. {FS ? Caller B}.
> Call is established between FS and caller B. (?api originate
> sofia/external/<another number>@<gateway IP address> 9999?)
> 3.       On receiving event message as ?Application: Answer?, I am sending
> another command to bridge call between A & B. (?api uuid_bridge <callerA
> UUID> <caller UUID>?)
> 4.       With this call is established between A & B, but there is a huge
> delay (appox 30 secs).
>
> I believe that FS is still in the call and might be this is creating  delay
> (not sure).
>
> Could you please tell me if I am doing something wrong or process to
> achieve this scenario working.
>
> I tried in to transfer the call instead of bridging  using the command
> (?uuid_transfer <callerA UUID> intercept: <caller UUID> inline?), but the
> response is same as above with huge delay.
>
> Thanks & Regards
> Suneel
>
>
> *************************************************************************
> Please consider the environment before printing this e-mail
> *************************************************************************
> This email and any files transmitted with it are confidential and
> intended solely for the use of the individual or entity to whom they
> are addressed. If you have received this email in error please notify
> the system manager.  http://www.mettoni.com
>
> Mettoni Ltd
> Registered in England and Wales: 4485956
> 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN
> *************************************************************************
>
> _______________________________________________
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100202/d7e6417b/attachment-0002.html 

From mbsip at gazeta.pl  Tue Feb  2 15:39:54 2010
From: mbsip at gazeta.pl (mbsip)
Date: Wed, 3 Feb 2010 00:39:54 +0100
Subject: [Freeswitch-users] voicemail_greeting_number - question
In-Reply-To: <87f2f3b91002011649p62ddff50o3f47bbb2b0be538a@mail.gmail.com>
References: <28f27f5d1001310905r41b16ca7r5ef1f236f76a070c@mail.gmail.com>
	<28f27f5d1002011522h31e03c0aoa71f26dabfa1d174@mail.gmail.com>
	<87f2f3b91002011649p62ddff50o3f47bbb2b0be538a@mail.gmail.com>
Message-ID: <28f27f5d1002021539l6a20be3bu774c5d7b32c791e1@mail.gmail.com>

>> > Hi ALL,
>> >
>> > I am playing around with VM and want to play user recorded greeting
>> > instead of default one.
>> > I've scaned wiki Mod_Voicemail and found proper parameter
>> > "voicemail_greeting_number".
>> > Unfortunately there is a lack of example hence i dont know if it is
>> > already working.
>> >
>> > Aforementioned param was placed in /conf/directory/default/1000.xml
>> > file (param name="voicemail_greeting_number", i tried many values)
>> > The effect is that the default greeting is played.
>> >
>> > Is this param embeeded into FS right now?
>> > How to use it?
>> > Is there any other place I should do the changes?
>> >
>> > I am running ?FreeSWITCH Version 1.0.trunk (16456).
>> >
>> > Thx in advance.
>> > Maciej
>> >
>>
>> Anyone knows how to use this param?
>>
>> Of course i may provide voicemail_default.db with proper greeting_path
>> manually but i am not sure if "voicemail_greeting_number" works the
>> same way and is somehow correlated?
>>
> Have you recorded vm greeting one, vm greeting two, etc. before changing the
> param?
> -MC

Here is what i did:
- recorded two files /tmp/vm1.wav and /tmp/vm2.wav
- inserted two entries into voicemail_default.db/voicemail_prefs
1000|XX.xx.XX.xx|1|/tmp/vm1.wav|
1000|XX.xx.XX.xx|2|/tmp/vm2.wav|
- inserted param
<param name="voicemail_greeting_number" value="1"/> after tests exchanged with
<param name="voicemail_greeting_number" value="2"/>

The result is that every call vm2.wav is played - so it only depends
on db entries (param is omitted)

Could sb tell me how to configure this or point me to detailed
description of this param.

Thx,
Maciej.



From jerry.richards at teotech.com  Tue Feb  2 15:55:07 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Tue, 2 Feb 2010 15:55:07 -0800
Subject: [Freeswitch-users] FS Core Dump
Message-ID: <40B8BCC90CB0486EA7730EB86BA1D1DB@greyhawk.tonecommander.com>

I spoke too soon.  Using Freeswitch version Feb 01, 2010 and Wanpipe Driver
ftp://ftp.sangoma.com/linux/custom/DavidYS/wanpipe-3.5.10.smg_pri.1.tgz, the
FS core will dump if I call my external cell phone, answer and then hangup.
See the pastebin: http://pastebin.freeswitch.org/12035.

This happens every time.
 
Best Regards,
Jerry
 




From max.bridgewater at gmail.com  Tue Feb  2 17:27:51 2010
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Tue, 2 Feb 2010 20:27:51 -0500
Subject: [Freeswitch-users] Ringback after before` Bridge
Message-ID: <c55d46951002021727v76e79adcg2281347ff8a0458f@mail.gmail.com>

Hi,

I'm trying to place a call to A and then bridge it to B. The problem I'm
having right now is that after A answers and while dialing B is being dialed
or rining, I want to send A a ringing tone. I don't succeed in doing this.
No tone/ringback is being sent to A.  Here is what i did using ESL:

 api originate
{ringback=\'%(400,200,400,450);%(400,2200,400,450)\',transfer_ringback=\'%(400,200,400,450);%(400,2200,400,450)\',origination_caller_id_number=4156781020}sofia/gateway/voipms/4152309090
&park()

To bridge, I then send the message:

sendmsg e9dae14c-e473-466e-9d65-704e36a82e5f
call-command: execute
execute-app-name: bridge
execute-app-arg:
{{ringback=\'%(400,200,400,450);%(400,2200,400,450\'},origination_caller_id_number=4152309090
}sofia/gateway/voipms/4156781020

any idea?

Max.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100202/a6da8143/attachment-0002.html 

From moises.silva at gmail.com  Tue Feb  2 17:42:20 2010
From: moises.silva at gmail.com (Moises Silva)
Date: Tue, 2 Feb 2010 20:42:20 -0500
Subject: [Freeswitch-users] FS Core Dump
In-Reply-To: <40B8BCC90CB0486EA7730EB86BA1D1DB@greyhawk.tonecommander.com>
References: <40B8BCC90CB0486EA7730EB86BA1D1DB@greyhawk.tonecommander.com>
Message-ID: <c4d05cbe1002021742m546286bcxc96930990e57d9e6@mail.gmail.com>

On Tue, Feb 2, 2010 at 6:55 PM, Jerry Richards
<jerry.richards at teotech.com>wrote:

> I spoke too soon.  Using Freeswitch version Feb 01, 2010 and Wanpipe Driver
> ftp://ftp.sangoma.com/linux/custom/DavidYS/wanpipe-3.5.10.smg_pri.1.tgz,
> the
> FS core will dump if I call my external cell phone, answer and then hangup.
> See the pastebin: http://pastebin.freeswitch.org/12035.
>


Make sure you have latest revision of openzap, a few revisions ago I saw a
commit that set the zchan to null on hangup and you have a core dump on
hangup where the zchan is null, however on the latest trunk zchan is no
longer set to null on SIGEVENT_STOP ... suspicious ... if you can reproduce
with latest openzap trunk (at least revision 1021) then let me know and I
will take a look.

 --
Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100202/d6c7e18c/attachment-0002.html 

From msc at freeswitch.org  Tue Feb  2 19:19:24 2010
From: msc at freeswitch.org (Michael S Collins)
Date: Tue, 2 Feb 2010 19:19:24 -0800
Subject: [Freeswitch-users] Ringback after before` Bridge
In-Reply-To: <c55d46951002021727v76e79adcg2281347ff8a0458f@mail.gmail.com>
References: <c55d46951002021727v76e79adcg2281347ff8a0458f@mail.gmail.com>
Message-ID: <91DD5271-AE00-454A-A5E5-9AA933E0B459@freeswitch.org>

Try the transfer_ringback var. Check the wiki for details.
-MC

Sent from my iPhone

On Feb 2, 2010, at 5:27 PM, Max Bridgewater  
<max.bridgewater at gmail.com> wrote:

> Hi,
>
> I'm trying to place a call to A and then bridge it to B. The problem  
> I'm having right now is that after A answers and while dialing B is  
> being dialed or rining, I want to send A a ringing tone. I don't  
> succeed in doing this. No tone/ringback is being sent to A.  Here is  
> what i did using ESL:
>
>  api originate {ringback=\'%(400,200,400,450); 
> %(400,2200,400,450)\',transfer_ringback=\'%(400,200,400,450); 
> %(400,2200,400,450)\',origination_caller_id_number=4156781020}sofia/ 
> gateway/voipms/4152309090 &park()
>
> To bridge, I then send the message:
>
> sendmsg e9dae14c-e473-466e-9d65-704e36a82e5f
> call-command: execute
> execute-app-name: bridge
> execute-app-arg: {{ringback=\'%(400,200,400,450); 
> %(400,2200,400,450\'},origination_caller_id_number=4152309090 }sofia/ 
> gateway/voipms/4156781020
>
> any idea?
>
> Max.
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org



From nagalenoj at gmail.com  Tue Feb  2 19:34:49 2010
From: nagalenoj at gmail.com (Nagalenoj H.)
Date: Wed, 3 Feb 2010 09:04:49 +0530
Subject: [Freeswitch-users] A leg disconnects after uuid_bridge?
Message-ID: <fe67a3341002021934m548cde89p4d48381c5abe02a0@mail.gmail.com>

Dear friends,
  In event socket, Why the session is closed for A leg when I do a
uuid_bridge with another uuid.
I've done the following operations(In nc),
  * Dial to he socket extension.
  * connect to the call.
  * Answered the call.
  * Bridged with extension X. When B leg terminates, A leg continues to be
alive.
  * Create an uuid.
  * Originated a call with origination_uuid and parked the leg.
  * I did a uuid_bridge for these 2 legs.
  * When B leg terminates the call, A leg is also getting exited.

I've tried setting the hangup_after_bridge to false explicitly. But, A leg
is getting exited.
I need it for further processing.

What is the way in which I can keep the A leg alive after uuid_bridge??

-- 
Regards,
Nagalenoj H.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100203/1ee810c1/attachment-0002.html 

From anthony.minessale at gmail.com  Tue Feb  2 21:08:02 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 2 Feb 2010 23:08:02 -0600
Subject: [Freeswitch-users] A leg disconnects after uuid_bridge?
In-Reply-To: <fe67a3341002021934m548cde89p4d48381c5abe02a0@mail.gmail.com>
References: <fe67a3341002021934m548cde89p4d48381c5abe02a0@mail.gmail.com>
Message-ID: <191c3a031002022108ta22287ev40fc36420abba2e5@mail.gmail.com>

Tell one leg to execute intercept on the other instead.

On Feb 2, 2010 9:40 PM, "Nagalenoj H." <nagalenoj at gmail.com> wrote:

Dear friends,
  In event socket, Why the session is closed for A leg when I do a
uuid_bridge with another uuid.
I've done the following operations(In nc),
  * Dial to he socket extension.
  * connect to the call.
  * Answered the call.
  * Bridged with extension X. When B leg terminates, A leg continues to be
alive.
  * Create an uuid.
  * Originated a call with origination_uuid and parked the leg.
  * I did a uuid_bridge for these 2 legs.
  * When B leg terminates the call, A leg is also getting exited.

I've tried setting the hangup_after_bridge to false explicitly. But, A leg
is getting exited.
I need it for further processing.

What is the way in which I can keep the A leg alive after uuid_bridge??

-- 
Regards,
Nagalenoj H.

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100202/d95e34e7/attachment-0002.html 

From thangappan143 at gmail.com  Tue Feb  2 21:22:25 2010
From: thangappan143 at gmail.com (Thangappan.M)
Date: Wed, 3 Feb 2010 10:52:25 +0530
Subject: [Freeswitch-users] Need Help to setup freeswitch with sangoma
	card
In-Reply-To: <7aa29e791001242150o3a582e2end69f1f5e9f1aef1c@mail.gmail.com>
References: <7aa29e791001192155l3c1f06e6w69769a69c9b761a1@mail.gmail.com>
	<7aa29e791001192313p73b20c8fk7e9b78e577c305a9@mail.gmail.com>
	<7aa29e791001200610m5a7b64cdx356e527ea7285b23@mail.gmail.com>
	<7aa29e791001202104n276ca48bo56d55ca82f3aac0f@mail.gmail.com>
	<7aa29e791001212045u20d85213wa0c8523761fee826@mail.gmail.com>
	<7aa29e791001222032t6f381e34x9f3c5eb099af9ba8@mail.gmail.com>
	<7aa29e791001242025y2adc91aes622bd22ea8d5ae1c@mail.gmail.com>
	<7aa29e791001242150o3a582e2end69f1f5e9f1aef1c@mail.gmail.com>
Message-ID: <7aa29e791002022122o289bc807p55f5ed20ccbd91b7@mail.gmail.com>

Any updates for this question. Still now I unable to make an outbound call
please help me...............

Or give the idea to change from boost to isdn?

On Mon, Jan 25, 2010 at 11:20 AM, Thangappan.M <thangappan143 at gmail.com>wrote:

> The following link have the openzap.conf,openzap.conf.xml ,smg_pri.conf, output of oz list
> and oz dump.
>
> http://www.pastebin.org/81929
>
>
>
> On Mon, Jan 25, 2010 at 9:55 AM, Thangappan.M <thangappan143 at gmail.com>wrote:
>
>> Here I mentioned the link which has the details of
>> /etc/wanpipe/smg_pri.conf
>> http://www.pastebin.org/81895
>>
>>
>> On Sat, Jan 23, 2010 at 10:02 AM, Thangappan.M <thangappan143 at gmail.com>wrote:
>>
>>> Updated the latest version of freeswitch ( 1.0.5-20100121-0400) and run
>>> the wanrouter then freeswitch. While executing the freeswtich it said the
>>> following error.
>>>
>>> [ERR] zap_io.c:2562 Error loading/usr/local/freeswitch/mod/ozmod_sangoma_boost.so
>>>
>>> [/usr/local/freeswitch/mod/ozmod_sangoma_boost.so: cannot open shared object
>>> file: No such file or directory]
>>> [ERR] zap_io.c:2722 can't find 'sangoma_boost
>>>
>>>
>>>
>>>
>>> Searched about this in freeswitch mailing list and found one post was
>>> there regarding the same problem. Finally found the problem. I missed to
>>> install the SCTP packages. Installed it and compiled the freeswitch again
>>> now the inbound call was landed on freeswitch.
>>>
>>> But I am unable to make a outbound call. When I was trying the following
>>> was get.
>>>
>>> freeswitch at internal> originate openzap/smg_prid/a/9940464753 at g1openzap/smg_prid/a/9940464753 at g1
>>> -ERR NORMAL_CIRCUIT_CONGESTION
>>>
>>> 2010-01-23 10:00:45.688854 [WARNING] ozmod_sangoma_boost.c:348 TX EVENT:
>>> CALL_START:(80) [w1g1] CSid=2 Seq=2 Cn=[FreeSWITCH] Cd=[9940464753]
>>> Ci=[0000000000] Rdnis=[]
>>> freeswitch at internal> 2010-01-23 10:00:46.709355 [WARNING]
>>> ozmod_sangoma_boost.c:1373 RX EVENT (N): CALL_START_NACK:(82) [w256g256]
>>> Rc=0 CSid=2 Seq=2
>>> 2010-01-23 10:00:46.709355 [WARNING] sangoma_boost_client.c:220 TX EVENT
>>> (N): CALL_START_NACK_ACK:(83) [w1g1] Rc=0 CSid=2 Seq=3
>>> 2010-01-23 10:00:46.709355 [ERR] mod_openzap.c:1162 No channels available
>>> 2010-01-23 10:00:46.709355 [ERR] switch_ivr_originate.c:2411 Cannot
>>> create outgoing channel of type [openzap] cause: [NORMAL_CIRCUIT_CONGESTION]
>>> 2010-01-23 10:00:46.709355 [DEBUG] switch_ivr_originate.c:3211 Originate
>>> Resulted in Error Cause: 34 [NORMAL_CIRCUIT_CONGESTION]
>>>
>>> Please help me...........
>>>
>>>
>>>
>>> On Fri, Jan 22, 2010 at 10:15 AM, Thangappan.M <thangappan143 at gmail.com>wrote:
>>>
>>>> The following link have the openzap.conf,openzap.conf.xml ,smg_prid.conf
>>>> , debug log of mod_openzap , oz list and oz dump 1 output.
>>>>
>>>> http://pastebin.org/80095
>>>>
>>>>
>>>>
>>>> On Thu, Jan 21, 2010 at 10:34 AM, Thangappan.M <thangappan143 at gmail.com
>>>> > wrote:
>>>>
>>>>> OpenZap is loading the ss7 signalling type. As per your concern openzap
>>>>> does not know the details of the signalling then how it is loading the
>>>>> ss7_boost libraries?
>>>>>
>>>>> FreeSWITCH log:
>>>>> -----------------------------
>>>>> 2010-01-21 10:10:46.707844 [INFO] zap_io.c:2374 Configured 30
>>>>> channel(s)
>>>>> 2010-01-21 10:10:46.708600 [INFO] zap_io.c:2468 Loading SIG from
>>>>> /usr/local/freeswitch/mod/ozmod_ss7_boost.so
>>>>> 2010-01-21 10:10:46.709031 [INFO] zap_io.c:2584 auto-loaded 'ss7_boost'
>>>>> 2010-01-21 10:10:46.709466 [DEBUG] ss7_boost_client.c:124 Creating L=
>>>>> 127.0.0.65:53000 R=127.0.0.66:53000
>>>>> 2010-01-21 10:10:46.709834 [DEBUG] ss7_boost_client.c:124 Creating L=
>>>>> 127.0.0.65:53001 R=127.0.0.66:53001
>>>>> 2010-01-21 10:10:46.710424 [WARNING] ss7_boost_client.c:244 TX EVENT
>>>>> (P): SYSTEM_RESTART:(87) [w1g1] Rc=0 CSid=0 Seq=0
>>>>>
>>>>> The signalling type might be anything but when I used the oz list
>>>>> command it showed the span details. But I am unable to make a inbound and
>>>>> outbound call.
>>>>>
>>>>> Outbound call result:
>>>>> ============
>>>>> >  originate openzap/smg_prid/a/9940464753
>>>>> openzap/smg_prid/a/9843171457
>>>>> -ERR NORMAL_CIRCUIT_CONGESTION
>>>>>
>>>>> 2010-01-21 10:26:14.304816 [CRIT] ozmod_ss7_boost.c:244 SPAN is not
>>>>> online.
>>>>> freeswitch at internal> 2010-01-21 10:26:14.304816 [ERR]
>>>>> mod_openzap.c:1043 No channels available
>>>>> 2010-01-21 10:26:14.304816 [ERR] switch_ivr_originate.c:1510 Cannot
>>>>> create outgoing channel of type [openzap] cause: [NORMAL_CIRCUIT_CONGESTION]
>>>>> 2010-01-21 10:26:14.304816 [DEBUG] switch_ivr_originate.c:2138
>>>>> Originate Resulted in Error Cause: 34 [NORMAL_CIRCUIT_CONGESTION]
>>>>>
>>>>> Inbound call result:
>>>>> -----------------------------
>>>>>
>>>>>      I made incoming call for the dial plan which is specified in the
>>>>> earlier post at that time it said the number is busy. We did the packet
>>>>> capture using the following command.
>>>>>
>>>>> wanpipemon -i w1g1 -pcap -pcap file isdn.pcap -port ISDN -full -systime
>>>>> -c trd
>>>>>
>>>>> Here I attached the pcap file for that.
>>>>>
>>>>>
>>>>> Where I did mistake or Did I miss any thing to do?
>>>>> Please help me.......
>>>>>
>>>>>
>>>>>
>>>>> On Wed, Jan 20, 2010 at 7:40 PM, Thangappan.M <thangappan143 at gmail.com
>>>>> > wrote:
>>>>>
>>>>>>
>>>>>>       I noticed the 'oz list' output  in that span type is 'ss7
>>>>>> (boost)'. How can I change this to isdn?
>>>>>>
>>>>>>
>>>>>>
>>>>>> On Wed, Jan 20, 2010 at 12:43 PM, Thangappan.M <
>>>>>> thangappan143 at gmail.com> wrote:
>>>>>>
>>>>>>> I found the error in it. The file name is used as openzap.conf.xml (
>>>>>>> smg_prid is specified here) and another file name as openzap.conf.wiki.xml (
>>>>>>> PRI_1 span is specified here ). FreeSWITCH referred the PRI_1 span from
>>>>>>> openzap.conf.wiki.xml  file.
>>>>>>>
>>>>>>> Now the another problem is raised here.
>>>>>>> When I was using oz list command , the details of the smg_prid shown.
>>>>>>> When I was using 'oz dump smg_prid' command it shows all the channels'
>>>>>>> details. But all the channels' states are DOWN. why? How can I make it the
>>>>>>> states to UP?
>>>>>>>
>>>>>>> When I was making the call , the number is busy message was get. The
>>>>>>> call was not  at all landed to the freeswitch.
>>>>>>>
>>>>>>> Dial plan Example:
>>>>>>> -------------------------------
>>>>>>>  <extension name="PRI_Testing">
>>>>>>>         <condition field="destination_number"
>>>>>>> expression="^39114600$">
>>>>>>>                 <action application="playback"
>>>>>>> data="ivr-welcome_to_freeswitch"/>
>>>>>>>         </condition>
>>>>>>>   </extension>
>>>>>>>
>>>>>>> Please help me...........
>>>>>>>
>>>>>>> *Output Reference:*
>>>>>>> http://pastebin.org/79074
>>>>>>>
>>>>>>>
>>>>>>> On Wed, Jan 20, 2010 at 11:25 AM, Thangappan.M <
>>>>>>> thangappan143 at gmail.com> wrote:
>>>>>>>
>>>>>>>> Dear all,
>>>>>>>>
>>>>>>>>          I have successfully configured wanpipe with freeswitch.
>>>>>>>> When I was the running wancfg_fs script the following files openzap.conf ,
>>>>>>>> autoload_confg/openzap.conf.xml , /etc/wanpipe/wanpipe1.xml, smg_pri.conf
>>>>>>>> are created.
>>>>>>>>
>>>>>>>>          I started the wanrouter command then executed the
>>>>>>>> freeswitch.
>>>>>>>>          When I was executing freeswitch mod_openzap.c said the
>>>>>>>> error as  "Error  for finding the span id. name:PRI_1".
>>>>>>>>          But in the openzap.conf and openzap.conf.xml files the span
>>>>>>>> name is smg_prid.
>>>>>>>>
>>>>>>>>          Why the freeswitch is referring the span name as PRI_1 ?
>>>>>>>>          Whether this has to configured in anywhere?
>>>>>>>>
>>>>>>>>          In the freeswitch CLI using oz command I tried to dump the
>>>>>>>> PRI_1 span id but it said te error  as "PRI_1 is not found".  When I was
>>>>>>>> trying the command 'oz dump smg_prid'  all the channel states and details
>>>>>>>> shown.
>>>>>>>>
>>>>>>>>         It seems that smg_prid span configured in openzap perfectly
>>>>>>>> (Its my assumption). Then Why freeswitch is referring the span name as
>>>>>>>> PRI_1.
>>>>>>>>
>>>>>>>> DID I MAKE ANY MISTAKE OR DID I MISS ANYTHING TO DO?
>>>>>>>>
>>>>>>>> Could anyone please help me?
>>>>>>>>
>>>>>>>> REFERENCE:
>>>>>>>>
>>>>>>>> openzap.conf
>>>>>>>> [span wanpipe smg_prid]
>>>>>>>> name => smg_prid
>>>>>>>> trunk_type =>e1
>>>>>>>> b-channel => 1:1-15
>>>>>>>> b-channel => 1:17-31
>>>>>>>>
>>>>>>>>
>>>>>>>> openzap.conf.xml
>>>>>>>> <configuration name="openzap.conf" description="OpenZAP
>>>>>>>> Configuration">
>>>>>>>>   <settings>
>>>>>>>>     <param name="debug" value="0"/>
>>>>>>>>     <!--<param name="hold-music" value="$${moh_uri}"/>-->
>>>>>>>>     <!--<param name="enable-analog-option" value="call-swap"/>-->
>>>>>>>>     <!--<param name="enable-analog-option" value="3-way"/>-->
>>>>>>>>   </settings>
>>>>>>>>   <boost_spans>
>>>>>>>>         <span id="smg_prid">
>>>>>>>>         <!--<param name="hold-music" value="$${moh_uri}"/>-->
>>>>>>>>         <param name="dialplan" value="XML"/>
>>>>>>>>         <param name="context" value="default"/>
>>>>>>>>          <!-- regex to stop dialing when it matches -->
>>>>>>>>         <!--<param name="dial-regex" value="5555"/>-->
>>>>>>>>         <!-- regex to stop dialing when it does not match -->
>>>>>>>>         <!--<param name="fail-dial-regex" value="^5"/>-->
>>>>>>>>         </span>
>>>>>>>>  </boost_spans>
>>>>>>>>  <analog_spans>
>>>>>>>>  </analog_spans>
>>>>>>>> </configuration>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> --
>>>>>>>> Regards,
>>>>>>>> Thangappan.M
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> --
>>>>>>> Regards,
>>>>>>> Thangappan.M
>>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>> Regards,
>>>>>> Thangappan.M
>>>>>>
>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> Regards,
>>>>> Thangappan.M
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> Regards,
>>>> Thangappan.M
>>>>
>>>
>>>
>>>
>>> --
>>> Regards,
>>> Thangappan.M
>>>
>>
>>
>>
>> --
>> Regards,
>> Thangappan.M
>>
>
>
>
> --
> Regards,
> Thangappan.M
>



-- 
Regards,
Thangappan.M
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100203/7b84ebe7/attachment-0002.html 

From nagalenoj at gmail.com  Wed Feb  3 01:12:03 2010
From: nagalenoj at gmail.com (Nagalenoj H.)
Date: Wed, 3 Feb 2010 14:42:03 +0530
Subject: [Freeswitch-users] A leg disconnects after uuid_bridge?
In-Reply-To: <191c3a031002022108ta22287ev40fc36420abba2e5@mail.gmail.com>
References: <fe67a3341002021934m548cde89p4d48381c5abe02a0@mail.gmail.com>
	<191c3a031002022108ta22287ev40fc36420abba2e5@mail.gmail.com>
Message-ID: <fe67a3341002030112yeea379u90c22c482614c57@mail.gmail.com>

I've used intercept application instead of uuid_bridge. I got the call
bridged with the given uuid. But, similar to uuid_bridge, A leg is getting
disconnected when B leg terminates.
I've did the same steps posted above, but used intercept instead of
uuid_bridge. Am I right?!

Tried using the hangup_after_bridge=false, but didn't see any difference.

On Wed, Feb 3, 2010 at 10:38 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> Tell one leg to execute intercept on the other instead.
>
> On Feb 2, 2010 9:40 PM, "Nagalenoj H." <nagalenoj at gmail.com> wrote:
>
> Dear friends,
>   In event socket, Why the session is closed for A leg when I do a
> uuid_bridge with another uuid.
> I've done the following operations(In nc),
>   * Dial to he socket extension.
>   * connect to the call.
>   * Answered the call.
>   * Bridged with extension X. When B leg terminates, A leg continues to be
> alive.
>   * Create an uuid.
>   * Originated a call with origination_uuid and parked the leg.
>   * I did a uuid_bridge for these 2 legs.
>   * When B leg terminates the call, A leg is also getting exited.
>
> I've tried setting the hangup_after_bridge to false explicitly. But, A leg
> is getting exited.
> I need it for further processing.
>
> What is the way in which I can keep the A leg alive after uuid_bridge??
>
> --
> Regards,
> Nagalenoj H.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100203/5cc696ce/attachment-0002.html 

From Suneel.Papineni at mettoni.com  Wed Feb  3 02:56:21 2010
From: Suneel.Papineni at mettoni.com (Suneel Papineni)
Date: Wed, 3 Feb 2010 10:56:21 -0000
Subject: [Freeswitch-users] Attendant call transfer
In-Reply-To: <07247F72-DE53-4028-AFBF-BB5EB23FEA7B@freeswitch.org>
References: <3181A30B8C35AB4AA8577B78DDF461380668B5D3@nickel.mettonigroup.com>
	<07247F72-DE53-4028-AFBF-BB5EB23FEA7B@freeswitch.org>
Message-ID: <3181A30B8C35AB4AA8577B78DDF461380668B6E7@nickel.mettonigroup.com>

Hi Brian,

 

My apologies  for not sending required information.

 

I am using freeswitch 1.0.5_20100104-0400 and running on a Windows XP machine. I have written a .NET application to communicate with FS through Event Sockets.

 

In the scenario, once call is connected between Caller A and Caller B, I am expecting FS to come out of loop. 

 

Thanks in advance for helping me.

 

Thanks & Regards

Suneel

 

 

From: Brian West [mailto:brian at freeswitch.org] 
Sent: 02 February 2010 18:17
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Attendant call transfer

 

Suneel,

            After printing 100 copies of this email It dawned on me that you failed to include any details about what SVN revision you're using.  If you can reply with that info I can promptly print out 100 more copies and see if we can find your problem.

 

Thanks,

Brian

PS: just kidding about the printing part, but the svn rev would be helpful.

 

On Feb 2, 2010, at 12:09 PM, Suneel Papineni wrote:





Hi,

 

I am trying to establish attendant call transfer using event sockets.

1.       A call has come into Freeswitch from an external Gateway  and this call is parked (it is configured to park all calls coming to freeswitch) {Caller A ? FS}

2.       Once the call is parked, I am sending a command to originate a call to another number connected to external gateway. {FS ? Caller B}. Call is established between FS and caller B. ("api originate sofia/external/<another number>@<gateway IP address> 9999")

3.       On receiving event message as "Application: Answer", I am sending another command to bridge call between A & B. ("api uuid_bridge <callerA UUID> <caller UUID>")

4.       With this call is established between A & B, but there is a huge delay (appox 30 secs).

 

I believe that FS is still in the call and might be this is creating  delay (not sure).

 

Could you please tell me if I am doing something wrong or process to achieve this scenario working.

 

I tried in to transfer the call instead of bridging  using the command ("uuid_transfer <callerA UUID> intercept: <caller UUID> inline"), but the response is same as above with huge delay.

 

Thanks & Regards

Suneel

 

*************************************************************************
Please consider the environment before printing this e-mail
*************************************************************************
This email and any files transmitted with it are confidential and
intended solely for the use of the individual or entity to whom they
are addressed. If you have received this email in error please notify
the system manager.  http://www.mettoni.com
 
Mettoni Ltd
Registered in England and Wales: 4485956
9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN
*************************************************************************

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

 


*************************************************************************
Please consider the environment before printing this e-mail
*************************************************************************
This email and any files transmitted with it are confidential and
intended solely for the use of the individual or entity to whom they
are addressed. If you have received this email in error please notify
the system manager.  http://www.mettoni.com

Mettoni Ltd
Registered in England and Wales: 4485956
9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN
*************************************************************************

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100203/6871558f/attachment-0002.html 

From nicolas at medularis.com  Wed Feb  3 03:11:49 2010
From: nicolas at medularis.com (Nicolas Brenner)
Date: Wed, 3 Feb 2010 08:11:49 -0300
Subject: [Freeswitch-users] Ringback after before` Bridge
In-Reply-To: <91DD5271-AE00-454A-A5E5-9AA933E0B459@freeswitch.org>
References: <c55d46951002021727v76e79adcg2281347ff8a0458f@mail.gmail.com>
	<91DD5271-AE00-454A-A5E5-9AA933E0B459@freeswitch.org>
Message-ID: <1b46b4e81002030311x48db5d8et4822bd8bd97c0094@mail.gmail.com>

Not sure about using ESL, but here's an example on how to do it with Lua:

http://wiki.freeswitch.org/wiki/Bridging_two_calls_with_retry

Simplifying the code on that example though, here are the basics:

session1 = freeswitch.Session(ostr1);
if (session1:ready()) then
-- Set ringback
session1:setVariable("ringback", "%(2000,4000,440,480)");

session2 = freeswitch.Session(ostr2, session1);
if (session2:ready()) then
freeswitch.bridge(session1, session2);

-- Hangup session2 if session1 is over
if (session2:ready()) then session2:hangup(); end
end
 -- hangup when done
    if (session1:ready()) then session1:hangup(); end
end


ostr1 and ostr2 should be your dialstrings, something like:
{ignore_early_media=true,originate_timeout=90,hangup_after_bridge=true}sofia/gateway/yourgateway/phonenumber


On Wed, Feb 3, 2010 at 12:19 AM, Michael S Collins <msc at freeswitch.org>wrote:

> Try the transfer_ringback var. Check the wiki for details.
> -MC
>
> Sent from my iPhone
>
> On Feb 2, 2010, at 5:27 PM, Max Bridgewater
> <max.bridgewater at gmail.com> wrote:
>
> > Hi,
> >
> > I'm trying to place a call to A and then bridge it to B. The problem
> > I'm having right now is that after A answers and while dialing B is
> > being dialed or rining, I want to send A a ringing tone. I don't
> > succeed in doing this. No tone/ringback is being sent to A.  Here is
> > what i did using ESL:
> >
> >  api originate {ringback=\'%(400,200,400,450);
> > %(400,2200,400,450)\',transfer_ringback=\'%(400,200,400,450);
> > %(400,2200,400,450)\',origination_caller_id_number=4156781020}sofia/
> > gateway/voipms/4152309090 &park()
> >
> > To bridge, I then send the message:
> >
> > sendmsg e9dae14c-e473-466e-9d65-704e36a82e5f
> > call-command: execute
> > execute-app-name: bridge
> > execute-app-arg: {{ringback=\'%(400,200,400,450);
> > %(400,2200,400,450\'},origination_caller_id_number=4152309090 }sofia/
> > gateway/voipms/4156781020
> >
> > any idea?
> >
> > Max.
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> > users
> > http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100203/9e7f221e/attachment-0002.html 

From moizchinoy at gmail.com  Wed Feb  3 04:12:02 2010
From: moizchinoy at gmail.com (Moiz Chinoy)
Date: Wed, 3 Feb 2010 16:12:02 +0400
Subject: [Freeswitch-users] Dialogic Board (JCT & DM3) Supported?
In-Reply-To: <D7BC4112-CA0B-4825-8ADA-A8F228C2F201@jerris.com>
References: <29b888f81002010308uf875b53h838453b6e77e9a9e@mail.gmail.com>
	<D7BC4112-CA0B-4825-8ADA-A8F228C2F201@jerris.com>
Message-ID: <29b888f81002030412x7856f1d4p3612ce49c9d23d8f@mail.gmail.com>

Sometime back it was posted that following cards are supported by
Freeswitch. Can anyone please guide me. We have a JCT and DMV card
available so I was thinking if we can do anything useful with it.

Analog cards: D/41JCT-LS, D/120JCT-LS  (jct serie)
Digital cards: D/600JCT-1E1 and DMV serie.

On Tue, Feb 2, 2010 at 12:14 PM, Michael Jerris <mike at jerris.com> wrote:
> Nope
>
> On Feb 1, 2010, at 6:08 AM, Moiz Chinoy wrote:
>> Can anyone please advise that whether Dialogic boards (JCT and DM3)
>> are supported by FS.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Regards,
Moiz Chinoy.



From steveu at coppice.org  Wed Feb  3 04:38:11 2010
From: steveu at coppice.org (Steve Underwood)
Date: Wed, 03 Feb 2010 20:38:11 +0800
Subject: [Freeswitch-users] Dialogic Board (JCT & DM3) Supported?
In-Reply-To: <29b888f81002030412x7856f1d4p3612ce49c9d23d8f@mail.gmail.com>
References: <29b888f81002010308uf875b53h838453b6e77e9a9e@mail.gmail.com>	<D7BC4112-CA0B-4825-8ADA-A8F228C2F201@jerris.com>
	<29b888f81002030412x7856f1d4p3612ce49c9d23d8f@mail.gmail.com>
Message-ID: <4B696E33.5040103@coppice.org>

Hi Moiz,

On 02/03/2010 08:12 PM, Moiz Chinoy wrote:
> Sometime back it was posted that following cards are supported by
> Freeswitch. Can anyone please guide me. We have a JCT and DMV card
> available so I was thinking if we can do anything useful with it.
>
> Analog cards: D/41JCT-LS, D/120JCT-LS  (jct serie)
> Digital cards: D/600JCT-1E1 and DMV serie.
>    
"For it is written" :-)

Someone randomly posting rubbish to the mailing list doesn't make it 
true. Those Dialogic cards have never been supported by Freeswitch.
> On Tue, Feb 2, 2010 at 12:14 PM, Michael Jerris<mike at jerris.com>  wrote:
>    
>> Nope
>>
>> On Feb 1, 2010, at 6:08 AM, Moiz Chinoy wrote:
>>      
>>> Can anyone please advise that whether Dialogic boards (JCT and DM3)
>>> are supported by FS.
>>>        
Steve




From dftoro at yahoo.com  Wed Feb  3 05:42:38 2010
From: dftoro at yahoo.com (Diego Toro)
Date: Wed, 3 Feb 2010 05:42:38 -0800 (PST)
Subject: [Freeswitch-users] Dialogic Board (JCT & DM3) Supported?
In-Reply-To: <29b888f81002030412x7856f1d4p3612ce49c9d23d8f@mail.gmail.com>
Message-ID: <963917.92014.qm@web33505.mail.mud.yahoo.com>

This would be a good opportunity to start the support of this hardware. The problem is that the "System Release" (API) has cost of licensing, this to avoid having to start from scratch. Now, this compared with good quality Sangoma hardware, Dialogic would not be competitive. Would be good to ask to Eicom people if they would be interested in the subject.

Diego Toro
http://lacarretade.blogspot.com/


--- On Wed, 2/3/10, Moiz Chinoy <moizchinoy at gmail.com> wrote:

> From: Moiz Chinoy <moizchinoy at gmail.com>
> Subject: Re: [Freeswitch-users] Dialogic Board (JCT & DM3) Supported?
> To: freeswitch-users at lists.freeswitch.org
> Date: Wednesday, February 3, 2010, 7:12 AM
> Sometime back it was posted that
> following cards are supported by
> Freeswitch. Can anyone please guide me. We have a JCT and
> DMV card
> available so I was thinking if we can do anything useful
> with it.
> 
> Analog cards: D/41JCT-LS, D/120JCT-LS? (jct serie)
> Digital cards: D/600JCT-1E1 and DMV serie.
> 
> On Tue, Feb 2, 2010 at 12:14 PM, Michael Jerris <mike at jerris.com>
> wrote:
> > Nope
> >
> > On Feb 1, 2010, at 6:08 AM, Moiz Chinoy wrote:
> >> Can anyone please advise that whether Dialogic
> boards (JCT and DM3)
> >> are supported by FS.
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> 
> 
> 
> -- 
> Regards,
> Moiz Chinoy.
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 


      



From msc at freeswitch.org  Wed Feb  3 06:05:16 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 3 Feb 2010 06:05:16 -0800
Subject: [Freeswitch-users] A leg disconnects after uuid_bridge?
In-Reply-To: <fe67a3341002030112yeea379u90c22c482614c57@mail.gmail.com>
References: <fe67a3341002021934m548cde89p4d48381c5abe02a0@mail.gmail.com>
	<191c3a031002022108ta22287ev40fc36420abba2e5@mail.gmail.com>
	<fe67a3341002030112yeea379u90c22c482614c57@mail.gmail.com>
Message-ID: <87f2f3b91002030605j7a1f9a02g665ba043c6ed7f7b@mail.gmail.com>

On Wed, Feb 3, 2010 at 1:12 AM, Nagalenoj H. <nagalenoj at gmail.com> wrote:

> I've used intercept application instead of uuid_bridge. I got the call
> bridged with the given uuid. But, similar to uuid_bridge, A leg is getting
> disconnected when B leg terminates.
> I've did the same steps posted above, but used intercept instead of
> uuid_bridge. Am I right?!
>
> Tried using the hangup_after_bridge=false, but didn't see any difference.
>
> Are you on the latest SVN of FreeSWITCH? Be sure to "make current" and try
again. If the issue remains then capture a complete debug log of the call
from start to finish and also pastebin your script so others can test and
analyze.
-MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100203/45071da9/attachment-0002.html 

From marketing at cluecon.com  Wed Feb  3 06:28:04 2010
From: marketing at cluecon.com (Michael Collins)
Date: Wed, 3 Feb 2010 06:28:04 -0800
Subject: [Freeswitch-users] ClueCon MMX - Call For Speakers!
Message-ID: <87f2f3b91002030628n3b3bf512x68947758cf042ea@mail.gmail.com>

Hello everyone!

We are gearing up for ClueCon 2010 in August later this year. We are making
the necessary arrangements for the conference facilities and rooming. Things
are beginning to fall into place. Now we need to hear from you.

We would like to put out a call for speakers for this year's event. Please
contact us if you or your organization would like to give a presentation at
ClueCon this year. We want to get the speakers scheduled as early as
possible. Keep in mind that those organizations which sponsor ClueCon will
be given the highest priority when it comes to scheduling. Please contact
Brian West to discuss sponsorship opportunities for this year's event.

We also would like to hear from the conference attendees: what would you
like to see this year? Please give us your input. ClueCon is, of course, "By
Developers, For Developers." However, developers come in all shapes and
sizes and we would like have something for everyone. Please tell us what
would make ClueCon MMX the best conference of the year!

Stay tuned for more announcements. We look forward to hearing from you and
seeing everyone this August in Chicago.

-ClueCon team
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100203/24dc1efc/attachment-0002.html 

From msc at freeswitch.org  Wed Feb  3 06:55:47 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 3 Feb 2010 06:55:47 -0800
Subject: [Freeswitch-users] Let's buy the FreeSWITCH developers dinner!
Message-ID: <87f2f3b91002030655x315159ads102f269dfdab200d@mail.gmail.com>

Hello all!

This is a reminder that next week the FreeSWITCH development team is
gathering together in one location for the final push to get version 1.0.5
released. You can help facilitate the timely release of the latest version
by helping to buy dinner for the FreeSWITCH developers. Remember, it's not
just the three core developers who are meeting together. There will be eight
FreeSWITCH team members gathering. Let's all pitch in a few dollars each and
give them a nice dinner! It's a great way to say thanks for all the hard
work they've done: building FreeSWITCH, answering questions on the mailing
list, and spending many hours in the IRC channel.

You can use the PayPal link on the main website. (http://www.freeswitch.org)
Alternatively, if PayPal is not available to you then please contact Brian
West (brian at freeswitch.org) to discuss alternate ways of donating.

Let's really pull together and support the guys for all of their hard work.
A nice meal paid for by the community would be greatly appreciated!

-Michael
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100203/277e82ea/attachment-0002.html 

From moises.silva at gmail.com  Wed Feb  3 07:06:59 2010
From: moises.silva at gmail.com (Moises Silva)
Date: Wed, 3 Feb 2010 10:06:59 -0500
Subject: [Freeswitch-users] Need Help to setup freeswitch with sangoma
	card
In-Reply-To: <7aa29e791002022122o289bc807p55f5ed20ccbd91b7@mail.gmail.com>
References: <7aa29e791001192155l3c1f06e6w69769a69c9b761a1@mail.gmail.com>
	<7aa29e791001192313p73b20c8fk7e9b78e577c305a9@mail.gmail.com>
	<7aa29e791001200610m5a7b64cdx356e527ea7285b23@mail.gmail.com>
	<7aa29e791001202104n276ca48bo56d55ca82f3aac0f@mail.gmail.com>
	<7aa29e791001212045u20d85213wa0c8523761fee826@mail.gmail.com>
	<7aa29e791001222032t6f381e34x9f3c5eb099af9ba8@mail.gmail.com>
	<7aa29e791001242025y2adc91aes622bd22ea8d5ae1c@mail.gmail.com>
	<7aa29e791001242150o3a582e2end69f1f5e9f1aef1c@mail.gmail.com>
	<7aa29e791002022122o289bc807p55f5ed20ccbd91b7@mail.gmail.com>
Message-ID: <c4d05cbe1002030706x61342ee4hc9d5fe0389d123fc@mail.gmail.com>

Boost has been working fine, so there is no point in switching.

Try removing the d-channel from openzap.conf, boost spans do not need
d-channel declared, because that is done through the signaling binary
(sangoma_prid), declaring it is causing 2 different processes to open the
same channel, at best, data will be missing from one process or the other.

If you still cannot make calls after removing the d-channel from
openzap.conf (and restarting FreeSWITCH and sangoma_prid using smg_ctrl
script), then pastebin a debug log for both FreeSWITCH and sangoma_prid
after a single call attempt.

-- 
Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com

On Wed, Feb 3, 2010 at 12:22 AM, Thangappan.M <thangappan143 at gmail.com>wrote:

> Any updates for this question. Still now I unable to make an outbound call
> please help me...............
>
> Or give the idea to change from boost to isdn?
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100203/847db260/attachment-0002.html 

From jerry.richards at teotech.com  Wed Feb  3 08:40:30 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Wed, 3 Feb 2010 08:40:30 -0800
Subject: [Freeswitch-users] FS Core Dump
In-Reply-To: <c4d05cbe1002021742m546286bcxc96930990e57d9e6@mail.gmail.com>
References: <40B8BCC90CB0486EA7730EB86BA1D1DB@greyhawk.tonecommander.com>
	<c4d05cbe1002021742m546286bcxc96930990e57d9e6@mail.gmail.com>
Message-ID: <C1EEC6F570BD4302ABDB7D4A0BEE7803@greyhawk.tonecommander.com>

Okay. I will get freeswitch-1.0.5-20100202-0400.tar.gz, which is one day
later than what I currently have.  I don't get openzap separately, because
it is included with this tarball.  True?
 
Thanks And Best Regards,
Jerry


  _____  

From: Moises Silva [mailto:moises.silva at gmail.com] 
Sent: Tuesday, February 02, 2010 5:42 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] FS Core Dump


On Tue, Feb 2, 2010 at 6:55 PM, Jerry Richards <jerry.richards at teotech.com>
wrote:


I spoke too soon.  Using Freeswitch version Feb 01, 2010 and Wanpipe Driver
ftp://ftp.sangoma.com/linux/custom/DavidYS/wanpipe-3.5.10.smg_pri.1.tgz, the
FS core will dump if I call my external cell phone, answer and then hangup.
See the pastebin: http://pastebin.freeswitch.org/12035.




Make sure you have latest revision of openzap, a few revisions ago I saw a
commit that set the zchan to null on hangup and you have a core dump on
hangup where the zchan is null, however on the latest trunk zchan is no
longer set to null on SIGEVENT_STOP ... suspicious ... if you can reproduce
with latest openzap trunk (at least revision 1021) then let me know and I
will take a look.

 -- 
Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com


-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100203/cd871bd6/attachment-0002.html 

From m.sobkow at marketelsystems.com  Wed Feb  3 10:49:11 2010
From: m.sobkow at marketelsystems.com (Mark Sobkow)
Date: Wed, 03 Feb 2010 12:49:11 -0600
Subject: [Freeswitch-users] Has anyone had Freeswitch crash on loading
 mod_voicemail?
In-Reply-To: <4B5E4A35.3060803@marketelsystems.com>
References: <4B5E4A35.3060803@marketelsystems.com>
Message-ID: <4B69C527.4090808@marketelsystems.com>

Mark Sobkow wrote:
> I tried a "load mod_voicemail" in fs_cli, hoping to see what 
> configuration section it requested from Erlang, but instead of loading 
> the module, Freeswitch crashed without any error messages.  SVN 15188 
> built on Ubuntu Hardy 32-bit.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>   
Updated Freeswitch to svn16561 this morning, captured the backtrace, and 
attached it to JIRA as requested.

-- 
Mark Sobkow
Senior Developer
MarkeTel Multi-Line Dialing Systems LTD.
428 Victoria Ave
Regina, SK S4N-0P6
Toll-Free: 800-289-8616-X533
Local: 306-359-6893-X533
Fax: 306-359-6879
Email: m.sobkow at marketelsystems.com
Web: http://www.marketelsystems.com 




From anthony.minessale at gmail.com  Wed Feb  3 11:19:30 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 3 Feb 2010 13:19:30 -0600
Subject: [Freeswitch-users] Has anyone had Freeswitch crash on loading
	mod_voicemail?
In-Reply-To: <4B69C527.4090808@marketelsystems.com>
References: <4B5E4A35.3060803@marketelsystems.com>
	<4B69C527.4090808@marketelsystems.com>
Message-ID: <191c3a031002031119q1efc36bwd5484aa0d640682d@mail.gmail.com>

Clear bug in mod_erlang.  Author has been notified.


On Wed, Feb 3, 2010 at 12:49 PM, Mark Sobkow
<m.sobkow at marketelsystems.com>wrote:

> Mark Sobkow wrote:
> > I tried a "load mod_voicemail" in fs_cli, hoping to see what
> > configuration section it requested from Erlang, but instead of loading
> > the module, Freeswitch crashed without any error messages.  SVN 15188
> > built on Ubuntu Hardy 32-bit.
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> Updated Freeswitch to svn16561 this morning, captured the backtrace, and
> attached it to JIRA as requested.
>
> --
> Mark Sobkow
> Senior Developer
> MarkeTel Multi-Line Dialing Systems LTD.
> 428 Victoria Ave
> Regina, SK S4N-0P6
> Toll-Free: 800-289-8616-X533
> Local: 306-359-6893-X533
> Fax: 306-359-6879
> Email: m.sobkow at marketelsystems.com
> Web: http://www.marketelsystems.com
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100203/96aa9f73/attachment-0002.html 

From jerry.richards at teotech.com  Wed Feb  3 11:23:05 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Wed, 3 Feb 2010 11:23:05 -0800
Subject: [Freeswitch-users] FS Core Dump
In-Reply-To: <C1EEC6F570BD4302ABDB7D4A0BEE7803@greyhawk.tonecommander.com>
References: <40B8BCC90CB0486EA7730EB86BA1D1DB@greyhawk.tonecommander.com><c4d05cbe1002021742m546286bcxc96930990e57d9e6@mail.gmail.com>
	<C1EEC6F570BD4302ABDB7D4A0BEE7803@greyhawk.tonecommander.com>
Message-ID: <F940A2D994FA41EF812D9956EEB9A039@greyhawk.tonecommander.com>

Okay.  I got the latest trunk at 9:39AM PST and FS does not crash when I
call my cell phone, answer, and hangup.  I am using the
<ftp://ftp.sangoma.com/linux/custom/DavidYS/wanpipe-3.5.10.smg_pri.1.tgz>
ftp://ftp.sangoma.com/linux/custom/DavidYS/wanpipe-3.5.10.smg_pri.1.tgz
driver.
 
Thanks and Best Regards,
Jerry
 



  _____  

From: Jerry Richards [mailto:jerry.richards at teotech.com] 
Sent: Wednesday, February 03, 2010 8:41 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] FS Core Dump


Okay. I will get freeswitch-1.0.5-20100202-0400.tar.gz, which is one day
later than what I currently have.  I don't get openzap separately, because
it is included with this tarball.  True?
 
Thanks And Best Regards,
Jerry


  _____  

From: Moises Silva [mailto:moises.silva at gmail.com] 
Sent: Tuesday, February 02, 2010 5:42 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] FS Core Dump


On Tue, Feb 2, 2010 at 6:55 PM, Jerry Richards <jerry.richards at teotech.com>
wrote:


I spoke too soon.  Using Freeswitch version Feb 01, 2010 and Wanpipe Driver
ftp://ftp.sangoma.com/linux/custom/DavidYS/wanpipe-3.5.10.smg_pri.1.tgz, the
FS core will dump if I call my external cell phone, answer and then hangup.
See the pastebin: http://pastebin.freeswitch.org/12035.




Make sure you have latest revision of openzap, a few revisions ago I saw a
commit that set the zchan to null on hangup and you have a core dump on
hangup where the zchan is null, however on the latest trunk zchan is no
longer set to null on SIGEVENT_STOP ... suspicious ... if you can reproduce
with latest openzap trunk (at least revision 1021) then let me know and I
will take a look.

 -- 
Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com


-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100203/a56882f6/attachment-0002.html 

From m.sobkow at marketelsystems.com  Wed Feb  3 11:40:45 2010
From: m.sobkow at marketelsystems.com (Mark Sobkow)
Date: Wed, 03 Feb 2010 13:40:45 -0600
Subject: [Freeswitch-users] Has anyone had Freeswitch crash on loading
 mod_voicemail?
In-Reply-To: <4B5E4A35.3060803@marketelsystems.com>
References: <4B5E4A35.3060803@marketelsystems.com>
Message-ID: <4B69D13D.7030205@marketelsystems.com>

Mark Sobkow wrote:
> I tried a "load mod_voicemail" in fs_cli, hoping to see what 
> configuration section it requested from Erlang, but instead of loading 
> the module, Freeswitch crashed without any error messages.  SVN 15188 
> built on Ubuntu Hardy 32-bit.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>   
Did a little digging through the traceback for this problem, and it 
looks to me like Freeswitch is passing an invalid UUID to the code that 
tries to get the configuration from Erlang.

http://jira.freeswitch.org/browse/FSCORE-542



-- 
Mark Sobkow
Senior Developer
MarkeTel Multi-Line Dialing Systems LTD.
428 Victoria Ave
Regina, SK S4N-0P6
Toll-Free: 800-289-8616-X533
Local: 306-359-6893-X533
Fax: 306-359-6879
Email: m.sobkow at marketelsystems.com
Web: http://www.marketelsystems.com 




From fvillarroel at yahoo.com  Wed Feb  3 12:29:54 2010
From: fvillarroel at yahoo.com (FERNANDO VILLARROEL)
Date: Wed, 3 Feb 2010 12:29:54 -0800 (PST)
Subject: [Freeswitch-users] max calls from a gateway
Message-ID: <867692.53487.qm@web34301.mail.mud.yahoo.com>

Dear.

How i can do for limits inbound calls from a gateway:


My config is like this:

~sip_profiles/external/

gateway1.xml

<include>
    <gateway name="foo">
      <param name="username" value="foo"/>
      <param name="password" value="foo"/>
      <param name="realm" value="xx.xxx.xxx.xx"/>
      <param name="proxy" value="xx.xxx.xxx.xx/>
      <param name="register" value="false"/>
      <param name="context" value="public"/>
      <param name="max_calls" value="30"/>-->
    </gateway>
</include>

It?s fine?

Fernando


      



From john at acsol.net  Wed Feb  3 14:30:37 2010
From: john at acsol.net (John)
Date: Wed, 03 Feb 2010 15:30:37 -0700
Subject: [Freeswitch-users] PAP2T issue
Message-ID: <4B69F90D.1070503@acsol.net>

I can register SNOM phones fine, one the same network, I am trying to 
get a Linksys PAP2T-NA registered. Anyone have a configuration example? 
Thanks




From christian.loeschenkohl at xpirio.com  Wed Feb  3 14:33:17 2010
From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=)
Date: Wed, 03 Feb 2010 23:33:17 +0100
Subject: [Freeswitch-users] adding sip header without X-
Message-ID: <4B69F9AD.8090904@xpirio.com>

hello

do anybody know a way to add "Alert-Info: <http://www.notused.com>;info=alert-group;x-line-id=0"
as a custom sip header to the invite message?

<action application="set" data="sip_h_X-Alert-Info=<http://www.notused.com>\;info=alert-internal\;x-line-id=0"/>
works but "X-Alert-Info" as a header isn't usefull at all

the purpose is to distinguish internal and external calls in a pbx. it is also described here
http://wiki.snom.com/Web_Interface/V7/Preferences/Alert-Info_Ringer

there was also a posting on the list by Kristian Kielhofner who suggested a very flexible solution

-----
mail from: 2009-10-09 21:10

2)  Make the behavior configurable with a channel variable and/or
     sofia config option: {sip_pass_headers=all|none|X-MyCustomHeaderByName}
-----

br


-- 
Ing. Christian L?schenkohl
Technische Leitung, Forschung & Entwicklung VoIP

xpirio
Telekommunikation & Service GmbH
Lakeside B04
9020 Klagenfurt
Austria

T  +43 (0) 5 77 11 - 1000
F  +43 (0) 5 77 11 - 1002
E  christian.loeschenkohl at xpirio.com



From john at acsol.net  Wed Feb  3 15:23:05 2010
From: john at acsol.net (John)
Date: Wed, 03 Feb 2010 16:23:05 -0700
Subject: [Freeswitch-users] PAP2T issue
In-Reply-To: <4B69F90D.1070503@acsol.net>
References: <4B69F90D.1070503@acsol.net>
Message-ID: <4B6A0559.2040200@acsol.net>

Found two issues and solved the problem. By default, the PAP2T has 
provisioning turned on, and won't try to register if the provisioning 
server isn't found. Turned that off. Secondly, turned off Stun and 
enabled rport and it registered fine. Hope this helps someone else.

thanks John
On 2/3/2010 3:30 PM, John wrote:
> I can register SNOM phones fine, one the same network, I am trying to
> get a Linksys PAP2T-NA registered. Anyone have a configuration example?
> Thanks
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>    




From kristian.kielhofner at gmail.com  Wed Feb  3 15:55:48 2010
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Wed, 3 Feb 2010 18:55:48 -0500
Subject: [Freeswitch-users] adding sip header without X-
In-Reply-To: <4B69F9AD.8090904@xpirio.com>
References: <4B69F9AD.8090904@xpirio.com>
Message-ID: <2d9149cd1002031555o150b63a2hfd76dac20725b06a@mail.gmail.com>

Hello Christian,

  Reading through the code (sofia_glue.c) it looks like all you have
to do is set the alert_info channel variable.  I also don't think you
have to include the X- when using set with sip_h.  It's just always
good (RFC compliant) to prefix "custom" headers with X-.

  Regarding my suggestion for that configuration option/channel
variable.  That was only to be used on bridged channels to make it
configurable which headers FreeSWITCH passes from the a leg to the b
leg when using bridge.  A "proper" B2BUA (according to me) should
completely rewrite the outgoing INVITE, including stripping any custom
headers.

2010/2/3 Christian L?schenkohl <christian.loeschenkohl at xpirio.com>:
> hello
>
> do anybody know a way to add "Alert-Info: <http://www.notused.com>;info=alert-group;x-line-id=0"
> as a custom sip header to the invite message?
>
> <action application="set" data="sip_h_X-Alert-Info=<http://www.notused.com>\;info=alert-internal\;x-line-id=0"/>
> works but "X-Alert-Info" as a header isn't usefull at all
>
> the purpose is to distinguish internal and external calls in a pbx. it is also described here
> http://wiki.snom.com/Web_Interface/V7/Preferences/Alert-Info_Ringer
>
> there was also a posting on the list by Kristian Kielhofner who suggested a very flexible solution
>
> -----
> mail from: 2009-10-09 21:10
>
> 2) ?Make the behavior configurable with a channel variable and/or
> ? ? sofia config option: {sip_pass_headers=all|none|X-MyCustomHeaderByName}
> -----
>
> br
>
>
> --
> Ing. Christian L?schenkohl
> Technische Leitung, Forschung & Entwicklung VoIP
>
> xpirio
> Telekommunikation & Service GmbH
> Lakeside B04
> 9020 Klagenfurt
> Austria
>
> T ?+43 (0) 5 77 11 - 1000
> F ?+43 (0) 5 77 11 - 1002
> E ?christian.loeschenkohl at xpirio.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com



From christian.loeschenkohl at xpirio.com  Wed Feb  3 16:19:35 2010
From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=)
Date: Thu, 04 Feb 2010 01:19:35 +0100
Subject: [Freeswitch-users] adding sip header without X-
In-Reply-To: <2d9149cd1002031555o150b63a2hfd76dac20725b06a@mail.gmail.com>
References: <4B69F9AD.8090904@xpirio.com>
	<2d9149cd1002031555o150b63a2hfd76dac20725b06a@mail.gmail.com>
Message-ID: <4B6A1297.9090802@xpirio.com>

hello

thank you for your answer
setting only sip_h_Alert-Info was my first try, didn't work at all

i use trun rev. 16456 in proxy mode - if it helps
every header i add with X-... shows up in the invite to the sip endpoint (snom phone)

when i read sofia_glue.c i do see the function sofia_glue_set_extra_headers
which only adds header starting with X- or P- (line 1407 for me)

br

Kristian Kielhofner wrote:

> Hello Christian,
> 
>   Reading through the code (sofia_glue.c) it looks like all you have
> to do is set the alert_info channel variable.  I also don't think you
> have to include the X- when using set with sip_h.  It's just always
> good (RFC compliant) to prefix "custom" headers with X-.
> 
>   Regarding my suggestion for that configuration option/channel
> variable.  That was only to be used on bridged channels to make it
> configurable which headers FreeSWITCH passes from the a leg to the b
> leg when using bridge.  A "proper" B2BUA (according to me) should
> completely rewrite the outgoing INVITE, including stripping any custom
> headers.
> 
> 2010/2/3 Christian L?schenkohl <christian.loeschenkohl at xpirio.com>:
>> hello
>>
>> do anybody know a way to add "Alert-Info: <http://www.notused.com>;info=alert-group;x-line-id=0"
>> as a custom sip header to the invite message?
>>
>> <action application="set" data="sip_h_X-Alert-Info=<http://www.notused.com>\;info=alert-internal\;x-line-id=0"/>
>> works but "X-Alert-Info" as a header isn't usefull at all
>>
>> the purpose is to distinguish internal and external calls in a pbx. it is also described here
>> http://wiki.snom.com/Web_Interface/V7/Preferences/Alert-Info_Ringer
>>
>> there was also a posting on the list by Kristian Kielhofner who suggested a very flexible solution
>>
>> -----
>> mail from: 2009-10-09 21:10
>>
>> 2)  Make the behavior configurable with a channel variable and/or
>>     sofia config option: {sip_pass_headers=all|none|X-MyCustomHeaderByName}
>> -----
>>
>> br
>>
>>
>> --
>> Ing. Christian L?schenkohl
>> Technische Leitung, Forschung & Entwicklung VoIP
>>
>> xpirio
>> Telekommunikation & Service GmbH
>> Lakeside B04
>> 9020 Klagenfurt
>> Austria
>>
>> T  +43 (0) 5 77 11 - 1000
>> F  +43 (0) 5 77 11 - 1002
>> E  christian.loeschenkohl at xpirio.com
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
> 
> 
> 


-- 
Ing. Christian L?schenkohl
Technische Leitung, Forschung & Entwicklung VoIP

xpirio
Telekommunikation & Service GmbH
Lakeside B04
9020 Klagenfurt
Austria

T  +43 (0) 5 77 11 - 1000
F  +43 (0) 5 77 11 - 1002
E  christian.loeschenkohl at xpirio.com



From mrene_lists at avgs.ca  Wed Feb  3 16:28:21 2010
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Wed, 3 Feb 2010 19:28:21 -0500
Subject: [Freeswitch-users] adding sip header without X-
In-Reply-To: <4B6A1297.9090802@xpirio.com>
References: <4B69F9AD.8090904@xpirio.com>
	<2d9149cd1002031555o150b63a2hfd76dac20725b06a@mail.gmail.com>
	<4B6A1297.9090802@xpirio.com>
Message-ID: <CA18157A-7FBB-47CB-83CC-9B60D1E5E28B@avgs.ca>

If you look carefully, you'll see that it imports X- headers as  
variables on an incoming invite, but it'll send out sip_h_<anything>  
anyways.

For alert info, FS uses a channel variable called "alert_info", its  
implemented directly, without using sip_h_*.

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 3-Feb-10, at 7:19 PM, Christian L?schenkohl wrote:

> hello
>
> thank you for your answer
> setting only sip_h_Alert-Info was my first try, didn't work at all
>
> i use trun rev. 16456 in proxy mode - if it helps
> every header i add with X-... shows up in the invite to the sip  
> endpoint (snom phone)
>
> when i read sofia_glue.c i do see the function  
> sofia_glue_set_extra_headers
> which only adds header starting with X- or P- (line 1407 for me)
>
> br
>
> Kristian Kielhofner wrote:
>
>> Hello Christian,
>>
>>  Reading through the code (sofia_glue.c) it looks like all you have
>> to do is set the alert_info channel variable.  I also don't think you
>> have to include the X- when using set with sip_h.  It's just always
>> good (RFC compliant) to prefix "custom" headers with X-.
>>
>>  Regarding my suggestion for that configuration option/channel
>> variable.  That was only to be used on bridged channels to make it
>> configurable which headers FreeSWITCH passes from the a leg to the b
>> leg when using bridge.  A "proper" B2BUA (according to me) should
>> completely rewrite the outgoing INVITE, including stripping any  
>> custom
>> headers.
>>
>> 2010/2/3 Christian L?schenkohl <christian.loeschenkohl at xpirio.com>:
>>> hello
>>>
>>> do anybody know a way to add "Alert-Info: <http:// 
>>> www.notused.com>;info=alert-group;x-line-id=0"
>>> as a custom sip header to the invite message?
>>>
>>> <action application="set" data="sip_h_X-Alert-Info=<http://www.notused.com 
>>> >\;info=alert-internal\;x-line-id=0"/>
>>> works but "X-Alert-Info" as a header isn't usefull at all
>>>
>>> the purpose is to distinguish internal and external calls in a  
>>> pbx. it is also described here
>>> http://wiki.snom.com/Web_Interface/V7/Preferences/Alert-Info_Ringer
>>>
>>> there was also a posting on the list by Kristian Kielhofner who  
>>> suggested a very flexible solution
>>>
>>> -----
>>> mail from: 2009-10-09 21:10
>>>
>>> 2)  Make the behavior configurable with a channel variable and/or
>>>    sofia config option: {sip_pass_headers=all|none|X- 
>>> MyCustomHeaderByName}
>>> -----
>>>
>>> br
>>>
>>>
>>> --
>>> Ing. Christian L?schenkohl
>>> Technische Leitung, Forschung & Entwicklung VoIP
>>>
>>> xpirio
>>> Telekommunikation & Service GmbH
>>> Lakeside B04
>>> 9020 Klagenfurt
>>> Austria
>>>
>>> T  +43 (0) 5 77 11 - 1000
>>> F  +43 (0) 5 77 11 - 1002
>>> E  christian.loeschenkohl at xpirio.com
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>
>
> -- 
> Ing. Christian L?schenkohl
> Technische Leitung, Forschung & Entwicklung VoIP
>
> xpirio
> Telekommunikation & Service GmbH
> Lakeside B04
> 9020 Klagenfurt
> Austria
>
> T  +43 (0) 5 77 11 - 1000
> F  +43 (0) 5 77 11 - 1002
> E  christian.loeschenkohl at xpirio.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From kristian.kielhofner at gmail.com  Wed Feb  3 16:29:55 2010
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Wed, 3 Feb 2010 19:29:55 -0500
Subject: [Freeswitch-users] adding sip header without X-
In-Reply-To: <4B6A1297.9090802@xpirio.com>
References: <4B69F9AD.8090904@xpirio.com>
	<2d9149cd1002031555o150b63a2hfd76dac20725b06a@mail.gmail.com>
	<4B6A1297.9090802@xpirio.com>
Message-ID: <2d9149cd1002031629k6353b8b9v1874da556fb1c245@mail.gmail.com>

Have you tried setting channel variable alert_info?

2010/2/3 Christian L?schenkohl <christian.loeschenkohl at xpirio.com>:
> hello
>
> thank you for your answer
> setting only sip_h_Alert-Info was my first try, didn't work at all
>
> i use trun rev. 16456 in proxy mode - if it helps
> every header i add with X-... shows up in the invite to the sip endpoint (snom phone)
>
> when i read sofia_glue.c i do see the function sofia_glue_set_extra_headers
> which only adds header starting with X- or P- (line 1407 for me)
>
> br
>

-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com



From msc at freeswitch.org  Wed Feb  3 16:54:54 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 3 Feb 2010 16:54:54 -0800
Subject: [Freeswitch-users] PAP2T issue
In-Reply-To: <4B6A0559.2040200@acsol.net>
References: <4B69F90D.1070503@acsol.net> <4B6A0559.2040200@acsol.net>
Message-ID: <87f2f3b91002031654o1838868bk49d75df6866bf351@mail.gmail.com>

Would you mind creating a small wiki page and linking to it here?
http://wiki.freeswitch.org/wiki/Interop_List#Linksys_PAP2T

It would be nice to have a known working config on the wiki. If you have any
issues let me know.

Thanks,
MC

On Wed, Feb 3, 2010 at 3:23 PM, John <john at acsol.net> wrote:

> Found two issues and solved the problem. By default, the PAP2T has
> provisioning turned on, and won't try to register if the provisioning
> server isn't found. Turned that off. Secondly, turned off Stun and
> enabled rport and it registered fine. Hope this helps someone else.
>
> thanks John
> On 2/3/2010 3:30 PM, John wrote:
> > I can register SNOM phones fine, one the same network, I am trying to
> > get a Linksys PAP2T-NA registered. Anyone have a configuration example?
> > Thanks
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100203/7e5e6f5c/attachment-0002.html 

From christian.loeschenkohl at xpirio.com  Wed Feb  3 16:59:59 2010
From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=)
Date: Thu, 04 Feb 2010 01:59:59 +0100
Subject: [Freeswitch-users] adding sip header without X-
In-Reply-To: <2d9149cd1002031629k6353b8b9v1874da556fb1c245@mail.gmail.com>
References: <4B69F9AD.8090904@xpirio.com>	<2d9149cd1002031555o150b63a2hfd76dac20725b06a@mail.gmail.com>	<4B6A1297.9090802@xpirio.com>
	<2d9149cd1002031629k6353b8b9v1874da556fb1c245@mail.gmail.com>
Message-ID: <4B6A1C0F.9040806@xpirio.com>

now i did and i works

<action application="export" data="alert_info=http://www.notused.com;info=alert-internal;x-line-id=0"/>
did the job

thank you two
br

Kristian Kielhofner wrote:

> Have you tried setting channel variable alert_info?
> 
> 2010/2/3 Christian L?schenkohl <christian.loeschenkohl at xpirio.com>:
>> hello
>>
>> thank you for your answer
>> setting only sip_h_Alert-Info was my first try, didn't work at all
>>
>> i use trun rev. 16456 in proxy mode - if it helps
>> every header i add with X-... shows up in the invite to the sip endpoint (snom phone)
>>
>> when i read sofia_glue.c i do see the function sofia_glue_set_extra_headers
>> which only adds header starting with X- or P- (line 1407 for me)
>>
>> br
>>
> 


-- 
Ing. Christian L?schenkohl
Technische Leitung, Forschung & Entwicklung VoIP

xpirio
Telekommunikation & Service GmbH
Lakeside B04
9020 Klagenfurt
Austria

T  +43 (0) 5 77 11 - 1000
F  +43 (0) 5 77 11 - 1002
E  christian.loeschenkohl at xpirio.com



From scott.torr.fs at letterboxes.org  Wed Feb  3 23:16:10 2010
From: scott.torr.fs at letterboxes.org (Scott Torr)
Date: Thu, 04 Feb 2010 18:16:10 +1100
Subject: [Freeswitch-users] skypiax dtmf detection issue on Freeswitch
In-Reply-To: <A90E1E992A65E1428D12320969B90F2103042A99@USDALSMBS05.ad3.ad.alcatel.com>
References: <A90E1E992A65E1428D12320969B90F2103042A99@USDALSMBS05.ad3.ad.alcatel.com>
Message-ID: <1265267770.21239.1358206487@webmail.messagingengine.com>

Hi Majdi,

It is good to know that once the audio stream is in the right 'format',
Freeswitch is able to decode the DTMF's to digits. (as best it can given
the potential jitter and lossy quality of the signal)

I'm sure someone on the list has the answer and it will quite simple.

So here we go:


How do we convert the audio stream from mod_skypiax into to a suitable
format that <action application="start_dtmf" /> can attempt to decode
the 'Inband' DTMF tones into digits?

At present it seems that the decoder will not work on a 16Kbps stream.


Perhaps this has been explained and not understood?

Ideally Skype would decode the DTMF tones ingress to the Skype network
at the PSTN-Skype gateway and pass on as out of band signaling. Either
through bad design or intentionally they do not. It seems though at some
locations (New Zealand) they did, then did not?

Perhaps on their paid for (approx $9 AUD per month) SIP trunks they do?
Or is this just as hit and miss as well?
Does any body have any experience with those?


-Scott

On Wed, 03 Feb 2010 20:42 -0600, "BSOUL Majdi"
<Majdi.Bsoul at alcatel-lucent.com> wrote:
> I just saw your JIRA, so definitely by your comment date so close you
> might still blocked on that resolution.
> 
>  
> 
> I used SiptoSis with a single channel and because it passes the stream
> to Freeswitch as PCMU, it allows FS to detect correctly. And this proves
> that the inband DTMF quality coming from Skype is good enough.
> 
>  
> 
> I am trying to see if I can use stsTrunkBuilder to support multiple
> interfaces, but looks that it requires different skype account for each
> instance, rather than allowing me to use single instance for all
> instances as skypiax does.
> 
>  
> 
> Regards,
> 
>  
> 
> Majdi Bsoul
> 
> Mobile NGN R&D
> 
> Alcatel-Lucent
> 
> 3400 W. Plano Parkway
> 
> M/S 601-NGNRD
> 
> Plano, TX 75075  
> 
> Office: +1 972 477 0065
> 
> Fax: +1 972 519 3600
> Email: majdi.bsoul at alcatel-lucent.com
> 
> From: BSOUL Majdi 
> Sent: Wednesday, February 03, 2010 8:26 PM
> To: 'scott.torr.fs at letterboxes.org'
> Cc: 'mbsoul at hotmail.com'
> Subject: skypiax dtmf detection issue on Freeswitch
> 
>  
> 
> Hello Scott,
> 
>  
> 
> I thought I catch up with you and see if you had found a solution for
> the skypiax dtmf detection in FS issue that you reported in Dec?
> 
> I am running with the same issue.
> 
>  
> 
> Thanks,
> 
>  
> 
> Majdi Bsoul
> 
> Mobile NGN R&D
> 
> Alcatel-Lucent
> 
> 3400 W. Plano Parkway
> 
> M/S 601-NGNRD
> 
> Plano, TX 75075  
> 
> Office: +1 972 477 0065
> 
> Fax: +1 972 519 3600
> Email: majdi.bsoul at alcatel-lucent.com
> 



From nagalenoj at gmail.com  Thu Feb  4 01:32:58 2010
From: nagalenoj at gmail.com (Nagalenoj H.)
Date: Thu, 4 Feb 2010 15:02:58 +0530
Subject: [Freeswitch-users] A leg disconnects after uuid_bridge?
In-Reply-To: <87f2f3b91002030605j7a1f9a02g665ba043c6ed7f7b@mail.gmail.com>
References: <fe67a3341002021934m548cde89p4d48381c5abe02a0@mail.gmail.com>
	<191c3a031002022108ta22287ev40fc36420abba2e5@mail.gmail.com>
	<fe67a3341002030112yeea379u90c22c482614c57@mail.gmail.com>
	<87f2f3b91002030605j7a1f9a02g665ba043c6ed7f7b@mail.gmail.com>
Message-ID: <fe67a3341002040132n3aa2a897m60c5b9f1ca6b347f@mail.gmail.com>

On Wed, Feb 3, 2010 at 7:35 PM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Wed, Feb 3, 2010 at 1:12 AM, Nagalenoj H. <nagalenoj at gmail.com> wrote:
>
>> I've used intercept application instead of uuid_bridge. I got the call
>> bridged with the given uuid. But, similar to uuid_bridge, A leg is getting
>> disconnected when B leg terminates.
>> I've did the same steps posted above, but used intercept instead of
>> uuid_bridge. Am I right?!
>>
>> Tried using the hangup_after_bridge=false, but didn't see any difference.
>>
>> Are you on the latest SVN of FreeSWITCH? Be sure to "make current" and try
> again. If the issue remains then capture a complete debug log of the call
> from start to finish and also pastebin your script so others can test and
> analyze.
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
I've updated to latest SVN. Now, the version is 'FreeSWITCH Version
1.0.trunk (16565)'.
I tested intercept in this version and the problem persists here.

Freeswitch debug log:
     http://pastebin.freeswitch.org/12043

Script:
     Not done any scripts, tested through nc command. Used the following
commands,

sendmsg
call-command: execute
execute-app-name: answer

api originate
{origination_uuid=c6d2e0e2-0b2f-11df-9e84-fb15c3cd8565}user/1001 &park

sendmsg
call-command: execute
execute-app-name:intercept
execute-app-arg: c6d2e0e2-0b2f-11df-9e84-fb15c3cd8565

Steps:
* Make a call to the socket extension.
* Answer the call.
* Originate a call and park it.
* Intercept the originated call's uuid.
* Disconnect the 'B' leg.
* You will notice the 'A' leg is also getting disconnected.
-- 
Regards,
Nagalenoj H.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100204/b46049fb/attachment-0002.html 

From nagalenoj at gmail.com  Thu Feb  4 01:47:44 2010
From: nagalenoj at gmail.com (Nagalenoj H.)
Date: Thu, 4 Feb 2010 15:17:44 +0530
Subject: [Freeswitch-users] uuid_bridge isn't working
Message-ID: <fe67a3341002040147i5892d5b8md65e327be6dea7c5@mail.gmail.com>

Dear friends,
    After upgrading to 'FreeSWITCH Version 1.0.trunk (16565)', uuid_bridge
isn't working. When I give uuid_bridge, both the legs are not bridged, and
they got disconnected.

Did the following,
* Made a call to socket extension.
* Answered the call.
* Originated a call and parked it.
* Did uuid_bridge with the uuids of the originated call and caller's uuid.

* Didn't get the legs bridged, instead both got disconnected,

Freeswitch debug log:
      http://pastebin.freeswitch.org/12044

Facing this problem only after upgrading to this trunk version.

-- 
Regards,
Nagalenoj H.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100204/cc2ad2f9/attachment-0002.html 

From nagalenoj at gmail.com  Thu Feb  4 02:01:57 2010
From: nagalenoj at gmail.com (Nagalenoj H.)
Date: Thu, 4 Feb 2010 15:31:57 +0530
Subject: [Freeswitch-users] Event socket: filter delete isn't working
In-Reply-To: <191c3a031001271702l3e77c952lc8aba35bacbf3d58@mail.gmail.com>
References: <fe67a3341001270514g2715de2ag6aa2e3b099fa7816@mail.gmail.com>
	<191c3a031001271702l3e77c952lc8aba35bacbf3d58@mail.gmail.com>
Message-ID: <fe67a3341002040201i1b02a8bekdad0e649fdb323bf@mail.gmail.com>

I've now upgraded to 16565 trunk.  I've tested it and now it is working
fine.

But, when I give only 'filter delete' without the next parameters,
freeswitch is getting core dumped.

Freeswitch debug log:
   http://pastebin.freeswitch.org/12045

On Thu, Jan 28, 2010 at 6:32 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> in the future please report issues to jira http://jira.freeswitch.org
>
> please try svn trunk 16527 or higher
>
> This was not a bug but I made it work the way you describe since it made
> sense.
>
> you should have done
>
> filter delete unique-id
>
> which would have delete all the unique-id filters that was the only option
>
>
> you should be able to now say
>
> filter delete unique-id <value>
>
> To delete entry with specific value
>
> or
>
> filter delete unique-id
>
> to delete all entries with matching key
>
>
>
>
>
> On Wed, Jan 27, 2010 at 7:14 AM, Nagalenoj H. <nagalenoj at gmail.com> wrote:
>
>> Dear friends,
>>    I've tried to delete the filter which I applied for an unique id. But,
>> it doesn't work. After executing 'filter delete', I am receiving the events
>> from that uuid.
>>    I used the command as 'filter delete unique-id
>> c6d2e0e2-0b2f-11df-9e84-fb15c3cd8565'.
>>
>> I did the following operations.
>>     Made call to the event socket.
>>     Registered events for all. (events plain all).
>>     Applied filter for the uuid. (filter unique-id
>> aa3cb8ea-0b2f-11df-9e84-fb15c3cd8565).
>>     I've got a new uuid by using create_uuid.
>>     Applied filter for this new uuid. (filter unique-id
>> c6d2e0e2-0b2f-11df-9e84-fb15c3cd8565)
>>     Originated a call with that uuid.
>>     Now, I could receive events from both uuids. (Tested by giving DTMFs
>> in both end and checked unique-id in event header).
>>     Then, I wanted to delete a uuid from the filter. (filter delete
>> unique-id c6d2e0e2-0b2f-11df-9e84-fb15c3cd8565).
>>     I thought, i won't receive the events from this deleted unique-id.
>> But, I received the dtmfs from both unique-id.
>>
>> I'm using 'FreeSWITCH Version 1.0.trunk (15982)'.
>> --
>> Regards,
>> Nagalenoj H.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Regards,
Nagalenoj H.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100204/840d3f5d/attachment-0002.html 

From xanlich at gmail.com  Thu Feb  4 03:33:54 2010
From: xanlich at gmail.com (Chia-Yen Wu)
Date: Thu, 4 Feb 2010 19:33:54 +0800
Subject: [Freeswitch-users] about bgapi
Message-ID: <314dc3f81002040333n28c15d92l5dd0aeef4dc5aa92@mail.gmail.com>

I tried to run a javascript by background API (bgapi jsrun test.js)

this javascript (test.js) wont automatically stop, I tried to kill it by
"uuid_kill" command with Job-UUID

but return "-ERR no such channel!", how can I kill this bgapi? btw my FS run
in windows.

thank you
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100204/3b5cb1b6/attachment-0002.html 

From nepaligas at yahoo.com  Tue Feb  2 23:27:18 2010
From: nepaligas at yahoo.com (Prabin Shrestha)
Date: Tue, 2 Feb 2010 23:27:18 -0800 (PST)
Subject: [Freeswitch-users] need some hints on Softswitch deployment of
	FreeSwitch
Message-ID: <858472.38298.qm@web65402.mail.ac4.yahoo.com>

Dear all,

I had been browsing through all the wikis of freeswitch, googling more than 1 week and couldn't figure out where to start. 
I have been finding so many problems and IRC thing I don't understand. Basically, I am just a average linux user running Ubuntu, trying to build a softswitch. It there was some book on freeswitch it would have been much easier for newbie like me. 
Here are some problems I have been facing. 
After installation, I found freeswitch in /opt/freeswitch directory.
only creating freeswitch user, I can access it's fs_cli, and I have yet to learn the power of it. 
doing ps -A, I found freeswitch is running in background.

Now comes the hard part.
I wanted to test it using SPA3000 device with fxo and fxs ports, which after following guides in net,  is not working for me.

My reqirement is, 
1. to run freeswitch as a softswitch which can route calls from voip call providers to Quintum gateways.
2. to have complete CDR reports generated to sql database.

Some light on this matter will be highly appreciated.

Prabin.



      Your Mail works best with the New Yahoo Optimized IE8. Get it NOW! http://downloads.yahoo.com/in/internetexplorer/
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100202/fcbda4d8/attachment-0002.html 
-------------- next part --------------
A non-text attachment was scrubbed...
Name: not available
Type: image/gif
Size: 21362 bytes
Desc: not available
Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100202/fcbda4d8/attachment-0002.gif 

From nepaligas at yahoo.com  Wed Feb  3 22:00:33 2010
From: nepaligas at yahoo.com (Prabin Shrestha)
Date: Wed, 3 Feb 2010 22:00:33 -0800 (PST)
Subject: [Freeswitch-users] Fw: need some hints on Softswitch deployment of
	FreeSwitch
Message-ID: <609766.80114.qm@web65415.mail.ac4.yahoo.com>



--- On Tue, 2/2/10, Prabin Shrestha <nepaligas at yahoo.com> wrote:

From: Prabin Shrestha <nepaligas at yahoo.com>
Subject: need some hints on Softswitch deployment of FreeSwitch
To: freeswitch-users at lists.freeswitch.org
Date: Tuesday, 2 February, 2010, 11:27 PM

 
 
Dear all,

I had been browsing through all the wikis of freeswitch, googling more than 1 week and couldn't
 figure out where to start. 
I have been finding so many problems and IRC thing I don't understand. Basically, I am just a average linux user running Ubuntu, trying to build a softswitch. It there was some book on freeswitch it would have been much easier for newbie like me. 
Here are some problems I have been facing. 
After installation, I found freeswitch in /opt/freeswitch directory.
only creating freeswitch user, I can access it's fs_cli, and I have yet to learn the power of it. 
doing ps -A, I found freeswitch is running in background.

Now comes the hard part.
I wanted to test it using SPA3000 device with fxo and fxs ports, which after following guides in net,? is not working for me.

My reqirement is, 
1. to run freeswitch as a softswitch which can route calls from voip call providers to Quintum gateways.
2. to have complete CDR reports generated to sql database.

Some light on this matter will be
 highly appreciated.

Prabin.


 
 
 


       
Your Mail works best with the New Yahoo Optimized IE8. Get it NOW!.


      The INTERNET now has a personality. YOURS! See your Yahoo! Homepage. http://in.yahoo.com/
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100203/41a45589/attachment-0002.html 
-------------- next part --------------
A non-text attachment was scrubbed...
Name: not available
Type: image/gif
Size: 21362 bytes
Desc: not available
Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100203/41a45589/attachment-0002.gif 

From brian at freeswitch.org  Thu Feb  4 04:12:25 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 4 Feb 2010 06:12:25 -0600
Subject: [Freeswitch-users] A leg disconnects after uuid_bridge?
In-Reply-To: <fe67a3341002040132n3aa2a897m60c5b9f1ca6b347f@mail.gmail.com>
References: <fe67a3341002021934m548cde89p4d48381c5abe02a0@mail.gmail.com>
	<191c3a031002022108ta22287ev40fc36420abba2e5@mail.gmail.com>
	<fe67a3341002030112yeea379u90c22c482614c57@mail.gmail.com>
	<87f2f3b91002030605j7a1f9a02g665ba043c6ed7f7b@mail.gmail.com>
	<fe67a3341002040132n3aa2a897m60c5b9f1ca6b347f@mail.gmail.com>
Message-ID: <0A571E14-BB58-4A3D-996A-9D5205668E85@freeswitch.org>

Where are you getting this UUID?

/b

On Feb 4, 2010, at 3:32 AM, Nagalenoj H. wrote:

> api originate {origination_uuid=c6d2e0e2-0b2f-11df-9e84-fb15c3cd8565}user/1001 &park




From ustcorporation at yahoo.com  Wed Feb  3 18:12:09 2010
From: ustcorporation at yahoo.com (Darren C.)
Date: Wed, 3 Feb 2010 18:12:09 -0800 (PST)
Subject: [Freeswitch-users] Seeking Advice on SIP Phones like Aastra 6739i
	or Snom 870 that have good interoperability with FreeSWITCH
Message-ID: <894614.36103.qm@web33003.mail.mud.yahoo.com>

Hello,

I'm working on a project and are interested in using one of the newer SIP Phones with color displays, perhaps touchscreen, etc.?to implement PBX-like features like voice mail messages waiting, visual?address book,?conference call setup, and more.? 
?
We want to send some information about a call to the SIP Phone either via FS or our own Web Service.? These two phones have XML browsers that we may be able to utilize.? I'm concerned that these phones may work OK with Asterisk but not sure about FreeSWITCH.? There is a great guide for?XML Services for?Aastra 6739i but only mention Asterisk in the examples:
?
http://www.aastratelecom.com/cps/rde/xbcr/SID-3D8CCB6A-93A834D7/03/XML_Free_Services_-_PA-001005-00-05.pdf
?
I prefer to use phones that work well with FS but these are too new to show up on interoperability page.? Anyone have experience using these, any advise would be appreciated:


Aastra 6739i
http://www.aastratelecom.com/cps/rde/xbcr/SID-3D8CCB6A-EAA51F83/03/6739i_pds_en_1209.pdf
http://www.voipsupply.com/aastra-6739i?
?
SNOM 870
http://www.snom.com/uploads/docu/data_snom870_en.pdf
http://www.snom.com/products/ip-phones/snom-870-touchscreen-voip-phone
http://www.voipsupply.com/snom-870

Hoping someone has some positive or negative experiences working with these phones.? 
?
Thanks,

teldev



      
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100203/72c60834/attachment-0002.html 

From nagalenoj at gmail.com  Thu Feb  4 04:52:35 2010
From: nagalenoj at gmail.com (Nagalenoj H.)
Date: Thu, 4 Feb 2010 18:22:35 +0530
Subject: [Freeswitch-users] A leg disconnects after uuid_bridge?
In-Reply-To: <0A571E14-BB58-4A3D-996A-9D5205668E85@freeswitch.org>
References: <fe67a3341002021934m548cde89p4d48381c5abe02a0@mail.gmail.com>
	<191c3a031002022108ta22287ev40fc36420abba2e5@mail.gmail.com>
	<fe67a3341002030112yeea379u90c22c482614c57@mail.gmail.com>
	<87f2f3b91002030605j7a1f9a02g665ba043c6ed7f7b@mail.gmail.com>
	<fe67a3341002040132n3aa2a897m60c5b9f1ca6b347f@mail.gmail.com>
	<0A571E14-BB58-4A3D-996A-9D5205668E85@freeswitch.org>
Message-ID: <fe67a3341002040452v5468fd1fh96d1c5062091474b@mail.gmail.com>

By using create_uuid. I've also tried without giving origination_uuid. But,
the result is same.

-- 
Regards,
Nagalenoj H.

On Thu, Feb 4, 2010 at 5:42 PM, Brian West <brian at freeswitch.org> wrote:

> Where are you getting this UUID?
>
> /b
>
> On Feb 4, 2010, at 3:32 AM, Nagalenoj H. wrote:
>
> > api originate
> {origination_uuid=c6d2e0e2-0b2f-11df-9e84-fb15c3cd8565}user/1001 &park
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100204/70acd9b7/attachment-0002.html 

From christian.loeschenkohl at xpirio.com  Thu Feb  4 04:59:06 2010
From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=)
Date: Thu, 04 Feb 2010 13:59:06 +0100
Subject: [Freeswitch-users] presence with event socket
Message-ID: <4B6AC49A.8030409@xpirio.com>

hello

i have a little problem with setting the presence information manually (on and off)

with an outbound socket script i do send a event like this

sendevent PRESENCE_IN
proto: sip
from: xxxx at mydomain.com
login: xxxx at mydomain.com
event_type: presence
alt_event_type: dialog
Presence-Call-Direction: outbound
answer-state: confirmed

i monitor the estension on a snom phone with bfl (function key mode)

the problem is now that the led on the monitoring snom gets lit very short an the is off again
if i put a sleep in the outbound socket script the led lights up as long as i set the timeout.

the expected behaviour is that the led lights up and can be switched off with another
event (then answer-state: terminated)

we do use trunk rev. 16456 in proxy mode
br

-- 
Ing. Christian L?schenkohl
Technische Leitung, Forschung & Entwicklung VoIP

xpirio
Telekommunikation & Service GmbH
Lakeside B04
9020 Klagenfurt
Austria

T  +43 (0) 5 77 11 - 1000
F  +43 (0) 5 77 11 - 1002
E  christian.loeschenkohl at xpirio.com



From rupa at rupa.com  Thu Feb  4 05:19:21 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Thu, 4 Feb 2010 07:19:21 -0600
Subject: [Freeswitch-users] about bgapi
In-Reply-To: <314dc3f81002040333n28c15d92l5dd0aeef4dc5aa92@mail.gmail.com>
References: <314dc3f81002040333n28c15d92l5dd0aeef4dc5aa92@mail.gmail.com>
Message-ID: <b39273531002040519g71fa8a37uc9f0819c0a19772f@mail.gmail.com>

you can't.  If you want to terminate it, then you should set a global var
that the script periodically checks and if set the script should terminate
itself.

On Thu, Feb 4, 2010 at 5:33 AM, Chia-Yen Wu <xanlich at gmail.com> wrote:

> I tried to run a javascript by background API (bgapi jsrun test.js)
>
> this javascript (test.js) wont automatically stop, I tried to kill it by
> "uuid_kill" command with Job-UUID
>
> but return "-ERR no such channel!", how can I kill this bgapi? btw my FS
> run in windows.
>
> thank you
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100204/9c988b86/attachment-0002.html 

From peder at networkoblivion.com  Thu Feb  4 05:57:20 2010
From: peder at networkoblivion.com (Peder)
Date: Thu, 4 Feb 2010 07:57:20 -0600
Subject: [Freeswitch-users] need some hints on Softswitch deployment
	of	FreeSwitch
In-Reply-To: <858472.38298.qm@web65402.mail.ac4.yahoo.com>
References: <858472.38298.qm@web65402.mail.ac4.yahoo.com>
Message-ID: <039401caa5a1$f8256b40$e87041c0$@com>

You will need to ask more specific questions, such as ?when I call from 1000 to 1001, I get xxx on the console and it doesn?t work?.  Just saying ?I need help? is too big of a topic for anyone to respond.  Have you tried the wiki?

 

http://wiki.freeswitch.org/wiki/Main_Page

 

 

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Prabin Shrestha
Sent: Wednesday, February 03, 2010 1:27 AM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] need some hints on Softswitch deployment of FreeSwitch

 


	
	


Dear all,

I had been browsing through all the wikis of freeswitch, googling more than 1 week and couldn't figure out where to start. 
I have been finding so many problems and IRC thing I don't understand. Basically, I am just a average linux user running Ubuntu, trying to build a softswitch. It there was some book on freeswitch it would have been much easier for newbie like me. 
Here are some problems I have been facing. 
After installation, I found freeswitch in /opt/freeswitch directory.
only creating freeswitch user, I can access it's fs_cli, and I have yet to learn the power of it. 
doing ps -A, I found freeswitch is running in background.

Now comes the hard part.
I wanted to test it using SPA3000 device with fxo and fxs ports, which after following guides in net,  is not working for me.

My reqirement is, 
1. to run freeswitch as a softswitch which can route calls from voip call providers to Quintum gateways.
2. to have complete CDR reports generated to sql database.

Some light on this matter will be highly appreciated.

Prabin.

	
	
	





  _____  

Your Mail works best with the New Yahoo Optimized IE8. Get it NOW! <http://in.rd.yahoo.com/tagline_ie8_new/*http:/downloads.yahoo.com/in/internetexplorer/> .

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100204/2d53841c/attachment-0002.html 

From maciej.aniserowicz at gmail.com  Thu Feb  4 06:53:10 2010
From: maciej.aniserowicz at gmail.com (Maciej Aniserowicz)
Date: Thu, 4 Feb 2010 06:53:10 -0800 (PST)
Subject: [Freeswitch-users] Trunk compilation error on Windows "Cannot open
 source file: '..\..\celt-0.7.0-1\libcelt\kfft_single.c'"
Message-ID: <1265295190863-4513865.post@n2.nabble.com>


Hi,
I downloaded the latest bits from svn but have troubles compiling FS. I get
21 similar errors:
Cannot open source file: '..\..\celt-0.7.0-1\libcelt\kfft_single.c'

Error 40 fatal error C1083: Cannot open source file:
'..\..\celt-0.7.0-1\libcelt\bands.c': No such file or directory
File: c1
Project: libcelt

I just opened Freeswitch.2008.sln in VS, changed configuration to Release
and hit Build. What am I missing?

Regards,
Maciej Aniserowicz
-- 
View this message in context: http://n2.nabble.com/Trunk-compilation-error-on-Windows-Cannot-open-source-file-celt-0-7-0-1-libcelt-kfft-single-c-tp4513865p4513865.html
Sent from the freeswitch-users mailing list archive at Nabble.com.



From brian at freeswitch.org  Thu Feb  4 07:02:31 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 4 Feb 2010 09:02:31 -0600
Subject: [Freeswitch-users] Trunk compilation error on Windows "Cannot
	open source file: '..\..\celt-0.7.0-1\libcelt\kfft_single.c'"
In-Reply-To: <1265295190863-4513865.post@n2.nabble.com>
References: <1265295190863-4513865.post@n2.nabble.com>
Message-ID: <FC7358A4-5486-4E61-BABE-8FDD5D8D52BF@freeswitch.org>

Please update.  That was already fixed... remember ALWAYS update.. check and then email... we move fast around these parts!  :P

/b

On Feb 4, 2010, at 8:53 AM, Maciej Aniserowicz wrote:

> 
> Hi,
> I downloaded the latest bits from svn but have troubles compiling FS. I get
> 21 similar errors:
> Cannot open source file: '..\..\celt-0.7.0-1\libcelt\kfft_single.c'
> 
> Error 40 fatal error C1083: Cannot open source file:
> '..\..\celt-0.7.0-1\libcelt\bands.c': No such file or directory
> File: c1
> Project: libcelt
> 
> I just opened Freeswitch.2008.sln in VS, changed configuration to Release
> and hit Build. What am I missing?
> 
> Regards,
> Maciej Aniserowicz
> -- 
> View this message in context: http://n2.nabble.com/Trunk-compilation-error-on-Windows-Cannot-open-source-file-celt-0-7-0-1-libcelt-kfft-single-c-tp4513865p4513865.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From tim at communicatefreely.net  Thu Feb  4 07:05:37 2010
From: tim at communicatefreely.net (Tim St. Pierre)
Date: Thu, 04 Feb 2010 10:05:37 -0500
Subject: [Freeswitch-users] Seeking Advice on SIP Phones like Aastra
 6739i or Snom 870 that have good interoperability with FreeSWITCH
In-Reply-To: <894614.36103.qm@web33003.mail.mud.yahoo.com>
References: <894614.36103.qm@web33003.mail.mud.yahoo.com>
Message-ID: <4B6AE241.4040106@communicatefreely.net>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hi Darren,

While I can't vouch for the 6739i yet, I have been doing all of my FS development with Aastra
phones.  I had a 57i connected up, and it worked just fine.  I could use the wideband codecs, BLFs
all worked correctly, I have intercom and distinctive ring working properly too.

As far as the XML features are concerned, these are all done outside of Freeswitch.  I have built
some simple applications that use the XML browser for or own network, and it was pretty straight
forward.  The phone interacts with our web server, that in turn manipulates the database used for
call handling and other parameters.  In some cases, the php script on the web server fires events to
the switch to initiate calls, or do some other call handling.

Right now, our production system is Asterisk, but I'm trying to migrate everything to freeswitch.
As long as everything is in a database, it's pretty easy to make some great features.  In many ways,
I have found freeswitch easier to integrate with.  You may want to enable core odbc, it opens up a
lot, since you can do a DB query to find out what voicemail messages a user has, or what calls are
active, etc.  I find it a lot easier to interact with than something like an event socket, and it
clusters better.

I ended up not using any of the Aastra supplied scripts, since they are very specific to a single
office Asterisk setup.  You will have to build your own for freeswitch, but they are easy to write.
 The Aastra php classes are still useful functions, just not their pre-built voicemail and other
Asterisk feature tools.

Good luck!

- -Tim

Darren C. wrote:
> Hello,
> 
> I'm working on a project and are interested in using one of the newer
> SIP Phones with color displays, perhaps touchscreen, etc. to implement
> PBX-like features like voice mail messages waiting, visual address
> book, conference call setup, and more. 
> 
>  
> 
> We want to send some information about a call to the SIP Phone either
> via FS or our own Web Service.  These two phones have XML browsers that
> we may be able to utilize.  I'm concerned that these phones may work OK
> with Asterisk but not sure about FreeSWITCH.  There is a great guide
> for XML Services for Aastra 6739i but only mention Asterisk in the examples:
> 
>  
> 
> http://www.aastratelecom.com/cps/rde/xbcr/SID-3D8CCB6A-93A834D7/03/XML_Free_Services_-_PA-001005-00-05.pdf
> 
>  
> 
> I prefer to use phones that work well with FS but these are too new to
> show up on interoperability page.  Anyone have experience using these,
> any advise would be appreciated:
> 
> *Aastra 6739i*
> 
> http://www.aastratelecom.com/cps/rde/xbcr/SID-3D8CCB6A-EAA51F83/03/6739i_pds_en_1209.pdf
> 
> http://www.voipsupply.com/aastra-6739i* *
> 
> * *
> 
> *SNOM 870*
> 
> http://www.snom.com/uploads/docu/data_snom870_en.pdf
> 
> http://www.snom.com/products/ip-phones/snom-870-touchscreen-voip-phone
> 
> http://www.voipsupply.com/snom-870
> 
> 
> Hoping someone has some positive or negative experiences working with
> these phones. 
> 
>  
> 
> Thanks,
> 
> teldev
> 
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

- --
Tim St. Pierre
IP Voice technician
Communicate Freely
1-877-291-8647 x5101
sip:5101 at communicatefreely.net
tim at communicatefreely.net
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v2.0.4 (FreeBSD)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org

iQCVAwUBS2riQYqVcvNCnHOrAQKQLgQAk9thDiDNp89QhI32cv4G3/RmPET2nFns
WHRovrKxo/gGLJV0eC7XAB9vysQqa/A7VGXQBjvrNZ/EylsRoKcGDHr95Tndz6jI
w82rqWUdiCl58ABae2jYW03sRUG47y9rwp6ujERO1XQ8iyUxLPdTju5pve0gmdyI
qO0PnOkWkhk=
=e6DD
-----END PGP SIGNATURE-----



From anthony.minessale at gmail.com  Thu Feb  4 07:46:25 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 4 Feb 2010 09:46:25 -0600
Subject: [Freeswitch-users] presence with event socket
In-Reply-To: <4B6AC49A.8030409@xpirio.com>
References: <4B6AC49A.8030409@xpirio.com>
Message-ID: <191c3a031002040746i494d5fe4s163f512afd34da79@mail.gmail.com>

the presence stuff is automatic so when the phone re-registers it probably
turns it back off.
snom does not do the nonce count thing right so it re-challenges every time
and causes a presence event.


2010/2/4 Christian L?schenkohl <christian.loeschenkohl at xpirio.com>

> hello
>
> i have a little problem with setting the presence information manually (on
> and off)
>
> with an outbound socket script i do send a event like this
>
> sendevent PRESENCE_IN
> proto: sip
> from: xxxx at mydomain.com
> login: xxxx at mydomain.com
> event_type: presence
> alt_event_type: dialog
> Presence-Call-Direction: outbound
> answer-state: confirmed
>
> i monitor the estension on a snom phone with bfl (function key mode)
>
> the problem is now that the led on the monitoring snom gets lit very short
> an the is off again
> if i put a sleep in the outbound socket script the led lights up as long as
> i set the timeout.
>
> the expected behaviour is that the led lights up and can be switched off
> with another
> event (then answer-state: terminated)
>
> we do use trunk rev. 16456 in proxy mode
> br
>
> --
> Ing. Christian L?schenkohl
> Technische Leitung, Forschung & Entwicklung VoIP
>
> xpirio
> Telekommunikation & Service GmbH
> Lakeside B04
> 9020 Klagenfurt
> Austria
>
> T  +43 (0) 5 77 11 - 1000
> F  +43 (0) 5 77 11 - 1002
> E  christian.loeschenkohl at xpirio.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100204/5956864d/attachment-0002.html 

From anthony.minessale at gmail.com  Thu Feb  4 07:50:27 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 4 Feb 2010 09:50:27 -0600
Subject: [Freeswitch-users] Event socket: filter delete isn't working
In-Reply-To: <fe67a3341002040201i1b02a8bekdad0e649fdb323bf@mail.gmail.com>
References: <fe67a3341001270514g2715de2ag6aa2e3b099fa7816@mail.gmail.com>
	<191c3a031001271702l3e77c952lc8aba35bacbf3d58@mail.gmail.com>
	<fe67a3341002040201i1b02a8bekdad0e649fdb323bf@mail.gmail.com>
Message-ID: <191c3a031002040750x5c94f701p2f5dd7a6a8598cd4@mail.gmail.com>

not anymore, fixed r16568

On Thu, Feb 4, 2010 at 4:01 AM, Nagalenoj H. <nagalenoj at gmail.com> wrote:

> I've now upgraded to 16565 trunk.  I've tested it and now it is working
> fine.
>
> But, when I give only 'filter delete' without the next parameters,
> freeswitch is getting core dumped.
>
> Freeswitch debug log:
>    http://pastebin.freeswitch.org/12045
>
>
> On Thu, Jan 28, 2010 at 6:32 AM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> in the future please report issues to jira http://jira.freeswitch.org
>>
>> please try svn trunk 16527 or higher
>>
>> This was not a bug but I made it work the way you describe since it made
>> sense.
>>
>> you should have done
>>
>> filter delete unique-id
>>
>> which would have delete all the unique-id filters that was the only option
>>
>>
>> you should be able to now say
>>
>> filter delete unique-id <value>
>>
>> To delete entry with specific value
>>
>> or
>>
>> filter delete unique-id
>>
>> to delete all entries with matching key
>>
>>
>>
>>
>>
>> On Wed, Jan 27, 2010 at 7:14 AM, Nagalenoj H. <nagalenoj at gmail.com>wrote:
>>
>>> Dear friends,
>>>    I've tried to delete the filter which I applied for an unique id. But,
>>> it doesn't work. After executing 'filter delete', I am receiving the events
>>> from that uuid.
>>>    I used the command as 'filter delete unique-id
>>> c6d2e0e2-0b2f-11df-9e84-fb15c3cd8565'.
>>>
>>> I did the following operations.
>>>     Made call to the event socket.
>>>     Registered events for all. (events plain all).
>>>     Applied filter for the uuid. (filter unique-id
>>> aa3cb8ea-0b2f-11df-9e84-fb15c3cd8565).
>>>     I've got a new uuid by using create_uuid.
>>>     Applied filter for this new uuid. (filter unique-id
>>> c6d2e0e2-0b2f-11df-9e84-fb15c3cd8565)
>>>     Originated a call with that uuid.
>>>     Now, I could receive events from both uuids. (Tested by giving DTMFs
>>> in both end and checked unique-id in event header).
>>>     Then, I wanted to delete a uuid from the filter. (filter delete
>>> unique-id c6d2e0e2-0b2f-11df-9e84-fb15c3cd8565).
>>>     I thought, i won't receive the events from this deleted unique-id.
>>> But, I received the dtmfs from both unique-id.
>>>
>>> I'm using 'FreeSWITCH Version 1.0.trunk (15982)'.
>>> --
>>> Regards,
>>> Nagalenoj H.
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Regards,
> Nagalenoj H.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100204/5184adb5/attachment-0002.html 

From anthony.minessale at gmail.com  Thu Feb  4 07:54:22 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 4 Feb 2010 09:54:22 -0600
Subject: [Freeswitch-users] A leg disconnects after uuid_bridge?
In-Reply-To: <fe67a3341002040452v5468fd1fh96d1c5062091474b@mail.gmail.com>
References: <fe67a3341002021934m548cde89p4d48381c5abe02a0@mail.gmail.com>
	<191c3a031002022108ta22287ev40fc36420abba2e5@mail.gmail.com>
	<fe67a3341002030112yeea379u90c22c482614c57@mail.gmail.com>
	<87f2f3b91002030605j7a1f9a02g665ba043c6ed7f7b@mail.gmail.com>
	<fe67a3341002040132n3aa2a897m60c5b9f1ca6b347f@mail.gmail.com>
	<0A571E14-BB58-4A3D-996A-9D5205668E85@freeswitch.org>
	<fe67a3341002040452v5468fd1fh96d1c5062091474b@mail.gmail.com>
Message-ID: <191c3a031002040754r2987a03fo18ee2e1ceb239f6a@mail.gmail.com>

   1. 2010-02-04 14:30:09.574084 [DEBUG] switch_core_state_machine.c:354 (
   sofia/internal/1010 at 192.168.1.222) State SOFT_EXECUTE going to sleep
   2. 2010-02-04 14:30:09.574084 [DEBUG] switch_core_state_machine.c:314 (
   sofia/internal/1010 at 192.168.1.222) Running State Change CS_EXECUTE
   3. 2010-02-04 14:30:09.574084 [DEBUG] switch_core_state_machine.c:348 (
   sofia/internal/1010 at 192.168.1.222) State EXECUTE
   4. 2010-02-04 14:30:09.574084 [DEBUG] mod_sofia.c:181 sofia/internal/1010
   @192.168.1.222 SOFIA EXECUTE
   5. 2010-02-04 14:30:09.574084 [DEBUG]
switch_core_state_machine.c:159sofia/internal/
   1010 at 192.168.1.222 Standard EXECUTE
   6. 2010-02-04 14:30:09.574084 [NOTICE]
switch_core_state_machine.c:187Hangup sofia/internal/
   1010 at 192.168.1.222 [CS_EXECUTE] [NORMAL_CLEARING]



Your channel went back to EXECUTE as expected then it hungup because there
were no more instructions in your dial plan for it to execute.  So it is
working as expected.

Consider using transfer_after_bridge variable or park_after bridge to make
it stay around when the call is over.



On Thu, Feb 4, 2010 at 6:52 AM, Nagalenoj H. <nagalenoj at gmail.com> wrote:

> By using create_uuid. I've also tried without giving origination_uuid. But,
> the result is same.
>
> --
> Regards,
> Nagalenoj H.
>
>
> On Thu, Feb 4, 2010 at 5:42 PM, Brian West <brian at freeswitch.org> wrote:
>
>> Where are you getting this UUID?
>>
>> /b
>>
>> On Feb 4, 2010, at 3:32 AM, Nagalenoj H. wrote:
>>
>> > api originate
>> {origination_uuid=c6d2e0e2-0b2f-11df-9e84-fb15c3cd8565}user/1001 &park
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100204/7cc6eeff/attachment-0002.html 

From rupa at rupa.com  Thu Feb  4 07:58:17 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Thu, 4 Feb 2010 09:58:17 -0600
Subject: [Freeswitch-users] max calls from a gateway
In-Reply-To: <867692.53487.qm@web34301.mail.mud.yahoo.com>
References: <867692.53487.qm@web34301.mail.mud.yahoo.com>
Message-ID: <b39273531002040758y6ef0491bo7d57accc83b1864b@mail.gmail.com>

If that works, let me know.  I just grepped the source for:

 ack-grep max[-_]calls

and while setting that will set profile->max_calls, I don't see anything
that actually acts on that.

My standard recommendation for setting limits is to use...  umm..  mod_limit
:)

On Wed, Feb 3, 2010 at 2:29 PM, FERNANDO VILLARROEL
<fvillarroel at yahoo.com>wrote:

> Dear.
>
> How i can do for limits inbound calls from a gateway:
>
>
> My config is like this:
>
> ~sip_profiles/external/
>
> gateway1.xml
>
> <include>
>    <gateway name="foo">
>      <param name="username" value="foo"/>
>      <param name="password" value="foo"/>
>      <param name="realm" value="xx.xxx.xxx.xx"/>
>      <param name="proxy" value="xx.xxx.xxx.xx/>
>      <param name="register" value="false"/>
>      <param name="context" value="public"/>
>      <param name="max_calls" value="30"/>-->
>    </gateway>
> </include>
>
> It?s fine?
>
> Fernando
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
-Rupa
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100204/59177e37/attachment-0002.html 

From maciej.aniserowicz at gmail.com  Thu Feb  4 09:56:14 2010
From: maciej.aniserowicz at gmail.com (Maciej Aniserowicz)
Date: Thu, 4 Feb 2010 09:56:14 -0800 (PST)
Subject: [Freeswitch-users] Trunk compilation error on Windows "Cannot
 open source file: '..\..\celt-0.7.0-1\libcelt\kfft_single.c'"
In-Reply-To: <FC7358A4-5486-4E61-BABE-8FDD5D8D52BF@freeswitch.org>
References: <1265295190863-4513865.post@n2.nabble.com>
	<FC7358A4-5486-4E61-BABE-8FDD5D8D52BF@freeswitch.org>
Message-ID: <1265306174527-4515042.post@n2.nabble.com>


I updated, checked, took a break and then sent email -> the break should not
be there I guess :). Thanks, it works now.
-- 
View this message in context: http://n2.nabble.com/Trunk-compilation-error-on-Windows-Cannot-open-source-file-celt-0-7-0-1-libcelt-kfft-single-c-tp4513865p4515042.html
Sent from the freeswitch-users mailing list archive at Nabble.com.



From jerry.richards at teotech.com  Thu Feb  4 11:29:21 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Thu, 4 Feb 2010 11:29:21 -0800
Subject: [Freeswitch-users] Stripping Leading Digits of 10-digit Inbound
	Number
Message-ID: <59D52ECCA7234EB9ABC3EC66E241656A@greyhawk.tonecommander.com>

I am using a Sangoma PRI card.  When an inbound call is received, where do
the leading digits get stripped off?  For example, if the inbound called
number is 4257405381, I notice the call is routed to 5381 extension, but I
don't know what is stripping off the 425740 digits.

Best Regards,
Jerry




From msc at freeswitch.org  Thu Feb  4 11:45:05 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 4 Feb 2010 11:45:05 -0800
Subject: [Freeswitch-users] Fw: need some hints on Softswitch deployment
	of FreeSwitch
In-Reply-To: <609766.80114.qm@web65415.mail.ac4.yahoo.com>
References: <609766.80114.qm@web65415.mail.ac4.yahoo.com>
Message-ID: <87f2f3b91002041145y6b5e8a61m9ad86f330e9f952f@mail.gmail.com>

On Wed, Feb 3, 2010 at 10:00 PM, Prabin Shrestha <nepaligas at yahoo.com>wrote:

>
>
> --- On *Tue, 2/2/10, Prabin Shrestha <nepaligas at yahoo.com>* wrote:
>
>
> From: Prabin Shrestha <nepaligas at yahoo.com>
> Subject: need some hints on Softswitch deployment of FreeSwitch
> To: freeswitch-users at lists.freeswitch.org
> Date: Tuesday, 2 February, 2010, 11:27 PM
>
>
>
> Dear all,
>
> I had been browsing through all the wikis of freeswitch, googling more than
> 1 week and couldn't figure out where to start.
>
> Welcome to FreeSWITCH. Yes, it is overwhelming at first. Telephony and VoIP
are deep subjects. Give yourself a lot of time to learn - you've got a long
road ahead.

> I have been finding so many problems and IRC thing I don't understand.
> Basically, I am just a average linux user running Ubuntu, trying to build a
> softswitch. It there was some book on freeswitch it would have been much
> easier for newbie like me.
>
> Be careful what you wish for! ;) In the meantime you can check out this
article from Linux Pro magazine:
http://bit.ly/EpVrv

Also, are you installing from the source? We much prefer to see people on
the SVN trunk and installing from source. Using the Q&D install is very easy
and is very predictable. Check this out:
http://wiki.freeswitch.org/wiki/Quick_and_Dirty_Install

Here are some problems I have been facing.
> After installation, I found freeswitch in /opt/freeswitch directory.
> only creating freeswitch user, I can access it's fs_cli, and I have yet to
> learn the power of it.
> doing ps -A, I found freeswitch is running in background.
>
> Now comes the hard part.
> I wanted to test it using SPA3000 device with fxo and fxs ports, which
> after following guides in net,  is not working for me.
>
> Is the SPA3000 the old Sipura version of the newer Linksys/Cisco SPA3102? 1
FXO and 1 FXS port, right? The FXS ports are easy to configure - you just
need to have the Phone port register to FreeSWITCH as a user. The FXO port
is a bit more challenging because you have to handle inbound and outbound
calls differently. It would take too much time to write up how to do it here
in this email. I would look at the principles found here:
http://wiki.freeswitch.org/wiki/SPA400_FreeSwitch_HowTo


> My reqirement is,
> 1. to run freeswitch as a softswitch which can route calls from voip call
> providers to Quintum gateways.
>
> Should be doable. Anything crazy or unusual about the Quintum gateways?

> 2. to have complete CDR reports generated to sql database.
>
> Totally doable, just you need to do it after the call completes. FS always
writes CSV CDRs to disk. You can use the SQL CDR template to create easily
loadable CDR records that can drop right into a MySQL/PostgreSQL/etc. db
backed. (You can also use mod_xml_cdr but that's a much more difficult
proposition, so get to know FS first before going to the deep end of the
pool.)

>
> Some light on this matter will be highly appreciated.
>
> Relax and breath. This is a really long and drawn out process. It just
takes a lot of time and patience to learn it all. Sorry, there just aren't
any shortcuts.
-MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100204/acaa80b2/attachment-0002.html 

From msc at freeswitch.org  Thu Feb  4 11:48:30 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 4 Feb 2010 11:48:30 -0800
Subject: [Freeswitch-users] Stripping Leading Digits of 10-digit Inbound
	Number
In-Reply-To: <59D52ECCA7234EB9ABC3EC66E241656A@greyhawk.tonecommander.com>
References: <59D52ECCA7234EB9ABC3EC66E241656A@greyhawk.tonecommander.com>
Message-ID: <87f2f3b91002041148s33bc755p57d211282f3c8cd2@mail.gmail.com>

On Thu, Feb 4, 2010 at 11:29 AM, Jerry Richards
<jerry.richards at teotech.com>wrote:

> I am using a Sangoma PRI card.  When an inbound call is received, where do
> the leading digits get stripped off?  For example, if the inbound called
> number is 4257405381, I notice the call is routed to 5381 extension, but I
> don't know what is stripping off the 425740 digits.
>
> Are you receiving all 10 digits from the carrier? You might only be
receiving four digits. The log will show you. I think it's a green log line.
If you need help then capture the debug output and drop into pastebin.
-MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100204/06a86f9b/attachment-0002.html 

From brian at freeswitch.org  Thu Feb  4 12:06:31 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 4 Feb 2010 14:06:31 -0600
Subject: [Freeswitch-users] Stripping Leading Digits of 10-digit Inbound
	Number
In-Reply-To: <87f2f3b91002041148s33bc755p57d211282f3c8cd2@mail.gmail.com>
References: <59D52ECCA7234EB9ABC3EC66E241656A@greyhawk.tonecommander.com>
	<87f2f3b91002041148s33bc755p57d211282f3c8cd2@mail.gmail.com>
Message-ID: <0A76C7F3-343A-4F2E-BF1A-65216E97CFD0@freeswitch.org>

Sounds like your provider just isn't delivering them to you.

/b

On Feb 4, 2010, at 1:48 PM, Michael Collins wrote:

> On Thu, Feb 4, 2010 at 11:29 AM, Jerry Richards <jerry.richards at teotech.com> wrote:
> I am using a Sangoma PRI card.  When an inbound call is received, where do
> the leading digits get stripped off?  For example, if the inbound called
> number is 4257405381, I notice the call is routed to 5381 extension, but I
> don't know what is stripping off the 425740 digits.
> 
> Are you receiving all 10 digits from the carrier? You might only be receiving four digits. The log will show you. I think it's a green log line. If you need help then capture the debug output and drop into pastebin.
> -MC

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100204/73d4c89d/attachment-0002.html 

From jerry.richards at teotech.com  Thu Feb  4 15:56:05 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Thu, 4 Feb 2010 15:56:05 -0800
Subject: [Freeswitch-users] Difference Between "bridge" and "transfer"
Message-ID: <D123D09A29B942D988B3456528DD44AD@greyhawk.tonecommander.com>

What is the difference between "bridge" and "transfer"?  I'm looking at the
demo IVRs.

Thanks,
Jerry




From msc at freeswitch.org  Thu Feb  4 16:32:40 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 4 Feb 2010 16:32:40 -0800
Subject: [Freeswitch-users] Difference Between "bridge" and "transfer"
In-Reply-To: <D123D09A29B942D988B3456528DD44AD@greyhawk.tonecommander.com>
References: <D123D09A29B942D988B3456528DD44AD@greyhawk.tonecommander.com>
Message-ID: <87f2f3b91002041632t19f17aefoaf11e048150816c5@mail.gmail.com>

On Thu, Feb 4, 2010 at 3:56 PM, Jerry Richards
<jerry.richards at teotech.com>wrote:

> What is the difference between "bridge" and "transfer"?  I'm looking at the
> demo IVRs.
>
>
bridge will connect two endpoints together while transfer sends the endpoint
back through the dialplan again...

-MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100204/fa5fab9c/attachment-0002.html 

From brian at freeswitch.org  Thu Feb  4 17:08:10 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 4 Feb 2010 19:08:10 -0600
Subject: [Freeswitch-users] Difference Between "bridge" and "transfer"
In-Reply-To: <87f2f3b91002041632t19f17aefoaf11e048150816c5@mail.gmail.com>
References: <D123D09A29B942D988B3456528DD44AD@greyhawk.tonecommander.com>
	<87f2f3b91002041632t19f17aefoaf11e048150816c5@mail.gmail.com>
Message-ID: <0F758845-5AE6-47C8-B1C8-F13C4CC1C756@freeswitch.org>

which then can result in a bridge being called again.

/b

On Feb 4, 2010, at 6:32 PM, Michael Collins wrote:

> bridge will connect two endpoints together while transfer sends the endpoint back through the dialplan again...




From andrew at hijacked.us  Thu Feb  4 17:28:24 2010
From: andrew at hijacked.us (Andrew Thompson)
Date: Thu, 4 Feb 2010 20:28:24 -0500
Subject: [Freeswitch-users] Difference Between "bridge" and "transfer"
In-Reply-To: <0F758845-5AE6-47C8-B1C8-F13C4CC1C756@freeswitch.org>
References: <D123D09A29B942D988B3456528DD44AD@greyhawk.tonecommander.com>
	<87f2f3b91002041632t19f17aefoaf11e048150816c5@mail.gmail.com>
	<0F758845-5AE6-47C8-B1C8-F13C4CC1C756@freeswitch.org>
Message-ID: <20100205012824.GD21394@hijacked.us>

On Thu, Feb 04, 2010 at 07:08:10PM -0600, Brian West wrote:
> which then can result in a bridge being called again.
> 
> /b
> 
> On Feb 4, 2010, at 6:32 PM, Michael Collins wrote:
> 
> > bridge will connect two endpoints together while transfer sends the endpoint back through the dialplan again...
> 
> 
Don't forget that the dialplan can be an endpoint too (via mod_loopback)
:P.

Andrew



From lists at redbonez.net  Thu Feb  4 18:09:59 2010
From: lists at redbonez.net (Adam Ford)
Date: Thu, 4 Feb 2010 19:09:59 -0700
Subject: [Freeswitch-users] Passing channel variables to mod_fifo and xml_cdr
Message-ID: <004301caa608$534747d0$f9d5d770$@net>

When sending a call through mod_fifo I seem to be losing my custom channel
variables that were assigned during prior processing of the call.  In my
example, I am trying to assign a unique identifier at the time the call
enters my FreeSWITCH system in order to more easily tie the xml_cdr logs
together.  This works great, until a call is processed through mod_fifo,
which drops my custom channel variable in the calls that it generates.  Is
it likely that I have something wrong with my config? Or does mod_fifo not
support the passing of custom channel variables?

 

The overall problem I am trying to solve is that mod_fifo generates a
separate a-leg for every time it rings an agent.  If the agent answers, the
a-leg log gets tied to the associated b-leg log with the uuids and I am able
to see the entire call in xml_cdr.  However, if the agent rejects the call
or doesn't answer, the a-leg is abandoned with seemingly no association back
to a b-leg log.  Anyone have a better suggestion for tying all these xml_cdr
logs together for a full picture of the call?  

 

-Adam

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100204/068d58f2/attachment-0002.html 

From dujinfang at gmail.com  Thu Feb  4 18:46:01 2010
From: dujinfang at gmail.com (Seven Du)
Date: Fri, 5 Feb 2010 10:46:01 +0800
Subject: [Freeswitch-users] about bgapi
In-Reply-To: <b39273531002040519g71fa8a37uc9f0819c0a19772f@mail.gmail.com>
References: <314dc3f81002040333n28c15d92l5dd0aeef4dc5aa92@mail.gmail.com>
	<b39273531002040519g71fa8a37uc9f0819c0a19772f@mail.gmail.com>
Message-ID: <23f91031002041846n2ab49e30gb1e3e37a9b7d69ba@mail.gmail.com>

Or wait for a certain event.

http://fisheye.freeswitch.org/browse/~raw,r=14500/FreeSWITCH/contrib/seven/lua/gateway_report.lua

2010/2/4 Rupa Schomaker <rupa at rupa.com>:
> you can't. ?If you want to terminate it, then you should set a global var
> that the script periodically checks and if set the script should terminate
> itself.
>
> On Thu, Feb 4, 2010 at 5:33 AM, Chia-Yen Wu <xanlich at gmail.com> wrote:
>>
>> I tried to run a javascript by background API (bgapi jsrun test.js)
>> this javascript (test.js) wont automatically stop, I tried to kill it by
>> "uuid_kill" command with Job-UUID
>> but return "-ERR no such channel!", how can I kill this bgapi? btw my FS
>> run in windows.
>> thank you
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From wiltingtree at gmail.com  Thu Feb  4 18:48:07 2010
From: wiltingtree at gmail.com (Adam Wilt)
Date: Thu, 4 Feb 2010 21:48:07 -0500
Subject: [Freeswitch-users] Running a session asynchronously from a Lua
	script
Message-ID: <eb7797d71002041848h21e70a13t5ab3ba0004c6075a@mail.gmail.com>

Hi all,

I want to park an inbound call and play hold music while I simultaneously
place another outbound call.
But the hold music doesn't play while the lua script is placing the second
call. When the lua script ends, the hold music finally starts.
Here's my example code:

#!/usr/local/bin/lua

session:answer()
api = freeswitch.API()
api:executeString("bgapi uuid_park " .. tostring(session.uuid))
api:executeString("bgapi uuid_broadcast " .. tostring(session.uuid)  .. "
/freeswitch/sounds/music/8000/danza-espanola-op-37-h-142-xii-arabesca.wav")
local new_session =
freeswitch.Session("sofia/gateway/myprovider/15555555555")


So it seems like the script is blocking the original session, despite the
fact that I'm using bgapi.  I'd really appreciate if somebody could help me
with this.

By the way, I'm using FreeSWITCH 1.0.4 in Windows.

Thanks,
Adam
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100204/f4f4ea15/attachment-0002.html 

From brian at freeswitch.org  Thu Feb  4 19:23:46 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 4 Feb 2010 21:23:46 -0600
Subject: [Freeswitch-users] Running a session asynchronously from a Lua
	script
In-Reply-To: <eb7797d71002041848h21e70a13t5ab3ba0004c6075a@mail.gmail.com>
References: <eb7797d71002041848h21e70a13t5ab3ba0004c6075a@mail.gmail.com>
Message-ID: <0BD2ED36-2279-412A-AF43-554E15B596E2@freeswitch.org>

I wouldn't do it like that...

I would let FreeSWITCH do what its good at and stop trying to create and manage sessions manually.

-- You can optionally answer but you need to set transfer_ringback instead of ringback.
session:setVariable("ringback", "local_stream://moh");                                                                                                          
session:setVariable("ignore_early_media", "true");                                                                                                              
session:execute("bridge","user/1007");  

Which can also be expressed in pure XML as such:

<extension name="1234">
  <condition field="destination_number" expression="^1234$">                                                                                                
    <action application="set" data="ringback=local_stream://moh"/>
    <action application="set" data="ignore_early_media=true"/>
    <action application="bridge" data="user/1007"/>
  </condition>
</extension>

There is really no need to do this with Lua.  The XML dialplan can do some VERY complex things once you wrap your head around it.

In addition I would recommend you get the latest 1.0.5 build http://files-sync.freeswitch.org/windows_installer/  Its tagged as 1.0.4 and shouldn't be.

/b



On Feb 4, 2010, at 8:48 PM, Adam Wilt wrote:

> Hi all,
> 
> I want to park an inbound call and play hold music while I simultaneously place another outbound call.
> But the hold music doesn't play while the lua script is placing the second call. When the lua script ends, the hold music finally starts. 
> Here's my example code:
> 
> #!/usr/local/bin/lua
>    
> session:answer()
> api = freeswitch.API()
> api:executeString("bgapi uuid_park " .. tostring(session.uuid))
> api:executeString("bgapi uuid_broadcast " .. tostring(session.uuid)  .. " /freeswitch/sounds/music/8000/danza-espanola-op-37-h-142-xii-arabesca.wav")
> local new_session = freeswitch.Session("sofia/gateway/myprovider/15555555555")
> 
> So it seems like the script is blocking the original session, despite the fact that I'm using bgapi.  I'd really appreciate if somebody could help me with this.
> 
> By the way, I'm using FreeSWITCH 1.0.4 in Windows.
> 
> Thanks,
> Adam

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100204/c4c1fc03/attachment-0002.html 

From ustcorporation at yahoo.com  Thu Feb  4 19:38:53 2010
From: ustcorporation at yahoo.com (Darren C.)
Date: Thu, 4 Feb 2010 19:38:53 -0800 (PST)
Subject: [Freeswitch-users]  Seeking Advice on SIP Phones like Aastra
Message-ID: <845952.61278.qm@web33007.mail.mud.yahoo.com>

Tim,
 
Many thanks for your response.  I posted this message on the Dev list and all I heard was crickets.  I would think a web GUI for a phone would be in demand by the FS community....
 
We are doing something similar to what you described.  We?re developing a rather complex IVR/Switching application and it currently does all its database writes via our Web Service to an MS SQL database.  We have a web site that is updated with call details via the web service?s backend database.  From this web interface a user can see counts of voicemails, see call activity, play voicemails, see calls in progress, record calls, etc.  It?s a specialized application so it doesn?t have every PBX feature but this is what we wanted to do with a high-end SIP Phone to replace our office PBX.
 
Currently we have an ESI (Estech) E-Class PBX that uses normal digital phones as well as proprietary VOIP (non-SIP) phones.  I think these really nice SIP phones with huge color touchscreens would be much better than even high-end proprietary digital phones + we?d get all the benefits of FS.  We?ll just need to add all the basic PBX capabilities to the phone?s GUI to see how many voicemails are waiting, how many lines are in use, button for call transfer, etc.
 
As you mentioned:  ?you can do a DB query to find out what voicemail messages a user has, or what calls are active, etc.?  We are keeping track of all this ourselves via our web service?s backend database.  I?m not sure I need to do this for everything but we are.  It?s a multi-tenant system with hundreds of tenants so I?m guessing I might lose some needed relationships by querying FS but I?m going to re-visit this?I will make sure I can?t just query FS like you?re doing for some of these things?we?ve never turned on ODBC or even looked for the documentation as to what FS stores.  We have to run FS in Linux for some Sangoma stuff but we?re Windows people so that is another reason we store via web services to an MS SQLbackend.
 
I was worried I?d get one of these fancy phones and find out it doesn?t support important SIP/FS features rendering the color touchscreen useless.  I?ve never owned one of these SIP phones, just used various softphones.  But thanks to you I?ll get an Aastra 6739i and give it a try.
 
I have a fulltime programmer working on this system on and off now for over a year but this has been mostly developing the IVR.  We haven't made any attempt at using FS as a PBX.  So based on your comments I think I?ll purchase an Aastra 6739i and develop a custom SIP Phone GUI interface with FS.
 
If you or others would like to collaborate on an Aastra 6739i phone GUI for FS, feel free to contact me.  We can try to make it extensible for other phones as well.  My email is ustcor<remove this junk>poration at yah<remove this too>oo.com.  
 
Thanks,

teldev


      
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100204/de6721c7/attachment-0002.html 

From wiltingtree at gmail.com  Thu Feb  4 19:49:36 2010
From: wiltingtree at gmail.com (Adam Wilt)
Date: Thu, 4 Feb 2010 22:49:36 -0500
Subject: [Freeswitch-users] Running a session asynchronously from a Lua
	script
In-Reply-To: <0BD2ED36-2279-412A-AF43-554E15B596E2@freeswitch.org>
References: <eb7797d71002041848h21e70a13t5ab3ba0004c6075a@mail.gmail.com>
	<0BD2ED36-2279-412A-AF43-554E15B596E2@freeswitch.org>
Message-ID: <eb7797d71002041949t7a3fd84dgcfa3f63c69465612@mail.gmail.com>

Brian,

Thanks for the reply.
The problem is I don't want the two parties to speak to each other. I want
one party to wait on hold while the system interacts with the other party.
Then the system will hang-up on the second party and start interacting with
the first party again.

Thanks,
Adam


On Thu, Feb 4, 2010 at 10:23 PM, Brian West <brian at freeswitch.org> wrote:

> I wouldn't do it like that...
>
> I would let FreeSWITCH do what its good at and stop trying to create and
> manage sessions manually.
>
> -- You can optionally answer but you need to set transfer_ringback instead
> of ringback.
> session:setVariable("ringback", "local_stream://moh");
>
>
> session:setVariable("ignore_early_media", "true");
>
>
> session:execute("bridge","user/1007");
>
> Which can also be expressed in pure XML as such:
>
> <extension name="1234">
>   <condition field="destination_number" expression="^1234$">
>
>
>     <action application="set" data="ringback=local_stream://moh"/>
>     <action application="set" data="ignore_early_media=true"/>
>     <action application="bridge" data="user/1007"/>
>   </condition>
> </extension>
>
> There is really no need to do this with Lua.  The XML dialplan can do some
> VERY complex things once you wrap your head around it.
>
> In addition I would recommend you get the latest 1.0.5 build
> http://files-sync.freeswitch.org/windows_installer/  Its tagged as 1.0.4
> and shouldn't be.
>
> /b
>
>
>
> On Feb 4, 2010, at 8:48 PM, Adam Wilt wrote:
>
> Hi all,
>
> I want to park an inbound call and play hold music while I simultaneously
> place another outbound call.
> But the hold music doesn't play while the lua script is placing the second
> call. When the lua script ends, the hold music finally starts.
> Here's my example code:
>
> #!/usr/local/bin/lua
>
> session:answer()
> api = freeswitch.API()
> api:executeString("bgapi uuid_park " .. tostring(session.uuid))
> api:executeString("bgapi uuid_broadcast " .. tostring(session.uuid)  .. "
> /freeswitch/sounds/music/8000/danza-espanola-op-37-h-142-xii-arabesca.wav")
> local new_session =
> freeswitch.Session("sofia/gateway/myprovider/15555555555")
>
>
> So it seems like the script is blocking the original session, despite the
> fact that I'm using bgapi.  I'd really appreciate if somebody could help me
> with this.
>
> By the way, I'm using FreeSWITCH 1.0.4 in Windows.
>
> Thanks,
> Adam
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100204/22e157d3/attachment-0002.html 

From brian at freeswitch.org  Thu Feb  4 19:57:24 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 4 Feb 2010 21:57:24 -0600
Subject: [Freeswitch-users] Running a session asynchronously from a Lua
	script
In-Reply-To: <eb7797d71002041949t7a3fd84dgcfa3f63c69465612@mail.gmail.com>
References: <eb7797d71002041848h21e70a13t5ab3ba0004c6075a@mail.gmail.com>
	<0BD2ED36-2279-412A-AF43-554E15B596E2@freeswitch.org>
	<eb7797d71002041949t7a3fd84dgcfa3f63c69465612@mail.gmail.com>
Message-ID: <DD852357-BDC6-425D-81ED-1AE807EE2F48@freeswitch.org>

ok the use the api to "originate" the call to say an extension that plays music.  But if they never talk to each other what is the point of the two calls in the same script?

/b

On Feb 4, 2010, at 9:49 PM, Adam Wilt wrote:

> Brian, 
> 
> Thanks for the reply.  
> The problem is I don't want the two parties to speak to each other. I want one party to wait on hold while the system interacts with the other party. Then the system will hang-up on the second party and start interacting with the first party again.
> 
> Thanks,
> Adam




From nagalenoj at gmail.com  Thu Feb  4 20:41:39 2010
From: nagalenoj at gmail.com (Nagalenoj H.)
Date: Fri, 5 Feb 2010 10:11:39 +0530
Subject: [Freeswitch-users] A leg disconnects after uuid_bridge?
In-Reply-To: <191c3a031002040754r2987a03fo18ee2e1ceb239f6a@mail.gmail.com>
References: <fe67a3341002021934m548cde89p4d48381c5abe02a0@mail.gmail.com>
	<191c3a031002022108ta22287ev40fc36420abba2e5@mail.gmail.com>
	<fe67a3341002030112yeea379u90c22c482614c57@mail.gmail.com>
	<87f2f3b91002030605j7a1f9a02g665ba043c6ed7f7b@mail.gmail.com>
	<fe67a3341002040132n3aa2a897m60c5b9f1ca6b347f@mail.gmail.com>
	<0A571E14-BB58-4A3D-996A-9D5205668E85@freeswitch.org>
	<fe67a3341002040452v5468fd1fh96d1c5062091474b@mail.gmail.com>
	<191c3a031002040754r2987a03fo18ee2e1ceb239f6a@mail.gmail.com>
Message-ID: <fe67a3341002042041u1ee70b86u3bc91adbb6fb3d52@mail.gmail.com>

Sorry., I couldn't understand its behavior.

Let me ask the same question in this way.

* hangup_after_bridge is set to false.
* In outbound socket, first I answer the call.
* When I do a bridge to a extension (1001), after 1001 disconnects the call.
I am able to make another call.
      sendmsg
      call-command: execute
      execute-app-name: bridge
      execute-app-arg: user/1001

* When I originate a call to extension (1001), after 1001 disconnects the
call. I'm unable to make another call, because my session is also getting
closed.
      api originate user/1001 &park

      Content-Type: api/response
      Content-Length: 41

     +OK 1fac17ce-120b-11df-a878-d9c7fbcf71c4

      sendmsg
      call-command: execute
      execute-app-name: intercept
      execute-app-arg: 1fac17ce-120b-11df-a878-d9c7fbcf71c4

* In both the case, the call is getting bridged to an extension and
hangup_after_bridge is false.
* When bridge doesn't need any other variables to set to continue, why
intercept needs a explicit park after bridge.?

Hope, this has some clarity.,

On Thu, Feb 4, 2010 at 9:24 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

>
>
>    1. 2010-02-04 14:30:09.574084 [DEBUG] switch_core_state_machine.c:354 (
>    sofia/internal/1010 at 192.168.1.222) State SOFT_EXECUTE going to sleep
>    2. 2010-02-04 14:30:09.574084 [DEBUG] switch_core_state_machine.c:314 (
>    sofia/internal/1010 at 192.168.1.222) Running State Change CS_EXECUTE
>    3. 2010-02-04 14:30:09.574084 [DEBUG] switch_core_state_machine.c:348 (
>    sofia/internal/1010 at 192.168.1.222) State EXECUTE
>    4. 2010-02-04 14:30:09.574084 [DEBUG] mod_sofia.c:181 sofia/internal/
>    1010 at 192.168.1.222 SOFIA EXECUTE
>    5. 2010-02-04 14:30:09.574084 [DEBUG] switch_core_state_machine.c:159sofia/internal/
>    1010 at 192.168.1.222 Standard EXECUTE
>    6. 2010-02-04 14:30:09.574084 [NOTICE] switch_core_state_machine.c:187Hangup sofia/internal/
>    1010 at 192.168.1.222 [CS_EXECUTE] [NORMAL_CLEARING]
>
>
>
> Your channel went back to EXECUTE as expected then it hungup because there
> were no more instructions in your dial plan for it to execute.  So it is
> working as expected.
>
> Consider using transfer_after_bridge variable or park_after bridge to make
> it stay around when the call is over.
>
>
>
>
> On Thu, Feb 4, 2010 at 6:52 AM, Nagalenoj H. <nagalenoj at gmail.com> wrote:
>
>> By using create_uuid. I've also tried without giving origination_uuid.
>> But, the result is same.
>>
>> --
>> Regards,
>> Nagalenoj H.
>>
>>
>> On Thu, Feb 4, 2010 at 5:42 PM, Brian West <brian at freeswitch.org> wrote:
>>
>>> Where are you getting this UUID?
>>>
>>> /b
>>>
>>> On Feb 4, 2010, at 3:32 AM, Nagalenoj H. wrote:
>>>
>>> > api originate
>>> {origination_uuid=c6d2e0e2-0b2f-11df-9e84-fb15c3cd8565}user/1001 &park
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Regards,
Nagalenoj H.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100205/203be6f6/attachment-0002.html 

From matt at webcontracts.co.uk  Thu Feb  4 04:22:13 2010
From: matt at webcontracts.co.uk (Matthew Law)
Date: Thu, 4 Feb 2010 12:22:13 -0000
Subject: [Freeswitch-users] Simple IAX2 setup - help with converting from
 asterisk to freeswitch
Message-ID: <91093d24f2166d220acf2ab19115f95d.squirrel@www.webcontracts.co.uk>

Hi,

I have compiled freeswitch trunk from svn on debian.  I am trying to
convert the following simple working asterisk config to freeswitch and I
would be really grateful if someone could point me in the right direction:

pbx:/etc/asterisk# cat iax.conf
[general]
bindport=4569
bindaddr=my.ip.address
jitterbuffer=yes
disallow=all
allow=ulaw
allow=alaw
context=deadend

[voiptalk]
type=peer
username=XXXXXX
secret=XXXXXX
host=iax.voiptalk.org

[0843XXXXXX]
type=friend
username=08433XXXXXX
context=incoming
requirecalltoken=auto

[1000]
type=friend
host=dynamic
mailbox=1000
secret=XXXXXX
context=phones
requirecalltoken=auto

pbx:/etc/asterisk# cat extensions.conf
[general]
autofallthrough=yes

[outgoing]
exten => _0[1-9].,1,Dial(IAX2/XXXXXX at voiptalk/44${EXTEN:1})
exten => _00.,1,Dial(IAX2/XXXXXX at voiptalk/${EXTEN:2})

[internal]
exten => 901,1,VoiceMailMain()
exten => 901,2,Hangup()
exten => 902,1,MeetMe(1234,cdM)
exten => 902,2,Hangup()

[incoming]
exten => XXXXXXXX,1,Dial(IAX2/1000,30)
exten => XXXXXXXX,2,VoiceMail(1000 at internal)

[phones]
include => internal
include => outgoing

pbx:/etc/asterisk# cat voicemail.conf
[internal]
1000 => XXXX,Matt,me at myomain.com
format=wav49
maxsilence=0


Sorry to be a bonehead, but I'm struggling with the wiki docs (especially
with regard to IAX2) and also concerned about security.  I installed the
samples when I compiled freeswitch but wondering if that is a security
risk as the PBX box is on the public internet?

Many thanks,

Matt






From tim at novion.ru  Thu Feb  4 11:47:57 2010
From: tim at novion.ru (=?KOI8-R?B?9MnN1dIg98HMydvF1w==?=)
Date: Thu, 4 Feb 2010 22:47:57 +0300
Subject: [Freeswitch-users] No reINVITE when bridging two sessions from
	JavaScript with bypass_media_after_bridge=true
Message-ID: <8e9d67561002041147q6c135546u1f7e1887529a916d@mail.gmail.com>

Dear colleagues,

The task is to start two sessions from JS script and then bridge them
in no-media mode. Unfirtunately, FreeSwitch does not reINVITE the
peers after bridging.

Here is my script:

SCRIPT #1 - 'callback-session.js' <<<<<<<<<<<<<<<<<<<<<<<<<<
session = new Session("{bypass_media_after_bridge=true,ignore_early_media=true,hangup_after_bridge=true}user/1001");
session2 = new Session("{ignore_early_media=true}user/1001");
bridge(session, session2);
>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>

Then I run this script from the freeswitch console:

freeswitch at internal> jsrun callback-session.js

And there is no re-invites between peers, peers get connected and the
traffic goes through FS

BUT!!! Here is another scipt:

SCRIPT #2 - 'callback-bridge.js' <<<<<<<<<<<<<<<<<<<<<<<<<<
session = new Session("{bypass_media_after_bridge=true,continue_on_fail=true,ignore_early_media=true}user/1001");
session.execute("bridge","{ignore_early_media=false,originate_timeout=90}user/1001");
>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>

When I run this script, FreeSwitch successfuly sends reINVITES to both
users after bridge, so they exchange media directly, not through FS.
In my task, I need to have control when B-leg establishes (to start
billing correctly), so I need to get the first scenario working.

What I've already tried and did not succeed:
1) set bypass_media=true, on A leg only, on B leg only, on both legs
2) set bypass_media_after_bridge=true, on A leg only, on B leg only,
on both legs

Is there something I do wrong in the first script? What should I do to
make FS reINVITE peers? Many thanks in advance!

Best regards,
Timur Valishev
sip:tim at novion.ru



From christian at officepools.com  Thu Feb  4 14:08:56 2010
From: christian at officepools.com (Christain Jensen)
Date: Thu, 4 Feb 2010 14:08:56 -0800
Subject: [Freeswitch-users] Looking for some good/cheap desktop phones
Message-ID: <3f2e071b5aea5e11bb37a37e435a33b7@mail.gmail.com>

Hi,



I am looking for a vendor for some (3-5) desktop voip phones. Any
suggestions?



Christian
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100204/e206ac28/attachment-0002.html 

From tim at novion.ru  Thu Feb  4 21:46:00 2010
From: tim at novion.ru (=?KOI8-R?B?9MnN1dIg98HMydvF1w==?=)
Date: Fri, 5 Feb 2010 08:46:00 +0300
Subject: [Freeswitch-users] No reINVITE when bridging two sessions from
	JavaScript with bypass_media_after_bridge=true
In-Reply-To: <8e9d67561002041147q6c135546u1f7e1887529a916d@mail.gmail.com>
References: <8e9d67561002041147q6c135546u1f7e1887529a916d@mail.gmail.com>
Message-ID: <8e9d67561002042146l3b4265f9sb87270d5b0adac68@mail.gmail.com>

Dear colleagues,

The task is to start two sessions from JS script and then bridge them
in no-media mode. Unfirtunately, FreeSwitch does not reINVITE the
peers after bridging.

Here is my script:

SCRIPT #1 - 'callback-session.js' <<<<<<<<<<<<<<<<<<<<<<<<<<
session = new Session("{bypass_media_after_bridge=true,ignore_early_media=true,hangup_after_bridge=true}user/1001");
session2 = new Session("{ignore_early_media=true}user/1001");
bridge(session, session2);
>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>

Then I run this script from the freeswitch console:

freeswitch at internal> jsrun callback-session.js

And there is no re-invites between peers, peers get connected and the
traffic goes through FS

BUT!!! Here is another scipt:

SCRIPT #2 - 'callback-bridge.js' <<<<<<<<<<<<<<<<<<<<<<<<<<
session = new Session("{bypass_media_after_bridge=true,continue_on_fail=true,ignore_early_media=true}user/1001");
session.execute("bridge","{ignore_early_media=false,originate_timeout=90}user/1001");
>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>

When I run this script, FreeSwitch successfuly sends reINVITES to both
users after bridge, so they exchange media directly, not through FS.
In my task, I need to have control when B-leg establishes (to start
billing correctly), so I need to get the first scenario working.

What I've already tried and did not succeed:
1) set bypass_media=true, on A leg only, on B leg only, on both legs
2) set bypass_media_after_bridge=true, on A leg only, on B leg only,
on both legs

Is there something I do wrong in the first script? What should I do to
make FS reINVITE peers? Many thanks in advance!

Best regards,
Timur Valishev
sip:tim at novion.ru



From tim at novion.ru  Thu Feb  4 22:20:41 2010
From: tim at novion.ru (=?KOI8-R?B?9MnN1dIg98HMydvF1w==?=)
Date: Fri, 5 Feb 2010 09:20:41 +0300
Subject: [Freeswitch-users] No reINVITE when bridging two sessions from
	JavaScript with bypass_media_after_bridge=true
In-Reply-To: <8e9d67561002041147q6c135546u1f7e1887529a916d@mail.gmail.com>
References: <8e9d67561002041147q6c135546u1f7e1887529a916d@mail.gmail.com>
Message-ID: <8e9d67561002042220t59dfbf97x50c2a0bf804c5e5f@mail.gmail.com>

Dear colleagues,

The task is to start two sessions from JS script and then bridge them
in no-media mode. Unfirtunately, FreeSwitch does not reINVITE the
peers after bridging.

Here is my script:

SCRIPT #1 - 'callback-session.js' <<<<<<<<<<<<<<<<<<<<<<<<<<
session = new Session("{bypass_media_after_bridge=true,ignore_early_media=true,hangup_after_bridge=true}user/1001");
session2 = new Session("{ignore_early_media=true}user/1001");
bridge(session, session2);
>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>

Then I run this script from the freeswitch console:

freeswitch at internal> jsrun callback-session.js

And there is no re-invites between peers, peers get connected and the
traffic goes through FS. Is there something I do wrong in the script?
What should I do to
make FS reINVITE peers? Many thanks in advance!

Best regards,
Timur Valishev



From tim at novion.ru  Thu Feb  4 22:36:19 2010
From: tim at novion.ru (=?KOI8-R?B?9MnN1dIg98HMydvF1w==?=)
Date: Fri, 5 Feb 2010 09:36:19 +0300
Subject: [Freeswitch-users] Looking for some good/cheap desktop phones
In-Reply-To: <3f2e071b5aea5e11bb37a37e435a33b7@mail.gmail.com>
References: <3f2e071b5aea5e11bb37a37e435a33b7@mail.gmail.com>
Message-ID: <8e9d67561002042236s434e0643od322551de8720be1@mail.gmail.com>

Have a look at Yealink (Skypemate) and Fanvill

2010/2/5 Christain Jensen <christian at officepools.com>:
> Hi,
>
>
>
> I am looking for a vendor for some (3-5) desktop voip phones. Any
> suggestions?
>
>
>
> Christian
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From brian at freeswitch.org  Thu Feb  4 22:41:27 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 5 Feb 2010 00:41:27 -0600
Subject: [Freeswitch-users] No reINVITE when bridging two sessions from
	JavaScript with bypass_media_after_bridge=true
In-Reply-To: <8e9d67561002041147q6c135546u1f7e1887529a916d@mail.gmail.com>
References: <8e9d67561002041147q6c135546u1f7e1887529a916d@mail.gmail.com>
Message-ID: <3D739D7F-F5C6-4266-B04E-9413DCFDA6AF@freeswitch.org>

http://wiki.freeswitch.org/wiki/Bypass_Media

Also make sure you're on SVN trunk.

/b

On Feb 4, 2010, at 1:47 PM, ????? ??????? wrote:

> Is there something I do wrong in the first script? What should I do to
> make FS reINVITE peers? Many thanks in advance!




From mike at jerris.com  Thu Feb  4 22:43:00 2010
From: mike at jerris.com (Michael Jerris)
Date: Fri, 5 Feb 2010 01:43:00 -0500
Subject: [Freeswitch-users] Simple IAX2 setup - help with converting
	from asterisk to freeswitch
In-Reply-To: <91093d24f2166d220acf2ab19115f95d.squirrel@www.webcontracts.co.uk>
References: <91093d24f2166d220acf2ab19115f95d.squirrel@www.webcontracts.co.uk>
Message-ID: <E1E3F6EF-1A1E-4891-B371-7A3BF0988E95@jerris.com>

iax2 support has been removed from FreeSWITCH in current trunk and will not be in the 1.0.5 release.

On Feb 4, 2010, at 7:22 AM, Matthew Law wrote:

> Hi,
> 
> I have compiled freeswitch trunk from svn on debian.  I am trying to
> convert the following simple working asterisk config to freeswitch and I
> would be really grateful if someone could point me in the right direction:
> 
<snip>

> Sorry to be a bonehead, but I'm struggling with the wiki docs (especially
> with regard to IAX2) and also concerned about security.  I installed the
> samples when I compiled freeswitch but wondering if that is a security
> risk as the PBX box is on the public internet?

Mike




From brian at freeswitch.org  Thu Feb  4 22:43:37 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 5 Feb 2010 00:43:37 -0600
Subject: [Freeswitch-users] Looking for some good/cheap desktop phones
In-Reply-To: <8e9d67561002042236s434e0643od322551de8720be1@mail.gmail.com>
References: <3f2e071b5aea5e11bb37a37e435a33b7@mail.gmail.com>
	<8e9d67561002042236s434e0643od322551de8720be1@mail.gmail.com>
Message-ID: <409D0ACF-ADA4-4D02-8B7E-71EC1B1BAFBC@freeswitch.org>

And all of those are awful phones.  They don't even make good paper weights.

You can't have good and cheap in the same sentence when talking about VoIP phones.  You have to take your pick between quality (good) and price (cheap) you can't have both at once.

/b


On Feb 5, 2010, at 12:36 AM, ????? ??????? wrote:

> Have a look at Yealink (Skypemate) and Fanvill




From tim at novion.ru  Thu Feb  4 23:01:42 2010
From: tim at novion.ru (=?KOI8-R?B?9MnN1dIg98HMydvF1w==?=)
Date: Fri, 5 Feb 2010 10:01:42 +0300
Subject: [Freeswitch-users] Looking for some good/cheap desktop phones
In-Reply-To: <409D0ACF-ADA4-4D02-8B7E-71EC1B1BAFBC@freeswitch.org>
References: <3f2e071b5aea5e11bb37a37e435a33b7@mail.gmail.com>
	<8e9d67561002042236s434e0643od322551de8720be1@mail.gmail.com>
	<409D0ACF-ADA4-4D02-8B7E-71EC1B1BAFBC@freeswitch.org>
Message-ID: <8e9d67561002042301v76129abfj3dcbc70ed073752c@mail.gmail.com>

Sure, those phones do not deliver superior usability, but they at
least give the best sound among budget models.

2010/2/5 Brian West <brian at freeswitch.org>:
> And all of those are awful phones. ?They don't even make good paper weights.
>
> You can't have good and cheap in the same sentence when talking about VoIP phones. ?You have to take your pick between quality (good) and price (cheap) you can't have both at once.
>
> /b
>
>
> On Feb 5, 2010, at 12:36 AM, ????? ??????? wrote:
>
>> Have a look at Yealink (Skypemate) and Fanvill
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From ustcorporation at yahoo.com  Thu Feb  4 23:11:54 2010
From: ustcorporation at yahoo.com (Darren C.)
Date: Thu, 4 Feb 2010 23:11:54 -0800 (PST)
Subject: [Freeswitch-users] Seeking Advice on SIP Phones like Aastra 6739i
	or Snom 870 that have good interoperability with FreeSWITCH
Message-ID: <174266.76158.qm@web33008.mail.mud.yahoo.com>

Tim,

Many thanks for your response.  My first response became an orphan/new post on the list...I think if subject line is too long reply looses thread.

We are doing something similar to what you described.  We?re developing a rather complex IVR/Switching application and it currently does all its database writes via our Web Service to an MS SQL database.  We have a web site that is updated with call details via the web service?s backend database.  From this web interface a user can see counts of voicemails, see call activity, play voicemails, see calls in progress, record calls, etc.  It?s a specialized application so it doesn?t have every PBX feature but this is what we wanted to do with a high-end SIP Phone to replace our office PBX.

Currently we have an ESI (Estech) E-Class PBX that uses normal digital phones as well as proprietary VOIP (non-SIP) phones.  I think these really nice SIP phones with huge color touchscreens would be much better than even high-end proprietary digital phones + we?d get all the benefits of FS.  We?ll just need to add all the basic PBX capabilities to the phone?s GUI to see how many voicemails are waiting, how many lines are in use, button for call transfer, etc.

As you mentioned:  ?you can do a DB query to find out what voicemail messages a user has, or what calls are active, etc.?  We are keeping track of all this ourselves via our web service?s backend database.  I?m not sure I need to do this for everything but we are.  It?s a multi-tenant system with hundreds of tenants so I?m guessing I might lose some needed relationships by querying FS but I?m going to re-visit this?I will make sure I can?t just query FS like you?re doing for some of these things?we?ve never turned on ODBC or even looked for the documentation as to what FS stores.  We have to run FS in Linux for some Sangoma stuff but we?re Windows people so that is another reason we store via web services to an MS SQLbackend.

I was worried I?d get one of these fancy phones and find out it doesn?t support important SIP/FS features rendering the color touchscreen useless.  I?ve never owned one of these SIP phones, just used various softphones.  But thanks to you I?ll get an Aastra 6739i and give it a try.

I have a fulltime programmer working on this system on and off now for over a year but this has been mostly developing the IVR.  We haven't made any attempt at using FS as a PBX.  So based on your comments I think I?ll purchase an Aastra 6739i and develop a custom SIP Phone GUI interface with FS.

If you or others would like to collaborate on an Aastra 6739i phone GUI for FS, feel free to contact me.  We can try to make it extensible for other phones as well.  My email is ustcor<remove this junk>poration at yah<remove this too>oo.com.  

Thanks,

teldev


      
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100204/eb1b26d0/attachment-0002.html 

From tim at novion.ru  Fri Feb  5 00:18:15 2010
From: tim at novion.ru (Timur Valishev)
Date: Fri, 5 Feb 2010 11:18:15 +0300
Subject: [Freeswitch-users] No reINVITE when bridging two sessions from
	JavaScript with bypass_media_after_bridge=true
In-Reply-To: <3D739D7F-F5C6-4266-B04E-9413DCFDA6AF@freeswitch.org>
References: <8e9d67561002041147q6c135546u1f7e1887529a916d@mail.gmail.com>
	<3D739D7F-F5C6-4266-B04E-9413DCFDA6AF@freeswitch.org>
Message-ID: <8e9d67561002050018q24e3a806kc8807b55268e759b@mail.gmail.com>

Thank you for reply, Brian!

http://wiki.freeswitch.org/wiki/Bypass_Media says:
*>Can I use bypass media when executing the bridge application from a
javascript? *
*>Of course you can, all it takes is setting the bypass_media session
variable to true before the bridge: *
*>session.setVariable('bypass_media', 'true'*);

I've modified my script to make sure: <<<<<<<<<<<<<<<<<<<<<<<<<<<<,
*session = new Session(*
*"{ignore_early_media=true,hangup_after_bridge=true}sofia/external/
timwork at novion.ru"*
*);*
*session2 = new Session(*
*"{ignore_early_media=true}sofia/external/timwork at novion.ru"*
*);*
*session.setVariable('bypass_media', 'true');*
*session2.setVariable('bypass_media', 'true');*
*bridge(session, session2);*
>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>

But there is still no reINVITE =(

SVN trunk I'm building from seems to be fresh: <<<<<<<<<<<<<<<
*[root at sip freeswitch.trunk]# svn info*
*Path: .*
*URL: http://svn.freeswitch.org:/svn/freeswitch/trunk*
*Repository Root: http://svn.freeswitch.org:/svn*
*Repository UUID: d0543943-73ff-0310-b7d9-9358b9ac24b2*
*Revision: 16561*
*Node Kind: directory*
*Schedule: normal*
*Last Changed Author: mcollins*
*Last Changed Rev: 16561*
*Last Changed Date: 2010-02-03 04:53:31 +0300 (Wed, 03 Feb 2010)*
>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>

Do you have any ideas how to make FS reINVITE in this scenario?

2010/2/5 Brian West <brian at freeswitch.org>:
> http://wiki.freeswitch.org/wiki/Bypass_Media
>
> Also make sure you're on SVN trunk.
>
> /b
>
> On Feb 4, 2010, at 1:47 PM, ????? ??????? wrote:
>
>> Is there something I do wrong in the first script? What should I do to
>> make FS reINVITE peers? Many thanks in advance!
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100205/f5f42fee/attachment-0002.html 

From tayeb.meftah at gmail.com  Fri Feb  5 04:09:08 2010
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Fri, 05 Feb 2010 13:09:08 +0100
Subject: [Freeswitch-users] Looking for some good/cheap desktop phones
In-Reply-To: <3f2e071b5aea5e11bb37a37e435a33b7@mail.gmail.com>
References: <3f2e071b5aea5e11bb37a37e435a33b7@mail.gmail.com>
Message-ID: <4B6C0A64.3060500@gmail.com>

hi
try linksys SPA901
Le 04/02/2010 23:08, Christain Jensen a ?crit :
>
> Hi,
>
> I am looking for a vendor for some (3-5) desktop voip phones. Any 
> suggestions?
>
> Christian
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>    

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100205/0b83da11/attachment-0002.html 

From jcasale at activenetwerx.com  Fri Feb  5 04:41:41 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Fri, 5 Feb 2010 12:41:41 +0000
Subject: [Freeswitch-users] Looking for some good/cheap desktop phones
In-Reply-To: <4B6C0A64.3060500@gmail.com>
References: <3f2e071b5aea5e11bb37a37e435a33b7@mail.gmail.com>
	<4B6C0A64.3060500@gmail.com>
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E90CD9E1@prato.activenetwerx.local>

>try linksys SPA901

Use Linksys support just once, then tell me if you still want
any of their product...



From dave at 3c.co.uk  Fri Feb  5 05:04:05 2010
From: dave at 3c.co.uk (David Knell)
Date: Fri, 05 Feb 2010 06:04:05 -0700
Subject: [Freeswitch-users] Looking for some good/cheap desktop phones
In-Reply-To: <8e9d67561002042301v76129abfj3dcbc70ed073752c@mail.gmail.com>
References: <3f2e071b5aea5e11bb37a37e435a33b7@mail.gmail.com>
	<8e9d67561002042236s434e0643od322551de8720be1@mail.gmail.com>
	<409D0ACF-ADA4-4D02-8B7E-71EC1B1BAFBC@freeswitch.org>
	<8e9d67561002042301v76129abfj3dcbc70ed073752c@mail.gmail.com>
Message-ID: <1265375045.12871.27.camel@local.freepabx.com>

Some notes from a grumpy old luddite:

I have one of the Yealink USB desk speakerphones, and I don't think you
can get a better usability and audio quality to price ratio anywhere on
the market.  And it just worked.

The Aastra 6757i (also on my somewhat cluttered desk) cost more than ten
times as much, was a PITA to set up (its UPnP sporadically crashed my
WiFi router which was entertaining until I worked out what was going on
and turned it off), has a configuration interface that would send Steve
Jobs out hunting those responsible with an elephant gun were it an Apple
product and, once it was finally configured and working, gave me really
no more usable functionality on the deskphone than the el cheapo one
above.  The cordless handset's nice to have, though.

--Dave

> Sure, those phones do not deliver superior usability, but they at
> least give the best sound among budget models.
> 
> 2010/2/5 Brian West <brian at freeswitch.org>:
> > And all of those are awful phones.  They don't even make good paper weights.
> >
> > You can't have good and cheap in the same sentence when talking about VoIP phones.  You have to take your pick between quality (good) and price (cheap) you can't have both at once.
> >
> > /b
> >
> >
> > On Feb 5, 2010, at 12:36 AM, ????? ??????? wrote:
> >
> >> Have a look at Yealink (Skypemate) and Fanvill
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From yehavi.bourvine at gmail.com  Fri Feb  5 05:07:09 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Fri, 5 Feb 2010 15:07:09 +0200
Subject: [Freeswitch-users] Looking for some good/cheap desktop phones
In-Reply-To: <8e9d67561002042301v76129abfj3dcbc70ed073752c@mail.gmail.com>
References: <3f2e071b5aea5e11bb37a37e435a33b7@mail.gmail.com>
	<8e9d67561002042236s434e0643od322551de8720be1@mail.gmail.com>
	<409D0ACF-ADA4-4D02-8B7E-71EC1B1BAFBC@freeswitch.org>
	<8e9d67561002042301v76129abfj3dcbc70ed073752c@mail.gmail.com>
Message-ID: <d361ca7b1002050507n5e564621oa6cf7ace005dd5b8@mail.gmail.com>

>From my experience Polycom and SNOM are expensive but give you what you
need. Polycom is more intutive to the users but more cumbersome for the
manager to deploy; SNOM is somewhat less intuitive to the user but
everything can be set via the WEB interface.

If you talk about 4-5 phones, then probably SNOM is the choice. It also
depends about the specific functions you want to use. I our specific
environment (high use of BLF and shared lines) Polycom wins because it
handles these functions just as the user expects.

I did not try Aastra so cannot testify. We did test Yealink, Thomson,
Asterphone, SipTip and maybe others I forgot. Cisco also seems good but
Cisco does not supply the required socumentation to make them fully working.

                    Regards, __Yehavi:

2010/2/5 ????? ??????? <tim at novion.ru>

> Sure, those phones do not deliver superior usability, but they at
> least give the best sound among budget models.
>
> 2010/2/5 Brian West <brian at freeswitch.org>:
>  > And all of those are awful phones.  They don't even make good paper
> weights.
> >
> > You can't have good and cheap in the same sentence when talking about
> VoIP phones.  You have to take your pick between quality (good) and price
> (cheap) you can't have both at once.
> >
> > /b
> >
> >
> > On Feb 5, 2010, at 12:36 AM, ????? ??????? wrote:
> >
> >> Have a look at Yealink (Skypemate) and Fanvill
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100205/183cf9e4/attachment-0002.html 

From tculjaga at gmail.com  Fri Feb  5 05:21:25 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Fri, 5 Feb 2010 14:21:25 +0100
Subject: [Freeswitch-users] Looking for some good/cheap desktop phones
In-Reply-To: <d361ca7b1002050507n5e564621oa6cf7ace005dd5b8@mail.gmail.com>
References: <3f2e071b5aea5e11bb37a37e435a33b7@mail.gmail.com>
	<8e9d67561002042236s434e0643od322551de8720be1@mail.gmail.com>
	<409D0ACF-ADA4-4D02-8B7E-71EC1B1BAFBC@freeswitch.org>
	<8e9d67561002042301v76129abfj3dcbc70ed073752c@mail.gmail.com>
	<d361ca7b1002050507n5e564621oa6cf7ace005dd5b8@mail.gmail.com>
Message-ID: <65d96fc81002050521w746707b1q4398e6d8a0692d1e@mail.gmail.com>

Atcom AT-620 (
http://www.atcom.cn/AT620.html#~tab-small_midsized_large_enterprises<http://www.atcom.cn/AT620.html#%7Etab-small_midsized_large_enterprises>)
is quite ok and cheap (~30$)... also we have been talking to Atcom to add a
sort of auto-provissioning (dhcp/http) and this is going to happen next
week.


T.


2010/2/5 Yehavi Bourvine <yehavi.bourvine at gmail.com>

> From my experience Polycom and SNOM are expensive but give you what you
> need. Polycom is more intutive to the users but more cumbersome for the
> manager to deploy; SNOM is somewhat less intuitive to the user but
> everything can be set via the WEB interface.
>
> If you talk about 4-5 phones, then probably SNOM is the choice. It also
> depends about the specific functions you want to use. I our specific
> environment (high use of BLF and shared lines) Polycom wins because it
> handles these functions just as the user expects.
>
> I did not try Aastra so cannot testify. We did test Yealink, Thomson,
> Asterphone, SipTip and maybe others I forgot. Cisco also seems good but
> Cisco does not supply the required socumentation to make them fully working.
>
>                     Regards, __Yehavi:
>
> 2010/2/5 ????? ??????? <tim at novion.ru>
>
> Sure, those phones do not deliver superior usability, but they at
>> least give the best sound among budget models.
>>
>> 2010/2/5 Brian West <brian at freeswitch.org>:
>>  > And all of those are awful phones.  They don't even make good paper
>> weights.
>> >
>> > You can't have good and cheap in the same sentence when talking about
>> VoIP phones.  You have to take your pick between quality (good) and price
>> (cheap) you can't have both at once.
>> >
>> > /b
>> >
>> >
>> > On Feb 5, 2010, at 12:36 AM, ????? ??????? wrote:
>> >
>> >> Have a look at Yealink (Skypemate) and Fanvill
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100205/df4afa81/attachment-0002.html 

From Prometheus001 at gmx.net  Fri Feb  5 05:27:39 2010
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Fri, 05 Feb 2010 14:27:39 +0100
Subject: [Freeswitch-users] mod_skypiax - help needed with
 modified	snd-dummy (one way audio)
In-Reply-To: <7b197bef1001290520x37179cb1o63273410e9ef5bc@mail.gmail.com>
References: <4B60555B.2020004@gmx.net> <4B607944.4040700@gmx.net>
	<4B608014.4030902@gmx.net>	<7b197bef1001271215o64918f56s3d16e51528fc5f66@mail.gmail.com>
	<4B61ECE0.10409@gmx.net>	<7b197bef1001281210u43a907edi13afe794cf0e1a2e@mail.gmail.com>
	<4B61FCAB.5040707@gmx.net>	<7b197bef1001281341h7ee2c58fj79b3886630901f29@mail.gmail.com>
	<4B62C962.7000601@gmx.net>	<8213d6071001290403g2c08fe39gd92eb0446bb6485a@mail.gmail.com>
	<7b197bef1001290520x37179cb1o63273410e9ef5bc@mail.gmail.com>
Message-ID: <4B6C1CCB.4080606@gmx.net>

Hello Giovanni,

as I couldn't even get skype again working again with the standard alsa
driver, I would like to setup the machine from scratch based on a
working machine.
The latest errors I received from Skype was:
  snd_pcm_avail_update() returned a value that is exceptionally large:
715706624 bytes (3727638 ms).
  Most likely this is a bug in the ALSA driver. Please report this issue
to the ALSA developers.
I think that may be the reason for one-way-audio.

For setting up my machine from scratch, please advise:
- which OS you are you using und recommending exactly?
- I would like to use 64bit OS in order to use 8GB of memory, does this
work?
- any other hints?

Best regards
Peter

Giovanni Maruzzelli schrieb:
> Peter,
>
> Can you connect on IRC (irc.freenode.net #freeswitch)? I'm gmaruzz there.
>
> Can you restate your problems? I've lost connection :)
>
> with snd-dummy custom you can create *one only* snd-dummy instance, so
> *one only* fake soundcard. If you create more, will not work. But with
> that one fake soundcard you can use 64 skype client instances, all
> with the same soundcard hardware device (hw:n).
>
> with original snd-dummy you can create a max of 8 instances, so 8 fake
> soundcards, and with each fake soundcard you can  use a max of 8 skype
> client instances.
>
> use the hardware devices, not the default devices (use the "hw:n")
>
> -giovanni
>
> On Fri, Jan 29, 2010 at 1:03 PM, Ghulam Mustafa <mustafa.pk at gmail.com> wrote:
>   
>> did you enable debug mode while compiling custom snd-dummy? if  yes
>> try re-compiling with debug mode disabled.
>>
>> -m
>>
>> On Fri, Jan 29, 2010 at 4:41 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>     
>>> I now reinstalled the original sound drivers
>>> Unfortunaltely the sound problems remain, not that worse but they are there:
>>> Audio is still (almost) one way. Almost means:
>>>
>>>    * SIP -> Skype ok
>>>    * Skype=> SIP I hear only some scratching on very loud audio
>>>
>>> Could it be a volume problem? But snd-dummy should have no volume
>>> properties, right?
>>>
>>> Best regards
>>> Peter
>>>
>>>
>>> Giovanni Maruzzelli schrieb:
>>>       
>>>> with three instances you will assign the hw:0 device to skype client
>>>> 0...7, hw:1 to skype client 8...15 and hw:2 to skype client 16...23.
>>>> Must work. Pay attention to assign the same device name to all devices
>>>> needed by a skype instance (sound devices window): playback, capture
>>>> AND ring.
>>>>
>>>> Or maybe is a bug of ALSA on Debian...
>>>>
>>>> -giovanni
>>>>
>>>> On Thu, Jan 28, 2010 at 10:07 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>
>>>>         
>>>>> I crated 3 instances of snd-dummy, this worked. I assigned then Instance
>>>>> #2 to the Skype accounts. Still no sound.
>>>>> On the frist call there is one way audio, on the following calls there
>>>>> is no audio at all.
>>>>> This is weird.
>>>>>
>>>>> Best regards
>>>>> Peter
>>>>>
>>>>> Giovanni Maruzzelli schrieb:
>>>>>
>>>>>           
>>>>>> Ciao Peter,
>>>>>>
>>>>>> Never tested on Debian 5.
>>>>>>
>>>>>> When you write "same problem" you are referring to the audio going one
>>>>>> way only (btw, which way?) with the custom audio driver?
>>>>>>
>>>>>> Have you tried with multiple instances of the regular Debian
>>>>>> snd-dummy, as I wrote in a mail before?
>>>>>>
>>>>>> -gm
>>>>>>
>>>>>>
>>>>>>
>>>>>> On Thu, Jan 28, 2010 at 9:00 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>>
>>>>>>
>>>>>>             
>>>>>>> Hello Giovanni,
>>>>>>>
>>>>>>> I did so but the same problem again.
>>>>>>>
>>>>>>> Did you ever test in on Debian 5.0?
>>>>>>>
>>>>>>> Best reards
>>>>>>> Peter
>>>>>>>
>>>>>>> Giovanni Maruzzelli schrieb:
>>>>>>>
>>>>>>>
>>>>>>>               
>>>>>>>> good, so you have only one sound device, the right one.
>>>>>>>>
>>>>>>>> Use the one with hw:0 in the window that skype gives you to set sound devices
>>>>>>>>
>>>>>>>> -gm
>>>>>>>>
>>>>>>>> On Wed, Jan 27, 2010 at 7:04 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>                 
>>>>>>>>> I installed alsa-utile,
>>>>>>>>>
>>>>>>>>> now I get:
>>>>>>>>>
>>>>>>>>> skype:/var/cache/apt/archives# aplay -l
>>>>>>>>> **** List of PLAYBACK Hardware Devices ****
>>>>>>>>> card 0: Dummy [Dummy], device 0: Dummy PCM [Dummy PCM]
>>>>>>>>>  Subdevices: 127/128
>>>>>>>>>  Subdevice #0: subdevice #0
>>>>>>>>>  Subdevice #1: subdevice #1
>>>>>>>>>  Subdevice #2: subdevice #2
>>>>>>>>>  Subdevice #3: subdevice #3
>>>>>>>>>  Subdevice #4: subdevice #4
>>>>>>>>>  Subdevice #5: subdevice #5
>>>>>>>>>  Subdevice #6: subdevice #6
>>>>>>>>>  Subdevice #7: subdevice #7
>>>>>>>>>  Subdevice #8: subdevice #8
>>>>>>>>>  Subdevice #9: subdevice #9
>>>>>>>>>  Subdevice #10: subdevice #10
>>>>>>>>>  Subdevice #11: subdevice #11
>>>>>>>>>  Subdevice #12: subdevice #12
>>>>>>>>>  Subdevice #13: subdevice #13
>>>>>>>>>  Subdevice #14: subdevice #14
>>>>>>>>>  Subdevice #15: subdevice #15
>>>>>>>>>  Subdevice #16: subdevice #16
>>>>>>>>>  Subdevice #17: subdevice #17
>>>>>>>>>  Subdevice #18: subdevice #18
>>>>>>>>>  Subdevice #19: subdevice #19
>>>>>>>>>  Subdevice #20: subdevice #20
>>>>>>>>>  Subdevice #21: subdevice #21
>>>>>>>>>  Subdevice #22: subdevice #22
>>>>>>>>>  Subdevice #23: subdevice #23
>>>>>>>>>  Subdevice #24: subdevice #24
>>>>>>>>>  Subdevice #25: subdevice #25
>>>>>>>>>  Subdevice #26: subdevice #26
>>>>>>>>>  Subdevice #27: subdevice #27
>>>>>>>>>  Subdevice #28: subdevice #28
>>>>>>>>>  Subdevice #29: subdevice #29
>>>>>>>>>  Subdevice #30: subdevice #30
>>>>>>>>>  Subdevice #31: subdevice #31
>>>>>>>>>  Subdevice #32: subdevice #32
>>>>>>>>>  Subdevice #33: subdevice #33
>>>>>>>>>  Subdevice #34: subdevice #34
>>>>>>>>>  Subdevice #35: subdevice #35
>>>>>>>>>  Subdevice #36: subdevice #36
>>>>>>>>>  Subdevice #37: subdevice #37
>>>>>>>>>  Subdevice #38: subdevice #38
>>>>>>>>>  Subdevice #39: subdevice #39
>>>>>>>>>  Subdevice #40: subdevice #40
>>>>>>>>>  Subdevice #41: subdevice #41
>>>>>>>>>  Subdevice #42: subdevice #42
>>>>>>>>>  Subdevice #43: subdevice #43
>>>>>>>>>  Subdevice #44: subdevice #44
>>>>>>>>>  Subdevice #45: subdevice #45
>>>>>>>>>  Subdevice #46: subdevice #46
>>>>>>>>>  Subdevice #47: subdevice #47
>>>>>>>>>  Subdevice #48: subdevice #48
>>>>>>>>>  Subdevice #49: subdevice #49
>>>>>>>>>  Subdevice #50: subdevice #50
>>>>>>>>>  Subdevice #51: subdevice #51
>>>>>>>>>  Subdevice #52: subdevice #52
>>>>>>>>>  Subdevice #53: subdevice #53
>>>>>>>>>  Subdevice #54: subdevice #54
>>>>>>>>>  Subdevice #55: subdevice #55
>>>>>>>>>  Subdevice #56: subdevice #56
>>>>>>>>>  Subdevice #57: subdevice #57
>>>>>>>>>  Subdevice #58: subdevice #58
>>>>>>>>>  Subdevice #59: subdevice #59
>>>>>>>>>  Subdevice #60: subdevice #60
>>>>>>>>>  Subdevice #61: subdevice #61
>>>>>>>>>  Subdevice #62: subdevice #62
>>>>>>>>>  Subdevice #63: subdevice #63
>>>>>>>>>  Subdevice #64: subdevice #64
>>>>>>>>>  Subdevice #65: subdevice #65
>>>>>>>>>  Subdevice #66: subdevice #66
>>>>>>>>>  Subdevice #67: subdevice #67
>>>>>>>>>  Subdevice #68: subdevice #68
>>>>>>>>>  Subdevice #69: subdevice #69
>>>>>>>>>  Subdevice #70: subdevice #70
>>>>>>>>>  Subdevice #71: subdevice #71
>>>>>>>>>  Subdevice #72: subdevice #72
>>>>>>>>>  Subdevice #73: subdevice #73
>>>>>>>>>  Subdevice #74: subdevice #74
>>>>>>>>>  Subdevice #75: subdevice #75
>>>>>>>>>  Subdevice #76: subdevice #76
>>>>>>>>>  Subdevice #77: subdevice #77
>>>>>>>>>  Subdevice #78: subdevice #78
>>>>>>>>>  Subdevice #79: subdevice #79
>>>>>>>>>  Subdevice #80: subdevice #80
>>>>>>>>>  Subdevice #81: subdevice #81
>>>>>>>>>  Subdevice #82: subdevice #82
>>>>>>>>>  Subdevice #83: subdevice #83
>>>>>>>>>  Subdevice #84: subdevice #84
>>>>>>>>>  Subdevice #85: subdevice #85
>>>>>>>>>  Subdevice #86: subdevice #86
>>>>>>>>>  Subdevice #87: subdevice #87
>>>>>>>>>  Subdevice #88: subdevice #88
>>>>>>>>>  Subdevice #89: subdevice #89
>>>>>>>>>  Subdevice #90: subdevice #90
>>>>>>>>>  Subdevice #91: subdevice #91
>>>>>>>>>  Subdevice #92: subdevice #92
>>>>>>>>>  Subdevice #93: subdevice #93
>>>>>>>>>  Subdevice #94: subdevice #94
>>>>>>>>>  Subdevice #95: subdevice #95
>>>>>>>>>  Subdevice #96: subdevice #96
>>>>>>>>>  Subdevice #97: subdevice #97
>>>>>>>>>  Subdevice #98: subdevice #98
>>>>>>>>>  Subdevice #99: subdevice #99
>>>>>>>>>  Subdevice #100: subdevice #100
>>>>>>>>>  Subdevice #101: subdevice #101
>>>>>>>>>  Subdevice #102: subdevice #102
>>>>>>>>>  Subdevice #103: subdevice #103
>>>>>>>>>  Subdevice #104: subdevice #104
>>>>>>>>>  Subdevice #105: subdevice #105
>>>>>>>>>  Subdevice #106: subdevice #106
>>>>>>>>>  Subdevice #107: subdevice #107
>>>>>>>>>  Subdevice #108: subdevice #108
>>>>>>>>>  Subdevice #109: subdevice #109
>>>>>>>>>  Subdevice #110: subdevice #110
>>>>>>>>>  Subdevice #111: subdevice #111
>>>>>>>>>  Subdevice #112: subdevice #112
>>>>>>>>>  Subdevice #113: subdevice #113
>>>>>>>>>  Subdevice #114: subdevice #114
>>>>>>>>>  Subdevice #115: subdevice #115
>>>>>>>>>  Subdevice #116: subdevice #116
>>>>>>>>>  Subdevice #117: subdevice #117
>>>>>>>>>  Subdevice #118: subdevice #118
>>>>>>>>>  Subdevice #119: subdevice #119
>>>>>>>>>  Subdevice #120: subdevice #120
>>>>>>>>>  Subdevice #121: subdevice #121
>>>>>>>>>  Subdevice #122: subdevice #122
>>>>>>>>>  Subdevice #123: subdevice #123
>>>>>>>>>  Subdevice #124: subdevice #124
>>>>>>>>>  Subdevice #125: subdevice #125
>>>>>>>>>  Subdevice #126: subdevice #126
>>>>>>>>>  Subdevice #127: subdevice #127
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> Peter P GMX schrieb:
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>                   
>>>>>>>>>> Her's the output:
>>>>>>>>>>
>>>>>>>>>> skype:~# aplay -l
>>>>>>>>>> bash: aplay: command not found
>>>>>>>>>>
>>>>>>>>>> Giovanni Maruzzelli schrieb:
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>                     
>>>>>>>>>>> I don't think you got two snd-dummy loaded (but maybe yes)
>>>>>>>>>>> what's the output of:
>>>>>>>>>>>
>>>>>>>>>>> aplay -l
>>>>>>>>>>>
>>>>>>>>>>> ?
>>>>>>>>>>>
>>>>>>>>>>> If instead you are referring to the choices that skype clients offers
>>>>>>>>>>> you in the "set audio devices" window, choose Dummy PCM (hw0:0)
>>>>>>>>>>>
>>>>>>>>>>> Eg: not the "default", but the "hardware" one
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> On Wed, Jan 27, 2010 at 5:58 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>                       
>>>>>>>>>>>> Thanks Giovanni,
>>>>>>>>>>>>
>>>>>>>>>>>> I think there may be the problem, that I have 2 sound devices now:
>>>>>>>>>>>> - Dummy PCM (hw0:0) (this is from debian install)
>>>>>>>>>>>> - Dummy PCM Default Audio device (defauzlt: CARD=Dummy) (this is new
>>>>>>>>>>>> since I compiled alsa newly)
>>>>>>>>>>>>
>>>>>>>>>>>> I tried both, but both do not work. How do I get rid of the old alsa device?
>>>>>>>>>>>> By the way: I uninstalled Alsa before I installed the new driver
>>>>>>>>>>>> (apt-get remove alsa-utils alsa-base).
>>>>>>>>>>>>
>>>>>>>>>>>> Best regards
>>>>>>>>>>>> Peter
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>> Giovanni Maruzzelli schrieb:
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>                         
>>>>>>>>>>>>> This warning is harmless:
>>>>>>>>>>>>>
>>>>>>>>>>>>> Jan 27 15:28:50 skype kernel: [ 3994.795786] process `skype' is using
>>>>>>>>>>>>> obsolete setsockopt SO_BSDCOMPAT
>>>>>>>>>>>>>
>>>>>>>>>>>>> On Wed, Jan 27, 2010 at 4:26 PM, Giovanni Maruzzelli
>>>>>>>>>>>>> <gmaruzz at celliax.org> wrote:
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>                           
>>>>>>>>>>>>>> Ciao Peter
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> one instance of snd-dummy "customized" is enough for 64 instances of
>>>>>>>>>>>>>> skype clients, no need (and do not works) with more instances of
>>>>>>>>>>>>>> snd-dummy-customized.
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> Maybe you got the one-way problem because of kernel at 250HZ (don't
>>>>>>>>>>>>>> know). It uses to works well on a tickless kernel at 100HZ (eg: ubuntu
>>>>>>>>>>>>>> 8.04).
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> Or maybe you have to check and modify which sound devices the skype
>>>>>>>>>>>>>> clients are using (try to check that with snd-summy-custom loaded,
>>>>>>>>>>>>>> maybe with the ssh -X trick (as in the wiki page).
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> To load more than one snd-dummy-original (the non modified one), you
>>>>>>>>>>>>>> do this with the modprobe command, as in:
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> rmmod snd-dummy
>>>>>>>>>>>>>> modprobe snd-dummy enable=1,1,1
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> this command will enable three instances of snd-dummy original, so
>>>>>>>>>>>>>> you'll have three fake soundcards, and you'll have to setup each group
>>>>>>>>>>>>>> of 8 skype instances to use sound devices from one fake soundcard, RG:
>>>>>>>>>>>>>> no more than 8 skype client instances can use one instance of fake
>>>>>>>>>>>>>> soundcard.
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> Also, please update the mod_skypiax code (svn up in its directory) I
>>>>>>>>>>>>>> just committed some improvements.
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> If you have any other doubts, or need more info, don't hesitate to
>>>>>>>>>>>>>> write the mailing list again,
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> ciao for now,
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> -giovanni
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> On Wed, Jan 27, 2010 at 4:01 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>                             
>>>>>>>>>>>>>>> I have mod_skypiax working nicely so far with 2 Skype channels. Thanks
>>>>>>>>>>>>>>> to all contributors, excellent work!
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> In order to have more than 8 channels working, I have followed the
>>>>>>>>>>>>>>> instructions in
>>>>>>>>>>>>>>> http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk
>>>>>>>>>>>>>>> and compiled alsa-driver-1.0.20 with the modified dummy.c file. (System
>>>>>>>>>>>>>>> ist Debian 5.0R3)
>>>>>>>>>>>>>>> It compiled well however when I start snd-dummy I only have
>>>>>>>>>>>>>>> one-way-audio and my logs show
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> Jan 27 15:28:41 skype kernel: [ 3984.318403] snd-dummy skypiax driver,
>>>>>>>>>>>>>>> /usr/src/alsa-driver-1.0.20/drivers/../alsa-kernel/drivers/dummy.c:920
>>>>>>>>>>>>>>> working on a machine with 250HZ kernel
>>>>>>>>>>>>>>> Jan 27 15:28:50 skype kernel: [ 3994.795786] process `skype' is using
>>>>>>>>>>>>>>> obsolete setsockopt SO_BSDCOMPAT
>>>>>>>>>>>>>>> Jan 27 15:28:56 skype kernel: [ 4005.289907] __ratelimit: 490 messages
>>>>>>>>>>>>>>> suppressed
>>>>>>>>>>>>>>> Jan 27 15:29:01 skype kernel: [ 4012.458310] process `skype' is using
>>>>>>>>>>>>>>> obsolete setsockopt SO_BSDCOMPAT
>>>>>>>>>>>>>>> Jan 27 15:29:01 skype kernel: [ 4013.326290] __ratelimit: 499 messages
>>>>>>>>>>>>>>> suppressed
>>>>>>>>>>>>>>> If I reinstall alsa from deb everything sworks fine again (of course
>>>>>>>>>>>>>>> with the current limitations).
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> First question: Has anybody had this issue before? How can I solve this?
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> Second question:
>>>>>>>>>>>>>>> As I do not need 64 channels or more: how do I manage, that Skype
>>>>>>>>>>>>>>> instances 9..15 use a second instance of snd-dummy as addressed in the wiki?
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> Best regards
>>>>>>>>>>>>>>> Peter
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>                               
>>>>>>>>>>>>>> --
>>>>>>>>>>>>>> Sincerely,
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> Giovanni Maruzzelli
>>>>>>>>>>>>>> Cell : +39-347-2665618
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>                             
>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>                         
>>>>>>>>> _______________________________________________
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>                   
>>>>>>>>                 
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>               
>>>>>>
>>>>>>             
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>>           
>>>>
>>>>
>>>>         
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>       
>>
>> --
>> Ghulam Mustafa
>> cell: +92 333.611.7681
>> sip: cyrenity at ekiga.net
>> mail: mustafa.pk at gmail.com
>> web: cyrenity.wordpress.com
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>     
>
>
>
>   



From gmaruzz at celliax.org  Fri Feb  5 05:36:28 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Fri, 5 Feb 2010 14:36:28 +0100
Subject: [Freeswitch-users] mod_skypiax - help needed with modified
	snd-dummy (one way audio)
In-Reply-To: <4B6C1CCB.4080606@gmx.net>
References: <4B60555B.2020004@gmx.net>
	<7b197bef1001271215o64918f56s3d16e51528fc5f66@mail.gmail.com> 
	<4B61ECE0.10409@gmx.net>
	<7b197bef1001281210u43a907edi13afe794cf0e1a2e@mail.gmail.com> 
	<4B61FCAB.5040707@gmx.net>
	<7b197bef1001281341h7ee2c58fj79b3886630901f29@mail.gmail.com> 
	<4B62C962.7000601@gmx.net>
	<8213d6071001290403g2c08fe39gd92eb0446bb6485a@mail.gmail.com> 
	<7b197bef1001290520x37179cb1o63273410e9ef5bc@mail.gmail.com> 
	<4B6C1CCB.4080606@gmx.net>
Message-ID: <7b197bef1002050536j7ea09684v913434296ae52b97@mail.gmail.com>

Ciao Peter,

I would use ubuntu 8.04 (hardy) LTS server 64bit or CentOS 5.4 64bit.

-giovanni

On Fri, Feb 5, 2010 at 2:27 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
> Hello Giovanni,
>
> as I couldn't even get skype again working again with the standard alsa
> driver, I would like to setup the machine from scratch based on a
> working machine.
> The latest errors I received from Skype was:
> ?snd_pcm_avail_update() returned a value that is exceptionally large:
> 715706624 bytes (3727638 ms).
> ?Most likely this is a bug in the ALSA driver. Please report this issue
> to the ALSA developers.
> I think that may be the reason for one-way-audio.
>
> For setting up my machine from scratch, please advise:
> - which OS you are you using und recommending exactly?
> - I would like to use 64bit OS in order to use 8GB of memory, does this
> work?
> - any other hints?
>
> Best regards
> Peter
>
> Giovanni Maruzzelli schrieb:
>> Peter,
>>
>> Can you connect on IRC (irc.freenode.net #freeswitch)? I'm gmaruzz there.
>>
>> Can you restate your problems? I've lost connection :)
>>
>> with snd-dummy custom you can create *one only* snd-dummy instance, so
>> *one only* fake soundcard. If you create more, will not work. But with
>> that one fake soundcard you can use 64 skype client instances, all
>> with the same soundcard hardware device (hw:n).
>>
>> with original snd-dummy you can create a max of 8 instances, so 8 fake
>> soundcards, and with each fake soundcard you can ?use a max of 8 skype
>> client instances.
>>
>> use the hardware devices, not the default devices (use the "hw:n")
>>
>> -giovanni
>>
>> On Fri, Jan 29, 2010 at 1:03 PM, Ghulam Mustafa <mustafa.pk at gmail.com> wrote:
>>
>>> did you enable debug mode while compiling custom snd-dummy? if ?yes
>>> try re-compiling with debug mode disabled.
>>>
>>> -m
>>>
>>> On Fri, Jan 29, 2010 at 4:41 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>
>>>> I now reinstalled the original sound drivers
>>>> Unfortunaltely the sound problems remain, not that worse but they are there:
>>>> Audio is still (almost) one way. Almost means:
>>>>
>>>> ? ?* SIP -> Skype ok
>>>> ? ?* Skype=> SIP I hear only some scratching on very loud audio
>>>>
>>>> Could it be a volume problem? But snd-dummy should have no volume
>>>> properties, right?
>>>>
>>>> Best regards
>>>> Peter
>>>>
>>>>
>>>> Giovanni Maruzzelli schrieb:
>>>>
>>>>> with three instances you will assign the hw:0 device to skype client
>>>>> 0...7, hw:1 to skype client 8...15 and hw:2 to skype client 16...23.
>>>>> Must work. Pay attention to assign the same device name to all devices
>>>>> needed by a skype instance (sound devices window): playback, capture
>>>>> AND ring.
>>>>>
>>>>> Or maybe is a bug of ALSA on Debian...
>>>>>
>>>>> -giovanni
>>>>>
>>>>> On Thu, Jan 28, 2010 at 10:07 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>
>>>>>
>>>>>> I crated 3 instances of snd-dummy, this worked. I assigned then Instance
>>>>>> #2 to the Skype accounts. Still no sound.
>>>>>> On the frist call there is one way audio, on the following calls there
>>>>>> is no audio at all.
>>>>>> This is weird.
>>>>>>
>>>>>> Best regards
>>>>>> Peter
>>>>>>
>>>>>> Giovanni Maruzzelli schrieb:
>>>>>>
>>>>>>
>>>>>>> Ciao Peter,
>>>>>>>
>>>>>>> Never tested on Debian 5.
>>>>>>>
>>>>>>> When you write "same problem" you are referring to the audio going one
>>>>>>> way only (btw, which way?) with the custom audio driver?
>>>>>>>
>>>>>>> Have you tried with multiple instances of the regular Debian
>>>>>>> snd-dummy, as I wrote in a mail before?
>>>>>>>
>>>>>>> -gm
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> On Thu, Jan 28, 2010 at 9:00 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>> Hello Giovanni,
>>>>>>>>
>>>>>>>> I did so but the same problem again.
>>>>>>>>
>>>>>>>> Did you ever test in on Debian 5.0?
>>>>>>>>
>>>>>>>> Best reards
>>>>>>>> Peter
>>>>>>>>
>>>>>>>> Giovanni Maruzzelli schrieb:
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>> good, so you have only one sound device, the right one.
>>>>>>>>>
>>>>>>>>> Use the one with hw:0 in the window that skype gives you to set sound devices
>>>>>>>>>
>>>>>>>>> -gm
>>>>>>>>>
>>>>>>>>> On Wed, Jan 27, 2010 at 7:04 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>> I installed alsa-utile,
>>>>>>>>>>
>>>>>>>>>> now I get:
>>>>>>>>>>
>>>>>>>>>> skype:/var/cache/apt/archives# aplay -l
>>>>>>>>>> **** List of PLAYBACK Hardware Devices ****
>>>>>>>>>> card 0: Dummy [Dummy], device 0: Dummy PCM [Dummy PCM]
>>>>>>>>>> ?Subdevices: 127/128
>>>>>>>>>> ?Subdevice #0: subdevice #0
>>>>>>>>>> ?Subdevice #1: subdevice #1
>>>>>>>>>> ?Subdevice #2: subdevice #2
>>>>>>>>>> ?Subdevice #3: subdevice #3
>>>>>>>>>> ?Subdevice #4: subdevice #4
>>>>>>>>>> ?Subdevice #5: subdevice #5
>>>>>>>>>> ?Subdevice #6: subdevice #6
>>>>>>>>>> ?Subdevice #7: subdevice #7
>>>>>>>>>> ?Subdevice #8: subdevice #8
>>>>>>>>>> ?Subdevice #9: subdevice #9
>>>>>>>>>> ?Subdevice #10: subdevice #10
>>>>>>>>>> ?Subdevice #11: subdevice #11
>>>>>>>>>> ?Subdevice #12: subdevice #12
>>>>>>>>>> ?Subdevice #13: subdevice #13
>>>>>>>>>> ?Subdevice #14: subdevice #14
>>>>>>>>>> ?Subdevice #15: subdevice #15
>>>>>>>>>> ?Subdevice #16: subdevice #16
>>>>>>>>>> ?Subdevice #17: subdevice #17
>>>>>>>>>> ?Subdevice #18: subdevice #18
>>>>>>>>>> ?Subdevice #19: subdevice #19
>>>>>>>>>> ?Subdevice #20: subdevice #20
>>>>>>>>>> ?Subdevice #21: subdevice #21
>>>>>>>>>> ?Subdevice #22: subdevice #22
>>>>>>>>>> ?Subdevice #23: subdevice #23
>>>>>>>>>> ?Subdevice #24: subdevice #24
>>>>>>>>>> ?Subdevice #25: subdevice #25
>>>>>>>>>> ?Subdevice #26: subdevice #26
>>>>>>>>>> ?Subdevice #27: subdevice #27
>>>>>>>>>> ?Subdevice #28: subdevice #28
>>>>>>>>>> ?Subdevice #29: subdevice #29
>>>>>>>>>> ?Subdevice #30: subdevice #30
>>>>>>>>>> ?Subdevice #31: subdevice #31
>>>>>>>>>> ?Subdevice #32: subdevice #32
>>>>>>>>>> ?Subdevice #33: subdevice #33
>>>>>>>>>> ?Subdevice #34: subdevice #34
>>>>>>>>>> ?Subdevice #35: subdevice #35
>>>>>>>>>> ?Subdevice #36: subdevice #36
>>>>>>>>>> ?Subdevice #37: subdevice #37
>>>>>>>>>> ?Subdevice #38: subdevice #38
>>>>>>>>>> ?Subdevice #39: subdevice #39
>>>>>>>>>> ?Subdevice #40: subdevice #40
>>>>>>>>>> ?Subdevice #41: subdevice #41
>>>>>>>>>> ?Subdevice #42: subdevice #42
>>>>>>>>>> ?Subdevice #43: subdevice #43
>>>>>>>>>> ?Subdevice #44: subdevice #44
>>>>>>>>>> ?Subdevice #45: subdevice #45
>>>>>>>>>> ?Subdevice #46: subdevice #46
>>>>>>>>>> ?Subdevice #47: subdevice #47
>>>>>>>>>> ?Subdevice #48: subdevice #48
>>>>>>>>>> ?Subdevice #49: subdevice #49
>>>>>>>>>> ?Subdevice #50: subdevice #50
>>>>>>>>>> ?Subdevice #51: subdevice #51
>>>>>>>>>> ?Subdevice #52: subdevice #52
>>>>>>>>>> ?Subdevice #53: subdevice #53
>>>>>>>>>> ?Subdevice #54: subdevice #54
>>>>>>>>>> ?Subdevice #55: subdevice #55
>>>>>>>>>> ?Subdevice #56: subdevice #56
>>>>>>>>>> ?Subdevice #57: subdevice #57
>>>>>>>>>> ?Subdevice #58: subdevice #58
>>>>>>>>>> ?Subdevice #59: subdevice #59
>>>>>>>>>> ?Subdevice #60: subdevice #60
>>>>>>>>>> ?Subdevice #61: subdevice #61
>>>>>>>>>> ?Subdevice #62: subdevice #62
>>>>>>>>>> ?Subdevice #63: subdevice #63
>>>>>>>>>> ?Subdevice #64: subdevice #64
>>>>>>>>>> ?Subdevice #65: subdevice #65
>>>>>>>>>> ?Subdevice #66: subdevice #66
>>>>>>>>>> ?Subdevice #67: subdevice #67
>>>>>>>>>> ?Subdevice #68: subdevice #68
>>>>>>>>>> ?Subdevice #69: subdevice #69
>>>>>>>>>> ?Subdevice #70: subdevice #70
>>>>>>>>>> ?Subdevice #71: subdevice #71
>>>>>>>>>> ?Subdevice #72: subdevice #72
>>>>>>>>>> ?Subdevice #73: subdevice #73
>>>>>>>>>> ?Subdevice #74: subdevice #74
>>>>>>>>>> ?Subdevice #75: subdevice #75
>>>>>>>>>> ?Subdevice #76: subdevice #76
>>>>>>>>>> ?Subdevice #77: subdevice #77
>>>>>>>>>> ?Subdevice #78: subdevice #78
>>>>>>>>>> ?Subdevice #79: subdevice #79
>>>>>>>>>> ?Subdevice #80: subdevice #80
>>>>>>>>>> ?Subdevice #81: subdevice #81
>>>>>>>>>> ?Subdevice #82: subdevice #82
>>>>>>>>>> ?Subdevice #83: subdevice #83
>>>>>>>>>> ?Subdevice #84: subdevice #84
>>>>>>>>>> ?Subdevice #85: subdevice #85
>>>>>>>>>> ?Subdevice #86: subdevice #86
>>>>>>>>>> ?Subdevice #87: subdevice #87
>>>>>>>>>> ?Subdevice #88: subdevice #88
>>>>>>>>>> ?Subdevice #89: subdevice #89
>>>>>>>>>> ?Subdevice #90: subdevice #90
>>>>>>>>>> ?Subdevice #91: subdevice #91
>>>>>>>>>> ?Subdevice #92: subdevice #92
>>>>>>>>>> ?Subdevice #93: subdevice #93
>>>>>>>>>> ?Subdevice #94: subdevice #94
>>>>>>>>>> ?Subdevice #95: subdevice #95
>>>>>>>>>> ?Subdevice #96: subdevice #96
>>>>>>>>>> ?Subdevice #97: subdevice #97
>>>>>>>>>> ?Subdevice #98: subdevice #98
>>>>>>>>>> ?Subdevice #99: subdevice #99
>>>>>>>>>> ?Subdevice #100: subdevice #100
>>>>>>>>>> ?Subdevice #101: subdevice #101
>>>>>>>>>> ?Subdevice #102: subdevice #102
>>>>>>>>>> ?Subdevice #103: subdevice #103
>>>>>>>>>> ?Subdevice #104: subdevice #104
>>>>>>>>>> ?Subdevice #105: subdevice #105
>>>>>>>>>> ?Subdevice #106: subdevice #106
>>>>>>>>>> ?Subdevice #107: subdevice #107
>>>>>>>>>> ?Subdevice #108: subdevice #108
>>>>>>>>>> ?Subdevice #109: subdevice #109
>>>>>>>>>> ?Subdevice #110: subdevice #110
>>>>>>>>>> ?Subdevice #111: subdevice #111
>>>>>>>>>> ?Subdevice #112: subdevice #112
>>>>>>>>>> ?Subdevice #113: subdevice #113
>>>>>>>>>> ?Subdevice #114: subdevice #114
>>>>>>>>>> ?Subdevice #115: subdevice #115
>>>>>>>>>> ?Subdevice #116: subdevice #116
>>>>>>>>>> ?Subdevice #117: subdevice #117
>>>>>>>>>> ?Subdevice #118: subdevice #118
>>>>>>>>>> ?Subdevice #119: subdevice #119
>>>>>>>>>> ?Subdevice #120: subdevice #120
>>>>>>>>>> ?Subdevice #121: subdevice #121
>>>>>>>>>> ?Subdevice #122: subdevice #122
>>>>>>>>>> ?Subdevice #123: subdevice #123
>>>>>>>>>> ?Subdevice #124: subdevice #124
>>>>>>>>>> ?Subdevice #125: subdevice #125
>>>>>>>>>> ?Subdevice #126: subdevice #126
>>>>>>>>>> ?Subdevice #127: subdevice #127
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> Peter P GMX schrieb:
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>> Her's the output:
>>>>>>>>>>>
>>>>>>>>>>> skype:~# aplay -l
>>>>>>>>>>> bash: aplay: command not found
>>>>>>>>>>>
>>>>>>>>>>> Giovanni Maruzzelli schrieb:
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>> I don't think you got two snd-dummy loaded (but maybe yes)
>>>>>>>>>>>> what's the output of:
>>>>>>>>>>>>
>>>>>>>>>>>> aplay -l
>>>>>>>>>>>>
>>>>>>>>>>>> ?
>>>>>>>>>>>>
>>>>>>>>>>>> If instead you are referring to the choices that skype clients offers
>>>>>>>>>>>> you in the "set audio devices" window, choose Dummy PCM (hw0:0)
>>>>>>>>>>>>
>>>>>>>>>>>> Eg: not the "default", but the "hardware" one
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>> On Wed, Jan 27, 2010 at 5:58 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>> Thanks Giovanni,
>>>>>>>>>>>>>
>>>>>>>>>>>>> I think there may be the problem, that I have 2 sound devices now:
>>>>>>>>>>>>> - Dummy PCM (hw0:0) (this is from debian install)
>>>>>>>>>>>>> - Dummy PCM Default Audio device (defauzlt: CARD=Dummy) (this is new
>>>>>>>>>>>>> since I compiled alsa newly)
>>>>>>>>>>>>>
>>>>>>>>>>>>> I tried both, but both do not work. How do I get rid of the old alsa device?
>>>>>>>>>>>>> By the way: I uninstalled Alsa before I installed the new driver
>>>>>>>>>>>>> (apt-get remove alsa-utils alsa-base).
>>>>>>>>>>>>>
>>>>>>>>>>>>> Best regards
>>>>>>>>>>>>> Peter
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>> Giovanni Maruzzelli schrieb:
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>> This warning is harmless:
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> Jan 27 15:28:50 skype kernel: [ 3994.795786] process `skype' is using
>>>>>>>>>>>>>> obsolete setsockopt SO_BSDCOMPAT
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> On Wed, Jan 27, 2010 at 4:26 PM, Giovanni Maruzzelli
>>>>>>>>>>>>>> <gmaruzz at celliax.org> wrote:
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> Ciao Peter
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> one instance of snd-dummy "customized" is enough for 64 instances of
>>>>>>>>>>>>>>> skype clients, no need (and do not works) with more instances of
>>>>>>>>>>>>>>> snd-dummy-customized.
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> Maybe you got the one-way problem because of kernel at 250HZ (don't
>>>>>>>>>>>>>>> know). It uses to works well on a tickless kernel at 100HZ (eg: ubuntu
>>>>>>>>>>>>>>> 8.04).
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> Or maybe you have to check and modify which sound devices the skype
>>>>>>>>>>>>>>> clients are using (try to check that with snd-summy-custom loaded,
>>>>>>>>>>>>>>> maybe with the ssh -X trick (as in the wiki page).
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> To load more than one snd-dummy-original (the non modified one), you
>>>>>>>>>>>>>>> do this with the modprobe command, as in:
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> rmmod snd-dummy
>>>>>>>>>>>>>>> modprobe snd-dummy enable=1,1,1
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> this command will enable three instances of snd-dummy original, so
>>>>>>>>>>>>>>> you'll have three fake soundcards, and you'll have to setup each group
>>>>>>>>>>>>>>> of 8 skype instances to use sound devices from one fake soundcard, RG:
>>>>>>>>>>>>>>> no more than 8 skype client instances can use one instance of fake
>>>>>>>>>>>>>>> soundcard.
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> Also, please update the mod_skypiax code (svn up in its directory) I
>>>>>>>>>>>>>>> just committed some improvements.
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> If you have any other doubts, or need more info, don't hesitate to
>>>>>>>>>>>>>>> write the mailing list again,
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> ciao for now,
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> -giovanni
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> On Wed, Jan 27, 2010 at 4:01 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> I have mod_skypiax working nicely so far with 2 Skype channels. Thanks
>>>>>>>>>>>>>>>> to all contributors, excellent work!
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> In order to have more than 8 channels working, I have followed the
>>>>>>>>>>>>>>>> instructions in
>>>>>>>>>>>>>>>> http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk
>>>>>>>>>>>>>>>> and compiled alsa-driver-1.0.20 with the modified dummy.c file. (System
>>>>>>>>>>>>>>>> ist Debian 5.0R3)
>>>>>>>>>>>>>>>> It compiled well however when I start snd-dummy I only have
>>>>>>>>>>>>>>>> one-way-audio and my logs show
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> Jan 27 15:28:41 skype kernel: [ 3984.318403] snd-dummy skypiax driver,
>>>>>>>>>>>>>>>> /usr/src/alsa-driver-1.0.20/drivers/../alsa-kernel/drivers/dummy.c:920
>>>>>>>>>>>>>>>> working on a machine with 250HZ kernel
>>>>>>>>>>>>>>>> Jan 27 15:28:50 skype kernel: [ 3994.795786] process `skype' is using
>>>>>>>>>>>>>>>> obsolete setsockopt SO_BSDCOMPAT
>>>>>>>>>>>>>>>> Jan 27 15:28:56 skype kernel: [ 4005.289907] __ratelimit: 490 messages
>>>>>>>>>>>>>>>> suppressed
>>>>>>>>>>>>>>>> Jan 27 15:29:01 skype kernel: [ 4012.458310] process `skype' is using
>>>>>>>>>>>>>>>> obsolete setsockopt SO_BSDCOMPAT
>>>>>>>>>>>>>>>> Jan 27 15:29:01 skype kernel: [ 4013.326290] __ratelimit: 499 messages
>>>>>>>>>>>>>>>> suppressed
>>>>>>>>>>>>>>>> If I reinstall alsa from deb everything sworks fine again (of course
>>>>>>>>>>>>>>>> with the current limitations).
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> First question: Has anybody had this issue before? How can I solve this?
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> Second question:
>>>>>>>>>>>>>>>> As I do not need 64 channels or more: how do I manage, that Skype
>>>>>>>>>>>>>>>> instances 9..15 use a second instance of snd-dummy as addressed in the wiki?
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> Best regards
>>>>>>>>>>>>>>>> Peter
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> --
>>>>>>>>>>>>>>> Sincerely,
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> Giovanni Maruzzelli
>>>>>>>>>>>>>>> Cell : +39-347-2665618
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>> _______________________________________________
>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> --
>>> Ghulam Mustafa
>>> cell: +92 333.611.7681
>>> sip: cyrenity at ekiga.net
>>> mail: mustafa.pk at gmail.com
>>> web: cyrenity.wordpress.com
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618



From matt at webcontracts.co.uk  Fri Feb  5 05:53:49 2010
From: matt at webcontracts.co.uk (Matthew Law)
Date: Fri, 5 Feb 2010 13:53:49 -0000
Subject: [Freeswitch-users] Simple IAX2 setup - help with converting
 from asterisk to freeswitch
In-Reply-To: <E1E3F6EF-1A1E-4891-B371-7A3BF0988E95@jerris.com>
References: <91093d24f2166d220acf2ab19115f95d.squirrel@www.webcontracts.co.uk>
	<E1E3F6EF-1A1E-4891-B371-7A3BF0988E95@jerris.com>
Message-ID: <d2976552e9ba0ce1ed095b7cef22e1ad.squirrel@www.webcontracts.co.uk>

Why is that? - a lot of web pages I have read claim that IAX is more
secure and efficient.  I have no problem with using SIP whatsoever and it
certainly appears to be ubiquitous.  I am a complete newcomer to VoIP and
I am trying to do this as securely as possible since I want to run
freeswitch on a Xen VPS which will be visible on the internet.

Anyway, since my first question, I have worked my way through the wiki,
read a lot of example configs and made some sense of the docs.  I now have
a very basic config working (with SIP) on a local vmware image using the
'quick and dirty' Makefile method.  I removed all of the example configs
from the xml files (those default extensions and passwords scared me) and
added just one for extension 1000, plus my dialplan and inbound/outbound
settings.

One question: is there any reason not to use a smaller extension number
range, like 200-210, for example?

Now to figure out how to get time based roaming working...


Thanks,

Matt.


On Fri, February 5, 2010 6:43 am, Michael Jerris wrote:
> iax2 support has been removed from FreeSWITCH in current trunk and will
> not be in the 1.0.5 release.
>
> Mike




From Prometheus001 at gmx.net  Fri Feb  5 06:23:26 2010
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Fri, 05 Feb 2010 15:23:26 +0100
Subject: [Freeswitch-users] mod_skypiax - help needed with
 modified	snd-dummy (one way audio)
In-Reply-To: <7b197bef1002050536j7ea09684v913434296ae52b97@mail.gmail.com>
References: <4B60555B.2020004@gmx.net>	<7b197bef1001271215o64918f56s3d16e51528fc5f66@mail.gmail.com>
	<4B61ECE0.10409@gmx.net>	<7b197bef1001281210u43a907edi13afe794cf0e1a2e@mail.gmail.com>
	<4B61FCAB.5040707@gmx.net>	<7b197bef1001281341h7ee2c58fj79b3886630901f29@mail.gmail.com>
	<4B62C962.7000601@gmx.net>	<8213d6071001290403g2c08fe39gd92eb0446bb6485a@mail.gmail.com>
	<7b197bef1001290520x37179cb1o63273410e9ef5bc@mail.gmail.com>
	<4B6C1CCB.4080606@gmx.net>
	<7b197bef1002050536j7ea09684v913434296ae52b97@mail.gmail.com>
Message-ID: <4B6C29DE.3010107@gmx.net>

Hello Giovanni,

I will then try ubuntu 8.04 (hardy) LTS server 64bit now 
and report the results.

Best regards
Peter



Giovanni Maruzzelli schrieb:
> Ciao Peter,
>
> I would use ubuntu 8.04 (hardy) LTS server 64bit or CentOS 5.4 64bit.
>
> -giovanni
>
> On Fri, Feb 5, 2010 at 2:27 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>   
>> Hello Giovanni,
>>
>> as I couldn't even get skype again working again with the standard alsa
>> driver, I would like to setup the machine from scratch based on a
>> working machine.
>> The latest errors I received from Skype was:
>>  snd_pcm_avail_update() returned a value that is exceptionally large:
>> 715706624 bytes (3727638 ms).
>>  Most likely this is a bug in the ALSA driver. Please report this issue
>> to the ALSA developers.
>> I think that may be the reason for one-way-audio.
>>
>> For setting up my machine from scratch, please advise:
>> - which OS you are you using und recommending exactly?
>> - I would like to use 64bit OS in order to use 8GB of memory, does this
>> work?
>> - any other hints?
>>
>> Best regards
>> Peter
>>
>> Giovanni Maruzzelli schrieb:
>>     
>>> Peter,
>>>
>>> Can you connect on IRC (irc.freenode.net #freeswitch)? I'm gmaruzz there.
>>>
>>> Can you restate your problems? I've lost connection :)
>>>
>>> with snd-dummy custom you can create *one only* snd-dummy instance, so
>>> *one only* fake soundcard. If you create more, will not work. But with
>>> that one fake soundcard you can use 64 skype client instances, all
>>> with the same soundcard hardware device (hw:n).
>>>
>>> with original snd-dummy you can create a max of 8 instances, so 8 fake
>>> soundcards, and with each fake soundcard you can  use a max of 8 skype
>>> client instances.
>>>
>>> use the hardware devices, not the default devices (use the "hw:n")
>>>
>>> -giovanni
>>>
>>> On Fri, Jan 29, 2010 at 1:03 PM, Ghulam Mustafa <mustafa.pk at gmail.com> wrote:
>>>
>>>       
>>>> did you enable debug mode while compiling custom snd-dummy? if  yes
>>>> try re-compiling with debug mode disabled.
>>>>
>>>> -m
>>>>
>>>> On Fri, Jan 29, 2010 at 4:41 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>
>>>>         
>>>>> I now reinstalled the original sound drivers
>>>>> Unfortunaltely the sound problems remain, not that worse but they are there:
>>>>> Audio is still (almost) one way. Almost means:
>>>>>
>>>>>    * SIP -> Skype ok
>>>>>    * Skype=> SIP I hear only some scratching on very loud audio
>>>>>
>>>>> Could it be a volume problem? But snd-dummy should have no volume
>>>>> properties, right?
>>>>>
>>>>> Best regards
>>>>> Peter
>>>>>
>>>>>
>>>>> Giovanni Maruzzelli schrieb:
>>>>>
>>>>>           
>>>>>> with three instances you will assign the hw:0 device to skype client
>>>>>> 0...7, hw:1 to skype client 8...15 and hw:2 to skype client 16...23.
>>>>>> Must work. Pay attention to assign the same device name to all devices
>>>>>> needed by a skype instance (sound devices window): playback, capture
>>>>>> AND ring.
>>>>>>
>>>>>> Or maybe is a bug of ALSA on Debian...
>>>>>>
>>>>>> -giovanni
>>>>>>
>>>>>> On Thu, Jan 28, 2010 at 10:07 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>>
>>>>>>
>>>>>>             
>>>>>>> I crated 3 instances of snd-dummy, this worked. I assigned then Instance
>>>>>>> #2 to the Skype accounts. Still no sound.
>>>>>>> On the frist call there is one way audio, on the following calls there
>>>>>>> is no audio at all.
>>>>>>> This is weird.
>>>>>>>
>>>>>>> Best regards
>>>>>>> Peter
>>>>>>>
>>>>>>> Giovanni Maruzzelli schrieb:
>>>>>>>
>>>>>>>
>>>>>>>               
>>>>>>>> Ciao Peter,
>>>>>>>>
>>>>>>>> Never tested on Debian 5.
>>>>>>>>
>>>>>>>> When you write "same problem" you are referring to the audio going one
>>>>>>>> way only (btw, which way?) with the custom audio driver?
>>>>>>>>
>>>>>>>> Have you tried with multiple instances of the regular Debian
>>>>>>>> snd-dummy, as I wrote in a mail before?
>>>>>>>>
>>>>>>>> -gm
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> On Thu, Jan 28, 2010 at 9:00 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>                 
>>>>>>>>> Hello Giovanni,
>>>>>>>>>
>>>>>>>>> I did so but the same problem again.
>>>>>>>>>
>>>>>>>>> Did you ever test in on Debian 5.0?
>>>>>>>>>
>>>>>>>>> Best reards
>>>>>>>>> Peter
>>>>>>>>>
>>>>>>>>> Giovanni Maruzzelli schrieb:
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>                   
>>>>>>>>>> good, so you have only one sound device, the right one.
>>>>>>>>>>
>>>>>>>>>> Use the one with hw:0 in the window that skype gives you to set sound devices
>>>>>>>>>>
>>>>>>>>>> -gm
>>>>>>>>>>
>>>>>>>>>> On Wed, Jan 27, 2010 at 7:04 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>                     
>>>>>>>>>>> I installed alsa-utile,
>>>>>>>>>>>
>>>>>>>>>>> now I get:
>>>>>>>>>>>
>>>>>>>>>>> skype:/var/cache/apt/archives# aplay -l
>>>>>>>>>>> **** List of PLAYBACK Hardware Devices ****
>>>>>>>>>>> card 0: Dummy [Dummy], device 0: Dummy PCM [Dummy PCM]
>>>>>>>>>>>  Subdevices: 127/128
>>>>>>>>>>>  Subdevice #0: subdevice #0
>>>>>>>>>>>  Subdevice #1: subdevice #1
>>>>>>>>>>>  Subdevice #2: subdevice #2
>>>>>>>>>>>  Subdevice #3: subdevice #3
>>>>>>>>>>>  Subdevice #4: subdevice #4
>>>>>>>>>>>  Subdevice #5: subdevice #5
>>>>>>>>>>>  Subdevice #6: subdevice #6
>>>>>>>>>>>  Subdevice #7: subdevice #7
>>>>>>>>>>>  Subdevice #8: subdevice #8
>>>>>>>>>>>  Subdevice #9: subdevice #9
>>>>>>>>>>>  Subdevice #10: subdevice #10
>>>>>>>>>>>  Subdevice #11: subdevice #11
>>>>>>>>>>>  Subdevice #12: subdevice #12
>>>>>>>>>>>  Subdevice #13: subdevice #13
>>>>>>>>>>>  Subdevice #14: subdevice #14
>>>>>>>>>>>  Subdevice #15: subdevice #15
>>>>>>>>>>>  Subdevice #16: subdevice #16
>>>>>>>>>>>  Subdevice #17: subdevice #17
>>>>>>>>>>>  Subdevice #18: subdevice #18
>>>>>>>>>>>  Subdevice #19: subdevice #19
>>>>>>>>>>>  Subdevice #20: subdevice #20
>>>>>>>>>>>  Subdevice #21: subdevice #21
>>>>>>>>>>>  Subdevice #22: subdevice #22
>>>>>>>>>>>  Subdevice #23: subdevice #23
>>>>>>>>>>>  Subdevice #24: subdevice #24
>>>>>>>>>>>  Subdevice #25: subdevice #25
>>>>>>>>>>>  Subdevice #26: subdevice #26
>>>>>>>>>>>  Subdevice #27: subdevice #27
>>>>>>>>>>>  Subdevice #28: subdevice #28
>>>>>>>>>>>  Subdevice #29: subdevice #29
>>>>>>>>>>>  Subdevice #30: subdevice #30
>>>>>>>>>>>  Subdevice #31: subdevice #31
>>>>>>>>>>>  Subdevice #32: subdevice #32
>>>>>>>>>>>  Subdevice #33: subdevice #33
>>>>>>>>>>>  Subdevice #34: subdevice #34
>>>>>>>>>>>  Subdevice #35: subdevice #35
>>>>>>>>>>>  Subdevice #36: subdevice #36
>>>>>>>>>>>  Subdevice #37: subdevice #37
>>>>>>>>>>>  Subdevice #38: subdevice #38
>>>>>>>>>>>  Subdevice #39: subdevice #39
>>>>>>>>>>>  Subdevice #40: subdevice #40
>>>>>>>>>>>  Subdevice #41: subdevice #41
>>>>>>>>>>>  Subdevice #42: subdevice #42
>>>>>>>>>>>  Subdevice #43: subdevice #43
>>>>>>>>>>>  Subdevice #44: subdevice #44
>>>>>>>>>>>  Subdevice #45: subdevice #45
>>>>>>>>>>>  Subdevice #46: subdevice #46
>>>>>>>>>>>  Subdevice #47: subdevice #47
>>>>>>>>>>>  Subdevice #48: subdevice #48
>>>>>>>>>>>  Subdevice #49: subdevice #49
>>>>>>>>>>>  Subdevice #50: subdevice #50
>>>>>>>>>>>  Subdevice #51: subdevice #51
>>>>>>>>>>>  Subdevice #52: subdevice #52
>>>>>>>>>>>  Subdevice #53: subdevice #53
>>>>>>>>>>>  Subdevice #54: subdevice #54
>>>>>>>>>>>  Subdevice #55: subdevice #55
>>>>>>>>>>>  Subdevice #56: subdevice #56
>>>>>>>>>>>  Subdevice #57: subdevice #57
>>>>>>>>>>>  Subdevice #58: subdevice #58
>>>>>>>>>>>  Subdevice #59: subdevice #59
>>>>>>>>>>>  Subdevice #60: subdevice #60
>>>>>>>>>>>  Subdevice #61: subdevice #61
>>>>>>>>>>>  Subdevice #62: subdevice #62
>>>>>>>>>>>  Subdevice #63: subdevice #63
>>>>>>>>>>>  Subdevice #64: subdevice #64
>>>>>>>>>>>  Subdevice #65: subdevice #65
>>>>>>>>>>>  Subdevice #66: subdevice #66
>>>>>>>>>>>  Subdevice #67: subdevice #67
>>>>>>>>>>>  Subdevice #68: subdevice #68
>>>>>>>>>>>  Subdevice #69: subdevice #69
>>>>>>>>>>>  Subdevice #70: subdevice #70
>>>>>>>>>>>  Subdevice #71: subdevice #71
>>>>>>>>>>>  Subdevice #72: subdevice #72
>>>>>>>>>>>  Subdevice #73: subdevice #73
>>>>>>>>>>>  Subdevice #74: subdevice #74
>>>>>>>>>>>  Subdevice #75: subdevice #75
>>>>>>>>>>>  Subdevice #76: subdevice #76
>>>>>>>>>>>  Subdevice #77: subdevice #77
>>>>>>>>>>>  Subdevice #78: subdevice #78
>>>>>>>>>>>  Subdevice #79: subdevice #79
>>>>>>>>>>>  Subdevice #80: subdevice #80
>>>>>>>>>>>  Subdevice #81: subdevice #81
>>>>>>>>>>>  Subdevice #82: subdevice #82
>>>>>>>>>>>  Subdevice #83: subdevice #83
>>>>>>>>>>>  Subdevice #84: subdevice #84
>>>>>>>>>>>  Subdevice #85: subdevice #85
>>>>>>>>>>>  Subdevice #86: subdevice #86
>>>>>>>>>>>  Subdevice #87: subdevice #87
>>>>>>>>>>>  Subdevice #88: subdevice #88
>>>>>>>>>>>  Subdevice #89: subdevice #89
>>>>>>>>>>>  Subdevice #90: subdevice #90
>>>>>>>>>>>  Subdevice #91: subdevice #91
>>>>>>>>>>>  Subdevice #92: subdevice #92
>>>>>>>>>>>  Subdevice #93: subdevice #93
>>>>>>>>>>>  Subdevice #94: subdevice #94
>>>>>>>>>>>  Subdevice #95: subdevice #95
>>>>>>>>>>>  Subdevice #96: subdevice #96
>>>>>>>>>>>  Subdevice #97: subdevice #97
>>>>>>>>>>>  Subdevice #98: subdevice #98
>>>>>>>>>>>  Subdevice #99: subdevice #99
>>>>>>>>>>>  Subdevice #100: subdevice #100
>>>>>>>>>>>  Subdevice #101: subdevice #101
>>>>>>>>>>>  Subdevice #102: subdevice #102
>>>>>>>>>>>  Subdevice #103: subdevice #103
>>>>>>>>>>>  Subdevice #104: subdevice #104
>>>>>>>>>>>  Subdevice #105: subdevice #105
>>>>>>>>>>>  Subdevice #106: subdevice #106
>>>>>>>>>>>  Subdevice #107: subdevice #107
>>>>>>>>>>>  Subdevice #108: subdevice #108
>>>>>>>>>>>  Subdevice #109: subdevice #109
>>>>>>>>>>>  Subdevice #110: subdevice #110
>>>>>>>>>>>  Subdevice #111: subdevice #111
>>>>>>>>>>>  Subdevice #112: subdevice #112
>>>>>>>>>>>  Subdevice #113: subdevice #113
>>>>>>>>>>>  Subdevice #114: subdevice #114
>>>>>>>>>>>  Subdevice #115: subdevice #115
>>>>>>>>>>>  Subdevice #116: subdevice #116
>>>>>>>>>>>  Subdevice #117: subdevice #117
>>>>>>>>>>>  Subdevice #118: subdevice #118
>>>>>>>>>>>  Subdevice #119: subdevice #119
>>>>>>>>>>>  Subdevice #120: subdevice #120
>>>>>>>>>>>  Subdevice #121: subdevice #121
>>>>>>>>>>>  Subdevice #122: subdevice #122
>>>>>>>>>>>  Subdevice #123: subdevice #123
>>>>>>>>>>>  Subdevice #124: subdevice #124
>>>>>>>>>>>  Subdevice #125: subdevice #125
>>>>>>>>>>>  Subdevice #126: subdevice #126
>>>>>>>>>>>  Subdevice #127: subdevice #127
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> Peter P GMX schrieb:
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>                       
>>>>>>>>>>>> Her's the output:
>>>>>>>>>>>>
>>>>>>>>>>>> skype:~# aplay -l
>>>>>>>>>>>> bash: aplay: command not found
>>>>>>>>>>>>
>>>>>>>>>>>> Giovanni Maruzzelli schrieb:
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>                         
>>>>>>>>>>>>> I don't think you got two snd-dummy loaded (but maybe yes)
>>>>>>>>>>>>> what's the output of:
>>>>>>>>>>>>>
>>>>>>>>>>>>> aplay -l
>>>>>>>>>>>>>
>>>>>>>>>>>>> ?
>>>>>>>>>>>>>
>>>>>>>>>>>>> If instead you are referring to the choices that skype clients offers
>>>>>>>>>>>>> you in the "set audio devices" window, choose Dummy PCM (hw0:0)
>>>>>>>>>>>>>
>>>>>>>>>>>>> Eg: not the "default", but the "hardware" one
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>> On Wed, Jan 27, 2010 at 5:58 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>                           
>>>>>>>>>>>>>> Thanks Giovanni,
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> I think there may be the problem, that I have 2 sound devices now:
>>>>>>>>>>>>>> - Dummy PCM (hw0:0) (this is from debian install)
>>>>>>>>>>>>>> - Dummy PCM Default Audio device (defauzlt: CARD=Dummy) (this is new
>>>>>>>>>>>>>> since I compiled alsa newly)
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> I tried both, but both do not work. How do I get rid of the old alsa device?
>>>>>>>>>>>>>> By the way: I uninstalled Alsa before I installed the new driver
>>>>>>>>>>>>>> (apt-get remove alsa-utils alsa-base).
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> Best regards
>>>>>>>>>>>>>> Peter
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> Giovanni Maruzzelli schrieb:
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>                             
>>>>>>>>>>>>>>> This warning is harmless:
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> Jan 27 15:28:50 skype kernel: [ 3994.795786] process `skype' is using
>>>>>>>>>>>>>>> obsolete setsockopt SO_BSDCOMPAT
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> On Wed, Jan 27, 2010 at 4:26 PM, Giovanni Maruzzelli
>>>>>>>>>>>>>>> <gmaruzz at celliax.org> wrote:
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>                               
>>>>>>>>>>>>>>>> Ciao Peter
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> one instance of snd-dummy "customized" is enough for 64 instances of
>>>>>>>>>>>>>>>> skype clients, no need (and do not works) with more instances of
>>>>>>>>>>>>>>>> snd-dummy-customized.
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> Maybe you got the one-way problem because of kernel at 250HZ (don't
>>>>>>>>>>>>>>>> know). It uses to works well on a tickless kernel at 100HZ (eg: ubuntu
>>>>>>>>>>>>>>>> 8.04).
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> Or maybe you have to check and modify which sound devices the skype
>>>>>>>>>>>>>>>> clients are using (try to check that with snd-summy-custom loaded,
>>>>>>>>>>>>>>>> maybe with the ssh -X trick (as in the wiki page).
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> To load more than one snd-dummy-original (the non modified one), you
>>>>>>>>>>>>>>>> do this with the modprobe command, as in:
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> rmmod snd-dummy
>>>>>>>>>>>>>>>> modprobe snd-dummy enable=1,1,1
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> this command will enable three instances of snd-dummy original, so
>>>>>>>>>>>>>>>> you'll have three fake soundcards, and you'll have to setup each group
>>>>>>>>>>>>>>>> of 8 skype instances to use sound devices from one fake soundcard, RG:
>>>>>>>>>>>>>>>> no more than 8 skype client instances can use one instance of fake
>>>>>>>>>>>>>>>> soundcard.
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> Also, please update the mod_skypiax code (svn up in its directory) I
>>>>>>>>>>>>>>>> just committed some improvements.
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> If you have any other doubts, or need more info, don't hesitate to
>>>>>>>>>>>>>>>> write the mailing list again,
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> ciao for now,
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> -giovanni
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> On Wed, Jan 27, 2010 at 4:01 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>                                 
>>>>>>>>>>>>>>>>> I have mod_skypiax working nicely so far with 2 Skype channels. Thanks
>>>>>>>>>>>>>>>>> to all contributors, excellent work!
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> In order to have more than 8 channels working, I have followed the
>>>>>>>>>>>>>>>>> instructions in
>>>>>>>>>>>>>>>>> http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk
>>>>>>>>>>>>>>>>> and compiled alsa-driver-1.0.20 with the modified dummy.c file. (System
>>>>>>>>>>>>>>>>> ist Debian 5.0R3)
>>>>>>>>>>>>>>>>> It compiled well however when I start snd-dummy I only have
>>>>>>>>>>>>>>>>> one-way-audio and my logs show
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> Jan 27 15:28:41 skype kernel: [ 3984.318403] snd-dummy skypiax driver,
>>>>>>>>>>>>>>>>> /usr/src/alsa-driver-1.0.20/drivers/../alsa-kernel/drivers/dummy.c:920
>>>>>>>>>>>>>>>>> working on a machine with 250HZ kernel
>>>>>>>>>>>>>>>>> Jan 27 15:28:50 skype kernel: [ 3994.795786] process `skype' is using
>>>>>>>>>>>>>>>>> obsolete setsockopt SO_BSDCOMPAT
>>>>>>>>>>>>>>>>> Jan 27 15:28:56 skype kernel: [ 4005.289907] __ratelimit: 490 messages
>>>>>>>>>>>>>>>>> suppressed
>>>>>>>>>>>>>>>>> Jan 27 15:29:01 skype kernel: [ 4012.458310] process `skype' is using
>>>>>>>>>>>>>>>>> obsolete setsockopt SO_BSDCOMPAT
>>>>>>>>>>>>>>>>> Jan 27 15:29:01 skype kernel: [ 4013.326290] __ratelimit: 499 messages
>>>>>>>>>>>>>>>>> suppressed
>>>>>>>>>>>>>>>>> If I reinstall alsa from deb everything sworks fine again (of course
>>>>>>>>>>>>>>>>> with the current limitations).
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> First question: Has anybody had this issue before? How can I solve this?
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> Second question:
>>>>>>>>>>>>>>>>> As I do not need 64 channels or more: how do I manage, that Skype
>>>>>>>>>>>>>>>>> instances 9..15 use a second instance of snd-dummy as addressed in the wiki?
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> Best regards
>>>>>>>>>>>>>>>>> Peter
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>                                   
>>>>>>>>>>>>>>>> --
>>>>>>>>>>>>>>>> Sincerely,
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> Giovanni Maruzzelli
>>>>>>>>>>>>>>>> Cell : +39-347-2665618
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>                                 
>>>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>                             
>>>>>>>>>>> _______________________________________________
>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>                       
>>>>>>>>> _______________________________________________
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>                   
>>>>>>>>                 
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>               
>>>>>>
>>>>>>             
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>>           
>>>> --
>>>> Ghulam Mustafa
>>>> cell: +92 333.611.7681
>>>> sip: cyrenity at ekiga.net
>>>> mail: mustafa.pk at gmail.com
>>>> web: cyrenity.wordpress.com
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>         
>>>
>>>
>>>       
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>     
>
>
>
>   



From rupa at rupa.com  Fri Feb  5 06:57:18 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Fri, 5 Feb 2010 08:57:18 -0600
Subject: [Freeswitch-users] Simple IAX2 setup - help with converting
	from asterisk to freeswitch
In-Reply-To: <d2976552e9ba0ce1ed095b7cef22e1ad.squirrel@www.webcontracts.co.uk>
References: <91093d24f2166d220acf2ab19115f95d.squirrel@www.webcontracts.co.uk>
	<E1E3F6EF-1A1E-4891-B371-7A3BF0988E95@jerris.com>
	<d2976552e9ba0ce1ed095b7cef22e1ad.squirrel@www.webcontracts.co.uk>
Message-ID: <b39273531002050657x4f0f921euce8a683b8018ed63@mail.gmail.com>

the lib that we used to provide iax support is pretty much abandonware (no
longer updated) and newer iax implementations (like latest asterisk) can
cause it to crash.  There are no license compatible iax implementations that
work, so.. mod_iax has been moved to the unsupported column.

Default passwords -- that is a single var in vars.xml that controls the
passwords.

number ranges - up to you.  The sample configs supplied are just that,
samples.  I use a smaller range personally.

On Fri, Feb 5, 2010 at 7:53 AM, Matthew Law <matt at webcontracts.co.uk> wrote:

> Why is that? - a lot of web pages I have read claim that IAX is more
> secure and efficient.  I have no problem with using SIP whatsoever and it
> certainly appears to be ubiquitous.  I am a complete newcomer to VoIP and
> I am trying to do this as securely as possible since I want to run
> freeswitch on a Xen VPS which will be visible on the internet.
>
> Anyway, since my first question, I have worked my way through the wiki,
> read a lot of example configs and made some sense of the docs.  I now have
> a very basic config working (with SIP) on a local vmware image using the
> 'quick and dirty' Makefile method.  I removed all of the example configs
> from the xml files (those default extensions and passwords scared me) and
> added just one for extension 1000, plus my dialplan and inbound/outbound
> settings.
>
> One question: is there any reason not to use a smaller extension number
> range, like 200-210, for example?
>
> Now to figure out how to get time based roaming working...
>
>
> Thanks,
>
> Matt.
>
>
> On Fri, February 5, 2010 6:43 am, Michael Jerris wrote:
> > iax2 support has been removed from FreeSWITCH in current trunk and will
> > not be in the 1.0.5 release.
> >
> > Mike
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
-Rupa
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100205/61672bcc/attachment-0002.html 

From brian at freeswitch.org  Fri Feb  5 07:09:39 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 5 Feb 2010 09:09:39 -0600
Subject: [Freeswitch-users] No reINVITE when bridging two sessions from
	JavaScript with bypass_media_after_bridge=true
In-Reply-To: <8e9d67561002050018q24e3a806kc8807b55268e759b@mail.gmail.com>
References: <8e9d67561002041147q6c135546u1f7e1887529a916d@mail.gmail.com>
	<3D739D7F-F5C6-4266-B04E-9413DCFDA6AF@freeswitch.org>
	<8e9d67561002050018q24e3a806kc8807b55268e759b@mail.gmail.com>
Message-ID: <B66289FF-C2A9-4AC4-A56C-66E51B4726F4@freeswitch.org>

set it inside each of the {} for each session you create its not set after the fact the call is up already...  you're setting it too late.

you an also issue uuid_media off <uuid>

/b

On Feb 5, 2010, at 2:18 AM, Timur Valishev wrote:

> I've modified my script to make sure: <<<<<<<<<<<<<<<<<<<<<<<<<<<<,
> session = new Session(
> "{ignore_early_media=true,hangup_after_bridge=true}sofia/external/timwork at novion.ru"
> );
> session2 = new Session(
> "{ignore_early_media=true}sofia/external/timwork at novion.ru"
> );
> session.setVariable('bypass_media', 'true');
> session2.setVariable('bypass_media', 'true');
> bridge(session, session2);

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100205/48a0ce6f/attachment-0002.html 

From john at acsol.net  Fri Feb  5 07:11:26 2010
From: john at acsol.net (John)
Date: Fri, 05 Feb 2010 08:11:26 -0700
Subject: [Freeswitch-users] Switch Security
Message-ID: <4B6C351E.6080608@acsol.net>

Freeswitch is to be used by phones external to my lan. Many of the 
phones will be coming from DSL connections without static IP. I have 
disabled the default password. What is a good procedure to secure the 
switch from non-customers registering? I know that I could use an ACL; 
however it's difficult with all the non-static users.

Thanks John



From brian at freeswitch.org  Fri Feb  5 07:13:29 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 5 Feb 2010 09:13:29 -0600
Subject: [Freeswitch-users] Looking for some good/cheap desktop phones
In-Reply-To: <65d96fc81002050521w746707b1q4398e6d8a0692d1e@mail.gmail.com>
References: <3f2e071b5aea5e11bb37a37e435a33b7@mail.gmail.com>
	<8e9d67561002042236s434e0643od322551de8720be1@mail.gmail.com>
	<409D0ACF-ADA4-4D02-8B7E-71EC1B1BAFBC@freeswitch.org>
	<8e9d67561002042301v76129abfj3dcbc70ed073752c@mail.gmail.com>
	<d361ca7b1002050507n5e564621oa6cf7ace005dd5b8@mail.gmail.com>
	<65d96fc81002050521w746707b1q4398e6d8a0692d1e@mail.gmail.com>
Message-ID: <6B520D1B-EDD6-44F7-9DE5-A384D3286138@freeswitch.org>

Sigh... When is someone actually going to build an open platform voip hardware phone... Its just a linux box that happens to be shaped like a phone, with a touch screen, 48kHz sound card... and possibly video too.

/b
PS: Most of those eastern made phones are crap. 

On Feb 5, 2010, at 7:21 AM, Tihomir Culjaga wrote:

> Atcom AT-620 (http://www.atcom.cn/AT620.html#~tab-small_midsized_large_enterprises) is quite ok and cheap (~30$)... also we have been talking to Atcom to add a sort of auto-provissioning (dhcp/http) and this is going to happen next week.
> 
> 
> T.

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100205/b1a26745/attachment-0002.html 

From brian at freeswitch.org  Fri Feb  5 07:28:42 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 5 Feb 2010 09:28:42 -0600
Subject: [Freeswitch-users] Switch Security
In-Reply-To: <4B6C351E.6080608@acsol.net>
References: <4B6C351E.6080608@acsol.net>
Message-ID: <2ED7083F-F294-40F8-B46C-1D2E0E46509F@freeswitch.org>

Give them passwords... install fail2ban... http://wiki.freeswitch.org/wiki/Fail2ban

/b

On Feb 5, 2010, at 9:11 AM, John wrote:

> Freeswitch is to be used by phones external to my lan. Many of the 
> phones will be coming from DSL connections without static IP. I have 
> disabled the default password. What is a good procedure to secure the 
> switch from non-customers registering? I know that I could use an ACL; 
> however it's difficult with all the non-static users.
> 
> Thanks John




From dave at 3c.co.uk  Fri Feb  5 07:31:11 2010
From: dave at 3c.co.uk (David Knell)
Date: Fri, 05 Feb 2010 08:31:11 -0700
Subject: [Freeswitch-users] Looking for some good/cheap desktop phones
In-Reply-To: <6B520D1B-EDD6-44F7-9DE5-A384D3286138@freeswitch.org>
References: <3f2e071b5aea5e11bb37a37e435a33b7@mail.gmail.com>
	<8e9d67561002042236s434e0643od322551de8720be1@mail.gmail.com>
	<409D0ACF-ADA4-4D02-8B7E-71EC1B1BAFBC@freeswitch.org>
	<8e9d67561002042301v76129abfj3dcbc70ed073752c@mail.gmail.com>
	<d361ca7b1002050507n5e564621oa6cf7ace005dd5b8@mail.gmail.com>
	<65d96fc81002050521w746707b1q4398e6d8a0692d1e@mail.gmail.com>
	<6B520D1B-EDD6-44F7-9DE5-A384D3286138@freeswitch.org>
Message-ID: <1265383871.12871.71.camel@local.freepabx.com>

Hi Brian,

This is a start:
http://www.digitmat.com/ - you need to follow some links, but it's open
source.

--Dave

> Sigh... When is someone actually going to build an open platform voip
> hardware phone... Its just a linux box that happens to be shaped like
> a phone, with a touch screen, 48kHz sound card... and possibly video
> too.
> 
> 
> /b
> PS: Most of those eastern made phones are crap. 
> 
> On Feb 5, 2010, at 7:21 AM, Tihomir Culjaga wrote:
> 
> > Atcom AT-620
> > (http://www.atcom.cn/AT620.html#~tab-small_midsized_large_enterprises) is quite ok and cheap (~30$)... also we have been talking to Atcom to add a sort of auto-provissioning (dhcp/http) and this is going to happen next week.
> > 
> > 
> > T.
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From mike at jerris.com  Fri Feb  5 07:54:09 2010
From: mike at jerris.com (Michael Jerris)
Date: Fri, 5 Feb 2010 10:54:09 -0500
Subject: [Freeswitch-users] Simple IAX2 setup - help with converting
	from asterisk to freeswitch
In-Reply-To: <d2976552e9ba0ce1ed095b7cef22e1ad.squirrel@www.webcontracts.co.uk>
References: <91093d24f2166d220acf2ab19115f95d.squirrel@www.webcontracts.co.uk>
	<E1E3F6EF-1A1E-4891-B371-7A3BF0988E95@jerris.com>
	<d2976552e9ba0ce1ed095b7cef22e1ad.squirrel@www.webcontracts.co.uk>
Message-ID: <4D6421C5-9336-40D2-B54C-F773B2E6BA0E@jerris.com>

I find the secure and efficiency claims on IAX to be pretty much a farce.  IAX offers less overhead on rtp, but the same thing can be accomplished with rtp using packetization, as for security, I don't see any credible claim on that.  IAX also forces the program to sort out a ton of audio for different users going to 1 socket, something that a network stack is quite good at when using different ports, but is a lot more work where we are getting the packets.  As for default passwords and users, of course I wouldn't use those in production, those are for you to see how the pieces work together out of the box.  I wouldn't however quickly scrap the entire default config, just read through them and think about what you need and do not.  The extension ranges you use is totally at your discretion.

Mike


On Feb 5, 2010, at 8:53 AM, Matthew Law wrote:

> Why is that? - a lot of web pages I have read claim that IAX is more
> secure and efficient.  I have no problem with using SIP whatsoever and it
> certainly appears to be ubiquitous.  I am a complete newcomer to VoIP and
> I am trying to do this as securely as possible since I want to run
> freeswitch on a Xen VPS which will be visible on the internet.
> 
> Anyway, since my first question, I have worked my way through the wiki,
> read a lot of example configs and made some sense of the docs.  I now have
> a very basic config working (with SIP) on a local vmware image using the
> 'quick and dirty' Makefile method.  I removed all of the example configs
> from the xml files (those default extensions and passwords scared me) and
> added just one for extension 1000, plus my dialplan and inbound/outbound
> settings.
> 
> One question: is there any reason not to use a smaller extension number
> range, like 200-210, for example?
> 
> Now to figure out how to get time based roaming working?




From dave at 3c.co.uk  Fri Feb  5 08:03:54 2010
From: dave at 3c.co.uk (David Knell)
Date: Fri, 05 Feb 2010 09:03:54 -0700
Subject: [Freeswitch-users] Simple IAX2 setup - help with
	converting	from asterisk to freeswitch
In-Reply-To: <b39273531002050657x4f0f921euce8a683b8018ed63@mail.gmail.com>
References: <91093d24f2166d220acf2ab19115f95d.squirrel@www.webcontracts.co.uk>
	<E1E3F6EF-1A1E-4891-B371-7A3BF0988E95@jerris.com>
	<d2976552e9ba0ce1ed095b7cef22e1ad.squirrel@www.webcontracts.co.uk>
	<b39273531002050657x4f0f921euce8a683b8018ed63@mail.gmail.com>
Message-ID: <1265385834.12871.83.camel@local.freepabx.com>

There's a fairly simple solution to IAX needs, which is to run Asterisk,
probably on the same box, as a protocol converter - you just need to
tell it to use a non-standard port in sip.conf so that it doesn't clash
with FreeSWITCH.

--Dave

> the lib that we used to provide iax support is pretty much abandonware
> (no longer updated) and newer iax implementations (like latest
> asterisk) can cause it to crash.  There are no license compatible iax
> implementations that work, so.. mod_iax has been moved to the
> unsupported column.
> 
> 
> Default passwords -- that is a single var in vars.xml that controls
> the passwords.
> 
> 
> number ranges - up to you.  The sample configs supplied are just that,
> samples.  I use a smaller range personally.
> 
> On Fri, Feb 5, 2010 at 7:53 AM, Matthew Law <matt at webcontracts.co.uk>
> wrote:
>         Why is that? - a lot of web pages I have read claim that IAX
>         is more
>         secure and efficient.  I have no problem with using SIP
>         whatsoever and it
>         certainly appears to be ubiquitous.  I am a complete newcomer
>         to VoIP and
>         I am trying to do this as securely as possible since I want to
>         run
>         freeswitch on a Xen VPS which will be visible on the internet.
>         
>         Anyway, since my first question, I have worked my way through
>         the wiki,
>         read a lot of example configs and made some sense of the
>         docs.  I now have
>         a very basic config working (with SIP) on a local vmware image
>         using the
>         'quick and dirty' Makefile method.  I removed all of the
>         example configs
>         from the xml files (those default extensions and passwords
>         scared me) and
>         added just one for extension 1000, plus my dialplan and
>         inbound/outbound
>         settings.
>         
>         One question: is there any reason not to use a smaller
>         extension number
>         range, like 200-210, for example?
>         
>         Now to figure out how to get time based roaming working...
>         
>         
>         Thanks,
>         
>         Matt.
>         
>         
>         On Fri, February 5, 2010 6:43 am, Michael Jerris wrote:
>         > iax2 support has been removed from FreeSWITCH in current
>         trunk and will
>         > not be in the 1.0.5 release.
>         >
>         
>         
>         > Mike
>         
>         
>         _______________________________________________
>         FreeSWITCH-users mailing list
>         FreeSWITCH-users at lists.freeswitch.org
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         http://www.freeswitch.org
>         
> 
> 
> 
> -- 
> -Rupa
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From rupa at rupa.com  Fri Feb  5 08:18:24 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Fri, 5 Feb 2010 10:18:24 -0600
Subject: [Freeswitch-users] Switch Security
In-Reply-To: <2ED7083F-F294-40F8-B46C-1D2E0E46509F@freeswitch.org>
References: <4B6C351E.6080608@acsol.net>
	<2ED7083F-F294-40F8-B46C-1D2E0E46509F@freeswitch.org>
Message-ID: <b39273531002050818m39b2b45er762ffc491b0e80cd@mail.gmail.com>

That wiki needs to actually have the fail2ban rules for freeswitch
documented....

hmm... ok, I finished up the documentation with what is needed to actually
configure/verify fail2ban.

On Fri, Feb 5, 2010 at 9:28 AM, Brian West <brian at freeswitch.org> wrote:

> Give them passwords... install fail2ban...
> http://wiki.freeswitch.org/wiki/Fail2ban
>
> /b
>
> On Feb 5, 2010, at 9:11 AM, John wrote:
>
> > Freeswitch is to be used by phones external to my lan. Many of the
> > phones will be coming from DSL connections without static IP. I have
> > disabled the default password. What is a good procedure to secure the
> > switch from non-customers registering? I know that I could use an ACL;
> > however it's difficult with all the non-static users.
> >
> > Thanks John
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
-Rupa
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100205/f6f118e9/attachment-0002.html 

From jmesquita at freeswitch.org  Fri Feb  5 08:23:02 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Fri, 5 Feb 2010 14:23:02 -0200
Subject: [Freeswitch-users] Switch Security
In-Reply-To: <b39273531002050818m39b2b45er762ffc491b0e80cd@mail.gmail.com>
References: <4B6C351E.6080608@acsol.net>
	<2ED7083F-F294-40F8-B46C-1D2E0E46509F@freeswitch.org>
	<b39273531002050818m39b2b45er762ffc491b0e80cd@mail.gmail.com>
Message-ID: <ebbc2d551002050823s39f17451vd21cbe46a9509a19@mail.gmail.com>

Rupa, like usual, thank you.

Regards,
Jo?o Mesquita


On Fri, Feb 5, 2010 at 2:18 PM, Rupa Schomaker <rupa at rupa.com> wrote:

> That wiki needs to actually have the fail2ban rules for freeswitch
> documented....
>
> hmm... ok, I finished up the documentation with what is needed to actually
> configure/verify fail2ban.
>
>
> On Fri, Feb 5, 2010 at 9:28 AM, Brian West <brian at freeswitch.org> wrote:
>
>> Give them passwords... install fail2ban...
>> http://wiki.freeswitch.org/wiki/Fail2ban
>>
>> /b
>>
>> On Feb 5, 2010, at 9:11 AM, John wrote:
>>
>> > Freeswitch is to be used by phones external to my lan. Many of the
>> > phones will be coming from DSL connections without static IP. I have
>> > disabled the default password. What is a good procedure to secure the
>> > switch from non-customers registering? I know that I could use an ACL;
>> > however it's difficult with all the non-static users.
>> >
>> > Thanks John
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100205/eee5c222/attachment-0002.html 

From tculjaga at gmail.com  Fri Feb  5 08:42:06 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Fri, 5 Feb 2010 17:42:06 +0100
Subject: [Freeswitch-users] Looking for some good/cheap desktop phones
In-Reply-To: <1265383871.12871.71.camel@local.freepabx.com>
References: <3f2e071b5aea5e11bb37a37e435a33b7@mail.gmail.com>
	<8e9d67561002042236s434e0643od322551de8720be1@mail.gmail.com>
	<409D0ACF-ADA4-4D02-8B7E-71EC1B1BAFBC@freeswitch.org>
	<8e9d67561002042301v76129abfj3dcbc70ed073752c@mail.gmail.com>
	<d361ca7b1002050507n5e564621oa6cf7ace005dd5b8@mail.gmail.com>
	<65d96fc81002050521w746707b1q4398e6d8a0692d1e@mail.gmail.com>
	<6B520D1B-EDD6-44F7-9DE5-A384D3286138@freeswitch.org>
	<1265383871.12871.71.camel@local.freepabx.com>
Message-ID: <65d96fc81002050842l544d012eg2b3d43aba1e0d8dc@mail.gmail.com>

impressive!

This really looks nice: http://www.digitmat.com/res.html

i'm tempted to five it a shot :)


T.

On Fri, Feb 5, 2010 at 4:31 PM, David Knell <dave at 3c.co.uk> wrote:

> Hi Brian,
>
> This is a start:
> http://www.digitmat.com/ - you need to follow some links, but it's open
> source.
>
> --Dave
>
> > Sigh... When is someone actually going to build an open platform voip
> > hardware phone... Its just a linux box that happens to be shaped like
> > a phone, with a touch screen, 48kHz sound card... and possibly video
> > too.
> >
> >
> > /b
> > PS: Most of those eastern made phones are crap.
> >
> > On Feb 5, 2010, at 7:21 AM, Tihomir Culjaga wrote:
> >
> > > Atcom AT-620
> > > (http://www.atcom.cn/AT620.html#~tab-small_midsized_large_enterprises<http://www.atcom.cn/AT620.html#%7Etab-small_midsized_large_enterprises>)
> is quite ok and cheap (~30$)... also we have been talking to Atcom to add a
> sort of auto-provissioning (dhcp/http) and this is going to happen next
> week.
> > >
> > >
> > > T.
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100205/83717744/attachment-0002.html 

From msc at freeswitch.org  Fri Feb  5 08:45:04 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 5 Feb 2010 08:45:04 -0800
Subject: [Freeswitch-users] FreeSWITCH Conference Call Starting Shortly!
Message-ID: <87f2f3b91002050845w70bb52s434dff55c11fec92@mail.gmail.com>

Come join us today!

http://wiki.freeswitch.org/wiki/FS_weekly_2010_02_5

-Michael
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100205/74f048cf/attachment-0002.html 

From jerry.richards at teotech.com  Fri Feb  5 09:16:21 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Fri, 5 Feb 2010 09:16:21 -0800
Subject: [Freeswitch-users] Can Freeswitch and OpenSER Co-reside on Same
	Server?
Message-ID: <C4DC856C75094358AEDBE3FA0E242741@greyhawk.tonecommander.com>

If I use OpenSER for a session border controller, does anyone see an issue
if it resides on the same server as Freeswitch?  So I would have a LAN and
WAN socket?  Are there any drawbacks (other than loading) to worry about?

Thanks And Best Regards,
Jerry




From steveu at coppice.org  Fri Feb  5 09:44:33 2010
From: steveu at coppice.org (Steve Underwood)
Date: Sat, 06 Feb 2010 01:44:33 +0800
Subject: [Freeswitch-users] Looking for some good/cheap desktop phones
In-Reply-To: <1265383871.12871.71.camel@local.freepabx.com>
References: <3f2e071b5aea5e11bb37a37e435a33b7@mail.gmail.com>	<8e9d67561002042236s434e0643od322551de8720be1@mail.gmail.com>	<409D0ACF-ADA4-4D02-8B7E-71EC1B1BAFBC@freeswitch.org>	<8e9d67561002042301v76129abfj3dcbc70ed073752c@mail.gmail.com>	<d361ca7b1002050507n5e564621oa6cf7ace005dd5b8@mail.gmail.com>	<65d96fc81002050521w746707b1q4398e6d8a0692d1e@mail.gmail.com>	<6B520D1B-EDD6-44F7-9DE5-A384D3286138@freeswitch.org>
	<1265383871.12871.71.camel@local.freepabx.com>
Message-ID: <4B6C5901.1010509@coppice.org>

On 02/05/2010 11:31 PM, David Knell wrote:
> Hi Brian,
>
> This is a start:
> http://www.digitmat.com/ - you need to follow some links, but it's open
> source.
>
> --Dave
>    
Those people have a troubled history. They were the people behind many 
of the early cheap, but less than stellar, VoIP phones, with a chip that 
isn't made any more - the PA1688. They seem to have regrouped, and have 
a newer chip. I wonder if they have got their act together this time.
>> Sigh... When is someone actually going to build an open platform voip
>> hardware phone... Its just a linux box that happens to be shaped like
>> a phone, with a touch screen, 48kHz sound card... and possibly video
>> too.
>>
>>
>> /b
>> PS: Most of those eastern made phones are crap.
>>
>> On Feb 5, 2010, at 7:21 AM, Tihomir Culjaga wrote:
>>
>>      
>>> Atcom AT-620
>>> (http://www.atcom.cn/AT620.html#~tab-small_midsized_large_enterprises) is quite ok and cheap (~30$)... also we have been talking to Atcom to add a sort of auto-provissioning (dhcp/http) and this is going to happen next week.
>>>
>>>
>>> T.
>>>        
>>
ATCOM are amenable to the idea of supplying hardware for other people to 
put their own software on. Some of their phones use Infineon chip sets, 
and I think those are running software based on the Infineon Linux 
reference platform. All it takes is a few good people with the 
commitment to actually do something. I expect there are other Chinese 
makers who would be delighted to supply bare hardware to people, and a 
lot of the cheaper current Chinese phones have very similar Infineon 
based designs.

Steve




From joel.sisko at iconverged.com  Fri Feb  5 10:08:12 2010
From: joel.sisko at iconverged.com (joel.sisko at iconverged.com)
Date: Fri, 5 Feb 2010 12:08:12 -0600 (CST)
Subject: [Freeswitch-users] Use of mod_Conference
In-Reply-To: <472380364.66881265393034269.JavaMail.root@mail-2.01.com>
Message-ID: <648878714.68411265393292092.JavaMail.root@mail-2.01.com>

Group, 

I have a simple question I think about mod_Conference, does each conference room have to be created dynamically? My question is born from how Asterisk MeetMe room can work where you can have all the meetme rooms programmed statically in the meetme.conf file. 

What I am looking to do is rather than using the Asterisk meetme room application, I want to use FreeSwitch to provide this functionality. So I am trying to figure out what the integration would look like from a 10,000 foot view. 

Thanks for the help in advance. 

Joel 



-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100205/e52cbc2b/attachment-0002.html 

From kristian.kielhofner at gmail.com  Fri Feb  5 10:11:35 2010
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Fri, 5 Feb 2010 13:11:35 -0500
Subject: [Freeswitch-users] Can Freeswitch and OpenSER Co-reside on Same
	Server?
In-Reply-To: <C4DC856C75094358AEDBE3FA0E242741@greyhawk.tonecommander.com>
References: <C4DC856C75094358AEDBE3FA0E242741@greyhawk.tonecommander.com>
Message-ID: <2d9149cd1002051011x3b4f548dqabe615ed5f991687@mail.gmail.com>

On Fri, Feb 5, 2010 at 12:16 PM, Jerry Richards
<jerry.richards at teotech.com> wrote:
> If I use OpenSER for a session border controller, does anyone see an issue
> if it resides on the same server as Freeswitch? ?So I would have a LAN and
> WAN socket? ?Are there any drawbacks (other than loading) to worry about?
>
> Thanks And Best Regards,
> Jerry
>

You can use different IP addresses or ports.  I do this all of the time.

I question why you are using OpenSER (OpenSIPS?) as a SBC.  FreeSWITCH
is actually more well suited to most of the functions served by
something called* a "session border controller".

For example, FreeSWITCH in bypass media mode is a signaling only SBC
where you can (cleanly) do the header rewriting, number formatting,
and SIP topology hiding typically done by a SBC without touching the
media.  Proxy media mode can do the same while proxying media
(traversing NAT and hiding real RTP addresses).  FreeSWITCH in normal
bridging mode can transcode, convert between different types of DTMF
and do everything else mentioned above.

OpenSER as a proxy can't even (per RFC3261) rewrite To or From, nor
will it hide topology (it simple adds Record-Route/Via).

* Session borders controllers are very ill-defined and mean different
things to different people.

-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com



From kristian.kielhofner at gmail.com  Fri Feb  5 10:20:01 2010
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Fri, 5 Feb 2010 13:20:01 -0500
Subject: [Freeswitch-users] Use of mod_Conference
In-Reply-To: <648878714.68411265393292092.JavaMail.root@mail-2.01.com>
References: <472380364.66881265393034269.JavaMail.root@mail-2.01.com>
	<648878714.68411265393292092.JavaMail.root@mail-2.01.com>
Message-ID: <2d9149cd1002051020n2035e423y701f9608b098d1c8@mail.gmail.com>

On Fri, Feb 5, 2010 at 1:08 PM,  <joel.sisko at iconverged.com> wrote:
> Group,
>
> I have a simple question I think about mod_Conference, does each conference
> room have to be created dynamically? My question is born from how Asterisk
> MeetMe room can work where you can have all the meetme rooms programmed
> statically in the meetme.conf file.
>
> What I am looking to do is rather than using the Asterisk meetme room
> application, I want to use FreeSwitch to provide this functionality. So I am
> trying to figure out what the integration would look like from a 10,000 foot
> view.
>
> Thanks for the help in advance.
>
> Joel

Joel,

  I think what you are looking for is "profile" support in
mod_conference, where you can specify most of the configuration
parameters for a specific conference statically:

http://wiki.freeswitch.org/wiki/Mod_conference#.3Cprofiles.3E

-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com



From anthony.minessale at gmail.com  Fri Feb  5 10:24:10 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 5 Feb 2010 12:24:10 -0600
Subject: [Freeswitch-users] A leg disconnects after uuid_bridge?
In-Reply-To: <fe67a3341002042041u1ee70b86u3bc91adbb6fb3d52@mail.gmail.com>
References: <fe67a3341002021934m548cde89p4d48381c5abe02a0@mail.gmail.com>
	<191c3a031002022108ta22287ev40fc36420abba2e5@mail.gmail.com>
	<fe67a3341002030112yeea379u90c22c482614c57@mail.gmail.com>
	<87f2f3b91002030605j7a1f9a02g665ba043c6ed7f7b@mail.gmail.com>
	<fe67a3341002040132n3aa2a897m60c5b9f1ca6b347f@mail.gmail.com>
	<0A571E14-BB58-4A3D-996A-9D5205668E85@freeswitch.org>
	<fe67a3341002040452v5468fd1fh96d1c5062091474b@mail.gmail.com>
	<191c3a031002040754r2987a03fo18ee2e1ceb239f6a@mail.gmail.com>
	<fe67a3341002042041u1ee70b86u3bc91adbb6fb3d52@mail.gmail.com>
Message-ID: <191c3a031002051024w1f484febj34a34d9e06421c52@mail.gmail.com>

try latest trunk i think your issue is fixed.


On Thu, Feb 4, 2010 at 10:41 PM, Nagalenoj H. <nagalenoj at gmail.com> wrote:

> Sorry., I couldn't understand its behavior.
>
> Let me ask the same question in this way.
>
> * hangup_after_bridge is set to false.
> * In outbound socket, first I answer the call.
> * When I do a bridge to a extension (1001), after 1001 disconnects the
> call. I am able to make another call.
>       sendmsg
>       call-command: execute
>       execute-app-name: bridge
>       execute-app-arg: user/1001
>
> * When I originate a call to extension (1001), after 1001 disconnects the
> call. I'm unable to make another call, because my session is also getting
> closed.
>       api originate user/1001 &park
>
>       Content-Type: api/response
>       Content-Length: 41
>
>      +OK 1fac17ce-120b-11df-a878-d9c7fbcf71c4
>
>
>       sendmsg
>       call-command: execute
>       execute-app-name: intercept
>       execute-app-arg: 1fac17ce-120b-11df-a878-d9c7fbcf71c4
>
> * In both the case, the call is getting bridged to an extension and
> hangup_after_bridge is false.
> * When bridge doesn't need any other variables to set to continue, why
> intercept needs a explicit park after bridge.?
>
> Hope, this has some clarity.,
>
>
> On Thu, Feb 4, 2010 at 9:24 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>>
>>
>>    1. 2010-02-04 14:30:09.574084 [DEBUG] switch_core_state_machine.c:354
>>    (sofia/internal/1010 at 192.168.1.222) State SOFT_EXECUTE going to sleep
>>    2. 2010-02-04 14:30:09.574084 [DEBUG] switch_core_state_machine.c:314
>>    (sofia/internal/1010 at 192.168.1.222) Running State Change CS_EXECUTE
>>    3. 2010-02-04 14:30:09.574084 [DEBUG] switch_core_state_machine.c:348
>>    (sofia/internal/1010 at 192.168.1.222) State EXECUTE
>>    4. 2010-02-04 14:30:09.574084 [DEBUG] mod_sofia.c:181 sofia/internal/
>>    1010 at 192.168.1.222 SOFIA EXECUTE
>>    5. 2010-02-04 14:30:09.574084 [DEBUG] switch_core_state_machine.c:159sofia/internal/
>>    1010 at 192.168.1.222 Standard EXECUTE
>>    6. 2010-02-04 14:30:09.574084 [NOTICE] switch_core_state_machine.c:187Hangup sofia/internal/
>>    1010 at 192.168.1.222 [CS_EXECUTE] [NORMAL_CLEARING]
>>
>>
>>
>> Your channel went back to EXECUTE as expected then it hungup because there
>> were no more instructions in your dial plan for it to execute.  So it is
>> working as expected.
>>
>> Consider using transfer_after_bridge variable or park_after bridge to make
>> it stay around when the call is over.
>>
>>
>>
>>
>> On Thu, Feb 4, 2010 at 6:52 AM, Nagalenoj H. <nagalenoj at gmail.com> wrote:
>>
>>> By using create_uuid. I've also tried without giving origination_uuid.
>>> But, the result is same.
>>>
>>> --
>>> Regards,
>>> Nagalenoj H.
>>>
>>>
>>> On Thu, Feb 4, 2010 at 5:42 PM, Brian West <brian at freeswitch.org> wrote:
>>>
>>>> Where are you getting this UUID?
>>>>
>>>> /b
>>>>
>>>> On Feb 4, 2010, at 3:32 AM, Nagalenoj H. wrote:
>>>>
>>>> > api originate
>>>> {origination_uuid=c6d2e0e2-0b2f-11df-9e84-fb15c3cd8565}user/1001 &park
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Regards,
> Nagalenoj H.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100205/3696c4c0/attachment-0002.html 

From red.rain.seven at gmail.com  Fri Feb  5 10:36:13 2010
From: red.rain.seven at gmail.com (Henry Huang)
Date: Sat, 6 Feb 2010 02:36:13 +0800
Subject: [Freeswitch-users] Can Freeswitch and OpenSER Co-reside on Same
	Server?
In-Reply-To: <2d9149cd1002051011x3b4f548dqabe615ed5f991687@mail.gmail.com>
References: <C4DC856C75094358AEDBE3FA0E242741@greyhawk.tonecommander.com>
	<2d9149cd1002051011x3b4f548dqabe615ed5f991687@mail.gmail.com>
Message-ID: <59ad9ca11002051036w5817127ai66cad23046c100c1@mail.gmail.com>

Kristian:

Can you point me to the wiki link where it describes how to do the header
rewriting and number formatting and topology hiding?
I am also looking into OpenSIPS to be a session boarder controller, but if
freeswitch is already able to do all these, then I think it's easier for me
to stick with it since I already learn and do a lot with it already.

Thanks,

On Sat, Feb 6, 2010 at 2:11 AM, Kristian Kielhofner <
kristian.kielhofner at gmail.com> wrote:

> On Fri, Feb 5, 2010 at 12:16 PM, Jerry Richards
> <jerry.richards at teotech.com> wrote:
> > If I use OpenSER for a session border controller, does anyone see an
> issue
> > if it resides on the same server as Freeswitch?  So I would have a LAN
> and
> > WAN socket?  Are there any drawbacks (other than loading) to worry about?
> >
> > Thanks And Best Regards,
> > Jerry
> >
>
> You can use different IP addresses or ports.  I do this all of the time.
>
> I question why you are using OpenSER (OpenSIPS?) as a SBC.  FreeSWITCH
> is actually more well suited to most of the functions served by
> something called* a "session border controller".
>
> For example, FreeSWITCH in bypass media mode is a signaling only SBC
> where you can (cleanly) do the header rewriting, number formatting,
> and SIP topology hiding typically done by a SBC without touching the
> media.  Proxy media mode can do the same while proxying media
> (traversing NAT and hiding real RTP addresses).  FreeSWITCH in normal
> bridging mode can transcode, convert between different types of DTMF
> and do everything else mentioned above.
>
> OpenSER as a proxy can't even (per RFC3261) rewrite To or From, nor
> will it hide topology (it simple adds Record-Route/Via).
>
> * Session borders controllers are very ill-defined and mean different
> things to different people.
>
> --
> Kristian Kielhofner
> http://www.astlinux.org
> http://blog.krisk.org
> http://www.star2star.com
> http://www.submityoursip.com
> http://www.voalte.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Henry Huang
UniC Solution - Communication Unified
VoIP & Open Source software Consultant
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100206/675cf29a/attachment-0002.html 

From costa.zikalala at gmail.com  Fri Feb  5 10:44:42 2010
From: costa.zikalala at gmail.com (Costa Zikalala)
Date: Fri, 5 Feb 2010 20:44:42 +0200
Subject: [Freeswitch-users] Difference Between "bridge" and "transfer"
In-Reply-To: <87f2f3b91002041632t19f17aefoaf11e048150816c5@mail.gmail.com>
References: <D123D09A29B942D988B3456528DD44AD@greyhawk.tonecommander.com> 
	<87f2f3b91002041632t19f17aefoaf11e048150816c5@mail.gmail.com>
Message-ID: <59daa2cd1002051044p1bee852ele8d0d972c303ea0e@mail.gmail.com>

Whilst on this subject, if I receive a call from PSTN and I 'bridge' it to
another PSTN extension, will the PSTN Provider charge me for the 'b-leg' or
will the original caller be charged the entire call?



On 5 February 2010 02:32, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Thu, Feb 4, 2010 at 3:56 PM, Jerry Richards <jerry.richards at teotech.com
> > wrote:
>
>> What is the difference between "bridge" and "transfer"?  I'm looking at
>> the
>> demo IVRs.
>>
>>
> bridge will connect two endpoints together while transfer sends the
> endpoint back through the dialplan again...
>
> -MC
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100205/9ac3de36/attachment-0002.html 

From joel.sisko at iconverged.com  Fri Feb  5 10:59:31 2010
From: joel.sisko at iconverged.com (Joel Sisko)
Date: Fri, 5 Feb 2010 12:59:31 -0600 (CST)
Subject: [Freeswitch-users] Use of mod_Conference
In-Reply-To: <2d9149cd1002051020n2035e423y701f9608b098d1c8@mail.gmail.com>
Message-ID: <875390421.80391265396371614.JavaMail.root@mail-2.01.com>

Kristian,

Thank you for the quick reply.

The profile looks like it meant to be specific to any or all conferences that are created (a template).

I think I have a different way to clarify my question:

Lets assume on my Asterisk system that someone wants to enter conference room 201. I want to pass them into conference room 201 on my FreeSwitch server, so to over simplify I will pass the caller to conference201 at myFreeSwitch.server.com, but if  the conference room has not been created then there is nothing to transfer to.(?) So is there a way that to use  a static configuration so the conference room is already created? Or must I create that conference room prior to my first request for that room, then transfer the caller after the creation of the conference room?

Joel
----- Original Message -----
From: "Kristian Kielhofner" <kristian.kielhofner at gmail.com>
To: freeswitch-users at lists.freeswitch.org
Sent: Friday, February 5, 2010 10:20:01 AM GMT -08:00 US/Canada Pacific
Subject: Re: [Freeswitch-users] Use of mod_Conference

On Fri, Feb 5, 2010 at 1:08 PM,  <joel.sisko at iconverged.com> wrote:
> Group,
>
> I have a simple question I think about mod_Conference, does each conference
> room have to be created dynamically? My question is born from how Asterisk
> MeetMe room can work where you can have all the meetme rooms programmed
> statically in the meetme.conf file.
>
> What I am looking to do is rather than using the Asterisk meetme room
> application, I want to use FreeSwitch to provide this functionality. So I am
> trying to figure out what the integration would look like from a 10,000 foot
> view.
>
> Thanks for the help in advance.
>
> Joel

Joel,

  I think what you are looking for is "profile" support in
mod_conference, where you can specify most of the configuration
parameters for a specific conference statically:

http://wiki.freeswitch.org/wiki/Mod_conference#.3Cprofiles.3E

-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



From kristian.kielhofner at gmail.com  Fri Feb  5 11:12:53 2010
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Fri, 5 Feb 2010 14:12:53 -0500
Subject: [Freeswitch-users] Can Freeswitch and OpenSER Co-reside on Same
	Server?
In-Reply-To: <59ad9ca11002051036w5817127ai66cad23046c100c1@mail.gmail.com>
References: <C4DC856C75094358AEDBE3FA0E242741@greyhawk.tonecommander.com>
	<2d9149cd1002051011x3b4f548dqabe615ed5f991687@mail.gmail.com>
	<59ad9ca11002051036w5817127ai66cad23046c100c1@mail.gmail.com>
Message-ID: <2d9149cd1002051112y4c82bb77m25aa43d70c1005d3@mail.gmail.com>

On Fri, Feb 5, 2010 at 1:36 PM, Henry Huang <red.rain.seven at gmail.com> wrote:
> Kristian:
>
> Can you point me to the wiki link where it describes how to do the header
> rewriting and number formatting and topology hiding?
> I am also looking into OpenSIPS to be a session boarder controller, but if
> freeswitch is already able to do all these, then I think it's easier for me
> to stick with it since I already learn and do a lot with it already.
>
> Thanks,
>

Henry,

  By default (when using bridge) FreeSWITCH will generate a new leg
with a fresh set of headers.  Various channel variables can be used to
influence the values of some of these:

Request URI: based on bridge string (411 at sip.provider.com) - can also
include transport, port, uri params, etc
To: same as Request URI (possible minus some of the params)
From: Determined by gateway config (if used: use-callerid-in-from) and
effective_caller_id_number/effective_caller_id_name
RPID/PAI: sip_cid_type

  The other params will be taken from the Sofia config (session
timers, etc).  Use multiple profiles for internal and external
networks (just like in the samples).

  FreeSWITCH does topology hiding by default - as a B2BUA (regardless
of "mode") it will do topology hiding by creating a new channel/leg.
None of the IP addresses etc, from the original channel are visible.
Note that this doesn't count the SDP if you are using bypass media or
proxy media.  If you want *full* SBC style topology hiding with media
you can't use these modes but you'll pay for it in performance.

  Unless you use the uac/uas modules and/or some textops based
manipulation in OpenSER, all OpenSER can do is copy most of the
headers/body, add Record-Route and Via headers, and forward the
message to the next hop while leaving all IP addresses, etc intact.
OpenSER/OpenSIPS can be a fairly general purpose SIP server but it's
main function is a fairly strict RFC 3261 compliant proxy.

  I should also point out that OpenSIPS does have a B2BUA module.  I
myself would much rather just use FreeSWITCH.  You know - the best
tool for the job.  They're both EXCELLENT pieces of software and
between the two of them you can build incredible VoIP solutions and
networks.

  I use FreeSWITCH to interface with each of my carriers.  FreeSWITCH
runs on the same machine as our main SIP proxy running in
bypass_media.  The SIP proxy (OpenSER) handles all of the requests for
our servers/customers (servers we provision and control) while
FreeSWITCH (usually in bypass media) interfaces with each of our
carriers to make everyone happy at the signaling level.  I have a
profile for our network and a profile for each carrier.  No need to
worry about different caller id formats, number formats (e.164),
transports, caller id, etc.  On one machine FreeSWITCH regularly does
over 1200 channels using about %20 CPU.  It's a four year old Dell
1850 :).

-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com



From lists at redbonez.net  Fri Feb  5 11:21:17 2010
From: lists at redbonez.net (Adam Ford)
Date: Fri, 5 Feb 2010 12:21:17 -0700
Subject: [Freeswitch-users] Looking for some good/cheap desktop phones
In-Reply-To: <d361ca7b1002050507n5e564621oa6cf7ace005dd5b8@mail.gmail.com>
References: <3f2e071b5aea5e11bb37a37e435a33b7@mail.gmail.com>	<8e9d67561002042236s434e0643od322551de8720be1@mail.gmail.com>	<409D0ACF-ADA4-4D02-8B7E-71EC1B1BAFBC@freeswitch.org>	<8e9d67561002042301v76129abfj3dcbc70ed073752c@mail.gmail.com>
	<d361ca7b1002050507n5e564621oa6cf7ace005dd5b8@mail.gmail.com>
Message-ID: <00ae01caa698$65b570f0$312052d0$@net>

I just picked up old model Polycoms.  You can get the IP301's for ~$60-70
new and the IP501s for ~$100 new.  They don't have some of the fancier
features of the new Polycoms, but they carry the same quality and
configurability(with the exception of NAT).

 

-Adam

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Yehavi
Bourvine
Sent: Friday, February 05, 2010 6:07 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Looking for some good/cheap desktop phones

 

>From my experience Polycom and SNOM are expensive but give you what you
need. Polycom is more intutive to the users but more cumbersome for the
manager to deploy; SNOM is somewhat less intuitive to the user but
everything can be set via the WEB interface.

 

If you talk about 4-5 phones, then probably SNOM is the choice. It also
depends about the specific functions you want to use. I our specific
environment (high use of BLF and shared lines) Polycom wins because it
handles these functions just as the user expects.

 

I did not try Aastra so cannot testify. We did test Yealink, Thomson,
Asterphone, SipTip and maybe others I forgot. Cisco also seems good but
Cisco does not supply the required socumentation to make them fully working.

 

                    Regards, __Yehavi:

2010/2/5 ????? ??????? <tim at novion.ru>

Sure, those phones do not deliver superior usability, but they at
least give the best sound among budget models.


2010/2/5 Brian West <brian at freeswitch.org>:

> And all of those are awful phones.  They don't even make good paper
weights.
>
> You can't have good and cheap in the same sentence when talking about VoIP
phones.  You have to take your pick between quality (good) and price (cheap)
you can't have both at once.
>
> /b
>
>
> On Feb 5, 2010, at 12:36 AM, ????? ??????? wrote:
>
>> Have a look at Yealink (Skypemate) and Fanvill
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org <http://www.freeswitch.org/> 
>

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org <http://www.freeswitch.org/> 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100205/4417ceba/attachment-0002.html 

From msc at freeswitch.org  Fri Feb  5 11:25:10 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 5 Feb 2010 11:25:10 -0800
Subject: [Freeswitch-users] Difference Between "bridge" and "transfer"
In-Reply-To: <59daa2cd1002051044p1bee852ele8d0d972c303ea0e@mail.gmail.com>
References: <D123D09A29B942D988B3456528DD44AD@greyhawk.tonecommander.com>
	<87f2f3b91002041632t19f17aefoaf11e048150816c5@mail.gmail.com>
	<59daa2cd1002051044p1bee852ele8d0d972c303ea0e@mail.gmail.com>
Message-ID: <87f2f3b91002051125u31dbf290h531d34d078c0f42e@mail.gmail.com>

On Fri, Feb 5, 2010 at 10:44 AM, Costa Zikalala <costa.zikalala at gmail.com>wrote:

> Whilst on this subject, if I receive a call from PSTN and I 'bridge' it to
> another PSTN extension, will the PSTN Provider charge me for the 'b-leg' or
> will the original caller be charged the entire call?
>

That depends... is the "other" leg an outbound call? Is the other leg an
inbound call to a toll-free number?
-MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100205/b1f33908/attachment-0002.html 

From kristian.kielhofner at gmail.com  Fri Feb  5 11:25:11 2010
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Fri, 5 Feb 2010 14:25:11 -0500
Subject: [Freeswitch-users] Use of mod_Conference
In-Reply-To: <875390421.80391265396371614.JavaMail.root@mail-2.01.com>
References: <2d9149cd1002051020n2035e423y701f9608b098d1c8@mail.gmail.com>
	<875390421.80391265396371614.JavaMail.root@mail-2.01.com>
Message-ID: <2d9149cd1002051125u20eef2c8jf9ce39a0223238b@mail.gmail.com>

On Fri, Feb 5, 2010 at 1:59 PM, Joel Sisko <joel.sisko at iconverged.com> wrote:
> Kristian,
>
> Thank you for the quick reply.
>
> The profile looks like it meant to be specific to any or all conferences that are created (a template).
>
> I think I have a different way to clarify my question:
>
> Lets assume on my Asterisk system that someone wants to enter conference room 201. I want to pass them into conference room 201 on my FreeSwitch server, so to over simplify I will pass the caller to conference201 at myFreeSwitch.server.com, but if ?the conference room has not been created then there is nothing to transfer to.(?) So is there a way that to use ?a static configuration so the conference room is already created? Or must I create that conference room prior to my first request for that room, then transfer the caller after the creation of the conference room?
>
> Joel

Joel,

  Let me try to explain using a sample taken from the default dialplan:

    <extension name="conference201">
      <condition field="destination_number" expression="^conference201$">
        <action application="answer"/>
        <action application="conference" data="201 at default"/>
      </condition>
    </extension>

  (Yes I could use variables here but I wanted to keep it simple).

  This would create the conference with the first caller that called
in.  The conference number would be 201 and it would use the "default"
profile from the conference configuration.  Any number of conferences
can be defined in the dialplan with different settings, pins, etc
depending on the profile used and the arguments passed to conference
via data=.

-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com



From jerry.richards at teotech.com  Fri Feb  5 11:38:09 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Fri, 5 Feb 2010 11:38:09 -0800
Subject: [Freeswitch-users] Presence PUBLISH Not Updating After Softphone
	OffLine Then Available
Message-ID: <31FEE9072F104AE18B26812978FCA341@greyhawk.tonecommander.com>

I found a scenario where presence status is not distributed to subscribers.
This is using the latest changes (as of Feb 03, 2010).  The scenario
follows:

1) Register two Bria softphones (A and B), which each have the other as a
contact.
2) Set softphone B's presence status to 'Appear Offline'.
3) Softphone A correctly reflects contact B is offline.
4) Set softphone B's presence status to 'Available'.
5) ******* There is no change to contact B's status at softphone A *******

I posted a log at http://pastebin.freeswitch.org/12054.  At line 773, there
is an error when FS is processing the PUBLISH from softphone B:

773.2010-02-05 10:29:21.254221 [ERR] sofia_presence.c:674 DUMP PRESENCE SQL:

I did notice that after about 30 minutes, softphone B's status gets
reflected at softphone A.

Thanks and Best Regards,
Jerry




From joel.sisko at iconverged.com  Fri Feb  5 11:39:59 2010
From: joel.sisko at iconverged.com (Joel Sisko)
Date: Fri, 5 Feb 2010 13:39:59 -0600 (CST)
Subject: [Freeswitch-users] Use of mod_Conference
In-Reply-To: <2d9149cd1002051125u20eef2c8jf9ce39a0223238b@mail.gmail.com>
Message-ID: <1852904235.89881265398799084.JavaMail.root@mail-2.01.com>

Thanks for the insight.

Joel
----- Original Message -----
From: "Kristian Kielhofner" <kristian.kielhofner at gmail.com>
To: freeswitch-users at lists.freeswitch.org
Sent: Friday, February 5, 2010 11:25:11 AM GMT -08:00 US/Canada Pacific
Subject: Re: [Freeswitch-users] Use of mod_Conference

On Fri, Feb 5, 2010 at 1:59 PM, Joel Sisko <joel.sisko at iconverged.com> wrote:
> Kristian,
>
> Thank you for the quick reply.
>
> The profile looks like it meant to be specific to any or all conferences that are created (a template).
>
> I think I have a different way to clarify my question:
>
> Lets assume on my Asterisk system that someone wants to enter conference room 201. I want to pass them into conference room 201 on my FreeSwitch server, so to over simplify I will pass the caller to conference201 at myFreeSwitch.server.com, but if ?the conference room has not been created then there is nothing to transfer to.(?) So is there a way that to use ?a static configuration so the conference room is already created? Or must I create that conference room prior to my first request for that room, then transfer the caller after the creation of the conference room?
>
> Joel

Joel,

  Let me try to explain using a sample taken from the default dialplan:

    <extension name="conference201">
      <condition field="destination_number" expression="^conference201$">
        <action application="answer"/>
        <action application="conference" data="201 at default"/>
      </condition>
    </extension>

  (Yes I could use variables here but I wanted to keep it simple).

  This would create the conference with the first caller that called
in.  The conference number would be 201 and it would use the "default"
profile from the conference configuration.  Any number of conferences
can be defined in the dialplan with different settings, pins, etc
depending on the profile used and the arguments passed to conference
via data=.

-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



From jerry.richards at teotech.com  Fri Feb  5 11:40:06 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Fri, 5 Feb 2010 11:40:06 -0800
Subject: [Freeswitch-users] Can Freeswitch and OpenSER Co-reside on
	SameServer?
In-Reply-To: <2d9149cd1002051011x3b4f548dqabe615ed5f991687@mail.gmail.com>
References: <C4DC856C75094358AEDBE3FA0E242741@greyhawk.tonecommander.com>
	<2d9149cd1002051011x3b4f548dqabe615ed5f991687@mail.gmail.com>
Message-ID: <0203851F538F44B18BFD40EC383EBEBF@greyhawk.tonecommander.com>

So do you build your server with two FS instances running?  One as the SBC
and one as Proxy/PBX?

Thanks,
Jerry
 

-----Original Message-----
From: Kristian Kielhofner [mailto:kristian.kielhofner at gmail.com] 
Sent: Friday, February 05, 2010 10:12 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Can Freeswitch and OpenSER Co-reside on
SameServer?

On Fri, Feb 5, 2010 at 12:16 PM, Jerry Richards <jerry.richards at teotech.com>
wrote:
> If I use OpenSER for a session border controller, does anyone see an 
> issue if it resides on the same server as Freeswitch? ?So I would have 
> a LAN and WAN socket? ?Are there any drawbacks (other than loading) to
worry about?
>
> Thanks And Best Regards,
> Jerry
>

You can use different IP addresses or ports.  I do this all of the time.

I question why you are using OpenSER (OpenSIPS?) as a SBC.  FreeSWITCH is
actually more well suited to most of the functions served by something
called* a "session border controller".

For example, FreeSWITCH in bypass media mode is a signaling only SBC where
you can (cleanly) do the header rewriting, number formatting, and SIP
topology hiding typically done by a SBC without touching the media.  Proxy
media mode can do the same while proxying media (traversing NAT and hiding
real RTP addresses).  FreeSWITCH in normal bridging mode can transcode,
convert between different types of DTMF and do everything else mentioned
above.

OpenSER as a proxy can't even (per RFC3261) rewrite To or From, nor will it
hide topology (it simple adds Record-Route/Via).

* Session borders controllers are very ill-defined and mean different things
to different people.

--
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com






From lon at kickasspixels.com  Fri Feb  5 11:42:50 2010
From: lon at kickasspixels.com (Lon Baker)
Date: Fri, 5 Feb 2010 11:42:50 -0800
Subject: [Freeswitch-users] Testing Config Changes
Message-ID: <5d3e0dc61002051142h65c3f533t9524ef0510af02fd@mail.gmail.com>

I'm looking for the best practice for testing configuration changes on
a live Fresswitch server.

Is it best to use reloadxml in the cli? Will that alert us to issues
with syntax and other errors without bringing down the server?

Lon



From kristian.kielhofner at gmail.com  Fri Feb  5 11:50:00 2010
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Fri, 5 Feb 2010 14:50:00 -0500
Subject: [Freeswitch-users] Can Freeswitch and OpenSER Co-reside on
	SameServer?
In-Reply-To: <0203851F538F44B18BFD40EC383EBEBF@greyhawk.tonecommander.com>
References: <C4DC856C75094358AEDBE3FA0E242741@greyhawk.tonecommander.com>
	<2d9149cd1002051011x3b4f548dqabe615ed5f991687@mail.gmail.com>
	<0203851F538F44B18BFD40EC383EBEBF@greyhawk.tonecommander.com>
Message-ID: <2d9149cd1002051150i3fba0945s4332af51d261274c@mail.gmail.com>

On Fri, Feb 5, 2010 at 2:40 PM, Jerry Richards
<jerry.richards at teotech.com> wrote:
> So do you build your server with two FS instances running? ?One as the SBC
> and one as Proxy/PBX?
>
> Thanks,
> Jerry

Jerry,

  No.  One instance of FreeSWITCH and one instance of OpenSER.  As I
said, just make sure they use separate IPs and/or ports.  I prefer
standard ports and separate IPs because then (in the future) if I need
to split them (scaling, redundancy, etc) all I have to do is bring up
the second IP on a different host and move the software/config.

-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com



From brian at freeswitch.org  Fri Feb  5 11:53:30 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 5 Feb 2010 13:53:30 -0600
Subject: [Freeswitch-users] Testing Config Changes
In-Reply-To: <5d3e0dc61002051142h65c3f533t9524ef0510af02fd@mail.gmail.com>
References: <5d3e0dc61002051142h65c3f533t9524ef0510af02fd@mail.gmail.com>
Message-ID: <14964F2E-0D87-4E40-8CB9-E7ACA815FB34@freeswitch.org>

This all depends on what all you're changing.. some things you can't change no matter how hard you try.

/b

On Feb 5, 2010, at 1:42 PM, Lon Baker wrote:

> I'm looking for the best practice for testing configuration changes on
> a live Fresswitch server.
> 
> Is it best to use reloadxml in the cli? Will that alert us to issues
> with syntax and other errors without bringing down the server?
> 
> Lon




From kristian.kielhofner at gmail.com  Fri Feb  5 11:54:30 2010
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Fri, 5 Feb 2010 14:54:30 -0500
Subject: [Freeswitch-users] Use of mod_Conference
In-Reply-To: <1852904235.89881265398799084.JavaMail.root@mail-2.01.com>
References: <2d9149cd1002051125u20eef2c8jf9ce39a0223238b@mail.gmail.com>
	<1852904235.89881265398799084.JavaMail.root@mail-2.01.com>
Message-ID: <2d9149cd1002051154w507c31e9y8c7dfc9c64956542@mail.gmail.com>

On Fri, Feb 5, 2010 at 2:39 PM, Joel Sisko <joel.sisko at iconverged.com> wrote:
> Thanks for the insight.
>
> Joel

No problem.  Maybe we'll be able to put this stuff in writing someday
if O'Reilly ever publishes a FreeSWITCH book...  If you recall, you
and I worked on "VoIP Hacks" and "The Future of Asterisk".

P.S. - Ted Wallingford friend requested me on Facebook today.  What
are the chances? ;)

-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com



From rupa at rupa.com  Fri Feb  5 11:54:40 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Fri, 5 Feb 2010 13:54:40 -0600
Subject: [Freeswitch-users] Looking for some good/cheap desktop phones
In-Reply-To: <00ae01caa698$65b570f0$312052d0$@net>
References: <3f2e071b5aea5e11bb37a37e435a33b7@mail.gmail.com>
	<8e9d67561002042236s434e0643od322551de8720be1@mail.gmail.com>
	<409D0ACF-ADA4-4D02-8B7E-71EC1B1BAFBC@freeswitch.org>
	<8e9d67561002042301v76129abfj3dcbc70ed073752c@mail.gmail.com>
	<d361ca7b1002050507n5e564621oa6cf7ace005dd5b8@mail.gmail.com>
	<00ae01caa698$65b570f0$312052d0$@net>
Message-ID: <b39273531002051154h5132faa9h9fb024ebdfb15b77@mail.gmail.com>

Also be aware they are EOL so no new firmware for them.  The 301 also is not
backlit which can be a pain depending on environment.

2010/2/5 Adam Ford <lists at redbonez.net>

>  I just picked up old model Polycoms.  You can get the IP301's for ~$60-70
> new and the IP501s for ~$100 new.  They don't have some of the fancier
> features of the new Polycoms, but they carry the same quality and
> configurability(with the exception of NAT).
>
>
>
> -Adam
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Yehavi
> Bourvine
> *Sent:* Friday, February 05, 2010 6:07 AM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Looking for some good/cheap desktop
> phones
>
>
>
> From my experience Polycom and SNOM are expensive but give you what you
> need. Polycom is more intutive to the users but more cumbersome for the
> manager to deploy; SNOM is somewhat less intuitive to the user but
> everything can be set via the WEB interface.
>
>
>
> If you talk about 4-5 phones, then probably SNOM is the choice. It also
> depends about the specific functions you want to use. I our specific
> environment (high use of BLF and shared lines) Polycom wins because it
> handles these functions just as the user expects.
>
>
>
> I did not try Aastra so cannot testify. We did test Yealink, Thomson,
> Asterphone, SipTip and maybe others I forgot. Cisco also seems good but
> Cisco does not supply the required socumentation to make them fully working.
>
>
>
>                     Regards, __Yehavi:
>
> 2010/2/5 ????? ??????? <tim at novion.ru>
>
> Sure, those phones do not deliver superior usability, but they at
> least give the best sound among budget models.
>
>
> 2010/2/5 Brian West <brian at freeswitch.org>:
>
> > And all of those are awful phones.  They don't even make good paper
> weights.
> >
> > You can't have good and cheap in the same sentence when talking about
> VoIP phones.  You have to take your pick between quality (good) and price
> (cheap) you can't have both at once.
> >
> > /b
> >
> >
> > On Feb 5, 2010, at 12:36 AM, ????? ??????? wrote:
> >
> >> Have a look at Yealink (Skypemate) and Fanvill
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100205/260ede3d/attachment-0002.html 

From rupa at rupa.com  Fri Feb  5 11:58:13 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Fri, 5 Feb 2010 13:58:13 -0600
Subject: [Freeswitch-users] Testing Config Changes
In-Reply-To: <5d3e0dc61002051142h65c3f533t9524ef0510af02fd@mail.gmail.com>
References: <5d3e0dc61002051142h65c3f533t9524ef0510af02fd@mail.gmail.com>
Message-ID: <b39273531002051158y18ea21a5ka5b174927778a24f@mail.gmail.com>

it'll alert you to syntax errors, but pretty much any other mistake you'll
find out when calls start failing or misbehaving.  Better to have a staging
server where you test stuff out (with a test plan!).

On Fri, Feb 5, 2010 at 1:42 PM, Lon Baker <lon at kickasspixels.com> wrote:

> I'm looking for the best practice for testing configuration changes on
> a live Fresswitch server.
>
> Is it best to use reloadxml in the cli? Will that alert us to issues
> with syntax and other errors without bringing down the server?
>
> Lon
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
-Rupa
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100205/ff170705/attachment-0002.html 

From tim at novion.ru  Fri Feb  5 12:02:38 2010
From: tim at novion.ru (Timur Valishev)
Date: Fri, 5 Feb 2010 23:02:38 +0300
Subject: [Freeswitch-users] No reINVITE when bridging two sessions from
	JavaScript with bypass_media_after_bridge=true
In-Reply-To: <B66289FF-C2A9-4AC4-A56C-66E51B4726F4@freeswitch.org>
References: <8e9d67561002041147q6c135546u1f7e1887529a916d@mail.gmail.com>
	<3D739D7F-F5C6-4266-B04E-9413DCFDA6AF@freeswitch.org>
	<8e9d67561002050018q24e3a806kc8807b55268e759b@mail.gmail.com>
	<B66289FF-C2A9-4AC4-A56C-66E51B4726F4@freeswitch.org>
Message-ID: <8e9d67561002051202qeaf1551v8fdd29565f8c02fd@mail.gmail.com>

I think we are on the right way) still does not work, but there is hope)

First of all, this script does not produce any reinvite either (even if
replace bypass_media to bypass_media_after_bridge, or set bypass_media only
on one channel):

<<<<<<<<<<<<<<<<<<<<<<<<<<<<<
session = new
Session("{bypass_media=true,ignore_early_media=true,hangup_after_bridge=true}user/1001");
session2 = new
Session("{bypass_media=true,ignore_early_media=true} user/1001");
bridge(session, session2);
>>>>>>>>>>>>>>>>>>>>>>>>>>>>>

BUT! if I run the following script:
<<<<<<<<<<<<<<<<<<<<<<<<<<<<<
session = new
Session("{bypass_media=true,ignore_early_media=true,hangup_after_bridge=true}
user/1001");
session2 = new
Session("{bypass_media=true,ignore_early_media=true}user/1001");
>>>>>>>>>>>>>>>>>>>>>>>>>>>>>

And then manually type in the console
uuid_media off <uuid of my session>

- then I get the reINVITE!

BUT! When I try to write it to the script:
<<<<<<<<<<<<<<<<<<<<<<<<<<<<<
session = new
Session("{bypass_media=true,ignore_early_media=true,hangup_after_bridge=true}sofia/external/
timwork at novion.ru");
session2 = new
Session("{bypass_media=true,ignore_early_media=true}sofia/external/
timwork at novion.ru");
bridge(session, session2);
apiExecute('uuid_media off '+session.uuid); // <-- this line is not
executed, because bridge hangs up untill BYE
>>>>>>>>>>>>>>>>>>>>>>>>>>>>>

the last line is not executed, because bridge hangs up untill BYE

Then I've tried to do like this:
<<<<<<<<<<<<<<<<<<<<<<<<<<<<<
session = new
Session("{bypass_media=true,ignore_early_media=true,hangup_after_bridge=true}user/1001");
session2 = new
Session("{bypass_media=true,ignore_early_media=true}user/1001");

session.setAutoHangup(false)
session2.setAutoHangup(false)

apiExecute("uuid_bridge "+session.uuid+" "+session2.uuid);
apiExecute('uuid_media off '+session.uuid);
>>>>>>>>>>>>>>>>>>>>>>>>>>>>>

But sessions do not get bridged -( Even if I insert session.ready() after
each call.

Any ideas on how to call the functions correctly to get the reINVITE?

Best regards,
Timur Valishev

2010/2/5 Brian West <brian at freeswitch.org>

> set it inside each of the {} for each session you create its not set after
> the fact the call is up already...  you're setting it too late.
>
> you an also issue uuid_media off <uuid>
>
> /b
>
> On Feb 5, 2010, at 2:18 AM, Timur Valishev wrote:
>
> I've modified my script to make sure: <<<<<<<<<<<<<<<<<<<<<<<<<<<<,
> *session = new Session(*
> *"{ignore_early_media=true,hangup_after_bridge=true}sofia/external/
> timwork at novion.ru"*
> *);*
> *session2 = new Session(*
> *"{ignore_early_media=true}sofia/external/timwork at novion.ru"*
> *);*
> *session.setVariable('bypass_media', 'true');*
> *session2.setVariable('bypass_media', 'true');*
> *bridge(session, session2);*
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100205/3bc58a3a/attachment-0002.html 

From lon at kickasspixels.com  Fri Feb  5 12:15:37 2010
From: lon at kickasspixels.com (Lon Baker)
Date: Fri, 5 Feb 2010 12:15:37 -0800
Subject: [Freeswitch-users] Testing Config Changes
In-Reply-To: <b39273531002051158y18ea21a5ka5b174927778a24f@mail.gmail.com>
References: <5d3e0dc61002051142h65c3f533t9524ef0510af02fd@mail.gmail.com> 
	<b39273531002051158y18ea21a5ka5b174927778a24f@mail.gmail.com>
Message-ID: <5d3e0dc61002051215i619d7cb2t6293edbe6397e1bc@mail.gmail.com>

I agree with the staging and test plan, this is something we do.

I want to implement a procedure on the production servers to help
insure that we are as bullet proof as possible.

On Fri, Feb 5, 2010 at 11:58 AM, Rupa Schomaker <rupa at rupa.com> wrote:
> it'll alert you to syntax errors, but pretty much any other mistake you'll
> find out when calls start failing or misbehaving. ?Better to have a staging
> server where you test stuff out (with a test plan!).
>
> On Fri, Feb 5, 2010 at 1:42 PM, Lon Baker <lon at kickasspixels.com> wrote:
>>
>> I'm looking for the best practice for testing configuration changes on
>> a live Fresswitch server.
>>
>> Is it best to use reloadxml in the cli? Will that alert us to issues
>> with syntax and other errors without bringing down the server?
>>
>> Lon
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From shyjuk at live.com  Fri Feb  5 12:33:19 2010
From: shyjuk at live.com (Shyju Kanaprath)
Date: Sat, 6 Feb 2010 02:03:19 +0530
Subject: [Freeswitch-users] Looking for some good/cheap desktop phones
In-Reply-To: <b39273531002051154h5132faa9h9fb024ebdfb15b77@mail.gmail.com>
References: <3f2e071b5aea5e11bb37a37e435a33b7@mail.gmail.com>,
	<8e9d67561002042236s434e0643od322551de8720be1@mail.gmail.com>,
	<409D0ACF-ADA4-4D02-8B7E-71EC1B1BAFBC@freeswitch.org>,
	<8e9d67561002042301v76129abfj3dcbc70ed073752c@mail.gmail.com>,
	<d361ca7b1002050507n5e564621oa6cf7ace005dd5b8@mail.gmail.com>,
	<00ae01caa698$65b570f0$312052d0$@net>,
	<b39273531002051154h5132faa9h9fb024ebdfb15b77@mail.gmail.com>
Message-ID: <SNT119-W1223C808A5E092C7E2A590B5540@phx.gbl>


Grandstream is also a good choice.. GXP2000 has got 5-6 blf/speed dial buttons and is very user friendly.
Configuring through web interface is also very easy.

Regards,
Shyju
 		 	   		  
_________________________________________________________________
Post free property ads on Yello Classifieds now! www.yello.in
http://ss1.richmedia.in/recurl.asp?pid=219
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100206/0be9922e/attachment-0002.html 

From Prometheus001 at gmx.net  Fri Feb  5 12:58:51 2010
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Fri, 05 Feb 2010 21:58:51 +0100
Subject: [Freeswitch-users] mod_skypiax - help needed with
 modified	snd-dummy (one way audio)
In-Reply-To: <7b197bef1002050536j7ea09684v913434296ae52b97@mail.gmail.com>
References: <4B60555B.2020004@gmx.net>	<7b197bef1001271215o64918f56s3d16e51528fc5f66@mail.gmail.com>
	<4B61ECE0.10409@gmx.net>	<7b197bef1001281210u43a907edi13afe794cf0e1a2e@mail.gmail.com>
	<4B61FCAB.5040707@gmx.net>	<7b197bef1001281341h7ee2c58fj79b3886630901f29@mail.gmail.com>
	<4B62C962.7000601@gmx.net>	<8213d6071001290403g2c08fe39gd92eb0446bb6485a@mail.gmail.com>
	<7b197bef1001290520x37179cb1o63273410e9ef5bc@mail.gmail.com>
	<4B6C1CCB.4080606@gmx.net>
	<7b197bef1002050536j7ea09684v913434296ae52b97@mail.gmail.com>
Message-ID: <4B6C868B.3040406@gmx.net>

Hello Giovanni,

I am now at the point to install Skype. But there is only an Intrepid
version available (no 8.04 version).
The current verison crashed on 8.04x because of dbus error.
    process 8408: D-Bus library appears to be incorrectly set up; failed
to read machine uuid: Failed to open "/var/lib/dbus/machine-id": No such
file or directory
    See the manual page for dbus-uuidgen to correct this issue.
    /usr/bin/skype: symbol lookup error: /usr/bin/skype: undefined
symbol: _ZN10QBoxLayout13addSpacerItemEP11QSpacerIte

Any idea where I can download the older version for 8.04?

Best regards
Peter


Giovanni Maruzzelli schrieb:
> Ciao Peter,
>
> I would use ubuntu 8.04 (hardy) LTS server 64bit or CentOS 5.4 64bit.
>
> -giovanni
>
> On Fri, Feb 5, 2010 at 2:27 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>   
>> Hello Giovanni,
>>
>> as I couldn't even get skype again working again with the standard alsa
>> driver, I would like to setup the machine from scratch based on a
>> working machine.
>> The latest errors I received from Skype was:
>>  snd_pcm_avail_update() returned a value that is exceptionally large:
>> 715706624 bytes (3727638 ms).
>>  Most likely this is a bug in the ALSA driver. Please report this issue
>> to the ALSA developers.
>> I think that may be the reason for one-way-audio.
>>
>> For setting up my machine from scratch, please advise:
>> - which OS you are you using und recommending exactly?
>> - I would like to use 64bit OS in order to use 8GB of memory, does this
>> work?
>> - any other hints?
>>
>> Best regards
>> Peter
>>
>> Giovanni Maruzzelli schrieb:
>>     
>>> Peter,
>>>
>>> Can you connect on IRC (irc.freenode.net #freeswitch)? I'm gmaruzz there.
>>>
>>> Can you restate your problems? I've lost connection :)
>>>
>>> with snd-dummy custom you can create *one only* snd-dummy instance, so
>>> *one only* fake soundcard. If you create more, will not work. But with
>>> that one fake soundcard you can use 64 skype client instances, all
>>> with the same soundcard hardware device (hw:n).
>>>
>>> with original snd-dummy you can create a max of 8 instances, so 8 fake
>>> soundcards, and with each fake soundcard you can  use a max of 8 skype
>>> client instances.
>>>
>>> use the hardware devices, not the default devices (use the "hw:n")
>>>
>>> -giovanni
>>>
>>> On Fri, Jan 29, 2010 at 1:03 PM, Ghulam Mustafa <mustafa.pk at gmail.com> wrote:
>>>
>>>       
>>>> did you enable debug mode while compiling custom snd-dummy? if  yes
>>>> try re-compiling with debug mode disabled.
>>>>
>>>> -m
>>>>
>>>> On Fri, Jan 29, 2010 at 4:41 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>
>>>>         
>>>>> I now reinstalled the original sound drivers
>>>>> Unfortunaltely the sound problems remain, not that worse but they are there:
>>>>> Audio is still (almost) one way. Almost means:
>>>>>
>>>>>    * SIP -> Skype ok
>>>>>    * Skype=> SIP I hear only some scratching on very loud audio
>>>>>
>>>>> Could it be a volume problem? But snd-dummy should have no volume
>>>>> properties, right?
>>>>>
>>>>> Best regards
>>>>> Peter
>>>>>
>>>>>
>>>>> Giovanni Maruzzelli schrieb:
>>>>>
>>>>>           
>>>>>> with three instances you will assign the hw:0 device to skype client
>>>>>> 0...7, hw:1 to skype client 8...15 and hw:2 to skype client 16...23.
>>>>>> Must work. Pay attention to assign the same device name to all devices
>>>>>> needed by a skype instance (sound devices window): playback, capture
>>>>>> AND ring.
>>>>>>
>>>>>> Or maybe is a bug of ALSA on Debian...
>>>>>>
>>>>>> -giovanni
>>>>>>
>>>>>> On Thu, Jan 28, 2010 at 10:07 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>>
>>>>>>
>>>>>>             
>>>>>>> I crated 3 instances of snd-dummy, this worked. I assigned then Instance
>>>>>>> #2 to the Skype accounts. Still no sound.
>>>>>>> On the frist call there is one way audio, on the following calls there
>>>>>>> is no audio at all.
>>>>>>> This is weird.
>>>>>>>
>>>>>>> Best regards
>>>>>>> Peter
>>>>>>>
>>>>>>> Giovanni Maruzzelli schrieb:
>>>>>>>
>>>>>>>
>>>>>>>               
>>>>>>>> Ciao Peter,
>>>>>>>>
>>>>>>>> Never tested on Debian 5.
>>>>>>>>
>>>>>>>> When you write "same problem" you are referring to the audio going one
>>>>>>>> way only (btw, which way?) with the custom audio driver?
>>>>>>>>
>>>>>>>> Have you tried with multiple instances of the regular Debian
>>>>>>>> snd-dummy, as I wrote in a mail before?
>>>>>>>>
>>>>>>>> -gm
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> On Thu, Jan 28, 2010 at 9:00 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>                 
>>>>>>>>> Hello Giovanni,
>>>>>>>>>
>>>>>>>>> I did so but the same problem again.
>>>>>>>>>
>>>>>>>>> Did you ever test in on Debian 5.0?
>>>>>>>>>
>>>>>>>>> Best reards
>>>>>>>>> Peter
>>>>>>>>>
>>>>>>>>> Giovanni Maruzzelli schrieb:
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>                   
>>>>>>>>>> good, so you have only one sound device, the right one.
>>>>>>>>>>
>>>>>>>>>> Use the one with hw:0 in the window that skype gives you to set sound devices
>>>>>>>>>>
>>>>>>>>>> -gm
>>>>>>>>>>
>>>>>>>>>> On Wed, Jan 27, 2010 at 7:04 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>                     
>>>>>>>>>>> I installed alsa-utile,
>>>>>>>>>>>
>>>>>>>>>>> now I get:
>>>>>>>>>>>
>>>>>>>>>>> skype:/var/cache/apt/archives# aplay -l
>>>>>>>>>>> **** List of PLAYBACK Hardware Devices ****
>>>>>>>>>>> card 0: Dummy [Dummy], device 0: Dummy PCM [Dummy PCM]
>>>>>>>>>>>  Subdevices: 127/128
>>>>>>>>>>>  Subdevice #0: subdevice #0
>>>>>>>>>>>  Subdevice #1: subdevice #1
>>>>>>>>>>>  Subdevice #2: subdevice #2
>>>>>>>>>>>  Subdevice #3: subdevice #3
>>>>>>>>>>>  Subdevice #4: subdevice #4
>>>>>>>>>>>  Subdevice #5: subdevice #5
>>>>>>>>>>>  Subdevice #6: subdevice #6
>>>>>>>>>>>  Subdevice #7: subdevice #7
>>>>>>>>>>>  Subdevice #8: subdevice #8
>>>>>>>>>>>  Subdevice #9: subdevice #9
>>>>>>>>>>>  Subdevice #10: subdevice #10
>>>>>>>>>>>  Subdevice #11: subdevice #11
>>>>>>>>>>>  Subdevice #12: subdevice #12
>>>>>>>>>>>  Subdevice #13: subdevice #13
>>>>>>>>>>>  Subdevice #14: subdevice #14
>>>>>>>>>>>  Subdevice #15: subdevice #15
>>>>>>>>>>>  Subdevice #16: subdevice #16
>>>>>>>>>>>  Subdevice #17: subdevice #17
>>>>>>>>>>>  Subdevice #18: subdevice #18
>>>>>>>>>>>  Subdevice #19: subdevice #19
>>>>>>>>>>>  Subdevice #20: subdevice #20
>>>>>>>>>>>  Subdevice #21: subdevice #21
>>>>>>>>>>>  Subdevice #22: subdevice #22
>>>>>>>>>>>  Subdevice #23: subdevice #23
>>>>>>>>>>>  Subdevice #24: subdevice #24
>>>>>>>>>>>  Subdevice #25: subdevice #25
>>>>>>>>>>>  Subdevice #26: subdevice #26
>>>>>>>>>>>  Subdevice #27: subdevice #27
>>>>>>>>>>>  Subdevice #28: subdevice #28
>>>>>>>>>>>  Subdevice #29: subdevice #29
>>>>>>>>>>>  Subdevice #30: subdevice #30
>>>>>>>>>>>  Subdevice #31: subdevice #31
>>>>>>>>>>>  Subdevice #32: subdevice #32
>>>>>>>>>>>  Subdevice #33: subdevice #33
>>>>>>>>>>>  Subdevice #34: subdevice #34
>>>>>>>>>>>  Subdevice #35: subdevice #35
>>>>>>>>>>>  Subdevice #36: subdevice #36
>>>>>>>>>>>  Subdevice #37: subdevice #37
>>>>>>>>>>>  Subdevice #38: subdevice #38
>>>>>>>>>>>  Subdevice #39: subdevice #39
>>>>>>>>>>>  Subdevice #40: subdevice #40
>>>>>>>>>>>  Subdevice #41: subdevice #41
>>>>>>>>>>>  Subdevice #42: subdevice #42
>>>>>>>>>>>  Subdevice #43: subdevice #43
>>>>>>>>>>>  Subdevice #44: subdevice #44
>>>>>>>>>>>  Subdevice #45: subdevice #45
>>>>>>>>>>>  Subdevice #46: subdevice #46
>>>>>>>>>>>  Subdevice #47: subdevice #47
>>>>>>>>>>>  Subdevice #48: subdevice #48
>>>>>>>>>>>  Subdevice #49: subdevice #49
>>>>>>>>>>>  Subdevice #50: subdevice #50
>>>>>>>>>>>  Subdevice #51: subdevice #51
>>>>>>>>>>>  Subdevice #52: subdevice #52
>>>>>>>>>>>  Subdevice #53: subdevice #53
>>>>>>>>>>>  Subdevice #54: subdevice #54
>>>>>>>>>>>  Subdevice #55: subdevice #55
>>>>>>>>>>>  Subdevice #56: subdevice #56
>>>>>>>>>>>  Subdevice #57: subdevice #57
>>>>>>>>>>>  Subdevice #58: subdevice #58
>>>>>>>>>>>  Subdevice #59: subdevice #59
>>>>>>>>>>>  Subdevice #60: subdevice #60
>>>>>>>>>>>  Subdevice #61: subdevice #61
>>>>>>>>>>>  Subdevice #62: subdevice #62
>>>>>>>>>>>  Subdevice #63: subdevice #63
>>>>>>>>>>>  Subdevice #64: subdevice #64
>>>>>>>>>>>  Subdevice #65: subdevice #65
>>>>>>>>>>>  Subdevice #66: subdevice #66
>>>>>>>>>>>  Subdevice #67: subdevice #67
>>>>>>>>>>>  Subdevice #68: subdevice #68
>>>>>>>>>>>  Subdevice #69: subdevice #69
>>>>>>>>>>>  Subdevice #70: subdevice #70
>>>>>>>>>>>  Subdevice #71: subdevice #71
>>>>>>>>>>>  Subdevice #72: subdevice #72
>>>>>>>>>>>  Subdevice #73: subdevice #73
>>>>>>>>>>>  Subdevice #74: subdevice #74
>>>>>>>>>>>  Subdevice #75: subdevice #75
>>>>>>>>>>>  Subdevice #76: subdevice #76
>>>>>>>>>>>  Subdevice #77: subdevice #77
>>>>>>>>>>>  Subdevice #78: subdevice #78
>>>>>>>>>>>  Subdevice #79: subdevice #79
>>>>>>>>>>>  Subdevice #80: subdevice #80
>>>>>>>>>>>  Subdevice #81: subdevice #81
>>>>>>>>>>>  Subdevice #82: subdevice #82
>>>>>>>>>>>  Subdevice #83: subdevice #83
>>>>>>>>>>>  Subdevice #84: subdevice #84
>>>>>>>>>>>  Subdevice #85: subdevice #85
>>>>>>>>>>>  Subdevice #86: subdevice #86
>>>>>>>>>>>  Subdevice #87: subdevice #87
>>>>>>>>>>>  Subdevice #88: subdevice #88
>>>>>>>>>>>  Subdevice #89: subdevice #89
>>>>>>>>>>>  Subdevice #90: subdevice #90
>>>>>>>>>>>  Subdevice #91: subdevice #91
>>>>>>>>>>>  Subdevice #92: subdevice #92
>>>>>>>>>>>  Subdevice #93: subdevice #93
>>>>>>>>>>>  Subdevice #94: subdevice #94
>>>>>>>>>>>  Subdevice #95: subdevice #95
>>>>>>>>>>>  Subdevice #96: subdevice #96
>>>>>>>>>>>  Subdevice #97: subdevice #97
>>>>>>>>>>>  Subdevice #98: subdevice #98
>>>>>>>>>>>  Subdevice #99: subdevice #99
>>>>>>>>>>>  Subdevice #100: subdevice #100
>>>>>>>>>>>  Subdevice #101: subdevice #101
>>>>>>>>>>>  Subdevice #102: subdevice #102
>>>>>>>>>>>  Subdevice #103: subdevice #103
>>>>>>>>>>>  Subdevice #104: subdevice #104
>>>>>>>>>>>  Subdevice #105: subdevice #105
>>>>>>>>>>>  Subdevice #106: subdevice #106
>>>>>>>>>>>  Subdevice #107: subdevice #107
>>>>>>>>>>>  Subdevice #108: subdevice #108
>>>>>>>>>>>  Subdevice #109: subdevice #109
>>>>>>>>>>>  Subdevice #110: subdevice #110
>>>>>>>>>>>  Subdevice #111: subdevice #111
>>>>>>>>>>>  Subdevice #112: subdevice #112
>>>>>>>>>>>  Subdevice #113: subdevice #113
>>>>>>>>>>>  Subdevice #114: subdevice #114
>>>>>>>>>>>  Subdevice #115: subdevice #115
>>>>>>>>>>>  Subdevice #116: subdevice #116
>>>>>>>>>>>  Subdevice #117: subdevice #117
>>>>>>>>>>>  Subdevice #118: subdevice #118
>>>>>>>>>>>  Subdevice #119: subdevice #119
>>>>>>>>>>>  Subdevice #120: subdevice #120
>>>>>>>>>>>  Subdevice #121: subdevice #121
>>>>>>>>>>>  Subdevice #122: subdevice #122
>>>>>>>>>>>  Subdevice #123: subdevice #123
>>>>>>>>>>>  Subdevice #124: subdevice #124
>>>>>>>>>>>  Subdevice #125: subdevice #125
>>>>>>>>>>>  Subdevice #126: subdevice #126
>>>>>>>>>>>  Subdevice #127: subdevice #127
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> Peter P GMX schrieb:
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>                       
>>>>>>>>>>>> Her's the output:
>>>>>>>>>>>>
>>>>>>>>>>>> skype:~# aplay -l
>>>>>>>>>>>> bash: aplay: command not found
>>>>>>>>>>>>
>>>>>>>>>>>> Giovanni Maruzzelli schrieb:
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>                         
>>>>>>>>>>>>> I don't think you got two snd-dummy loaded (but maybe yes)
>>>>>>>>>>>>> what's the output of:
>>>>>>>>>>>>>
>>>>>>>>>>>>> aplay -l
>>>>>>>>>>>>>
>>>>>>>>>>>>> ?
>>>>>>>>>>>>>
>>>>>>>>>>>>> If instead you are referring to the choices that skype clients offers
>>>>>>>>>>>>> you in the "set audio devices" window, choose Dummy PCM (hw0:0)
>>>>>>>>>>>>>
>>>>>>>>>>>>> Eg: not the "default", but the "hardware" one
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>> On Wed, Jan 27, 2010 at 5:58 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>                           
>>>>>>>>>>>>>> Thanks Giovanni,
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> I think there may be the problem, that I have 2 sound devices now:
>>>>>>>>>>>>>> - Dummy PCM (hw0:0) (this is from debian install)
>>>>>>>>>>>>>> - Dummy PCM Default Audio device (defauzlt: CARD=Dummy) (this is new
>>>>>>>>>>>>>> since I compiled alsa newly)
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> I tried both, but both do not work. How do I get rid of the old alsa device?
>>>>>>>>>>>>>> By the way: I uninstalled Alsa before I installed the new driver
>>>>>>>>>>>>>> (apt-get remove alsa-utils alsa-base).
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> Best regards
>>>>>>>>>>>>>> Peter
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> Giovanni Maruzzelli schrieb:
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>                             
>>>>>>>>>>>>>>> This warning is harmless:
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> Jan 27 15:28:50 skype kernel: [ 3994.795786] process `skype' is using
>>>>>>>>>>>>>>> obsolete setsockopt SO_BSDCOMPAT
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> On Wed, Jan 27, 2010 at 4:26 PM, Giovanni Maruzzelli
>>>>>>>>>>>>>>> <gmaruzz at celliax.org> wrote:
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>                               
>>>>>>>>>>>>>>>> Ciao Peter
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> one instance of snd-dummy "customized" is enough for 64 instances of
>>>>>>>>>>>>>>>> skype clients, no need (and do not works) with more instances of
>>>>>>>>>>>>>>>> snd-dummy-customized.
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> Maybe you got the one-way problem because of kernel at 250HZ (don't
>>>>>>>>>>>>>>>> know). It uses to works well on a tickless kernel at 100HZ (eg: ubuntu
>>>>>>>>>>>>>>>> 8.04).
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> Or maybe you have to check and modify which sound devices the skype
>>>>>>>>>>>>>>>> clients are using (try to check that with snd-summy-custom loaded,
>>>>>>>>>>>>>>>> maybe with the ssh -X trick (as in the wiki page).
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> To load more than one snd-dummy-original (the non modified one), you
>>>>>>>>>>>>>>>> do this with the modprobe command, as in:
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> rmmod snd-dummy
>>>>>>>>>>>>>>>> modprobe snd-dummy enable=1,1,1
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> this command will enable three instances of snd-dummy original, so
>>>>>>>>>>>>>>>> you'll have three fake soundcards, and you'll have to setup each group
>>>>>>>>>>>>>>>> of 8 skype instances to use sound devices from one fake soundcard, RG:
>>>>>>>>>>>>>>>> no more than 8 skype client instances can use one instance of fake
>>>>>>>>>>>>>>>> soundcard.
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> Also, please update the mod_skypiax code (svn up in its directory) I
>>>>>>>>>>>>>>>> just committed some improvements.
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> If you have any other doubts, or need more info, don't hesitate to
>>>>>>>>>>>>>>>> write the mailing list again,
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> ciao for now,
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> -giovanni
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> On Wed, Jan 27, 2010 at 4:01 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>                                 
>>>>>>>>>>>>>>>>> I have mod_skypiax working nicely so far with 2 Skype channels. Thanks
>>>>>>>>>>>>>>>>> to all contributors, excellent work!
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> In order to have more than 8 channels working, I have followed the
>>>>>>>>>>>>>>>>> instructions in
>>>>>>>>>>>>>>>>> http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk
>>>>>>>>>>>>>>>>> and compiled alsa-driver-1.0.20 with the modified dummy.c file. (System
>>>>>>>>>>>>>>>>> ist Debian 5.0R3)
>>>>>>>>>>>>>>>>> It compiled well however when I start snd-dummy I only have
>>>>>>>>>>>>>>>>> one-way-audio and my logs show
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> Jan 27 15:28:41 skype kernel: [ 3984.318403] snd-dummy skypiax driver,
>>>>>>>>>>>>>>>>> /usr/src/alsa-driver-1.0.20/drivers/../alsa-kernel/drivers/dummy.c:920
>>>>>>>>>>>>>>>>> working on a machine with 250HZ kernel
>>>>>>>>>>>>>>>>> Jan 27 15:28:50 skype kernel: [ 3994.795786] process `skype' is using
>>>>>>>>>>>>>>>>> obsolete setsockopt SO_BSDCOMPAT
>>>>>>>>>>>>>>>>> Jan 27 15:28:56 skype kernel: [ 4005.289907] __ratelimit: 490 messages
>>>>>>>>>>>>>>>>> suppressed
>>>>>>>>>>>>>>>>> Jan 27 15:29:01 skype kernel: [ 4012.458310] process `skype' is using
>>>>>>>>>>>>>>>>> obsolete setsockopt SO_BSDCOMPAT
>>>>>>>>>>>>>>>>> Jan 27 15:29:01 skype kernel: [ 4013.326290] __ratelimit: 499 messages
>>>>>>>>>>>>>>>>> suppressed
>>>>>>>>>>>>>>>>> If I reinstall alsa from deb everything sworks fine again (of course
>>>>>>>>>>>>>>>>> with the current limitations).
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> First question: Has anybody had this issue before? How can I solve this?
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> Second question:
>>>>>>>>>>>>>>>>> As I do not need 64 channels or more: how do I manage, that Skype
>>>>>>>>>>>>>>>>> instances 9..15 use a second instance of snd-dummy as addressed in the wiki?
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> Best regards
>>>>>>>>>>>>>>>>> Peter
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>                                   
>>>>>>>>>>>>>>>> --
>>>>>>>>>>>>>>>> Sincerely,
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> Giovanni Maruzzelli
>>>>>>>>>>>>>>>> Cell : +39-347-2665618
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>                                 
>>>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>                             
>>>>>>>>>>> _______________________________________________
>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>                       
>>>>>>>>> _______________________________________________
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>                   
>>>>>>>>                 
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>               
>>>>>>
>>>>>>             
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>>           
>>>> --
>>>> Ghulam Mustafa
>>>> cell: +92 333.611.7681
>>>> sip: cyrenity at ekiga.net
>>>> mail: mustafa.pk at gmail.com
>>>> web: cyrenity.wordpress.com
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>         
>>>
>>>
>>>       
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>     
>
>
>
>   



From tayeb.meftah at gmail.com  Fri Feb  5 13:05:46 2010
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Fri, 05 Feb 2010 22:05:46 +0100
Subject: [Freeswitch-users] Switch Security
In-Reply-To: <4B6C351E.6080608@acsol.net>
References: <4B6C351E.6080608@acsol.net>
Message-ID: <4B6C882A.4050101@gmail.com>

hi
you can use acl with mod_xml_curl
thanks
Le 05/02/2010 16:11, John a ?crit :
> Freeswitch is to be used by phones external to my lan. Many of the
> phones will be coming from DSL connections without static IP. I have
> disabled the default password. What is a good procedure to secure the
> switch from non-customers registering? I know that I could use an ACL;
> however it's difficult with all the non-static users.
>
> Thanks John
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>    




From tayeb.meftah at gmail.com  Fri Feb  5 13:15:44 2010
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Fri, 05 Feb 2010 22:15:44 +0100
Subject: [Freeswitch-users] Simple IAX2 setup - help with
 converting	from asterisk to freeswitch
In-Reply-To: <1265385834.12871.83.camel@local.freepabx.com>
References: <91093d24f2166d220acf2ab19115f95d.squirrel@www.webcontracts.co.uk>	<E1E3F6EF-1A1E-4891-B371-7A3BF0988E95@jerris.com>	<d2976552e9ba0ce1ed095b7cef22e1ad.squirrel@www.webcontracts.co.uk>	<b39273531002050657x4f0f921euce8a683b8018ed63@mail.gmail.com>
	<1265385834.12871.83.camel@local.freepabx.com>
Message-ID: <4B6C8A80.9050700@gmail.com>

hi,
iax2 is secure
but, is not a good idea to avoid rtp and pass all packet including audio 
and signalisation troug the same port
and digium added some change to the IAX2 protocol so freeswitch is not 
up to date
no one want to update the iax2 stack in fs
so fs mod_iax have bean removedfrom the trunk
Le 05/02/2010 17:03, David Knell a ?crit :
> There's a fairly simple solution to IAX needs, which is to run Asterisk,
> probably on the same box, as a protocol converter - you just need to
> tell it to use a non-standard port in sip.conf so that it doesn't clash
> with FreeSWITCH.
>
> --Dave
>
>    
>> the lib that we used to provide iax support is pretty much abandonware
>> (no longer updated) and newer iax implementations (like latest
>> asterisk) can cause it to crash.  There are no license compatible iax
>> implementations that work, so.. mod_iax has been moved to the
>> unsupported column.
>>
>>
>> Default passwords -- that is a single var in vars.xml that controls
>> the passwords.
>>
>>
>> number ranges - up to you.  The sample configs supplied are just that,
>> samples.  I use a smaller range personally.
>>
>> On Fri, Feb 5, 2010 at 7:53 AM, Matthew Law<matt at webcontracts.co.uk>
>> wrote:
>>          Why is that? - a lot of web pages I have read claim that IAX
>>          is more
>>          secure and efficient.  I have no problem with using SIP
>>          whatsoever and it
>>          certainly appears to be ubiquitous.  I am a complete newcomer
>>          to VoIP and
>>          I am trying to do this as securely as possible since I want to
>>          run
>>          freeswitch on a Xen VPS which will be visible on the internet.
>>
>>          Anyway, since my first question, I have worked my way through
>>          the wiki,
>>          read a lot of example configs and made some sense of the
>>          docs.  I now have
>>          a very basic config working (with SIP) on a local vmware image
>>          using the
>>          'quick and dirty' Makefile method.  I removed all of the
>>          example configs
>>          from the xml files (those default extensions and passwords
>>          scared me) and
>>          added just one for extension 1000, plus my dialplan and
>>          inbound/outbound
>>          settings.
>>
>>          One question: is there any reason not to use a smaller
>>          extension number
>>          range, like 200-210, for example?
>>
>>          Now to figure out how to get time based roaming working...
>>
>>
>>          Thanks,
>>
>>          Matt.
>>
>>
>>          On Fri, February 5, 2010 6:43 am, Michael Jerris wrote:
>>          >  iax2 support has been removed from FreeSWITCH in current
>>          trunk and will
>>          >  not be in the 1.0.5 release.
>>          >
>>
>>
>>          >  Mike
>>
>>
>>          _______________________________________________
>>          FreeSWITCH-users mailing list
>>          FreeSWITCH-users at lists.freeswitch.org
>>          http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>          UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>          http://www.freeswitch.org
>>
>>
>>
>>
>> -- 
>> -Rupa
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>      
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>    




From gmaruzz at celliax.org  Fri Feb  5 13:19:24 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Fri, 5 Feb 2010 22:19:24 +0100
Subject: [Freeswitch-users] mod_skypiax - help needed with modified
	snd-dummy (one way audio)
In-Reply-To: <4B6C868B.3040406@gmx.net>
References: <4B60555B.2020004@gmx.net>
	<7b197bef1001281210u43a907edi13afe794cf0e1a2e@mail.gmail.com> 
	<4B61FCAB.5040707@gmx.net>
	<7b197bef1001281341h7ee2c58fj79b3886630901f29@mail.gmail.com> 
	<4B62C962.7000601@gmx.net>
	<8213d6071001290403g2c08fe39gd92eb0446bb6485a@mail.gmail.com> 
	<7b197bef1001290520x37179cb1o63273410e9ef5bc@mail.gmail.com> 
	<4B6C1CCB.4080606@gmx.net>
	<7b197bef1002050536j7ea09684v913434296ae52b97@mail.gmail.com> 
	<4B6C868B.3040406@gmx.net>
Message-ID: <7b197bef1002051319l2fcb5617p1691692f58c65372@mail.gmail.com>

that's not at all a fatal error.
I believe it works the same.
Are you sure it does not work?

-gm


On Fri, Feb 5, 2010 at 9:58 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
> Hello Giovanni,
>
> I am now at the point to install Skype. But there is only an Intrepid
> version available (no 8.04 version).
> The current verison crashed on 8.04x because of dbus error.
> ? ?process 8408: D-Bus library appears to be incorrectly set up; failed
> to read machine uuid: Failed to open "/var/lib/dbus/machine-id": No such
> file or directory
> ? ?See the manual page for dbus-uuidgen to correct this issue.
> ? ?/usr/bin/skype: symbol lookup error: /usr/bin/skype: undefined
> symbol: _ZN10QBoxLayout13addSpacerItemEP11QSpacerIte
>
> Any idea where I can download the older version for 8.04?
>
> Best regards
> Peter
>
>
> Giovanni Maruzzelli schrieb:
>> Ciao Peter,
>>
>> I would use ubuntu 8.04 (hardy) LTS server 64bit or CentOS 5.4 64bit.
>>
>> -giovanni
>>
>> On Fri, Feb 5, 2010 at 2:27 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>
>>> Hello Giovanni,
>>>
>>> as I couldn't even get skype again working again with the standard alsa
>>> driver, I would like to setup the machine from scratch based on a
>>> working machine.
>>> The latest errors I received from Skype was:
>>> ?snd_pcm_avail_update() returned a value that is exceptionally large:
>>> 715706624 bytes (3727638 ms).
>>> ?Most likely this is a bug in the ALSA driver. Please report this issue
>>> to the ALSA developers.
>>> I think that may be the reason for one-way-audio.
>>>
>>> For setting up my machine from scratch, please advise:
>>> - which OS you are you using und recommending exactly?
>>> - I would like to use 64bit OS in order to use 8GB of memory, does this
>>> work?
>>> - any other hints?
>>>
>>> Best regards
>>> Peter
>>>
>>> Giovanni Maruzzelli schrieb:
>>>
>>>> Peter,
>>>>
>>>> Can you connect on IRC (irc.freenode.net #freeswitch)? I'm gmaruzz there.
>>>>
>>>> Can you restate your problems? I've lost connection :)
>>>>
>>>> with snd-dummy custom you can create *one only* snd-dummy instance, so
>>>> *one only* fake soundcard. If you create more, will not work. But with
>>>> that one fake soundcard you can use 64 skype client instances, all
>>>> with the same soundcard hardware device (hw:n).
>>>>
>>>> with original snd-dummy you can create a max of 8 instances, so 8 fake
>>>> soundcards, and with each fake soundcard you can ?use a max of 8 skype
>>>> client instances.
>>>>
>>>> use the hardware devices, not the default devices (use the "hw:n")
>>>>
>>>> -giovanni
>>>>
>>>> On Fri, Jan 29, 2010 at 1:03 PM, Ghulam Mustafa <mustafa.pk at gmail.com> wrote:
>>>>
>>>>
>>>>> did you enable debug mode while compiling custom snd-dummy? if ?yes
>>>>> try re-compiling with debug mode disabled.
>>>>>
>>>>> -m
>>>>>
>>>>> On Fri, Jan 29, 2010 at 4:41 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>
>>>>>
>>>>>> I now reinstalled the original sound drivers
>>>>>> Unfortunaltely the sound problems remain, not that worse but they are there:
>>>>>> Audio is still (almost) one way. Almost means:
>>>>>>
>>>>>> ? ?* SIP -> Skype ok
>>>>>> ? ?* Skype=> SIP I hear only some scratching on very loud audio
>>>>>>
>>>>>> Could it be a volume problem? But snd-dummy should have no volume
>>>>>> properties, right?
>>>>>>
>>>>>> Best regards
>>>>>> Peter
>>>>>>
>>>>>>
>>>>>> Giovanni Maruzzelli schrieb:
>>>>>>
>>>>>>
>>>>>>> with three instances you will assign the hw:0 device to skype client
>>>>>>> 0...7, hw:1 to skype client 8...15 and hw:2 to skype client 16...23.
>>>>>>> Must work. Pay attention to assign the same device name to all devices
>>>>>>> needed by a skype instance (sound devices window): playback, capture
>>>>>>> AND ring.
>>>>>>>
>>>>>>> Or maybe is a bug of ALSA on Debian...
>>>>>>>
>>>>>>> -giovanni
>>>>>>>
>>>>>>> On Thu, Jan 28, 2010 at 10:07 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>> I crated 3 instances of snd-dummy, this worked. I assigned then Instance
>>>>>>>> #2 to the Skype accounts. Still no sound.
>>>>>>>> On the frist call there is one way audio, on the following calls there
>>>>>>>> is no audio at all.
>>>>>>>> This is weird.
>>>>>>>>
>>>>>>>> Best regards
>>>>>>>> Peter
>>>>>>>>
>>>>>>>> Giovanni Maruzzelli schrieb:
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>> Ciao Peter,
>>>>>>>>>
>>>>>>>>> Never tested on Debian 5.
>>>>>>>>>
>>>>>>>>> When you write "same problem" you are referring to the audio going one
>>>>>>>>> way only (btw, which way?) with the custom audio driver?
>>>>>>>>>
>>>>>>>>> Have you tried with multiple instances of the regular Debian
>>>>>>>>> snd-dummy, as I wrote in a mail before?
>>>>>>>>>
>>>>>>>>> -gm
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> On Thu, Jan 28, 2010 at 9:00 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>> Hello Giovanni,
>>>>>>>>>>
>>>>>>>>>> I did so but the same problem again.
>>>>>>>>>>
>>>>>>>>>> Did you ever test in on Debian 5.0?
>>>>>>>>>>
>>>>>>>>>> Best reards
>>>>>>>>>> Peter
>>>>>>>>>>
>>>>>>>>>> Giovanni Maruzzelli schrieb:
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>> good, so you have only one sound device, the right one.
>>>>>>>>>>>
>>>>>>>>>>> Use the one with hw:0 in the window that skype gives you to set sound devices
>>>>>>>>>>>
>>>>>>>>>>> -gm
>>>>>>>>>>>
>>>>>>>>>>> On Wed, Jan 27, 2010 at 7:04 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>> I installed alsa-utile,
>>>>>>>>>>>>
>>>>>>>>>>>> now I get:
>>>>>>>>>>>>
>>>>>>>>>>>> skype:/var/cache/apt/archives# aplay -l
>>>>>>>>>>>> **** List of PLAYBACK Hardware Devices ****
>>>>>>>>>>>> card 0: Dummy [Dummy], device 0: Dummy PCM [Dummy PCM]
>>>>>>>>>>>> ?Subdevices: 127/128
>>>>>>>>>>>> ?Subdevice #0: subdevice #0
>>>>>>>>>>>> ?Subdevice #1: subdevice #1
>>>>>>>>>>>> ?Subdevice #2: subdevice #2
>>>>>>>>>>>> ?Subdevice #3: subdevice #3
>>>>>>>>>>>> ?Subdevice #4: subdevice #4
>>>>>>>>>>>> ?Subdevice #5: subdevice #5
>>>>>>>>>>>> ?Subdevice #6: subdevice #6
>>>>>>>>>>>> ?Subdevice #7: subdevice #7
>>>>>>>>>>>> ?Subdevice #8: subdevice #8
>>>>>>>>>>>> ?Subdevice #9: subdevice #9
>>>>>>>>>>>> ?Subdevice #10: subdevice #10
>>>>>>>>>>>> ?Subdevice #11: subdevice #11
>>>>>>>>>>>> ?Subdevice #12: subdevice #12
>>>>>>>>>>>> ?Subdevice #13: subdevice #13
>>>>>>>>>>>> ?Subdevice #14: subdevice #14
>>>>>>>>>>>> ?Subdevice #15: subdevice #15
>>>>>>>>>>>> ?Subdevice #16: subdevice #16
>>>>>>>>>>>> ?Subdevice #17: subdevice #17
>>>>>>>>>>>> ?Subdevice #18: subdevice #18
>>>>>>>>>>>> ?Subdevice #19: subdevice #19
>>>>>>>>>>>> ?Subdevice #20: subdevice #20
>>>>>>>>>>>> ?Subdevice #21: subdevice #21
>>>>>>>>>>>> ?Subdevice #22: subdevice #22
>>>>>>>>>>>> ?Subdevice #23: subdevice #23
>>>>>>>>>>>> ?Subdevice #24: subdevice #24
>>>>>>>>>>>> ?Subdevice #25: subdevice #25
>>>>>>>>>>>> ?Subdevice #26: subdevice #26
>>>>>>>>>>>> ?Subdevice #27: subdevice #27
>>>>>>>>>>>> ?Subdevice #28: subdevice #28
>>>>>>>>>>>> ?Subdevice #29: subdevice #29
>>>>>>>>>>>> ?Subdevice #30: subdevice #30
>>>>>>>>>>>> ?Subdevice #31: subdevice #31
>>>>>>>>>>>> ?Subdevice #32: subdevice #32
>>>>>>>>>>>> ?Subdevice #33: subdevice #33
>>>>>>>>>>>> ?Subdevice #34: subdevice #34
>>>>>>>>>>>> ?Subdevice #35: subdevice #35
>>>>>>>>>>>> ?Subdevice #36: subdevice #36
>>>>>>>>>>>> ?Subdevice #37: subdevice #37
>>>>>>>>>>>> ?Subdevice #38: subdevice #38
>>>>>>>>>>>> ?Subdevice #39: subdevice #39
>>>>>>>>>>>> ?Subdevice #40: subdevice #40
>>>>>>>>>>>> ?Subdevice #41: subdevice #41
>>>>>>>>>>>> ?Subdevice #42: subdevice #42
>>>>>>>>>>>> ?Subdevice #43: subdevice #43
>>>>>>>>>>>> ?Subdevice #44: subdevice #44
>>>>>>>>>>>> ?Subdevice #45: subdevice #45
>>>>>>>>>>>> ?Subdevice #46: subdevice #46
>>>>>>>>>>>> ?Subdevice #47: subdevice #47
>>>>>>>>>>>> ?Subdevice #48: subdevice #48
>>>>>>>>>>>> ?Subdevice #49: subdevice #49
>>>>>>>>>>>> ?Subdevice #50: subdevice #50
>>>>>>>>>>>> ?Subdevice #51: subdevice #51
>>>>>>>>>>>> ?Subdevice #52: subdevice #52
>>>>>>>>>>>> ?Subdevice #53: subdevice #53
>>>>>>>>>>>> ?Subdevice #54: subdevice #54
>>>>>>>>>>>> ?Subdevice #55: subdevice #55
>>>>>>>>>>>> ?Subdevice #56: subdevice #56
>>>>>>>>>>>> ?Subdevice #57: subdevice #57
>>>>>>>>>>>> ?Subdevice #58: subdevice #58
>>>>>>>>>>>> ?Subdevice #59: subdevice #59
>>>>>>>>>>>> ?Subdevice #60: subdevice #60
>>>>>>>>>>>> ?Subdevice #61: subdevice #61
>>>>>>>>>>>> ?Subdevice #62: subdevice #62
>>>>>>>>>>>> ?Subdevice #63: subdevice #63
>>>>>>>>>>>> ?Subdevice #64: subdevice #64
>>>>>>>>>>>> ?Subdevice #65: subdevice #65
>>>>>>>>>>>> ?Subdevice #66: subdevice #66
>>>>>>>>>>>> ?Subdevice #67: subdevice #67
>>>>>>>>>>>> ?Subdevice #68: subdevice #68
>>>>>>>>>>>> ?Subdevice #69: subdevice #69
>>>>>>>>>>>> ?Subdevice #70: subdevice #70
>>>>>>>>>>>> ?Subdevice #71: subdevice #71
>>>>>>>>>>>> ?Subdevice #72: subdevice #72
>>>>>>>>>>>> ?Subdevice #73: subdevice #73
>>>>>>>>>>>> ?Subdevice #74: subdevice #74
>>>>>>>>>>>> ?Subdevice #75: subdevice #75
>>>>>>>>>>>> ?Subdevice #76: subdevice #76
>>>>>>>>>>>> ?Subdevice #77: subdevice #77
>>>>>>>>>>>> ?Subdevice #78: subdevice #78
>>>>>>>>>>>> ?Subdevice #79: subdevice #79
>>>>>>>>>>>> ?Subdevice #80: subdevice #80
>>>>>>>>>>>> ?Subdevice #81: subdevice #81
>>>>>>>>>>>> ?Subdevice #82: subdevice #82
>>>>>>>>>>>> ?Subdevice #83: subdevice #83
>>>>>>>>>>>> ?Subdevice #84: subdevice #84
>>>>>>>>>>>> ?Subdevice #85: subdevice #85
>>>>>>>>>>>> ?Subdevice #86: subdevice #86
>>>>>>>>>>>> ?Subdevice #87: subdevice #87
>>>>>>>>>>>> ?Subdevice #88: subdevice #88
>>>>>>>>>>>> ?Subdevice #89: subdevice #89
>>>>>>>>>>>> ?Subdevice #90: subdevice #90
>>>>>>>>>>>> ?Subdevice #91: subdevice #91
>>>>>>>>>>>> ?Subdevice #92: subdevice #92
>>>>>>>>>>>> ?Subdevice #93: subdevice #93
>>>>>>>>>>>> ?Subdevice #94: subdevice #94
>>>>>>>>>>>> ?Subdevice #95: subdevice #95
>>>>>>>>>>>> ?Subdevice #96: subdevice #96
>>>>>>>>>>>> ?Subdevice #97: subdevice #97
>>>>>>>>>>>> ?Subdevice #98: subdevice #98
>>>>>>>>>>>> ?Subdevice #99: subdevice #99
>>>>>>>>>>>> ?Subdevice #100: subdevice #100
>>>>>>>>>>>> ?Subdevice #101: subdevice #101
>>>>>>>>>>>> ?Subdevice #102: subdevice #102
>>>>>>>>>>>> ?Subdevice #103: subdevice #103
>>>>>>>>>>>> ?Subdevice #104: subdevice #104
>>>>>>>>>>>> ?Subdevice #105: subdevice #105
>>>>>>>>>>>> ?Subdevice #106: subdevice #106
>>>>>>>>>>>> ?Subdevice #107: subdevice #107
>>>>>>>>>>>> ?Subdevice #108: subdevice #108
>>>>>>>>>>>> ?Subdevice #109: subdevice #109
>>>>>>>>>>>> ?Subdevice #110: subdevice #110
>>>>>>>>>>>> ?Subdevice #111: subdevice #111
>>>>>>>>>>>> ?Subdevice #112: subdevice #112
>>>>>>>>>>>> ?Subdevice #113: subdevice #113
>>>>>>>>>>>> ?Subdevice #114: subdevice #114
>>>>>>>>>>>> ?Subdevice #115: subdevice #115
>>>>>>>>>>>> ?Subdevice #116: subdevice #116
>>>>>>>>>>>> ?Subdevice #117: subdevice #117
>>>>>>>>>>>> ?Subdevice #118: subdevice #118
>>>>>>>>>>>> ?Subdevice #119: subdevice #119
>>>>>>>>>>>> ?Subdevice #120: subdevice #120
>>>>>>>>>>>> ?Subdevice #121: subdevice #121
>>>>>>>>>>>> ?Subdevice #122: subdevice #122
>>>>>>>>>>>> ?Subdevice #123: subdevice #123
>>>>>>>>>>>> ?Subdevice #124: subdevice #124
>>>>>>>>>>>> ?Subdevice #125: subdevice #125
>>>>>>>>>>>> ?Subdevice #126: subdevice #126
>>>>>>>>>>>> ?Subdevice #127: subdevice #127
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>> Peter P GMX schrieb:
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>> Her's the output:
>>>>>>>>>>>>>
>>>>>>>>>>>>> skype:~# aplay -l
>>>>>>>>>>>>> bash: aplay: command not found
>>>>>>>>>>>>>
>>>>>>>>>>>>> Giovanni Maruzzelli schrieb:
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>> I don't think you got two snd-dummy loaded (but maybe yes)
>>>>>>>>>>>>>> what's the output of:
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> aplay -l
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> ?
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> If instead you are referring to the choices that skype clients offers
>>>>>>>>>>>>>> you in the "set audio devices" window, choose Dummy PCM (hw0:0)
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> Eg: not the "default", but the "hardware" one
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> On Wed, Jan 27, 2010 at 5:58 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> Thanks Giovanni,
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> I think there may be the problem, that I have 2 sound devices now:
>>>>>>>>>>>>>>> - Dummy PCM (hw0:0) (this is from debian install)
>>>>>>>>>>>>>>> - Dummy PCM Default Audio device (defauzlt: CARD=Dummy) (this is new
>>>>>>>>>>>>>>> since I compiled alsa newly)
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> I tried both, but both do not work. How do I get rid of the old alsa device?
>>>>>>>>>>>>>>> By the way: I uninstalled Alsa before I installed the new driver
>>>>>>>>>>>>>>> (apt-get remove alsa-utils alsa-base).
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> Best regards
>>>>>>>>>>>>>>> Peter
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> Giovanni Maruzzelli schrieb:
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> This warning is harmless:
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> Jan 27 15:28:50 skype kernel: [ 3994.795786] process `skype' is using
>>>>>>>>>>>>>>>> obsolete setsockopt SO_BSDCOMPAT
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> On Wed, Jan 27, 2010 at 4:26 PM, Giovanni Maruzzelli
>>>>>>>>>>>>>>>> <gmaruzz at celliax.org> wrote:
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> Ciao Peter
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> one instance of snd-dummy "customized" is enough for 64 instances of
>>>>>>>>>>>>>>>>> skype clients, no need (and do not works) with more instances of
>>>>>>>>>>>>>>>>> snd-dummy-customized.
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> Maybe you got the one-way problem because of kernel at 250HZ (don't
>>>>>>>>>>>>>>>>> know). It uses to works well on a tickless kernel at 100HZ (eg: ubuntu
>>>>>>>>>>>>>>>>> 8.04).
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> Or maybe you have to check and modify which sound devices the skype
>>>>>>>>>>>>>>>>> clients are using (try to check that with snd-summy-custom loaded,
>>>>>>>>>>>>>>>>> maybe with the ssh -X trick (as in the wiki page).
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> To load more than one snd-dummy-original (the non modified one), you
>>>>>>>>>>>>>>>>> do this with the modprobe command, as in:
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> rmmod snd-dummy
>>>>>>>>>>>>>>>>> modprobe snd-dummy enable=1,1,1
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> this command will enable three instances of snd-dummy original, so
>>>>>>>>>>>>>>>>> you'll have three fake soundcards, and you'll have to setup each group
>>>>>>>>>>>>>>>>> of 8 skype instances to use sound devices from one fake soundcard, RG:
>>>>>>>>>>>>>>>>> no more than 8 skype client instances can use one instance of fake
>>>>>>>>>>>>>>>>> soundcard.
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> Also, please update the mod_skypiax code (svn up in its directory) I
>>>>>>>>>>>>>>>>> just committed some improvements.
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> If you have any other doubts, or need more info, don't hesitate to
>>>>>>>>>>>>>>>>> write the mailing list again,
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> ciao for now,
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> -giovanni
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> On Wed, Jan 27, 2010 at 4:01 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>> I have mod_skypiax working nicely so far with 2 Skype channels. Thanks
>>>>>>>>>>>>>>>>>> to all contributors, excellent work!
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>> In order to have more than 8 channels working, I have followed the
>>>>>>>>>>>>>>>>>> instructions in
>>>>>>>>>>>>>>>>>> http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk
>>>>>>>>>>>>>>>>>> and compiled alsa-driver-1.0.20 with the modified dummy.c file. (System
>>>>>>>>>>>>>>>>>> ist Debian 5.0R3)
>>>>>>>>>>>>>>>>>> It compiled well however when I start snd-dummy I only have
>>>>>>>>>>>>>>>>>> one-way-audio and my logs show
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>> Jan 27 15:28:41 skype kernel: [ 3984.318403] snd-dummy skypiax driver,
>>>>>>>>>>>>>>>>>> /usr/src/alsa-driver-1.0.20/drivers/../alsa-kernel/drivers/dummy.c:920
>>>>>>>>>>>>>>>>>> working on a machine with 250HZ kernel
>>>>>>>>>>>>>>>>>> Jan 27 15:28:50 skype kernel: [ 3994.795786] process `skype' is using
>>>>>>>>>>>>>>>>>> obsolete setsockopt SO_BSDCOMPAT
>>>>>>>>>>>>>>>>>> Jan 27 15:28:56 skype kernel: [ 4005.289907] __ratelimit: 490 messages
>>>>>>>>>>>>>>>>>> suppressed
>>>>>>>>>>>>>>>>>> Jan 27 15:29:01 skype kernel: [ 4012.458310] process `skype' is using
>>>>>>>>>>>>>>>>>> obsolete setsockopt SO_BSDCOMPAT
>>>>>>>>>>>>>>>>>> Jan 27 15:29:01 skype kernel: [ 4013.326290] __ratelimit: 499 messages
>>>>>>>>>>>>>>>>>> suppressed
>>>>>>>>>>>>>>>>>> If I reinstall alsa from deb everything sworks fine again (of course
>>>>>>>>>>>>>>>>>> with the current limitations).
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>> First question: Has anybody had this issue before? How can I solve this?
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>> Second question:
>>>>>>>>>>>>>>>>>> As I do not need 64 channels or more: how do I manage, that Skype
>>>>>>>>>>>>>>>>>> instances 9..15 use a second instance of snd-dummy as addressed in the wiki?
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>> Best regards
>>>>>>>>>>>>>>>>>> Peter
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> --
>>>>>>>>>>>>>>>>> Sincerely,
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> Giovanni Maruzzelli
>>>>>>>>>>>>>>>>> Cell : +39-347-2665618
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>> _______________________________________________
>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>>
>>>>> --
>>>>> Ghulam Mustafa
>>>>> cell: +92 333.611.7681
>>>>> sip: cyrenity at ekiga.net
>>>>> mail: mustafa.pk at gmail.com
>>>>> web: cyrenity.wordpress.com
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>>
>>>>
>>>>
>>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618



From brian at freeswitch.org  Fri Feb  5 13:23:41 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 5 Feb 2010 15:23:41 -0600
Subject: [Freeswitch-users] Simple IAX2 setup - help with
	converting	from asterisk to freeswitch
In-Reply-To: <4B6C8A80.9050700@gmail.com>
References: <91093d24f2166d220acf2ab19115f95d.squirrel@www.webcontracts.co.uk>	<E1E3F6EF-1A1E-4891-B371-7A3BF0988E95@jerris.com>	<d2976552e9ba0ce1ed095b7cef22e1ad.squirrel@www.webcontracts.co.uk>	<b39273531002050657x4f0f921euce8a683b8018ed63@mail.gmail.com>
	<1265385834.12871.83.camel@local.freepabx.com>
	<4B6C8A80.9050700@gmail.com>
Message-ID: <0C996EBC-3E28-404B-9160-3692080D6A19@freeswitch.org>

Its not that we didn't want to... nobody stepped up to help out so we had no choice but to tag it as unsupported. 

/b

On Feb 5, 2010, at 3:15 PM, Meftah Tayeb wrote:

> no one want to update the iax2 stack in fs
> so fs mod_iax have bean removedfrom the trunk




From jerry.richards at teotech.com  Fri Feb  5 13:24:13 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Fri, 5 Feb 2010 13:24:13 -0800
Subject: [Freeswitch-users] Can Freeswitch and OpenSER Co-reside
	onSameServer?
In-Reply-To: <2d9149cd1002051150i3fba0945s4332af51d261274c@mail.gmail.com>
References: <C4DC856C75094358AEDBE3FA0E242741@greyhawk.tonecommander.com><2d9149cd1002051011x3b4f548dqabe615ed5f991687@mail.gmail.com><0203851F538F44B18BFD40EC383EBEBF@greyhawk.tonecommander.com>
	<2d9149cd1002051150i3fba0945s4332af51d261274c@mail.gmail.com>
Message-ID: <7C537AE7D6064AF080EA20CD9C3D43D6@greyhawk.tonecommander.com>

Okay, so you use both FreeSWITCH and OpenSER in one box.  But just to be
clear, if I want to I should be able to use two FreeSWITCH instances in the
same box, one as a SBC and one as a PBX.  True?

Jerry


-----Original Message-----
From: Kristian Kielhofner [mailto:kristian.kielhofner at gmail.com] 
Sent: Friday, February 05, 2010 11:50 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Can Freeswitch and OpenSER Co-reside
onSameServer?

On Fri, Feb 5, 2010 at 2:40 PM, Jerry Richards <jerry.richards at teotech.com>
wrote:
> So do you build your server with two FS instances running? ?One as the 
> SBC and one as Proxy/PBX?
>
> Thanks,
> Jerry

Jerry,

  No.  One instance of FreeSWITCH and one instance of OpenSER.  As I said,
just make sure they use separate IPs and/or ports.  I prefer standard ports
and separate IPs because then (in the future) if I need to split them
(scaling, redundancy, etc) all I have to do is bring up the second IP on a
different host and move the software/config.

--
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com






From sos at sokhapkin.dyndns.org  Fri Feb  5 13:32:57 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Fri, 5 Feb 2010 16:32:57 -0500
Subject: [Freeswitch-users]
	=?iso-8859-1?q?Simple_IAX2_setup_-_help_with_c?=
	=?iso-8859-1?q?onverting=09from_asterisk_to_freeswitch?=
In-Reply-To: <4B6C8A80.9050700@gmail.com>
References: <91093d24f2166d220acf2ab19115f95d.squirrel@www.webcontracts.co.uk>
	<1265385834.12871.83.camel@local.freepabx.com>
	<4B6C8A80.9050700@gmail.com>
Message-ID: <201002051632.57936.sos@sokhapkin.dyndns.org>

I had random crashes on IAX outgoing calls in mod_iax (all calls went to the 
same provider). I gave up and now use asterisk as protocol converter.

On Friday 05 February 2010, Meftah Tayeb wrote:
> hi,
> iax2 is secure
> but, is not a good idea to avoid rtp and pass all packet including audio
> and signalisation troug the same port
> and digium added some change to the IAX2 protocol so freeswitch is not
> up to date
> no one want to update the iax2 stack in fs
> so fs mod_iax have bean removedfrom the trunk
>
> Le 05/02/2010 17:03, David Knell a ?crit :
> > There's a fairly simple solution to IAX needs, which is to run Asterisk,
> > probably on the same box, as a protocol converter - you just need to
> > tell it to use a non-standard port in sip.conf so that it doesn't clash
> > with FreeSWITCH.
> >
> > --Dave
> >
> >> the lib that we used to provide iax support is pretty much abandonware
> >> (no longer updated) and newer iax implementations (like latest
> >> asterisk) can cause it to crash.  There are no license compatible iax
> >> implementations that work, so.. mod_iax has been moved to the
> >> unsupported column.
> >>
> >>
> >> Default passwords -- that is a single var in vars.xml that controls
> >> the passwords.
> >>
> >>
> >> number ranges - up to you.  The sample configs supplied are just that,
> >> samples.  I use a smaller range personally.
> >>
> >> On Fri, Feb 5, 2010 at 7:53 AM, Matthew Law<matt at webcontracts.co.uk>
> >> wrote:
> >>          Why is that? - a lot of web pages I have read claim that IAX
> >>          is more
> >>          secure and efficient.  I have no problem with using SIP
> >>          whatsoever and it
> >>          certainly appears to be ubiquitous.  I am a complete newcomer
> >>          to VoIP and
> >>          I am trying to do this as securely as possible since I want to
> >>          run
> >>          freeswitch on a Xen VPS which will be visible on the internet.
> >>
> >>          Anyway, since my first question, I have worked my way through
> >>          the wiki,
> >>          read a lot of example configs and made some sense of the
> >>          docs.  I now have
> >>          a very basic config working (with SIP) on a local vmware image
> >>          using the
> >>          'quick and dirty' Makefile method.  I removed all of the
> >>          example configs
> >>          from the xml files (those default extensions and passwords
> >>          scared me) and
> >>          added just one for extension 1000, plus my dialplan and
> >>          inbound/outbound
> >>          settings.
> >>
> >>          One question: is there any reason not to use a smaller
> >>          extension number
> >>          range, like 200-210, for example?
> >>
> >>          Now to figure out how to get time based roaming working...
> >>
> >>
> >>          Thanks,
> >>
> >>          Matt.
> >>
> >>          On Fri, February 5, 2010 6:43 am, Michael Jerris wrote:
> >>          >  iax2 support has been removed from FreeSWITCH in current
> >>
> >>          trunk and will
> >>
> >>          >  not be in the 1.0.5 release.
> >>          >
> >>          >
> >>          >
> >>          >  Mike
> >>
> >>          _______________________________________________
> >>          FreeSWITCH-users mailing list
> >>          FreeSWITCH-users at lists.freeswitch.org
> >>          http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>         
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >>
> >>
> >> --
> >> -Rupa
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org





From jerry.richards at teotech.com  Fri Feb  5 13:34:27 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Fri, 5 Feb 2010 13:34:27 -0800
Subject: [Freeswitch-users] Blind Transfer Not Working
Message-ID: <6E6877B337EC4FF683536961971841D8@greyhawk.tonecommander.com>

Does anyone know why my blind transfer is not working?  I posted a trace in
th pastebin at http://pastebin.freeswitch.org/12065.  Attended transfer is
not working either.

Thanks And Best Regards,
Jerry




From sos at sokhapkin.dyndns.org  Fri Feb  5 13:37:43 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Fri, 5 Feb 2010 16:37:43 -0500
Subject: [Freeswitch-users] Can Freeswitch and OpenSER Co-reside
	onSameServer?
In-Reply-To: <7C537AE7D6064AF080EA20CD9C3D43D6@greyhawk.tonecommander.com>
References: <C4DC856C75094358AEDBE3FA0E242741@greyhawk.tonecommander.com>
	<2d9149cd1002051150i3fba0945s4332af51d261274c@mail.gmail.com>
	<7C537AE7D6064AF080EA20CD9C3D43D6@greyhawk.tonecommander.com>
Message-ID: <201002051637.43994.sos@sokhapkin.dyndns.org>

You can use either multiple FS instances on the same box, or use different SIP 
profiles of single FS instance to perform different functions.

I use openser as registrar/load balancer and multiple FS boxes for call 
handling and billing.

On Friday 05 February 2010, Jerry Richards wrote:
> Okay, so you use both FreeSWITCH and OpenSER in one box.  But just to be
> clear, if I want to I should be able to use two FreeSWITCH instances in the
> same box, one as a SBC and one as a PBX.  True?
>
> Jerry
>
>
> -----Original Message-----
> From: Kristian Kielhofner [mailto:kristian.kielhofner at gmail.com]
> Sent: Friday, February 05, 2010 11:50 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Can Freeswitch and OpenSER Co-reside
> onSameServer?
>
> On Fri, Feb 5, 2010 at 2:40 PM, Jerry Richards <jerry.richards at teotech.com>
>
> wrote:
> > So do you build your server with two FS instances running? ?One as the
> > SBC and one as Proxy/PBX?
> >
> > Thanks,
> > Jerry
>
> Jerry,
>
>   No.  One instance of FreeSWITCH and one instance of OpenSER.  As I said,
> just make sure they use separate IPs and/or ports.  I prefer standard ports
> and separate IPs because then (in the future) if I need to split them
> (scaling, redundancy, etc) all I have to do is bring up the second IP on a
> different host and move the software/config.
>
> --
> Kristian Kielhofner
> http://www.astlinux.org
> http://blog.krisk.org
> http://www.star2star.com
> http://www.submityoursip.com
> http://www.voalte.com
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org





From anthony.minessale at gmail.com  Fri Feb  5 13:39:51 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 5 Feb 2010 15:39:51 -0600
Subject: [Freeswitch-users] Simple IAX2 setup - help with converting
	from asterisk to freeswitch
In-Reply-To: <201002051632.57936.sos@sokhapkin.dyndns.org>
References: <91093d24f2166d220acf2ab19115f95d.squirrel@www.webcontracts.co.uk>
	<1265385834.12871.83.camel@local.freepabx.com>
	<4B6C8A80.9050700@gmail.com>
	<201002051632.57936.sos@sokhapkin.dyndns.org>
Message-ID: <191c3a031002051339g786fc85dwdbd139d9d097d70a@mail.gmail.com>

exactly,

The random crashes started happening by themselves, the protocol on the
other end has exploited it's ancestor code and we really don't feel like
bothering to fix it.


On Fri, Feb 5, 2010 at 3:32 PM, Sergey Okhapkin <sos at sokhapkin.dyndns.org>wrote:

> I had random crashes on IAX outgoing calls in mod_iax (all calls went to
> the
> same provider). I gave up and now use asterisk as protocol converter.
>
> On Friday 05 February 2010, Meftah Tayeb wrote:
> > hi,
> > iax2 is secure
> > but, is not a good idea to avoid rtp and pass all packet including audio
> > and signalisation troug the same port
> > and digium added some change to the IAX2 protocol so freeswitch is not
> > up to date
> > no one want to update the iax2 stack in fs
> > so fs mod_iax have bean removedfrom the trunk
> >
> > Le 05/02/2010 17:03, David Knell a ?crit :
> > > There's a fairly simple solution to IAX needs, which is to run
> Asterisk,
> > > probably on the same box, as a protocol converter - you just need to
> > > tell it to use a non-standard port in sip.conf so that it doesn't clash
> > > with FreeSWITCH.
> > >
> > > --Dave
> > >
> > >> the lib that we used to provide iax support is pretty much abandonware
> > >> (no longer updated) and newer iax implementations (like latest
> > >> asterisk) can cause it to crash.  There are no license compatible iax
> > >> implementations that work, so.. mod_iax has been moved to the
> > >> unsupported column.
> > >>
> > >>
> > >> Default passwords -- that is a single var in vars.xml that controls
> > >> the passwords.
> > >>
> > >>
> > >> number ranges - up to you.  The sample configs supplied are just that,
> > >> samples.  I use a smaller range personally.
> > >>
> > >> On Fri, Feb 5, 2010 at 7:53 AM, Matthew Law<matt at webcontracts.co.uk>
> > >> wrote:
> > >>          Why is that? - a lot of web pages I have read claim that IAX
> > >>          is more
> > >>          secure and efficient.  I have no problem with using SIP
> > >>          whatsoever and it
> > >>          certainly appears to be ubiquitous.  I am a complete newcomer
> > >>          to VoIP and
> > >>          I am trying to do this as securely as possible since I want
> to
> > >>          run
> > >>          freeswitch on a Xen VPS which will be visible on the
> internet.
> > >>
> > >>          Anyway, since my first question, I have worked my way through
> > >>          the wiki,
> > >>          read a lot of example configs and made some sense of the
> > >>          docs.  I now have
> > >>          a very basic config working (with SIP) on a local vmware
> image
> > >>          using the
> > >>          'quick and dirty' Makefile method.  I removed all of the
> > >>          example configs
> > >>          from the xml files (those default extensions and passwords
> > >>          scared me) and
> > >>          added just one for extension 1000, plus my dialplan and
> > >>          inbound/outbound
> > >>          settings.
> > >>
> > >>          One question: is there any reason not to use a smaller
> > >>          extension number
> > >>          range, like 200-210, for example?
> > >>
> > >>          Now to figure out how to get time based roaming working...
> > >>
> > >>
> > >>          Thanks,
> > >>
> > >>          Matt.
> > >>
> > >>          On Fri, February 5, 2010 6:43 am, Michael Jerris wrote:
> > >>          >  iax2 support has been removed from FreeSWITCH in current
> > >>
> > >>          trunk and will
> > >>
> > >>          >  not be in the 1.0.5 release.
> > >>          >
> > >>          >
> > >>          >
> > >>          >  Mike
> > >>
> > >>          _______________________________________________
> > >>          FreeSWITCH-users mailing list
> > >>          FreeSWITCH-users at lists.freeswitch.org
> > >>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >>
> > >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > >> http://www.freeswitch.org
> > >>
> > >>
> > >>
> > >>
> > >> --
> > >> -Rupa
> > >>
> > >> _______________________________________________
> > >> FreeSWITCH-users mailing list
> > >> FreeSWITCH-users at lists.freeswitch.org
> > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > >> http://www.freeswitch.org
> > >
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100205/6b7da7e7/attachment-0002.html 

From kristian.kielhofner at gmail.com  Fri Feb  5 13:44:12 2010
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Fri, 5 Feb 2010 16:44:12 -0500
Subject: [Freeswitch-users] Can Freeswitch and OpenSER Co-reside
	onSameServer?
In-Reply-To: <201002051637.43994.sos@sokhapkin.dyndns.org>
References: <C4DC856C75094358AEDBE3FA0E242741@greyhawk.tonecommander.com>
	<2d9149cd1002051150i3fba0945s4332af51d261274c@mail.gmail.com>
	<7C537AE7D6064AF080EA20CD9C3D43D6@greyhawk.tonecommander.com>
	<201002051637.43994.sos@sokhapkin.dyndns.org>
Message-ID: <2d9149cd1002051344r4cbe8d7ha95149ec6e392e15@mail.gmail.com>

On Fri, Feb 5, 2010 at 4:37 PM, Sergey Okhapkin
<sos at sokhapkin.dyndns.org> wrote:
> You can use either multiple FS instances on the same box, or use different SIP
> profiles of single FS instance to perform different functions.
>

Exactly.

-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com



From carlos.talbot at gmail.com  Fri Feb  5 13:46:41 2010
From: carlos.talbot at gmail.com (Carlos Talbot)
Date: Fri, 5 Feb 2010 15:46:41 -0600
Subject: [Freeswitch-users] SIP over TCP with Sipdroid, an Android SIP client
Message-ID: <5800526b1002051346g3890f152o1d939faa054811e6@mail.gmail.com>

Anyone use sipdroid on their Andorid phone? For the most part it works with
the exception of when using SIP over TCP. For some reason, after 30 seconds
into a call FreeSWITCH sends a bye and drops the call. Why use TCP? The
author claims significantly increased standby times using SIP TCP over 3g:
http://code.google.com/p/sipdroid/wiki/NewStandbyTechnique

According to Brian it might be because the phone is not setting a transport
in the contact field and FS is falling back to UDP.

This is on r16557. Here's a sip trace along with call graph:

http://pastebin.freeswitch.org/12064

regards,

Carlos
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100205/a529ca80/attachment-0002.html 

From jimthomasembedded at yahoo.com  Fri Feb  5 13:51:50 2010
From: jimthomasembedded at yahoo.com (Jim Thomas)
Date: Fri, 5 Feb 2010 13:51:50 -0800 (PST)
Subject: [Freeswitch-users] Can Freeswitch and OpenSER Co-reside on Same
	Server?
In-Reply-To: <2d9149cd1002051011x3b4f548dqabe615ed5f991687@mail.gmail.com>
References: <C4DC856C75094358AEDBE3FA0E242741@greyhawk.tonecommander.com>
	<2d9149cd1002051011x3b4f548dqabe615ed5f991687@mail.gmail.com>
Message-ID: <970116.27641.qm@web44811.mail.sp1.yahoo.com>

Kristian,

I enjoyed your recent blog praising FreeSWITCH.

The world needs an O'Reilly book about FreeSWITCH.? Perhaps you could write one in your spare time?

Whoever gets there first and does it well is likely to own that shelf space for a long time to come.

Just an idea.

Jim



----- Original Message ----
From: Kristian Kielhofner <kristian.kielhofner at gmail.com>
To: freeswitch-users at lists.freeswitch.org
Sent: Fri, February 5, 2010 12:11:35 PM
Subject: Re: [Freeswitch-users] Can Freeswitch and OpenSER Co-reside on Same Server?

On Fri, Feb 5, 2010 at 12:16 PM, Jerry Richards
<jerry.richards at teotech.com> wrote:
> If I use OpenSER for a session border controller, does anyone see an issue
> if it resides on the same server as Freeswitch? ?So I would have a LAN and
> WAN socket? ?Are there any drawbacks (other than loading) to worry about?
>
> Thanks And Best Regards,
> Jerry
>

You can use different IP addresses or ports.? I do this all of the time.

I question why you are using OpenSER (OpenSIPS?) as a SBC.? FreeSWITCH
is actually more well suited to most of the functions served by
something called* a "session border controller".

For example, FreeSWITCH in bypass media mode is a signaling only SBC
where you can (cleanly) do the header rewriting, number formatting,
and SIP topology hiding typically done by a SBC without touching the
media.? Proxy media mode can do the same while proxying media
(traversing NAT and hiding real RTP addresses).? FreeSWITCH in normal
bridging mode can transcode, convert between different types of DTMF
and do everything else mentioned above.

OpenSER as a proxy can't even (per RFC3261) rewrite To or From, nor
will it hide topology (it simple adds Record-Route/Via).

* Session borders controllers are very ill-defined and mean different
things to different people.

-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



      




From anthony.minessale at gmail.com  Fri Feb  5 13:53:27 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 5 Feb 2010 15:53:27 -0600
Subject: [Freeswitch-users] Blind Transfer Not Working
In-Reply-To: <6E6877B337EC4FF683536961971841D8@greyhawk.tonecommander.com>
References: <6E6877B337EC4FF683536961971841D8@greyhawk.tonecommander.com>
Message-ID: <191c3a031002051353u78163bafv1fcd2cd4fae244a7@mail.gmail.com>

look at line 481


   1.  v=0
   2.    o=TC 251112858 251112858 IN IP4 192.168.72.58
   3.    s=session rtp/2
   4.    c=IN IP4 192.168.72.58
   5.    t=0 0
   6.    m=audio 1760 RTP/AVP 101
   7.    a=rtpmap:101 telephone-event/8000/1


the sdp on the remote end does not have any codec info.



On Fri, Feb 5, 2010 at 3:34 PM, Jerry Richards
<jerry.richards at teotech.com>wrote:

> Does anyone know why my blind transfer is not working?  I posted a trace in
> th pastebin at http://pastebin.freeswitch.org/12065.  Attended transfer is
> not working either.
>
> Thanks And Best Regards,
> Jerry
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100205/33a535b5/attachment-0002.html 

From sos at sokhapkin.dyndns.org  Fri Feb  5 13:58:49 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Fri, 5 Feb 2010 16:58:49 -0500
Subject: [Freeswitch-users] Simple IAX2 setup - help with converting
	from asterisk to freeswitch
In-Reply-To: <191c3a031002051339g786fc85dwdbd139d9d097d70a@mail.gmail.com>
References: <91093d24f2166d220acf2ab19115f95d.squirrel@www.webcontracts.co.uk>
	<201002051632.57936.sos@sokhapkin.dyndns.org>
	<191c3a031002051339g786fc85dwdbd139d9d097d70a@mail.gmail.com>
Message-ID: <201002051658.49267.sos@sokhapkin.dyndns.org>

From the beginning Digium made a big mistake when released libiax as 
independent  from chan_iax code. Protocol changes/bug fixes in chan_iax never 
(or maybe rarely) went back to libiax. This ended up in 2 different IAX 
protocol implementations.

I wish Digium make chan_iax as a wrapper on top of supported and actively 
developed libiax...


On Friday 05 February 2010, Anthony Minessale wrote:
> exactly,
>
> The random crashes started happening by themselves, the protocol on the
> other end has exploited it's ancestor code and we really don't feel like
> bothering to fix it.
>
> On Fri, Feb 5, 2010 at 3:32 PM, Sergey Okhapkin 
<sos at sokhapkin.dyndns.org>wrote:
> > I had random crashes on IAX outgoing calls in mod_iax (all calls went to
> > the
> > same provider). I gave up and now use asterisk as protocol converter.
> >
> > On Friday 05 February 2010, Meftah Tayeb wrote:
> > > hi,
> > > iax2 is secure
> > > but, is not a good idea to avoid rtp and pass all packet including
> > > audio and signalisation troug the same port
> > > and digium added some change to the IAX2 protocol so freeswitch is not
> > > up to date
> > > no one want to update the iax2 stack in fs
> > > so fs mod_iax have bean removedfrom the trunk
> > >
> > > Le 05/02/2010 17:03, David Knell a ?crit :
> > > > There's a fairly simple solution to IAX needs, which is to run
> >
> > Asterisk,
> >
> > > > probably on the same box, as a protocol converter - you just need to
> > > > tell it to use a non-standard port in sip.conf so that it doesn't
> > > > clash with FreeSWITCH.
> > > >
> > > > --Dave
> > > >
> > > >> the lib that we used to provide iax support is pretty much
> > > >> abandonware (no longer updated) and newer iax implementations (like
> > > >> latest asterisk) can cause it to crash.  There are no license
> > > >> compatible iax implementations that work, so.. mod_iax has been
> > > >> moved to the unsupported column.
> > > >>
> > > >>
> > > >> Default passwords -- that is a single var in vars.xml that controls
> > > >> the passwords.
> > > >>
> > > >>
> > > >> number ranges - up to you.  The sample configs supplied are just
> > > >> that, samples.  I use a smaller range personally.
> > > >>
> > > >> On Fri, Feb 5, 2010 at 7:53 AM, Matthew Law<matt at webcontracts.co.uk>
> > > >> wrote:
> > > >>          Why is that? - a lot of web pages I have read claim that
> > > >> IAX is more
> > > >>          secure and efficient.  I have no problem with using SIP
> > > >>          whatsoever and it
> > > >>          certainly appears to be ubiquitous.  I am a complete
> > > >> newcomer to VoIP and
> > > >>          I am trying to do this as securely as possible since I want
> >
> > to
> >
> > > >>          run
> > > >>          freeswitch on a Xen VPS which will be visible on the
> >
> > internet.
> >
> > > >>          Anyway, since my first question, I have worked my way
> > > >> through the wiki,
> > > >>          read a lot of example configs and made some sense of the
> > > >>          docs.  I now have
> > > >>          a very basic config working (with SIP) on a local vmware
> >
> > image
> >
> > > >>          using the
> > > >>          'quick and dirty' Makefile method.  I removed all of the
> > > >>          example configs
> > > >>          from the xml files (those default extensions and passwords
> > > >>          scared me) and
> > > >>          added just one for extension 1000, plus my dialplan and
> > > >>          inbound/outbound
> > > >>          settings.
> > > >>
> > > >>          One question: is there any reason not to use a smaller
> > > >>          extension number
> > > >>          range, like 200-210, for example?
> > > >>
> > > >>          Now to figure out how to get time based roaming working...
> > > >>
> > > >>
> > > >>          Thanks,
> > > >>
> > > >>          Matt.
> > > >>
> > > >>          On Fri, February 5, 2010 6:43 am, Michael Jerris wrote:
> > > >>          >  iax2 support has been removed from FreeSWITCH in current
> > > >>
> > > >>          trunk and will
> > > >>
> > > >>          >  not be in the 1.0.5 release.
> > > >>          >
> > > >>          >
> > > >>          >
> > > >>          >  Mike
> > > >>
> > > >>          _______________________________________________
> > > >>          FreeSWITCH-users mailing list
> > > >>          FreeSWITCH-users at lists.freeswitch.org
> >
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >
> > > >> UNSUBSCRIBE:
> >
> > http://lists.freeswitch.org/mailman/options/freeswitch-users
> >
> > > >> http://www.freeswitch.org
> > > >>
> > > >>
> > > >>
> > > >>
> > > >> --
> > > >> -Rupa
> > > >>
> > > >> _______________________________________________
> > > >> FreeSWITCH-users mailing list
> > > >> FreeSWITCH-users at lists.freeswitch.org
> > > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > >> UNSUBSCRIBE:
> >
> > http://lists.freeswitch.org/mailman/options/freeswitch-users
> >
> > > >> http://www.freeswitch.org
> > > >
> > > > _______________________________________________
> > > > FreeSWITCH-users mailing list
> > > > FreeSWITCH-users at lists.freeswitch.org
> > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > > UNSUBSCRIBE:
> >
> > http://lists.freeswitch.org/mailman/options/freeswitch-users
> >
> > > > http://www.freeswitch.org
> > >
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-user
> > >s http://www.freeswitch.org
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org





From msc at freeswitch.org  Fri Feb  5 14:07:10 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 5 Feb 2010 14:07:10 -0800
Subject: [Freeswitch-users] Can Freeswitch and OpenSER Co-reside on Same
	Server?
In-Reply-To: <970116.27641.qm@web44811.mail.sp1.yahoo.com>
References: <C4DC856C75094358AEDBE3FA0E242741@greyhawk.tonecommander.com>
	<2d9149cd1002051011x3b4f548dqabe615ed5f991687@mail.gmail.com>
	<970116.27641.qm@web44811.mail.sp1.yahoo.com>
Message-ID: <87f2f3b91002051407r2ed4eb26hf6d8ff14f3a61c89@mail.gmail.com>

On Fri, Feb 5, 2010 at 1:51 PM, Jim Thomas <jimthomasembedded at yahoo.com>wrote:

> Kristian,
>
> I enjoyed your recent blog praising FreeSWITCH.
>
> The world needs an O'Reilly book about FreeSWITCH.  Perhaps you could write
> one in your spare time?
>
>
Would you guys settle for a FreeSWITCH book from Packt Publishing? Maybe we
could get them to put an animal on the cover... :)
-MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100205/33956e5d/attachment-0002.html 

From kristian.kielhofner at gmail.com  Fri Feb  5 14:10:05 2010
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Fri, 5 Feb 2010 17:10:05 -0500
Subject: [Freeswitch-users] Can Freeswitch and OpenSER Co-reside on Same
	Server?
In-Reply-To: <970116.27641.qm@web44811.mail.sp1.yahoo.com>
References: <C4DC856C75094358AEDBE3FA0E242741@greyhawk.tonecommander.com>
	<2d9149cd1002051011x3b4f548dqabe615ed5f991687@mail.gmail.com>
	<970116.27641.qm@web44811.mail.sp1.yahoo.com>
Message-ID: <2d9149cd1002051410o73076e0fs94ca76b4d2458801@mail.gmail.com>

On Fri, Feb 5, 2010 at 4:51 PM, Jim Thomas <jimthomasembedded at yahoo.com> wrote:
> Kristian,
>
> I enjoyed your recent blog praising FreeSWITCH.

  Thanks.

> The world needs an O'Reilly book about FreeSWITCH.? Perhaps you could write one in your spare time?

  O'Reilly authors have a joke: "The only people that get rich writing
books are Stephen King and J.K. Rowling".  I'm not picking on
O'Reilly; it doesn't matter who the publisher is.  Writing a tech book
takes an immense amount of time and compared to consulting and other
ways to spend your limited time it (often) doesn't pay well enough.

  Certainly some people are in different situation.  In the next
couple of days I'll be reviewing Packt Publishing's new OpenSIPS book
written by Flavio Goncalves.  Flavio has an OpenSIPS training business
so writing a book is perfect for him.  You get to train with the guy
who literally wrote the book!

> Whoever gets there first and does it well is likely to own that shelf space for a long time to come.

  There are some people working on a FreeSWITCH book and it makes
sense for them too.  I'm sure they'll do a great job.

-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com



From costa.zikalala at gmail.com  Fri Feb  5 15:52:48 2010
From: costa.zikalala at gmail.com (Costa Zikalala)
Date: Sat, 6 Feb 2010 01:52:48 +0200
Subject: [Freeswitch-users] Difference Between "bridge" and "transfer"
In-Reply-To: <87f2f3b91002051125u31dbf290h531d34d078c0f42e@mail.gmail.com>
References: <D123D09A29B942D988B3456528DD44AD@greyhawk.tonecommander.com> 
	<87f2f3b91002041632t19f17aefoaf11e048150816c5@mail.gmail.com> 
	<59daa2cd1002051044p1bee852ele8d0d972c303ea0e@mail.gmail.com> 
	<87f2f3b91002051125u31dbf290h531d34d078c0f42e@mail.gmail.com>
Message-ID: <59daa2cd1002051552l5d349c92w9a26baa55e5dbd1e@mail.gmail.com>

The incoming call is to a normal DID number, but I don't bridge to that
internal extension instead to a normal external PSTN number.
Will I then be charged for the b-leg?

Thanks
Costa


On 5 February 2010 21:25, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Fri, Feb 5, 2010 at 10:44 AM, Costa Zikalala <costa.zikalala at gmail.com>wrote:
>
>> Whilst on this subject, if I receive a call from PSTN and I 'bridge' it to
>> another PSTN extension, will the PSTN Provider charge me for the 'b-leg' or
>> will the original caller be charged the entire call?
>>
>
> That depends... is the "other" leg an outbound call? Is the other leg an
> inbound call to a toll-free number?
> -MC
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100206/76b43aa6/attachment-0002.html 

From jmesquita at freeswitch.org  Fri Feb  5 16:00:16 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Fri, 5 Feb 2010 22:00:16 -0200
Subject: [Freeswitch-users] Can Freeswitch and OpenSER Co-reside on Same
	Server?
In-Reply-To: <2d9149cd1002051410o73076e0fs94ca76b4d2458801@mail.gmail.com>
References: <C4DC856C75094358AEDBE3FA0E242741@greyhawk.tonecommander.com>
	<2d9149cd1002051011x3b4f548dqabe615ed5f991687@mail.gmail.com>
	<970116.27641.qm@web44811.mail.sp1.yahoo.com>
	<2d9149cd1002051410o73076e0fs94ca76b4d2458801@mail.gmail.com>
Message-ID: <ebbc2d551002051600yedce872m1c949db909c3be3b@mail.gmail.com>

Kudos for Flavio! Vamos brazukas! :-)

Abra?os,
Jo?o Mesquita


On Fri, Feb 5, 2010 at 8:10 PM, Kristian Kielhofner <
kristian.kielhofner at gmail.com> wrote:

> On Fri, Feb 5, 2010 at 4:51 PM, Jim Thomas <jimthomasembedded at yahoo.com>
> wrote:
> > Kristian,
> >
> > I enjoyed your recent blog praising FreeSWITCH.
>
>   Thanks.
>
> > The world needs an O'Reilly book about FreeSWITCH.  Perhaps you could
> write one in your spare time?
>
>   O'Reilly authors have a joke: "The only people that get rich writing
> books are Stephen King and J.K. Rowling".  I'm not picking on
> O'Reilly; it doesn't matter who the publisher is.  Writing a tech book
> takes an immense amount of time and compared to consulting and other
> ways to spend your limited time it (often) doesn't pay well enough.
>
>  Certainly some people are in different situation.  In the next
> couple of days I'll be reviewing Packt Publishing's new OpenSIPS book
> written by Flavio Goncalves.  Flavio has an OpenSIPS training business
> so writing a book is perfect for him.  You get to train with the guy
> who literally wrote the book!
>
> > Whoever gets there first and does it well is likely to own that shelf
> space for a long time to come.
>
>   There are some people working on a FreeSWITCH book and it makes
> sense for them too.  I'm sure they'll do a great job.
>
> --
> Kristian Kielhofner
> http://www.astlinux.org
> http://blog.krisk.org
> http://www.star2star.com
> http://www.submityoursip.com
> http://www.voalte.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100205/2b03afc0/attachment-0002.html 

From brian at freeswitch.org  Fri Feb  5 16:27:51 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 5 Feb 2010 18:27:51 -0600
Subject: [Freeswitch-users] Difference Between "bridge" and "transfer"
In-Reply-To: <59daa2cd1002051552l5d349c92w9a26baa55e5dbd1e@mail.gmail.com>
References: <D123D09A29B942D988B3456528DD44AD@greyhawk.tonecommander.com>
	<87f2f3b91002041632t19f17aefoaf11e048150816c5@mail.gmail.com>
	<59daa2cd1002051044p1bee852ele8d0d972c303ea0e@mail.gmail.com>
	<87f2f3b91002051125u31dbf290h531d34d078c0f42e@mail.gmail.com>
	<59daa2cd1002051552l5d349c92w9a26baa55e5dbd1e@mail.gmail.com>
Message-ID: <B074C6EB-C1FC-45D4-9577-AE01556CA343@freeswitch.org>

Depends dos your provider charge you for outbound calls?  Most do... so I suspect YES.

/b

On Feb 5, 2010, at 5:52 PM, Costa Zikalala wrote:

> The incoming call is to a normal DID number, but I don't bridge to that internal extension instead to a normal external PSTN number. 
> Will I then be charged for the b-leg? 
> 
> Thanks 
> Costa




From Prometheus001 at gmx.net  Fri Feb  5 17:36:18 2010
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Sat, 06 Feb 2010 02:36:18 +0100
Subject: [Freeswitch-users] mod_skypiax - help needed with
 modified	snd-dummy (one way audio)
In-Reply-To: <7b197bef1002051319l2fcb5617p1691692f58c65372@mail.gmail.com>
References: <4B60555B.2020004@gmx.net>	<7b197bef1001281210u43a907edi13afe794cf0e1a2e@mail.gmail.com>
	<4B61FCAB.5040707@gmx.net>	<7b197bef1001281341h7ee2c58fj79b3886630901f29@mail.gmail.com>
	<4B62C962.7000601@gmx.net>	<8213d6071001290403g2c08fe39gd92eb0446bb6485a@mail.gmail.com>
	<7b197bef1001290520x37179cb1o63273410e9ef5bc@mail.gmail.com>
	<4B6C1CCB.4080606@gmx.net>	<7b197bef1002050536j7ea09684v913434296ae52b97@mail.gmail.com>
	<4B6C868B.3040406@gmx.net>
	<7b197bef1002051319l2fcb5617p1691692f58c65372@mail.gmail.com>
Message-ID: <4B6CC792.5060608@gmx.net>

Skype starts, but as soon as it receives a call it crashes with:
  /usr/bin/skype: symbol lookup error: /usr/bin/skype: undefined symbol:
_ZN10QBoxLayout13addSpacerItemEP11QSpacerItem

I think the 8.10 version dos not work with8.04.

Any hints, where I may get an older Skype client? I may also try the static skype client.

Best regards
Peter



.
Giovanni Maruzzelli schrieb:
> that's not at all a fatal error.
> I believe it works the same.
> Are you sure it does not work?
>
> -gm
>
>
> On Fri, Feb 5, 2010 at 9:58 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>   
>> Hello Giovanni,
>>
>> I am now at the point to install Skype. But there is only an Intrepid
>> version available (no 8.04 version).
>> The current verison crashed on 8.04x because of dbus error.
>>    process 8408: D-Bus library appears to be incorrectly set up; failed
>> to read machine uuid: Failed to open "/var/lib/dbus/machine-id": No such
>> file or directory
>>    See the manual page for dbus-uuidgen to correct this issue.
>>    /usr/bin/skype: symbol lookup error: /usr/bin/skype: undefined
>> symbol: _ZN10QBoxLayout13addSpacerItemEP11QSpacerIte
>>
>> Any idea where I can download the older version for 8.04?
>>
>> Best regards
>> Peter
>>
>>
>> Giovanni Maruzzelli schrieb:
>>     
>>> Ciao Peter,
>>>
>>> I would use ubuntu 8.04 (hardy) LTS server 64bit or CentOS 5.4 64bit.
>>>
>>> -giovanni
>>>
>>> On Fri, Feb 5, 2010 at 2:27 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>
>>>       
>>>> Hello Giovanni,
>>>>
>>>> as I couldn't even get skype again working again with the standard alsa
>>>> driver, I would like to setup the machine from scratch based on a
>>>> working machine.
>>>> The latest errors I received from Skype was:
>>>>  snd_pcm_avail_update() returned a value that is exceptionally large:
>>>> 715706624 bytes (3727638 ms).
>>>>  Most likely this is a bug in the ALSA driver. Please report this issue
>>>> to the ALSA developers.
>>>> I think that may be the reason for one-way-audio.
>>>>
>>>> For setting up my machine from scratch, please advise:
>>>> - which OS you are you using und recommending exactly?
>>>> - I would like to use 64bit OS in order to use 8GB of memory, does this
>>>> work?
>>>> - any other hints?
>>>>
>>>> Best regards
>>>> Peter
>>>>
>>>> Giovanni Maruzzelli schrieb:
>>>>
>>>>         
>>>>> Peter,
>>>>>
>>>>> Can you connect on IRC (irc.freenode.net #freeswitch)? I'm gmaruzz there.
>>>>>
>>>>> Can you restate your problems? I've lost connection :)
>>>>>
>>>>> with snd-dummy custom you can create *one only* snd-dummy instance, so
>>>>> *one only* fake soundcard. If you create more, will not work. But with
>>>>> that one fake soundcard you can use 64 skype client instances, all
>>>>> with the same soundcard hardware device (hw:n).
>>>>>
>>>>> with original snd-dummy you can create a max of 8 instances, so 8 fake
>>>>> soundcards, and with each fake soundcard you can  use a max of 8 skype
>>>>> client instances.
>>>>>
>>>>> use the hardware devices, not the default devices (use the "hw:n")
>>>>>
>>>>> -giovanni
>>>>>
>>>>> On Fri, Jan 29, 2010 at 1:03 PM, Ghulam Mustafa <mustafa.pk at gmail.com> wrote:
>>>>>
>>>>>
>>>>>           
>>>>>> did you enable debug mode while compiling custom snd-dummy? if  yes
>>>>>> try re-compiling with debug mode disabled.
>>>>>>
>>>>>> -m
>>>>>>
>>>>>> On Fri, Jan 29, 2010 at 4:41 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>>
>>>>>>
>>>>>>             
>>>>>>> I now reinstalled the original sound drivers
>>>>>>> Unfortunaltely the sound problems remain, not that worse but they are there:
>>>>>>> Audio is still (almost) one way. Almost means:
>>>>>>>
>>>>>>>    * SIP -> Skype ok
>>>>>>>    * Skype=> SIP I hear only some scratching on very loud audio
>>>>>>>
>>>>>>> Could it be a volume problem? But snd-dummy should have no volume
>>>>>>> properties, right?
>>>>>>>
>>>>>>> Best regards
>>>>>>> Peter
>>>>>>>
>>>>>>>
>>>>>>> Giovanni Maruzzelli schrieb:
>>>>>>>
>>>>>>>
>>>>>>>               
>>>>>>>> with three instances you will assign the hw:0 device to skype client
>>>>>>>> 0...7, hw:1 to skype client 8...15 and hw:2 to skype client 16...23.
>>>>>>>> Must work. Pay attention to assign the same device name to all devices
>>>>>>>> needed by a skype instance (sound devices window): playback, capture
>>>>>>>> AND ring.
>>>>>>>>
>>>>>>>> Or maybe is a bug of ALSA on Debian...
>>>>>>>>
>>>>>>>> -giovanni
>>>>>>>>
>>>>>>>> On Thu, Jan 28, 2010 at 10:07 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>                 
>>>>>>>>> I crated 3 instances of snd-dummy, this worked. I assigned then Instance
>>>>>>>>> #2 to the Skype accounts. Still no sound.
>>>>>>>>> On the frist call there is one way audio, on the following calls there
>>>>>>>>> is no audio at all.
>>>>>>>>> This is weird.
>>>>>>>>>
>>>>>>>>> Best regards
>>>>>>>>> Peter
>>>>>>>>>
>>>>>>>>> Giovanni Maruzzelli schrieb:
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>                   
>>>>>>>>>> Ciao Peter,
>>>>>>>>>>
>>>>>>>>>> Never tested on Debian 5.
>>>>>>>>>>
>>>>>>>>>> When you write "same problem" you are referring to the audio going one
>>>>>>>>>> way only (btw, which way?) with the custom audio driver?
>>>>>>>>>>
>>>>>>>>>> Have you tried with multiple instances of the regular Debian
>>>>>>>>>> snd-dummy, as I wrote in a mail before?
>>>>>>>>>>
>>>>>>>>>> -gm
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> On Thu, Jan 28, 2010 at 9:00 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>                     
>>>>>>>>>>> Hello Giovanni,
>>>>>>>>>>>
>>>>>>>>>>> I did so but the same problem again.
>>>>>>>>>>>
>>>>>>>>>>> Did you ever test in on Debian 5.0?
>>>>>>>>>>>
>>>>>>>>>>> Best reards
>>>>>>>>>>> Peter
>>>>>>>>>>>
>>>>>>>>>>> Giovanni Maruzzelli schrieb:
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>                       
>>>>>>>>>>>> good, so you have only one sound device, the right one.
>>>>>>>>>>>>
>>>>>>>>>>>> Use the one with hw:0 in the window that skype gives you to set sound devices
>>>>>>>>>>>>
>>>>>>>>>>>> -gm
>>>>>>>>>>>>
>>>>>>>>>>>> On Wed, Jan 27, 2010 at 7:04 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>                         
>>>>>>>>>>>>> I installed alsa-utile,
>>>>>>>>>>>>>
>>>>>>>>>>>>> now I get:
>>>>>>>>>>>>>
>>>>>>>>>>>>> skype:/var/cache/apt/archives# aplay -l
>>>>>>>>>>>>> **** List of PLAYBACK Hardware Devices ****
>>>>>>>>>>>>> card 0: Dummy [Dummy], device 0: Dummy PCM [Dummy PCM]
>>>>>>>>>>>>>  Subdevices: 127/128
>>>>>>>>>>>>>  Subdevice #0: subdevice #0
>>>>>>>>>>>>>  Subdevice #1: subdevice #1
>>>>>>>>>>>>>  Subdevice #2: subdevice #2
>>>>>>>>>>>>>  Subdevice #3: subdevice #3
>>>>>>>>>>>>>  Subdevice #4: subdevice #4
>>>>>>>>>>>>>  Subdevice #5: subdevice #5
>>>>>>>>>>>>>  Subdevice #6: subdevice #6
>>>>>>>>>>>>>  Subdevice #7: subdevice #7
>>>>>>>>>>>>>  Subdevice #8: subdevice #8
>>>>>>>>>>>>>  Subdevice #9: subdevice #9
>>>>>>>>>>>>>  Subdevice #10: subdevice #10
>>>>>>>>>>>>>  Subdevice #11: subdevice #11
>>>>>>>>>>>>>  Subdevice #12: subdevice #12
>>>>>>>>>>>>>  Subdevice #13: subdevice #13
>>>>>>>>>>>>>  Subdevice #14: subdevice #14
>>>>>>>>>>>>>  Subdevice #15: subdevice #15
>>>>>>>>>>>>>  Subdevice #16: subdevice #16
>>>>>>>>>>>>>  Subdevice #17: subdevice #17
>>>>>>>>>>>>>  Subdevice #18: subdevice #18
>>>>>>>>>>>>>  Subdevice #19: subdevice #19
>>>>>>>>>>>>>  Subdevice #20: subdevice #20
>>>>>>>>>>>>>  Subdevice #21: subdevice #21
>>>>>>>>>>>>>  Subdevice #22: subdevice #22
>>>>>>>>>>>>>  Subdevice #23: subdevice #23
>>>>>>>>>>>>>  Subdevice #24: subdevice #24
>>>>>>>>>>>>>  Subdevice #25: subdevice #25
>>>>>>>>>>>>>  Subdevice #26: subdevice #26
>>>>>>>>>>>>>  Subdevice #27: subdevice #27
>>>>>>>>>>>>>  Subdevice #28: subdevice #28
>>>>>>>>>>>>>  Subdevice #29: subdevice #29
>>>>>>>>>>>>>  Subdevice #30: subdevice #30
>>>>>>>>>>>>>  Subdevice #31: subdevice #31
>>>>>>>>>>>>>  Subdevice #32: subdevice #32
>>>>>>>>>>>>>  Subdevice #33: subdevice #33
>>>>>>>>>>>>>  Subdevice #34: subdevice #34
>>>>>>>>>>>>>  Subdevice #35: subdevice #35
>>>>>>>>>>>>>  Subdevice #36: subdevice #36
>>>>>>>>>>>>>  Subdevice #37: subdevice #37
>>>>>>>>>>>>>  Subdevice #38: subdevice #38
>>>>>>>>>>>>>  Subdevice #39: subdevice #39
>>>>>>>>>>>>>  Subdevice #40: subdevice #40
>>>>>>>>>>>>>  Subdevice #41: subdevice #41
>>>>>>>>>>>>>  Subdevice #42: subdevice #42
>>>>>>>>>>>>>  Subdevice #43: subdevice #43
>>>>>>>>>>>>>  Subdevice #44: subdevice #44
>>>>>>>>>>>>>  Subdevice #45: subdevice #45
>>>>>>>>>>>>>  Subdevice #46: subdevice #46
>>>>>>>>>>>>>  Subdevice #47: subdevice #47
>>>>>>>>>>>>>  Subdevice #48: subdevice #48
>>>>>>>>>>>>>  Subdevice #49: subdevice #49
>>>>>>>>>>>>>  Subdevice #50: subdevice #50
>>>>>>>>>>>>>  Subdevice #51: subdevice #51
>>>>>>>>>>>>>  Subdevice #52: subdevice #52
>>>>>>>>>>>>>  Subdevice #53: subdevice #53
>>>>>>>>>>>>>  Subdevice #54: subdevice #54
>>>>>>>>>>>>>  Subdevice #55: subdevice #55
>>>>>>>>>>>>>  Subdevice #56: subdevice #56
>>>>>>>>>>>>>  Subdevice #57: subdevice #57
>>>>>>>>>>>>>  Subdevice #58: subdevice #58
>>>>>>>>>>>>>  Subdevice #59: subdevice #59
>>>>>>>>>>>>>  Subdevice #60: subdevice #60
>>>>>>>>>>>>>  Subdevice #61: subdevice #61
>>>>>>>>>>>>>  Subdevice #62: subdevice #62
>>>>>>>>>>>>>  Subdevice #63: subdevice #63
>>>>>>>>>>>>>  Subdevice #64: subdevice #64
>>>>>>>>>>>>>  Subdevice #65: subdevice #65
>>>>>>>>>>>>>  Subdevice #66: subdevice #66
>>>>>>>>>>>>>  Subdevice #67: subdevice #67
>>>>>>>>>>>>>  Subdevice #68: subdevice #68
>>>>>>>>>>>>>  Subdevice #69: subdevice #69
>>>>>>>>>>>>>  Subdevice #70: subdevice #70
>>>>>>>>>>>>>  Subdevice #71: subdevice #71
>>>>>>>>>>>>>  Subdevice #72: subdevice #72
>>>>>>>>>>>>>  Subdevice #73: subdevice #73
>>>>>>>>>>>>>  Subdevice #74: subdevice #74
>>>>>>>>>>>>>  Subdevice #75: subdevice #75
>>>>>>>>>>>>>  Subdevice #76: subdevice #76
>>>>>>>>>>>>>  Subdevice #77: subdevice #77
>>>>>>>>>>>>>  Subdevice #78: subdevice #78
>>>>>>>>>>>>>  Subdevice #79: subdevice #79
>>>>>>>>>>>>>  Subdevice #80: subdevice #80
>>>>>>>>>>>>>  Subdevice #81: subdevice #81
>>>>>>>>>>>>>  Subdevice #82: subdevice #82
>>>>>>>>>>>>>  Subdevice #83: subdevice #83
>>>>>>>>>>>>>  Subdevice #84: subdevice #84
>>>>>>>>>>>>>  Subdevice #85: subdevice #85
>>>>>>>>>>>>>  Subdevice #86: subdevice #86
>>>>>>>>>>>>>  Subdevice #87: subdevice #87
>>>>>>>>>>>>>  Subdevice #88: subdevice #88
>>>>>>>>>>>>>  Subdevice #89: subdevice #89
>>>>>>>>>>>>>  Subdevice #90: subdevice #90
>>>>>>>>>>>>>  Subdevice #91: subdevice #91
>>>>>>>>>>>>>  Subdevice #92: subdevice #92
>>>>>>>>>>>>>  Subdevice #93: subdevice #93
>>>>>>>>>>>>>  Subdevice #94: subdevice #94
>>>>>>>>>>>>>  Subdevice #95: subdevice #95
>>>>>>>>>>>>>  Subdevice #96: subdevice #96
>>>>>>>>>>>>>  Subdevice #97: subdevice #97
>>>>>>>>>>>>>  Subdevice #98: subdevice #98
>>>>>>>>>>>>>  Subdevice #99: subdevice #99
>>>>>>>>>>>>>  Subdevice #100: subdevice #100
>>>>>>>>>>>>>  Subdevice #101: subdevice #101
>>>>>>>>>>>>>  Subdevice #102: subdevice #102
>>>>>>>>>>>>>  Subdevice #103: subdevice #103
>>>>>>>>>>>>>  Subdevice #104: subdevice #104
>>>>>>>>>>>>>  Subdevice #105: subdevice #105
>>>>>>>>>>>>>  Subdevice #106: subdevice #106
>>>>>>>>>>>>>  Subdevice #107: subdevice #107
>>>>>>>>>>>>>  Subdevice #108: subdevice #108
>>>>>>>>>>>>>  Subdevice #109: subdevice #109
>>>>>>>>>>>>>  Subdevice #110: subdevice #110
>>>>>>>>>>>>>  Subdevice #111: subdevice #111
>>>>>>>>>>>>>  Subdevice #112: subdevice #112
>>>>>>>>>>>>>  Subdevice #113: subdevice #113
>>>>>>>>>>>>>  Subdevice #114: subdevice #114
>>>>>>>>>>>>>  Subdevice #115: subdevice #115
>>>>>>>>>>>>>  Subdevice #116: subdevice #116
>>>>>>>>>>>>>  Subdevice #117: subdevice #117
>>>>>>>>>>>>>  Subdevice #118: subdevice #118
>>>>>>>>>>>>>  Subdevice #119: subdevice #119
>>>>>>>>>>>>>  Subdevice #120: subdevice #120
>>>>>>>>>>>>>  Subdevice #121: subdevice #121
>>>>>>>>>>>>>  Subdevice #122: subdevice #122
>>>>>>>>>>>>>  Subdevice #123: subdevice #123
>>>>>>>>>>>>>  Subdevice #124: subdevice #124
>>>>>>>>>>>>>  Subdevice #125: subdevice #125
>>>>>>>>>>>>>  Subdevice #126: subdevice #126
>>>>>>>>>>>>>  Subdevice #127: subdevice #127
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>> Peter P GMX schrieb:
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>                           
>>>>>>>>>>>>>> Her's the output:
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> skype:~# aplay -l
>>>>>>>>>>>>>> bash: aplay: command not found
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> Giovanni Maruzzelli schrieb:
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>                             
>>>>>>>>>>>>>>> I don't think you got two snd-dummy loaded (but maybe yes)
>>>>>>>>>>>>>>> what's the output of:
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> aplay -l
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> ?
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> If instead you are referring to the choices that skype clients offers
>>>>>>>>>>>>>>> you in the "set audio devices" window, choose Dummy PCM (hw0:0)
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> Eg: not the "default", but the "hardware" one
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> On Wed, Jan 27, 2010 at 5:58 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>                               
>>>>>>>>>>>>>>>> Thanks Giovanni,
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> I think there may be the problem, that I have 2 sound devices now:
>>>>>>>>>>>>>>>> - Dummy PCM (hw0:0) (this is from debian install)
>>>>>>>>>>>>>>>> - Dummy PCM Default Audio device (defauzlt: CARD=Dummy) (this is new
>>>>>>>>>>>>>>>> since I compiled alsa newly)
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> I tried both, but both do not work. How do I get rid of the old alsa device?
>>>>>>>>>>>>>>>> By the way: I uninstalled Alsa before I installed the new driver
>>>>>>>>>>>>>>>> (apt-get remove alsa-utils alsa-base).
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> Best regards
>>>>>>>>>>>>>>>> Peter
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> Giovanni Maruzzelli schrieb:
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>                                 
>>>>>>>>>>>>>>>>> This warning is harmless:
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> Jan 27 15:28:50 skype kernel: [ 3994.795786] process `skype' is using
>>>>>>>>>>>>>>>>> obsolete setsockopt SO_BSDCOMPAT
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> On Wed, Jan 27, 2010 at 4:26 PM, Giovanni Maruzzelli
>>>>>>>>>>>>>>>>> <gmaruzz at celliax.org> wrote:
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>                                   
>>>>>>>>>>>>>>>>>> Ciao Peter
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>> one instance of snd-dummy "customized" is enough for 64 instances of
>>>>>>>>>>>>>>>>>> skype clients, no need (and do not works) with more instances of
>>>>>>>>>>>>>>>>>> snd-dummy-customized.
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>> Maybe you got the one-way problem because of kernel at 250HZ (don't
>>>>>>>>>>>>>>>>>> know). It uses to works well on a tickless kernel at 100HZ (eg: ubuntu
>>>>>>>>>>>>>>>>>> 8.04).
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>> Or maybe you have to check and modify which sound devices the skype
>>>>>>>>>>>>>>>>>> clients are using (try to check that with snd-summy-custom loaded,
>>>>>>>>>>>>>>>>>> maybe with the ssh -X trick (as in the wiki page).
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>> To load more than one snd-dummy-original (the non modified one), you
>>>>>>>>>>>>>>>>>> do this with the modprobe command, as in:
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>> rmmod snd-dummy
>>>>>>>>>>>>>>>>>> modprobe snd-dummy enable=1,1,1
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>> this command will enable three instances of snd-dummy original, so
>>>>>>>>>>>>>>>>>> you'll have three fake soundcards, and you'll have to setup each group
>>>>>>>>>>>>>>>>>> of 8 skype instances to use sound devices from one fake soundcard, RG:
>>>>>>>>>>>>>>>>>> no more than 8 skype client instances can use one instance of fake
>>>>>>>>>>>>>>>>>> soundcard.
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>> Also, please update the mod_skypiax code (svn up in its directory) I
>>>>>>>>>>>>>>>>>> just committed some improvements.
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>> If you have any other doubts, or need more info, don't hesitate to
>>>>>>>>>>>>>>>>>> write the mailing list again,
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>> ciao for now,
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>> -giovanni
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>> On Wed, Jan 27, 2010 at 4:01 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>                                     
>>>>>>>>>>>>>>>>>>> I have mod_skypiax working nicely so far with 2 Skype channels. Thanks
>>>>>>>>>>>>>>>>>>> to all contributors, excellent work!
>>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>> In order to have more than 8 channels working, I have followed the
>>>>>>>>>>>>>>>>>>> instructions in
>>>>>>>>>>>>>>>>>>> http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk
>>>>>>>>>>>>>>>>>>> and compiled alsa-driver-1.0.20 with the modified dummy.c file. (System
>>>>>>>>>>>>>>>>>>> ist Debian 5.0R3)
>>>>>>>>>>>>>>>>>>> It compiled well however when I start snd-dummy I only have
>>>>>>>>>>>>>>>>>>> one-way-audio and my logs show
>>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>> Jan 27 15:28:41 skype kernel: [ 3984.318403] snd-dummy skypiax driver,
>>>>>>>>>>>>>>>>>>> /usr/src/alsa-driver-1.0.20/drivers/../alsa-kernel/drivers/dummy.c:920
>>>>>>>>>>>>>>>>>>> working on a machine with 250HZ kernel
>>>>>>>>>>>>>>>>>>> Jan 27 15:28:50 skype kernel: [ 3994.795786] process `skype' is using
>>>>>>>>>>>>>>>>>>> obsolete setsockopt SO_BSDCOMPAT
>>>>>>>>>>>>>>>>>>> Jan 27 15:28:56 skype kernel: [ 4005.289907] __ratelimit: 490 messages
>>>>>>>>>>>>>>>>>>> suppressed
>>>>>>>>>>>>>>>>>>> Jan 27 15:29:01 skype kernel: [ 4012.458310] process `skype' is using
>>>>>>>>>>>>>>>>>>> obsolete setsockopt SO_BSDCOMPAT
>>>>>>>>>>>>>>>>>>> Jan 27 15:29:01 skype kernel: [ 4013.326290] __ratelimit: 499 messages
>>>>>>>>>>>>>>>>>>> suppressed
>>>>>>>>>>>>>>>>>>> If I reinstall alsa from deb everything sworks fine again (of course
>>>>>>>>>>>>>>>>>>> with the current limitations).
>>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>> First question: Has anybody had this issue before? How can I solve this?
>>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>> Second question:
>>>>>>>>>>>>>>>>>>> As I do not need 64 channels or more: how do I manage, that Skype
>>>>>>>>>>>>>>>>>>> instances 9..15 use a second instance of snd-dummy as addressed in the wiki?
>>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>> Best regards
>>>>>>>>>>>>>>>>>>> Peter
>>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>>                                       
>>>>>>>>>>>>>>>>>> --
>>>>>>>>>>>>>>>>>> Sincerely,
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>> Giovanni Maruzzelli
>>>>>>>>>>>>>>>>>> Cell : +39-347-2665618
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>                                     
>>>>>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>                                 
>>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>                           
>>>>>>>>>>> _______________________________________________
>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>                       
>>>>>>>>> _______________________________________________
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>                   
>>>>>>>>                 
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>               
>>>>>> --
>>>>>> Ghulam Mustafa
>>>>>> cell: +92 333.611.7681
>>>>>> sip: cyrenity at ekiga.net
>>>>>> mail: mustafa.pk at gmail.com
>>>>>> web: cyrenity.wordpress.com
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>>
>>>>>>             
>>>>>
>>>>>           
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>         
>>>
>>>
>>>       
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>     
>
>
>
>   



From Russell.Mosemann at cune.org  Fri Feb  5 18:51:57 2010
From: Russell.Mosemann at cune.org (Russell Mosemann)
Date: Fri, 5 Feb 2010 20:51:57 -0600
Subject: [Freeswitch-users] Difference Between "bridge" and "transfer"
In-Reply-To: <59daa2cd1002051552l5d349c92w9a26baa55e5dbd1e@mail.gmail.com>
References: <D123D09A29B942D988B3456528DD44AD@greyhawk.tonecommander.com>
	<87f2f3b91002041632t19f17aefoaf11e048150816c5@mail.gmail.com>
	<59daa2cd1002051044p1bee852ele8d0d972c303ea0e@mail.gmail.com>
	<87f2f3b91002051125u31dbf290h531d34d078c0f42e@mail.gmail.com>
	<59daa2cd1002051552l5d349c92w9a26baa55e5dbd1e@mail.gmail.com>
Message-ID: <1467DE0871D34F5DB81DC9C2ADE0463C@cune.pri>

Costa Zikalala asked

> The incoming call is to a normal DID number, but I don't bridge to that
> internal extension instead to a normal external PSTN number.
> Will I then be charged for the b-leg?

You are making the call on the b-leg. If you are normally charged for making a call, then you will be charged. A call is a call. It doesn't matter if your cousin is making the call, your grandmother is making the call or FS is making the call. It is all happening on your PSTN line. The telephone company doesn't care who is making the call. They only care that your line is being used to make a call.

--
Russell Mosemann




From nagalenoj at gmail.com  Fri Feb  5 22:48:36 2010
From: nagalenoj at gmail.com (Nagalenoj H.)
Date: Sat, 6 Feb 2010 12:18:36 +0530
Subject: [Freeswitch-users] A leg disconnects after uuid_bridge?
In-Reply-To: <191c3a031002051024w1f484febj34a34d9e06421c52@mail.gmail.com>
References: <fe67a3341002021934m548cde89p4d48381c5abe02a0@mail.gmail.com>
	<191c3a031002022108ta22287ev40fc36420abba2e5@mail.gmail.com>
	<fe67a3341002030112yeea379u90c22c482614c57@mail.gmail.com>
	<87f2f3b91002030605j7a1f9a02g665ba043c6ed7f7b@mail.gmail.com>
	<fe67a3341002040132n3aa2a897m60c5b9f1ca6b347f@mail.gmail.com>
	<0A571E14-BB58-4A3D-996A-9D5205668E85@freeswitch.org>
	<fe67a3341002040452v5468fd1fh96d1c5062091474b@mail.gmail.com>
	<191c3a031002040754r2987a03fo18ee2e1ceb239f6a@mail.gmail.com>
	<fe67a3341002042041u1ee70b86u3bc91adbb6fb3d52@mail.gmail.com>
	<191c3a031002051024w1f484febj34a34d9e06421c52@mail.gmail.com>
Message-ID: <fe67a3341002052248y3bff9d07g78d0a70e6a0ddb17@mail.gmail.com>

I've upgraded to trunk number - 16580. But, Intercept isn't working as per
the my understanding(in my last mail).

While doing 'make current', faced the following errors.. But, after
installing, I executed freeswitch and it showed the version as 16580.

installing mod_voicemail.so
make[6]: Leaving directory
`/root/files/freeswitch/src/mod/applications/mod_voicemail'
make[5]: Leaving directory
`/root/files/freeswitch/src/mod/applications/mod_voicemail'

making install mod_voipcodecs
make[5]: Entering directory
`/root/files/freeswitch/src/mod/codecs/mod_voipcodecs'
make[6]: Entering directory
`/root/files/freeswitch/src/mod/codecs/mod_voipcodecs'
make[7]: Entering directory `/root/files/freeswitch/libs/tiff-3.8.2'
 cd . && /bin/sh /root/files/freeswitch/libs/tiff-3.8.2/config/missing --run
automake-1.9 --foreign  Makefile
configure.ac: no proper invocation of AM_INIT_AUTOMAKE was found.
configure.ac: You should verify that configure.ac invokes AM_INIT_AUTOMAKE,
configure.ac: that aclocal.m4 is present in the top-level directory,
configure.ac: and that aclocal.m4 was recently regenerated (using aclocal).
configure.ac: required file `config/install-sh' not found
configure.ac:37: required file `config/config.guess' not found
configure.ac:37: required file `config/config.sub' not found
make[7]: *** [Makefile.in] Error 1
make[7]: Leaving directory `/root/files/freeswitch/libs/tiff-3.8.2'
make[6]: *** [/root/files/freeswitch/libs/tiff-3.8.2/libtiff/libtiff.la]
Error 2
make[6]: Leaving directory
`/root/files/freeswitch/src/mod/codecs/mod_voipcodecs'
make[5]: *** [install] Error 1
make[5]: Leaving directory
`/root/files/freeswitch/src/mod/codecs/mod_voipcodecs'
make[4]: *** [mod_voipcodecs-install] Error 1
make[4]: Leaving directory `/root/files/freeswitch/src/mod'
make[4]: Entering directory `/root/files/freeswitch/src'
make[4]: Leaving directory `/root/files/freeswitch/src'
make[3]: *** [install-recursive] Error 1
make[3]: Leaving directory `/root/files/freeswitch/src'
make[2]: Leaving directory `/root/files/freeswitch'
make[1]: Leaving directory `/root/files/freeswitch'


On Fri, Feb 5, 2010 at 11:54 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> try latest trunk i think your issue is fixed.
>
>
>
> On Thu, Feb 4, 2010 at 10:41 PM, Nagalenoj H. <nagalenoj at gmail.com> wrote:
>
>> Sorry., I couldn't understand its behavior.
>>
>> Let me ask the same question in this way.
>>
>> * hangup_after_bridge is set to false.
>> * In outbound socket, first I answer the call.
>> * When I do a bridge to a extension (1001), after 1001 disconnects the
>> call. I am able to make another call.
>>       sendmsg
>>       call-command: execute
>>       execute-app-name: bridge
>>       execute-app-arg: user/1001
>>
>> * When I originate a call to extension (1001), after 1001 disconnects the
>> call. I'm unable to make another call, because my session is also getting
>> closed.
>>       api originate user/1001 &park
>>
>>       Content-Type: api/response
>>       Content-Length: 41
>>
>>      +OK 1fac17ce-120b-11df-a878-d9c7fbcf71c4
>>
>>
>>       sendmsg
>>       call-command: execute
>>       execute-app-name: intercept
>>       execute-app-arg: 1fac17ce-120b-11df-a878-d9c7fbcf71c4
>>
>> * In both the case, the call is getting bridged to an extension and
>> hangup_after_bridge is false.
>> * When bridge doesn't need any other variables to set to continue, why
>> intercept needs a explicit park after bridge.?
>>
>> Hope, this has some clarity.,
>>
>>
>> On Thu, Feb 4, 2010 at 9:24 PM, Anthony Minessale <
>> anthony.minessale at gmail.com> wrote:
>>
>>>
>>>
>>>    1. 2010-02-04 14:30:09.574084 [DEBUG] switch_core_state_machine.c:354
>>>    (sofia/internal/1010 at 192.168.1.222) State SOFT_EXECUTE going to sleep
>>>    2. 2010-02-04 14:30:09.574084 [DEBUG] switch_core_state_machine.c:314
>>>    (sofia/internal/1010 at 192.168.1.222) Running State Change CS_EXECUTE
>>>    3. 2010-02-04 14:30:09.574084 [DEBUG] switch_core_state_machine.c:348
>>>    (sofia/internal/1010 at 192.168.1.222) State EXECUTE
>>>    4. 2010-02-04 14:30:09.574084 [DEBUG] mod_sofia.c:181 sofia/internal/
>>>    1010 at 192.168.1.222 SOFIA EXECUTE
>>>    5. 2010-02-04 14:30:09.574084 [DEBUG] switch_core_state_machine.c:159sofia/internal/
>>>    1010 at 192.168.1.222 Standard EXECUTE
>>>    6. 2010-02-04 14:30:09.574084 [NOTICE] switch_core_state_machine.c:
>>>    187 Hangup sofia/internal/1010 at 192.168.1.222 [CS_EXECUTE] [
>>>    NORMAL_CLEARING]
>>>
>>>
>>>
>>> Your channel went back to EXECUTE as expected then it hungup because
>>> there were no more instructions in your dial plan for it to execute.  So it
>>> is working as expected.
>>>
>>> Consider using transfer_after_bridge variable or park_after bridge to
>>> make it stay around when the call is over.
>>>
>>>
>>>
>>>
>>> On Thu, Feb 4, 2010 at 6:52 AM, Nagalenoj H. <nagalenoj at gmail.com>wrote:
>>>
>>>> By using create_uuid. I've also tried without giving origination_uuid.
>>>> But, the result is same.
>>>>
>>>> --
>>>> Regards,
>>>> Nagalenoj H.
>>>>
>>>>
>>>> On Thu, Feb 4, 2010 at 5:42 PM, Brian West <brian at freeswitch.org>wrote:
>>>>
>>>>> Where are you getting this UUID?
>>>>>
>>>>> /b
>>>>>
>>>>> On Feb 4, 2010, at 3:32 AM, Nagalenoj H. wrote:
>>>>>
>>>>> > api originate
>>>>> {origination_uuid=c6d2e0e2-0b2f-11df-9e84-fb15c3cd8565}user/1001 &park
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>> iax:guest at conference.freeswitch.org/888
>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> pstn:+19193869900
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Regards,
>> Nagalenoj H.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Regards,
Nagalenoj H.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100206/207a5e32/attachment-0002.html 

From oseslija at gmail.com  Sat Feb  6 02:55:01 2010
From: oseslija at gmail.com (Ognjen Seslija)
Date: Sat, 6 Feb 2010 11:55:01 +0100
Subject: [Freeswitch-users] Difference Between "bridge" and "transfer"
In-Reply-To: <59daa2cd1002051044p1bee852ele8d0d972c303ea0e@mail.gmail.com>
References: <D123D09A29B942D988B3456528DD44AD@greyhawk.tonecommander.com>
	<87f2f3b91002041632t19f17aefoaf11e048150816c5@mail.gmail.com>
	<59daa2cd1002051044p1bee852ele8d0d972c303ea0e@mail.gmail.com>
Message-ID: <4468a6771002060255l582856e7g577d63d0b596b173@mail.gmail.com>

This happens when most of my customers forward call from their sip phones to
pstn extensions (mobile phones i.e.).
FS will then bridge two pstn calls and hence you'll be charged for the b-leg
call. Also, most likely is that you will not be able to present the original
caller id of a-leg to the forwarded extension, because of pstn settings
(inbound acls for clid on your line).

There is a way on most ISDN lines to "deflect" call back to the telco switch
with the forward extension information, so the switch can call the b-leg
itself and properly present the original clid. FS then doesn't need to
anything but to respond with the information.
I'm pretty sure, though, that telco will charge you for that second call,
also.

Regards,
Ognjen

On Fri, Feb 5, 2010 at 7:44 PM, Costa Zikalala <costa.zikalala at gmail.com>wrote:

> Whilst on this subject, if I receive a call from PSTN and I 'bridge' it to
> another PSTN extension, will the PSTN Provider charge me for the 'b-leg' or
> will the original caller be charged the entire call?
>
>
>
> On 5 February 2010 02:32, Michael Collins <msc at freeswitch.org> wrote:
>
>>
>>
>> On Thu, Feb 4, 2010 at 3:56 PM, Jerry Richards <
>> jerry.richards at teotech.com> wrote:
>>
>>> What is the difference between "bridge" and "transfer"?  I'm looking at
>>> the
>>> demo IVRs.
>>>
>>>
>> bridge will connect two endpoints together while transfer sends the
>> endpoint back through the dialplan again...
>>
>> -MC
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100206/601917d3/attachment-0002.html 

From woodydickson at gmail.com  Sat Feb  6 05:06:38 2010
From: woodydickson at gmail.com (Woody Dickson)
Date: Sat, 6 Feb 2010 21:06:38 +0800
Subject: [Freeswitch-users] AUDIO RTP REPORTS ERROR: [Bind Error!]
Message-ID: <a44688a1002060506q8679d19u5e403207df073d30@mail.gmail.com>

Hi,

While running FreeSwitch with 300 CPS, I start to get the following error:


2010-02-05 01:08:39.756510 [ERR] sofia_glue.c:2503 AUDIO RTP REPORTS
ERROR: [Bind Error!]
2010-02-05 01:08:39.759524 [WARNING] switch_ivr_bridge.c:992 Bridge
Failed


Each time such an error pops up, the memory used by FS increases.
This is a potential memory leaking there.


Does anyone know what is the problem and how to solve it?


woody



From tayeb.meftah at gmail.com  Sat Feb  6 05:41:18 2010
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Sat, 06 Feb 2010 14:41:18 +0100
Subject: [Freeswitch-users] Billing IVR
In-Reply-To: <4468a6771002060255l582856e7g577d63d0b596b173@mail.gmail.com>
References: <D123D09A29B942D988B3456528DD44AD@greyhawk.tonecommander.com>	<87f2f3b91002041632t19f17aefoaf11e048150816c5@mail.gmail.com>	<59daa2cd1002051044p1bee852ele8d0d972c303ea0e@mail.gmail.com>
	<4468a6771002060255l582856e7g577d63d0b596b173@mail.gmail.com>
Message-ID: <4B6D717E.4000306@gmail.com>

hi,
  i am creating a simple prepay IVR to bill calls troug mod_nibblebill
     using javascript and some recorded sounds
the user call 222
call get answered and request a pin
the pin is the nibblebill account number
if the pin is corect, the user will heare the actual account cache
and if want to recharge, the user will enter the recharge card pin 
number (8 digites)
if is corecte, we move cache from the card table to account table and 
the account is debited
but if user inter the card pin number, the IVR is returning the user to 
the pin number prompt
any error in this
Pastebin <http://pastebin:freeswitch at pastebin.freeswitch.org/12068>
(http://pastebin.freeswitch.org/12068)
please help me folk
thank you
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100206/3a3403d2/attachment-0002.html 

From anthony.minessale at gmail.com  Sat Feb  6 05:44:27 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 6 Feb 2010 07:44:27 -0600
Subject: [Freeswitch-users] A leg disconnects after uuid_bridge?
In-Reply-To: <191c3a031002060543x27fb6ce8t6b96f05630dec77d@mail.gmail.com>
References: <fe67a3341002021934m548cde89p4d48381c5abe02a0@mail.gmail.com>
	<87f2f3b91002030605j7a1f9a02g665ba043c6ed7f7b@mail.gmail.com>
	<fe67a3341002040132n3aa2a897m60c5b9f1ca6b347f@mail.gmail.com>
	<0A571E14-BB58-4A3D-996A-9D5205668E85@freeswitch.org>
	<fe67a3341002040452v5468fd1fh96d1c5062091474b@mail.gmail.com>
	<191c3a031002040754r2987a03fo18ee2e1ceb239f6a@mail.gmail.com>
	<fe67a3341002042041u1ee70b86u3bc91adbb6fb3d52@mail.gmail.com>
	<191c3a031002051024w1f484febj34a34d9e06421c52@mail.gmail.com>
	<fe67a3341002052248y3bff9d07g78d0a70e6a0ddb17@mail.gmail.com>
	<191c3a031002060543x27fb6ce8t6b96f05630dec77d@mail.gmail.com>
Message-ID: <191c3a031002060544v4707e0abmbbc239e6469f0ba0@mail.gmail.com>

You probably need to do a fresh build those errors mean its not fully
building correctly.

On Feb 6, 2010 12:56 AM, "Nagalenoj H." <nagalenoj at gmail.com> wrote:

I've upgraded to trunk number - 16580. But, Intercept isn't working as per
the my understanding(in my last mail).

While doing 'make current', faced the following errors.. But, after
installing, I executed freeswitch and it showed the version as 16580.

installing mod_voicemail.so
make[6]: Leaving directory
`/root/files/freeswitch/src/mod/applications/mod_voicemail'
make[5]: Leaving directory
`/root/files/freeswitch/src/mod/applications/mod_voicemail'

making install mod_voipcodecs
make[5]: Entering directory
`/root/files/freeswitch/src/mod/codecs/mod_voipcodecs'
make[6]: Entering directory
`/root/files/freeswitch/src/mod/codecs/mod_voipcodecs'
make[7]: Entering directory `/root/files/freeswitch/libs/tiff-3.8.2'
 cd . && /bin/sh /root/files/freeswitch/libs/tiff-3.8.2/config/missing --run
automake-1.9 --foreign  Makefile
configure.ac: no proper invocation of AM_INIT_AUTOMAKE was found.
configure.ac: You should verify that configure.ac invokes AM_INIT_AUTOMAKE,
configure.ac: that aclocal.m4 is present in the top-level directory,
configure.ac: and that aclocal.m4 was recently regenerated (using aclocal).
configure.ac: required file `config/install-sh' not found
configure.ac:37: required file `config/config.guess' not found
configure.ac:37: required file `config/config.sub' not found
make[7]: *** [Makefile.in] Error 1
make[7]: Leaving directory `/root/files/freeswitch/libs/tiff-3.8.2'
make[6]: *** [/root/files/freeswitch/libs/tiff-3.8.2/libtiff/libtiff.la]
Error 2
make[6]: Leaving directory
`/root/files/freeswitch/src/mod/codecs/mod_voipcodecs'
make[5]: *** [install] Error 1
make[5]: Leaving directory
`/root/files/freeswitch/src/mod/codecs/mod_voipcodecs'
make[4]: *** [mod_voipcodecs-install] Error 1
make[4]: Leaving directory `/root/files/freeswitch/src/mod'
make[4]: Entering directory `/root/files/freeswitch/src'
make[4]: Leaving directory `/root/files/freeswitch/src'
make[3]: *** [install-recursive] Error 1
make[3]: Leaving directory `/root/files/freeswitch/src'
make[2]: Leaving directory `/root/files/freeswitch'
make[1]: Leaving directory `/root/files/freeswitch'




On Fri, Feb 5, 2010 at 11:54 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:
>
> try l...

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100206/9aa099a6/attachment-0002.html 

From anthony.minessale at gmail.com  Sat Feb  6 05:55:01 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 6 Feb 2010 07:55:01 -0600
Subject: [Freeswitch-users] AUDIO RTP REPORTS ERROR: [Bind Error!]
In-Reply-To: <191c3a031002060552g4f241f83wb4d881ed07de7dd5@mail.gmail.com>
References: <a44688a1002060506q8679d19u5e403207df073d30@mail.gmail.com>
	<191c3a031002060549h57c256fapb1b7e56aaab8a12f@mail.gmail.com>
	<191c3a031002060550h1fb60638q16b201536685b532@mail.gmail.com>
	<191c3a031002060552g4f241f83wb4d881ed07de7dd5@mail.gmail.com>
Message-ID: <191c3a031002060555x153637d9iec16fb1d8dc44d7c@mail.gmail.com>

That's pretty funny.
Let me stop laughing about the nonchalant expectation of 300cps
first.........................

Ok so,

You ran out of rtp ports.
The configured range is from 16384 to 32768 evens so do the math, 8192 simo
ports.

At that rate you are going to run out or possibly collide with other rtp
enabled devices on the same box.

If you are just using sipp to torture fs expect it to back up if you go
beyond the limitations of your hardware.

That said we do not entertain load testing threads......

On Feb 6, 2010 7:13 AM, "Woody Dickson" <woodydickson at gmail.com> wrote:

Hi,

While running FreeSwitch with 300 CPS, I start to get the following error:


2010-02-05 01:08:39.756510 [ERR] sofia_glue.c:2503 AUDIO RTP REPORTS
ERROR: [Bind Error!]
2010-02-05 01:08:39.759524 [WARNING] switch_ivr_bridge.c:992 Bridge
Failed


Each time such an error pops up, the memory used by FS increases.
This is a potential memory leaking there.


Does anyone know what is the problem and how to solve it?


woody

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100206/09c70177/attachment-0002.html 

From anthony.minessale at gmail.com  Sat Feb  6 05:58:48 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 6 Feb 2010 07:58:48 -0600
Subject: [Freeswitch-users] mod_skypiax - help needed with modified
	snd-dummy (one way audio)
In-Reply-To: <4B6CC792.5060608@gmx.net>
References: <4B60555B.2020004@gmx.net>
	<7b197bef1001281341h7ee2c58fj79b3886630901f29@mail.gmail.com>
	<4B62C962.7000601@gmx.net>
	<8213d6071001290403g2c08fe39gd92eb0446bb6485a@mail.gmail.com>
	<7b197bef1001290520x37179cb1o63273410e9ef5bc@mail.gmail.com>
	<4B6C1CCB.4080606@gmx.net>
	<7b197bef1002050536j7ea09684v913434296ae52b97@mail.gmail.com>
	<4B6C868B.3040406@gmx.net>
	<7b197bef1002051319l2fcb5617p1691692f58c65372@mail.gmail.com>
	<4B6CC792.5060608@gmx.net>
Message-ID: <191c3a031002060558r36947d93vcbb46df9e0a7dfde@mail.gmail.com>

Quick fyi centos 5.4 has some memory bugs in libc don't go beyond 5.3 until
its fixed.

On Feb 5, 2010 7:42 PM, "Peter P GMX" <Prometheus001 at gmx.net> wrote:

Skype starts, but as soon as it receives a call it crashes with:

/usr/bin/skype: symbol lookup error: /usr/bin/skype: undefined symbol:
_ZN10QBoxLayout13addSpacerItemEP11QSpacerItem

I think the 8.10 version dos not work with8.04.

Any hints, where I may get an older Skype client? I may also try the static
skype client.


Best regards
Peter



.
Giovanni Maruzzelli schrieb:
> that's not at all a fatal error.
> I believe...
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100206/7472dca5/attachment-0002.html 

From steveu at coppice.org  Sat Feb  6 10:21:13 2010
From: steveu at coppice.org (Steve Underwood)
Date: Sun, 07 Feb 2010 02:21:13 +0800
Subject: [Freeswitch-users] The brilliance of hyper-links
Message-ID: <4B6DB319.2090308@coppice.org>

I would like to strongly object to a story posted on the front page at 
www.freeswitch.org, about patent reexamination. Its the kind of thing 
that gets proponents of patent reform a bad name. It says "Did you know 
that someone actually got a patent on the oh-so-clever concept of the 
hyperlink 
<http://patft.uspto.gov/netacgi/nph-Parser?Sect1=PTO1&Sect2=HITOFF&d=PALL&p=1&u=%2Fnetahtml%2FPTO%2Fsrchnum.htm&r=1&f=G&l=50&s1=4,873,662.PN.&OS=PN/4,873,662&RS=PN/4,873,662>? 
Enough said.". If you can't see the brilliance of the hyperlink you must 
be a fool. When Doug Engelbart demostrated that concept in 1968 it was 
so brilliant it went over the heads of most of the audience, some of 
them making dumb remarks that entirely missed the point. That's pretty 
much conclusive proof that it was not something obvious to practitioners 
in the art.

The patent on hyperlinks was not bad because the idea was obvious. It 
was bad because it was applied for in 1980, 12 years after the concept 
was demonstrated. Luckily, video exists of the 1968 demo, and the patent 
was shot down.

If you can't get your act together about where real innovation lies, 
just shut up. It just makes arguing a meaningful case for a better 
patent system hard for the rest of us.

Steve




From jimthomasembedded at yahoo.com  Sat Feb  6 11:14:19 2010
From: jimthomasembedded at yahoo.com (Jim Thomas)
Date: Sat, 6 Feb 2010 11:14:19 -0800 (PST)
Subject: [Freeswitch-users] Can Freeswitch and OpenSER Co-reside on Same
	Server?
In-Reply-To: <87f2f3b91002051407r2ed4eb26hf6d8ff14f3a61c89@mail.gmail.com>
References: <C4DC856C75094358AEDBE3FA0E242741@greyhawk.tonecommander.com>
	<2d9149cd1002051011x3b4f548dqabe615ed5f991687@mail.gmail.com>
	<970116.27641.qm@web44811.mail.sp1.yahoo.com>
	<87f2f3b91002051407r2ed4eb26hf6d8ff14f3a61c89@mail.gmail.com>
Message-ID: <344660.64876.qm@web44803.mail.sp1.yahoo.com>

Packt would be fine, and kudos to Packt for being so present in the open source telephony publishing space.

Actually, I would like to see two FreeSWITCH books, one for users (how to employ it as a turnkey element) and another for developers (FreeSWITCH internal architecture, structure, Sofia-SIP, etc.).



________________________________
From: Michael Collins <msc at freeswitch.org>
To: freeswitch-users at lists.freeswitch.org
Sent: Fri, February 5, 2010 4:07:10 PM
Subject: Re: [Freeswitch-users] Can Freeswitch and OpenSER Co-reside on Same Server?




On Fri, Feb 5, 2010 at 1:51 PM, Jim Thomas <jimthomasembedded at yahoo.com> wrote:

>Kristian,
>
>>I enjoyed your recent blog praising FreeSWITCH.
>
>>The world needs an O'Reilly book about FreeSWITCH.  Perhaps you could write one in your spare time?
>
>

Would you guys settle for a FreeSWITCH book from Packt Publishing? Maybe we could get them to put an animal on the cover... :)
-MC


      
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100206/89974506/attachment-0002.html 

From christian at officepools.com  Sat Feb  6 11:24:05 2010
From: christian at officepools.com (Christian Jensen)
Date: Sat, 6 Feb 2010 11:24:05 -0800
Subject: [Freeswitch-users] Looking for some good/cheap desktop phones
In-Reply-To: <SNT119-W1223C808A5E092C7E2A590B5540@phx.gbl>
References: <3f2e071b5aea5e11bb37a37e435a33b7@mail.gmail.com>
	<8e9d67561002042236s434e0643od322551de8720be1@mail.gmail.com>
	<409D0ACF-ADA4-4D02-8B7E-71EC1B1BAFBC@freeswitch.org>
	<8e9d67561002042301v76129abfj3dcbc70ed073752c@mail.gmail.com>
	<d361ca7b1002050507n5e564621oa6cf7ace005dd5b8@mail.gmail.com>
	<00ae01caa698$65b570f0$312052d0$@net>
	<b39273531002051154h5132faa9h9fb024ebdfb15b77@mail.gmail.com>
	<SNT119-W1223C808A5E092C7E2A590B5540@phx.gbl>
Message-ID: <e90351e71002061124v129b081cg501fb0c6ae6a17dd@mail.gmail.com>

The response on this thread has been great!

I wonder if there should be a link off the FreeSwitch site for vendors for
these phones.... hint hint.

>From what I have been hearing here and elsewhere, the preference is Polycom,
Cisco, Grandstream, Aastra in that order.

Does anyone have a URL for where I could buy the units without crazy stupid
shipping costs? VoipSupply quoted me ~$60 for a shipping a single phone to
Vancouver, Canada, Yick!

I looked on EBay but there isn't anything I could consider a steal.

Did I hear that someone was putting FS into a phone or something like that?
Cluecon and the secure sip guy was talking about this I think.

Christian
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100206/78214d55/attachment-0002.html 

From frank at carmickle.com  Sat Feb  6 12:05:47 2010
From: frank at carmickle.com (Frank Carmickle)
Date: Sat, 6 Feb 2010 15:05:47 -0500
Subject: [Freeswitch-users] Looking for some good/cheap desktop phones
In-Reply-To: <e90351e71002061124v129b081cg501fb0c6ae6a17dd@mail.gmail.com>
References: <3f2e071b5aea5e11bb37a37e435a33b7@mail.gmail.com>
	<8e9d67561002042236s434e0643od322551de8720be1@mail.gmail.com>
	<409D0ACF-ADA4-4D02-8B7E-71EC1B1BAFBC@freeswitch.org>
	<8e9d67561002042301v76129abfj3dcbc70ed073752c@mail.gmail.com>
	<d361ca7b1002050507n5e564621oa6cf7ace005dd5b8@mail.gmail.com>
	<00ae01caa698$65b570f0$312052d0$@net>
	<b39273531002051154h5132faa9h9fb024ebdfb15b77@mail.gmail.com>
	<SNT119-W1223C808A5E092C7E2A590B5540@phx.gbl>
	<e90351e71002061124v129b081cg501fb0c6ae6a17dd@mail.gmail.com>
Message-ID: <20100206200547.GD31942@base.carmickle.com>

On Sat, Feb 06, Christian Jensen wrote:
> The response on this thread has been great!
> 
> I wonder if there should be a link off the FreeSwitch site for vendors for
> these phones.... hint hint.
> 
> >From what I have been hearing here and elsewhere, the preference is Polycom,
> Cisco, Grandstream, Aastra in that order.

Hmmm...  That's not the impression that I got at all.  From what I picked up on this thread it would look more like

polycom snom aastra cisco grandstream

I'm sure others can speak to how Cisco doesn't do presence right.  They are harder to provision then Snoms and Polycoms and in my opinion don't sound as good either.  To me they sound dull and fuzzy on transmit.  I've heard about four different versions of 7940/7960.  

The Snom 3x0 series have suffered in the speaker phone department until version 7 firmware.  Version 8 betas have sounded even better yet.  Make sure that if your going to be using headsets with Snoms that you ground the phone to something.  This can be done by either having a shielded network cable attached to a grounded switch or computer, or order them with the grounded power adaptor.  They give you the not so grounded adaptor by default.  Finding shielded ethernet can be kind of tricky but it is nice to have.

Polycom speech quality is my favorite if you can handle the ringy under water sounds of the background noise especially on speaker.  It sounds like there's a fish tank running near by if there's any white noise.  I believe this to be the case because the way they split the audio up in to many bands to process it.  Maybe less steep filters would help.  

It's hard to believe how many people just don't care about call quality.  It starts with a good sounding phone that doesn't echo or hum.  I'd like to hope that we are doing our part to help the world sound a little better all the time.

--FC



From msc at freeswitch.org  Sat Feb  6 13:48:20 2010
From: msc at freeswitch.org (Michael S Collins)
Date: Sat, 6 Feb 2010 13:48:20 -0800
Subject: [Freeswitch-users] The brilliance of hyper-links
In-Reply-To: <4B6DB319.2090308@coppice.org>
References: <4B6DB319.2090308@coppice.org>
Message-ID: <7B471A70-40F2-4F8D-A3EA-009DDA20FCD6@freeswitch.org>

Sir,

You've gotten unnecessarily upset. A poor choice of words written in a  
bit of a hurry. You are exactly right about the reason *why* the  
patent is bogus. "Clever" was the wrong word here. It should have been  
"the oh so new, original, never-been-done-before, not even a hint of  
prior art, concept of the hyperlink."

No offense intended to the real originator(s) of the very useful  
hyperlink.

-MC

Sent from my iPhone

On Feb 6, 2010, at 10:21 AM, Steve Underwood <steveu at coppice.org> wrote:

> I would like to strongly object to a story posted on the front page at
> www.freeswitch.org, about patent reexamination. Its the kind of thing
> that gets proponents of patent reform a bad name. It says "Did you  
> know
> that someone actually got a patent on the oh-so-clever concept of the
> hyperlink
> <http://patft.uspto.gov/netacgi/nph-Parser?Sect1=PTO1&Sect2=HITOFF&d=PALL&p=1&u=%2Fnetahtml%2FPTO%2Fsrchnum.htm&r=1&f=G&l=50&s1=4,873,662.PN.&OS=PN/4,873,662&RS=PN/4,873,662 
> >?
> Enough said.". If you can't see the brilliance of the hyperlink you  
> must
> be a fool. When Doug Engelbart demostrated that concept in 1968 it was
> so brilliant it went over the heads of most of the audience, some of
> them making dumb remarks that entirely missed the point. That's pretty
> much conclusive proof that it was not something obvious to  
> practitioners
> in the art.
>
> The patent on hyperlinks was not bad because the idea was obvious. It
> was bad because it was applied for in 1980, 12 years after the concept
> was demonstrated. Luckily, video exists of the 1968 demo, and the  
> patent
> was shot down.
>
> If you can't get your act together about where real innovation lies,
> just shut up. It just makes arguing a meaningful case for a better
> patent system hard for the rest of us.
>
> Steve
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org



From derek at indranet.co.nz  Sat Feb  6 15:11:05 2010
From: derek at indranet.co.nz (Derek Smithies)
Date: Sun, 7 Feb 2010 12:11:05 +1300 (NZDT)
Subject: [Freeswitch-users] Simple IAX2 setup - help with converting
 from asterisk to freeswitch
In-Reply-To: <d2976552e9ba0ce1ed095b7cef22e1ad.squirrel@www.webcontracts.co.uk>
References: <91093d24f2166d220acf2ab19115f95d.squirrel@www.webcontracts.co.uk>
	<E1E3F6EF-1A1E-4891-B371-7A3BF0988E95@jerris.com>
	<d2976552e9ba0ce1ed095b7cef22e1ad.squirrel@www.webcontracts.co.uk>
Message-ID: <alpine.DEB.2.00.1002071152520.4493@monster>

On Fri, 5 Feb 2010, Matthew Law wrote:

> Why is that? - a lot of web pages I have read claim that IAX is more
> secure and efficient.  I have no problem with using SIP whatsoever and it
> certainly appears to be ubiquitous.  I am a complete newcomer to VoIP and
> I am trying to do this as securely as possible since I want to run
> freeswitch on a Xen VPS which will be visible on the internet.
>

Sigh.
  the web pages are wrong.

I have implemented the earlier version of IAX2 inside the opal library. At 
one stage, I could make a voip call using IAX2 to digiums test server.

The standard IAX2 codebase did not (at one stage) support silence 
suppression. With silence suppression, the bandwidth usage is halved. 
Using efficient packing of audio makes a few percent of difference. Doing 
silence suppression makes 50% of difference to bandwidth.

IAX2 is more efficient? Most(>90%) of the cpu work when doing voip is in 
the codec. By using a non standard approach (iax2 audio packets) for 
carrying audio, you will have a minimal gain in effiency (or cpu usage). 
Remember that iax2 uses the same codecs as in H.323/SIP. Oh - you are 
right, g711 is a codec, and has trivial cpu cost. However, with g711 any 
bandwidth you save on the header is negligible compared to the size of the 
encoded audio block.

Secure? Security protocols (HIP, ZRTP, etc) take years of careful 
development to get something that works well. I do not recall seeing 
evidence of years of development going into IAX2 security. Either the 
author of the security in IAX2 is a pure genius, able to do in 
days/weeks what other do in years,,,, or the iax2 security is 
average (at best).

Yes, all the iax2 packets go to the same port. This has huge advantages 
for getting through firewalls, and setting up firewalls to accept incoming 
calls. It does (slightly) increase the complexity inside the iax2 code.

===================================

if you want iax2 inside freeswitch, my suggestion is that the opal library 
needs to have the iax2 code there brought up to spec, and then used (in 
the same way as the H.323 component of the opal library).

Derek.
-- 
Derek Smithies Ph.D.
IndraNet Technologies Ltd.
Email: derek at indranet.co.nz
ph +64 3 365 6485
Web: http://www.indranet-technologies.com/




From gavin.henry at gmail.com  Sat Feb  6 15:47:01 2010
From: gavin.henry at gmail.com (Gavin Henry)
Date: Sat, 6 Feb 2010 23:47:01 +0000
Subject: [Freeswitch-users] Looking for some good/cheap desktop phones
In-Reply-To: <20100206200547.GD31942@base.carmickle.com>
References: <3f2e071b5aea5e11bb37a37e435a33b7@mail.gmail.com>
	<8e9d67561002042236s434e0643od322551de8720be1@mail.gmail.com>
	<409D0ACF-ADA4-4D02-8B7E-71EC1B1BAFBC@freeswitch.org>
	<8e9d67561002042301v76129abfj3dcbc70ed073752c@mail.gmail.com>
	<d361ca7b1002050507n5e564621oa6cf7ace005dd5b8@mail.gmail.com>
	<00ae01caa698$65b570f0$312052d0$@net>
	<b39273531002051154h5132faa9h9fb024ebdfb15b77@mail.gmail.com>
	<SNT119-W1223C808A5E092C7E2A590B5540@phx.gbl>
	<e90351e71002061124v129b081cg501fb0c6ae6a17dd@mail.gmail.com>
	<20100206200547.GD31942@base.carmickle.com>
Message-ID: <13ca621c1002061547n22353b27tefe78f28c66f073d@mail.gmail.com>

I would put Aastra before a snom, but then again Aastra always wants a reboot!

On 06/02/2010, Frank Carmickle <frank at carmickle.com> wrote:
> On Sat, Feb 06, Christian Jensen wrote:
>> The response on this thread has been great!
>>
>> I wonder if there should be a link off the FreeSwitch site for vendors for
>> these phones.... hint hint.
>>
>> >From what I have been hearing here and elsewhere, the preference is
>> Polycom,
>> Cisco, Grandstream, Aastra in that order.
>
> Hmmm...  That's not the impression that I got at all.  From what I picked up
> on this thread it would look more like
>
> polycom snom aastra cisco grandstream
>
> I'm sure others can speak to how Cisco doesn't do presence right.  They are
> harder to provision then Snoms and Polycoms and in my opinion don't sound as
> good either.  To me they sound dull and fuzzy on transmit.  I've heard about
> four different versions of 7940/7960.
>
> The Snom 3x0 series have suffered in the speaker phone department until
> version 7 firmware.  Version 8 betas have sounded even better yet.  Make
> sure that if your going to be using headsets with Snoms that you ground the
> phone to something.  This can be done by either having a shielded network
> cable attached to a grounded switch or computer, or order them with the
> grounded power adaptor.  They give you the not so grounded adaptor by
> default.  Finding shielded ethernet can be kind of tricky but it is nice to
> have.
>
> Polycom speech quality is my favorite if you can handle the ringy under
> water sounds of the background noise especially on speaker.  It sounds like
> there's a fish tank running near by if there's any white noise.  I believe
> this to be the case because the way they split the audio up in to many bands
> to process it.  Maybe less steep filters would help.
>
> It's hard to believe how many people just don't care about call quality.  It
> starts with a good sounding phone that doesn't echo or hum.  I'd like to
> hope that we are doing our part to help the world sound a little better all
> the time.
>
> --FC
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>

-- 
Sent from my mobile device

http://www.suretecsystems.com/services/openldap/
http://www.suretectelecom.com



From gavin.henry at gmail.com  Sat Feb  6 16:00:07 2010
From: gavin.henry at gmail.com (Gavin Henry)
Date: Sun, 7 Feb 2010 00:00:07 +0000
Subject: [Freeswitch-users] Can Freeswitch and OpenSER Co-reside on Same
	Server?
In-Reply-To: <2d9149cd1002051410o73076e0fs94ca76b4d2458801@mail.gmail.com>
References: <C4DC856C75094358AEDBE3FA0E242741@greyhawk.tonecommander.com>
	<2d9149cd1002051011x3b4f548dqabe615ed5f991687@mail.gmail.com>
	<970116.27641.qm@web44811.mail.sp1.yahoo.com>
	<2d9149cd1002051410o73076e0fs94ca76b4d2458801@mail.gmail.com>
Message-ID: <13ca621c1002061600x2bfe7ee4l166e6d5cec18f1ad@mail.gmail.com>

Hi,

I got asked to review this too as I submitted a few bugs in the book.
Their review guidelines are long and I don't have enough time to write
such an in depth review. I'll watch your blog though!

I hope the index is better this time as Packt always let me down that way.

Gavin.


On 05/02/2010, Kristian Kielhofner <kristian.kielhofner at gmail.com> wrote:
> On Fri, Feb 5, 2010 at 4:51 PM, Jim Thomas <jimthomasembedded at yahoo.com>
> wrote:
>> Kristian,
>>
>> I enjoyed your recent blog praising FreeSWITCH.
>
>   Thanks.
>
>> The world needs an O'Reilly book about FreeSWITCH.? Perhaps you could
>> write one in your spare time?
>
>   O'Reilly authors have a joke: "The only people that get rich writing
> books are Stephen King and J.K. Rowling".  I'm not picking on
> O'Reilly; it doesn't matter who the publisher is.  Writing a tech book
> takes an immense amount of time and compared to consulting and other
> ways to spend your limited time it (often) doesn't pay well enough.
>
>   Certainly some people are in different situation.  In the next
> couple of days I'll be reviewing Packt Publishing's new OpenSIPS book
> written by Flavio Goncalves.  Flavio has an OpenSIPS training business
> so writing a book is perfect for him.  You get to train with the guy
> who literally wrote the book!
>
>> Whoever gets there first and does it well is likely to own that shelf
>> space for a long time to come.
>
>   There are some people working on a FreeSWITCH book and it makes
> sense for them too.  I'm sure they'll do a great job.
>
> --
> Kristian Kielhofner
> http://www.astlinux.org
> http://blog.krisk.org
> http://www.star2star.com
> http://www.submityoursip.com
> http://www.voalte.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>

-- 
Sent from my mobile device

http://www.suretecsystems.com/services/openldap/
http://www.suretectelecom.com



From matt at webcontracts.co.uk  Sat Feb  6 16:05:47 2010
From: matt at webcontracts.co.uk (Matthew Law)
Date: Sun, 7 Feb 2010 00:05:47 -0000
Subject: [Freeswitch-users] Simple IAX2 setup - help with converting
 from asterisk to freeswitch
In-Reply-To: <alpine.DEB.2.00.1002071152520.4493@monster>
References: <91093d24f2166d220acf2ab19115f95d.squirrel@www.webcontracts.co.uk>
	<E1E3F6EF-1A1E-4891-B371-7A3BF0988E95@jerris.com>
	<d2976552e9ba0ce1ed095b7cef22e1ad.squirrel@www.webcontracts.co.uk>
	<alpine.DEB.2.00.1002071152520.4493@monster>
Message-ID: <65e4725010ae312e26887c33624c1bb9.squirrel@www.webcontracts.co.uk>

Thanks for the comprehensive reply.  I have switched to using plain SIP
and the experiments continue...

Matt

On Sat, February 6, 2010 11:11 pm, Derek Smithies wrote:
> Sigh.
>   the web pages are wrong.
>
> I have implemented the earlier version of IAX2 inside the opal library. At
> one stage, I could make a voip call using IAX2 to digiums test server.
>
> The standard IAX2 codebase did not (at one stage) support silence
> suppression. With silence suppression, the bandwidth usage is halved.
> Using efficient packing of audio makes a few percent of difference. Doing
> silence suppression makes 50% of difference to bandwidth.
>
> IAX2 is more efficient? Most(>90%) of the cpu work when doing voip is in
> the codec. By using a non standard approach (iax2 audio packets) for
> carrying audio, you will have a minimal gain in effiency (or cpu usage).
> Remember that iax2 uses the same codecs as in H.323/SIP. Oh - you are
> right, g711 is a codec, and has trivial cpu cost. However, with g711 any
> bandwidth you save on the header is negligible compared to the size of the
> encoded audio block.
>
> Secure? Security protocols (HIP, ZRTP, etc) take years of careful
> development to get something that works well. I do not recall seeing
> evidence of years of development going into IAX2 security. Either the
> author of the security in IAX2 is a pure genius, able to do in
> days/weeks what other do in years,,,, or the iax2 security is
> average (at best).
>
> Yes, all the iax2 packets go to the same port. This has huge advantages
> for getting through firewalls, and setting up firewalls to accept incoming
> calls. It does (slightly) increase the complexity inside the iax2 code.
>
> ===================================
>
> if you want iax2 inside freeswitch, my suggestion is that the opal library
> needs to have the iax2 code there brought up to spec, and then used (in
> the same way as the H.323 component of the opal library).




From matt at webcontracts.co.uk  Sat Feb  6 16:24:39 2010
From: matt at webcontracts.co.uk (Matthew Law)
Date: Sun, 7 Feb 2010 00:24:39 -0000
Subject: [Freeswitch-users] ACL question and js error
Message-ID: <ea05d45dca9be2d979fb386f960bc077.squirrel@www.webcontracts.co.uk>

After some more experiments I have a working replacement for the asterisk
box we were using before, which is great.

I had problems getting incoming calls to work. Changing the entry in
acl.conf.xml from:

<list name="domains" default="deny">
   <node type="allow" domain="$${domain}"/>
</list>

to:

<list name="domains" default="allow">
   <node type="allow" domain="$${domain}"/>
</list>

and reloading xml works but this gets reverted every time FS starts up. 
I've scanned the wiki docs and can't see anything pertaining to that. 
Why/where is this happening and how do I make it the default?  Actually,
the question should probably be is it sensible to do that? - the box is
out on the internet and I really only want to take incoming calls from
voiptalk.org, but I can't find a list of IPs on their site which I could
create an acl from...

Second question: I have tried this example for an answer machine (mainly
because it looked the shortest and simplest of the examples listed):

http://wiki.freeswitch.org/wiki/Examples_answermachine

I get this error:

2010-02-06 19:01:26.799118 [ERR] answermachine.js:135  ReferenceError:
email is not defined

Which is relating to this line:

email(eMailFrom, eMailTo, "Subject: " + tmp_eMailSubject, eMailBody,
tmp_Filename);

The script says it requires mod_spidermonkey_etpan, but I can't find any
reference to that anywhere (I'm using svn trunk).  Where is it?

Thanks,

Matt - roughly 1% up the FreeSWITCH learning curve and climbing...

PS: a nice FreeSWITCH 'cookbook' which starts out at the most simple
example and goes on to add more juicy, *working* features would get my
money any day. Hint, hint! :-)




From costa.zikalala at gmail.com  Sun Feb  7 06:36:43 2010
From: costa.zikalala at gmail.com (Costa Zikalala)
Date: Sun, 7 Feb 2010 16:36:43 +0200
Subject: [Freeswitch-users] Difference Between "bridge" and "transfer"
In-Reply-To: <4468a6771002060255l582856e7g577d63d0b596b173@mail.gmail.com>
References: <D123D09A29B942D988B3456528DD44AD@greyhawk.tonecommander.com> 
	<87f2f3b91002041632t19f17aefoaf11e048150816c5@mail.gmail.com> 
	<59daa2cd1002051044p1bee852ele8d0d972c303ea0e@mail.gmail.com> 
	<4468a6771002060255l582856e7g577d63d0b596b173@mail.gmail.com>
Message-ID: <59daa2cd1002070636p7d28b1b3s4bae8ee7c9c69fbf@mail.gmail.com>

Thanks
Thanks Ognjen for that extra bit of info.


On 6 February 2010 12:55, Ognjen Seslija <oseslija at gmail.com> wrote:
>
> There is a way on most ISDN lines to "deflect" call back to the telco
> switch with the forward extension information, so the switch can call the
> b-leg itself and properly present the original clid. FS then doesn't need to
> anything but to respond with the information.
>
>
> On Fri, Feb 5, 2010 at 7:44 PM, Costa Zikalala <costa.zikalala at gmail.com>wrote:
>
>> Whilst on this subject, if I receive a call from PSTN and I 'bridge' it to
>> another PSTN extension, will the PSTN Provider charge me for the 'b-leg' or
>> will the original caller be charged the entire call?
>>
>>
>>
>> On 5 February 2010 02:32, Michael Collins <msc at freeswitch.org> wrote:
>>
>>>
>>>
>>> On Thu, Feb 4, 2010 at 3:56 PM, Jerry Richards <
>>> jerry.richards at teotech.com> wrote:
>>>
>>>> What is the difference between "bridge" and "transfer"?  I'm looking at
>>>> the
>>>> demo IVRs.
>>>>
>>>>
>>> bridge will connect two endpoints together while transfer sends the
>>> endpoint back through the dialplan again...
>>>
>>> -MC
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100207/0393f070/attachment-0002.html 

From frank at carmickle.com  Sun Feb  7 06:59:07 2010
From: frank at carmickle.com (Frank Carmickle)
Date: Sun, 7 Feb 2010 09:59:07 -0500
Subject: [Freeswitch-users] ACL question and js error
In-Reply-To: <ea05d45dca9be2d979fb386f960bc077.squirrel@www.webcontracts.co.uk>
References: <ea05d45dca9be2d979fb386f960bc077.squirrel@www.webcontracts.co.uk>
Message-ID: <20100207145907.GF31942@base.carmickle.com>

On Sun, Feb 07, Matthew Law wrote:
> After some more experiments I have a working replacement for the asterisk
> box we were using before, which is great.
> 
> I had problems getting incoming calls to work. Changing the entry in
> acl.conf.xml from:
> 
> <list name="domains" default="deny">
>    <node type="allow" domain="$${domain}"/>
> </list>
> 
> to:
> 
> <list name="domains" default="allow">
>    <node type="allow" domain="$${domain}"/>
> </list>
> 
> and reloading xml works but this gets reverted every time FS starts up. 
> I've scanned the wiki docs and can't see anything pertaining to that. 
> Why/where is this happening and how do I make it the default?  Actually,
> the question should probably be is it sensible to do that? - the box is
> out on the internet and I really only want to take incoming calls from
> voiptalk.org, but I can't find a list of IPs on their site which I could
> create an acl from...

This is what gateway definitions are for in sofia.

> 
> Second question: I have tried this example for an answer machine (mainly
> because it looked the shortest and simplest of the examples listed):
> 
> http://wiki.freeswitch.org/wiki/Examples_answermachine

Is voicemail not what your looking for?

I understand the frustration of trying to get things working first run.  I found reading rereading and rereading the wiki to be most helpful.  You start to get a sense for how things work.  There are usually 100 ways to accomplish the same task in fs.  Over time you'll start to figure which ones work best for your setup.  You should jump in the weekly conf call.  Lots of people there can give you a hand.

--FC



From christian at officepools.com  Sun Feb  7 09:16:08 2010
From: christian at officepools.com (Christian Jensen)
Date: Sun, 7 Feb 2010 09:16:08 -0800
Subject: [Freeswitch-users] Looking for some good/cheap desktop phones
In-Reply-To: <00ae01caa698$65b570f0$312052d0$@net>
References: <3f2e071b5aea5e11bb37a37e435a33b7@mail.gmail.com>	
	<8e9d67561002042236s434e0643od322551de8720be1@mail.gmail.com>	
	<409D0ACF-ADA4-4D02-8B7E-71EC1B1BAFBC@freeswitch.org>	<8e9d67561002042301v76129abfj3dcbc70ed073752c@mail.gmail.com>	
	<d361ca7b1002050507n5e564621oa6cf7ace005dd5b8@mail.gmail.com> 
	<00ae01caa698$65b570f0$312052d0$@net>
Message-ID: <c7987ee2c6cbc7c0c148302dee8ee343@mail.gmail.com>

Where did you get these cheap phones?



*From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Adam Ford
*Sent:* Friday, February 05, 2010 11:21 AM
*To:* freeswitch-users at lists.freeswitch.org
*Subject:* Re: [Freeswitch-users] Looking for some good/cheap desktop phones



I just picked up old model Polycoms.  You can get the IP301?s for ~$60-70
new and the IP501s for ~$100 new.  They don?t have some of the fancier
features of the new Polycoms, but they carry the same quality and
configurability(with the exception of NAT).



-Adam



*From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Yehavi
Bourvine
*Sent:* Friday, February 05, 2010 6:07 AM
*To:* freeswitch-users at lists.freeswitch.org
*Subject:* Re: [Freeswitch-users] Looking for some good/cheap desktop phones



>From my experience Polycom and SNOM are expensive but give you what you
need. Polycom is more intutive to the users but more cumbersome for the
manager to deploy; SNOM is somewhat less intuitive to the user but
everything can be set via the WEB interface.



If you talk about 4-5 phones, then probably SNOM is the choice. It also
depends about the specific functions you want to use. I our specific
environment (high use of BLF and shared lines) Polycom wins because it
handles these functions just as the user expects.



I did not try Aastra so cannot testify. We did test Yealink, Thomson,
Asterphone, SipTip and maybe others I forgot. Cisco also seems good but
Cisco does not supply the required socumentation to make them fully working.



                    Regards, __Yehavi:

2010/2/5 ????? ??????? <tim at novion.ru>

Sure, those phones do not deliver superior usability, but they at
least give the best sound among budget models.


2010/2/5 Brian West <brian at freeswitch.org>:

> And all of those are awful phones.  They don't even make good paper
weights.
>
> You can't have good and cheap in the same sentence when talking about VoIP
phones.  You have to take your pick between quality (good) and price (cheap)
you can't have both at once.
>
> /b
>
>
> On Feb 5, 2010, at 12:36 AM, ????? ??????? wrote:
>
>> Have a look at Yealink (Skypemate) and Fanvill
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100207/4444caf4/attachment-0002.html 

From vmaruani at interwise.com  Sun Feb  7 07:00:55 2010
From: vmaruani at interwise.com (Victor Maruani)
Date: Sun, 7 Feb 2010 17:00:55 +0200
Subject: [Freeswitch-users] Bypass-media and REFER method
Message-ID: <B816EF85C732094BA2016B2112F3DC2B0402C777@isr-brass.interwise.com>

Hi,

 

I'm trying to do a POC using FS, the goal is to have FS handle REFERs
containing proprietary data. 

I want to have some logic on top of FS and also use the fail over
mechanism.

in short, I have something like this:

(third party) A side  --- FS ---- B side (IVR server)

 

the IVR the sends a REFER to FS. I don't want A to deal with it. 

now say B refers to C, it would be considered as a "group" C1, C2 ... to
which I want FS to failover.

only when one has answered should A be updated (REINVITE) and B notified
and disconnected.

if all fails I would expect B to be notified of the failure and proceed
as I wish without "losing" A.

 

from what I've read FS should be OK for the job but I have a couple
issues:

 

1 ) I have some issues getting FS handle a REFER while in bypass-media
mode.

(I tried with the release and some revisions including latest)

first when I bridge A and B everything is fine and media is bypassed.

When B sends REFER to C:

-           FS immediately NOTIFY B of success and send a reinvite to A
with SDP containing its own media IP/port.

-          then it does INVITE C with A's SDP.

-          B gets disconnected. A is not updated with C's sdp.

so at this point A sends RTP to FS and C sends RTP to A. ...

 

I basically have one extension for B: (set bypass-media and bridge to B)

and another extension to C which does the same actions.

what do you think I do wrong? 

 

 

2 ) how can I catch the REFER and set variables from it? (like ref-by or
ref-to)

in the dial plan I do catch the INVITE sent to C, but how to do it with
the REFER itself?

 

 

thanks for your help!

 

 

Best Regards,

Victor.

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100207/2992971b/attachment-0002.html 

From tculjaga at gmail.com  Sun Feb  7 13:17:40 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Sun, 7 Feb 2010 22:17:40 +0100
Subject: [Freeswitch-users] Looking for some good/cheap desktop phones
In-Reply-To: <20100206200547.GD31942@base.carmickle.com>
References: <3f2e071b5aea5e11bb37a37e435a33b7@mail.gmail.com>
	<8e9d67561002042236s434e0643od322551de8720be1@mail.gmail.com>
	<409D0ACF-ADA4-4D02-8B7E-71EC1B1BAFBC@freeswitch.org>
	<8e9d67561002042301v76129abfj3dcbc70ed073752c@mail.gmail.com>
	<d361ca7b1002050507n5e564621oa6cf7ace005dd5b8@mail.gmail.com>
	<00ae01caa698$65b570f0$312052d0$@net>
	<b39273531002051154h5132faa9h9fb024ebdfb15b77@mail.gmail.com>
	<SNT119-W1223C808A5E092C7E2A590B5540@phx.gbl>
	<e90351e71002061124v129b081cg501fb0c6ae6a17dd@mail.gmail.com>
	<20100206200547.GD31942@base.carmickle.com>
Message-ID: <65d96fc81002071317l69f8d8ccw8f6abee2f4265bd4@mail.gmail.com>

On Sat, Feb 6, 2010 at 9:05 PM, Frank Carmickle <frank at carmickle.com> wrote:

> On Sat, Feb 06, Christian Jensen wrote:
> > The response on this thread has been great!
> >
> > I wonder if there should be a link off the FreeSwitch site for vendors
> for
> > these phones.... hint hint.
> >
> > >From what I have been hearing here and elsewhere, the preference is
> Polycom,
> > Cisco, Grandstream, Aastra in that order.
>
> Hmmm...  That's not the impression that I got at all.  From what I picked
> up on this thread it would look more like
>
> polycom snom aastra cisco grandstream
>

Try Avaya 9600 SIP it is really great but it costs :)
It's always the same story ... there is not cheap and good ... it just
depends of what you can be satisfied for..
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100207/b1f6ec7e/attachment-0002.html 

From matt at webcontracts.co.uk  Sun Feb  7 14:23:15 2010
From: matt at webcontracts.co.uk (Matthew Law)
Date: Sun, 7 Feb 2010 22:23:15 -0000
Subject: [Freeswitch-users] ACL question and js error
In-Reply-To: <20100207145907.GF31942@base.carmickle.com>
References: <ea05d45dca9be2d979fb386f960bc077.squirrel@www.webcontracts.co.uk>
	<20100207145907.GF31942@base.carmickle.com>
Message-ID: <f14a5f708124b6c1ca692e2ef390749e.squirrel@www.webcontracts.co.uk>


On Sun, February 7, 2010 2:59 pm, Frank Carmickle wrote:
> This is what gateway definitions are for in sofia.
>
>>
>> Second question: I have tried this example for an answer machine (mainly
>> because it looked the shortest and simplest of the examples listed):
>>
>> http://wiki.freeswitch.org/wiki/Examples_answermachine
>
> Is voicemail not what your looking for?
>
> I understand the frustration of trying to get things working first run.  I
> found reading rereading and rereading the wiki to be most helpful.  You
> start to get a sense for how things work.  There are usually 100 ways to
> accomplish the same task in fs.  Over time you'll start to figure which
> ones work best for your setup.  You should jump in the weekly conf call.
> Lots of people there can give you a hand.

I'll read the wiki again :-)

What I would like to do for the moment is route all calls to extension 200
if it is logged in, ring for 20 seconds then go to the answermachine.  If
200 is not logged in, then go straight to answermachine.  Answermachine in
the current context could, I suppose, be a custom voicemail message for
200, but I do need it to be emailed to our group address.

Eventually I want several extensions in a group called 'support'.  Each
one with their own voicemail so they can receive messages from other
extensions but I want external callers to go to the 'answermachine'
voicemail as before.

What would be really cool is if I could wire that to RT so that it raises
a support ticket with the message attached if the caller ID was recognised
as one of our customers.  If not, it should just get emailed to our group
email address to be picked up by someone as per usual.

Yes, I am finding FS difficult mainly because all of the information is
there somewhere, it is just difficult to piece it all together.  One thing
is becoming clear: it is very powerful indeed and like many things in Unix
land, it is the way you can chain each individual component together that
makes it so capable.

Thanks,

Matt.






From Prometheus001 at gmx.net  Sun Feb  7 14:28:42 2010
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Sun, 07 Feb 2010 23:28:42 +0100
Subject: [Freeswitch-users] mod_skypiax - help needed with
 modified	snd-dummy (one way audio)
In-Reply-To: <191c3a031002060558r36947d93vcbb46df9e0a7dfde@mail.gmail.com>
References: <4B60555B.2020004@gmx.net>	<7b197bef1001281341h7ee2c58fj79b3886630901f29@mail.gmail.com>	<4B62C962.7000601@gmx.net>	<8213d6071001290403g2c08fe39gd92eb0446bb6485a@mail.gmail.com>	<7b197bef1001290520x37179cb1o63273410e9ef5bc@mail.gmail.com>	<4B6C1CCB.4080606@gmx.net>	<7b197bef1002050536j7ea09684v913434296ae52b97@mail.gmail.com>	<4B6C868B.3040406@gmx.net>	<7b197bef1002051319l2fcb5617p1691692f58c65372@mail.gmail.com>	<4B6CC792.5060608@gmx.net>
	<191c3a031002060558r36947d93vcbb46df9e0a7dfde@mail.gmail.com>
Message-ID: <4B6F3E9A.2020103@gmx.net>

I now used the static Skype binary in order to avoid missing constraints
to other libraries: It still crashes
1st it starts with:
  process 15431: D-Bus library appears to be incorrectly set up; failed
to read machine uuid: Failed to open "/var/lib/dbus/machine-id": No such
file or directory
  See the manual page for dbus-uuidgen to correct this issue.
After calling this client it crashes with:
   /usr/bin/skype: symbol lookup error: /usr/bin/skype: undefined
symbol: _ZN10QBoxLayout13addSpacerItemEP11QSpacerItem

Any hints, where I may get an older Skype client?

Best regards
Peter

Anthony Minessale schrieb:
>
> Quick fyi centos 5.4 has some memory bugs in libc don't go beyond 5.3
> until its fixed.
>
>> On Feb 5, 2010 7:42 PM, "Peter P GMX" <Prometheus001 at gmx.net
>> <mailto:Prometheus001 at gmx.net>> wrote:
>>
>> Skype starts, but as soon as it receives a call it crashes with:
>>
>> /usr/bin/skype: symbol lookup error: /usr/bin/skype: undefined symbol:
>>
>> _ZN10QBoxLayout13addSpacerItemEP11QSpacerItem
>>
>> I think the 8.10 version dos not work with8.04.
>>
>> Any hints, where I may get an older Skype client? I may also try the
>> static skype client.
>>
>>
>> Best regards
>> Peter
>>
>>
>>
>> .
>> Giovanni Maruzzelli schrieb:
>> > that's not at all a fatal error.
>> > I believe...
>>
> ------------------------------------------------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   



From wiltingtree at gmail.com  Sun Feb  7 18:31:40 2010
From: wiltingtree at gmail.com (Adam Wilt)
Date: Sun, 7 Feb 2010 21:31:40 -0500
Subject: [Freeswitch-users] Bridging sessions from two separate lua scripts;
	uuid_bridge? 	intercept?
Message-ID: <eb7797d71002071831w27301c5dva767b82b38c2ce6f@mail.gmail.com>

Hi. I have two sessions running in two separate Lua scripts, and I want to
bridge them so that the bridged call is being controlled by the first
(a-leg) script.
If I simply use uuid_bridge, I get no error but the calls don't bridge.
I've tried intercept, but I don't understand how it should be used; nothing
I try seems to work.
Here's what I have:

function bridge_calls(session,api,b_leg_uuid, call_len)
  session:setAutoHangup(false)
  session:execute("sched_hangup","+" .. tostring(call_len) .. " " ..
tostring(session.uuid))
  session:execute("set","continue_on_fail=true")
  api:executeString("intercept -bleg " .. tostring(b_leg_uuid))
  api:executeString("uuid_bridge " .. tostring(session.uuid) .. " " ..
tostring(b_leg_uuid))
end

I'd really appreciate any help.

Thanks,
Adam
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100207/115218cc/attachment-0002.html 

From msc at freeswitch.org  Sun Feb  7 18:50:33 2010
From: msc at freeswitch.org (Michael S Collins)
Date: Sun, 7 Feb 2010 20:50:33 -0600
Subject: [Freeswitch-users] Bridging sessions from two separate lua
	scripts; uuid_bridge? 	intercept?
In-Reply-To: <eb7797d71002071831w27301c5dva767b82b38c2ce6f@mail.gmail.com>
References: <eb7797d71002071831w27301c5dva767b82b38c2ce6f@mail.gmail.com>
Message-ID: <C23AA196-E851-4D5B-B1B5-2C2F32F6BB68@freeswitch.org>

Pastebin a debug log so we can see what is happening when the script  
runs.

-MC

Sent from my iPhone

On Feb 7, 2010, at 8:31 PM, Adam Wilt <wiltingtree at gmail.com> wrote:

> Hi. I have two sessions running in two separate Lua scripts, and I  
> want to bridge them so that the bridged call is being controlled by  
> the first (a-leg) script.
> If I simply use uuid_bridge, I get no error but the calls don't  
> bridge.
> I've tried intercept, but I don't understand how it should be used;  
> nothing I try seems to work.
> Here's what I have:
>
> function bridge_calls(session,api,b_leg_uuid, call_len)
>   session:setAutoHangup(false)
>   session:execute("sched_hangup","+" .. tostring(call_len) .. " " ..  
> tostring(session.uuid))
>   session:execute("set","continue_on_fail=true")
>   api:executeString("intercept -bleg " .. tostring(b_leg_uuid))
>   api:executeString("uuid_bridge " .. tostring(session.uuid) .. "  
> " .. tostring(b_leg_uuid))
> end
>
> I'd really appreciate any help.
>
> Thanks,
> Adam
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100207/655fb680/attachment-0002.html 

From wiltingtree at gmail.com  Sun Feb  7 19:29:30 2010
From: wiltingtree at gmail.com (Adam Wilt)
Date: Sun, 7 Feb 2010 22:29:30 -0500
Subject: [Freeswitch-users] Bridging sessions from two separate lua
	scripts; uuid_bridge? intercept?
In-Reply-To: <C23AA196-E851-4D5B-B1B5-2C2F32F6BB68@freeswitch.org>
References: <eb7797d71002071831w27301c5dva767b82b38c2ce6f@mail.gmail.com>
	<C23AA196-E851-4D5B-B1B5-2C2F32F6BB68@freeswitch.org>
Message-ID: <eb7797d71002071929pb53e1adue2e91e6bdb598251@mail.gmail.com>

Thanks Michael for the reply.
Here's the pastebin link: http://pastebin.freeswitch.org/12084


On Sun, Feb 7, 2010 at 9:50 PM, Michael S Collins <msc at freeswitch.org>wrote:

> Pastebin a debug log so we can see what is happening when the script runs.
>
> -MC
>
> Sent from my iPhone
>
> On Feb 7, 2010, at 8:31 PM, Adam Wilt <wiltingtree at gmail.com> wrote:
>
> Hi. I have two sessions running in two separate Lua scripts, and I want to
> bridge them so that the bridged call is being controlled by the first
> (a-leg) script.
> If I simply use uuid_bridge, I get no error but the calls don't bridge.
> I've tried intercept, but I don't understand how it should be used; nothing
> I try seems to work.
> Here's what I have:
>
> function bridge_calls(session,api,b_leg_uuid, call_len)
>   session:setAutoHangup(false)
>   session:execute("sched_hangup","+" .. tostring(call_len) .. " " ..
> tostring(session.uuid))
>   session:execute("set","continue_on_fail=true")
>   api:executeString("intercept -bleg " .. tostring(b_leg_uuid))
>   api:executeString("uuid_bridge " .. tostring(session.uuid) .. " " ..
> tostring(b_leg_uuid))
> end
>
> I'd really appreciate any help.
>
> Thanks,
> Adam
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100207/079392c4/attachment-0002.html 

From kristian.kielhofner at gmail.com  Sun Feb  7 21:06:03 2010
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Mon, 8 Feb 2010 00:06:03 -0500
Subject: [Freeswitch-users] mod_limit requires media?
Message-ID: <2d9149cd1002072106n7272cb80se3329700f0074d52@mail.gmail.com>

Hello everyone,

  I was playing with mod_limit earlier tonight and I noticed that it
essentially stopped hashing/tracking calls once bypass media was set.
Is this by design?  Is there some reason mod_limit requires media?
Other than that mod_limit looks to be very well implemented (no
surprise there) and I'm excited to put it to more use.

Thanks!

-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com



From codecomplete at free.fr  Mon Feb  8 03:07:05 2010
From: codecomplete at free.fr (Fred-145)
Date: Mon, 08 Feb 2010 12:07:05 +0100
Subject: [Freeswitch-users] Driving peripherals through Freeswitch?
Message-ID: <2srvm5945qgcno44oetn9ngii0u3aed73p@4ax.com>

Hello

I don't know anything about this, but I was wondering if someone had
successfully used a Freeswitch server to drive peripherals like
switching on a heater by sending an SMS or calling an extension, etc.?

I'm thinking of tools like X10 to drive peripherals from a PC.

Has someone played with this kind of tool and could tell me what is
technically possible?

Thank you for any feedback.




From codecomplete at free.fr  Mon Feb  8 03:14:31 2010
From: codecomplete at free.fr (Fred-145)
Date: Mon, 08 Feb 2010 12:14:31 +0100
Subject: [Freeswitch-users] Looking for some good/cheap desktop phones
References: <3f2e071b5aea5e11bb37a37e435a33b7@mail.gmail.com>
Message-ID: <2asvm5105hn2d6pdtdqlnrd8hmc3btg80a@4ax.com>

On Thu, 4 Feb 2010 14:08:56 -0800, Christain Jensen
<christian at officepools.com> wrote:
>I am looking for a vendor for some (3-5) desktop voip phones. Any
>suggestions?

Based on positive feedback from VoIP forums, I bought the DECT Siemens
A580IP a couple of weeks ago for about 70? (sales tax excluded).

Don't know how well it compares to higher-end solutions, but it's been
working fine so far once I was told how to solve some compatibility
issue to get it working with Eyebeam/XLite.




From codecomplete at free.fr  Mon Feb  8 03:10:55 2010
From: codecomplete at free.fr (Fred-145)
Date: Mon, 08 Feb 2010 12:10:55 +0100
Subject: [Freeswitch-users] Looking for some good/cheap desktop phones
References: <8e9d67561002042236s434e0643od322551de8720be1@mail.gmail.com>
	<409D0ACF-ADA4-4D02-8B7E-71EC1B1BAFBC@freeswitch.org>
	<8e9d67561002042301v76129abfj3dcbc70ed073752c@mail.gmail.com>
	<d361ca7b1002050507n5e564621oa6cf7ace005dd5b8@mail.gmail.com>
	<65d96fc81002050521w746707b1q4398e6d8a0692d1e@mail.gmail.com>
	<65d96fc81002050521w746707b1q4398e6d8a0692d1e-JsoAwUIsXosN+BqQ9rBEUg@public.gmane.org>
	<6B520D1B-EDD6-44F7-9DE5-A384D3286138@freeswitch.org>
Message-ID: <u6svm5t18meap4cd6t673o03f8gvosvgos@4ax.com>

On Fri, 5 Feb 2010 09:13:29 -0600, Brian West
<brian at freeswitch.org> wrote:
>Sigh... When is someone actually going to build an open platform voip hardware phone

I guess David Rowe is the person who could pull this off

http://www.rowetel.com/blog/?page_id=2




From mike at yes.net.ua  Mon Feb  8 03:19:09 2010
From: mike at yes.net.ua (Mike Tkachuk)
Date: Mon, 8 Feb 2010 13:19:09 +0200
Subject: [Freeswitch-users] Dialplan search order
Message-ID: <798899361.20100208131909@yes.net.ua>

Hello FS gurus,

  I'm using xml_curl for external dialplan fetch, but I like to
  split static and dynamic parts of configuration, so for example,
  leave call unloop logic, and voicemail extension in static xml file
  while having all other parts dynamic. It will allow to avoid
  unnecessary call to costly external source and also avoid xml parse
  of content that is static.

  Currently FS first look in xml_curl and only after that falls back
  to static files. Is that behavior possible to change, so FS will work
  like that:

  1 - Look in static xml file and execute all extensions that have
  'continue="true"'
  2 - If previous step didn't stop on matching extension than look in
  xml_curl or other source specified in dialplan param of sofia
  config.

  Looks like <param name="dialplan" value="XML:/path_to_dialplan_default.xml,XML"/>
  don't do the trick.

  I'm sorry if my question is really noobish or goes against FS
  logic. Thanks in advance.


--
Mike Tkachuk




From gmaruzz at celliax.org  Mon Feb  8 05:29:36 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Mon, 8 Feb 2010 14:29:36 +0100
Subject: [Freeswitch-users] mod_skypiax - help needed with modified
	snd-dummy (one way audio)
In-Reply-To: <4B6F3E9A.2020103@gmx.net>
References: <4B60555B.2020004@gmx.net>
	<8213d6071001290403g2c08fe39gd92eb0446bb6485a@mail.gmail.com> 
	<7b197bef1001290520x37179cb1o63273410e9ef5bc@mail.gmail.com> 
	<4B6C1CCB.4080606@gmx.net>
	<7b197bef1002050536j7ea09684v913434296ae52b97@mail.gmail.com> 
	<4B6C868B.3040406@gmx.net>
	<7b197bef1002051319l2fcb5617p1691692f58c65372@mail.gmail.com> 
	<4B6CC792.5060608@gmx.net>
	<191c3a031002060558r36947d93vcbb46df9e0a7dfde@mail.gmail.com> 
	<4B6F3E9A.2020103@gmx.net>
Message-ID: <7b197bef1002080529u1afc7f9av492b0c221ee2d6af@mail.gmail.com>

Peter,

I just tested with the static build you find on skype.com

I never tested for performances or other issues, there may be (it's a beta).

But it do not crash on me.

I have no problem at all.

If you can give me ssh access I can try to understand why you have so
many problems.

Or, alternatively, try to follow the wiki. You know, I've not heard
about those problems.

root at hardy64:~/skype.beta/skype_static-2.1.0.81# ldd skype
        linux-gate.so.1 =>  (0xffffe000)
        libasound.so.2 => /usr/lib32/libasound.so.2 (0xf7eb7000)
        libXv.so.1 => /usr/lib32/libXv.so.1 (0xf7eb2000)
        libXss.so.1 => /usr/lib32/libXss.so.1 (0xf7eae000)
        libSM.so.6 => /usr/lib32/libSM.so.6 (0xf7ea6000)
        libICE.so.6 => /usr/lib32/libICE.so.6 (0xf7e8e000)
        libXi.so.6 => /usr/lib32/libXi.so.6 (0xf7e86000)
        libXrender.so.1 => /usr/lib32/libXrender.so.1 (0xf7e7e000)
        libXrandr.so.2 => /usr/lib32/libXrandr.so.2 (0xf7e78000)
        libfreetype.so.6 => /usr/lib32/libfreetype.so.6 (0xf7e0a000)
        libfontconfig.so.1 => /usr/lib32/libfontconfig.so.1 (0xf7de0000)
        libXext.so.6 => /usr/lib32/libXext.so.6 (0xf7dd2000)
        libX11.so.6 => /usr/lib32/libX11.so.6 (0xf7ceb000)
        libz.so.1 => /usr/lib32/libz.so.1 (0xf7cd6000)
        libgthread-2.0.so.0 => /usr/lib32/libgthread-2.0.so.0 (0xf7cd1000)
        libglib-2.0.so.0 => /usr/lib32/libglib-2.0.so.0 (0xf7c1f000)
        librt.so.1 => /lib32/librt.so.1 (0xf7c16000)
        libdl.so.2 => /lib32/libdl.so.2 (0xf7c12000)
        libpthread.so.0 => /lib32/libpthread.so.0 (0xf7bfa000)
        libstdc++.so.6 => /usr/lib32/libstdc++.so.6 (0xf7b07000)
        libm.so.6 => /lib32/libm.so.6 (0xf7ae2000)
        libgcc_s.so.1 => /usr/lib32/libgcc_s.so.1 (0xf7ad6000)
        libc.so.6 => /lib32/libc.so.6 (0xf7987000)
        libexpat.so.1 => /usr/lib32/libexpat.so.1 (0xf7966000)
        libXau.so.6 => /usr/lib32/libXau.so.6 (0xf7963000)
        libxcb-xlib.so.0 => /usr/lib32/libxcb-xlib.so.0 (0xf7961000)
        libxcb.so.1 => /usr/lib32/libxcb.so.1 (0xf7948000)
        libpcre.so.3 => /usr/lib32/libpcre.so.3 (0xf7921000)
        /lib/ld-linux.so.2 (0xf7f86000)
        libXdmcp.so.6 => /usr/lib32/libXdmcp.so.6 (0xf791c000)


On Sun, Feb 7, 2010 at 11:28 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
> I now used the static Skype binary in order to avoid missing constraints
> to other libraries: It still crashes
> 1st it starts with:
> ?process 15431: D-Bus library appears to be incorrectly set up; failed
> to read machine uuid: Failed to open "/var/lib/dbus/machine-id": No such
> file or directory
> ?See the manual page for dbus-uuidgen to correct this issue.
> After calling this client it crashes with:
> ? /usr/bin/skype: symbol lookup error: /usr/bin/skype: undefined
> symbol: _ZN10QBoxLayout13addSpacerItemEP11QSpacerItem
>
> Any hints, where I may get an older Skype client?
>
> Best regards
> Peter
>
> Anthony Minessale schrieb:
>>
>> Quick fyi centos 5.4 has some memory bugs in libc don't go beyond 5.3
>> until its fixed.
>>
>>> On Feb 5, 2010 7:42 PM, "Peter P GMX" <Prometheus001 at gmx.net
>>> <mailto:Prometheus001 at gmx.net>> wrote:
>>>
>>> Skype starts, but as soon as it receives a call it crashes with:
>>>
>>> /usr/bin/skype: symbol lookup error: /usr/bin/skype: undefined symbol:
>>>
>>> _ZN10QBoxLayout13addSpacerItemEP11QSpacerItem
>>>
>>> I think the 8.10 version dos not work with8.04.
>>>
>>> Any hints, where I may get an older Skype client? I may also try the
>>> static skype client.
>>>
>>>
>>> Best regards
>>> Peter
>>>
>>>
>>>
>>> .
>>> Giovanni Maruzzelli schrieb:
>>> > that's not at all a fatal error.
>>> > I believe...
>>>
>> ------------------------------------------------------------------------
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618



From max.bridgewater at gmail.com  Mon Feb  8 05:46:42 2010
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Mon, 8 Feb 2010 08:46:42 -0500
Subject: [Freeswitch-users] mod_skypiax - help needed with modified
	snd-dummy (one way audio)
In-Reply-To: <7b197bef1002080529u1afc7f9av492b0c221ee2d6af@mail.gmail.com>
References: <4B60555B.2020004@gmx.net>
	<7b197bef1001290520x37179cb1o63273410e9ef5bc@mail.gmail.com>
	<4B6C1CCB.4080606@gmx.net>
	<7b197bef1002050536j7ea09684v913434296ae52b97@mail.gmail.com>
	<4B6C868B.3040406@gmx.net>
	<7b197bef1002051319l2fcb5617p1691692f58c65372@mail.gmail.com>
	<4B6CC792.5060608@gmx.net>
	<191c3a031002060558r36947d93vcbb46df9e0a7dfde@mail.gmail.com>
	<4B6F3E9A.2020103@gmx.net>
	<7b197bef1002080529u1afc7f9av492b0c221ee2d6af@mail.gmail.com>
Message-ID: <c55d46951002080546p829a881w256bfdb19334c548@mail.gmail.com>

Interesting; a while back I tried to install Skypiax with the latest
static build on Skype.com. I had QT library compatibility problem on a
CentOS 5.4 (Final). Giovannni, may I ask what Linux distro you are
using?

Thanks,
max.

On Mon, Feb 8, 2010 at 8:29 AM, Giovanni Maruzzelli <gmaruzz at celliax.org> wrote:
> Peter,
>
> I just tested with the static build you find on skype.com
>
> I never tested for performances or other issues, there may be (it's a beta).
>
> But it do not crash on me.
>
> I have no problem at all.
>
> If you can give me ssh access I can try to understand why you have so
> many problems.
>
> Or, alternatively, try to follow the wiki. You know, I've not heard
> about those problems.
>
> root at hardy64:~/skype.beta/skype_static-2.1.0.81# ldd skype
> ? ? ? ?linux-gate.so.1 => ?(0xffffe000)
> ? ? ? ?libasound.so.2 => /usr/lib32/libasound.so.2 (0xf7eb7000)
> ? ? ? ?libXv.so.1 => /usr/lib32/libXv.so.1 (0xf7eb2000)
> ? ? ? ?libXss.so.1 => /usr/lib32/libXss.so.1 (0xf7eae000)
> ? ? ? ?libSM.so.6 => /usr/lib32/libSM.so.6 (0xf7ea6000)
> ? ? ? ?libICE.so.6 => /usr/lib32/libICE.so.6 (0xf7e8e000)
> ? ? ? ?libXi.so.6 => /usr/lib32/libXi.so.6 (0xf7e86000)
> ? ? ? ?libXrender.so.1 => /usr/lib32/libXrender.so.1 (0xf7e7e000)
> ? ? ? ?libXrandr.so.2 => /usr/lib32/libXrandr.so.2 (0xf7e78000)
> ? ? ? ?libfreetype.so.6 => /usr/lib32/libfreetype.so.6 (0xf7e0a000)
> ? ? ? ?libfontconfig.so.1 => /usr/lib32/libfontconfig.so.1 (0xf7de0000)
> ? ? ? ?libXext.so.6 => /usr/lib32/libXext.so.6 (0xf7dd2000)
> ? ? ? ?libX11.so.6 => /usr/lib32/libX11.so.6 (0xf7ceb000)
> ? ? ? ?libz.so.1 => /usr/lib32/libz.so.1 (0xf7cd6000)
> ? ? ? ?libgthread-2.0.so.0 => /usr/lib32/libgthread-2.0.so.0 (0xf7cd1000)
> ? ? ? ?libglib-2.0.so.0 => /usr/lib32/libglib-2.0.so.0 (0xf7c1f000)
> ? ? ? ?librt.so.1 => /lib32/librt.so.1 (0xf7c16000)
> ? ? ? ?libdl.so.2 => /lib32/libdl.so.2 (0xf7c12000)
> ? ? ? ?libpthread.so.0 => /lib32/libpthread.so.0 (0xf7bfa000)
> ? ? ? ?libstdc++.so.6 => /usr/lib32/libstdc++.so.6 (0xf7b07000)
> ? ? ? ?libm.so.6 => /lib32/libm.so.6 (0xf7ae2000)
> ? ? ? ?libgcc_s.so.1 => /usr/lib32/libgcc_s.so.1 (0xf7ad6000)
> ? ? ? ?libc.so.6 => /lib32/libc.so.6 (0xf7987000)
> ? ? ? ?libexpat.so.1 => /usr/lib32/libexpat.so.1 (0xf7966000)
> ? ? ? ?libXau.so.6 => /usr/lib32/libXau.so.6 (0xf7963000)
> ? ? ? ?libxcb-xlib.so.0 => /usr/lib32/libxcb-xlib.so.0 (0xf7961000)
> ? ? ? ?libxcb.so.1 => /usr/lib32/libxcb.so.1 (0xf7948000)
> ? ? ? ?libpcre.so.3 => /usr/lib32/libpcre.so.3 (0xf7921000)
> ? ? ? ?/lib/ld-linux.so.2 (0xf7f86000)
> ? ? ? ?libXdmcp.so.6 => /usr/lib32/libXdmcp.so.6 (0xf791c000)
>
>
> On Sun, Feb 7, 2010 at 11:28 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>> I now used the static Skype binary in order to avoid missing constraints
>> to other libraries: It still crashes
>> 1st it starts with:
>> ?process 15431: D-Bus library appears to be incorrectly set up; failed
>> to read machine uuid: Failed to open "/var/lib/dbus/machine-id": No such
>> file or directory
>> ?See the manual page for dbus-uuidgen to correct this issue.
>> After calling this client it crashes with:
>> ? /usr/bin/skype: symbol lookup error: /usr/bin/skype: undefined
>> symbol: _ZN10QBoxLayout13addSpacerItemEP11QSpacerItem
>>
>> Any hints, where I may get an older Skype client?
>>
>> Best regards
>> Peter
>>
>> Anthony Minessale schrieb:
>>>
>>> Quick fyi centos 5.4 has some memory bugs in libc don't go beyond 5.3
>>> until its fixed.
>>>
>>>> On Feb 5, 2010 7:42 PM, "Peter P GMX" <Prometheus001 at gmx.net
>>>> <mailto:Prometheus001 at gmx.net>> wrote:
>>>>
>>>> Skype starts, but as soon as it receives a call it crashes with:
>>>>
>>>> /usr/bin/skype: symbol lookup error: /usr/bin/skype: undefined symbol:
>>>>
>>>> _ZN10QBoxLayout13addSpacerItemEP11QSpacerItem
>>>>
>>>> I think the 8.10 version dos not work with8.04.
>>>>
>>>> Any hints, where I may get an older Skype client? I may also try the
>>>> static skype client.
>>>>
>>>>
>>>> Best regards
>>>> Peter
>>>>
>>>>
>>>>
>>>> .
>>>> Giovanni Maruzzelli schrieb:
>>>> > that's not at all a fatal error.
>>>> > I believe...
>>>>
>>> ------------------------------------------------------------------------
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From gmaruzz at celliax.org  Mon Feb  8 05:53:04 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Mon, 8 Feb 2010 14:53:04 +0100
Subject: [Freeswitch-users] mod_skypiax - help needed with modified
	snd-dummy (one way audio)
In-Reply-To: <c55d46951002080546p829a881w256bfdb19334c548@mail.gmail.com>
References: <4B60555B.2020004@gmx.net> <4B6C1CCB.4080606@gmx.net> 
	<7b197bef1002050536j7ea09684v913434296ae52b97@mail.gmail.com> 
	<4B6C868B.3040406@gmx.net>
	<7b197bef1002051319l2fcb5617p1691692f58c65372@mail.gmail.com> 
	<4B6CC792.5060608@gmx.net>
	<191c3a031002060558r36947d93vcbb46df9e0a7dfde@mail.gmail.com> 
	<4B6F3E9A.2020103@gmx.net>
	<7b197bef1002080529u1afc7f9av492b0c221ee2d6af@mail.gmail.com> 
	<c55d46951002080546p829a881w256bfdb19334c548@mail.gmail.com>
Message-ID: <7b197bef1002080553qd1cb42dla26425942de83c43@mail.gmail.com>

Peter is using hardy 64 bit. I checked on that.

But, let me understand: if you're using a static build, why you have a
problem with QT?
Is actually Qt to be statically linked...

what is the results of:

ldd skype

Anyway, in a short while I'll check that on centos 5.4 too, and I'll report here

-giovanni

On Mon, Feb 8, 2010 at 2:46 PM, Max Bridgewater
<max.bridgewater at gmail.com> wrote:
> Interesting; a while back I tried to install Skypiax with the latest
> static build on Skype.com. I had QT library compatibility problem on a
> CentOS 5.4 (Final). Giovannni, may I ask what Linux distro you are
> using?
>
> Thanks,
> max.
>
> On Mon, Feb 8, 2010 at 8:29 AM, Giovanni Maruzzelli <gmaruzz at celliax.org> wrote:
>> Peter,
>>
>> I just tested with the static build you find on skype.com
>>
>> I never tested for performances or other issues, there may be (it's a beta).
>>
>> But it do not crash on me.
>>
>> I have no problem at all.
>>
>> If you can give me ssh access I can try to understand why you have so
>> many problems.
>>
>> Or, alternatively, try to follow the wiki. You know, I've not heard
>> about those problems.
>>
>> root at hardy64:~/skype.beta/skype_static-2.1.0.81# ldd skype
>> ? ? ? ?linux-gate.so.1 => ?(0xffffe000)
>> ? ? ? ?libasound.so.2 => /usr/lib32/libasound.so.2 (0xf7eb7000)
>> ? ? ? ?libXv.so.1 => /usr/lib32/libXv.so.1 (0xf7eb2000)
>> ? ? ? ?libXss.so.1 => /usr/lib32/libXss.so.1 (0xf7eae000)
>> ? ? ? ?libSM.so.6 => /usr/lib32/libSM.so.6 (0xf7ea6000)
>> ? ? ? ?libICE.so.6 => /usr/lib32/libICE.so.6 (0xf7e8e000)
>> ? ? ? ?libXi.so.6 => /usr/lib32/libXi.so.6 (0xf7e86000)
>> ? ? ? ?libXrender.so.1 => /usr/lib32/libXrender.so.1 (0xf7e7e000)
>> ? ? ? ?libXrandr.so.2 => /usr/lib32/libXrandr.so.2 (0xf7e78000)
>> ? ? ? ?libfreetype.so.6 => /usr/lib32/libfreetype.so.6 (0xf7e0a000)
>> ? ? ? ?libfontconfig.so.1 => /usr/lib32/libfontconfig.so.1 (0xf7de0000)
>> ? ? ? ?libXext.so.6 => /usr/lib32/libXext.so.6 (0xf7dd2000)
>> ? ? ? ?libX11.so.6 => /usr/lib32/libX11.so.6 (0xf7ceb000)
>> ? ? ? ?libz.so.1 => /usr/lib32/libz.so.1 (0xf7cd6000)
>> ? ? ? ?libgthread-2.0.so.0 => /usr/lib32/libgthread-2.0.so.0 (0xf7cd1000)
>> ? ? ? ?libglib-2.0.so.0 => /usr/lib32/libglib-2.0.so.0 (0xf7c1f000)
>> ? ? ? ?librt.so.1 => /lib32/librt.so.1 (0xf7c16000)
>> ? ? ? ?libdl.so.2 => /lib32/libdl.so.2 (0xf7c12000)
>> ? ? ? ?libpthread.so.0 => /lib32/libpthread.so.0 (0xf7bfa000)
>> ? ? ? ?libstdc++.so.6 => /usr/lib32/libstdc++.so.6 (0xf7b07000)
>> ? ? ? ?libm.so.6 => /lib32/libm.so.6 (0xf7ae2000)
>> ? ? ? ?libgcc_s.so.1 => /usr/lib32/libgcc_s.so.1 (0xf7ad6000)
>> ? ? ? ?libc.so.6 => /lib32/libc.so.6 (0xf7987000)
>> ? ? ? ?libexpat.so.1 => /usr/lib32/libexpat.so.1 (0xf7966000)
>> ? ? ? ?libXau.so.6 => /usr/lib32/libXau.so.6 (0xf7963000)
>> ? ? ? ?libxcb-xlib.so.0 => /usr/lib32/libxcb-xlib.so.0 (0xf7961000)
>> ? ? ? ?libxcb.so.1 => /usr/lib32/libxcb.so.1 (0xf7948000)
>> ? ? ? ?libpcre.so.3 => /usr/lib32/libpcre.so.3 (0xf7921000)
>> ? ? ? ?/lib/ld-linux.so.2 (0xf7f86000)
>> ? ? ? ?libXdmcp.so.6 => /usr/lib32/libXdmcp.so.6 (0xf791c000)
>>
>>
>> On Sun, Feb 7, 2010 at 11:28 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>> I now used the static Skype binary in order to avoid missing constraints
>>> to other libraries: It still crashes
>>> 1st it starts with:
>>> ?process 15431: D-Bus library appears to be incorrectly set up; failed
>>> to read machine uuid: Failed to open "/var/lib/dbus/machine-id": No such
>>> file or directory
>>> ?See the manual page for dbus-uuidgen to correct this issue.
>>> After calling this client it crashes with:
>>> ? /usr/bin/skype: symbol lookup error: /usr/bin/skype: undefined
>>> symbol: _ZN10QBoxLayout13addSpacerItemEP11QSpacerItem
>>>
>>> Any hints, where I may get an older Skype client?
>>>
>>> Best regards
>>> Peter
>>>
>>> Anthony Minessale schrieb:
>>>>
>>>> Quick fyi centos 5.4 has some memory bugs in libc don't go beyond 5.3
>>>> until its fixed.
>>>>
>>>>> On Feb 5, 2010 7:42 PM, "Peter P GMX" <Prometheus001 at gmx.net
>>>>> <mailto:Prometheus001 at gmx.net>> wrote:
>>>>>
>>>>> Skype starts, but as soon as it receives a call it crashes with:
>>>>>
>>>>> /usr/bin/skype: symbol lookup error: /usr/bin/skype: undefined symbol:
>>>>>
>>>>> _ZN10QBoxLayout13addSpacerItemEP11QSpacerItem
>>>>>
>>>>> I think the 8.10 version dos not work with8.04.
>>>>>
>>>>> Any hints, where I may get an older Skype client? I may also try the
>>>>> static skype client.
>>>>>
>>>>>
>>>>> Best regards
>>>>> Peter
>>>>>
>>>>>
>>>>>
>>>>> .
>>>>> Giovanni Maruzzelli schrieb:
>>>>> > that's not at all a fatal error.
>>>>> > I believe...
>>>>>
>>>> ------------------------------------------------------------------------
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> Cell : +39-347-2665618
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618



From max.bridgewater at gmail.com  Mon Feb  8 06:04:15 2010
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Mon, 8 Feb 2010 09:04:15 -0500
Subject: [Freeswitch-users] mod_skypiax - help needed with modified
	snd-dummy (one way audio)
In-Reply-To: <7b197bef1002080553qd1cb42dla26425942de83c43@mail.gmail.com>
References: <4B60555B.2020004@gmx.net>
	<7b197bef1002050536j7ea09684v913434296ae52b97@mail.gmail.com>
	<4B6C868B.3040406@gmx.net>
	<7b197bef1002051319l2fcb5617p1691692f58c65372@mail.gmail.com>
	<4B6CC792.5060608@gmx.net>
	<191c3a031002060558r36947d93vcbb46df9e0a7dfde@mail.gmail.com>
	<4B6F3E9A.2020103@gmx.net>
	<7b197bef1002080529u1afc7f9av492b0c221ee2d6af@mail.gmail.com>
	<c55d46951002080546p829a881w256bfdb19334c548@mail.gmail.com>
	<7b197bef1002080553qd1cb42dla26425942de83c43@mail.gmail.com>
Message-ID: <c55d46951002080604k400344b0i8add00b5c6014958@mail.gmail.com>

Well, it was a couple of weeks ago. I then switched back to 2.0.4. I'm
going to try it again and let you know.

Max.

On Mon, Feb 8, 2010 at 8:53 AM, Giovanni Maruzzelli <gmaruzz at celliax.org> wrote:
> Peter is using hardy 64 bit. I checked on that.
>
> But, let me understand: if you're using a static build, why you have a
> problem with QT?
> Is actually Qt to be statically linked...
>
> what is the results of:
>
> ldd skype
>
> Anyway, in a short while I'll check that on centos 5.4 too, and I'll report here
>
> -giovanni
>
> On Mon, Feb 8, 2010 at 2:46 PM, Max Bridgewater
> <max.bridgewater at gmail.com> wrote:
>> Interesting; a while back I tried to install Skypiax with the latest
>> static build on Skype.com. I had QT library compatibility problem on a
>> CentOS 5.4 (Final). Giovannni, may I ask what Linux distro you are
>> using?
>>
>> Thanks,
>> max.
>>
>> On Mon, Feb 8, 2010 at 8:29 AM, Giovanni Maruzzelli <gmaruzz at celliax.org> wrote:
>>> Peter,
>>>
>>> I just tested with the static build you find on skype.com
>>>
>>> I never tested for performances or other issues, there may be (it's a beta).
>>>
>>> But it do not crash on me.
>>>
>>> I have no problem at all.
>>>
>>> If you can give me ssh access I can try to understand why you have so
>>> many problems.
>>>
>>> Or, alternatively, try to follow the wiki. You know, I've not heard
>>> about those problems.
>>>
>>> root at hardy64:~/skype.beta/skype_static-2.1.0.81# ldd skype
>>> ? ? ? ?linux-gate.so.1 => ?(0xffffe000)
>>> ? ? ? ?libasound.so.2 => /usr/lib32/libasound.so.2 (0xf7eb7000)
>>> ? ? ? ?libXv.so.1 => /usr/lib32/libXv.so.1 (0xf7eb2000)
>>> ? ? ? ?libXss.so.1 => /usr/lib32/libXss.so.1 (0xf7eae000)
>>> ? ? ? ?libSM.so.6 => /usr/lib32/libSM.so.6 (0xf7ea6000)
>>> ? ? ? ?libICE.so.6 => /usr/lib32/libICE.so.6 (0xf7e8e000)
>>> ? ? ? ?libXi.so.6 => /usr/lib32/libXi.so.6 (0xf7e86000)
>>> ? ? ? ?libXrender.so.1 => /usr/lib32/libXrender.so.1 (0xf7e7e000)
>>> ? ? ? ?libXrandr.so.2 => /usr/lib32/libXrandr.so.2 (0xf7e78000)
>>> ? ? ? ?libfreetype.so.6 => /usr/lib32/libfreetype.so.6 (0xf7e0a000)
>>> ? ? ? ?libfontconfig.so.1 => /usr/lib32/libfontconfig.so.1 (0xf7de0000)
>>> ? ? ? ?libXext.so.6 => /usr/lib32/libXext.so.6 (0xf7dd2000)
>>> ? ? ? ?libX11.so.6 => /usr/lib32/libX11.so.6 (0xf7ceb000)
>>> ? ? ? ?libz.so.1 => /usr/lib32/libz.so.1 (0xf7cd6000)
>>> ? ? ? ?libgthread-2.0.so.0 => /usr/lib32/libgthread-2.0.so.0 (0xf7cd1000)
>>> ? ? ? ?libglib-2.0.so.0 => /usr/lib32/libglib-2.0.so.0 (0xf7c1f000)
>>> ? ? ? ?librt.so.1 => /lib32/librt.so.1 (0xf7c16000)
>>> ? ? ? ?libdl.so.2 => /lib32/libdl.so.2 (0xf7c12000)
>>> ? ? ? ?libpthread.so.0 => /lib32/libpthread.so.0 (0xf7bfa000)
>>> ? ? ? ?libstdc++.so.6 => /usr/lib32/libstdc++.so.6 (0xf7b07000)
>>> ? ? ? ?libm.so.6 => /lib32/libm.so.6 (0xf7ae2000)
>>> ? ? ? ?libgcc_s.so.1 => /usr/lib32/libgcc_s.so.1 (0xf7ad6000)
>>> ? ? ? ?libc.so.6 => /lib32/libc.so.6 (0xf7987000)
>>> ? ? ? ?libexpat.so.1 => /usr/lib32/libexpat.so.1 (0xf7966000)
>>> ? ? ? ?libXau.so.6 => /usr/lib32/libXau.so.6 (0xf7963000)
>>> ? ? ? ?libxcb-xlib.so.0 => /usr/lib32/libxcb-xlib.so.0 (0xf7961000)
>>> ? ? ? ?libxcb.so.1 => /usr/lib32/libxcb.so.1 (0xf7948000)
>>> ? ? ? ?libpcre.so.3 => /usr/lib32/libpcre.so.3 (0xf7921000)
>>> ? ? ? ?/lib/ld-linux.so.2 (0xf7f86000)
>>> ? ? ? ?libXdmcp.so.6 => /usr/lib32/libXdmcp.so.6 (0xf791c000)
>>>
>>>
>>> On Sun, Feb 7, 2010 at 11:28 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>> I now used the static Skype binary in order to avoid missing constraints
>>>> to other libraries: It still crashes
>>>> 1st it starts with:
>>>> ?process 15431: D-Bus library appears to be incorrectly set up; failed
>>>> to read machine uuid: Failed to open "/var/lib/dbus/machine-id": No such
>>>> file or directory
>>>> ?See the manual page for dbus-uuidgen to correct this issue.
>>>> After calling this client it crashes with:
>>>> ? /usr/bin/skype: symbol lookup error: /usr/bin/skype: undefined
>>>> symbol: _ZN10QBoxLayout13addSpacerItemEP11QSpacerItem
>>>>
>>>> Any hints, where I may get an older Skype client?
>>>>
>>>> Best regards
>>>> Peter
>>>>
>>>> Anthony Minessale schrieb:
>>>>>
>>>>> Quick fyi centos 5.4 has some memory bugs in libc don't go beyond 5.3
>>>>> until its fixed.
>>>>>
>>>>>> On Feb 5, 2010 7:42 PM, "Peter P GMX" <Prometheus001 at gmx.net
>>>>>> <mailto:Prometheus001 at gmx.net>> wrote:
>>>>>>
>>>>>> Skype starts, but as soon as it receives a call it crashes with:
>>>>>>
>>>>>> /usr/bin/skype: symbol lookup error: /usr/bin/skype: undefined symbol:
>>>>>>
>>>>>> _ZN10QBoxLayout13addSpacerItemEP11QSpacerItem
>>>>>>
>>>>>> I think the 8.10 version dos not work with8.04.
>>>>>>
>>>>>> Any hints, where I may get an older Skype client? I may also try the
>>>>>> static skype client.
>>>>>>
>>>>>>
>>>>>> Best regards
>>>>>> Peter
>>>>>>
>>>>>>
>>>>>>
>>>>>> .
>>>>>> Giovanni Maruzzelli schrieb:
>>>>>> > that's not at all a fatal error.
>>>>>> > I believe...
>>>>>>
>>>>> ------------------------------------------------------------------------
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>> Sincerely,
>>>
>>> Giovanni Maruzzelli
>>> Cell : +39-347-2665618
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From max.bridgewater at gmail.com  Mon Feb  8 07:06:22 2010
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Mon, 8 Feb 2010 10:06:22 -0500
Subject: [Freeswitch-users] CentOS 5.3 (Final) with Skype Static Build
Message-ID: <c55d46951002080706t6bb8531i4f12cb35af12e6de@mail.gmail.com>

Hi Giovanni,

Let me start a new thread so I don't hijack Peter's one. Here is what
I get for ldd /usr/bin/skype. What is the deal with these libXv.so.1,
libXss.so.1?

        linux-gate.so.1 =>  (0x00697000)
        libasound.so.2 => /lib/libasound.so.2 (0x008b7000)
        libXv.so.1 => not found
        libXss.so.1 => not found
        libSM.so.6 => /usr/lib/libSM.so.6 (0x00d02000)
        libICE.so.6 => /usr/lib/libICE.so.6 (0x0036b000)
        libXi.so.6 => /usr/lib/libXi.so.6 (0x00111000)
        libXrender.so.1 => /usr/lib/libXrender.so.1 (0x00119000)
        libXrandr.so.2 => /usr/lib/libXrandr.so.2 (0x00b77000)
        libfreetype.so.6 => /usr/lib/libfreetype.so.6 (0x00122000)
        libfontconfig.so.1 => /usr/lib/libfontconfig.so.1 (0x009fb000)
        libXext.so.6 => /usr/lib/libXext.so.6 (0x00eb0000)
        libX11.so.6 => /usr/lib/libX11.so.6 (0x001a2000)
        libz.so.1 => /usr/lib/libz.so.1 (0x002a5000)
        libgthread-2.0.so.0 => /lib/libgthread-2.0.so.0 (0x006bd000)
        libglib-2.0.so.0 => /lib/libglib-2.0.so.0 (0x002be000)
        librt.so.1 => /lib/librt.so.1 (0x009a1000)
        libdl.so.2 => /lib/libdl.so.2 (0x00604000)
        libpthread.so.0 => /lib/libpthread.so.0 (0x004e6000)
        libstdc++.so.6 => /usr/lib/libstdc++.so.6 (0x004fd000)
        libm.so.6 => /lib/libm.so.6 (0x00385000)
        libgcc_s.so.1 => /lib/libgcc_s.so.1 (0x00bcc000)
        libc.so.6 => /lib/libc.so.6 (0x006c2000)
        libexpat.so.0 => /lib/libexpat.so.0 (0x00c2a000)
        libXau.so.6 => /usr/lib/libXau.so.6 (0x002ba000)
        libXdmcp.so.6 => /usr/lib/libXdmcp.so.6 (0x0087e000)
        /lib/ld-linux.so.2 (0x00476000)

Max



From m.sobkow at marketelsystems.com  Mon Feb  8 07:06:55 2010
From: m.sobkow at marketelsystems.com (Mark Sobkow)
Date: Mon, 08 Feb 2010 09:06:55 -0600
Subject: [Freeswitch-users] IVR and Erlang
Message-ID: <4B70288F.6040003@marketelsystems.com>

I'm having a little trouble with Erlang and IVRs.  Specifically, the IVR 
configuration file is auto-loaded before the initialization code has a 
chance to tell Freeswitch to get it's configs from Erlang.  As a result, 
it's not querying Erlang for the configuration file.

Also, because there is no mod_ivr, I can't just do a "reload mod_ivr" 
after the Erlang communications has been initialized.

Any idea how to get Freeswitch to load it's IVR configs from Erlang?

-- 
Mark Sobkow
Senior Developer
MarkeTel Multi-Line Dialing Systems LTD.
428 Victoria Ave
Regina, SK S4N-0P6
Toll-Free: 800-289-8616-X533
Local: 306-359-6893-X533
Fax: 306-359-6879
Email: m.sobkow at marketelsystems.com
Web: http://www.marketelsystems.com 




From wiltingtree at gmail.com  Mon Feb  8 07:37:59 2010
From: wiltingtree at gmail.com (Adam Wilt)
Date: Mon, 8 Feb 2010 10:37:59 -0500
Subject: [Freeswitch-users] Bridging sessions from two separate lua
	scripts; uuid_bridge? intercept?
In-Reply-To: <eb7797d71002071929pb53e1adue2e91e6bdb598251@mail.gmail.com>
References: <eb7797d71002071831w27301c5dva767b82b38c2ce6f@mail.gmail.com>
	<C23AA196-E851-4D5B-B1B5-2C2F32F6BB68@freeswitch.org>
	<eb7797d71002071929pb53e1adue2e91e6bdb598251@mail.gmail.com>
Message-ID: <eb7797d71002080737k3f8fa5c9m8bf3ef2fcf42881e@mail.gmail.com>

One other thing I should mention. I'm running FreeSWITCH version 1.4 (build
14460) in Windows.
Brian suggested I upgrade to the build in the
http://files-sync.freeswitch.org/windows_installer/  folder, but it turned
out to be the exact same build I already had. I'd love to try upgrade to 1.5
in case this problem has been fixed already.


On Sun, Feb 7, 2010 at 10:29 PM, Adam Wilt <wiltingtree at gmail.com> wrote:

> Thanks Michael for the reply.
> Here's the pastebin link: http://pastebin.freeswitch.org/12084
>
>
> On Sun, Feb 7, 2010 at 9:50 PM, Michael S Collins <msc at freeswitch.org>wrote:
>
>> Pastebin a debug log so we can see what is happening when the script
>> runs.
>>
>> -MC
>>
>> Sent from my iPhone
>>
>> On Feb 7, 2010, at 8:31 PM, Adam Wilt <wiltingtree at gmail.com> wrote:
>>
>> Hi. I have two sessions running in two separate Lua scripts, and I want to
>> bridge them so that the bridged call is being controlled by the first
>> (a-leg) script.
>> If I simply use uuid_bridge, I get no error but the calls don't bridge.
>> I've tried intercept, but I don't understand how it should be used;
>> nothing I try seems to work.
>> Here's what I have:
>>
>> function bridge_calls(session,api,b_leg_uuid, call_len)
>>   session:setAutoHangup(false)
>>   session:execute("sched_hangup","+" .. tostring(call_len) .. " " ..
>> tostring(session.uuid))
>>   session:execute("set","continue_on_fail=true")
>>   api:executeString("intercept -bleg " .. tostring(b_leg_uuid))
>>   api:executeString("uuid_bridge " .. tostring(session.uuid) .. " " ..
>> tostring(b_leg_uuid))
>> end
>>
>> I'd really appreciate any help.
>>
>> Thanks,
>> Adam
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100208/e88d0652/attachment-0002.html 

From m.sobkow at marketelsystems.com  Mon Feb  8 08:17:37 2010
From: m.sobkow at marketelsystems.com (Mark Sobkow)
Date: Mon, 08 Feb 2010 10:17:37 -0600
Subject: [Freeswitch-users] IVR and Erlang
In-Reply-To: <4B70288F.6040003@marketelsystems.com>
References: <4B70288F.6040003@marketelsystems.com>
Message-ID: <4B703921.4080909@marketelsystems.com>

Mark Sobkow wrote:
> I'm having a little trouble with Erlang and IVRs.  Specifically, the IVR 
> configuration file is auto-loaded before the initialization code has a 
> chance to tell Freeswitch to get it's configs from Erlang.  As a result, 
> it's not querying Erlang for the configuration file.
>
> Also, because there is no mod_ivr, I can't just do a "reload mod_ivr" 
> after the Erlang communications has been initialized.
>
> Any idea how to get Freeswitch to load it's IVR configs from Erlang?
>
>   
Mucking about this morning, I tried doing a "reloadxml", but that only 
seems to reload the timezones.conf from Erlang.

-- 
Mark Sobkow
Senior Developer
MarkeTel Multi-Line Dialing Systems LTD.
428 Victoria Ave
Regina, SK S4N-0P6
Toll-Free: 800-289-8616-X533
Local: 306-359-6893-X533
Fax: 306-359-6879
Email: m.sobkow at marketelsystems.com
Web: http://www.marketelsystems.com 




From Prometheus001 at gmx.net  Mon Feb  8 08:24:48 2010
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Mon, 08 Feb 2010 17:24:48 +0100
Subject: [Freeswitch-users] mod_skypiax - help needed with
 modified	snd-dummy (one way audio)
In-Reply-To: <c55d46951002080604k400344b0i8add00b5c6014958@mail.gmail.com>
References: <4B60555B.2020004@gmx.net>	<7b197bef1002050536j7ea09684v913434296ae52b97@mail.gmail.com>	<4B6C868B.3040406@gmx.net>	<7b197bef1002051319l2fcb5617p1691692f58c65372@mail.gmail.com>	<4B6CC792.5060608@gmx.net>	<191c3a031002060558r36947d93vcbb46df9e0a7dfde@mail.gmail.com>	<4B6F3E9A.2020103@gmx.net>	<7b197bef1002080529u1afc7f9av492b0c221ee2d6af@mail.gmail.com>	<c55d46951002080546p829a881w256bfdb19334c548@mail.gmail.com>	<7b197bef1002080553qd1cb42dla26425942de83c43@mail.gmail.com>
	<c55d46951002080604k400344b0i8add00b5c6014958@mail.gmail.com>
Message-ID: <4B703AD0.2080909@gmx.net>

I got it working now with static build and an older version of skype
(skype_static-2.1.0.47).

But I still have a problem ongoing with sound quality, resp. one way audio:
With original Alsa driver: When Skype calls mod_skypiax => SIP Phone,
then the sound from the SIP phone is interupted regularly 2 times a second.
Example:
When a person on the sip phone speaks
"aaaaaaaaaaaaaaaaaaaaaa"
the other site hears
"aaatataaaatataaaatataaaa"
With "t" meaning the interruption of the sound.

So I compliled and installed the modified alsa driver as described in
the wiki (configure, make and make install, remove old ubuntu sound dir
in /lib/modules/2.6.24-24-server/ubuntu/sound + depmod -a + reboot the
server.

Now the SIP phone is heard loud and clearly without interruption.
However the other direction is not heard, so we're at the beginning of
the post (one way audio). Only when I really scratch the microphone then
I hear some parts of this scratching on the SIP side.

So, some more hints are needed.

Here's the log, when I start the skype client:
  su root -c "/bin/echo 'username password'| DISPLAY=:101 
/usr/bin/skype1 --pipelogin &" &
  /usr/bin/Xvfb :102 -ac &
  error opening security policy file /etc/X11/xserver/SecurityPolicy
  expected keysym, got XF86KbdLightOnOff: line 70 of pc
  expected keysym, got XF86KbdBrightnessDown: line 71 of pc 
  expected keysym, got XF86KbdBrightnessUp: line 72 of pc
  Could not init font path element /usr/share/fonts/X11/cyrillic,
removing from list!
  Could not init font path element
/var/lib/defoma/x-ttcidfont-conf.d/dirs/TrueType, removing from list!
But these messages are not critical, right?

Best regards
Peter


Max Bridgewater schrieb:
> Well, it was a couple of weeks ago. I then switched back to 2.0.4. I'm
> going to try it again and let you know.
>
> Max.
>
> On Mon, Feb 8, 2010 at 8:53 AM, Giovanni Maruzzelli <gmaruzz at celliax.org> wrote:
>   
>> Peter is using hardy 64 bit. I checked on that.
>>
>> But, let me understand: if you're using a static build, why you have a
>> problem with QT?
>> Is actually Qt to be statically linked...
>>
>> what is the results of:
>>
>> ldd skype
>>
>> Anyway, in a short while I'll check that on centos 5.4 too, and I'll report here
>>
>> -giovanni
>>
>> On Mon, Feb 8, 2010 at 2:46 PM, Max Bridgewater
>> <max.bridgewater at gmail.com> wrote:
>>     
>>> Interesting; a while back I tried to install Skypiax with the latest
>>> static build on Skype.com. I had QT library compatibility problem on a
>>> CentOS 5.4 (Final). Giovannni, may I ask what Linux distro you are
>>> using?
>>>
>>> Thanks,
>>> max.
>>>
>>> On Mon, Feb 8, 2010 at 8:29 AM, Giovanni Maruzzelli <gmaruzz at celliax.org> wrote:
>>>       
>>>> Peter,
>>>>
>>>> I just tested with the static build you find on skype.com
>>>>
>>>> I never tested for performances or other issues, there may be (it's a beta).
>>>>
>>>> But it do not crash on me.
>>>>
>>>> I have no problem at all.
>>>>
>>>> If you can give me ssh access I can try to understand why you have so
>>>> many problems.
>>>>
>>>> Or, alternatively, try to follow the wiki. You know, I've not heard
>>>> about those problems.
>>>>
>>>> root at hardy64:~/skype.beta/skype_static-2.1.0.81# ldd skype
>>>>        linux-gate.so.1 =>  (0xffffe000)
>>>>        libasound.so.2 => /usr/lib32/libasound.so.2 (0xf7eb7000)
>>>>        libXv.so.1 => /usr/lib32/libXv.so.1 (0xf7eb2000)
>>>>        libXss.so.1 => /usr/lib32/libXss.so.1 (0xf7eae000)
>>>>        libSM.so.6 => /usr/lib32/libSM.so.6 (0xf7ea6000)
>>>>        libICE.so.6 => /usr/lib32/libICE.so.6 (0xf7e8e000)
>>>>        libXi.so.6 => /usr/lib32/libXi.so.6 (0xf7e86000)
>>>>        libXrender.so.1 => /usr/lib32/libXrender.so.1 (0xf7e7e000)
>>>>        libXrandr.so.2 => /usr/lib32/libXrandr.so.2 (0xf7e78000)
>>>>        libfreetype.so.6 => /usr/lib32/libfreetype.so.6 (0xf7e0a000)
>>>>        libfontconfig.so.1 => /usr/lib32/libfontconfig.so.1 (0xf7de0000)
>>>>        libXext.so.6 => /usr/lib32/libXext.so.6 (0xf7dd2000)
>>>>        libX11.so.6 => /usr/lib32/libX11.so.6 (0xf7ceb000)
>>>>        libz.so.1 => /usr/lib32/libz.so.1 (0xf7cd6000)
>>>>        libgthread-2.0.so.0 => /usr/lib32/libgthread-2.0.so.0 (0xf7cd1000)
>>>>        libglib-2.0.so.0 => /usr/lib32/libglib-2.0.so.0 (0xf7c1f000)
>>>>        librt.so.1 => /lib32/librt.so.1 (0xf7c16000)
>>>>        libdl.so.2 => /lib32/libdl.so.2 (0xf7c12000)
>>>>        libpthread.so.0 => /lib32/libpthread.so.0 (0xf7bfa000)
>>>>        libstdc++.so.6 => /usr/lib32/libstdc++.so.6 (0xf7b07000)
>>>>        libm.so.6 => /lib32/libm.so.6 (0xf7ae2000)
>>>>        libgcc_s.so.1 => /usr/lib32/libgcc_s.so.1 (0xf7ad6000)
>>>>        libc.so.6 => /lib32/libc.so.6 (0xf7987000)
>>>>        libexpat.so.1 => /usr/lib32/libexpat.so.1 (0xf7966000)
>>>>        libXau.so.6 => /usr/lib32/libXau.so.6 (0xf7963000)
>>>>        libxcb-xlib.so.0 => /usr/lib32/libxcb-xlib.so.0 (0xf7961000)
>>>>        libxcb.so.1 => /usr/lib32/libxcb.so.1 (0xf7948000)
>>>>        libpcre.so.3 => /usr/lib32/libpcre.so.3 (0xf7921000)
>>>>        /lib/ld-linux.so.2 (0xf7f86000)
>>>>        libXdmcp.so.6 => /usr/lib32/libXdmcp.so.6 (0xf791c000)
>>>>
>>>>
>>>> On Sun, Feb 7, 2010 at 11:28 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>         
>>>>> I now used the static Skype binary in order to avoid missing constraints
>>>>> to other libraries: It still crashes
>>>>> 1st it starts with:
>>>>>  process 15431: D-Bus library appears to be incorrectly set up; failed
>>>>> to read machine uuid: Failed to open "/var/lib/dbus/machine-id": No such
>>>>> file or directory
>>>>>  See the manual page for dbus-uuidgen to correct this issue.
>>>>> After calling this client it crashes with:
>>>>>   /usr/bin/skype: symbol lookup error: /usr/bin/skype: undefined
>>>>> symbol: _ZN10QBoxLayout13addSpacerItemEP11QSpacerItem
>>>>>
>>>>> Any hints, where I may get an older Skype client?
>>>>>
>>>>> Best regards
>>>>> Peter
>>>>>
>>>>> Anthony Minessale schrieb:
>>>>>           
>>>>>> Quick fyi centos 5.4 has some memory bugs in libc don't go beyond 5.3
>>>>>> until its fixed.
>>>>>>
>>>>>>             
>>>>>>> On Feb 5, 2010 7:42 PM, "Peter P GMX" <Prometheus001 at gmx.net
>>>>>>> <mailto:Prometheus001 at gmx.net>> wrote:
>>>>>>>
>>>>>>> Skype starts, but as soon as it receives a call it crashes with:
>>>>>>>
>>>>>>> /usr/bin/skype: symbol lookup error: /usr/bin/skype: undefined symbol:
>>>>>>>
>>>>>>> _ZN10QBoxLayout13addSpacerItemEP11QSpacerItem
>>>>>>>
>>>>>>> I think the 8.10 version dos not work with8.04.
>>>>>>>
>>>>>>> Any hints, where I may get an older Skype client? I may also try the
>>>>>>> static skype client.
>>>>>>>
>>>>>>>
>>>>>>> Best regards
>>>>>>> Peter
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> .
>>>>>>> Giovanni Maruzzelli schrieb:
>>>>>>>               
>>>>>>>> that's not at all a fatal error.
>>>>>>>> I believe...
>>>>>>>>                 
>>>>>> ------------------------------------------------------------------------
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>             
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>           
>>>>
>>>> --
>>>> Sincerely,
>>>>
>>>> Giovanni Maruzzelli
>>>> Cell : +39-347-2665618
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>         
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>       
>>
>> --
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> Cell : +39-347-2665618
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>     
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>   



From gmaruzz at celliax.org  Mon Feb  8 08:39:54 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Mon, 8 Feb 2010 17:39:54 +0100
Subject: [Freeswitch-users] mod_skypiax - help needed with modified
	snd-dummy (one way audio)
In-Reply-To: <4B703AD0.2080909@gmx.net>
References: <4B60555B.2020004@gmx.net>
	<7b197bef1002051319l2fcb5617p1691692f58c65372@mail.gmail.com> 
	<4B6CC792.5060608@gmx.net>
	<191c3a031002060558r36947d93vcbb46df9e0a7dfde@mail.gmail.com> 
	<4B6F3E9A.2020103@gmx.net>
	<7b197bef1002080529u1afc7f9av492b0c221ee2d6af@mail.gmail.com> 
	<c55d46951002080546p829a881w256bfdb19334c548@mail.gmail.com> 
	<7b197bef1002080553qd1cb42dla26425942de83c43@mail.gmail.com> 
	<c55d46951002080604k400344b0i8add00b5c6014958@mail.gmail.com> 
	<4B703AD0.2080909@gmx.net>
Message-ID: <7b197bef1002080839o61b4b1b5k6d5cc39bb05d9a32@mail.gmail.com>

Peter,

excuse me but I really do not follow you.

Why you have the normal static build not working?

Also, this is really taking too much of my time. You continue to
change things, and report issues, without waiting for solutions you
ask for, then you report something else, and so on... we'll never get
at the end of this.

If you want, please connect via IRC, and contact me (gmaruzz).
Or let me connect ssh to your machine.

-gm


On Mon, Feb 8, 2010 at 5:24 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
> I got it working now with static build and an older version of skype
> (skype_static-2.1.0.47).
>
> But I still have a problem ongoing with sound quality, resp. one way audio:
> With original Alsa driver: When Skype calls mod_skypiax => SIP Phone,
> then the sound from the SIP phone is interupted regularly 2 times a second.
> Example:
> When a person on the sip phone speaks
> "aaaaaaaaaaaaaaaaaaaaaa"
> the other site hears
> "aaatataaaatataaaatataaaa"
> With "t" meaning the interruption of the sound.
>
> So I compliled and installed the modified alsa driver as described in
> the wiki (configure, make and make install, remove old ubuntu sound dir
> in /lib/modules/2.6.24-24-server/ubuntu/sound + depmod -a + reboot the
> server.
>
> Now the SIP phone is heard loud and clearly without interruption.
> However the other direction is not heard, so we're at the beginning of
> the post (one way audio). Only when I really scratch the microphone then
> I hear some parts of this scratching on the SIP side.
>
> So, some more hints are needed.
>
> Here's the log, when I start the skype client:
> ?su root -c "/bin/echo 'username password'| DISPLAY=:101
> /usr/bin/skype1 --pipelogin &" &
> ?/usr/bin/Xvfb :102 -ac &
> ?error opening security policy file /etc/X11/xserver/SecurityPolicy
> ?expected keysym, got XF86KbdLightOnOff: line 70 of pc
> ?expected keysym, got XF86KbdBrightnessDown: line 71 of pc
> ?expected keysym, got XF86KbdBrightnessUp: line 72 of pc
> ?Could not init font path element /usr/share/fonts/X11/cyrillic,
> removing from list!
> ?Could not init font path element
> /var/lib/defoma/x-ttcidfont-conf.d/dirs/TrueType, removing from list!
> But these messages are not critical, right?
>
> Best regards
> Peter
>
>
> Max Bridgewater schrieb:
>> Well, it was a couple of weeks ago. I then switched back to 2.0.4. I'm
>> going to try it again and let you know.
>>
>> Max.
>>
>> On Mon, Feb 8, 2010 at 8:53 AM, Giovanni Maruzzelli <gmaruzz at celliax.org> wrote:
>>
>>> Peter is using hardy 64 bit. I checked on that.
>>>
>>> But, let me understand: if you're using a static build, why you have a
>>> problem with QT?
>>> Is actually Qt to be statically linked...
>>>
>>> what is the results of:
>>>
>>> ldd skype
>>>
>>> Anyway, in a short while I'll check that on centos 5.4 too, and I'll report here
>>>
>>> -giovanni
>>>
>>> On Mon, Feb 8, 2010 at 2:46 PM, Max Bridgewater
>>> <max.bridgewater at gmail.com> wrote:
>>>
>>>> Interesting; a while back I tried to install Skypiax with the latest
>>>> static build on Skype.com. I had QT library compatibility problem on a
>>>> CentOS 5.4 (Final). Giovannni, may I ask what Linux distro you are
>>>> using?
>>>>
>>>> Thanks,
>>>> max.
>>>>
>>>> On Mon, Feb 8, 2010 at 8:29 AM, Giovanni Maruzzelli <gmaruzz at celliax.org> wrote:
>>>>
>>>>> Peter,
>>>>>
>>>>> I just tested with the static build you find on skype.com
>>>>>
>>>>> I never tested for performances or other issues, there may be (it's a beta).
>>>>>
>>>>> But it do not crash on me.
>>>>>
>>>>> I have no problem at all.
>>>>>
>>>>> If you can give me ssh access I can try to understand why you have so
>>>>> many problems.
>>>>>
>>>>> Or, alternatively, try to follow the wiki. You know, I've not heard
>>>>> about those problems.
>>>>>
>>>>> root at hardy64:~/skype.beta/skype_static-2.1.0.81# ldd skype
>>>>> ? ? ? ?linux-gate.so.1 => ?(0xffffe000)
>>>>> ? ? ? ?libasound.so.2 => /usr/lib32/libasound.so.2 (0xf7eb7000)
>>>>> ? ? ? ?libXv.so.1 => /usr/lib32/libXv.so.1 (0xf7eb2000)
>>>>> ? ? ? ?libXss.so.1 => /usr/lib32/libXss.so.1 (0xf7eae000)
>>>>> ? ? ? ?libSM.so.6 => /usr/lib32/libSM.so.6 (0xf7ea6000)
>>>>> ? ? ? ?libICE.so.6 => /usr/lib32/libICE.so.6 (0xf7e8e000)
>>>>> ? ? ? ?libXi.so.6 => /usr/lib32/libXi.so.6 (0xf7e86000)
>>>>> ? ? ? ?libXrender.so.1 => /usr/lib32/libXrender.so.1 (0xf7e7e000)
>>>>> ? ? ? ?libXrandr.so.2 => /usr/lib32/libXrandr.so.2 (0xf7e78000)
>>>>> ? ? ? ?libfreetype.so.6 => /usr/lib32/libfreetype.so.6 (0xf7e0a000)
>>>>> ? ? ? ?libfontconfig.so.1 => /usr/lib32/libfontconfig.so.1 (0xf7de0000)
>>>>> ? ? ? ?libXext.so.6 => /usr/lib32/libXext.so.6 (0xf7dd2000)
>>>>> ? ? ? ?libX11.so.6 => /usr/lib32/libX11.so.6 (0xf7ceb000)
>>>>> ? ? ? ?libz.so.1 => /usr/lib32/libz.so.1 (0xf7cd6000)
>>>>> ? ? ? ?libgthread-2.0.so.0 => /usr/lib32/libgthread-2.0.so.0 (0xf7cd1000)
>>>>> ? ? ? ?libglib-2.0.so.0 => /usr/lib32/libglib-2.0.so.0 (0xf7c1f000)
>>>>> ? ? ? ?librt.so.1 => /lib32/librt.so.1 (0xf7c16000)
>>>>> ? ? ? ?libdl.so.2 => /lib32/libdl.so.2 (0xf7c12000)
>>>>> ? ? ? ?libpthread.so.0 => /lib32/libpthread.so.0 (0xf7bfa000)
>>>>> ? ? ? ?libstdc++.so.6 => /usr/lib32/libstdc++.so.6 (0xf7b07000)
>>>>> ? ? ? ?libm.so.6 => /lib32/libm.so.6 (0xf7ae2000)
>>>>> ? ? ? ?libgcc_s.so.1 => /usr/lib32/libgcc_s.so.1 (0xf7ad6000)
>>>>> ? ? ? ?libc.so.6 => /lib32/libc.so.6 (0xf7987000)
>>>>> ? ? ? ?libexpat.so.1 => /usr/lib32/libexpat.so.1 (0xf7966000)
>>>>> ? ? ? ?libXau.so.6 => /usr/lib32/libXau.so.6 (0xf7963000)
>>>>> ? ? ? ?libxcb-xlib.so.0 => /usr/lib32/libxcb-xlib.so.0 (0xf7961000)
>>>>> ? ? ? ?libxcb.so.1 => /usr/lib32/libxcb.so.1 (0xf7948000)
>>>>> ? ? ? ?libpcre.so.3 => /usr/lib32/libpcre.so.3 (0xf7921000)
>>>>> ? ? ? ?/lib/ld-linux.so.2 (0xf7f86000)
>>>>> ? ? ? ?libXdmcp.so.6 => /usr/lib32/libXdmcp.so.6 (0xf791c000)
>>>>>
>>>>>
>>>>> On Sun, Feb 7, 2010 at 11:28 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>
>>>>>> I now used the static Skype binary in order to avoid missing constraints
>>>>>> to other libraries: It still crashes
>>>>>> 1st it starts with:
>>>>>> ?process 15431: D-Bus library appears to be incorrectly set up; failed
>>>>>> to read machine uuid: Failed to open "/var/lib/dbus/machine-id": No such
>>>>>> file or directory
>>>>>> ?See the manual page for dbus-uuidgen to correct this issue.
>>>>>> After calling this client it crashes with:
>>>>>> ? /usr/bin/skype: symbol lookup error: /usr/bin/skype: undefined
>>>>>> symbol: _ZN10QBoxLayout13addSpacerItemEP11QSpacerItem
>>>>>>
>>>>>> Any hints, where I may get an older Skype client?
>>>>>>
>>>>>> Best regards
>>>>>> Peter
>>>>>>
>>>>>> Anthony Minessale schrieb:
>>>>>>
>>>>>>> Quick fyi centos 5.4 has some memory bugs in libc don't go beyond 5.3
>>>>>>> until its fixed.
>>>>>>>
>>>>>>>
>>>>>>>> On Feb 5, 2010 7:42 PM, "Peter P GMX" <Prometheus001 at gmx.net
>>>>>>>> <mailto:Prometheus001 at gmx.net>> wrote:
>>>>>>>>
>>>>>>>> Skype starts, but as soon as it receives a call it crashes with:
>>>>>>>>
>>>>>>>> /usr/bin/skype: symbol lookup error: /usr/bin/skype: undefined symbol:
>>>>>>>>
>>>>>>>> _ZN10QBoxLayout13addSpacerItemEP11QSpacerItem
>>>>>>>>
>>>>>>>> I think the 8.10 version dos not work with8.04.
>>>>>>>>
>>>>>>>> Any hints, where I may get an older Skype client? I may also try the
>>>>>>>> static skype client.
>>>>>>>>
>>>>>>>>
>>>>>>>> Best regards
>>>>>>>> Peter
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> .
>>>>>>>> Giovanni Maruzzelli schrieb:
>>>>>>>>
>>>>>>>>> that's not at all a fatal error.
>>>>>>>>> I believe...
>>>>>>>>>
>>>>>>> ------------------------------------------------------------------------
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>> --
>>>>> Sincerely,
>>>>>
>>>>> Giovanni Maruzzelli
>>>>> Cell : +39-347-2665618
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> --
>>> Sincerely,
>>>
>>> Giovanni Maruzzelli
>>> Cell : +39-347-2665618
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618



From gmaruzz at celliax.org  Mon Feb  8 08:42:49 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Mon, 8 Feb 2010 17:42:49 +0100
Subject: [Freeswitch-users] mod_skypiax - help needed with modified
	snd-dummy (one way audio)
In-Reply-To: <c55d46951002080604k400344b0i8add00b5c6014958@mail.gmail.com>
References: <4B60555B.2020004@gmx.net> <4B6C868B.3040406@gmx.net> 
	<7b197bef1002051319l2fcb5617p1691692f58c65372@mail.gmail.com> 
	<4B6CC792.5060608@gmx.net>
	<191c3a031002060558r36947d93vcbb46df9e0a7dfde@mail.gmail.com> 
	<4B6F3E9A.2020103@gmx.net>
	<7b197bef1002080529u1afc7f9av492b0c221ee2d6af@mail.gmail.com> 
	<c55d46951002080546p829a881w256bfdb19334c548@mail.gmail.com> 
	<7b197bef1002080553qd1cb42dla26425942de83c43@mail.gmail.com> 
	<c55d46951002080604k400344b0i8add00b5c6014958@mail.gmail.com>
Message-ID: <7b197bef1002080842v7a970d00l19df72d4f48a6ef6@mail.gmail.com>

Max,

Just checked, the static build of skype-beta works on latest centos (5.4)

you have to install:

yum install libXv
yum install libXScrnSaver

then use the snd-dummy supplied with centos (the modified one do not
works at the moment - because the skype-beta does some alsa calls was
not doing in previous versions, that are not implemented, I'll fix it
soon).

-gm

On Mon, Feb 8, 2010 at 3:04 PM, Max Bridgewater
<max.bridgewater at gmail.com> wrote:
> Well, it was a couple of weeks ago. I then switched back to 2.0.4. I'm
> going to try it again and let you know.
>
> Max.
>
> On Mon, Feb 8, 2010 at 8:53 AM, Giovanni Maruzzelli <gmaruzz at celliax.org> wrote:
>> Peter is using hardy 64 bit. I checked on that.
>>
>> But, let me understand: if you're using a static build, why you have a
>> problem with QT?
>> Is actually Qt to be statically linked...
>>
>> what is the results of:
>>
>> ldd skype
>>
>> Anyway, in a short while I'll check that on centos 5.4 too, and I'll report here
>>
>> -giovanni
>>
>> On Mon, Feb 8, 2010 at 2:46 PM, Max Bridgewater
>> <max.bridgewater at gmail.com> wrote:
>>> Interesting; a while back I tried to install Skypiax with the latest
>>> static build on Skype.com. I had QT library compatibility problem on a
>>> CentOS 5.4 (Final). Giovannni, may I ask what Linux distro you are
>>> using?
>>>
>>> Thanks,
>>> max.
>>>
>>> On Mon, Feb 8, 2010 at 8:29 AM, Giovanni Maruzzelli <gmaruzz at celliax.org> wrote:
>>>> Peter,
>>>>
>>>> I just tested with the static build you find on skype.com
>>>>
>>>> I never tested for performances or other issues, there may be (it's a beta).
>>>>
>>>> But it do not crash on me.
>>>>
>>>> I have no problem at all.
>>>>
>>>> If you can give me ssh access I can try to understand why you have so
>>>> many problems.
>>>>
>>>> Or, alternatively, try to follow the wiki. You know, I've not heard
>>>> about those problems.
>>>>
>>>> root at hardy64:~/skype.beta/skype_static-2.1.0.81# ldd skype
>>>> ? ? ? ?linux-gate.so.1 => ?(0xffffe000)
>>>> ? ? ? ?libasound.so.2 => /usr/lib32/libasound.so.2 (0xf7eb7000)
>>>> ? ? ? ?libXv.so.1 => /usr/lib32/libXv.so.1 (0xf7eb2000)
>>>> ? ? ? ?libXss.so.1 => /usr/lib32/libXss.so.1 (0xf7eae000)
>>>> ? ? ? ?libSM.so.6 => /usr/lib32/libSM.so.6 (0xf7ea6000)
>>>> ? ? ? ?libICE.so.6 => /usr/lib32/libICE.so.6 (0xf7e8e000)
>>>> ? ? ? ?libXi.so.6 => /usr/lib32/libXi.so.6 (0xf7e86000)
>>>> ? ? ? ?libXrender.so.1 => /usr/lib32/libXrender.so.1 (0xf7e7e000)
>>>> ? ? ? ?libXrandr.so.2 => /usr/lib32/libXrandr.so.2 (0xf7e78000)
>>>> ? ? ? ?libfreetype.so.6 => /usr/lib32/libfreetype.so.6 (0xf7e0a000)
>>>> ? ? ? ?libfontconfig.so.1 => /usr/lib32/libfontconfig.so.1 (0xf7de0000)
>>>> ? ? ? ?libXext.so.6 => /usr/lib32/libXext.so.6 (0xf7dd2000)
>>>> ? ? ? ?libX11.so.6 => /usr/lib32/libX11.so.6 (0xf7ceb000)
>>>> ? ? ? ?libz.so.1 => /usr/lib32/libz.so.1 (0xf7cd6000)
>>>> ? ? ? ?libgthread-2.0.so.0 => /usr/lib32/libgthread-2.0.so.0 (0xf7cd1000)
>>>> ? ? ? ?libglib-2.0.so.0 => /usr/lib32/libglib-2.0.so.0 (0xf7c1f000)
>>>> ? ? ? ?librt.so.1 => /lib32/librt.so.1 (0xf7c16000)
>>>> ? ? ? ?libdl.so.2 => /lib32/libdl.so.2 (0xf7c12000)
>>>> ? ? ? ?libpthread.so.0 => /lib32/libpthread.so.0 (0xf7bfa000)
>>>> ? ? ? ?libstdc++.so.6 => /usr/lib32/libstdc++.so.6 (0xf7b07000)
>>>> ? ? ? ?libm.so.6 => /lib32/libm.so.6 (0xf7ae2000)
>>>> ? ? ? ?libgcc_s.so.1 => /usr/lib32/libgcc_s.so.1 (0xf7ad6000)
>>>> ? ? ? ?libc.so.6 => /lib32/libc.so.6 (0xf7987000)
>>>> ? ? ? ?libexpat.so.1 => /usr/lib32/libexpat.so.1 (0xf7966000)
>>>> ? ? ? ?libXau.so.6 => /usr/lib32/libXau.so.6 (0xf7963000)
>>>> ? ? ? ?libxcb-xlib.so.0 => /usr/lib32/libxcb-xlib.so.0 (0xf7961000)
>>>> ? ? ? ?libxcb.so.1 => /usr/lib32/libxcb.so.1 (0xf7948000)
>>>> ? ? ? ?libpcre.so.3 => /usr/lib32/libpcre.so.3 (0xf7921000)
>>>> ? ? ? ?/lib/ld-linux.so.2 (0xf7f86000)
>>>> ? ? ? ?libXdmcp.so.6 => /usr/lib32/libXdmcp.so.6 (0xf791c000)
>>>>
>>>>
>>>> On Sun, Feb 7, 2010 at 11:28 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>> I now used the static Skype binary in order to avoid missing constraints
>>>>> to other libraries: It still crashes
>>>>> 1st it starts with:
>>>>> ?process 15431: D-Bus library appears to be incorrectly set up; failed
>>>>> to read machine uuid: Failed to open "/var/lib/dbus/machine-id": No such
>>>>> file or directory
>>>>> ?See the manual page for dbus-uuidgen to correct this issue.
>>>>> After calling this client it crashes with:
>>>>> ? /usr/bin/skype: symbol lookup error: /usr/bin/skype: undefined
>>>>> symbol: _ZN10QBoxLayout13addSpacerItemEP11QSpacerItem
>>>>>
>>>>> Any hints, where I may get an older Skype client?
>>>>>
>>>>> Best regards
>>>>> Peter
>>>>>
>>>>> Anthony Minessale schrieb:
>>>>>>
>>>>>> Quick fyi centos 5.4 has some memory bugs in libc don't go beyond 5.3
>>>>>> until its fixed.
>>>>>>
>>>>>>> On Feb 5, 2010 7:42 PM, "Peter P GMX" <Prometheus001 at gmx.net
>>>>>>> <mailto:Prometheus001 at gmx.net>> wrote:
>>>>>>>
>>>>>>> Skype starts, but as soon as it receives a call it crashes with:
>>>>>>>
>>>>>>> /usr/bin/skype: symbol lookup error: /usr/bin/skype: undefined symbol:
>>>>>>>
>>>>>>> _ZN10QBoxLayout13addSpacerItemEP11QSpacerItem
>>>>>>>
>>>>>>> I think the 8.10 version dos not work with8.04.
>>>>>>>
>>>>>>> Any hints, where I may get an older Skype client? I may also try the
>>>>>>> static skype client.
>>>>>>>
>>>>>>>
>>>>>>> Best regards
>>>>>>> Peter
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> .
>>>>>>> Giovanni Maruzzelli schrieb:
>>>>>>> > that's not at all a fatal error.
>>>>>>> > I believe...
>>>>>>>
>>>>>> ------------------------------------------------------------------------
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> Sincerely,
>>>>
>>>> Giovanni Maruzzelli
>>>> Cell : +39-347-2665618
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> Cell : +39-347-2665618
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618



From gmaruzz at celliax.org  Mon Feb  8 08:47:03 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Mon, 8 Feb 2010 17:47:03 +0100
Subject: [Freeswitch-users] CentOS 5.3 (Final) with Skype Static Build
In-Reply-To: <c55d46951002080706t6bb8531i4f12cb35af12e6de@mail.gmail.com>
References: <c55d46951002080706t6bb8531i4f12cb35af12e6de@mail.gmail.com>
Message-ID: <7b197bef1002080847v2524f99dt638c1668ac71425a@mail.gmail.com>

Max,

Just checked, the static build of skype-beta works on latest centos (5.4)

But you can have the same results if just tried :).

ldd tells you which libraries are missing, the you install those libraries.
If you dont know which package a library is into, just do, eg:

yum search libXss

and you'll be told.

========================
So, all in all:

you have to install:

yum install libXv
yum install libXScrnSaver

then use the original snd-dummy supplied with centos (the modified one do not
works at the moment - because the skype-beta does some alsa calls it was
not doing in previous versions, that are not implemented in the
modified alsa driver, I'll fix it soon).

===============================================

-gm



On Mon, Feb 8, 2010 at 4:06 PM, Max Bridgewater
<max.bridgewater at gmail.com> wrote:
> Hi Giovanni,
>
> Let me start a new thread so I don't hijack Peter's one. Here is what
> I get for ldd /usr/bin/skype. What is the deal with these libXv.so.1,
> libXss.so.1?
>
> ? ? ? ?linux-gate.so.1 => ?(0x00697000)
> ? ? ? ?libasound.so.2 => /lib/libasound.so.2 (0x008b7000)
> ? ? ? ?libXv.so.1 => not found
> ? ? ? ?libXss.so.1 => not found
> ? ? ? ?libSM.so.6 => /usr/lib/libSM.so.6 (0x00d02000)
> ? ? ? ?libICE.so.6 => /usr/lib/libICE.so.6 (0x0036b000)
> ? ? ? ?libXi.so.6 => /usr/lib/libXi.so.6 (0x00111000)
> ? ? ? ?libXrender.so.1 => /usr/lib/libXrender.so.1 (0x00119000)
> ? ? ? ?libXrandr.so.2 => /usr/lib/libXrandr.so.2 (0x00b77000)
> ? ? ? ?libfreetype.so.6 => /usr/lib/libfreetype.so.6 (0x00122000)
> ? ? ? ?libfontconfig.so.1 => /usr/lib/libfontconfig.so.1 (0x009fb000)
> ? ? ? ?libXext.so.6 => /usr/lib/libXext.so.6 (0x00eb0000)
> ? ? ? ?libX11.so.6 => /usr/lib/libX11.so.6 (0x001a2000)
> ? ? ? ?libz.so.1 => /usr/lib/libz.so.1 (0x002a5000)
> ? ? ? ?libgthread-2.0.so.0 => /lib/libgthread-2.0.so.0 (0x006bd000)
> ? ? ? ?libglib-2.0.so.0 => /lib/libglib-2.0.so.0 (0x002be000)
> ? ? ? ?librt.so.1 => /lib/librt.so.1 (0x009a1000)
> ? ? ? ?libdl.so.2 => /lib/libdl.so.2 (0x00604000)
> ? ? ? ?libpthread.so.0 => /lib/libpthread.so.0 (0x004e6000)
> ? ? ? ?libstdc++.so.6 => /usr/lib/libstdc++.so.6 (0x004fd000)
> ? ? ? ?libm.so.6 => /lib/libm.so.6 (0x00385000)
> ? ? ? ?libgcc_s.so.1 => /lib/libgcc_s.so.1 (0x00bcc000)
> ? ? ? ?libc.so.6 => /lib/libc.so.6 (0x006c2000)
> ? ? ? ?libexpat.so.0 => /lib/libexpat.so.0 (0x00c2a000)
> ? ? ? ?libXau.so.6 => /usr/lib/libXau.so.6 (0x002ba000)
> ? ? ? ?libXdmcp.so.6 => /usr/lib/libXdmcp.so.6 (0x0087e000)
> ? ? ? ?/lib/ld-linux.so.2 (0x00476000)
>
> Max
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618



From max.bridgewater at gmail.com  Mon Feb  8 09:12:31 2010
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Mon, 8 Feb 2010 12:12:31 -0500
Subject: [Freeswitch-users] CentOS 5.3 (Final) with Skype Static Build
In-Reply-To: <7b197bef1002080847v2524f99dt638c1668ac71425a@mail.gmail.com>
References: <c55d46951002080706t6bb8531i4f12cb35af12e6de@mail.gmail.com>
	<7b197bef1002080847v2524f99dt638c1668ac71425a@mail.gmail.com>
Message-ID: <c55d46951002080912l4123a589qd5546462b7a6b30@mail.gmail.com>

Great thanks, Giovanni. Just completed the installation. No sound test
yet though.

Max.

On Mon, Feb 8, 2010 at 11:47 AM, Giovanni Maruzzelli
<gmaruzz at celliax.org> wrote:
> Max,
>
> Just checked, the static build of skype-beta works on latest centos (5.4)
>
> But you can have the same results if just tried :).
>
> ldd tells you which libraries are missing, the you install those libraries.
> If you dont know which package a library is into, just do, eg:
>
> yum search libXss
>
> and you'll be told.
>
> ========================
> So, all in all:
>
> you have to install:
>
> yum install libXv
> yum install libXScrnSaver
>
> then use the original snd-dummy supplied with centos (the modified one do not
> works at the moment - because the skype-beta does some alsa calls it was
> not doing in previous versions, that are not implemented in the
> modified alsa driver, I'll fix it soon).
>
> ===============================================
>
> -gm
>
>
>
> On Mon, Feb 8, 2010 at 4:06 PM, Max Bridgewater
> <max.bridgewater at gmail.com> wrote:
>> Hi Giovanni,
>>
>> Let me start a new thread so I don't hijack Peter's one. Here is what
>> I get for ldd /usr/bin/skype. What is the deal with these libXv.so.1,
>> libXss.so.1?
>>
>> ? ? ? ?linux-gate.so.1 => ?(0x00697000)
>> ? ? ? ?libasound.so.2 => /lib/libasound.so.2 (0x008b7000)
>> ? ? ? ?libXv.so.1 => not found
>> ? ? ? ?libXss.so.1 => not found
>> ? ? ? ?libSM.so.6 => /usr/lib/libSM.so.6 (0x00d02000)
>> ? ? ? ?libICE.so.6 => /usr/lib/libICE.so.6 (0x0036b000)
>> ? ? ? ?libXi.so.6 => /usr/lib/libXi.so.6 (0x00111000)
>> ? ? ? ?libXrender.so.1 => /usr/lib/libXrender.so.1 (0x00119000)
>> ? ? ? ?libXrandr.so.2 => /usr/lib/libXrandr.so.2 (0x00b77000)
>> ? ? ? ?libfreetype.so.6 => /usr/lib/libfreetype.so.6 (0x00122000)
>> ? ? ? ?libfontconfig.so.1 => /usr/lib/libfontconfig.so.1 (0x009fb000)
>> ? ? ? ?libXext.so.6 => /usr/lib/libXext.so.6 (0x00eb0000)
>> ? ? ? ?libX11.so.6 => /usr/lib/libX11.so.6 (0x001a2000)
>> ? ? ? ?libz.so.1 => /usr/lib/libz.so.1 (0x002a5000)
>> ? ? ? ?libgthread-2.0.so.0 => /lib/libgthread-2.0.so.0 (0x006bd000)
>> ? ? ? ?libglib-2.0.so.0 => /lib/libglib-2.0.so.0 (0x002be000)
>> ? ? ? ?librt.so.1 => /lib/librt.so.1 (0x009a1000)
>> ? ? ? ?libdl.so.2 => /lib/libdl.so.2 (0x00604000)
>> ? ? ? ?libpthread.so.0 => /lib/libpthread.so.0 (0x004e6000)
>> ? ? ? ?libstdc++.so.6 => /usr/lib/libstdc++.so.6 (0x004fd000)
>> ? ? ? ?libm.so.6 => /lib/libm.so.6 (0x00385000)
>> ? ? ? ?libgcc_s.so.1 => /lib/libgcc_s.so.1 (0x00bcc000)
>> ? ? ? ?libc.so.6 => /lib/libc.so.6 (0x006c2000)
>> ? ? ? ?libexpat.so.0 => /lib/libexpat.so.0 (0x00c2a000)
>> ? ? ? ?libXau.so.6 => /usr/lib/libXau.so.6 (0x002ba000)
>> ? ? ? ?libXdmcp.so.6 => /usr/lib/libXdmcp.so.6 (0x0087e000)
>> ? ? ? ?/lib/ld-linux.so.2 (0x00476000)
>>
>> Max
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From christian at officepools.com  Mon Feb  8 10:39:46 2010
From: christian at officepools.com (Christian Jensen)
Date: Mon, 8 Feb 2010 10:39:46 -0800
Subject: [Freeswitch-users] FS based Softphone?
Message-ID: <e90351e71002081039y7c7723a6x7181d9fdd208bdde@mail.gmail.com>

Hi,

I am setting up our office with phones and softphones. I heard that there is
a softphone that was based on FS - is this ready for primetime? If not, what
is the best softphone for use with a mostly windows but some ubuntu and mac
environment?

Thanks!
Christian
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100208/c938948c/attachment-0002.html 

From jmesquita at freeswitch.org  Mon Feb  8 11:09:13 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Mon, 8 Feb 2010 17:09:13 -0200
Subject: [Freeswitch-users] FS based Softphone?
In-Reply-To: <e90351e71002081039y7c7723a6x7181d9fdd208bdde@mail.gmail.com>
References: <e90351e71002081039y7c7723a6x7181d9fdd208bdde@mail.gmail.com>
Message-ID: <ebbc2d551002081109p2e451894y45aa9bfae3c548bb@mail.gmail.com>

Christian,

The project you are mentioning is called FSComm and can be found on the SVN
source. The project is still under intense development with very limited
resources and time. We are constantly looking for sponsors to make this
dream possible, if you are a potential one, please send me and email
offlist.

As for the project maturity, it is up to you to decide since only you know
the features needed. We have users that use it on a daily basis and we have
a couple of bugs filed that I still haven't had time to fix.

We don't have a mac installer but the wiki makes it pretty clear on how to
compile it: http://wiki.freeswitch.org/wiki/FSComm

We welcome any type of criticism or request.

Regards,

Jo?o Mesquita
FSComm Developer


On Mon, Feb 8, 2010 at 4:39 PM, Christian Jensen
<christian at officepools.com>wrote:

> Hi,
>
> I am setting up our office with phones and softphones. I heard that there
> is a softphone that was based on FS - is this ready for primetime? If not,
> what is the best softphone for use with a mostly windows but some ubuntu and
> mac environment?
>
> Thanks!
> Christian
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100208/edb3d29d/attachment-0002.html 

From mrene_lists at avgs.ca  Mon Feb  8 11:32:25 2010
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Mon, 8 Feb 2010 14:32:25 -0500
Subject: [Freeswitch-users] mod_limit requires media?
In-Reply-To: <2d9149cd1002072106n7272cb80se3329700f0074d52@mail.gmail.com>
References: <2d9149cd1002072106n7272cb80se3329700f0074d52@mail.gmail.com>
Message-ID: <AE10DF9B-B6AC-41D1-AC28-EFEF5946C44D@avgs.ca>

Hi,

mod_limit doesn't require any media. ca you post some logs of your  
problem?

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 8-Feb-10, at 12:06 AM, Kristian Kielhofner wrote:

> Hello everyone,
>
>  I was playing with mod_limit earlier tonight and I noticed that it
> essentially stopped hashing/tracking calls once bypass media was set.
> Is this by design?  Is there some reason mod_limit requires media?
> Other than that mod_limit looks to be very well implemented (no
> surprise there) and I'm excited to put it to more use.
>
> Thanks!
>
> -- 
> Kristian Kielhofner
> http://www.astlinux.org
> http://blog.krisk.org
> http://www.star2star.com
> http://www.submityoursip.com
> http://www.voalte.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From kristian.kielhofner at gmail.com  Mon Feb  8 12:10:56 2010
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Mon, 8 Feb 2010 15:10:56 -0500
Subject: [Freeswitch-users] mod_limit requires media?
In-Reply-To: <AE10DF9B-B6AC-41D1-AC28-EFEF5946C44D@avgs.ca>
References: <2d9149cd1002072106n7272cb80se3329700f0074d52@mail.gmail.com>
	<AE10DF9B-B6AC-41D1-AC28-EFEF5946C44D@avgs.ca>
Message-ID: <2d9149cd1002081210u3662ffa6h3eb3d1c479512e80@mail.gmail.com>

Hi,

  I just sent this to Tony off-list:

It doesn't seem like it should be but testing it shows there is some issue...

no bypass_media (working):
http://pastebin.freeswitch.org/12085

with bypass_media (not working):
http://pastebin.freeswitch.org/12086

This is SVN rev 16584 on my Mac running Snow Leopard.  I'm trying to
call my AstLinux conf from a local softphone, bridging through FS.
Completely default configs except I added the limit config shown in
the PBs in conf/dialplan/default/00_astlinux-conf.xml

You can see that with bypass media set when the call is up
limit_hash_usage outbound carrier1 shows 0.  Without bypass media set
it shows 1.  However in both cases the FS console reports that usage
is 1/1 at some point...

On Mon, Feb 8, 2010 at 2:32 PM, Mathieu Rene <mrene_lists at avgs.ca> wrote:
> Hi,
>
> mod_limit doesn't require any media. ca you post some logs of your
> problem?
>
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>
>
>
>
> On 8-Feb-10, at 12:06 AM, Kristian Kielhofner wrote:
>
>> Hello everyone,
>>
>> ?I was playing with mod_limit earlier tonight and I noticed that it
>> essentially stopped hashing/tracking calls once bypass media was set.
>> Is this by design? ?Is there some reason mod_limit requires media?
>> Other than that mod_limit looks to be very well implemented (no
>> surprise there) and I'm excited to put it to more use.
>>
>> Thanks!
>>
>> --
>> Kristian Kielhofner
>> http://www.astlinux.org
>> http://blog.krisk.org
>> http://www.star2star.com
>> http://www.submityoursip.com
>> http://www.voalte.com
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com



From gavin.henry at gmail.com  Mon Feb  8 12:40:57 2010
From: gavin.henry at gmail.com (Gavin Henry)
Date: Mon, 8 Feb 2010 20:40:57 +0000
Subject: [Freeswitch-users] mod_nibblebill, loading A-Z rates and CDRs
Message-ID: <13ca621c1002081240h223942e7vb4f9e1034822ee28@mail.gmail.com>

Hi all,

We're testing FS and mod_nibblebill for a wholesale platform. So far
it's working ok via ODBC. I'd like to know how you'd recommend loading
A-Z rates that have peak, offpeak and weekend rates.

I was thinking about using the conditions for destination to match the
time of day:

http://wiki.freeswitch.org/wiki/Dialplan_XML#Condition

so each rate would be listed a few times in order to match the time of day.

For the destinations I was going to do
(http://wiki.freeswitch.org/wiki/Mod_nibblebill#Different_Rates_per_Area_Code):

 <extension name="uk_landline">
   <condition field="destination_number" expression="^0([1-3]\d{9})$">
    <action application="set" data="nibble_account=${accountcode}"/>
    <action application="set" data="nibble_rate=rate_tbc"/>
    <action application="bridge"
data="sofia/gateway/sip.myprovider.co.uk/44$1"/>
   </condition>
 </extension>


I presume I can have (newbie here) multiple conditions for the time of day?

My last question is if there is a better way to load the A-Z rates
rather than a massive XML file, like from a DB. I'm still new to FS so
not sure how to pull in this data. I presume I could just pull it in
via mod_xml_curl and do the same with saving the CDRs to a RDBMS using
mod_xml_cdr?

Is this a best practice going to the RDBMS (I'm part of the OpenLDAP
project so always call it RDBMS so as not confuse a user thinking
about the OpenLDAP database backend) via mod_xml_curl? I would thought
going directly to the RDBMS is better? I suppose it depends on the
system architecture.

Thanks,

Gavin.


-- 
http://www.suretecsystems.com/services/openldap/
http://www.suretectelecom.com



From rupa at rupa.com  Mon Feb  8 13:10:49 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Mon, 8 Feb 2010 15:10:49 -0600
Subject: [Freeswitch-users] mod_nibblebill, loading A-Z rates and CDRs
In-Reply-To: <13ca621c1002081240h223942e7vb4f9e1034822ee28@mail.gmail.com>
References: <13ca621c1002081240h223942e7vb4f9e1034822ee28@mail.gmail.com>
Message-ID: <b39273531002081310m7d37d9f0gf700b1ecd9ee56dd@mail.gmail.com>

Look at using mod_lcr for doing your A-Z rates.  You can categorize your
peak/offpeak/wkend rates as different profiles and then lookup based on
those profiles. mod_lcr is very flexible in that you can also specify your
own custom sql so if you are familiar with sql you could even put your
peak/offpeak/wkend etc decisions in the SQL or in a stored procedure in the
db.

If you are going down the path of custom_sql, then use the patch in jira
since custom_sql behavior will change post 1.0.5 and the new behavior is
much more flexible.

On Mon, Feb 8, 2010 at 2:40 PM, Gavin Henry <gavin.henry at gmail.com> wrote:

> Hi all,
>
> We're testing FS and mod_nibblebill for a wholesale platform. So far
> it's working ok via ODBC. I'd like to know how you'd recommend loading
> A-Z rates that have peak, offpeak and weekend rates.
>
> I was thinking about using the conditions for destination to match the
> time of day:
>
> http://wiki.freeswitch.org/wiki/Dialplan_XML#Condition
>
> so each rate would be listed a few times in order to match the time of day.
>
> For the destinations I was going to do
> (
> http://wiki.freeswitch.org/wiki/Mod_nibblebill#Different_Rates_per_Area_Code
> ):
>
>  <extension name="uk_landline">
>   <condition field="destination_number" expression="^0([1-3]\d{9})$">
>    <action application="set" data="nibble_account=${accountcode}"/>
>    <action application="set" data="nibble_rate=rate_tbc"/>
>    <action application="bridge"
> data="sofia/gateway/sip.myprovider.co.uk/44$1"/>
>   </condition>
>  </extension>
>
>
> I presume I can have (newbie here) multiple conditions for the time of day?
>
> My last question is if there is a better way to load the A-Z rates
> rather than a massive XML file, like from a DB. I'm still new to FS so
> not sure how to pull in this data. I presume I could just pull it in
> via mod_xml_curl and do the same with saving the CDRs to a RDBMS using
> mod_xml_cdr?
>
> Is this a best practice going to the RDBMS (I'm part of the OpenLDAP
> project so always call it RDBMS so as not confuse a user thinking
> about the OpenLDAP database backend) via mod_xml_curl? I would thought
> going directly to the RDBMS is better? I suppose it depends on the
> system architecture.
>
> Thanks,
>
> Gavin.
>
>
> --
> http://www.suretecsystems.com/services/openldap/
> http://www.suretectelecom.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
-Rupa
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100208/3d2778ec/attachment-0002.html 

From gavin.henry at gmail.com  Mon Feb  8 13:20:44 2010
From: gavin.henry at gmail.com (Gavin Henry)
Date: Mon, 8 Feb 2010 21:20:44 +0000
Subject: [Freeswitch-users] mod_nibblebill, loading A-Z rates and CDRs
In-Reply-To: <b39273531002081310m7d37d9f0gf700b1ecd9ee56dd@mail.gmail.com>
References: <13ca621c1002081240h223942e7vb4f9e1034822ee28@mail.gmail.com>
	<b39273531002081310m7d37d9f0gf700b1ecd9ee56dd@mail.gmail.com>
Message-ID: <13ca621c1002081320m3b9a7779l6ee6daa6382ffe56@mail.gmail.com>

On 8 February 2010 21:10, Rupa Schomaker <rupa at rupa.com> wrote:
> Look at using mod_lcr for doing your A-Z rates. ?You can categorize your
> peak/offpeak/wkend rates as different profiles and then lookup based on
> those profiles. mod_lcr is very flexible in that you can also specify your
> own custom sql so if you are familiar with sql you could even put your
> peak/offpeak/wkend etc decisions in the SQL or in a stored procedure in the
> db.
> If you are going down the path of custom_sql, then use the patch in jira
> since custom_sql behavior will change post 1.0.5 and the new behavior is
> much more flexible.

OK, thanks I will do. What about the real time billing features of
nibblebill? Can you use them together?

Will do read the docs now for mod_lcr.

> On Mon, Feb 8, 2010 at 2:40 PM, Gavin Henry <gavin.henry at gmail.com> wrote:
>>
>> Hi all,
>>
>> We're testing FS and mod_nibblebill for a wholesale platform. So far
>> it's working ok via ODBC. I'd like to know how you'd recommend loading
>> A-Z rates that have peak, offpeak and weekend rates.
>>
>> I was thinking about using the conditions for destination to match the
>> time of day:
>>
>> http://wiki.freeswitch.org/wiki/Dialplan_XML#Condition
>>
>> so each rate would be listed a few times in order to match the time of
>> day.
>>
>> For the destinations I was going to do
>>
>> (http://wiki.freeswitch.org/wiki/Mod_nibblebill#Different_Rates_per_Area_Code):
>>
>> ?<extension name="uk_landline">
>> ? <condition field="destination_number" expression="^0([1-3]\d{9})$">
>> ? ?<action application="set" data="nibble_account=${accountcode}"/>
>> ? ?<action application="set" data="nibble_rate=rate_tbc"/>
>> ? ?<action application="bridge"
>> data="sofia/gateway/sip.myprovider.co.uk/44$1"/>
>> ? </condition>
>> ?</extension>
>>
>>
>> I presume I can have (newbie here) multiple conditions for the time of
>> day?
>>
>> My last question is if there is a better way to load the A-Z rates
>> rather than a massive XML file, like from a DB. I'm still new to FS so
>> not sure how to pull in this data. I presume I could just pull it in
>> via mod_xml_curl and do the same with saving the CDRs to a RDBMS using
>> mod_xml_cdr?
>>
>> Is this a best practice going to the RDBMS (I'm part of the OpenLDAP
>> project so always call it RDBMS so as not confuse a user thinking
>> about the OpenLDAP database backend) via mod_xml_curl? I would thought
>> going directly to the RDBMS is better? I suppose it depends on the
>> system architecture.
>>
>> Thanks,
>>
>> Gavin.
>>
>>
>> --
>> http://www.suretecsystems.com/services/openldap/
>> http://www.suretectelecom.com
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
http://www.suretecsystems.com/services/openldap/
http://www.suretectelecom.com



From gavin.henry at gmail.com  Mon Feb  8 13:39:38 2010
From: gavin.henry at gmail.com (Gavin Henry)
Date: Mon, 8 Feb 2010 21:39:38 +0000
Subject: [Freeswitch-users] mod_nibblebill, loading A-Z rates and CDRs
In-Reply-To: <13ca621c1002081320m3b9a7779l6ee6daa6382ffe56@mail.gmail.com>
References: <13ca621c1002081240h223942e7vb4f9e1034822ee28@mail.gmail.com>
	<b39273531002081310m7d37d9f0gf700b1ecd9ee56dd@mail.gmail.com>
	<13ca621c1002081320m3b9a7779l6ee6daa6382ffe56@mail.gmail.com>
Message-ID: <13ca621c1002081339y574aabdg1408052b2bb28bf@mail.gmail.com>

On 8 February 2010 21:20, Gavin Henry <gavin.henry at gmail.com> wrote:
> On 8 February 2010 21:10, Rupa Schomaker <rupa at rupa.com> wrote:
>> Look at using mod_lcr for doing your A-Z rates. ?You can categorize your
>> peak/offpeak/wkend rates as different profiles and then lookup based on
>> those profiles. mod_lcr is very flexible in that you can also specify your
>> own custom sql so if you are familiar with sql you could even put your
>> peak/offpeak/wkend etc decisions in the SQL or in a stored procedure in the
>> db.
>> If you are going down the path of custom_sql, then use the patch in jira
>> since custom_sql behavior will change post 1.0.5 and the new behavior is
>> much more flexible.
>
> OK, thanks I will do. What about the real time billing features of
> nibblebill? Can you use them together?

For others, there is some info at
http://wiki.freeswitch.org/wiki/Mod_lcr#User_Rates that mentions
nibblebill.

Thanks.

-- 
http://www.suretecsystems.com/services/openldap/
http://www.suretectelecom.com



From mrene_lists at avgs.ca  Mon Feb  8 14:27:08 2010
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Mon, 8 Feb 2010 17:27:08 -0500
Subject: [Freeswitch-users] mod_limit requires media?
In-Reply-To: <2d9149cd1002081210u3662ffa6h3eb3d1c479512e80@mail.gmail.com>
References: <2d9149cd1002072106n7272cb80se3329700f0074d52@mail.gmail.com>
	<AE10DF9B-B6AC-41D1-AC28-EFEF5946C44D@avgs.ca>
	<2d9149cd1002081210u3662ffa6h3eb3d1c479512e80@mail.gmail.com>
Message-ID: <A2EB2618-601A-487C-A754-7B258250D6FC@avgs.ca>

Try r16586

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 8-Feb-10, at 3:10 PM, Kristian Kielhofner wrote:

> Hi,
>
>  I just sent this to Tony off-list:
>
> It doesn't seem like it should be but testing it shows there is some  
> issue...
>
> no bypass_media (working):
> http://pastebin.freeswitch.org/12085
>
> with bypass_media (not working):
> http://pastebin.freeswitch.org/12086
>
> This is SVN rev 16584 on my Mac running Snow Leopard.  I'm trying to
> call my AstLinux conf from a local softphone, bridging through FS.
> Completely default configs except I added the limit config shown in
> the PBs in conf/dialplan/default/00_astlinux-conf.xml
>
> You can see that with bypass media set when the call is up
> limit_hash_usage outbound carrier1 shows 0.  Without bypass media set
> it shows 1.  However in both cases the FS console reports that usage
> is 1/1 at some point...
>
> On Mon, Feb 8, 2010 at 2:32 PM, Mathieu Rene <mrene_lists at avgs.ca>  
> wrote:
>> Hi,
>>
>> mod_limit doesn't require any media. ca you post some logs of your
>> problem?
>>
>> Mathieu Rene
>> Avant-Garde Solutions Inc
>> Office: + 1 (514) 664-1044 x100
>> Cell: +1 (514) 664-1044 x200
>> mrene at avgs.ca
>>
>>
>>
>>
>> On 8-Feb-10, at 12:06 AM, Kristian Kielhofner wrote:
>>
>>> Hello everyone,
>>>
>>>  I was playing with mod_limit earlier tonight and I noticed that it
>>> essentially stopped hashing/tracking calls once bypass media was  
>>> set.
>>> Is this by design?  Is there some reason mod_limit requires media?
>>> Other than that mod_limit looks to be very well implemented (no
>>> surprise there) and I'm excited to put it to more use.
>>>
>>> Thanks!
>>>
>>> --
>>> Kristian Kielhofner
>>> http://www.astlinux.org
>>> http://blog.krisk.org
>>> http://www.star2star.com
>>> http://www.submityoursip.com
>>> http://www.voalte.com
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> -- 
> Kristian Kielhofner
> http://www.astlinux.org
> http://blog.krisk.org
> http://www.star2star.com
> http://www.submityoursip.com
> http://www.voalte.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From kristian.kielhofner at gmail.com  Mon Feb  8 14:55:56 2010
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Mon, 8 Feb 2010 17:55:56 -0500
Subject: [Freeswitch-users] mod_limit requires media?
In-Reply-To: <A2EB2618-601A-487C-A754-7B258250D6FC@avgs.ca>
References: <2d9149cd1002072106n7272cb80se3329700f0074d52@mail.gmail.com>
	<AE10DF9B-B6AC-41D1-AC28-EFEF5946C44D@avgs.ca>
	<2d9149cd1002081210u3662ffa6h3eb3d1c479512e80@mail.gmail.com>
	<A2EB2618-601A-487C-A754-7B258250D6FC@avgs.ca>
Message-ID: <2d9149cd1002081455q6a84e61fpac46be8082efc3d3@mail.gmail.com>

I'm going to perform further testing but it looks good so far.

Thanks a lot!

On Mon, Feb 8, 2010 at 5:27 PM, Mathieu Rene <mrene_lists at avgs.ca> wrote:
> Try r16586
>
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>

-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com



From lon at kickasspixels.com  Mon Feb  8 17:05:04 2010
From: lon at kickasspixels.com (Lon Baker)
Date: Mon, 8 Feb 2010 17:05:04 -0800
Subject: [Freeswitch-users] Redirect during Bridge
Message-ID: <5d3e0dc61002081705n24992492yd28194b9a5791bac@mail.gmail.com>

Hi,

I need to bridge calls to a proxy that issues a redirect command. Its
an internal proxy that translates internal SIP URIs into outbound
URIs.

I have one interface on a private network where the proxy is running
and another interface on my public network.

If I am reading the debug information correctly, the bridge to the
proxy is getting the redirect correctly, but since it is on the
internal network (profile), the call never transitions to the public
network to complete the call.

Is it possible to do this?

Lon



From r.wilczynski at gmail.com  Mon Feb  8 11:37:40 2010
From: r.wilczynski at gmail.com (=?ISO-8859-2?Q?Robert_Wilczy=F1ski?=)
Date: Mon, 8 Feb 2010 20:37:40 +0100
Subject: [Freeswitch-users] Trunk Version Number
In-Reply-To: <8976E2C4-CF8C-433A-AC79-B58F65105D08@jerris.com>
References: <2360B060EDF44D368707DD12064BD416@bp1.ad.bp.com>
	<8976E2C4-CF8C-433A-AC79-B58F65105D08@jerris.com>
Message-ID: <56bfa1be1002081137q3f2aad44wc2742aa3505bf961@mail.gmail.com>

Hi Michael,

You mean a set of windows binaries matching those in w32\Library (this
seems to work for me)? Where should I email it?

Robert.

On Mon, Feb 1, 2010 at 7:11 AM, Michael Jerris <mike at jerris.com> wrote:
> it should. ?This can happen if you build from an svn checkout and the svn
> client your using is newer than our static linked svnversion.exe. ?If anyone
> can make me a newer stripped down version like that I would appreciate it I
> have not had the time.
> On Jan 31, 2010, at 9:30 AM, Dave Stevenson wrote:
>
> Hi,
>
> Running the latest SVN (16453)?under Windows, the console "Version" command
> displays :-
>
> "FreeSWITCH Version 1.0.trunk (UNKNOWN)"
>
> Should the version number not include a meaningful build version?in the
> brackets ?
>
> regards
> Dave
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From troy at tlainvestments.com  Mon Feb  8 20:21:01 2010
From: troy at tlainvestments.com (Troy Anderson)
Date: Mon, 8 Feb 2010 21:21:01 -0700
Subject: [Freeswitch-users] UPnP Timeout
Message-ID: <9706F7A8-7644-4602-B1DE-289A35F788AA@tlainvestments.com>

I have been using pfSense (1.2.3) and FS.  FS nicely uses UPnP to poke holes in the firewall, but it seems that the holes close after a while.  I cannot find any documentation in FS nor in pfSense as to what the timeout is.  Is there a setting in FS to do some kind of keep-alive thing with UPnP to keep, e.g. 5060, open?  Or is it already doing that and pfSense is the issue?

Thanks!


From yehavi.bourvine at gmail.com  Mon Feb  8 21:10:04 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Tue, 9 Feb 2010 07:10:04 +0200
Subject: [Freeswitch-users] Announcement: FreeSWITCH Add Broadsoft SCA
	Support
In-Reply-To: <87f2f3b91001111742k691966cfs82515db77c13d4d4@mail.gmail.com>
References: <87f2f3b91001111742k691966cfs82515db77c13d4d4@mail.gmail.com>
Message-ID: <d361ca7b1002082110h4e3c60d0p88412f753444ec3d@mail.gmail.com>

Hello,

  We currently use the "old" type of presence which is activated by
"manage-presence" coupled with "dbname" and "presence-hosts".
With the new method, does "manage-shared-presence" replace all of the above
or comes in addition?

                       Thanks! __Yehavi:

 2010/1/12 Michael Collins <msc at freeswitch.org>

> We want to let everyone know that FreeSWITCH now supports the Broadsoft SCA
> method of doing shared lines. The story is here:
>
> http://www.freeswitch.org/node/227
>
> Tony and Brian spent many hours laboring over this, so please be sure to
> show your appreciation to them for this new feature and all of the great
> things they do for the FreeSWITCH community and VoIP in general!
>
> -Michael
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100209/fc020b4c/attachment-0002.html 

From mrene_lists at avgs.ca  Mon Feb  8 21:36:55 2010
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Tue, 9 Feb 2010 00:36:55 -0500
Subject: [Freeswitch-users] Redirect during Bridge
In-Reply-To: <5d3e0dc61002081705n24992492yd28194b9a5791bac@mail.gmail.com>
References: <5d3e0dc61002081705n24992492yd28194b9a5791bac@mail.gmail.com>
Message-ID: <ABEBD044-4E95-4916-A125-2556F2E47ED6@avgs.ca>

Hi,

mod_sofia's default behavior is to process 302s within the sip stack,  
automatically, in some cases, like yours, this behavior isnt desired.
You can enable manual redirects which will make the call go back in  
the dialplan whenever an invite hits a 302.
Add the following to your sip profile:
<param name="manual-redirect" value="true" />

You can then, in your dialplan, set the sip_redirect_profile variable  
to your internal network's sip profile. When a call is redirected,   
the sip_redirect_dialstring variable will contain the dialstring you  
need to pass to bridge.

your outbound dialplan should look like this
<extension name="something">
  <condition field="destination_number" expression="....">
    <action application="set" data="sip_redirect_profile=internal" />
  </condition>
</extension>

Once a 302 is received, your call with redirect to the "redirected"  
dialplan context (or to the one specified in sip_redirect_context and  
sip_redirect_dialplan). It should like the following:

<context name="redirected">
  <extension name="redirected">
   <condition>
     <action application="bridge" data="${sip_redirect_dialstring}" />
   </condition>
  </extension>
</context>

Note: you can also check if the user is allowed to do redirects while  
using this method, so, lets say, one of your carrier can't 302 you to  
another of your carriers.


Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 8-Feb-10, at 8:05 PM, Lon Baker wrote:

> Hi,
>
> I need to bridge calls to a proxy that issues a redirect command. Its
> an internal proxy that translates internal SIP URIs into outbound
> URIs.
>
> I have one interface on a private network where the proxy is running
> and another interface on my public network.
>
> If I am reading the debug information correctly, the bridge to the
> proxy is getting the redirect correctly, but since it is on the
> internal network (profile), the call never transitions to the public
> network to complete the call.
>
> Is it possible to do this?
>
> Lon
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From dujinfang at gmail.com  Mon Feb  8 22:20:05 2010
From: dujinfang at gmail.com (Seven Du)
Date: Tue, 9 Feb 2010 14:20:05 +0800
Subject: [Freeswitch-users] CentOS 5.3 (Final) with Skype Static Build
In-Reply-To: <7b197bef1002080847v2524f99dt638c1668ac71425a@mail.gmail.com>
References: <c55d46951002080706t6bb8531i4f12cb35af12e6de@mail.gmail.com>
	<7b197bef1002080847v2524f99dt638c1668ac71425a@mail.gmail.com>
Message-ID: <23f91031002082220p770fcfdep9a3ae60296a44654@mail.gmail.com>

Ciao Giovanni,

I'm about to install a new server and will try to migrate to CentOS
from Ubuntu, and based on this list Anthony mentioned CentOS5.4 has
some problems on some tool chain, but I noticed that you said you are
on the latest 5.4 with skypiax, do you have any problems?

The new server will mainly be used to do SIP and skype gateway. And I
have no experience on CentOS 5.

Thanks.

2010/2/9 Giovanni Maruzzelli <gmaruzz at celliax.org>:
> Max,
>
> Just checked, the static build of skype-beta works on latest centos (5.4)
>
> But you can have the same results if just tried :).
>
> ldd tells you which libraries are missing, the you install those libraries.
> If you dont know which package a library is into, just do, eg:
>
> yum search libXss
>
> and you'll be told.
>
> ========================
> So, all in all:
>
> you have to install:
>
> yum install libXv
> yum install libXScrnSaver
>
> then use the original snd-dummy supplied with centos (the modified one do not
> works at the moment - because the skype-beta does some alsa calls it was
> not doing in previous versions, that are not implemented in the
> modified alsa driver, I'll fix it soon).
>
> ===============================================
>
> -gm
>
>
>
> On Mon, Feb 8, 2010 at 4:06 PM, Max Bridgewater
> <max.bridgewater at gmail.com> wrote:
>> Hi Giovanni,
>>
>> Let me start a new thread so I don't hijack Peter's one. Here is what
>> I get for ldd /usr/bin/skype. What is the deal with these libXv.so.1,
>> libXss.so.1?
>>
>> ? ? ? ?linux-gate.so.1 => ?(0x00697000)
>> ? ? ? ?libasound.so.2 => /lib/libasound.so.2 (0x008b7000)
>> ? ? ? ?libXv.so.1 => not found
>> ? ? ? ?libXss.so.1 => not found
>> ? ? ? ?libSM.so.6 => /usr/lib/libSM.so.6 (0x00d02000)
>> ? ? ? ?libICE.so.6 => /usr/lib/libICE.so.6 (0x0036b000)
>> ? ? ? ?libXi.so.6 => /usr/lib/libXi.so.6 (0x00111000)
>> ? ? ? ?libXrender.so.1 => /usr/lib/libXrender.so.1 (0x00119000)
>> ? ? ? ?libXrandr.so.2 => /usr/lib/libXrandr.so.2 (0x00b77000)
>> ? ? ? ?libfreetype.so.6 => /usr/lib/libfreetype.so.6 (0x00122000)
>> ? ? ? ?libfontconfig.so.1 => /usr/lib/libfontconfig.so.1 (0x009fb000)
>> ? ? ? ?libXext.so.6 => /usr/lib/libXext.so.6 (0x00eb0000)
>> ? ? ? ?libX11.so.6 => /usr/lib/libX11.so.6 (0x001a2000)
>> ? ? ? ?libz.so.1 => /usr/lib/libz.so.1 (0x002a5000)
>> ? ? ? ?libgthread-2.0.so.0 => /lib/libgthread-2.0.so.0 (0x006bd000)
>> ? ? ? ?libglib-2.0.so.0 => /lib/libglib-2.0.so.0 (0x002be000)
>> ? ? ? ?librt.so.1 => /lib/librt.so.1 (0x009a1000)
>> ? ? ? ?libdl.so.2 => /lib/libdl.so.2 (0x00604000)
>> ? ? ? ?libpthread.so.0 => /lib/libpthread.so.0 (0x004e6000)
>> ? ? ? ?libstdc++.so.6 => /usr/lib/libstdc++.so.6 (0x004fd000)
>> ? ? ? ?libm.so.6 => /lib/libm.so.6 (0x00385000)
>> ? ? ? ?libgcc_s.so.1 => /lib/libgcc_s.so.1 (0x00bcc000)
>> ? ? ? ?libc.so.6 => /lib/libc.so.6 (0x006c2000)
>> ? ? ? ?libexpat.so.0 => /lib/libexpat.so.0 (0x00c2a000)
>> ? ? ? ?libXau.so.6 => /usr/lib/libXau.so.6 (0x002ba000)
>> ? ? ? ?libXdmcp.so.6 => /usr/lib/libXdmcp.so.6 (0x0087e000)
>> ? ? ? ?/lib/ld-linux.so.2 (0x00476000)
>>
>> Max
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From dujinfang at gmail.com  Mon Feb  8 22:22:27 2010
From: dujinfang at gmail.com (Seven Du)
Date: Tue, 9 Feb 2010 14:22:27 +0800
Subject: [Freeswitch-users] CentOS 5.3 (Final) with Skype Static Build
In-Reply-To: <23f91031002082220p770fcfdep9a3ae60296a44654@mail.gmail.com>
References: <c55d46951002080706t6bb8531i4f12cb35af12e6de@mail.gmail.com>
	<7b197bef1002080847v2524f99dt638c1668ac71425a@mail.gmail.com>
	<23f91031002082220p770fcfdep9a3ae60296a44654@mail.gmail.com>
Message-ID: <23f91031002082222q75228efai24d0d89933713cdc@mail.gmail.com>

or I will try CentOS 5.3 and Skype2.1 beta.   But it sames skype has a
64bit native on Ubuntu 8.10 which not available on CentOS.

2010/2/9 Seven Du <dujinfang at gmail.com>:
> Ciao Giovanni,
>
> I'm about to install a new server and will try to migrate to CentOS
> from Ubuntu, and based on this list Anthony mentioned CentOS5.4 has
> some problems on some tool chain, but I noticed that you said you are
> on the latest 5.4 with skypiax, do you have any problems?
>
> The new server will mainly be used to do SIP and skype gateway. And I
> have no experience on CentOS 5.
>
> Thanks.
>
> 2010/2/9 Giovanni Maruzzelli <gmaruzz at celliax.org>:
>> Max,
>>
>> Just checked, the static build of skype-beta works on latest centos (5.4)
>>
>> But you can have the same results if just tried :).
>>
>> ldd tells you which libraries are missing, the you install those libraries.
>> If you dont know which package a library is into, just do, eg:
>>
>> yum search libXss
>>
>> and you'll be told.
>>
>> ========================
>> So, all in all:
>>
>> you have to install:
>>
>> yum install libXv
>> yum install libXScrnSaver
>>
>> then use the original snd-dummy supplied with centos (the modified one do not
>> works at the moment - because the skype-beta does some alsa calls it was
>> not doing in previous versions, that are not implemented in the
>> modified alsa driver, I'll fix it soon).
>>
>> ===============================================
>>
>> -gm
>>
>>
>>
>> On Mon, Feb 8, 2010 at 4:06 PM, Max Bridgewater
>> <max.bridgewater at gmail.com> wrote:
>>> Hi Giovanni,
>>>
>>> Let me start a new thread so I don't hijack Peter's one. Here is what
>>> I get for ldd /usr/bin/skype. What is the deal with these libXv.so.1,
>>> libXss.so.1?
>>>
>>> ? ? ? ?linux-gate.so.1 => ?(0x00697000)
>>> ? ? ? ?libasound.so.2 => /lib/libasound.so.2 (0x008b7000)
>>> ? ? ? ?libXv.so.1 => not found
>>> ? ? ? ?libXss.so.1 => not found
>>> ? ? ? ?libSM.so.6 => /usr/lib/libSM.so.6 (0x00d02000)
>>> ? ? ? ?libICE.so.6 => /usr/lib/libICE.so.6 (0x0036b000)
>>> ? ? ? ?libXi.so.6 => /usr/lib/libXi.so.6 (0x00111000)
>>> ? ? ? ?libXrender.so.1 => /usr/lib/libXrender.so.1 (0x00119000)
>>> ? ? ? ?libXrandr.so.2 => /usr/lib/libXrandr.so.2 (0x00b77000)
>>> ? ? ? ?libfreetype.so.6 => /usr/lib/libfreetype.so.6 (0x00122000)
>>> ? ? ? ?libfontconfig.so.1 => /usr/lib/libfontconfig.so.1 (0x009fb000)
>>> ? ? ? ?libXext.so.6 => /usr/lib/libXext.so.6 (0x00eb0000)
>>> ? ? ? ?libX11.so.6 => /usr/lib/libX11.so.6 (0x001a2000)
>>> ? ? ? ?libz.so.1 => /usr/lib/libz.so.1 (0x002a5000)
>>> ? ? ? ?libgthread-2.0.so.0 => /lib/libgthread-2.0.so.0 (0x006bd000)
>>> ? ? ? ?libglib-2.0.so.0 => /lib/libglib-2.0.so.0 (0x002be000)
>>> ? ? ? ?librt.so.1 => /lib/librt.so.1 (0x009a1000)
>>> ? ? ? ?libdl.so.2 => /lib/libdl.so.2 (0x00604000)
>>> ? ? ? ?libpthread.so.0 => /lib/libpthread.so.0 (0x004e6000)
>>> ? ? ? ?libstdc++.so.6 => /usr/lib/libstdc++.so.6 (0x004fd000)
>>> ? ? ? ?libm.so.6 => /lib/libm.so.6 (0x00385000)
>>> ? ? ? ?libgcc_s.so.1 => /lib/libgcc_s.so.1 (0x00bcc000)
>>> ? ? ? ?libc.so.6 => /lib/libc.so.6 (0x006c2000)
>>> ? ? ? ?libexpat.so.0 => /lib/libexpat.so.0 (0x00c2a000)
>>> ? ? ? ?libXau.so.6 => /usr/lib/libXau.so.6 (0x002ba000)
>>> ? ? ? ?libXdmcp.so.6 => /usr/lib/libXdmcp.so.6 (0x0087e000)
>>> ? ? ? ?/lib/ld-linux.so.2 (0x00476000)
>>>
>>> Max
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> Cell : +39-347-2665618
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>



From tayeb.meftah at gmail.com  Mon Feb  8 23:03:06 2010
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Tue, 09 Feb 2010 08:03:06 +0100
Subject: [Freeswitch-users] UPnP Timeout
In-Reply-To: <9706F7A8-7644-4602-B1DE-289A35F788AA@tlainvestments.com>
References: <9706F7A8-7644-4602-B1DE-289A35F788AA@tlainvestments.com>
Message-ID: <4B7108AA.9000101@gmail.com>

hi,
no problem in windows or debian GNU/Linux
mayb is a pfsense problem
if the problem percist start fs with: ./freeswitch -nonat and redirect 
your required ports;)
thanks
Le 09/02/2010 05:21, Troy Anderson a ?crit :
> I have been using pfSense (1.2.3) and FS.  FS nicely uses UPnP to poke holes in the firewall, but it seems that the holes close after a while.  I cannot find any documentation in FS nor in pfSense as to what the timeout is.  Is there a setting in FS to do some kind of keep-alive thing with UPnP to keep, e.g. 5060, open?  Or is it already doing that and pfSense is the issue?
>
> Thanks!
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>    




From matt at webcontracts.co.uk  Tue Feb  9 01:53:47 2010
From: matt at webcontracts.co.uk (Matthew Law)
Date: Tue, 9 Feb 2010 09:53:47 -0000
Subject: [Freeswitch-users] ACL question and js error
In-Reply-To: <20100207145907.GF31942@base.carmickle.com>
References: <ea05d45dca9be2d979fb386f960bc077.squirrel@www.webcontracts.co.uk>
	<20100207145907.GF31942@base.carmickle.com>
Message-ID: <cfe871e3305675257b8c74ed4fb5a11b.squirrel@www.webcontracts.co.uk>

On Sun, February 7, 2010 2:59 pm, Frank Carmickle wrote:
> On Sun, Feb 07, Matthew Law wrote:
>> After some more experiments I have a working replacement for the
>> asterisk
>> box we were using before, which is great.
>>
>> I had problems getting incoming calls to work. Changing the entry in
>> acl.conf.xml from:
>>
>> <list name="domains" default="deny">
>>    <node type="allow" domain="$${domain}"/>
>> </list>
>>
>> to:
>>
>> <list name="domains" default="allow">
>>    <node type="allow" domain="$${domain}"/>
>> </list>
>>
>> and reloading xml works but this gets reverted every time FS starts up.
>> I've scanned the wiki docs and can't see anything pertaining to that.
>> Why/where is this happening and how do I make it the default?  Actually,
>> the question should probably be is it sensible to do that? - the box is
>> out on the internet and I really only want to take incoming calls from
>> voiptalk.org, but I can't find a list of IPs on their site which I could
>> create an acl from...
>
> This is what gateway definitions are for in sofia.

I'm still struggling with this.  How where do I tell sofia to allow
incoming connections from this gateway?

Here's my sip_profiles/external/voiptalk.org.xml with the sensitive stuff
removed:

<include>
        <gateway name="voiptalk">
                <param name="username" value="XXXXXXX" />
                <param name="realm" value="voiptalk.org" />
                <param name="from-user" value="XXXXXXX"/>
                <param name="from-domain" value="voiptalk.org"/>
                <param name="password" value="XXXXXXX" />
                <param name="extension" value="voiptalk-incoming"/>
                <param name="proxy" value="voiptalk.org"/>
                <param name="register-proxy" value="voiptalk.org"/>
                <param name="expire-seconds" value="600"/>
                <param name="register" value="true"/>
                <param name="ping" value="25"/>
        </gateway>
</include>

Do I need to add something to this file or maybe sofia.conf.xml to allow
connections from this domain?  Most everything else is working now, just
banging my head on this.


Thanks,

Matt.




From rupa at rupa.com  Tue Feb  9 02:07:24 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Tue, 9 Feb 2010 04:07:24 -0600
Subject: [Freeswitch-users] UPnP Timeout
In-Reply-To: <9706F7A8-7644-4602-B1DE-289A35F788AA@tlainvestments.com>
References: <9706F7A8-7644-4602-B1DE-289A35F788AA@tlainvestments.com>
Message-ID: <b39273531002090207n11d54e69h8d4ff6f7e71c731c@mail.gmail.com>

I believe FS opens the ports with an indefinite timeout (never close).  I'd
have to double check.  In addition, FS refreshes the NAT mappings on every
keep-alive packet sent by the upnp gateway.  Have you done a nat_map status
once the ports are missing in pfsense to see if fs still thinks the ports
should be open?  What if you do a nat_map republish?  Do the maps get pushed
to pfsense and then stay open for a whlie?

Perhaps pfsense is sending a keep-alive packet that we don't process right
or is invalid? If so, I'd need a packet trace to do analysis.

On Mon, Feb 8, 2010 at 10:21 PM, Troy Anderson <troy at tlainvestments.com>wrote:

> I have been using pfSense (1.2.3) and FS.  FS nicely uses UPnP to poke
> holes in the firewall, but it seems that the holes close after a while.  I
> cannot find any documentation in FS nor in pfSense as to what the timeout
> is.  Is there a setting in FS to do some kind of keep-alive thing with UPnP
> to keep, e.g. 5060, open?  Or is it already doing that and pfSense is the
> issue?
>
> Thanks!
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
-Rupa
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100209/ae52bf49/attachment-0002.html 

From nagalenoj at gmail.com  Tue Feb  9 05:19:34 2010
From: nagalenoj at gmail.com (Nagalenoj H.)
Date: Tue, 9 Feb 2010 18:49:34 +0530
Subject: [Freeswitch-users] Play music to A leg.
Message-ID: <fe67a3341002090519r2cc6e8edm7702b8d012fc2be3@mail.gmail.com>

Dear friends,
  In event socket, I'm originating a call to a number from A leg and till
the person answers the call, I would want to play some music to the A leg,
till I bridge these A leg and originated call.

  I don't want to use bridge, in which I could use ringback.

  So, what is the way to do this??

-- 
Regards,
Nagalenoj H.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100209/67e63d5f/attachment-0002.html 

From peder at networkoblivion.com  Tue Feb  9 05:51:22 2010
From: peder at networkoblivion.com (Peder)
Date: Tue, 9 Feb 2010 07:51:22 -0600
Subject: [Freeswitch-users] Announcement: FreeSWITCH Add Broadsoft
	SCA	Support
In-Reply-To: <d361ca7b1002082110h4e3c60d0p88412f753444ec3d@mail.gmail.com>
References: <87f2f3b91001111742k691966cfs82515db77c13d4d4@mail.gmail.com>
	<d361ca7b1002082110h4e3c60d0p88412f753444ec3d@mail.gmail.com>
Message-ID: <016d01caa98e$f6df25f0$e49d71d0$@com>

It is in addition to the existing settings.  It is for SCA presence on
shared lines.  The "manage presence" setting is for regular registrations.

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Yehavi
Bourvine
Sent: Monday, February 08, 2010 11:10 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Announcement: FreeSWITCH Add Broadsoft SCA
Support

 

Hello,

 

  We currently use the "old" type of presence which is activated by
"manage-presence" coupled with "dbname" and "presence-hosts".

With the new method, does "manage-shared-presence" replace all of the above
or comes in addition?

 

                       Thanks! __Yehavi:

2010/1/12 Michael Collins <msc at freeswitch.org>

We want to let everyone know that FreeSWITCH now supports the Broadsoft SCA
method of doing shared lines. The story is here:

http://www.freeswitch.org/node/227

Tony and Brian spent many hours laboring over this, so please be sure to
show your appreciation to them for this new feature and all of the great
things they do for the FreeSWITCH community and VoIP in general!

-Michael


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org <http://www.freeswitch.org/> 

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100209/16814a2a/attachment-0002.html 

From Prometheus001 at gmx.net  Tue Feb  9 06:09:20 2010
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Tue, 09 Feb 2010 15:09:20 +0100
Subject: [Freeswitch-users] mod_skypiax - help needed with
 modified	snd-dummy (one way audio)
In-Reply-To: <7b197bef1002080839o61b4b1b5k6d5cc39bb05d9a32@mail.gmail.com>
References: <4B60555B.2020004@gmx.net>	<7b197bef1002051319l2fcb5617p1691692f58c65372@mail.gmail.com>
	<4B6CC792.5060608@gmx.net>	<191c3a031002060558r36947d93vcbb46df9e0a7dfde@mail.gmail.com>
	<4B6F3E9A.2020103@gmx.net>	<7b197bef1002080529u1afc7f9av492b0c221ee2d6af@mail.gmail.com>
	<c55d46951002080546p829a881w256bfdb19334c548@mail.gmail.com>
	<7b197bef1002080553qd1cb42dla26425942de83c43@mail.gmail.com>
	<c55d46951002080604k400344b0i8add00b5c6014958@mail.gmail.com>
	<4B703AD0.2080909@gmx.net>
	<7b197bef1002080839o61b4b1b5k6d5cc39bb05d9a32@mail.gmail.com>
Message-ID: <4B716C90.8070109@gmx.net>

Hello Giovanni,

I will try to contact you via IRC (stony)

Best regards
Peter

Giovanni Maruzzelli schrieb:
> Peter,
>
> excuse me but I really do not follow you.
>
> Why you have the normal static build not working?
>
> Also, this is really taking too much of my time. You continue to
> change things, and report issues, without waiting for solutions you
> ask for, then you report something else, and so on... we'll never get
> at the end of this.
>
> If you want, please connect via IRC, and contact me (gmaruzz).
> Or let me connect ssh to your machine.
>
> -gm
>
>
> On Mon, Feb 8, 2010 at 5:24 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>   
>> I got it working now with static build and an older version of skype
>> (skype_static-2.1.0.47).
>>
>> But I still have a problem ongoing with sound quality, resp. one way audio:
>> With original Alsa driver: When Skype calls mod_skypiax => SIP Phone,
>> then the sound from the SIP phone is interupted regularly 2 times a second.
>> Example:
>> When a person on the sip phone speaks
>> "aaaaaaaaaaaaaaaaaaaaaa"
>> the other site hears
>> "aaatataaaatataaaatataaaa"
>> With "t" meaning the interruption of the sound.
>>
>> So I compliled and installed the modified alsa driver as described in
>> the wiki (configure, make and make install, remove old ubuntu sound dir
>> in /lib/modules/2.6.24-24-server/ubuntu/sound + depmod -a + reboot the
>> server.
>>
>> Now the SIP phone is heard loud and clearly without interruption.
>> However the other direction is not heard, so we're at the beginning of
>> the post (one way audio). Only when I really scratch the microphone then
>> I hear some parts of this scratching on the SIP side.
>>
>> So, some more hints are needed.
>>
>> Here's the log, when I start the skype client:
>>  su root -c "/bin/echo 'username password'| DISPLAY=:101
>> /usr/bin/skype1 --pipelogin &" &
>>  /usr/bin/Xvfb :102 -ac &
>>  error opening security policy file /etc/X11/xserver/SecurityPolicy
>>  expected keysym, got XF86KbdLightOnOff: line 70 of pc
>>  expected keysym, got XF86KbdBrightnessDown: line 71 of pc
>>  expected keysym, got XF86KbdBrightnessUp: line 72 of pc
>>  Could not init font path element /usr/share/fonts/X11/cyrillic,
>> removing from list!
>>  Could not init font path element
>> /var/lib/defoma/x-ttcidfont-conf.d/dirs/TrueType, removing from list!
>> But these messages are not critical, right?
>>
>> Best regards
>> Peter
>>
>>
>> Max Bridgewater schrieb:
>>     
>>> Well, it was a couple of weeks ago. I then switched back to 2.0.4. I'm
>>> going to try it again and let you know.
>>>
>>> Max.
>>>
>>> On Mon, Feb 8, 2010 at 8:53 AM, Giovanni Maruzzelli <gmaruzz at celliax.org> wrote:
>>>
>>>       
>>>> Peter is using hardy 64 bit. I checked on that.
>>>>
>>>> But, let me understand: if you're using a static build, why you have a
>>>> problem with QT?
>>>> Is actually Qt to be statically linked...
>>>>
>>>> what is the results of:
>>>>
>>>> ldd skype
>>>>
>>>> Anyway, in a short while I'll check that on centos 5.4 too, and I'll report here
>>>>
>>>> -giovanni
>>>>
>>>> On Mon, Feb 8, 2010 at 2:46 PM, Max Bridgewater
>>>> <max.bridgewater at gmail.com> wrote:
>>>>
>>>>         
>>>>> Interesting; a while back I tried to install Skypiax with the latest
>>>>> static build on Skype.com. I had QT library compatibility problem on a
>>>>> CentOS 5.4 (Final). Giovannni, may I ask what Linux distro you are
>>>>> using?
>>>>>
>>>>> Thanks,
>>>>> max.
>>>>>
>>>>> On Mon, Feb 8, 2010 at 8:29 AM, Giovanni Maruzzelli <gmaruzz at celliax.org> wrote:
>>>>>
>>>>>           
>>>>>> Peter,
>>>>>>
>>>>>> I just tested with the static build you find on skype.com
>>>>>>
>>>>>> I never tested for performances or other issues, there may be (it's a beta).
>>>>>>
>>>>>> But it do not crash on me.
>>>>>>
>>>>>> I have no problem at all.
>>>>>>
>>>>>> If you can give me ssh access I can try to understand why you have so
>>>>>> many problems.
>>>>>>
>>>>>> Or, alternatively, try to follow the wiki. You know, I've not heard
>>>>>> about those problems.
>>>>>>
>>>>>> root at hardy64:~/skype.beta/skype_static-2.1.0.81# ldd skype
>>>>>>        linux-gate.so.1 =>  (0xffffe000)
>>>>>>        libasound.so.2 => /usr/lib32/libasound.so.2 (0xf7eb7000)
>>>>>>        libXv.so.1 => /usr/lib32/libXv.so.1 (0xf7eb2000)
>>>>>>        libXss.so.1 => /usr/lib32/libXss.so.1 (0xf7eae000)
>>>>>>        libSM.so.6 => /usr/lib32/libSM.so.6 (0xf7ea6000)
>>>>>>        libICE.so.6 => /usr/lib32/libICE.so.6 (0xf7e8e000)
>>>>>>        libXi.so.6 => /usr/lib32/libXi.so.6 (0xf7e86000)
>>>>>>        libXrender.so.1 => /usr/lib32/libXrender.so.1 (0xf7e7e000)
>>>>>>        libXrandr.so.2 => /usr/lib32/libXrandr.so.2 (0xf7e78000)
>>>>>>        libfreetype.so.6 => /usr/lib32/libfreetype.so.6 (0xf7e0a000)
>>>>>>        libfontconfig.so.1 => /usr/lib32/libfontconfig.so.1 (0xf7de0000)
>>>>>>        libXext.so.6 => /usr/lib32/libXext.so.6 (0xf7dd2000)
>>>>>>        libX11.so.6 => /usr/lib32/libX11.so.6 (0xf7ceb000)
>>>>>>        libz.so.1 => /usr/lib32/libz.so.1 (0xf7cd6000)
>>>>>>        libgthread-2.0.so.0 => /usr/lib32/libgthread-2.0.so.0 (0xf7cd1000)
>>>>>>        libglib-2.0.so.0 => /usr/lib32/libglib-2.0.so.0 (0xf7c1f000)
>>>>>>        librt.so.1 => /lib32/librt.so.1 (0xf7c16000)
>>>>>>        libdl.so.2 => /lib32/libdl.so.2 (0xf7c12000)
>>>>>>        libpthread.so.0 => /lib32/libpthread.so.0 (0xf7bfa000)
>>>>>>        libstdc++.so.6 => /usr/lib32/libstdc++.so.6 (0xf7b07000)
>>>>>>        libm.so.6 => /lib32/libm.so.6 (0xf7ae2000)
>>>>>>        libgcc_s.so.1 => /usr/lib32/libgcc_s.so.1 (0xf7ad6000)
>>>>>>        libc.so.6 => /lib32/libc.so.6 (0xf7987000)
>>>>>>        libexpat.so.1 => /usr/lib32/libexpat.so.1 (0xf7966000)
>>>>>>        libXau.so.6 => /usr/lib32/libXau.so.6 (0xf7963000)
>>>>>>        libxcb-xlib.so.0 => /usr/lib32/libxcb-xlib.so.0 (0xf7961000)
>>>>>>        libxcb.so.1 => /usr/lib32/libxcb.so.1 (0xf7948000)
>>>>>>        libpcre.so.3 => /usr/lib32/libpcre.so.3 (0xf7921000)
>>>>>>        /lib/ld-linux.so.2 (0xf7f86000)
>>>>>>        libXdmcp.so.6 => /usr/lib32/libXdmcp.so.6 (0xf791c000)
>>>>>>
>>>>>>
>>>>>> On Sun, Feb 7, 2010 at 11:28 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>>
>>>>>>             
>>>>>>> I now used the static Skype binary in order to avoid missing constraints
>>>>>>> to other libraries: It still crashes
>>>>>>> 1st it starts with:
>>>>>>>  process 15431: D-Bus library appears to be incorrectly set up; failed
>>>>>>> to read machine uuid: Failed to open "/var/lib/dbus/machine-id": No such
>>>>>>> file or directory
>>>>>>>  See the manual page for dbus-uuidgen to correct this issue.
>>>>>>> After calling this client it crashes with:
>>>>>>>   /usr/bin/skype: symbol lookup error: /usr/bin/skype: undefined
>>>>>>> symbol: _ZN10QBoxLayout13addSpacerItemEP11QSpacerItem
>>>>>>>
>>>>>>> Any hints, where I may get an older Skype client?
>>>>>>>
>>>>>>> Best regards
>>>>>>> Peter
>>>>>>>
>>>>>>> Anthony Minessale schrieb:
>>>>>>>
>>>>>>>               
>>>>>>>> Quick fyi centos 5.4 has some memory bugs in libc don't go beyond 5.3
>>>>>>>> until its fixed.
>>>>>>>>
>>>>>>>>
>>>>>>>>                 
>>>>>>>>> On Feb 5, 2010 7:42 PM, "Peter P GMX" <Prometheus001 at gmx.net
>>>>>>>>> <mailto:Prometheus001 at gmx.net>> wrote:
>>>>>>>>>
>>>>>>>>> Skype starts, but as soon as it receives a call it crashes with:
>>>>>>>>>
>>>>>>>>> /usr/bin/skype: symbol lookup error: /usr/bin/skype: undefined symbol:
>>>>>>>>>
>>>>>>>>> _ZN10QBoxLayout13addSpacerItemEP11QSpacerItem
>>>>>>>>>
>>>>>>>>> I think the 8.10 version dos not work with8.04.
>>>>>>>>>
>>>>>>>>> Any hints, where I may get an older Skype client? I may also try the
>>>>>>>>> static skype client.
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> Best regards
>>>>>>>>> Peter
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> .
>>>>>>>>> Giovanni Maruzzelli schrieb:
>>>>>>>>>
>>>>>>>>>                   
>>>>>>>>>> that's not at all a fatal error.
>>>>>>>>>> I believe...
>>>>>>>>>>
>>>>>>>>>>                     
>>>>>>>> ------------------------------------------------------------------------
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>>                 
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>>               
>>>>>> --
>>>>>> Sincerely,
>>>>>>
>>>>>> Giovanni Maruzzelli
>>>>>> Cell : +39-347-2665618
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>>             
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>>           
>>>> --
>>>> Sincerely,
>>>>
>>>> Giovanni Maruzzelli
>>>> Cell : +39-347-2665618
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>         
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>       
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>     
>
>
>
>   



From msc at freeswitch.org  Tue Feb  9 07:35:52 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 9 Feb 2010 07:35:52 -0800
Subject: [Freeswitch-users] Last call: buy the devs dinner!
Message-ID: <87f2f3b91002090735g1c6c69eby5274f6dfa9127fc5@mail.gmail.com>

Hey all,

Thanks so much for the donations that have come in already! We appreciate
your generosity. The dev team really wants to release 1.0.5 but they're
kinda hungry! :) Please hit the PayPal button on the main
freeswitch.orgpage to drop a few dollars in the hat. Also, keep in
mind that we have the
"extended family" of developers all here so it's not just Tony, Mike, and
Brian. Let's all pitch in and have a great dinner for them.

Thanks!
-Michael
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100209/31eaa19d/attachment-0002.html 

From msc at freeswitch.org  Tue Feb  9 07:37:13 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 9 Feb 2010 07:37:13 -0800
Subject: [Freeswitch-users] Play music to A leg.
In-Reply-To: <fe67a3341002090519r2cc6e8edm7702b8d012fc2be3@mail.gmail.com>
References: <fe67a3341002090519r2cc6e8edm7702b8d012fc2be3@mail.gmail.com>
Message-ID: <87f2f3b91002090737l4efa8en29bde9457fef7ea5@mail.gmail.com>

On Tue, Feb 9, 2010 at 5:19 AM, Nagalenoj H. <nagalenoj at gmail.com> wrote:

> Dear friends,
>   In event socket, I'm originating a call to a number from A leg and till
> the person answers the call, I would want to play some music to the A leg,
> till I bridge these A leg and originated call.
>
>   I don't want to use bridge, in which I could use ringback.
>
You don't want to use bridge because... why?
-MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100209/dc809e44/attachment-0002.html 

From troy at tlainvestments.com  Tue Feb  9 07:44:02 2010
From: troy at tlainvestments.com (Troy Anderson)
Date: Tue, 9 Feb 2010 08:44:02 -0700
Subject: [Freeswitch-users] UPnP Timeout
In-Reply-To: <b39273531002090207n11d54e69h8d4ff6f7e71c731c@mail.gmail.com>
References: <9706F7A8-7644-4602-B1DE-289A35F788AA@tlainvestments.com>
	<b39273531002090207n11d54e69h8d4ff6f7e71c731c@mail.gmail.com>
Message-ID: <47D355D1-CDF6-4D79-8A64-1134EBEB36BA@tlainvestments.com>

I did do a nap_map status when the ports were missing from pfSense and FS thought they were still open.  I didn't know about nat_map republish, but will try next time.  I think the timeframe is days, so this is kind of hard to diagnose.  I may add a periodic nat_map republish from fs_cli to our production systems.

In any case, I'll keep an eye on it and try nat_map republish next time pfSense drops the ports to be sure that is working in this environment.

In the meantime, which .c file(s) can I peruse to learn more?

Thanks!
Troy

On Feb 9, 2010, at 3:07 AM, Rupa Schomaker wrote:

> I believe FS opens the ports with an indefinite timeout (never close).  I'd have to double check.  In addition, FS refreshes the NAT mappings on every keep-alive packet sent by the upnp gateway.  Have you done a nat_map status once the ports are missing in pfsense to see if fs still thinks the ports should be open?  What if you do a nat_map republish?  Do the maps get pushed to pfsense and then stay open for a whlie?
> 
> Perhaps pfsense is sending a keep-alive packet that we don't process right or is invalid? If so, I'd need a packet trace to do analysis.  
> 
> On Mon, Feb 8, 2010 at 10:21 PM, Troy Anderson <troy at tlainvestments.com> wrote:
> I have been using pfSense (1.2.3) and FS.  FS nicely uses UPnP to poke holes in the firewall, but it seems that the holes close after a while.  I cannot find any documentation in FS nor in pfSense as to what the timeout is.  Is there a setting in FS to do some kind of keep-alive thing with UPnP to keep, e.g. 5060, open?  Or is it already doing that and pfSense is the issue?
> 
> Thanks!
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> -- 
> -Rupa
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100209/5ac7b97c/attachment-0002.html 

From msc at freeswitch.org  Tue Feb  9 07:45:26 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 9 Feb 2010 07:45:26 -0800
Subject: [Freeswitch-users] ACL question and js error
In-Reply-To: <cfe871e3305675257b8c74ed4fb5a11b.squirrel@www.webcontracts.co.uk>
References: <ea05d45dca9be2d979fb386f960bc077.squirrel@www.webcontracts.co.uk>
	<20100207145907.GF31942@base.carmickle.com>
	<cfe871e3305675257b8c74ed4fb5a11b.squirrel@www.webcontracts.co.uk>
Message-ID: <87f2f3b91002090745v2128714byf1f7574d75f4449c@mail.gmail.com>

On Tue, Feb 9, 2010 at 1:53 AM, Matthew Law <matt at webcontracts.co.uk> wrote:

> On Sun, February 7, 2010 2:59 pm, Frank Carmickle wrote:
> > On Sun, Feb 07, Matthew Law wrote:
> >> After some more experiments I have a working replacement for the
> >> asterisk
> >> box we were using before, which is great.
> >>
> >> I had problems getting incoming calls to work. Changing the entry in
> >> acl.conf.xml from:
> >>
> >> <list name="domains" default="deny">
> >>    <node type="allow" domain="$${domain}"/>
> >> </list>
> >>
> >> to:
> >>
> >> <list name="domains" default="allow">
> >>    <node type="allow" domain="$${domain}"/>
> >> </list>
> >>
> >> and reloading xml works but this gets reverted every time FS starts up.
> >> I've scanned the wiki docs and can't see anything pertaining to that.
> >> Why/where is this happening and how do I make it the default?  Actually,
> >> the question should probably be is it sensible to do that? - the box is
> >> out on the internet and I really only want to take incoming calls from
> >> voiptalk.org, but I can't find a list of IPs on their site which I
> could
> >> create an acl from...
> >
> > This is what gateway definitions are for in sofia.
>
> I'm still struggling with this.  How where do I tell sofia to allow
> incoming connections from this gateway?
>
> Here's my sip_profiles/external/voiptalk.org.xml with the sensitive stuff
> removed:
>
> <include>
>        <gateway name="voiptalk">
>                <param name="username" value="XXXXXXX" />
>                <param name="realm" value="voiptalk.org" />
>                <param name="from-user" value="XXXXXXX"/>
>                <param name="from-domain" value="voiptalk.org"/>
>                <param name="password" value="XXXXXXX" />
>                <param name="extension" value="voiptalk-incoming"/>
>                <param name="proxy" value="voiptalk.org"/>
>                <param name="register-proxy" value="voiptalk.org"/>
>                <param name="expire-seconds" value="600"/>
>                <param name="register" value="true"/>
>                <param name="ping" value="25"/>
>        </gateway>
> </include>
>
> Do I need to add something to this file or maybe sofia.conf.xml to allow
> connections from this domain?  Most everything else is working now, just
> banging my head on this.
>
> Matt,

Are you trying to let calls in from voiptalk.org? Do you want to auth all
inbound calls or do you just want to blanket allow them and handle them in
the dialplan? If you just want to allow calls in from the voiptalk.org IP
address then you need to use the cidr tag in acl.conf.xml:
<list name="domains" default="deny">
    <node type="allow" domain="$${domain}"/>
    <node type="allow" cidr="1.2.3.4/32"/>
 </list>

-MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100209/7685f314/attachment-0002.html 

From rupa at rupa.com  Tue Feb  9 08:06:11 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Tue, 9 Feb 2010 10:06:11 -0600
Subject: [Freeswitch-users] UPnP Timeout
In-Reply-To: <47D355D1-CDF6-4D79-8A64-1134EBEB36BA@tlainvestments.com>
References: <9706F7A8-7644-4602-B1DE-289A35F788AA@tlainvestments.com>
	<b39273531002090207n11d54e69h8d4ff6f7e71c731c@mail.gmail.com>
	<47D355D1-CDF6-4D79-8A64-1134EBEB36BA@tlainvestments.com>
Message-ID: <b39273531002090806k7fa7f02dyeabd79586736041b@mail.gmail.com>

perhaps pfSense isn't sending the keep-alive packets like we expect?

You can look in switch_nat.c for details.

On Tue, Feb 9, 2010 at 9:44 AM, Troy Anderson <troy at tlainvestments.com>wrote:

> I did do a nap_map status when the ports were missing from pfSense and FS
> thought they were still open.  I didn't know about nat_map republish, but
> will try next time.  I think the timeframe is days, so this is kind of hard
> to diagnose.  I may add a periodic nat_map republish from fs_cli to our
> production systems.
>
> In any case, I'll keep an eye on it and try nat_map republish next time
> pfSense drops the ports to be sure that is working in this environment.
>
> In the meantime, which .c file(s) can I peruse to learn more?
>
> Thanks!
> Troy
>
>
> On Feb 9, 2010, at 3:07 AM, Rupa Schomaker wrote:
>
> I believe FS opens the ports with an indefinite timeout (never close).  I'd
> have to double check.  In addition, FS refreshes the NAT mappings on every
> keep-alive packet sent by the upnp gateway.  Have you done a nat_map status
> once the ports are missing in pfsense to see if fs still thinks the ports
> should be open?  What if you do a nat_map republish?  Do the maps get pushed
> to pfsense and then stay open for a whlie?
>
> Perhaps pfsense is sending a keep-alive packet that we don't process right
> or is invalid? If so, I'd need a packet trace to do analysis.
>
> On Mon, Feb 8, 2010 at 10:21 PM, Troy Anderson <troy at tlainvestments.com>wrote:
>
>> I have been using pfSense (1.2.3) and FS.  FS nicely uses UPnP to poke
>> holes in the firewall, but it seems that the holes close after a while.  I
>> cannot find any documentation in FS nor in pfSense as to what the timeout
>> is.  Is there a setting in FS to do some kind of keep-alive thing with UPnP
>> to keep, e.g. 5060, open?  Or is it already doing that and pfSense is the
>> issue?
>>
>> Thanks!
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> -Rupa
>  _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100209/5b58daae/attachment-0002.html 

From Prometheus001 at gmx.net  Tue Feb  9 08:07:59 2010
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Tue, 09 Feb 2010 17:07:59 +0100
Subject: [Freeswitch-users] Last call: buy the devs dinner!
In-Reply-To: <87f2f3b91002090735g1c6c69eby5274f6dfa9127fc5@mail.gmail.com>
References: <87f2f3b91002090735g1c6c69eby5274f6dfa9127fc5@mail.gmail.com>
Message-ID: <4B71885F.5090908@gmx.net>

Hello Michael,

just hit the paypal button. Enjoy your dinner! I think it's not just
dinner, it will be also 50% work I think (discussing about issues and
new features etc.) which brings additional benefits to the copmmunity.
Thanks for the great work you all have done so far.

Best regards
Peter

Michael Collins schrieb:
> Hey all,
>
> Thanks so much for the donations that have come in already! We
> appreciate your generosity. The dev team really wants to release 1.0.5
> but they're kinda hungry! :) Please hit the PayPal button on the main
> freeswitch.org <http://freeswitch.org> page to drop a few dollars in
> the hat. Also, keep in mind that we have the "extended family" of
> developers all here so it's not just Tony, Mike, and Brian. Let's all
> pitch in and have a great dinner for them.
>
> Thanks!
> -Michael
> ------------------------------------------------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   



From yehavi.bourvine at gmail.com  Tue Feb  9 08:17:57 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Tue, 9 Feb 2010 18:17:57 +0200
Subject: [Freeswitch-users] Announcement: FreeSWITCH Add Broadsoft SCA
	Support
In-Reply-To: <016d01caa98e$f6df25f0$e49d71d0$@com>
References: <87f2f3b91001111742k691966cfs82515db77c13d4d4@mail.gmail.com>
	<d361ca7b1002082110h4e3c60d0p88412f753444ec3d@mail.gmail.com>
	<016d01caa98e$f6df25f0$e49d71d0$@com>
Message-ID: <d361ca7b1002090817h4aa3c13dka4d4db8b4adc7606@mail.gmail.com>

Thanks!
I've managed to make it work today with Polcyoms. It works inside a profile
but does not across profiles. In our case this is not a limitation.

                           Thanks! __Yehavi:

 2010/2/9 Peder <peder at networkoblivion.com>

>   It is in addition to the existing settings.  It is for SCA presence on
> shared lines.  The ?manage presence? setting is for regular registrations.
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Yehavi
> Bourvine
> *Sent:* Monday, February 08, 2010 11:10 PM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Announcement: FreeSWITCH Add Broadsoft
> SCA Support
>
>
>
> Hello,
>
>
>
>   We currently use the "old" type of presence which is activated by
> "manage-presence" coupled with "dbname" and "presence-hosts".
>
> With the new method, does "manage-shared-presence" replace all of the above
> or comes in addition?
>
>
>
>                        Thanks! __Yehavi:
>
> 2010/1/12 Michael Collins <msc at freeswitch.org>
>
>  We want to let everyone know that FreeSWITCH now supports the Broadsoft
> SCA method of doing shared lines. The story is here:
>
> http://www.freeswitch.org/node/227
>
> Tony and Brian spent many hours laboring over this, so please be sure to
> show your appreciation to them for this new feature and all of the great
> things they do for the FreeSWITCH community and VoIP in general!
>
> -Michael
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100209/dfb636c2/attachment-0002.html 

From brian at freeswitch.org  Tue Feb  9 08:21:41 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 9 Feb 2010 10:21:41 -0600
Subject: [Freeswitch-users] Announcement: FreeSWITCH Add Broadsoft SCA
	Support
In-Reply-To: <d361ca7b1002090817h4aa3c13dka4d4db8b4adc7606@mail.gmail.com>
References: <87f2f3b91001111742k691966cfs82515db77c13d4d4@mail.gmail.com>
	<d361ca7b1002082110h4e3c60d0p88412f753444ec3d@mail.gmail.com>
	<016d01caa98e$f6df25f0$e49d71d0$@com>
	<d361ca7b1002090817h4aa3c13dka4d4db8b4adc7606@mail.gmail.com>
Message-ID: <1C0A1BEE-3480-4A33-916F-52067AF5B9C9@freeswitch.org>

It will work across profiles if you bond them.  :P

/b

On Feb 9, 2010, at 10:17 AM, Yehavi Bourvine wrote:

> Thanks!
> I've managed to make it work today with Polcyoms. It works inside a profile but does not across profiles. In our case this is not a limitation.
>  
>                            Thanks! __Yehavi:




From yehavi.bourvine at gmail.com  Tue Feb  9 08:38:53 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Tue, 9 Feb 2010 18:38:53 +0200
Subject: [Freeswitch-users] Announcement: FreeSWITCH Add Broadsoft SCA
	Support
In-Reply-To: <1C0A1BEE-3480-4A33-916F-52067AF5B9C9@freeswitch.org>
References: <87f2f3b91001111742k691966cfs82515db77c13d4d4@mail.gmail.com>
	<d361ca7b1002082110h4e3c60d0p88412f753444ec3d@mail.gmail.com>
	<016d01caa98e$f6df25f0$e49d71d0$@com>
	<d361ca7b1002090817h4aa3c13dka4d4db8b4adc7606@mail.gmail.com>
	<1C0A1BEE-3480-4A33-916F-52067AF5B9C9@freeswitch.org>
Message-ID: <d361ca7b1002090838j6c42dcd2of8affb5f281900fd@mail.gmail.com>

What do you mean by "bonding them"?

             Thanks! __Yehavi:

2010/2/9 Brian West <brian at freeswitch.org>

> It will work across profiles if you bond them.  :P
>
> /b
>
> On Feb 9, 2010, at 10:17 AM, Yehavi Bourvine wrote:
>
> > Thanks!
> > I've managed to make it work today with Polcyoms. It works inside a
> profile but does not across profiles. In our case this is not a limitation.
> >
> >                            Thanks! __Yehavi:
>
>
>  _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100209/48f4b4b2/attachment-0002.html 

From carlos.talbot at gmail.com  Tue Feb  9 08:51:02 2010
From: carlos.talbot at gmail.com (Carlos Talbot)
Date: Tue, 9 Feb 2010 10:51:02 -0600
Subject: [Freeswitch-users] SIP over TCP with Sipdroid,
	an Android SIP client
In-Reply-To: <5800526b1002051346g3890f152o1d939faa054811e6@mail.gmail.com>
References: <5800526b1002051346g3890f152o1d939faa054811e6@mail.gmail.com>
Message-ID: <5800526b1002090851x578310ddy205fa5a373b3d8be@mail.gmail.com>

FYI,

for those interested this issue has been identified and fixed with a simple
patch:

http://code.google.com/p/sipdroid/issues/detail?id=311#c2
<http://code.google.com/p/sipdroid/issues/detail?id=311#c2>

On Fri, Feb 5, 2010 at 3:46 PM, Carlos Talbot <carlos.talbot at gmail.com>wrote:

>
> Anyone use sipdroid on their Andorid phone? For the most part it works with
> the exception of when using SIP over TCP. For some reason, after 30 seconds
> into a call FreeSWITCH sends a bye and drops the call. Why use TCP? The
> author claims significantly increased standby times using SIP TCP over 3g:
> http://code.google.com/p/sipdroid/wiki/NewStandbyTechnique
>
> According to Brian it might be because the phone is not setting a transport
> in the contact field and FS is falling back to UDP.
>
> This is on r16557. Here's a sip trace along with call graph:
>
> http://pastebin.freeswitch.org/12064
>
> regards,
>
> Carlos
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100209/73f5464d/attachment-0002.html 

From matt at webcontracts.co.uk  Tue Feb  9 08:59:39 2010
From: matt at webcontracts.co.uk (Matthew Law)
Date: Tue, 9 Feb 2010 16:59:39 -0000
Subject: [Freeswitch-users] ACL question and js error
In-Reply-To: <87f2f3b91002090745v2128714byf1f7574d75f4449c@mail.gmail.com>
References: <ea05d45dca9be2d979fb386f960bc077.squirrel@www.webcontracts.co.uk>
	<20100207145907.GF31942@base.carmickle.com>
	<cfe871e3305675257b8c74ed4fb5a11b.squirrel@www.webcontracts.co.uk>
	<87f2f3b91002090745v2128714byf1f7574d75f4449c@mail.gmail.com>
Message-ID: <1a2b9e340124c8ef35c7fa5991bf3a5f.squirrel@www.webcontracts.co.uk>

On Tue, February 9, 2010 3:45 pm, Michael Collins wrote:
> Are you trying to let calls in from voiptalk.org? Do you want to auth all
> inbound calls or do you just want to blanket allow them and handle them in
> the dialplan? If you just want to allow calls in from the voiptalk.org IP
> address then you need to use the cidr tag in acl.conf.xml:
> <list name="domains" default="deny">
>     <node type="allow" domain="$${domain}"/>
>     <node type="allow" cidr="1.2.3.4/32"/>
>  </list>

AFAIK, they don't publish their IP addresses (I see incoming calls from
lots of different IPs in various subnets).  So at the moment I just want
to allow all and filter in the dialplan (which I think I am doing now).  I
just need a config that will survive a reboot and changing acl.conf.xml
doesn't at the moment.


Thanks,

Matt.




From brian at freeswitch.org  Tue Feb  9 09:02:30 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 9 Feb 2010 11:02:30 -0600
Subject: [Freeswitch-users] Announcement: FreeSWITCH Add Broadsoft SCA
	Support
In-Reply-To: <d361ca7b1002090838j6c42dcd2of8affb5f281900fd@mail.gmail.com>
References: <87f2f3b91001111742k691966cfs82515db77c13d4d4@mail.gmail.com>
	<d361ca7b1002082110h4e3c60d0p88412f753444ec3d@mail.gmail.com>
	<016d01caa98e$f6df25f0$e49d71d0$@com>
	<d361ca7b1002090817h4aa3c13dka4d4db8b4adc7606@mail.gmail.com>
	<1C0A1BEE-3480-4A33-916F-52067AF5B9C9@freeswitch.org>
	<d361ca7b1002090838j6c42dcd2of8affb5f281900fd@mail.gmail.com>
Message-ID: <767B677E-0D19-43A9-89DE-D96EC4A715EC@freeswitch.org>

    <!--<param name="dbname" value="share_presence"/>-->
    <!--<param name="presence-hosts" value="$${domain}"/>-->


/b

On Feb 9, 2010, at 10:38 AM, Yehavi Bourvine wrote:

> What do you mean by "bonding them"?
>  
>              Thanks! __Yehavi:
> 
> 2010/2/9 Brian West <brian at freeswitch.org>
> It will work across profiles if you bond them.  :P
> 
> /b
> 
> On Feb 9, 2010, at 10:17 AM, Yehavi Bourvine wrote:
> 
> > Thanks!
> > I've managed to make it work today with Polcyoms. It works inside a profile but does not across profiles. In our case this is not a limitation.
> >
> >                            Thanks! __Yehavi:
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100209/4432324d/attachment-0002.html 

From m.sobkow at marketelsystems.com  Tue Feb  9 10:46:54 2010
From: m.sobkow at marketelsystems.com (Mark Sobkow)
Date: Tue, 09 Feb 2010 12:46:54 -0600
Subject: [Freeswitch-users] Having trouble establishing a call
Message-ID: <4B71AD9E.7090606@marketelsystems.com>

We're using Erlang to serve up the configurations to Freeswitch.  I've 
got things configured such that I can place a call from a SIP phone 
registered to extension 5000 to our "external" SIP provider (our 
Asterisk installation), but I can't place a call to extension 5001 from 
5000.  Below is the trace log Freeswitch produces when I attempt to do so.

Any suggestions as to what I should be looking at?  The directory seems 
to be getting served up correctly, as it provides the passwords both SIP 
softphones are using to register with Freeswitch.  I'd have thought that 
once they've registered with FS, the extension would automatically be 
recognized when an incoming call is placed or bridged, but such does not 
seem to be the case.



2010-02-09 12:43:10.394647 [NOTICE] switch_channel.c:660 New Channel 
sofia/external/5000 at testsrv.marketel [9854f2ea-cf04-4a48-a560-467cbd86bb1d]
2010-02-09 12:43:10.394647 [DEBUG] switch_core_state_machine.c:316 
(sofia/external/5000 at testsrv.marketel) Running State Change CS_NEW
2010-02-09 12:43:10.394647 [DEBUG] switch_core_state_machine.c:322 
(sofia/external/5000 at testsrv.marketel) State NEW
2010-02-09 12:43:10.394647 [DEBUG] sofia.c:4019 Channel 
sofia/external/5000 at testsrv.marketel entering state [received][100]
2010-02-09 12:43:10.394647 [DEBUG] sofia.c:4030 Remote SDP:
v=0
o=- 4197664938 0 IN IP4 10.77.0.126
s=SIPPER for phoner
c=IN IP4 10.77.0.126
t=0 0
m=audio 5062 RTP/AVP 8 0 2 3 97 9 111 112 113 114 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:9 G722/8000
a=rtpmap:111 speex/16000
a=rtpmap:112 G726-16/8000
a=rtpmap:113 G726-24/8000
a=rtpmap:114 G726-40/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

2010-02-09 12:43:10.394647 [DEBUG] sofia_glue.c:3388 Audio Codec Compare 
[PCMA:8:8000:20]/[PCMU:0:8000:20]
2010-02-09 12:43:10.394647 [DEBUG] sofia_glue.c:3388 Audio Codec Compare 
[PCMA:8:8000:20]/[PCMA:8:8000:20]
2010-02-09 12:43:10.394647 [DEBUG] sofia_glue.c:2211 Set Codec 
sofia/external/5000 at testsrv.marketel PCMA/8000 20 ms 160 samples
2010-02-09 12:43:10.394647 [DEBUG] sofia_glue.c:3344 Set 2833 dtmf 
payload to 101
2010-02-09 12:43:10.394647 [DEBUG] sofia.c:4178 
(sofia/external/5000 at testsrv.marketel) State Change CS_NEW -> CS_INIT
2010-02-09 12:43:10.394647 [DEBUG] switch_core_session.c:1019 Send 
signal sofia/external/5000 at testsrv.marketel [BREAK]
2010-02-09 12:43:10.394647 [DEBUG] switch_core_state_machine.c:316 
(sofia/external/5000 at testsrv.marketel) Running State Change CS_INIT
2010-02-09 12:43:10.394647 [DEBUG] switch_core_state_machine.c:340 
(sofia/external/5000 at testsrv.marketel) State INIT
2010-02-09 12:43:10.394647 [DEBUG] mod_sofia.c:83 
sofia/external/5000 at testsrv.marketel SOFIA INIT
2010-02-09 12:43:10.394647 [DEBUG] mod_sofia.c:111 
(sofia/external/5000 at testsrv.marketel) State Change CS_INIT -> CS_ROUTING
2010-02-09 12:43:10.394647 [DEBUG] switch_core_session.c:1019 Send 
signal sofia/external/5000 at testsrv.marketel [BREAK]
2010-02-09 12:43:10.394647 [DEBUG] switch_core_state_machine.c:340 
(sofia/external/5000 at testsrv.marketel) State INIT going to sleep
2010-02-09 12:43:10.394647 [DEBUG] switch_core_state_machine.c:316 
(sofia/external/5000 at testsrv.marketel) Running State Change CS_ROUTING
2010-02-09 12:43:10.394647 [DEBUG] switch_core_state_machine.c:343 
(sofia/external/5000 at testsrv.marketel) State ROUTING
2010-02-09 12:43:10.394647 [DEBUG] mod_sofia.c:132 
sofia/external/5000 at testsrv.marketel SOFIA ROUTING
2010-02-09 12:43:10.394647 [DEBUG] switch_core_state_machine.c:78 
sofia/external/5000 at testsrv.marketel Standard ROUTING
2010-02-09 12:43:10.394647 [INFO] mod_dialplan_xml.c:408 Processing MSS 
Testing->5001 in context public
2010-02-09 12:43:10.394647 [DEBUG] mod_erlang_event.c:387 looking for 
bindings
2010-02-09 12:43:10.394647 [DEBUG] mod_erlang_event.c:403 binding for 
(null) in section dialplan with key (null) and value (null) requested 
from node pursuit at testsrv
2010-02-09 12:43:10.406527 [DEBUG] handle_msg.c:191 Found waiting slot 
for 7077ba5c-0aae-44fd-87fd-e7aa48f62659
2010-02-09 12:43:10.406527 [DEBUG] mod_erlang_event.c:456 got data 
<document type="freeswitch/xml">
  <section name="dialplan" description="Dial Plan For FreeSwitch">
    <context name="public">
      <extension name="Operator">
        <condition field="destination_number" expression="^(0)$">
          <action application="bridge" 
data="sofia/external/160 at rats.marketel"/>
        </condition>
      </extension>
      <extension name="InternalFS">
        <condition field="destination_number" expression="^(\d\d\d\d)$">
          <action application="bridge" 
data="sofia/external/$0 at testsrv.marketel"/>
        </condition>
      </extension>
      <extension name="RatsExtension">
        <condition field="destination_number" expression="^(\d\d\d)$">
          <action application="bridge" 
data="sofia/external/$0 at rats.marketel"/>
        </condition>
      </extension>
      <extension name="LocalCall">
        <condition field="destination_number" 
expression="^(9\d\d\d\d\d\d\d)$">
          <action application="bridge" 
data="sofia/external/$0 at rats.marketel"/>
        </condition>
      </extension>
      <extension name="LongDistanceCall">
        <condition field="destination_number" 
expression="^91(\d\d\d\d\d\d\d\d\d\d)$">
          <action application="bridge" 
data="sofia/external/$0 at rats.marketel"/>
        </condition>
      </extension>
      <extension name="DialerConference">
        <condition field="destination_number" expression="^8(\d\d\d)$">
          <action application="conference" data="conf$0 at dialer"/>
        </condition>
      </extension>
    </context>
    <context name="default">
      <extension name="InternalFS">
        <condition field="destination_number" expression="^(\d\d\d\d)$">
          <action application="bridge" 
data="sofia/external/$0 at testsrv.marketel"/>
        </condition>
      </extension>
      <extension name="RatsExtension">
        <condition field="destination_number" expression="^(\d\d\d)$">
          <action application="bridge" 
data="sofia/external/$0 at rats.marketel"/>
        </condition>
      </extension>
      <extension name="LocalCall">
        <condition field="destination_number" 
expression="^(9\d\d\d\d\d\d\d)$">
          <action application="bridge" 
data="sofia/external/$0 at rats.marketel"/>
        </condition>
      </extension>
      <extension name="LongDistanceCall">
        <condition field="destination_number" 
expression="^91(\d\d\d\d\d\d\d\d\d\d)$">
          <action application="bridge" 
data="sofia/external/$0 at rats.marketel"/>
        </condition>
      </extension>
      <extension name="DialerConference">
        <condition field="destination_number" expression="^8(\d\d\d)$">
          <action application="conference" data="conf$0 at dialer"/>
        </condition>
      </extension>
    </context>
  </section>
</document> after 10 milliseconds!
2010-02-09 12:43:10.406527 [DEBUG] mod_erlang_event.c:463 XML parsed OK!
Dialplan: sofia/external/5000 at testsrv.marketel parsing 
[public->Operator] continue=false
Dialplan: sofia/external/5000 at testsrv.marketel Regex (FAIL) [Operator] 
destination_number(5001) =~ /^(0)$/ break=on-false
Dialplan: sofia/external/5000 at testsrv.marketel parsing 
[public->InternalFS] continue=false
Dialplan: sofia/external/5000 at testsrv.marketel Regex (PASS) [InternalFS] 
destination_number(5001) =~ /^(\d\d\d\d)$/ break=on-false
Dialplan: sofia/external/5000 at testsrv.marketel Action 
bridge(sofia/external/5001 at testsrv.marketel)
2010-02-09 12:43:10.406527 [DEBUG] switch_core_state_machine.c:122 
(sofia/external/5000 at testsrv.marketel) State Change CS_ROUTING -> CS_EXECUTE
2010-02-09 12:43:10.406527 [DEBUG] switch_core_session.c:1019 Send 
signal sofia/external/5000 at testsrv.marketel [BREAK]
2010-02-09 12:43:10.406527 [DEBUG] switch_core_state_machine.c:343 
(sofia/external/5000 at testsrv.marketel) State ROUTING going to sleep
2010-02-09 12:43:10.406527 [DEBUG] switch_core_state_machine.c:316 
(sofia/external/5000 at testsrv.marketel) Running State Change CS_EXECUTE
2010-02-09 12:43:10.406527 [DEBUG] switch_core_state_machine.c:350 
(sofia/external/5000 at testsrv.marketel) State EXECUTE
2010-02-09 12:43:10.406527 [DEBUG] mod_sofia.c:181 
sofia/external/5000 at testsrv.marketel SOFIA EXECUTE
2010-02-09 12:43:10.406527 [DEBUG] switch_core_state_machine.c:159 
sofia/external/5000 at testsrv.marketel Standard EXECUTE
EXECUTE sofia/external/5000 at testsrv.marketel 
bridge(sofia/external/5001 at testsrv.marketel)
2010-02-09 12:43:10.406527 [NOTICE] switch_channel.c:660 New Channel 
sofia/external/5001 at testsrv.marketel [813c2e55-8acf-486d-b3b6-3244edf5529a]
2010-02-09 12:43:10.406527 [DEBUG] mod_sofia.c:3317 
(sofia/external/5001 at testsrv.marketel) State Change CS_NEW -> CS_INIT
2010-02-09 12:43:10.406527 [DEBUG] switch_core_session.c:1019 Send 
signal sofia/external/5001 at testsrv.marketel [BREAK]
2010-02-09 12:43:10.414137 [DEBUG] switch_core_state_machine.c:316 
(sofia/external/5001 at testsrv.marketel) Running State Change CS_INIT
2010-02-09 12:43:10.414137 [DEBUG] switch_core_state_machine.c:340 
(sofia/external/5001 at testsrv.marketel) State INIT
2010-02-09 12:43:10.414137 [DEBUG] mod_sofia.c:83 
sofia/external/5001 at testsrv.marketel SOFIA INIT
2010-02-09 12:43:10.414137 [DEBUG] mod_sofia.c:111 
(sofia/external/5001 at testsrv.marketel) State Change CS_INIT -> CS_ROUTING
2010-02-09 12:43:10.414137 [DEBUG] switch_core_session.c:1019 Send 
signal sofia/external/5001 at testsrv.marketel [BREAK]
2010-02-09 12:43:10.414137 [DEBUG] switch_core_state_machine.c:340 
(sofia/external/5001 at testsrv.marketel) State INIT going to sleep
2010-02-09 12:43:10.414137 [DEBUG] switch_core_state_machine.c:316 
(sofia/external/5001 at testsrv.marketel) Running State Change CS_ROUTING
2010-02-09 12:43:10.414137 [DEBUG] switch_core_state_machine.c:343 
(sofia/external/5001 at testsrv.marketel) State ROUTING
2010-02-09 12:43:10.414137 [DEBUG] mod_sofia.c:132 
sofia/external/5001 at testsrv.marketel SOFIA ROUTING
2010-02-09 12:43:10.414137 [DEBUG] switch_ivr_originate.c:66 
(sofia/external/5001 at testsrv.marketel) State Change CS_ROUTING -> 
CS_CONSUME_MEDIA
2010-02-09 12:43:10.414137 [DEBUG] switch_core_session.c:1019 Send 
signal sofia/external/5001 at testsrv.marketel [BREAK]
2010-02-09 12:43:10.414137 [DEBUG] switch_core_state_machine.c:343 
(sofia/external/5001 at testsrv.marketel) State ROUTING going to sleep
2010-02-09 12:43:10.414137 [DEBUG] switch_core_state_machine.c:316 
(sofia/external/5001 at testsrv.marketel) Running State Change CS_CONSUME_MEDIA
2010-02-09 12:43:10.414137 [DEBUG] switch_core_state_machine.c:362 
(sofia/external/5001 at testsrv.marketel) State CONSUME_MEDIA
2010-02-09 12:43:10.414137 [DEBUG] switch_core_state_machine.c:362 
(sofia/external/5001 at testsrv.marketel) State CONSUME_MEDIA going to sleep
2010-02-09 12:43:10.414137 [DEBUG] sofia.c:4019 Channel 
sofia/external/5001 at testsrv.marketel entering state [calling][0]
2010-02-09 12:43:10.414137 [DEBUG] sofia.c:4019 Channel 
sofia/external/5001 at testsrv.marketel entering state [terminated][503]
2010-02-09 12:43:10.414137 [NOTICE] sofia.c:4663 Hangup 
sofia/external/5001 at testsrv.marketel [CS_CONSUME_MEDIA] 
[NORMAL_TEMPORARY_FAILURE]
2010-02-09 12:43:10.414137 [DEBUG] switch_ivr_originate.c:3209 Originate 
Resulted in Error Cause: 41 [NORMAL_TEMPORARY_FAILURE]
2010-02-09 12:43:10.414137 [DEBUG] switch_channel.c:1994 Send signal 
sofia/external/5001 at testsrv.marketel [KILL]
2010-02-09 12:43:10.414137 [DEBUG] switch_core_state_machine.c:316 
(sofia/external/5001 at testsrv.marketel) Running State Change CS_HANGUP
2010-02-09 12:43:10.414137 [INFO] mod_dptools.c:2346 Originate Failed.  
Cause: NORMAL_TEMPORARY_FAILURE
2010-02-09 12:43:10.414137 [NOTICE] mod_dptools.c:2409 Hangup 
sofia/external/5000 at testsrv.marketel [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE]
2010-02-09 12:43:10.414137 [DEBUG] switch_channel.c:1994 Send signal 
sofia/external/5000 at testsrv.marketel [KILL]
2010-02-09 12:43:10.414137 [DEBUG] switch_core_session.c:1019 Send 
signal sofia/external/5000 at testsrv.marketel [BREAK]
2010-02-09 12:43:10.414137 [DEBUG] switch_core_state_machine.c:350 
(sofia/external/5000 at testsrv.marketel) State EXECUTE going to sleep
2010-02-09 12:43:10.414137 [DEBUG] switch_core_state_machine.c:496 
(sofia/external/5001 at testsrv.marketel) State HANGUP
2010-02-09 12:43:10.414137 [DEBUG] switch_core_state_machine.c:316 
(sofia/external/5000 at testsrv.marketel) Running State Change CS_HANGUP
2010-02-09 12:43:10.414137 [DEBUG] switch_core_session.c:1019 Send 
signal sofia/external/5001 at testsrv.marketel [BREAK]
2010-02-09 12:43:10.414137 [DEBUG] mod_sofia.c:352 
sofia/external/5001 at testsrv.marketel Overriding SIP cause 503 with 503 
from the other leg
2010-02-09 12:43:10.414137 [DEBUG] mod_sofia.c:358 Channel 
sofia/external/5001 at testsrv.marketel hanging up, cause: 
NORMAL_TEMPORARY_FAILURE
2010-02-09 12:43:10.414137 [DEBUG] switch_core_state_machine.c:46 
sofia/external/5001 at testsrv.marketel Standard HANGUP, cause: 
NORMAL_TEMPORARY_FAILURE
2010-02-09 12:43:10.414137 [DEBUG] switch_core_state_machine.c:496 
(sofia/external/5001 at testsrv.marketel) State HANGUP going to sleep
2010-02-09 12:43:10.414137 [DEBUG] switch_core_state_machine.c:335 
(sofia/external/5001 at testsrv.marketel) State Change CS_HANGUP -> 
CS_REPORTING
2010-02-09 12:43:10.414137 [DEBUG] switch_core_session.c:1019 Send 
signal sofia/external/5001 at testsrv.marketel [BREAK]
2010-02-09 12:43:10.414137 [DEBUG] switch_core_state_machine.c:316 
(sofia/external/5001 at testsrv.marketel) Running State Change CS_REPORTING
2010-02-09 12:43:10.414137 [DEBUG] switch_core_state_machine.c:587 
(sofia/external/5001 at testsrv.marketel) State REPORTING
2010-02-09 12:43:10.414137 [DEBUG] switch_core_state_machine.c:53 
sofia/external/5001 at testsrv.marketel Standard REPORTING, cause: 
NORMAL_TEMPORARY_FAILURE
2010-02-09 12:43:10.414137 [DEBUG] switch_core_state_machine.c:587 
(sofia/external/5001 at testsrv.marketel) State REPORTING going to sleep
2010-02-09 12:43:10.414137 [DEBUG] switch_core_state_machine.c:329 
(sofia/external/5001 at testsrv.marketel) State Change CS_REPORTING -> 
CS_DESTROY
2010-02-09 12:43:10.414137 [DEBUG] switch_core_session.c:1019 Send 
signal sofia/external/5001 at testsrv.marketel [BREAK]
2010-02-09 12:43:10.414137 [DEBUG] switch_core_session.c:1161 Session 18 
(sofia/external/5001 at testsrv.marketel) Locked, Waiting on external entities
2010-02-09 12:43:10.414137 [NOTICE] switch_core_session.c:1179 Session 
18 (sofia/external/5001 at testsrv.marketel) Ended
2010-02-09 12:43:10.414137 [NOTICE] switch_core_session.c:1181 Close 
Channel sofia/external/5001 at testsrv.marketel [CS_DESTROY]
2010-02-09 12:43:10.414137 [DEBUG] switch_core_state_machine.c:425 
(sofia/external/5001 at testsrv.marketel) Running State Change CS_DESTROY
2010-02-09 12:43:10.414137 [DEBUG] switch_core_state_machine.c:436 
(sofia/external/5001 at testsrv.marketel) State DESTROY
2010-02-09 12:43:10.414137 [DEBUG] mod_sofia.c:293 
sofia/external/5001 at testsrv.marketel SOFIA DESTROY
2010-02-09 12:43:10.414137 [DEBUG] switch_core_state_machine.c:60 
sofia/external/5001 at testsrv.marketel Standard DESTROY
2010-02-09 12:43:10.414137 [DEBUG] switch_core_state_machine.c:436 
(sofia/external/5001 at testsrv.marketel) State DESTROY going to sleep
2010-02-09 12:43:10.414137 [DEBUG] switch_core_state_machine.c:496 
(sofia/external/5000 at testsrv.marketel) State HANGUP
2010-02-09 12:43:10.414137 [DEBUG] mod_sofia.c:352 
sofia/external/5000 at testsrv.marketel Overriding SIP cause 503 with 503 
from the other leg
2010-02-09 12:43:10.414137 [DEBUG] mod_sofia.c:358 Channel 
sofia/external/5000 at testsrv.marketel hanging up, cause: 
NORMAL_TEMPORARY_FAILURE
2010-02-09 12:43:10.414137 [DEBUG] mod_sofia.c:424 Responding to INVITE 
with: 503
2010-02-09 12:43:10.414137 [DEBUG] switch_core_state_machine.c:46 
sofia/external/5000 at testsrv.marketel Standard HANGUP, cause: 
NORMAL_TEMPORARY_FAILURE
2010-02-09 12:43:10.414137 [DEBUG] switch_core_state_machine.c:496 
(sofia/external/5000 at testsrv.marketel) State HANGUP going to sleep
2010-02-09 12:43:10.414137 [DEBUG] switch_core_state_machine.c:335 
(sofia/external/5000 at testsrv.marketel) State Change CS_HANGUP -> 
CS_REPORTING
2010-02-09 12:43:10.414137 [DEBUG] switch_core_session.c:1019 Send 
signal sofia/external/5000 at testsrv.marketel [BREAK]
2010-02-09 12:43:10.414137 [DEBUG] switch_core_state_machine.c:316 
(sofia/external/5000 at testsrv.marketel) Running State Change CS_REPORTING
2010-02-09 12:43:10.414137 [DEBUG] switch_core_state_machine.c:587 
(sofia/external/5000 at testsrv.marketel) State REPORTING
2010-02-09 12:43:10.414137 [DEBUG] switch_core_state_machine.c:53 
sofia/external/5000 at testsrv.marketel Standard REPORTING, cause: 
NORMAL_TEMPORARY_FAILURE
2010-02-09 12:43:10.414137 [DEBUG] switch_core_state_machine.c:587 
(sofia/external/5000 at testsrv.marketel) State REPORTING going to sleep
2010-02-09 12:43:10.414137 [DEBUG] switch_core_state_machine.c:329 
(sofia/external/5000 at testsrv.marketel) State Change CS_REPORTING -> 
CS_DESTROY
2010-02-09 12:43:10.414137 [DEBUG] switch_core_session.c:1019 Send 
signal sofia/external/5000 at testsrv.marketel [BREAK]
2010-02-09 12:43:10.414137 [DEBUG] switch_core_session.c:1161 Session 17 
(sofia/external/5000 at testsrv.marketel) Locked, Waiting on external entities
2010-02-09 12:43:10.414137 [NOTICE] switch_core_session.c:1179 Session 
17 (sofia/external/5000 at testsrv.marketel) Ended
2010-02-09 12:43:10.414137 [NOTICE] switch_core_session.c:1181 Close 
Channel sofia/external/5000 at testsrv.marketel [CS_DESTROY]
2010-02-09 12:43:10.414137 [DEBUG] switch_core_state_machine.c:425 
(sofia/external/5000 at testsrv.marketel) Running State Change CS_DESTROY
2010-02-09 12:43:10.414137 [DEBUG] switch_core_state_machine.c:436 
(sofia/external/5000 at testsrv.marketel) State DESTROY
2010-02-09 12:43:10.414137 [DEBUG] mod_sofia.c:293 
sofia/external/5000 at testsrv.marketel SOFIA DESTROY
2010-02-09 12:43:10.414137 [DEBUG] switch_core_state_machine.c:60 
sofia/external/5000 at testsrv.marketel Standard DESTROY
2010-02-09 12:43:10.414137 [DEBUG] switch_core_state_machine.c:436 
(sofia/external/5000 at testsrv.marketel) State DESTROY going to sleep


-- 
Mark Sobkow
Senior Developer
MarkeTel Multi-Line Dialing Systems LTD.
428 Victoria Ave
Regina, SK S4N-0P6
Toll-Free: 800-289-8616-X533
Local: 306-359-6893-X533
Fax: 306-359-6879
Email: m.sobkow at marketelsystems.com
Web: http://www.marketelsystems.com 




From jerry.richards at teotech.com  Tue Feb  9 11:15:26 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Tue, 9 Feb 2010 11:15:26 -0800
Subject: [Freeswitch-users] Deleting Single Registered Device in the FS
	Console
Message-ID: <326DBA7172E24B42A1B466E7BAC0B562@greyhawk.tonecommander.com>

How do I delete a single registered user at the FS Console?  I know how to
flush_inbound_reg (which is all user) using "sofia profile internal
flush_inbound_reg".

Best Regards,
Jerry




From jerry.richards at teotech.com  Tue Feb  9 11:19:50 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Tue, 9 Feb 2010 11:19:50 -0800
Subject: [Freeswitch-users] Deleting Single Registered Device in the FS
	Console
Message-ID: <7377B7F5DEF54DB8843C47A7EBC8888F@greyhawk.tonecommander.com>

Actually I found out how to do it using the Call-ID (which is very long),
but can't I do it using the user's extension number?

Best Regards,
Jerry


-----Original Message-----
From: Jerry Richards [mailto:jerry.richards at teotech.com] 
Sent: Tuesday, February 09, 2010 11:15 AM
To: 'freeswitch-users at lists.freeswitch.org'
Subject: Deleting Single Registered Device in the FS Console

How do I delete a single registered user at the FS Console?  I know how to
flush_inbound_reg (which is all user) using "sofia profile internal
flush_inbound_reg".

Best Regards,
Jerry




From brian at freeswitch.org  Tue Feb  9 11:21:12 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 9 Feb 2010 13:21:12 -0600
Subject: [Freeswitch-users] Deleting Single Registered Device in the FS
	Console
In-Reply-To: <326DBA7172E24B42A1B466E7BAC0B562@greyhawk.tonecommander.com>
References: <326DBA7172E24B42A1B466E7BAC0B562@greyhawk.tonecommander.com>
Message-ID: <B1554204-BDAA-44D4-AD40-3E4FD3CBA04F@freeswitch.org>

Read the sofia command... you can add the call-id of the registration to flush the single one.

/b

On Feb 9, 2010, at 1:15 PM, Jerry Richards wrote:

> How do I delete a single registered user at the FS Console?  I know how to
> flush_inbound_reg (which is all user) using "sofia profile internal
> flush_inbound_reg".
> 
> Best Regards,
> Jerry




From brian at freeswitch.org  Tue Feb  9 11:23:35 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 9 Feb 2010 13:23:35 -0600
Subject: [Freeswitch-users] Deleting Single Registered Device in the FS
	Console
In-Reply-To: <7377B7F5DEF54DB8843C47A7EBC8888F@greyhawk.tonecommander.com>
References: <7377B7F5DEF54DB8843C47A7EBC8888F@greyhawk.tonecommander.com>
Message-ID: <D2E44884-0DB3-4CE8-BAD0-68D88DC297D4@freeswitch.org>

what if they have 10 phones registered?  That kinda goes beyond a single registration record.

/b

On Feb 9, 2010, at 1:19 PM, Jerry Richards wrote:

> Actually I found out how to do it using the Call-ID (which is very long),
> but can't I do it using the user's extension number?
> 
> Best Regards,
> Jerry




From andrew at hijacked.us  Tue Feb  9 11:25:37 2010
From: andrew at hijacked.us (Andrew Thompson)
Date: Tue, 9 Feb 2010 14:25:37 -0500
Subject: [Freeswitch-users] Having trouble establishing a call
In-Reply-To: <4B71AD9E.7090606@marketelsystems.com>
References: <4B71AD9E.7090606@marketelsystems.com>
Message-ID: <20100209192537.GA24616@hijacked.us>

On Tue, Feb 09, 2010 at 12:46:54PM -0600, Mark Sobkow wrote:
> We're using Erlang to serve up the configurations to Freeswitch.  I've 
> got things configured such that I can place a call from a SIP phone 
> registered to extension 5000 to our "external" SIP provider (our 
> Asterisk installation), but I can't place a call to extension 5001 from 
> 5000.  Below is the trace log Freeswitch produces when I attempt to do so.
> 
> Any suggestions as to what I should be looking at?  The directory seems 
> to be getting served up correctly, as it provides the passwords both SIP 
> softphones are using to register with Freeswitch.  I'd have thought that 
> once they've registered with FS, the extension would automatically be 
> recognized when an incoming call is placed or bridged, but such does not 
> seem to be the case.
>

Looks like the other sip box (sofia/external/5001 at testsrv.marketel) is
returning the failure code. Check the config on the other side? Don't
you have a gateway setup for this other machine so you couls do
sofia/gateway/testsrv.marketel/5001 instead?

Andrew



From robert.hadley at teotech.com  Tue Feb  9 11:28:13 2010
From: robert.hadley at teotech.com (Robert Hadley)
Date: Tue, 9 Feb 2010 11:28:13 -0800
Subject: [Freeswitch-users] Where are new sounds_dir and recordings_dir
	variables set?
Message-ID: <28788FA27BE74B9FA6BFFE52BED11B60@greyhawk.tonecommander.com>

 

The XML conf files have been recently modified to replace
"$${base_dir}/sounds" with $${sounds_dir}.  The same replacement was done
for $${base_dir}/recordings and $${recordings_dir}.  In vars.xml, the
X-PRE-PROCESS cmd to setting the old variable sound_prefix was removed but
set commands were not added for the new variables sounds_dir and
recordings_dir.  How is FS finding determining the value of $${sounds_dir}
and $${recordings_dir} in the XML files?

 

I ask because I have 2 cloned FS servers where one can find sound files in
the IVR when an invalid extension is entered, but the other does not and is
using the wrong path to search for sound files.

 

Server 2 Fails:

2010-02-09 11:11:16.735745 [DEBUG] switch_ivr_menu.c:381 digits '9999'

2010-02-09 11:11:16.735745 [DEBUG] switch_ivr_menu.c:475 action regex [9999]
[/^((5[34][8901][0-9])||(10[0-5][0-9])||(30\d{2}))$/] [0]

2010-02-09 11:11:16.735745 [DEBUG] switch_ivr_menu.c:565 IVR menu 'teo_ivr'
caught invalid input '9999'

2010-02-09 11:11:16.735745 [ERR] mod_sndfile.c:194 Error Opening File
[/opt/teoswitch/sounds/ivr/ivr-that_was_an_invalid_entry.wav] [System error
: No such file or directory.]

 

Server 1 Works:

2010-02-09 11:17:25.085660 [DEBUG] switch_ivr_menu.c:378 digits '9999'

2010-02-09 11:17:25.085660 [DEBUG] switch_ivr_menu.c:472 action regex [9999]
[/^((5[34][8901][0-9])||(10[0-5][0-9])||(30\d{2}))$/] [0]

2010-02-09 11:17:25.085660 [DEBUG] switch_ivr_menu.c:562 IVR menu 'teo_ivr'
caught invalid input '9999'

2010-02-09 11:17:25.085660 [DEBUG] switch_ivr_play_say.c:1162 Codec
Activated L16 at 8000hz 1 channels 20ms

2010-02-09 11:17:25.085660 [DEBUG] switch_core_io.c:652
sofia/internal/1045 at 192.168.72.141:5060 receive message
[TRANSCODING_NECESSARY]

2010-02-09 11:17:26.765470 [DEBUG] switch_ivr_play_say.c:1454 done playing
file

 

I have compared the conf folders and they are nearly identical.  The only
difference I know of is Server 2 was originally compiled with mod_flite
enabled but it is not loaded at runtime.  Any suggestions why one server can
find sound files but the other looks in the wrong path?

 

Thanks,

Robert

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100209/a1c12974/attachment-0002.html 

From brian at freeswitch.org  Tue Feb  9 11:33:53 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 9 Feb 2010 13:33:53 -0600
Subject: [Freeswitch-users] Where are new sounds_dir and recordings_dir
	variables set?
In-Reply-To: <28788FA27BE74B9FA6BFFE52BED11B60@greyhawk.tonecommander.com>
References: <28788FA27BE74B9FA6BFFE52BED11B60@greyhawk.tonecommander.com>
Message-ID: <6B8839F5-DA53-4565-A7E0-876FC53BD6F7@freeswitch.org>

did you configure it with --prefix=/opt/teoswitch or did you move it?

/b

On Feb 9, 2010, at 1:28 PM, Robert Hadley wrote:

>  
> The XML conf files have been recently modified to replace ?$${base_dir}/sounds? with $${sounds_dir}.  The same replacement was done for $${base_dir}/recordings and $${recordings_dir}.  In vars.xml, the X-PRE-PROCESS cmd to setting the old variable sound_prefix was removed but set commands were not added for the new variables sounds_dir and recordings_dir.  How is FS finding determining the value of $${sounds_dir} and $${recordings_dir} in the XML files?
>  
> I ask because I have 2 cloned FS servers where one can find sound files in the IVR when an invalid extension is entered, but the other does not and is using the wrong path to search for sound files.
>  
> Server 2 Fails:
> 2010-02-09 11:11:16.735745 [DEBUG] switch_ivr_menu.c:381 digits '9999'
> 2010-02-09 11:11:16.735745 [DEBUG] switch_ivr_menu.c:475 action regex [9999] [/^((5[34][8901][0-9])||(10[0-5][0-9])||(30\d{2}))$/] [0]
> 2010-02-09 11:11:16.735745 [DEBUG] switch_ivr_menu.c:565 IVR menu 'teo_ivr' caught invalid input '9999'
> 2010-02-09 11:11:16.735745 [ERR] mod_sndfile.c:194 Error Opening File [/opt/teoswitch/sounds/ivr/ivr-that_was_an_invalid_entry.wav] [System error : No such file or directory.]
>  
> Server 1 Works:
> 2010-02-09 11:17:25.085660 [DEBUG] switch_ivr_menu.c:378 digits '9999'
> 2010-02-09 11:17:25.085660 [DEBUG] switch_ivr_menu.c:472 action regex [9999] [/^((5[34][8901][0-9])||(10[0-5][0-9])||(30\d{2}))$/] [0]
> 2010-02-09 11:17:25.085660 [DEBUG] switch_ivr_menu.c:562 IVR menu 'teo_ivr' caught invalid input '9999'
> 2010-02-09 11:17:25.085660 [DEBUG] switch_ivr_play_say.c:1162 Codec Activated L16 at 8000hz 1 channels 20ms
> 2010-02-09 11:17:25.085660 [DEBUG] switch_core_io.c:652 sofia/internal/1045 at 192.168.72.141:5060 receive message [TRANSCODING_NECESSARY]
> 2010-02-09 11:17:26.765470 [DEBUG] switch_ivr_play_say.c:1454 done playing file
>  
> I have compared the conf folders and they are nearly identical.  The only difference I know of is Server 2 was originally compiled with mod_flite enabled but it is not loaded at runtime.  Any suggestions why one server can find sound files but the other looks in the wrong path?
>  
> Thanks,
> Robert




From m.sobkow at marketelsystems.com  Tue Feb  9 11:54:04 2010
From: m.sobkow at marketelsystems.com (Mark Sobkow)
Date: Tue, 09 Feb 2010 13:54:04 -0600
Subject: [Freeswitch-users] Having trouble establishing a call
In-Reply-To: <20100209192537.GA24616@hijacked.us>
References: <4B71AD9E.7090606@marketelsystems.com>
	<20100209192537.GA24616@hijacked.us>
Message-ID: <4B71BD5C.1080201@marketelsystems.com>

Andrew Thompson wrote:
> On Tue, Feb 09, 2010 at 12:46:54PM -0600, Mark Sobkow wrote:
>   
>> We're using Erlang to serve up the configurations to Freeswitch.  I've 
>> got things configured such that I can place a call from a SIP phone 
>> registered to extension 5000 to our "external" SIP provider (our 
>> Asterisk installation), but I can't place a call to extension 5001 from 
>> 5000.  Below is the trace log Freeswitch produces when I attempt to do so.
>>
>> Any suggestions as to what I should be looking at?  The directory seems 
>> to be getting served up correctly, as it provides the passwords both SIP 
>> softphones are using to register with Freeswitch.  I'd have thought that 
>> once they've registered with FS, the extension would automatically be 
>> recognized when an incoming call is placed or bridged, but such does not 
>> seem to be the case.
>>
>>     
>
> Looks like the other sip box (sofia/external/5001 at testsrv.marketel) is
> returning the failure code. Check the config on the other side? Don't
> you have a gateway setup for this other machine so you couls do
> sofia/gateway/testsrv.marketel/5001 instead?
>
> Andrew
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>   

Actually rats.marketel is our Asterisk box, which acts as our SIP 
trunk.  testsrv.marketel is the one running Freeswitch.

Am I maybe using an incorrect syntax in the dialplan for specifying that 
the call should be routed to a local extension on the Freeswitch box?  I 
thought the sofia/<profile>/<number>@<server> syntax is supposed to be 
used for placing any SIP calls, not just "remote" ones.



From m.sobkow at marketelsystems.com  Tue Feb  9 12:26:09 2010
From: m.sobkow at marketelsystems.com (Mark Sobkow)
Date: Tue, 09 Feb 2010 14:26:09 -0600
Subject: [Freeswitch-users] Having trouble establishing a call
In-Reply-To: <20100209192537.GA24616@hijacked.us>
References: <4B71AD9E.7090606@marketelsystems.com>
	<20100209192537.GA24616@hijacked.us>
Message-ID: <4B71C4E1.705@marketelsystems.com>

Andrew Thompson wrote:
> On Tue, Feb 09, 2010 at 12:46:54PM -0600, Mark Sobkow wrote:
>   
>> We're using Erlang to serve up the configurations to Freeswitch.  I've 
>> got things configured such that I can place a call from a SIP phone 
>> registered to extension 5000 to our "external" SIP provider (our 
>> Asterisk installation), but I can't place a call to extension 5001 from 
>> 5000.  Below is the trace log Freeswitch produces when I attempt to do so.
>>
>> Any suggestions as to what I should be looking at?  The directory seems 
>> to be getting served up correctly, as it provides the passwords both SIP 
>> softphones are using to register with Freeswitch.  I'd have thought that 
>> once they've registered with FS, the extension would automatically be 
>> recognized when an incoming call is placed or bridged, but such does not 
>> seem to be the case.
>>
>>     
>
> Looks like the other sip box (sofia/external/5001 at testsrv.marketel) is
> returning the failure code. Check the config on the other side? Don't
> you have a gateway setup for this other machine so you couls do
> sofia/gateway/testsrv.marketel/5001 instead?
>
> Andrew
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>   
After about 6 hours of debugging and digging, I finally got the 
Freeswitch installation to dial extensions attached/registered to it.

Buried away in the Freeswitch wikis are 2-3 lines of example for 
mod_sofia that show using a % to separate the extension/number and the 
server name instead of the @ sign that's using in 99% of the 
documentation.  The % syntax means "local".

*sigh*

Now I can get on with working on the conference automated dialing code 
that I was originally trying to prototype through the command line.  
Those commands just weren't working 'cause I was using the @ syntax so 
it was attempting to find a remote server named testsrv.marketel instead 
of routing the call through the local registration list.

-- 
Mark Sobkow
Senior Developer
MarkeTel Multi-Line Dialing Systems LTD.
428 Victoria Ave
Regina, SK S4N-0P6
Toll-Free: 800-289-8616-X533
Local: 306-359-6893-X533
Fax: 306-359-6879
Email: m.sobkow at marketelsystems.com
Web: http://www.marketelsystems.com 




From jerry.richards at teotech.com  Tue Feb  9 12:26:57 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Tue, 9 Feb 2010 12:26:57 -0800
Subject: [Freeswitch-users] Any Known Dual-Registration Issue?
Message-ID: <5AD33F8B1F0848CF9C3A314A60039366@greyhawk.tonecommander.com>

I've noticed that sometimes my phones end up with two registrations with two
Call-IDs at Freeswitch.  Is there any known bug that would cause this?  I've
seen it on different phone models, so I'm thinking there is some timing
issue with Freeswitch.

Best Regards,
Jerry




From Prometheus001 at gmx.net  Tue Feb  9 12:27:40 2010
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Tue, 09 Feb 2010 21:27:40 +0100
Subject: [Freeswitch-users] mod_skypiax - help needed with
 modified	snd-dummy (one way audio)
In-Reply-To: <4B716C90.8070109@gmx.net>
References: <4B60555B.2020004@gmx.net>	<7b197bef1002051319l2fcb5617p1691692f58c65372@mail.gmail.com>	<4B6CC792.5060608@gmx.net>	<191c3a031002060558r36947d93vcbb46df9e0a7dfde@mail.gmail.com>	<4B6F3E9A.2020103@gmx.net>	<7b197bef1002080529u1afc7f9av492b0c221ee2d6af@mail.gmail.com>	<c55d46951002080546p829a881w256bfdb19334c548@mail.gmail.com>	<7b197bef1002080553qd1cb42dla26425942de83c43@mail.gmail.com>	<c55d46951002080604k400344b0i8add00b5c6014958@mail.gmail.com>	<4B703AD0.2080909@gmx.net>	<7b197bef1002080839o61b4b1b5k6d5cc39bb05d9a32@mail.gmail.com>
	<4B716C90.8070109@gmx.net>
Message-ID: <4B71C53C.2020202@gmx.net>

Hello,

thanks to Giovanni's help we solved the problem.
It was an incompatiblity with the current Skype 2.1 Beta with Ubuntu Hardy.
The version skype_static-2.0.0.72 works with good sound quality in both
directions.

Best regards
Peter



Peter P GMX schrieb:
> Hello Giovanni,
>
> I will try to contact you via IRC (stony)
>
> Best regards
> Peter
>
> Giovanni Maruzzelli schrieb:
>   
>> Peter,
>>
>> excuse me but I really do not follow you.
>>
>> Why you have the normal static build not working?
>>
>> Also, this is really taking too much of my time. You continue to
>> change things, and report issues, without waiting for solutions you
>> ask for, then you report something else, and so on... we'll never get
>> at the end of this.
>>
>> If you want, please connect via IRC, and contact me (gmaruzz).
>> Or let me connect ssh to your machine.
>>
>> -gm
>>
>>
>> On Mon, Feb 8, 2010 at 5:24 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>   
>>     
>>> I got it working now with static build and an older version of skype
>>> (skype_static-2.1.0.47).
>>>
>>> But I still have a problem ongoing with sound quality, resp. one way audio:
>>> With original Alsa driver: When Skype calls mod_skypiax => SIP Phone,
>>> then the sound from the SIP phone is interupted regularly 2 times a second.
>>> Example:
>>> When a person on the sip phone speaks
>>> "aaaaaaaaaaaaaaaaaaaaaa"
>>> the other site hears
>>> "aaatataaaatataaaatataaaa"
>>> With "t" meaning the interruption of the sound.
>>>
>>> So I compliled and installed the modified alsa driver as described in
>>> the wiki (configure, make and make install, remove old ubuntu sound dir
>>> in /lib/modules/2.6.24-24-server/ubuntu/sound + depmod -a + reboot the
>>> server.
>>>
>>> Now the SIP phone is heard loud and clearly without interruption.
>>> However the other direction is not heard, so we're at the beginning of
>>> the post (one way audio). Only when I really scratch the microphone then
>>> I hear some parts of this scratching on the SIP side.
>>>
>>> So, some more hints are needed.
>>>
>>> Here's the log, when I start the skype client:
>>>  su root -c "/bin/echo 'username password'| DISPLAY=:101
>>> /usr/bin/skype1 --pipelogin &" &
>>>  /usr/bin/Xvfb :102 -ac &
>>>  error opening security policy file /etc/X11/xserver/SecurityPolicy
>>>  expected keysym, got XF86KbdLightOnOff: line 70 of pc
>>>  expected keysym, got XF86KbdBrightnessDown: line 71 of pc
>>>  expected keysym, got XF86KbdBrightnessUp: line 72 of pc
>>>  Could not init font path element /usr/share/fonts/X11/cyrillic,
>>> removing from list!
>>>  Could not init font path element
>>> /var/lib/defoma/x-ttcidfont-conf.d/dirs/TrueType, removing from list!
>>> But these messages are not critical, right?
>>>
>>> Best regards
>>> Peter
>>>
>>>
>>> Max Bridgewater schrieb:
>>>     
>>>       
>>>> Well, it was a couple of weeks ago. I then switched back to 2.0.4. I'm
>>>> going to try it again and let you know.
>>>>
>>>> Max.
>>>>
>>>> On Mon, Feb 8, 2010 at 8:53 AM, Giovanni Maruzzelli <gmaruzz at celliax.org> wrote:
>>>>
>>>>       
>>>>         
>>>>> Peter is using hardy 64 bit. I checked on that.
>>>>>
>>>>> But, let me understand: if you're using a static build, why you have a
>>>>> problem with QT?
>>>>> Is actually Qt to be statically linked...
>>>>>
>>>>> what is the results of:
>>>>>
>>>>> ldd skype
>>>>>
>>>>> Anyway, in a short while I'll check that on centos 5.4 too, and I'll report here
>>>>>
>>>>> -giovanni
>>>>>
>>>>> On Mon, Feb 8, 2010 at 2:46 PM, Max Bridgewater
>>>>> <max.bridgewater at gmail.com> wrote:
>>>>>
>>>>>         
>>>>>           
>>>>>> Interesting; a while back I tried to install Skypiax with the latest
>>>>>> static build on Skype.com. I had QT library compatibility problem on a
>>>>>> CentOS 5.4 (Final). Giovannni, may I ask what Linux distro you are
>>>>>> using?
>>>>>>
>>>>>> Thanks,
>>>>>> max.
>>>>>>
>>>>>> On Mon, Feb 8, 2010 at 8:29 AM, Giovanni Maruzzelli <gmaruzz at celliax.org> wrote:
>>>>>>
>>>>>>           
>>>>>>             
>>>>>>> Peter,
>>>>>>>
>>>>>>> I just tested with the static build you find on skype.com
>>>>>>>
>>>>>>> I never tested for performances or other issues, there may be (it's a beta).
>>>>>>>
>>>>>>> But it do not crash on me.
>>>>>>>
>>>>>>> I have no problem at all.
>>>>>>>
>>>>>>> If you can give me ssh access I can try to understand why you have so
>>>>>>> many problems.
>>>>>>>
>>>>>>> Or, alternatively, try to follow the wiki. You know, I've not heard
>>>>>>> about those problems.
>>>>>>>
>>>>>>> root at hardy64:~/skype.beta/skype_static-2.1.0.81# ldd skype
>>>>>>>        linux-gate.so.1 =>  (0xffffe000)
>>>>>>>        libasound.so.2 => /usr/lib32/libasound.so.2 (0xf7eb7000)
>>>>>>>        libXv.so.1 => /usr/lib32/libXv.so.1 (0xf7eb2000)
>>>>>>>        libXss.so.1 => /usr/lib32/libXss.so.1 (0xf7eae000)
>>>>>>>        libSM.so.6 => /usr/lib32/libSM.so.6 (0xf7ea6000)
>>>>>>>        libICE.so.6 => /usr/lib32/libICE.so.6 (0xf7e8e000)
>>>>>>>        libXi.so.6 => /usr/lib32/libXi.so.6 (0xf7e86000)
>>>>>>>        libXrender.so.1 => /usr/lib32/libXrender.so.1 (0xf7e7e000)
>>>>>>>        libXrandr.so.2 => /usr/lib32/libXrandr.so.2 (0xf7e78000)
>>>>>>>        libfreetype.so.6 => /usr/lib32/libfreetype.so.6 (0xf7e0a000)
>>>>>>>        libfontconfig.so.1 => /usr/lib32/libfontconfig.so.1 (0xf7de0000)
>>>>>>>        libXext.so.6 => /usr/lib32/libXext.so.6 (0xf7dd2000)
>>>>>>>        libX11.so.6 => /usr/lib32/libX11.so.6 (0xf7ceb000)
>>>>>>>        libz.so.1 => /usr/lib32/libz.so.1 (0xf7cd6000)
>>>>>>>        libgthread-2.0.so.0 => /usr/lib32/libgthread-2.0.so.0 (0xf7cd1000)
>>>>>>>        libglib-2.0.so.0 => /usr/lib32/libglib-2.0.so.0 (0xf7c1f000)
>>>>>>>        librt.so.1 => /lib32/librt.so.1 (0xf7c16000)
>>>>>>>        libdl.so.2 => /lib32/libdl.so.2 (0xf7c12000)
>>>>>>>        libpthread.so.0 => /lib32/libpthread.so.0 (0xf7bfa000)
>>>>>>>        libstdc++.so.6 => /usr/lib32/libstdc++.so.6 (0xf7b07000)
>>>>>>>        libm.so.6 => /lib32/libm.so.6 (0xf7ae2000)
>>>>>>>        libgcc_s.so.1 => /usr/lib32/libgcc_s.so.1 (0xf7ad6000)
>>>>>>>        libc.so.6 => /lib32/libc.so.6 (0xf7987000)
>>>>>>>        libexpat.so.1 => /usr/lib32/libexpat.so.1 (0xf7966000)
>>>>>>>        libXau.so.6 => /usr/lib32/libXau.so.6 (0xf7963000)
>>>>>>>        libxcb-xlib.so.0 => /usr/lib32/libxcb-xlib.so.0 (0xf7961000)
>>>>>>>        libxcb.so.1 => /usr/lib32/libxcb.so.1 (0xf7948000)
>>>>>>>        libpcre.so.3 => /usr/lib32/libpcre.so.3 (0xf7921000)
>>>>>>>        /lib/ld-linux.so.2 (0xf7f86000)
>>>>>>>        libXdmcp.so.6 => /usr/lib32/libXdmcp.so.6 (0xf791c000)
>>>>>>>
>>>>>>>
>>>>>>> On Sun, Feb 7, 2010 at 11:28 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>>>
>>>>>>>             
>>>>>>>               
>>>>>>>> I now used the static Skype binary in order to avoid missing constraints
>>>>>>>> to other libraries: It still crashes
>>>>>>>> 1st it starts with:
>>>>>>>>  process 15431: D-Bus library appears to be incorrectly set up; failed
>>>>>>>> to read machine uuid: Failed to open "/var/lib/dbus/machine-id": No such
>>>>>>>> file or directory
>>>>>>>>  See the manual page for dbus-uuidgen to correct this issue.
>>>>>>>> After calling this client it crashes with:
>>>>>>>>   /usr/bin/skype: symbol lookup error: /usr/bin/skype: undefined
>>>>>>>> symbol: _ZN10QBoxLayout13addSpacerItemEP11QSpacerItem
>>>>>>>>
>>>>>>>> Any hints, where I may get an older Skype client?
>>>>>>>>
>>>>>>>> Best regards
>>>>>>>> Peter
>>>>>>>>
>>>>>>>> Anthony Minessale schrieb:
>>>>>>>>
>>>>>>>>               
>>>>>>>>                 
>>>>>>>>> Quick fyi centos 5.4 has some memory bugs in libc don't go beyond 5.3
>>>>>>>>> until its fixed.
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>                 
>>>>>>>>>                   
>>>>>>>>>> On Feb 5, 2010 7:42 PM, "Peter P GMX" <Prometheus001 at gmx.net
>>>>>>>>>> <mailto:Prometheus001 at gmx.net>> wrote:
>>>>>>>>>>
>>>>>>>>>> Skype starts, but as soon as it receives a call it crashes with:
>>>>>>>>>>
>>>>>>>>>> /usr/bin/skype: symbol lookup error: /usr/bin/skype: undefined symbol:
>>>>>>>>>>
>>>>>>>>>> _ZN10QBoxLayout13addSpacerItemEP11QSpacerItem
>>>>>>>>>>
>>>>>>>>>> I think the 8.10 version dos not work with8.04.
>>>>>>>>>>
>>>>>>>>>> Any hints, where I may get an older Skype client? I may also try the
>>>>>>>>>> static skype client.
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> Best regards
>>>>>>>>>> Peter
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> .
>>>>>>>>>> Giovanni Maruzzelli schrieb:
>>>>>>>>>>
>>>>>>>>>>                   
>>>>>>>>>>                     
>>>>>>>>>>> that's not at all a fatal error.
>>>>>>>>>>> I believe...
>>>>>>>>>>>
>>>>>>>>>>>                     
>>>>>>>>>>>                       
>>>>>>>>> ------------------------------------------------------------------------
>>>>>>>>>
>>>>>>>>> _______________________________________________
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>                 
>>>>>>>>>                   
>>>>>>>> _______________________________________________
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>>               
>>>>>>>>                 
>>>>>>> --
>>>>>>> Sincerely,
>>>>>>>
>>>>>>> Giovanni Maruzzelli
>>>>>>> Cell : +39-347-2665618
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>>             
>>>>>>>               
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>>           
>>>>>>             
>>>>> --
>>>>> Sincerely,
>>>>>
>>>>> Giovanni Maruzzelli
>>>>> Cell : +39-347-2665618
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>>         
>>>>>           
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>       
>>>>         
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>     
>>>       
>>
>>   
>>     
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>   



From brian at freeswitch.org  Tue Feb  9 12:36:19 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 9 Feb 2010 14:36:19 -0600
Subject: [Freeswitch-users] Any Known Dual-Registration Issue?
In-Reply-To: <5AD33F8B1F0848CF9C3A314A60039366@greyhawk.tonecommander.com>
References: <5AD33F8B1F0848CF9C3A314A60039366@greyhawk.tonecommander.com>
Message-ID: <0FF21E6C-39DE-4973-8790-9E02C5ED8BDC@freeswitch.org>

The phone is at fault.  It prob. uses a different call-id for each new reg... and or you restart the phone without it unregistering and you'll end up with TWO.

/b

On Feb 9, 2010, at 2:26 PM, Jerry Richards wrote:

> I've noticed that sometimes my phones end up with two registrations with two
> Call-IDs at Freeswitch.  Is there any known bug that would cause this?  I've
> seen it on different phone models, so I'm thinking there is some timing
> issue with Freeswitch.
> 
> Best Regards,
> Jerry




From brian at freeswitch.org  Tue Feb  9 12:36:52 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 9 Feb 2010 14:36:52 -0600
Subject: [Freeswitch-users] Having trouble establishing a call
In-Reply-To: <4B71C4E1.705@marketelsystems.com>
References: <4B71AD9E.7090606@marketelsystems.com>
	<20100209192537.GA24616@hijacked.us>
	<4B71C4E1.705@marketelsystems.com>
Message-ID: <FB6BCA40-7B12-4BF1-9A2D-892A4FDDA909@freeswitch.org>

Question 2 on the FAQ

http://wiki.freeswitch.org/wiki/FreeSwitch_FAQ#Q:_What_is_the_difference_between_using_a_.25_and_.40_in_a_sofia_dial_string.3F

/b

On Feb 9, 2010, at 2:26 PM, Mark Sobkow wrote:

> Andrew Thompson wrote:
>> On Tue, Feb 09, 2010 at 12:46:54PM -0600, Mark Sobkow wrote:
>> 
>>> We're using Erlang to serve up the configurations to Freeswitch.  I've 
>>> got things configured such that I can place a call from a SIP phone 
>>> registered to extension 5000 to our "external" SIP provider (our 
>>> Asterisk installation), but I can't place a call to extension 5001 from 
>>> 5000.  Below is the trace log Freeswitch produces when I attempt to do so.
>>> 
>>> Any suggestions as to what I should be looking at?  The directory seems 
>>> to be getting served up correctly, as it provides the passwords both SIP 
>>> softphones are using to register with Freeswitch.  I'd have thought that 
>>> once they've registered with FS, the extension would automatically be 
>>> recognized when an incoming call is placed or bridged, but such does not 
>>> seem to be the case.
>>> 
>>> 
>> 
>> Looks like the other sip box (sofia/external/5001 at testsrv.marketel) is
>> returning the failure code. Check the config on the other side? Don't
>> you have a gateway setup for this other machine so you couls do
>> sofia/gateway/testsrv.marketel/5001 instead?
>> 
>> Andrew
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
> After about 6 hours of debugging and digging, I finally got the 
> Freeswitch installation to dial extensions attached/registered to it.
> 
> Buried away in the Freeswitch wikis are 2-3 lines of example for 
> mod_sofia that show using a % to separate the extension/number and the 
> server name instead of the @ sign that's using in 99% of the 
> documentation.  The % syntax means "local".
> 
> *sigh*
> 
> Now I can get on with working on the conference automated dialing code 
> that I was originally trying to prototype through the command line.  
> Those commands just weren't working 'cause I was using the @ syntax so 
> it was attempting to find a remote server named testsrv.marketel instead 
> of routing the call through the local registration list.
> 
> -- 
> Mark Sobkow
> Senior Developer
> MarkeTel Multi-Line Dialing Systems LTD.
> 428 Victoria Ave
> Regina, SK S4N-0P6
> Toll-Free: 800-289-8616-X533
> Local: 306-359-6893-X533
> Fax: 306-359-6879
> Email: m.sobkow at marketelsystems.com
> Web: http://www.marketelsystems.com 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From peder at networkoblivion.com  Tue Feb  9 12:40:55 2010
From: peder at networkoblivion.com (Peder)
Date: Tue, 9 Feb 2010 14:40:55 -0600
Subject: [Freeswitch-users] Any Known Dual-Registration Issue?
In-Reply-To: <5AD33F8B1F0848CF9C3A314A60039366@greyhawk.tonecommander.com>
References: <5AD33F8B1F0848CF9C3A314A60039366@greyhawk.tonecommander.com>
Message-ID: <035001caa9c8$2dca81c0$895f8540$@com>

What kind of phones?  If you have multiple registartion, this can happen
sometimes if you reboot a phone.  Crappy phones, like Grandstream, don't
un-register when you reboot and then when they come back up, they register
again and thus two registrations until the lifetime of the registration ends
and it gets flushed.  Changing the multiple-registration to contact can help
as I believe that uses port and source IP as part of the registration info:

<param name="multiple-registrations" value="contact"/>



-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jerry
Richards
Sent: Tuesday, February 09, 2010 2:27 PM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Any Known Dual-Registration Issue?

I've noticed that sometimes my phones end up with two registrations with two
Call-IDs at Freeswitch.  Is there any known bug that would cause this?  I've
seen it on different phone models, so I'm thinking there is some timing
issue with Freeswitch.

Best Regards,
Jerry


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




From w8hdkim at gmail.com  Tue Feb  9 06:52:51 2010
From: w8hdkim at gmail.com (Kim Culhan)
Date: Tue, 9 Feb 2010 09:52:51 -0500
Subject: [Freeswitch-users] UPnP Timeout
Message-ID: <89dbfdc31002090652h884bc2av82d53c6958fdc409@mail.gmail.com>

On Mon, February 8, 2010 11:21 pm, Troy Anderson wrote:
> I have been using pfSense (1.2.3) and FS.  FS nicely uses UPnP to poke
> holes in the firewall, but it seems that the holes close after a while.  I
> cannot find any documentation in FS nor in pfSense as to what the timeout
> is.  Is there a setting in FS to do some kind of keep-alive thing with
> UPnP to keep, e.g. 5060, open?  Or is it already doing that and pfSense is
> the issue?

FS has provisions for keep-alive, see the bottom of the page for ping
time value:

http://wiki.freeswitch.org/wiki/SIP_Provider_Examples

To watch the pf firewall hole timing you can install pftop from
FreeBSD ports/sysutils
which displays the filter states 'and more'.

-kim



From w8hdkim at gmail.com  Tue Feb  9 08:22:05 2010
From: w8hdkim at gmail.com (Kim Culhan)
Date: Tue, 9 Feb 2010 11:22:05 -0500
Subject: [Freeswitch-users] UPnP Timeout
Message-ID: <89dbfdc31002090822r3f4fd352ncbeb90955b5fbc14@mail.gmail.com>

On Mon, February 8, 2010 11:21 pm, Troy Anderson wrote:
> I have been using pfSense (1.2.3) and FS.  FS nicely uses UPnP to poke
> holes in the firewall, but it seems that the holes close after a while.  I
> cannot find any documentation in FS nor in pfSense as to what the timeout
> is.  Is there a setting in FS to do some kind of keep-alive thing with
> UPnP to keep, e.g. 5060, open?  Or is it already doing that and pfSense is
> the issue?

FS has provisions for keep-alive, see the bottom of the page for ping
time value:

http://wiki.freeswitch.org/wiki/SIP_Provider_Examples

To watch the pf firewall hole timing you can install pftop from
FreeBSD ports/sysutils
which displays the filter states 'and more'.

-kim



From sergey.kobzar at mail.ru  Tue Feb  9 08:57:02 2010
From: sergey.kobzar at mail.ru (Sergey Kobzar)
Date: Tue, 9 Feb 2010 18:57:02 +0200
Subject: [Freeswitch-users] Video conferencing
Message-ID: <57499143.20100209185702@mail.ru>

Hello.

Does anybody have a success story of implementing video conferencing?

I've spend some time with Goole and found that Asterisk has many
limitations, hardware solutions are quite expensive. I played with FS
without luck.

Any ideas?

Thanks.

-- 
Sergey




From jbrucehopkins at gmail.com  Tue Feb  9 12:03:08 2010
From: jbrucehopkins at gmail.com (Bruce Hopkins)
Date: Tue, 9 Feb 2010 20:03:08 +0000
Subject: [Freeswitch-users] g.722 --> SPEEX/16000 or SPEEX/32000 transcoding
	issue
Message-ID: <d03a10e01002091203qde4caabi1a7565929e60918f@mail.gmail.com>

Hi,

Using the FusionPBX ISO (FreeSWITCH 1.0.4) I find I cannot make a call which
requires transcoding from g.722 (or other codec which declares 8kHz sampling
rate in the SDP) to SPEEX/16000 or SPEEX/32000.

If the calling extension uses only g.722, alaw, ulaw, etc, then only the
SPEEX/8000 narrowband variety of SPEEX is offered to the recipient extension
in the SDP of the SIP invite.

If the call is initiated the other way round - e.g. Client using SPEEX/32000
--> FreeSWITCH --> Client using g.722,  then the call is transcoded with no
problem.

I am wondering if this is the intended behaviour, to avoid transcoding
narrowband --> wideband.  However what I am finding is that transcodinig
g722/8000 (wideband) to SPEEX (wideband or ultrawideband) does not seem to
work.

I would be most grateful if anyone were able to let me know if there is a
configuration option I can set to alter this behavious and allow the full
range of SPEEX sampling rates to be offered in the SDP to the receiving
party, regardless of the codec used by the calling party.

Also, is this perhaps different in a more recent version?

Many thanks
Bruce
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100209/959a3d07/attachment-0002.html 

From mike at jerris.com  Tue Feb  9 12:57:26 2010
From: mike at jerris.com (Michael Jerris)
Date: Tue, 9 Feb 2010 15:57:26 -0500
Subject: [Freeswitch-users] uuid_bridge isn't working
In-Reply-To: <fe67a3341002040147i5892d5b8md65e327be6dea7c5@mail.gmail.com>
References: <fe67a3341002040147i5892d5b8md65e327be6dea7c5@mail.gmail.com>
Message-ID: <BDBBA43F-8BC9-4DE9-8A6A-AA808C4FDEA3@jerris.com>

Fixed in rev 16574.

Mike

On Feb 4, 2010, at 4:47 AM, Nagalenoj H. wrote:

> Dear friends,
>     After upgrading to 'FreeSWITCH Version 1.0.trunk (16565)', uuid_bridge isn't working. When I give uuid_bridge, both the legs are not bridged, and they got disconnected.
> 
> Did the following,
> * Made a call to socket extension.
> * Answered the call.
> * Originated a call and parked it.
> * Did uuid_bridge with the uuids of the originated call and caller's uuid.
> 
> * Didn't get the legs bridged, instead both got disconnected,
> 
> Freeswitch debug log:
>       http://pastebin.freeswitch.org/12044
> 
> Facing this problem only after upgrading to this trunk version.

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100209/636a886e/attachment-0002.html 

From robert.hadley at teotech.com  Tue Feb  9 12:58:25 2010
From: robert.hadley at teotech.com (Robert Hadley)
Date: Tue, 9 Feb 2010 12:58:25 -0800
Subject: [Freeswitch-users] Where are new sounds_dir and
	recordings_dirvariables set?
In-Reply-To: <6B8839F5-DA53-4565-A7E0-876FC53BD6F7@freeswitch.org>
References: <28788FA27BE74B9FA6BFFE52BED11B60@greyhawk.tonecommander.com>
	<6B8839F5-DA53-4565-A7E0-876FC53BD6F7@freeswitch.org>
Message-ID: <21D89D46DD7A47DFB60DDF25C3E86814@greyhawk.tonecommander.com>


Configured it with --prefix=/opt/teoswitch.
/r

-----Original Message-----
From: Brian West [mailto:brian at freeswitch.org] 
Sent: Tuesday, February 09, 2010 11:34 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Where are new sounds_dir and
recordings_dirvariables set?

did you configure it with --prefix=/opt/teoswitch or did you move it?

/b

On Feb 9, 2010, at 1:28 PM, Robert Hadley wrote:

>  
> The XML conf files have been recently modified to replace
"$${base_dir}/sounds" with $${sounds_dir}.  The same replacement was done
for $${base_dir}/recordings and $${recordings_dir}.  In vars.xml, the
X-PRE-PROCESS cmd to setting the old variable sound_prefix was removed but
set commands were not added for the new variables sounds_dir and
recordings_dir.  How is FS finding determining the value of $${sounds_dir}
and $${recordings_dir} in the XML files?
>  
> I ask because I have 2 cloned FS servers where one can find sound files in
the IVR when an invalid extension is entered, but the other does not and is
using the wrong path to search for sound files.
>  
> Server 2 Fails:
> 2010-02-09 11:11:16.735745 [DEBUG] switch_ivr_menu.c:381 digits '9999'
> 2010-02-09 11:11:16.735745 [DEBUG] switch_ivr_menu.c:475 action regex
[9999] [/^((5[34][8901][0-9])||(10[0-5][0-9])||(30\d{2}))$/] [0]
> 2010-02-09 11:11:16.735745 [DEBUG] switch_ivr_menu.c:565 IVR menu
'teo_ivr' caught invalid input '9999'
> 2010-02-09 11:11:16.735745 [ERR] mod_sndfile.c:194 Error Opening File
[/opt/teoswitch/sounds/ivr/ivr-that_was_an_invalid_entry.wav] [System error
: No such file or directory.]
>  
> Server 1 Works:
> 2010-02-09 11:17:25.085660 [DEBUG] switch_ivr_menu.c:378 digits '9999'
> 2010-02-09 11:17:25.085660 [DEBUG] switch_ivr_menu.c:472 action regex
[9999] [/^((5[34][8901][0-9])||(10[0-5][0-9])||(30\d{2}))$/] [0]
> 2010-02-09 11:17:25.085660 [DEBUG] switch_ivr_menu.c:562 IVR menu
'teo_ivr' caught invalid input '9999'
> 2010-02-09 11:17:25.085660 [DEBUG] switch_ivr_play_say.c:1162 Codec
Activated L16 at 8000hz 1 channels 20ms
> 2010-02-09 11:17:25.085660 [DEBUG] switch_core_io.c:652
sofia/internal/1045 at 192.168.72.141:5060 receive message
[TRANSCODING_NECESSARY]
> 2010-02-09 11:17:26.765470 [DEBUG] switch_ivr_play_say.c:1454 done playing
file
>  
> I have compared the conf folders and they are nearly identical.  The only
difference I know of is Server 2 was originally compiled with mod_flite
enabled but it is not loaded at runtime.  Any suggestions why one server can
find sound files but the other looks in the wrong path?
>  
> Thanks,
> Robert







From brian at freeswitch.org  Tue Feb  9 12:59:48 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 9 Feb 2010 14:59:48 -0600
Subject: [Freeswitch-users] g.722 --> SPEEX/16000 or SPEEX/32000
	transcoding issue
In-Reply-To: <d03a10e01002091203qde4caabi1a7565929e60918f@mail.gmail.com>
References: <d03a10e01002091203qde4caabi1a7565929e60918f@mail.gmail.com>
Message-ID: <03A2F6D0-754A-4067-A106-9D91B0D68A92@freeswitch.org>

I would recommend you contact the FusionPBX project as 1.0.4 is no longer supported by our team as we are about to release 1.0.5 this week.

/b

On Feb 9, 2010, at 2:03 PM, Bruce Hopkins wrote:

> 
> Using the FusionPBX ISO (FreeSWITCH 1.0.4) I find I cannot make a call which requires transcoding from g.722 (or other codec which declares 8kHz sampling rate in the SDP) to SPEEX/16000 or SPEEX/32000.




From mike at jerris.com  Tue Feb  9 13:02:30 2010
From: mike at jerris.com (Michael Jerris)
Date: Tue, 9 Feb 2010 16:02:30 -0500
Subject: [Freeswitch-users] Passing channel variables to mod_fifo and
	xml_cdr
In-Reply-To: <004301caa608$534747d0$f9d5d770$@net>
References: <004301caa608$534747d0$f9d5d770$@net>
Message-ID: <3B778B9C-8884-4E07-B856-2F7C0B317F36@jerris.com>

There is no association if the agent does not accept the call.  When fifo has calls waiting for agents, it will call as many agents as it needs for calls waiting to be handled, when those agents accept the call, only then does it go and grab a specific caller to attach to the agent.  What most people assume incorrectly is that it is calling the agent using the waiting caller.  There is no connection at all at the time the call is made.  This is the same reason that you do not get caller id of the caller until you answer.  This is how mod_fifo is architected and a behavior that can not be changed without writing a completely different call queue system.

Mike

On Feb 4, 2010, at 9:09 PM, Adam Ford wrote:

> When sending a call through mod_fifo I seem to be losing my custom channel variables that were assigned during prior processing of the call.  In my example, I am trying to assign a unique identifier at the time the call enters my FreeSWITCH system in order to more easily tie the xml_cdr logs together.  This works great, until a call is processed through mod_fifo, which drops my custom channel variable in the calls that it generates.  Is it likely that I have something wrong with my config? Or does mod_fifo not support the passing of custom channel variables?
>  
> The overall problem I am trying to solve is that mod_fifo generates a separate a-leg for every time it rings an agent.  If the agent answers, the a-leg log gets tied to the associated b-leg log with the uuids and I am able to see the entire call in xml_cdr.  However, if the agent rejects the call or doesn?t answer, the a-leg is abandoned with seemingly no association back to a b-leg log.  Anyone have a better suggestion for tying all these xml_cdr logs together for a full picture of the call?  
>  
> -Adam
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100209/12063867/attachment-0002.html 

From mike at jerris.com  Tue Feb  9 13:08:45 2010
From: mike at jerris.com (Michael Jerris)
Date: Tue, 9 Feb 2010 16:08:45 -0500
Subject: [Freeswitch-users] Seeking Advice on SIP Phones like Aastra
In-Reply-To: <845952.61278.qm@web33007.mail.mud.yahoo.com>
References: <845952.61278.qm@web33007.mail.mud.yahoo.com>
Message-ID: <9DFA265F-8632-466B-8910-1E2F9D956908@jerris.com>


On Feb 4, 2010, at 10:38 PM, Darren C. wrote:

> Tim,
>  
> Many thanks for your response.  I posted this message on the Dev list and all I heard was crickets.  I would think a web GUI for a phone would be in demand by the FS community....
>  
> We are doing something similar to what you described.  We?re developing a rather complex IVR/Switching application and it currently does all its database writes via our Web Service to an MS SQL database.  We have a web site that is updated with call details via the web service?s backend database.  From this web interface a user can see counts of voicemails, see call activity, play voicemails, see calls in progress, record calls, etc.  It?s a specialized application so it doesn?t have every PBX feature but this is what we wanted to do with a high-end SIP Phone to replace our office PBX.
>  
> Currently we have an ESI (Estech) E-Class PBX that uses normal digital phones as well as proprietary VOIP (non-SIP) phones.  I think these really nice SIP phones with huge color touchscreens would be much better than even high-end proprietary digital phones + we?d get all the benefits of FS.  We?ll just need to add all the basic PBX capabilities to the phone?s GUI to see how many voicemails are waiting, how many lines are in use, button for call transfer, etc.
>  
> As you mentioned:  ?you can do a DB query to find out what voicemail messages a user has, or what calls are active, etc.?  We are keeping track of all this ourselves via our web service?s backend database.  I?m not sure I need to do this for everything but we are.  It?s a multi-tenant system with hundreds of tenants so I?m guessing I might lose some needed relationships by querying FS but I?m going to re-visit this?I will make sure I can?t just query FS like you?re doing for some of these things?we?ve never turned on ODBC or even looked for the documentation as to what FS stores.  We have to run FS in Linux for some Sangoma stuff but we?re Windows people so that is another reason we store via web services to an MS SQL backend.

FreeSWITCH can use ODBC to talk directly to your db, and for this sort of setup, its probably a requirement to use a db other than the embedded database.  Sangoma wanpipe on windows I am told works or basically works.  I would talk to sangoma for more information.

> I was worried I?d get one of these fancy phones and find out it doesn?t support important SIP/FS features rendering the color touchscreen useless.  I?ve never owned one of these SIP phones, just used various softphones.  But thanks to you I?ll get an Aastra 6739i and give it a try.

Typically the applications on the phone are not related to sip at all.

> I have a fulltime programmer working on this system on and off now for over a year but this has been mostly developing the IVR.  We haven't made any attempt at using FS as a PBX.  So based on your comments I think I?ll purchase an Aastra 6739i and develop a custom SIP Phone GUI interface with FS.

I am not sure the phone gui has much to do with FreeSWITCH at all, other than pulling a little data from the databases (again, ODBC highly recommended)
 
> If you or others would like to collaborate on an Aastra 6739i phone GUI for FS, feel free to contact me.  We can try to make it extensible for other phones as well.  My email is ustcor<remove this junk>poration at yah<remove this too>oo.com. 

Can you post for all to see some idea of how these applications work, lanagages used, some samples so we can see if we can get some interest in this?

Mike

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100209/558fc211/attachment-0002.html 

From mike at jerris.com  Tue Feb  9 13:10:06 2010
From: mike at jerris.com (Michael Jerris)
Date: Tue, 9 Feb 2010 16:10:06 -0500
Subject: [Freeswitch-users] Presence PUBLISH Not Updating After
	Softphone OffLine Then Available
In-Reply-To: <31FEE9072F104AE18B26812978FCA341@greyhawk.tonecommander.com>
References: <31FEE9072F104AE18B26812978FCA341@greyhawk.tonecommander.com>
Message-ID: <45963EB5-DFE5-43FA-AFCC-857E1E719E2D@jerris.com>

Try this again, I think I saw changes go in for this issue.

Mike

On Feb 5, 2010, at 2:38 PM, Jerry Richards wrote:

> I found a scenario where presence status is not distributed to subscribers.
> This is using the latest changes (as of Feb 03, 2010).  The scenario
> follows:
> 
> 1) Register two Bria softphones (A and B), which each have the other as a
> contact.
> 2) Set softphone B's presence status to 'Appear Offline'.
> 3) Softphone A correctly reflects contact B is offline.
> 4) Set softphone B's presence status to 'Available'.
> 5) ******* There is no change to contact B's status at softphone A *******
> 
> I posted a log at http://pastebin.freeswitch.org/12054.  At line 773, there
> is an error when FS is processing the PUBLISH from softphone B:
> 
> 773.2010-02-05 10:29:21.254221 [ERR] sofia_presence.c:674 DUMP PRESENCE SQL:
> 
> I did notice that after about 30 minutes, softphone B's status gets
> reflected at softphone A.




From mike at jerris.com  Tue Feb  9 13:12:38 2010
From: mike at jerris.com (Michael Jerris)
Date: Tue, 9 Feb 2010 16:12:38 -0500
Subject: [Freeswitch-users] Driving peripherals through Freeswitch?
In-Reply-To: <2srvm5945qgcno44oetn9ngii0u3aed73p@4ax.com>
References: <2srvm5945qgcno44oetn9ngii0u3aed73p@4ax.com>
Message-ID: <20F1477A-E98A-4BC9-904D-CB313D8E7B4C@jerris.com>

The possibilities are limitless, but requires someone to code a module to interface, or external scripts using the system api or some socket based application.

Mike

On Feb 8, 2010, at 6:07 AM, Fred-145 wrote:
> I don't know anything about this, but I was wondering if someone had
> successfully used a Freeswitch server to drive peripherals like
> switching on a heater by sending an SMS or calling an extension, etc.?
> 
> I'm thinking of tools like X10 to drive peripherals from a PC.
> 
> Has someone played with this kind of tool and could tell me what is
> technically possible?




From mike at jerris.com  Tue Feb  9 13:14:42 2010
From: mike at jerris.com (Michael Jerris)
Date: Tue, 9 Feb 2010 16:14:42 -0500
Subject: [Freeswitch-users] Dialplan search order
In-Reply-To: <798899361.20100208131909@yes.net.ua>
References: <798899361.20100208131909@yes.net.ua>
Message-ID: <E79C2B30-921E-4DC9-A045-D4B550F65B02@jerris.com>


On Feb 8, 2010, at 6:19 AM, Mike Tkachuk wrote:
> 
>  I'm using xml_curl for external dialplan fetch, but I like to
>  split static and dynamic parts of configuration, so for example,
>  leave call unloop logic, and voicemail extension in static xml file
>  while having all other parts dynamic. It will allow to avoid
>  unnecessary call to costly external source and also avoid xml parse
>  of content that is static.
> 
>  Currently FS first look in xml_curl and only after that falls back
>  to static files. Is that behavior possible to change, so FS will work
>  like that:
> 
>  1 - Look in static xml file and execute all extensions that have
>  'continue="true"'
>  2 - If previous step didn't stop on matching extension than look in
>  xml_curl or other source specified in dialplan param of sofia
>  config.
> 
>  Looks like <param name="dialplan" value="XML:/path_to_dialplan_default.xml,XML"/>
>  don't do the trick.

This is exactly how you do this.  What is not working about it?

Mike
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100209/70365a7c/attachment-0002.html 

From jbrucehopkins at gmail.com  Tue Feb  9 13:15:05 2010
From: jbrucehopkins at gmail.com (Bruce Hopkins)
Date: Tue, 9 Feb 2010 21:15:05 +0000
Subject: [Freeswitch-users] g.722 --> SPEEX/16000 or SPEEX/32000
	transcoding issue
In-Reply-To: <03A2F6D0-754A-4067-A106-9D91B0D68A92@freeswitch.org>
References: <d03a10e01002091203qde4caabi1a7565929e60918f@mail.gmail.com>
	<03A2F6D0-754A-4067-A106-9D91B0D68A92@freeswitch.org>
Message-ID: <d03a10e01002091315n5fa9242eoe2a86c4edee1501d@mail.gmail.com>

OK, many thanks for the extremely swift response Brian.

I will try to get up and running as soon as I can with 1.0.5 and see if the
issue goes away.

thanks again
Bruce

On 9 February 2010 20:59, Brian West <brian at freeswitch.org> wrote:

> I would recommend you contact the FusionPBX project as 1.0.4 is no longer
> supported by our team as we are about to release 1.0.5 this week.
>
> /b
>
> On Feb 9, 2010, at 2:03 PM, Bruce Hopkins wrote:
>
> >
> > Using the FusionPBX ISO (FreeSWITCH 1.0.4) I find I cannot make a call
> which requires transcoding from g.722 (or other codec which declares 8kHz
> sampling rate in the SDP) to SPEEX/16000 or SPEEX/32000.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100209/9f32a266/attachment-0002.html 

From robert.hadley at teotech.com  Tue Feb  9 13:14:57 2010
From: robert.hadley at teotech.com (Robert Hadley)
Date: Tue, 9 Feb 2010 13:14:57 -0800
Subject: [Freeswitch-users] Way to pass max-members from dialplan into
	conference?
Message-ID: <C7136E29C3FD472BB367FF412E282BED@greyhawk.tonecommander.com>

I have setting max-members=10 in conference.conf.xml working.  However, is
there are way to pass in the max-members=10 from the dialplan/default.xml to
mod_conference?  I tried using action application="set"
data="max-members=10" but it didn't work.  Also tried action
application="export" data="max-members=10" but it didn't work either.

 

>From default.xml:

    <extension name="nb_conferences">

      <condition field="destination_number" expression="^(30\d{2})$">

       <action application="answer"/>

<!-- tried set and export here to no avail -->

       <action application="conference" data="$1-${domain_name}@default"/>

 

Thanks,

Robert

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100209/e18f8b40/attachment-0002.html 

From mike at jerris.com  Tue Feb  9 13:16:13 2010
From: mike at jerris.com (Michael Jerris)
Date: Tue, 9 Feb 2010 16:16:13 -0500
Subject: [Freeswitch-users] Video conferencing
In-Reply-To: <57499143.20100209185702@mail.ru>
References: <57499143.20100209185702@mail.ru>
Message-ID: <D9657862-D8A0-45C8-98E2-42814BE8D6A1@jerris.com>

Our video conference features "work" but the functionality is pretty limited.  We don;t have iframe detection and can not do any video transcoding, just video follow audio support.  This code needs some work for sure.

Mike

On Feb 9, 2010, at 11:57 AM, Sergey Kobzar wrote:

> Does anybody have a success story of implementing video conferencing?
> 
> I've spend some time with Goole and found that Asterisk has many
> limitations, hardware solutions are quite expensive. I played with FS
> without luck.




From brian at freeswitch.org  Tue Feb  9 13:18:33 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 9 Feb 2010 15:18:33 -0600
Subject: [Freeswitch-users] g.722 --> SPEEX/16000 or SPEEX/32000
	transcoding issue
In-Reply-To: <d03a10e01002091315n5fa9242eoe2a86c4edee1501d@mail.gmail.com>
References: <d03a10e01002091203qde4caabi1a7565929e60918f@mail.gmail.com>
	<03A2F6D0-754A-4067-A106-9D91B0D68A92@freeswitch.org>
	<d03a10e01002091315n5fa9242eoe2a86c4edee1501d@mail.gmail.com>
Message-ID: <F167991C-151B-4D40-BF27-80A4C3A44D1E@freeswitch.org>

What you're saying makes little or no sense to me even on 1.0.4, Can you pastebin your logs?

/b

On Feb 9, 2010, at 3:15 PM, Bruce Hopkins wrote:

> OK, many thanks for the extremely swift response Brian.
> 
> I will try to get up and running as soon as I can with 1.0.5 and see if the issue goes away.
> 
> thanks again
> Bruce




From mike at jerris.com  Tue Feb  9 13:25:35 2010
From: mike at jerris.com (Michael Jerris)
Date: Tue, 9 Feb 2010 16:25:35 -0500
Subject: [Freeswitch-users] No reINVITE when bridging two sessions from
	JavaScript with bypass_media_after_bridge=true
In-Reply-To: <8e9d67561002051202qeaf1551v8fdd29565f8c02fd@mail.gmail.com>
References: <8e9d67561002041147q6c135546u1f7e1887529a916d@mail.gmail.com>
	<3D739D7F-F5C6-4266-B04E-9413DCFDA6AF@freeswitch.org>
	<8e9d67561002050018q24e3a806kc8807b55268e759b@mail.gmail.com>
	<B66289FF-C2A9-4AC4-A56C-66E51B4726F4@freeswitch.org>
	<8e9d67561002051202qeaf1551v8fdd29565f8c02fd@mail.gmail.com>
Message-ID: <154214DA-5781-4FC1-81EA-B41392E6ACF6@jerris.com>

controlling multiple calls in a script like this is tricky, you need to use the first session to create the second one.  Why are you not just doing an originate to do all of this not even in a js file?  What exactly are you trying to accomplish

Mike

On Feb 5, 2010, at 3:02 PM, Timur Valishev wrote:

> I think we are on the right way) still does not work, but there is hope)
> 
> First of all, this script does not produce any reinvite either (even if replace bypass_media to bypass_media_after_bridge, or set bypass_media only on one channel):
> 
> <<<<<<<<<<<<<<<<<<<<<<<<<<<<<
> session = new Session("{bypass_media=true,ignore_early_media=true,hangup_after_bridge=true}user/1001");
> session2 = new Session("{bypass_media=true,ignore_early_media=true} user/1001");
> bridge(session, session2);
> >>>>>>>>>>>>>>>>>>>>>>>>>>>>>
> 
> BUT! if I run the following script:
> <<<<<<<<<<<<<<<<<<<<<<<<<<<<<
> session = new Session("{bypass_media=true,ignore_early_media=true,hangup_after_bridge=true} user/1001");
> session2 = new Session("{bypass_media=true,ignore_early_media=true}user/1001");
> >>>>>>>>>>>>>>>>>>>>>>>>>>>>>
> 
> And then manually type in the console
> uuid_media off <uuid of my session>
> 
> - then I get the reINVITE!
> 
> BUT! When I try to write it to the script:
> <<<<<<<<<<<<<<<<<<<<<<<<<<<<<
> session = new Session("{bypass_media=true,ignore_early_media=true,hangup_after_bridge=true}sofia/external/timwork at novion.ru");
> session2 = new Session("{bypass_media=true,ignore_early_media=true}sofia/external/timwork at novion.ru");
> bridge(session, session2);
> apiExecute('uuid_media off '+session.uuid); // <-- this line is not executed, because bridge hangs up untill BYE
> >>>>>>>>>>>>>>>>>>>>>>>>>>>>>
> 
> the last line is not executed, because bridge hangs up untill BYE
> 
> Then I've tried to do like this:
> <<<<<<<<<<<<<<<<<<<<<<<<<<<<<
> session = new Session("{bypass_media=true,ignore_early_media=true,hangup_after_bridge=true}user/1001");
> session2 = new Session("{bypass_media=true,ignore_early_media=true}user/1001");
> 
> session.setAutoHangup(false)
> session2.setAutoHangup(false)
> 
> apiExecute("uuid_bridge "+session.uuid+" "+session2.uuid);
> apiExecute('uuid_media off '+session.uuid);
> >>>>>>>>>>>>>>>>>>>>>>>>>>>>>
> 
> But sessions do not get bridged -( Even if I insert session.ready() after each call.
> 
> Any ideas on how to call the functions correctly to get the reINVITE?
> 
> Best regards,
> Timur Valishev
> 
> 2010/2/5 Brian West <brian at freeswitch.org>
> set it inside each of the {} for each session you create its not set after the fact the call is up already...  you're setting it too late.
> 
> you an also issue uuid_media off <uuid>
> 
> /b
> 
> On Feb 5, 2010, at 2:18 AM, Timur Valishev wrote:
> 
>> I've modified my script to make sure: <<<<<<<<<<<<<<<<<<<<<<<<<<<<,
>> session = new Session(
>> "{ignore_early_media=true,hangup_after_bridge=true}sofia/external/timwork at novion.ru"
>> );
>> session2 = new Session(
>> "{ignore_early_media=true}sofia/external/timwork at novion.ru"
>> );
>> session.setVariable('bypass_media', 'true');
>> session2.setVariable('bypass_media', 'true');
>> bridge(session, session2);
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100209/41bcb5e4/attachment-0002.html 

From jbrucehopkins at gmail.com  Tue Feb  9 13:55:13 2010
From: jbrucehopkins at gmail.com (Bruce Hopkins)
Date: Tue, 9 Feb 2010 21:55:13 +0000
Subject: [Freeswitch-users] g.722 --> SPEEX/16000 or SPEEX/32000
	transcoding issue
In-Reply-To: <F167991C-151B-4D40-BF27-80A4C3A44D1E@freeswitch.org>
References: <d03a10e01002091203qde4caabi1a7565929e60918f@mail.gmail.com>
	<03A2F6D0-754A-4067-A106-9D91B0D68A92@freeswitch.org>
	<d03a10e01002091315n5fa9242eoe2a86c4edee1501d@mail.gmail.com>
	<F167991C-151B-4D40-BF27-80A4C3A44D1E@freeswitch.org>
Message-ID: <d03a10e01002091355n1934627t3902e0d8af5c7dc3@mail.gmail.com>

Willdo,

To clarify in brief though, the scenario which occurs and causes the call to
fail is:

SIP client  1 (g.722 enabled only ) -----> INVITE (with SDP offer: Media
Attribute (a): rtpmap:9 g722/8000 ) -->FreeSWITCH

---> INVITE (with SDP offer including a bunch of codecs including rtpmap:
rtpmap:98 SPEEX/8000 but crucially *not* including SPEEX/16000 or
SPEEX/32000)

---> SIP client 2 (with only SPEEX/16000 or SPEEX/32000 enabled).

The second SIP client does not get offered a codec it can accept, so SIP
client 1 is sent a method 488 "Not Acceptable Here" message and the calling
party gets directed to the voicemail for the other SIP client.

By contrast, there is no problem calling SPEEX/32000 --> SPEEX/32000 or
calling SPEEX/16000 --> SPEEX/16000.

there is also no problem calling SPEEX/32000 --> g.722/8000.

I am wondering if the problem is that FreeSWITCH is interpreting g.722 as
being a narrowband (8kHz sample rate) codec, due to the historic anomaly of
it presenting g722/8000 in the SDP even though it in fact uses 16kHz
sampling, and for that reason not wanting to offer a 16kHz sample rate codec
to the second SIP client?

I suggest this as I also found trying to call alaw --> SPEEX/16000 does not
work, for example.



Here is the log file for the scenario which does not work (g.722 client
trying to call Speex wideband client).  Please let me know if a Wireshark
trace would be helpful.

2010-02-09 21:49:32.784387 [DEBUG] sofia.c:5224 0 acls to check for proxy
2010-02-09 21:49:32.784387 [DEBUG] sofia.c:5242 network ip is a proxy [0]
2010-02-09 21:49:32.784387 [DEBUG] sofia.c:5270 IP 192.168.10.131 Rejected
by acl "domains". Falling back to Digest auth.
2010-02-09 21:49:32.791074 [DEBUG] sofia.c:5224 0 acls to check for proxy
2010-02-09 21:49:32.791074 [DEBUG] sofia.c:5242 network ip is a proxy [0]
2010-02-09 21:49:32.791074 [DEBUG] sofia.c:5270 IP 192.168.10.131 Rejected
by acl "domains". Falling back to Digest auth.
2010-02-09 21:49:32.794074 [NOTICE] switch_channel.c:613 New Channel
sofia/internal/60002 at 192.168.10.30 [66610a8c-aec9-48ca-9b67-aafdcd3e3d08]
2010-02-09 21:49:32.794074 [DEBUG] sofia.c:3727 Channel sofia/internal/
60002 at 192.168.10.30 entering state [received][100]
2010-02-09 21:49:32.794074 [DEBUG] sofia.c:3738 Remote SDP:
v=0
o=- 5 2 IN IP4 192.168.10.131
s=CounterPath eyeBeam 1.5
c=IN IP4 192.168.10.131
t=0 0
m=audio 25592 RTP/AVP 9 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

2010-02-09 21:49:32.794074 [DEBUG] sofia_glue.c:3305 Audio Codec Compare
[G722:9:8000:20]/[G722:9:8000:20]
2010-02-09 21:49:32.795079 [DEBUG] sofia_glue.c:2143 Set Codec
sofia/internal/60002 at 192.168.10.30 G722/8000 20 ms 160 samples
2010-02-09 21:49:32.795079 [DEBUG] sofia_glue.c:3261 Set 2833 dtmf payload
to 101
2010-02-09 21:49:32.795079 [DEBUG] sofia.c:3885 (sofia/internal/
60002 at 192.168.10.30) State Change CS_NEW -> CS_INIT
2010-02-09 21:49:32.795079 [DEBUG] switch_core_session.c:999 Send signal
sofia/internal/60002 at 192.168.10.30 [BREAK]
2010-02-09 21:49:32.795079 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/60002 at 192.168.10.30) Running State Change CS_INIT
2010-02-09 21:49:32.795079 [DEBUG] switch_core_state_machine.c:338
(sofia/internal/60002 at 192.168.10.30) State INIT
2010-02-09 21:49:32.795079 [DEBUG] mod_sofia.c:83 sofia/internal/
60002 at 192.168.10.30 SOFIA INIT
2010-02-09 21:49:32.795079 [DEBUG] mod_sofia.c:111 (sofia/internal/
60002 at 192.168.10.30) State Change CS_INIT -> CS_ROUTING
2010-02-09 21:49:32.795079 [DEBUG] switch_core_session.c:999 Send signal
sofia/internal/60002 at 192.168.10.30 [BREAK]
2010-02-09 21:49:32.795079 [DEBUG] switch_core_state_machine.c:338
(sofia/internal/60002 at 192.168.10.30) State INIT going to sleep
2010-02-09 21:49:32.795079 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/60002 at 192.168.10.30) Running State Change CS_ROUTING
2010-02-09 21:49:32.795079 [DEBUG] switch_core_state_machine.c:341
(sofia/internal/60002 at 192.168.10.30) State ROUTING
2010-02-09 21:49:32.795079 [DEBUG] mod_sofia.c:132 sofia/internal/
60002 at 192.168.10.30 SOFIA ROUTING
2010-02-09 21:49:32.795079 [DEBUG] switch_core_state_machine.c:78
sofia/internal/60002 at 192.168.10.30 Standard ROUTING
2010-02-09 21:49:32.795079 [INFO] mod_dialplan_xml.c:408 Processing
60002->1001 in context default
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->unloop]
continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (PASS) [unloop]
${unroll_loops}(true) =~ /^true$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [unloop]
${sip_looped_call}() =~ /^true$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->tod_example]
continue=true
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (PASS) [tod_example]
${strftime(%w)}(2) =~ /^([1-5])$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [tod_example]
${strftime(%H%M)}(2149) =~ /^((09|1[0-7])[0-5][0-9]|1800)$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
[default->global-intercept] continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [global-intercept]
destination_number(1001) =~ /^\*886$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
[default->group-intercept] continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [group-intercept]
destination_number(1001) =~ /^\*8$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
[default->intercept-ext] continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [intercept-ext]
destination_number(1001) =~ /^\*\*(\d+)$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->redial]
continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [redial]
destination_number(1001) =~ /^\*870$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->global]
continue=true
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [global]
${network_addr}(192.168.10.131) =~ /^$/ break=never
Dialplan: sofia/internal/60002 at 192.168.10.30 ANTI-Action
set(use_profile=${cond(${acl(${network_addr} rfc1918)} == true ? nat :
default)})
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (PASS) [global]
${numbering_plan}() =~ /^$/ break=never
Dialplan: sofia/internal/60002 at 192.168.10.30 Action
set_user(default@${domain_name})

Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [global]
${call_debug}(false) =~ /^true$/ break=never
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [global]
${sip_has_crypto}() =~ /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/
break=never
Dialplan: sofia/internal/60002 at 192.168.10.30 Absolute Condition [global]
Dialplan: sofia/internal/60002 at 192.168.10.30 Action
db(insert/${domain_name}-spymap/${caller_id_number}/${uuid})
Dialplan: sofia/internal/60002 at 192.168.10.30 Action
db(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number})

Dialplan: sofia/internal/60002 at 192.168.10.30 Action
db(insert/${domain_name}-last_dial/global/${uuid})
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->snom-demo-2]
continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [snom-demo-2]
destination_number(1001) =~ /^\*9001$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->snom-demo-1]
continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [snom-demo-1]
destination_number(1001) =~ /^\*9000$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->eavesdrop]
continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [eavesdrop]
destination_number(1001) =~ /^\*88(.*)$|^\*0(.*)$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->eavesdrop]
continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [eavesdrop]
destination_number(1001) =~ /^\*779$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->call_privacy]
continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [call_privacy]
destination_number(1001) =~ /^\*67(\d+)$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->call_return]
continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [call_return]
destination_number(1001) =~ /^\*69$|^869$|^lcr$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->del-group]
continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [del-group]
destination_number(1001) =~ /^\*80(\d{2})$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->add-group]
continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [add-group]
destination_number(1001) =~ /^\*81(\d{2})$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
[default->call-group-simo] continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [call-group-simo]
destination_number(1001) =~ /^\*82(\d{2})$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
[default->call-group-order] continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [call-group-order]
destination_number(1001) =~ /^\*83(\d{2})$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
[default->extension-intercom] continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL)
[extension-intercom] destination_number(1001) =~ /^\*8(\d{4})$/
break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
[default->send_to_voicemail_5digits] continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL)
[send_to_voicemail_5digits] destination_number(1001) =~ /^\*99(\d{5})$/
break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
[default->send_to_voicemail_4digits] continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL)
[send_to_voicemail_4digits] destination_number(1001) =~ /^\*99(\d{4})$/
break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
[default->send_to_voicemail_3digits] continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL)
[send_to_voicemail_3digits] destination_number(1001) =~ /^\*99(\d{3})$/
break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->pizza_demo]
continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [pizza_demo]
destination_number(1001) =~ /^(pizza|74992)$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->2001]
continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [2001]
destination_number(1001) =~ /^2001$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->Call via
asterisk-pbx1] continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [Call via
asterisk-pbx1] destination_number(1001) =~ /269065/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->pizza_demo]
continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [pizza_demo]
destination_number(1001) =~ /^(pizza|74992)$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->5002]
continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [5002]
destination_number(1001) =~ /^5002$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->7002]
continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [7002]
destination_number(1001) =~ /^7002$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->DISA]
continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [DISA]
destination_number(1001) =~ /^\*(3472)$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->Recordings]
continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [Recordings]
destination_number(1001) =~ /^\*(732673)$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->7002.park]
continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [7002.park]
destination_number(1001) =~ /^\*7002$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
[default->group_dial_sales] continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [group_dial_sales]
destination_number(1001) =~ /^\*2000$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
[default->group_dial_support] continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL)
[group_dial_support] destination_number(1001) =~ /^\*2001$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
[default->group_dial_billing] continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL)
[group_dial_billing] destination_number(1001) =~ /^\*2002$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->vmain2]
continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [vmain2]
destination_number(1001) =~ /^vmain2$|^\*97$|^\*4000$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->vmain]
continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [vmain]
destination_number(1001) =~ /^vmain$|^\*98$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->sip_uri]
continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [sip_uri]
destination_number(1001) =~ /^sip:(.*)$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
[default->nb_conferences] continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [nb_conferences]
destination_number(1001) =~ /^\*(30\d{2})$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
[default->wb_conferences] continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [wb_conferences]
destination_number(1001) =~ /^\*(31\d{2})$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
[default->uwb_conferences] continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [uwb_conferences]
destination_number(1001) =~ /^\*(32\d{2})$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
[default->cdquality_conferences] continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL)
[cdquality_conferences] destination_number(1001) =~ /^\*(33\d{2})$/
break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
[default->freeswitch_public_conf_via_sip] continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL)
[freeswitch_public_conf_via_sip] destination_number(1001) =~
/^\*9(888|1616|3232)$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
[default->mad_boss_intercom] continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL)
[mad_boss_intercom] destination_number(1001) =~ /^\*0911$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
[default->mad_boss_intercom] continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL)
[mad_boss_intercom] destination_number(1001) =~ /^\*0912$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->mad_boss]
continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [mad_boss]
destination_number(1001) =~ /^\*0913$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->ivr_demo]
continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [ivr_demo]
destination_number(1001) =~ /^\*5000$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
[default->dynamic_conference] continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL)
[dynamic_conference] destination_number(1001) =~ /^\*5001$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
[default->rtp_multicast_page] continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL)
[rtp_multicast_page] destination_number(1001) =~ /^\*pagegroup$|^\*7243/
break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->park]
continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [park]
destination_number(1001) =~ /^\*5900$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->unpark]
continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [unpark]
destination_number(1001) =~ /^\*5901$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->park]
continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (PASS) [park]
source(mod_sofia) =~ /mod_sofia/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [park]
destination_number(1001) =~ /park\+(\d+)/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->unpark]
continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (PASS) [unpark]
source(mod_sofia) =~ /mod_sofia/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [unpark]
destination_number(1001) =~ /^parking$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->park]
continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (PASS) [park]
source(mod_sofia) =~ /mod_sofia/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [park]
destination_number(1001) =~ /callpark/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->unpark]
continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (PASS) [unpark]
source(mod_sofia) =~ /mod_sofia/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [unpark]
destination_number(1001) =~ /pickup/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->wait]
continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [wait]
destination_number(1001) =~ /^wait$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->fax_receive]
continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [fax_receive]
destination_number(1001) =~ /^\*9978$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->fax_transmit]
continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [fax_transmit]
destination_number(1001) =~ /^\*9979$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->ringback_180]
continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [ringback_180]
destination_number(1001) =~ /^\*9980$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
[default->ringback_183_uk_ring] continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL)
[ringback_183_uk_ring] destination_number(1001) =~ /^\*9981$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
[default->ringback_183_music_ring] continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL)
[ringback_183_music_ring] destination_number(1001) =~ /^\*9982$/
break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
[default->ringback_post_answer_uk_ring] continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL)
[ringback_post_answer_uk_ring] destination_number(1001) =~ /^\*9983$/
break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
[default->ringback_post_answer_music] continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL)
[ringback_post_answer_music] destination_number(1001) =~ /^\*9984$/
break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->ClueCon]
continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [ClueCon]
destination_number(1001) =~ /^\*9991$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->show_info]
continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [show_info]
destination_number(1001) =~ /^\*9992$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->video_record]
continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [video_record]
destination_number(1001) =~ /^\*9993$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
[default->video_playback] continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [video_playback]
destination_number(1001) =~ /^\*9994$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->delay_echo]
continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [delay_echo]
destination_number(1001) =~ /^\*9995$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->echo]
continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [echo]
destination_number(1001) =~ /^\*9996$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->milliwatt]
continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [milliwatt]
destination_number(1001) =~ /^\*9997$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->tone_stream]
continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [tone_stream]
destination_number(1001) =~ /^\*9998$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
[default->zrtp_enrollement] continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [zrtp_enrollement]
destination_number(1001) =~ /^\*9787$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->hold_music]
continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [hold_music]
destination_number(1001) =~ /^\*9999$/ break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
[default->Local_Extension] continue=false
Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (PASS) [Local_Extension]
destination_number(1001) =~ /(^\d{6}$|\d{5}$|^\d{4}$|^\d{3}$)/
break=on-false
Dialplan: sofia/internal/60002 at 192.168.10.30 Action
set(dialed_extension=1001)
Dialplan: sofia/internal/60002 at 192.168.10.30 Action
export(dialed_extension=1001)
Dialplan: sofia/internal/60002 at 192.168.10.30 Action bind_meta_app(1 b s
execute_extension::dx XML features)
Dialplan: sofia/internal/60002 at 192.168.10.30 Action bind_meta_app(2 b s
record_session::/usr/local/freeswitch/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav)

Dialplan: sofia/internal/60002 at 192.168.10.30 Action bind_meta_app(3 b s
execute_extension::cf XML features)
Dialplan: sofia/internal/60002 at 192.168.10.30 Action set(ringback=${us-ring})

Dialplan: sofia/internal/60002 at 192.168.10.30 Action
set(transfer_ringback=local_stream://moh)
Dialplan: sofia/internal/60002 at 192.168.10.30 Action set(call_timeout=30)
Dialplan: sofia/internal/60002 at 192.168.10.30 Action
set(hangup_after_bridge=true)
Dialplan: sofia/internal/60002 at 192.168.10.30 Action
set(continue_on_fail=true)
Dialplan: sofia/internal/60002 at 192.168.10.30 Action
db(insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number})

Dialplan: sofia/internal/60002 at 192.168.10.30 Action
db(insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid})
Dialplan: sofia/internal/60002 at 192.168.10.30 Action
set(called_party_callgroup=${user_data(${dialed_extension}@${domain_name}
var callgroup)})
Dialplan: sofia/internal/60002 at 192.168.10.30 Action
db(insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid})
Dialplan: sofia/internal/60002 at 192.168.10.30 Action
bridge(user/${dialed_extension}@${domain_name})
Dialplan: sofia/internal/60002 at 192.168.10.30 Action answer()
Dialplan: sofia/internal/60002 at 192.168.10.30 Action sleep(1000)
Dialplan: sofia/internal/60002 at 192.168.10.30 Action voicemail(default
${domain_name} ${dialed_extension})
2010-02-09 21:49:32.798073 [DEBUG] switch_core_state_machine.c:122
(sofia/internal/60002 at 192.168.10.30) State Change CS_ROUTING -> CS_EXECUTE
2010-02-09 21:49:32.798073 [DEBUG] switch_core_session.c:999 Send signal
sofia/internal/60002 at 192.168.10.30 [BREAK]
2010-02-09 21:49:32.798073 [DEBUG] switch_core_state_machine.c:341
(sofia/internal/60002 at 192.168.10.30) State ROUTING going to sleep
2010-02-09 21:49:32.798073 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/60002 at 192.168.10.30) Running State Change CS_EXECUTE
2010-02-09 21:49:32.798073 [DEBUG] switch_core_state_machine.c:348
(sofia/internal/60002 at 192.168.10.30) State EXECUTE
2010-02-09 21:49:32.798073 [DEBUG] mod_sofia.c:181 sofia/internal/
60002 at 192.168.10.30 SOFIA EXECUTE
2010-02-09 21:49:32.798073 [DEBUG] switch_core_state_machine.c:159
sofia/internal/60002 at 192.168.10.30 Standard EXECUTE
EXECUTE sofia/internal/60002 at 192.168.10.30 set(use_profile=default)
2010-02-09 21:49:32.798073 [DEBUG] mod_dptools.c:768 sofia/internal/
60002 at 192.168.10.30 SET [use_profile]=[default]
EXECUTE sofia/internal/60002 at 192.168.10.30 set_user(default at 192.168.10.30)
EXECUTE sofia/internal/60002 at 192.168.10.30db(insert/192.168.10.30-spymap/60002/66610a8c-aec9-48ca-9b67-aafdcd3e3d08)
EXECUTE sofia/internal/60002 at 192.168.10.30db(insert/192.168.10.30-last_dial/60002/1001)
EXECUTE sofia/internal/60002 at 192.168.10.30db(insert/192.168.10.30-last_dial/global/66610a8c-aec9-48ca-9b67-aafdcd3e3d08)
EXECUTE sofia/internal/60002 at 192.168.10.30 set(dialed_extension=1001)
2010-02-09 21:49:32.857126 [DEBUG] mod_dptools.c:768 sofia/internal/
60002 at 192.168.10.30 SET [dialed_extension]=[1001]
EXECUTE sofia/internal/60002 at 192.168.10.30 export(dialed_extension=1001)
2010-02-09 21:49:32.858075 [DEBUG] mod_dptools.c:851 EXPORT
[dialed_extension]=[1001]
EXECUTE sofia/internal/60002 at 192.168.10.30 bind_meta_app(1 b s
execute_extension::dx XML features)
2010-02-09 21:49:32.858075 [INFO] switch_ivr_async.c:2174 Bound B-Leg: 1
execute_extension::dx XML features
EXECUTE sofia/internal/60002 at 192.168.10.30 bind_meta_app(2 b s
record_session::/usr/local/freeswitch/recordings/60002.2010-02-09-21-49-32.wav)
2010-02-09 21:49:32.858075 [INFO] switch_ivr_async.c:2174 Bound B-Leg: 2
record_session::/usr/local/freeswitch/recordings/60002.2010-02-09-21-49-32.wav
EXECUTE sofia/internal/60002 at 192.168.10.30 bind_meta_app(3 b s
execute_extension::cf XML features)
2010-02-09 21:49:32.859074 [INFO] switch_ivr_async.c:2174 Bound B-Leg: 3
execute_extension::cf XML features
EXECUTE sofia/internal/60002 at 192.168.10.30 set(ringback=%(2000, 4000, 440.0,
480.0))
2010-02-09 21:49:32.859074 [DEBUG] mod_dptools.c:768 sofia/internal/
60002 at 192.168.10.30 SET [ringback]=[%(2000, 4000, 440.0, 480.0)]
EXECUTE sofia/internal/60002 at 192.168.10.30set(transfer_ringback=local_stream://moh)
2010-02-09 21:49:32.859074 [DEBUG] mod_dptools.c:768 sofia/internal/
60002 at 192.168.10.30 SET [transfer_ringback]=[local_stream://moh]
EXECUTE sofia/internal/60002 at 192.168.10.30 set(call_timeout=30)
2010-02-09 21:49:32.860074 [DEBUG] mod_dptools.c:768 sofia/internal/
60002 at 192.168.10.30 SET [call_timeout]=[30]
EXECUTE sofia/internal/60002 at 192.168.10.30 set(hangup_after_bridge=true)
2010-02-09 21:49:32.860074 [DEBUG] mod_dptools.c:768 sofia/internal/
60002 at 192.168.10.30 SET [hangup_after_bridge]=[true]
EXECUTE sofia/internal/60002 at 192.168.10.30 set(continue_on_fail=true)
2010-02-09 21:49:32.860074 [DEBUG] mod_dptools.c:768 sofia/internal/
60002 at 192.168.10.30 SET [continue_on_fail]=[true]
EXECUTE sofia/internal/60002 at 192.168.10.30db(insert/192.168.10.30-call_return/1001/60002)
EXECUTE sofia/internal/60002 at 192.168.10.30db(insert/192.168.10.30-last_dial_ext/1001/66610a8c-aec9-48ca-9b67-aafdcd3e3d08)
EXECUTE sofia/internal/60002 at 192.168.10.30 set(called_party_callgroup=)
2010-02-09 21:49:32.870074 [DEBUG] mod_dptools.c:768 sofia/internal/
60002 at 192.168.10.30 SET [called_party_callgroup]=[UNDEF]
EXECUTE sofia/internal/60002 at 192.168.10.30db(insert/192.168.10.30-last_dial//66610a8c-aec9-48ca-9b67-aafdcd3e3d08)
EXECUTE sofia/internal/60002 at 192.168.10.30 bridge(user/1001 at 192.168.10.30)
2010-02-09 21:49:32.905073 [DEBUG] switch_ivr_originate.c:1735 variable
string 0 = [presence_id=1001 at 192.168.10.30]
2010-02-09 21:49:32.905073 [NOTICE] switch_channel.c:613 New Channel
sofia/internal/sip:1001 at 192.168.10.192:41080[df7cf235-f0e4-406a-83a5-dd2d681bb278]
2010-02-09 21:49:32.905073 [DEBUG] mod_sofia.c:3142 (sofia/internal/
sip:1001 at 192.168.10.192:41080) State Change CS_NEW -> CS_INIT
2010-02-09 21:49:32.905073 [DEBUG] switch_core_session.c:999 Send signal
sofia/internal/sip:1001 at 192.168.10.192:41080 [BREAK]
2010-02-09 21:49:32.906075 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/sip:1001 at 192.168.10.192:41080) Running State Change CS_INIT
2010-02-09 21:49:32.906075 [DEBUG] switch_core_state_machine.c:338
(sofia/internal/sip:1001 at 192.168.10.192:41080) State INIT
2010-02-09 21:49:32.906075 [DEBUG] mod_sofia.c:83 sofia/internal/
sip:1001 at 192.168.10.192:41080 SOFIA INIT
2010-02-09 21:49:32.907184 [DEBUG] mod_sofia.c:111 (sofia/internal/
sip:1001 at 192.168.10.192:41080) State Change CS_INIT -> CS_ROUTING
2010-02-09 21:49:32.907184 [DEBUG] switch_core_session.c:999 Send signal
sofia/internal/sip:1001 at 192.168.10.192:41080 [BREAK]
2010-02-09 21:49:32.907184 [DEBUG] sofia.c:3727 Channel sofia/internal/
sip:1001 at 192.168.10.192:41080 entering state [calling][0]
2010-02-09 21:49:32.907184 [DEBUG] switch_core_state_machine.c:338
(sofia/internal/sip:1001 at 192.168.10.192:41080) State INIT going to sleep
2010-02-09 21:49:32.907184 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/sip:1001 at 192.168.10.192:41080) Running State Change
CS_ROUTING
2010-02-09 21:49:32.907184 [DEBUG] switch_core_state_machine.c:341
(sofia/internal/sip:1001 at 192.168.10.192:41080) State ROUTING
2010-02-09 21:49:32.907184 [DEBUG] mod_sofia.c:132 sofia/internal/
sip:1001 at 192.168.10.192:41080 SOFIA ROUTING
2010-02-09 21:49:32.907184 [DEBUG] switch_ivr_originate.c:66
(sofia/internal/sip:1001 at 192.168.10.192:41080) State Change CS_ROUTING ->
CS_CONSUME_MEDIA
2010-02-09 21:49:32.907184 [DEBUG] switch_core_session.c:999 Send signal
sofia/internal/sip:1001 at 192.168.10.192:41080 [BREAK]
2010-02-09 21:49:32.907184 [DEBUG] switch_core_state_machine.c:341
(sofia/internal/sip:1001 at 192.168.10.192:41080) State ROUTING going to sleep
2010-02-09 21:49:32.907184 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/sip:1001 at 192.168.10.192:41080) Running State Change
CS_CONSUME_MEDIA
2010-02-09 21:49:33.023071 [DEBUG] switch_core_state_machine.c:360
(sofia/internal/sip:1001 at 192.168.10.192:41080) State CONSUME_MEDIA
2010-02-09 21:49:33.023071 [DEBUG] switch_core_state_machine.c:360
(sofia/internal/sip:1001 at 192.168.10.192:41080) State CONSUME_MEDIA going to
sleep
2010-02-09 21:49:33.329319 [DEBUG] sofia.c:3727 Channel sofia/internal/
sip:1001 at 192.168.10.192:41080 entering state [terminated][488]
2010-02-09 21:49:33.329319 [NOTICE] sofia.c:4331 Hangup sofia/internal/
sip:1001 at 192.168.10.192:41080 [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION]
2010-02-09 21:49:33.329319 [DEBUG] switch_channel.c:1912 Send signal
sofia/internal/sip:1001 at 192.168.10.192:41080 [KILL]
2010-02-09 21:49:33.329319 [DEBUG] switch_core_session.c:999 Send signal
sofia/internal/sip:1001 at 192.168.10.192:41080 [BREAK]
2010-02-09 21:49:33.329319 [DEBUG] switch_core_state_machine.c:459
sofia/internal/sip:1001 at 192.168.10.192:41080 thread mismatch skipping state
handler.
2010-02-09 21:49:33.329319 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/sip:1001 at 192.168.10.192:41080) Running State Change
CS_HANGUP
2010-02-09 21:49:33.330075 [DEBUG] switch_core_state_machine.c:488
(sofia/internal/sip:1001 at 192.168.10.192:41080) State HANGUP
2010-02-09 21:49:33.330075 [DEBUG] mod_sofia.c:352 sofia/internal/
sip:1001 at 192.168.10.192:41080 Overriding SIP cause 488 with 488 from the
other leg
2010-02-09 21:49:33.330075 [DEBUG] mod_sofia.c:358 Channel sofia/internal/
sip:1001 at 192.168.10.192:41080 hanging up, cause: INCOMPATIBLE_DESTINATION
2010-02-09 21:49:33.330075 [DEBUG] switch_core_state_machine.c:46
sofia/internal/sip:1001 at 192.168.10.192:41080 Standard HANGUP, cause:
INCOMPATIBLE_DESTINATION
2010-02-09 21:49:33.330075 [DEBUG] switch_core_state_machine.c:488
(sofia/internal/sip:1001 at 192.168.10.192:41080) State HANGUP going to sleep
2010-02-09 21:49:33.330075 [DEBUG] switch_core_state_machine.c:333
(sofia/internal/sip:1001 at 192.168.10.192:41080) State Change CS_HANGUP ->
CS_REPORTING
2010-02-09 21:49:33.330075 [DEBUG] switch_core_session.c:999 Send signal
sofia/internal/sip:1001 at 192.168.10.192:41080 [BREAK]
2010-02-09 21:49:33.330075 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/sip:1001 at 192.168.10.192:41080) Running State Change
CS_REPORTING
2010-02-09 21:49:33.330075 [DEBUG] switch_core_state_machine.c:579
(sofia/internal/sip:1001 at 192.168.10.192:41080) State REPORTING
2010-02-09 21:49:33.330075 [DEBUG] switch_core_state_machine.c:53
sofia/internal/sip:1001 at 192.168.10.192:41080 Standard REPORTING, cause:
INCOMPATIBLE_DESTINATION
2010-02-09 21:49:33.330075 [DEBUG] switch_core_state_machine.c:579
(sofia/internal/sip:1001 at 192.168.10.192:41080) State REPORTING going to
sleep
2010-02-09 21:49:33.330075 [DEBUG] switch_core_state_machine.c:327
(sofia/internal/sip:1001 at 192.168.10.192:41080) State Change CS_REPORTING ->
CS_DESTROY
2010-02-09 21:49:33.330075 [DEBUG] switch_core_session.c:999 Send signal
sofia/internal/sip:1001 at 192.168.10.192:41080 [BREAK]
2010-02-09 21:49:33.330075 [DEBUG] switch_core_session.c:1136 Session 2
(sofia/internal/sip:1001 at 192.168.10.192:41080) Locked, Waiting on external
entities
2010-02-09 21:49:33.331072 [DEBUG] switch_ivr_originate.c:3009 Originate
Resulted in Error Cause: 88 [INCOMPATIBLE_DESTINATION]
2010-02-09 21:49:33.331072 [NOTICE] switch_core_session.c:1154 Session 2
(sofia/internal/sip:1001 at 192.168.10.192:41080) Ended
2010-02-09 21:49:33.331072 [NOTICE] switch_core_session.c:1156 Close Channel
sofia/internal/sip:1001 at 192.168.10.192:41080 [CS_DESTROY]
2010-02-09 21:49:33.331072 [DEBUG] switch_core_state_machine.c:423
(sofia/internal/sip:1001 at 192.168.10.192:41080) Running State Change
CS_DESTROY
2010-02-09 21:49:33.331072 [DEBUG] switch_core_state_machine.c:434
(sofia/internal/sip:1001 at 192.168.10.192:41080) State DESTROY
2010-02-09 21:49:33.331072 [DEBUG] mod_sofia.c:293 sofia/internal/
sip:1001 at 192.168.10.192:41080 SOFIA DESTROY
2010-02-09 21:49:33.331072 [DEBUG] switch_core_state_machine.c:60
sofia/internal/sip:1001 at 192.168.10.192:41080 Standard DESTROY
2010-02-09 21:49:33.331072 [DEBUG] switch_core_state_machine.c:434
(sofia/internal/sip:1001 at 192.168.10.192:41080) State DESTROY going to sleep
2010-02-09 21:49:33.331072 [ERR] switch_ivr_originate.c:2249 Cannot create
outgoing channel of type [user] cause: [INCOMPATIBLE_DESTINATION]
2010-02-09 21:49:33.331072 [DEBUG] switch_ivr_originate.c:3009 Originate
Resulted in Error Cause: 88 [INCOMPATIBLE_DESTINATION]
2010-02-09 21:49:33.331072 [INFO] mod_dptools.c:2294 Originate Failed.
Cause: INCOMPATIBLE_DESTINATION
EXECUTE sofia/internal/60002 at 192.168.10.30 answer()
2010-02-09 21:49:33.332075 [DEBUG] mod_dptools.c:658 sofia/internal/
60002 at 192.168.10.30 receive message [ANSWER]
2010-02-09 21:49:33.332075 [DEBUG] sofia_glue.c:2381 AUDIO RTP
[sofia/internal/60002 at 192.168.10.30] 192.168.10.30 port 27634 ->
192.168.10.131 port 25592 codec: 9 ms: 20
2010-02-09 21:49:33.332075 [DEBUG] switch_rtp.c:1167 Starting timer [soft]
160 bytes per 20ms
2010-02-09 21:49:33.333080 [DEBUG] mod_sofia.c:571 Local SDP sofia/internal/
60002 at 192.168.10.30:
v=0
o=FreeSWITCH 1265704739 1265704740 IN IP4 192.168.10.30
s=FreeSWITCH
c=IN IP4 192.168.10.30
t=0 0
m=audio 27634 RTP/AVP 9 101
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

2010-02-09 21:49:33.333080 [DEBUG] sofia.c:3727 Channel sofia/internal/
60002 at 192.168.10.30 entering state [completed][200]
2010-02-09 21:49:33.333080 [DEBUG] switch_core_session.c:645 Send signal
sofia/internal/60002 at 192.168.10.30 [BREAK]
2010-02-09 21:49:33.333080 [NOTICE] mod_dptools.c:658 Channel
[sofia/internal/60002 at 192.168.10.30] has been answered
2010-02-09 21:49:33.333080 [DEBUG] switch_channel.c:182 sofia/internal/
60002 at 192.168.10.30 receive message [AUDIO_SYNC]
EXECUTE sofia/internal/60002 at 192.168.10.30 sleep(1000)
2010-02-09 21:49:33.334081 [DEBUG] switch_channel.c:182 sofia/internal/
60002 at 192.168.10.30 receive message [AUDIO_SYNC]
2010-02-09 21:49:33.395069 [DEBUG] switch_rtp.c:2004 Correct ip/port
confirmed.
2010-02-09 21:49:33.455069 [DEBUG] sofia.c:3727 Channel sofia/internal/
60002 at 192.168.10.30 entering state [ready][200]
EXECUTE sofia/internal/60002 at 192.168.10.30 voicemail(default 192.168.10.30
1001)
2010-02-09 21:49:34.335070 [DEBUG] mod_voicemail.c:730 [default] rwlock
2010-02-09 21:49:34.370083 [DEBUG] switch_channel.c:182 sofia/internal/
60002 at 192.168.10.30 receive message [AUDIO_SYNC]
2010-02-09 21:49:34.475070 [DEBUG] switch_ivr_play_say.c:118 No language
specified - Using [en]
2010-02-09 21:49:34.475070 [DEBUG] switch_ivr_play_say.c:273 Handle
play-file:[voicemail/vm-person.wav] (en:en)
2010-02-09 21:49:34.523070 [WARNING] switch_core_file.c:177 Sample rate
doesn't match
2010-02-09 21:49:34.524073 [DEBUG] switch_ivr_play_say.c:1154 Codec
Activated L16 at 16000hz 1 channels 20ms
2010-02-09 21:49:34.526072 [DEBUG] switch_core_io.c:652 sofia/internal/
60002 at 192.168.10.30 receive message [TRANSCODING_NECESSARY]
2010-02-09 21:49:35.875063 [DEBUG] switch_ivr_play_say.c:1446 done playing
file
2010-02-09 21:49:35.995062 [DEBUG] switch_ivr_play_say.c:273 Handle
say:[1001] (en:en)
2010-02-09 21:49:36.027062 [WARNING] switch_core_file.c:177 Sample rate
doesn't match
2010-02-09 21:49:36.027062 [DEBUG] switch_ivr_play_say.c:1154 Codec
Activated L16 at 16000hz 1 channels 20ms
2010-02-09 21:49:36.028065 [DEBUG] switch_core_io.c:652 sofia/internal/
60002 at 192.168.10.30 receive message [TRANSCODING_NECESSARY]
2010-02-09 21:49:36.455072 [DEBUG] switch_ivr_play_say.c:1446 done playing
file
2010-02-09 21:49:36.460061 [WARNING] switch_core_file.c:177 Sample rate
doesn't match
2010-02-09 21:49:36.460061 [DEBUG] switch_ivr_play_say.c:1154 Codec
Activated L16 at 16000hz 1 channels 20ms
2010-02-09 21:49:36.461074 [DEBUG] switch_core_io.c:652 sofia/internal/
60002 at 192.168.10.30 receive message [TRANSCODING_NECESSARY]
2010-02-09 21:49:36.995060 [DEBUG] switch_ivr_play_say.c:1446 done playing
file
2010-02-09 21:49:36.995060 [WARNING] switch_core_file.c:177 Sample rate
doesn't match
2010-02-09 21:49:36.995060 [DEBUG] switch_ivr_play_say.c:1154 Codec
Activated L16 at 16000hz 1 channels 20ms
2010-02-09 21:49:36.995060 [DEBUG] switch_core_io.c:652 sofia/internal/
60002 at 192.168.10.30 receive message [TRANSCODING_NECESSARY]
2010-02-09 21:49:37.535061 [DEBUG] switch_ivr_play_say.c:1446 done playing
file
2010-02-09 21:49:37.535061 [WARNING] switch_core_file.c:177 Sample rate
doesn't match
2010-02-09 21:49:37.535061 [DEBUG] switch_ivr_play_say.c:1154 Codec
Activated L16 at 16000hz 1 channels 20ms
2010-02-09 21:49:37.535061 [DEBUG] switch_core_io.c:652 sofia/internal/
60002 at 192.168.10.30 receive message [TRANSCODING_NECESSARY]
2010-02-09 21:49:37.995060 [DEBUG] switch_ivr_play_say.c:1446 done playing
file
2010-02-09 21:49:38.095060 [DEBUG] switch_ivr_play_say.c:273 Handle
play-file:[voicemail/vm-not_available.wav] (en:en)
2010-02-09 21:49:38.118060 [WARNING] switch_core_file.c:177 Sample rate
doesn't match
2010-02-09 21:49:38.119063 [DEBUG] switch_ivr_play_say.c:1154 Codec
Activated L16 at 16000hz 1 channels 20ms
2010-02-09 21:49:38.120062 [DEBUG] switch_core_io.c:652 sofia/internal/
60002 at 192.168.10.30 receive message [TRANSCODING_NECESSARY]
2010-02-09 21:49:39.075057 [DEBUG] switch_ivr_play_say.c:1446 done playing
file
2010-02-09 21:49:39.175057 [DEBUG] switch_ivr_play_say.c:118 No language
specified - Using [en]
2010-02-09 21:49:39.177086 [DEBUG] switch_ivr_play_say.c:273 Handle
play-file:[voicemail/vm-record_message.wav] (en:en)
2010-02-09 21:49:39.181967 [WARNING] switch_core_file.c:177 Sample rate
doesn't match
2010-02-09 21:49:39.181967 [DEBUG] switch_ivr_play_say.c:1154 Codec
Activated L16 at 16000hz 1 channels 20ms
2010-02-09 21:49:39.192068 [DEBUG] switch_core_io.c:652 sofia/internal/
60002 at 192.168.10.30 receive message [TRANSCODING_NECESSARY]
2010-02-09 21:49:40.999151 [NOTICE] sofia.c:329 Hangup sofia/internal/
60002 at 192.168.10.30 [CS_EXECUTE] [NORMAL_CLEARING]
2010-02-09 21:49:40.999151 [DEBUG] switch_channel.c:1912 Send signal
sofia/internal/60002 at 192.168.10.30 [KILL]
2010-02-09 21:49:40.999151 [DEBUG] switch_core_session.c:999 Send signal
sofia/internal/60002 at 192.168.10.30 [BREAK]
2010-02-09 21:49:40.999151 [DEBUG] switch_core_state_machine.c:459
sofia/internal/60002 at 192.168.10.30 thread mismatch skipping state handler.
2010-02-09 21:49:41.015049 [DEBUG] switch_ivr_play_say.c:1446 done playing
file
2010-02-09 21:49:41.114049 [DEBUG] switch_core_state_machine.c:348
(sofia/internal/60002 at 192.168.10.30) State EXECUTE going to sleep
2010-02-09 21:49:41.114049 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/60002 at 192.168.10.30) Running State Change CS_HANGUP
2010-02-09 21:49:41.114049 [DEBUG] switch_core_state_machine.c:488
(sofia/internal/60002 at 192.168.10.30) State HANGUP
2010-02-09 21:49:41.114049 [DEBUG] mod_sofia.c:352 sofia/internal/
60002 at 192.168.10.30 Overriding SIP cause 480 with 488 from the other leg
2010-02-09 21:49:41.114049 [DEBUG] mod_sofia.c:358 Channel sofia/internal/
60002 at 192.168.10.30 hanging up, cause: NORMAL_CLEARING
2010-02-09 21:49:41.114049 [DEBUG] switch_core_state_machine.c:46
sofia/internal/60002 at 192.168.10.30 Standard HANGUP, cause: NORMAL_CLEARING
2010-02-09 21:49:41.114049 [DEBUG] switch_core_state_machine.c:488
(sofia/internal/60002 at 192.168.10.30) State HANGUP going to sleep
2010-02-09 21:49:41.114049 [DEBUG] switch_core_state_machine.c:333
(sofia/internal/60002 at 192.168.10.30) State Change CS_HANGUP -> CS_REPORTING
2010-02-09 21:49:41.114049 [DEBUG] switch_core_session.c:999 Send signal
sofia/internal/60002 at 192.168.10.30 [BREAK]
2010-02-09 21:49:41.114049 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/60002 at 192.168.10.30) Running State Change CS_REPORTING
2010-02-09 21:49:41.114049 [DEBUG] switch_core_state_machine.c:579
(sofia/internal/60002 at 192.168.10.30) State REPORTING
2010-02-09 21:49:41.129049 [DEBUG] switch_core_state_machine.c:53
sofia/internal/60002 at 192.168.10.30 Standard REPORTING, cause:
NORMAL_CLEARING
2010-02-09 21:49:41.129049 [DEBUG] switch_core_state_machine.c:579
(sofia/internal/60002 at 192.168.10.30) State REPORTING going to sleep
2010-02-09 21:49:41.129049 [DEBUG] switch_core_state_machine.c:327
(sofia/internal/60002 at 192.168.10.30) State Change CS_REPORTING -> CS_DESTROY
2010-02-09 21:49:41.129049 [DEBUG] switch_core_session.c:999 Send signal
sofia/internal/60002 at 192.168.10.30 [BREAK]
2010-02-09 21:49:41.129049 [DEBUG] switch_core_session.c:1136 Session 1
(sofia/internal/60002 at 192.168.10.30) Locked, Waiting on external entities
2010-02-09 21:49:41.129049 [NOTICE] switch_core_session.c:1154 Session 1
(sofia/internal/60002 at 192.168.10.30) Ended
2010-02-09 21:49:41.129049 [NOTICE] switch_core_session.c:1156 Close Channel
sofia/internal/60002 at 192.168.10.30 [CS_DESTROY]
2010-02-09 21:49:41.129049 [DEBUG] switch_core_state_machine.c:423
(sofia/internal/60002 at 192.168.10.30) Running State Change CS_DESTROY
2010-02-09 21:49:41.129049 [DEBUG] switch_core_state_machine.c:434
(sofia/internal/60002 at 192.168.10.30) State DESTROY
2010-02-09 21:49:41.129049 [DEBUG] mod_sofia.c:293 sofia/internal/
60002 at 192.168.10.30 SOFIA DESTROY
2010-02-09 21:49:41.129049 [DEBUG] switch_core_state_machine.c:60
sofia/internal/60002 at 192.168.10.30 Standard DESTROY
2010-02-09 21:49:41.129049 [DEBUG] switch_core_state_machine.c:434
(sofia/internal/60002 at 192.168.10.30) State DESTROY going to sleep






On 9 February 2010 21:18, Brian West <brian at freeswitch.org> wrote:

> What you're saying makes little or no sense to me even on 1.0.4, Can you
> pastebin your logs?
>
> /b
>
> On Feb 9, 2010, at 3:15 PM, Bruce Hopkins wrote:
>
> > OK, many thanks for the extremely swift response Brian.
> >
> > I will try to get up and running as soon as I can with 1.0.5 and see if
> the issue goes away.
> >
> > thanks again
> > Bruce
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100209/f036726e/attachment-0002.html 

From brian at freeswitch.org  Tue Feb  9 14:00:33 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 9 Feb 2010 16:00:33 -0600
Subject: [Freeswitch-users] g.722 --> SPEEX/16000 or SPEEX/32000
	transcoding issue
In-Reply-To: <d03a10e01002091355n1934627t3902e0d8af5c7dc3@mail.gmail.com>
References: <d03a10e01002091203qde4caabi1a7565929e60918f@mail.gmail.com>
	<03A2F6D0-754A-4067-A106-9D91B0D68A92@freeswitch.org>
	<d03a10e01002091315n5fa9242eoe2a86c4edee1501d@mail.gmail.com>
	<F167991C-151B-4D40-BF27-80A4C3A44D1E@freeswitch.org>
	<d03a10e01002091355n1934627t3902e0d8af5c7dc3@mail.gmail.com>
Message-ID: <51A80F87-F2B9-477E-8D14-99C40E24413D@freeswitch.org>

You need to allow SPEEX at 8000h,SPEEX at 16000h,SPEEX at 32000h

/b

On Feb 9, 2010, at 3:55 PM, Bruce Hopkins wrote:

> Willdo,
> 
> To clarify in brief though, the scenario which occurs and causes the call to fail is:
> 
> SIP client  1 (g.722 enabled only ) -----> INVITE (with SDP offer: Media Attribute (a): rtpmap:9 g722/8000 ) -->FreeSWITCH
> 
> ---> INVITE (with SDP offer including a bunch of codecs including rtpmap: rtpmap:98 SPEEX/8000 but crucially not including SPEEX/16000 or SPEEX/32000)
> 
> ---> SIP client 2 (with only SPEEX/16000 or SPEEX/32000 enabled).
> 
> The second SIP client does not get offered a codec it can accept, so SIP client 1 is sent a method 488 "Not Acceptable Here" message and the calling party gets directed to the voicemail for the other SIP client.
> 
> By contrast, there is no problem calling SPEEX/32000 --> SPEEX/32000 or calling SPEEX/16000 --> SPEEX/16000.
> 
> there is also no problem calling SPEEX/32000 --> g.722/8000.
> 
> I am wondering if the problem is that FreeSWITCH is interpreting g.722 as being a narrowband (8kHz sample rate) codec, due to the historic anomaly of it presenting g722/8000 in the SDP even though it in fact uses 16kHz sampling, and for that reason not wanting to offer a 16kHz sample rate codec to the second SIP client?
> 
> I suggest this as I also found trying to call alaw --> SPEEX/16000 does not work, for example.




From jbrucehopkins at gmail.com  Tue Feb  9 14:04:51 2010
From: jbrucehopkins at gmail.com (Bruce Hopkins)
Date: Tue, 9 Feb 2010 22:04:51 +0000
Subject: [Freeswitch-users] g.722 --> SPEEX/16000 or SPEEX/32000
	transcoding issue
In-Reply-To: <d03a10e01002091355n1934627t3902e0d8af5c7dc3@mail.gmail.com>
References: <d03a10e01002091203qde4caabi1a7565929e60918f@mail.gmail.com>
	<03A2F6D0-754A-4067-A106-9D91B0D68A92@freeswitch.org>
	<d03a10e01002091315n5fa9242eoe2a86c4edee1501d@mail.gmail.com>
	<F167991C-151B-4D40-BF27-80A4C3A44D1E@freeswitch.org>
	<d03a10e01002091355n1934627t3902e0d8af5c7dc3@mail.gmail.com>
Message-ID: <d03a10e01002091404i5e7aefaam7235bef68828d42b@mail.gmail.com>

I think I see the problem:

In the log is the following:

2010-02-09 21:49:36.995060 [WARNING] switch_core_file.c:177 Sample rate
doesn't match

However, surely this is incorrect, as g.722 and SPEEX/16000 in fact both do
use the same sample rate of 16kHz

g.722 might not look like it from the SDP which announces g722/8000 - but
this is a historical error in the RFC.

Bruce

On 9 February 2010 21:55, Bruce Hopkins <jbrucehopkins at gmail.com> wrote:

> Willdo,
>
> To clarify in brief though, the scenario which occurs and causes the call
> to fail is:
>
> SIP client  1 (g.722 enabled only ) -----> INVITE (with SDP offer: Media
> Attribute (a): rtpmap:9 g722/8000 ) -->FreeSWITCH
>
> ---> INVITE (with SDP offer including a bunch of codecs including rtpmap:
> rtpmap:98 SPEEX/8000 but crucially *not* including SPEEX/16000 or
> SPEEX/32000)
>
> ---> SIP client 2 (with only SPEEX/16000 or SPEEX/32000 enabled).
>
> The second SIP client does not get offered a codec it can accept, so SIP
> client 1 is sent a method 488 "Not Acceptable Here" message and the calling
> party gets directed to the voicemail for the other SIP client.
>
> By contrast, there is no problem calling SPEEX/32000 --> SPEEX/32000 or
> calling SPEEX/16000 --> SPEEX/16000.
>
> there is also no problem calling SPEEX/32000 --> g.722/8000.
>
> I am wondering if the problem is that FreeSWITCH is interpreting g.722 as
> being a narrowband (8kHz sample rate) codec, due to the historic anomaly of
> it presenting g722/8000 in the SDP even though it in fact uses 16kHz
> sampling, and for that reason not wanting to offer a 16kHz sample rate codec
> to the second SIP client?
>
> I suggest this as I also found trying to call alaw --> SPEEX/16000 does not
> work, for example.
>
>
>
> Here is the log file for the scenario which does not work (g.722 client
> trying to call Speex wideband client).  Please let me know if a Wireshark
> trace would be helpful.
>
> 2010-02-09 21:49:32.784387 [DEBUG] sofia.c:5224 0 acls to check for proxy
> 2010-02-09 21:49:32.784387 [DEBUG] sofia.c:5242 network ip is a proxy [0]
> 2010-02-09 21:49:32.784387 [DEBUG] sofia.c:5270 IP 192.168.10.131 Rejected
> by acl "domains". Falling back to Digest auth.
> 2010-02-09 21:49:32.791074 [DEBUG] sofia.c:5224 0 acls to check for proxy
> 2010-02-09 21:49:32.791074 [DEBUG] sofia.c:5242 network ip is a proxy [0]
> 2010-02-09 21:49:32.791074 [DEBUG] sofia.c:5270 IP 192.168.10.131 Rejected
> by acl "domains". Falling back to Digest auth.
> 2010-02-09 21:49:32.794074 [NOTICE] switch_channel.c:613 New Channel
> sofia/internal/60002 at 192.168.10.30 [66610a8c-aec9-48ca-9b67-aafdcd3e3d08]
> 2010-02-09 21:49:32.794074 [DEBUG] sofia.c:3727 Channel sofia/internal/
> 60002 at 192.168.10.30 entering state [received][100]
> 2010-02-09 21:49:32.794074 [DEBUG] sofia.c:3738 Remote SDP:
> v=0
> o=- 5 2 IN IP4 192.168.10.131
> s=CounterPath eyeBeam 1.5
> c=IN IP4 192.168.10.131
> t=0 0
> m=audio 25592 RTP/AVP 9 101
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
>
> 2010-02-09 21:49:32.794074 [DEBUG] sofia_glue.c:3305 Audio Codec Compare
> [G722:9:8000:20]/[G722:9:8000:20]
> 2010-02-09 21:49:32.795079 [DEBUG] sofia_glue.c:2143 Set Codec
> sofia/internal/60002 at 192.168.10.30 G722/8000 20 ms 160 samples
> 2010-02-09 21:49:32.795079 [DEBUG] sofia_glue.c:3261 Set 2833 dtmf payload
> to 101
> 2010-02-09 21:49:32.795079 [DEBUG] sofia.c:3885 (sofia/internal/
> 60002 at 192.168.10.30) State Change CS_NEW -> CS_INIT
> 2010-02-09 21:49:32.795079 [DEBUG] switch_core_session.c:999 Send signal
> sofia/internal/60002 at 192.168.10.30 [BREAK]
> 2010-02-09 21:49:32.795079 [DEBUG] switch_core_state_machine.c:314
> (sofia/internal/60002 at 192.168.10.30) Running State Change CS_INIT
> 2010-02-09 21:49:32.795079 [DEBUG] switch_core_state_machine.c:338
> (sofia/internal/60002 at 192.168.10.30) State INIT
> 2010-02-09 21:49:32.795079 [DEBUG] mod_sofia.c:83 sofia/internal/
> 60002 at 192.168.10.30 SOFIA INIT
> 2010-02-09 21:49:32.795079 [DEBUG] mod_sofia.c:111 (sofia/internal/
> 60002 at 192.168.10.30) State Change CS_INIT -> CS_ROUTING
> 2010-02-09 21:49:32.795079 [DEBUG] switch_core_session.c:999 Send signal
> sofia/internal/60002 at 192.168.10.30 [BREAK]
> 2010-02-09 21:49:32.795079 [DEBUG] switch_core_state_machine.c:338
> (sofia/internal/60002 at 192.168.10.30) State INIT going to sleep
> 2010-02-09 21:49:32.795079 [DEBUG] switch_core_state_machine.c:314
> (sofia/internal/60002 at 192.168.10.30) Running State Change CS_ROUTING
> 2010-02-09 21:49:32.795079 [DEBUG] switch_core_state_machine.c:341
> (sofia/internal/60002 at 192.168.10.30) State ROUTING
> 2010-02-09 21:49:32.795079 [DEBUG] mod_sofia.c:132 sofia/internal/
> 60002 at 192.168.10.30 SOFIA ROUTING
> 2010-02-09 21:49:32.795079 [DEBUG] switch_core_state_machine.c:78
> sofia/internal/60002 at 192.168.10.30 Standard ROUTING
> 2010-02-09 21:49:32.795079 [INFO] mod_dialplan_xml.c:408 Processing
> 60002->1001 in context default
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->unloop]
> continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (PASS) [unloop]
> ${unroll_loops}(true) =~ /^true$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [unloop]
> ${sip_looped_call}() =~ /^true$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
> [default->tod_example] continue=true
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (PASS) [tod_example]
> ${strftime(%w)}(2) =~ /^([1-5])$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [tod_example]
> ${strftime(%H%M)}(2149) =~ /^((09|1[0-7])[0-5][0-9]|1800)$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
> [default->global-intercept] continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL)
> [global-intercept] destination_number(1001) =~ /^\*886$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
> [default->group-intercept] continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL)
> [group-intercept] destination_number(1001) =~ /^\*8$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
> [default->intercept-ext] continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [intercept-ext]
> destination_number(1001) =~ /^\*\*(\d+)$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->redial]
> continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [redial]
> destination_number(1001) =~ /^\*870$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->global]
> continue=true
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [global]
> ${network_addr}(192.168.10.131) =~ /^$/ break=never
> Dialplan: sofia/internal/60002 at 192.168.10.30 ANTI-Action
> set(use_profile=${cond(${acl(${network_addr} rfc1918)} == true ? nat :
> default)})
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (PASS) [global]
> ${numbering_plan}() =~ /^$/ break=never
> Dialplan: sofia/internal/60002 at 192.168.10.30 Action set_user(default@${domain_name})
>
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [global]
> ${call_debug}(false) =~ /^true$/ break=never
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [global]
> ${sip_has_crypto}() =~ /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/
> break=never
> Dialplan: sofia/internal/60002 at 192.168.10.30 Absolute Condition [global]
> Dialplan: sofia/internal/60002 at 192.168.10.30 Action
> db(insert/${domain_name}-spymap/${caller_id_number}/${uuid})
> Dialplan: sofia/internal/60002 at 192.168.10.30 Action
> db(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number})
>
> Dialplan: sofia/internal/60002 at 192.168.10.30 Action
> db(insert/${domain_name}-last_dial/global/${uuid})
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
> [default->snom-demo-2] continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [snom-demo-2]
> destination_number(1001) =~ /^\*9001$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
> [default->snom-demo-1] continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [snom-demo-1]
> destination_number(1001) =~ /^\*9000$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->eavesdrop]
> continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [eavesdrop]
> destination_number(1001) =~ /^\*88(.*)$|^\*0(.*)$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->eavesdrop]
> continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [eavesdrop]
> destination_number(1001) =~ /^\*779$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
> [default->call_privacy] continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [call_privacy]
> destination_number(1001) =~ /^\*67(\d+)$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
> [default->call_return] continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [call_return]
> destination_number(1001) =~ /^\*69$|^869$|^lcr$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->del-group]
> continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [del-group]
> destination_number(1001) =~ /^\*80(\d{2})$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->add-group]
> continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [add-group]
> destination_number(1001) =~ /^\*81(\d{2})$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
> [default->call-group-simo] continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL)
> [call-group-simo] destination_number(1001) =~ /^\*82(\d{2})$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
> [default->call-group-order] continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL)
> [call-group-order] destination_number(1001) =~ /^\*83(\d{2})$/
> break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
> [default->extension-intercom] continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL)
> [extension-intercom] destination_number(1001) =~ /^\*8(\d{4})$/
> break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
> [default->send_to_voicemail_5digits] continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL)
> [send_to_voicemail_5digits] destination_number(1001) =~ /^\*99(\d{5})$/
> break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
> [default->send_to_voicemail_4digits] continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL)
> [send_to_voicemail_4digits] destination_number(1001) =~ /^\*99(\d{4})$/
> break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
> [default->send_to_voicemail_3digits] continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL)
> [send_to_voicemail_3digits] destination_number(1001) =~ /^\*99(\d{3})$/
> break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->pizza_demo]
> continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [pizza_demo]
> destination_number(1001) =~ /^(pizza|74992)$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->2001]
> continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [2001]
> destination_number(1001) =~ /^2001$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->Call via
> asterisk-pbx1] continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [Call via
> asterisk-pbx1] destination_number(1001) =~ /269065/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->pizza_demo]
> continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [pizza_demo]
> destination_number(1001) =~ /^(pizza|74992)$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->5002]
> continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [5002]
> destination_number(1001) =~ /^5002$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->7002]
> continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [7002]
> destination_number(1001) =~ /^7002$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->DISA]
> continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [DISA]
> destination_number(1001) =~ /^\*(3472)$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->Recordings]
> continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [Recordings]
> destination_number(1001) =~ /^\*(732673)$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->7002.park]
> continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [7002.park]
> destination_number(1001) =~ /^\*7002$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
> [default->group_dial_sales] continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL)
> [group_dial_sales] destination_number(1001) =~ /^\*2000$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
> [default->group_dial_support] continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL)
> [group_dial_support] destination_number(1001) =~ /^\*2001$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
> [default->group_dial_billing] continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL)
> [group_dial_billing] destination_number(1001) =~ /^\*2002$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->vmain2]
> continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [vmain2]
> destination_number(1001) =~ /^vmain2$|^\*97$|^\*4000$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->vmain]
> continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [vmain]
> destination_number(1001) =~ /^vmain$|^\*98$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->sip_uri]
> continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [sip_uri]
> destination_number(1001) =~ /^sip:(.*)$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
> [default->nb_conferences] continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [nb_conferences]
> destination_number(1001) =~ /^\*(30\d{2})$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
> [default->wb_conferences] continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [wb_conferences]
> destination_number(1001) =~ /^\*(31\d{2})$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
> [default->uwb_conferences] continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL)
> [uwb_conferences] destination_number(1001) =~ /^\*(32\d{2})$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
> [default->cdquality_conferences] continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL)
> [cdquality_conferences] destination_number(1001) =~ /^\*(33\d{2})$/
> break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
> [default->freeswitch_public_conf_via_sip] continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL)
> [freeswitch_public_conf_via_sip] destination_number(1001) =~
> /^\*9(888|1616|3232)$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
> [default->mad_boss_intercom] continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL)
> [mad_boss_intercom] destination_number(1001) =~ /^\*0911$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
> [default->mad_boss_intercom] continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL)
> [mad_boss_intercom] destination_number(1001) =~ /^\*0912$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->mad_boss]
> continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [mad_boss]
> destination_number(1001) =~ /^\*0913$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->ivr_demo]
> continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [ivr_demo]
> destination_number(1001) =~ /^\*5000$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
> [default->dynamic_conference] continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL)
> [dynamic_conference] destination_number(1001) =~ /^\*5001$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
> [default->rtp_multicast_page] continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL)
> [rtp_multicast_page] destination_number(1001) =~ /^\*pagegroup$|^\*7243/
> break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->park]
> continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [park]
> destination_number(1001) =~ /^\*5900$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->unpark]
> continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [unpark]
> destination_number(1001) =~ /^\*5901$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->park]
> continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (PASS) [park]
> source(mod_sofia) =~ /mod_sofia/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [park]
> destination_number(1001) =~ /park\+(\d+)/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->unpark]
> continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (PASS) [unpark]
> source(mod_sofia) =~ /mod_sofia/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [unpark]
> destination_number(1001) =~ /^parking$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->park]
> continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (PASS) [park]
> source(mod_sofia) =~ /mod_sofia/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [park]
> destination_number(1001) =~ /callpark/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->unpark]
> continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (PASS) [unpark]
> source(mod_sofia) =~ /mod_sofia/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [unpark]
> destination_number(1001) =~ /pickup/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->wait]
> continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [wait]
> destination_number(1001) =~ /^wait$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
> [default->fax_receive] continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [fax_receive]
> destination_number(1001) =~ /^\*9978$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
> [default->fax_transmit] continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [fax_transmit]
> destination_number(1001) =~ /^\*9979$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
> [default->ringback_180] continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [ringback_180]
> destination_number(1001) =~ /^\*9980$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
> [default->ringback_183_uk_ring] continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL)
> [ringback_183_uk_ring] destination_number(1001) =~ /^\*9981$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
> [default->ringback_183_music_ring] continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL)
> [ringback_183_music_ring] destination_number(1001) =~ /^\*9982$/
> break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
> [default->ringback_post_answer_uk_ring] continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL)
> [ringback_post_answer_uk_ring] destination_number(1001) =~ /^\*9983$/
> break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
> [default->ringback_post_answer_music] continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL)
> [ringback_post_answer_music] destination_number(1001) =~ /^\*9984$/
> break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->ClueCon]
> continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [ClueCon]
> destination_number(1001) =~ /^\*9991$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->show_info]
> continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [show_info]
> destination_number(1001) =~ /^\*9992$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
> [default->video_record] continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [video_record]
> destination_number(1001) =~ /^\*9993$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
> [default->video_playback] continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [video_playback]
> destination_number(1001) =~ /^\*9994$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->delay_echo]
> continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [delay_echo]
> destination_number(1001) =~ /^\*9995$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->echo]
> continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [echo]
> destination_number(1001) =~ /^\*9996$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->milliwatt]
> continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [milliwatt]
> destination_number(1001) =~ /^\*9997$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
> [default->tone_stream] continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [tone_stream]
> destination_number(1001) =~ /^\*9998$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
> [default->zrtp_enrollement] continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL)
> [zrtp_enrollement] destination_number(1001) =~ /^\*9787$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing [default->hold_music]
> continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (FAIL) [hold_music]
> destination_number(1001) =~ /^\*9999$/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 parsing
> [default->Local_Extension] continue=false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Regex (PASS)
> [Local_Extension] destination_number(1001) =~
> /(^\d{6}$|\d{5}$|^\d{4}$|^\d{3}$)/ break=on-false
> Dialplan: sofia/internal/60002 at 192.168.10.30 Action
> set(dialed_extension=1001)
> Dialplan: sofia/internal/60002 at 192.168.10.30 Action
> export(dialed_extension=1001)
> Dialplan: sofia/internal/60002 at 192.168.10.30 Action bind_meta_app(1 b s
> execute_extension::dx XML features)
> Dialplan: sofia/internal/60002 at 192.168.10.30 Action bind_meta_app(2 b s
> record_session::/usr/local/freeswitch/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav)
>
> Dialplan: sofia/internal/60002 at 192.168.10.30 Action bind_meta_app(3 b s
> execute_extension::cf XML features)
> Dialplan: sofia/internal/60002 at 192.168.10.30 Action
> set(ringback=${us-ring})
> Dialplan: sofia/internal/60002 at 192.168.10.30 Action
> set(transfer_ringback=local_stream://moh)
> Dialplan: sofia/internal/60002 at 192.168.10.30 Action set(call_timeout=30)
> Dialplan: sofia/internal/60002 at 192.168.10.30 Action
> set(hangup_after_bridge=true)
> Dialplan: sofia/internal/60002 at 192.168.10.30 Action
> set(continue_on_fail=true)
> Dialplan: sofia/internal/60002 at 192.168.10.30 Action
> db(insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number})
>
> Dialplan: sofia/internal/60002 at 192.168.10.30 Action
> db(insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid})
> Dialplan: sofia/internal/60002 at 192.168.10.30 Action
> set(called_party_callgroup=${user_data(${dialed_extension}@${domain_name}
> var callgroup)})
> Dialplan: sofia/internal/60002 at 192.168.10.30 Action
> db(insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid})
> Dialplan: sofia/internal/60002 at 192.168.10.30 Action
> bridge(user/${dialed_extension}@${domain_name})
> Dialplan: sofia/internal/60002 at 192.168.10.30 Action answer()
> Dialplan: sofia/internal/60002 at 192.168.10.30 Action sleep(1000)
> Dialplan: sofia/internal/60002 at 192.168.10.30 Action voicemail(default
> ${domain_name} ${dialed_extension})
> 2010-02-09 21:49:32.798073 [DEBUG] switch_core_state_machine.c:122
> (sofia/internal/60002 at 192.168.10.30) State Change CS_ROUTING -> CS_EXECUTE
> 2010-02-09 21:49:32.798073 [DEBUG] switch_core_session.c:999 Send signal
> sofia/internal/60002 at 192.168.10.30 [BREAK]
> 2010-02-09 21:49:32.798073 [DEBUG] switch_core_state_machine.c:341
> (sofia/internal/60002 at 192.168.10.30) State ROUTING going to sleep
> 2010-02-09 21:49:32.798073 [DEBUG] switch_core_state_machine.c:314
> (sofia/internal/60002 at 192.168.10.30) Running State Change CS_EXECUTE
> 2010-02-09 21:49:32.798073 [DEBUG] switch_core_state_machine.c:348
> (sofia/internal/60002 at 192.168.10.30) State EXECUTE
> 2010-02-09 21:49:32.798073 [DEBUG] mod_sofia.c:181 sofia/internal/
> 60002 at 192.168.10.30 SOFIA EXECUTE
> 2010-02-09 21:49:32.798073 [DEBUG] switch_core_state_machine.c:159
> sofia/internal/60002 at 192.168.10.30 Standard EXECUTE
> EXECUTE sofia/internal/60002 at 192.168.10.30 set(use_profile=default)
> 2010-02-09 21:49:32.798073 [DEBUG] mod_dptools.c:768 sofia/internal/
> 60002 at 192.168.10.30 SET [use_profile]=[default]
> EXECUTE sofia/internal/60002 at 192.168.10.30 set_user(default at 192.168.10.30)
> EXECUTE sofia/internal/60002 at 192.168.10.30db(insert/192.168.10.30-spymap/60002/66610a8c-aec9-48ca-9b67-aafdcd3e3d08)
> EXECUTE sofia/internal/60002 at 192.168.10.30db(insert/192.168.10.30-last_dial/60002/1001)
> EXECUTE sofia/internal/60002 at 192.168.10.30db(insert/192.168.10.30-last_dial/global/66610a8c-aec9-48ca-9b67-aafdcd3e3d08)
> EXECUTE sofia/internal/60002 at 192.168.10.30 set(dialed_extension=1001)
> 2010-02-09 21:49:32.857126 [DEBUG] mod_dptools.c:768 sofia/internal/
> 60002 at 192.168.10.30 SET [dialed_extension]=[1001]
> EXECUTE sofia/internal/60002 at 192.168.10.30 export(dialed_extension=1001)
> 2010-02-09 21:49:32.858075 [DEBUG] mod_dptools.c:851 EXPORT
> [dialed_extension]=[1001]
> EXECUTE sofia/internal/60002 at 192.168.10.30 bind_meta_app(1 b s
> execute_extension::dx XML features)
> 2010-02-09 21:49:32.858075 [INFO] switch_ivr_async.c:2174 Bound B-Leg: 1
> execute_extension::dx XML features
> EXECUTE sofia/internal/60002 at 192.168.10.30 bind_meta_app(2 b s
> record_session::/usr/local/freeswitch/recordings/60002.2010-02-09-21-49-32.wav)
> 2010-02-09 21:49:32.858075 [INFO] switch_ivr_async.c:2174 Bound B-Leg: 2
> record_session::/usr/local/freeswitch/recordings/60002.2010-02-09-21-49-32.wav
> EXECUTE sofia/internal/60002 at 192.168.10.30 bind_meta_app(3 b s
> execute_extension::cf XML features)
> 2010-02-09 21:49:32.859074 [INFO] switch_ivr_async.c:2174 Bound B-Leg: 3
> execute_extension::cf XML features
> EXECUTE sofia/internal/60002 at 192.168.10.30 set(ringback=%(2000, 4000,
> 440.0, 480.0))
> 2010-02-09 21:49:32.859074 [DEBUG] mod_dptools.c:768 sofia/internal/
> 60002 at 192.168.10.30 SET [ringback]=[%(2000, 4000, 440.0, 480.0)]
> EXECUTE sofia/internal/60002 at 192.168.10.30set(transfer_ringback=local_stream://moh)
> 2010-02-09 21:49:32.859074 [DEBUG] mod_dptools.c:768 sofia/internal/
> 60002 at 192.168.10.30 SET [transfer_ringback]=[local_stream://moh]
> EXECUTE sofia/internal/60002 at 192.168.10.30 set(call_timeout=30)
> 2010-02-09 21:49:32.860074 [DEBUG] mod_dptools.c:768 sofia/internal/
> 60002 at 192.168.10.30 SET [call_timeout]=[30]
> EXECUTE sofia/internal/60002 at 192.168.10.30 set(hangup_after_bridge=true)
> 2010-02-09 21:49:32.860074 [DEBUG] mod_dptools.c:768 sofia/internal/
> 60002 at 192.168.10.30 SET [hangup_after_bridge]=[true]
> EXECUTE sofia/internal/60002 at 192.168.10.30 set(continue_on_fail=true)
> 2010-02-09 21:49:32.860074 [DEBUG] mod_dptools.c:768 sofia/internal/
> 60002 at 192.168.10.30 SET [continue_on_fail]=[true]
> EXECUTE sofia/internal/60002 at 192.168.10.30db(insert/192.168.10.30-call_return/1001/60002)
> EXECUTE sofia/internal/60002 at 192.168.10.30db(insert/192.168.10.30-last_dial_ext/1001/66610a8c-aec9-48ca-9b67-aafdcd3e3d08)
> EXECUTE sofia/internal/60002 at 192.168.10.30 set(called_party_callgroup=)
> 2010-02-09 21:49:32.870074 [DEBUG] mod_dptools.c:768 sofia/internal/
> 60002 at 192.168.10.30 SET [called_party_callgroup]=[UNDEF]
> EXECUTE sofia/internal/60002 at 192.168.10.30db(insert/192.168.10.30-last_dial//66610a8c-aec9-48ca-9b67-aafdcd3e3d08)
> EXECUTE sofia/internal/60002 at 192.168.10.30 bridge(user/1001 at 192.168.10.30)
> 2010-02-09 21:49:32.905073 [DEBUG] switch_ivr_originate.c:1735 variable
> string 0 = [presence_id=1001 at 192.168.10.30]
> 2010-02-09 21:49:32.905073 [NOTICE] switch_channel.c:613 New Channel
> sofia/internal/sip:1001 at 192.168.10.192:41080[df7cf235-f0e4-406a-83a5-dd2d681bb278]
> 2010-02-09 21:49:32.905073 [DEBUG] mod_sofia.c:3142 (sofia/internal/
> sip:1001 at 192.168.10.192:41080) State Change CS_NEW -> CS_INIT
> 2010-02-09 21:49:32.905073 [DEBUG] switch_core_session.c:999 Send signal
> sofia/internal/sip:1001 at 192.168.10.192:41080 [BREAK]
> 2010-02-09 21:49:32.906075 [DEBUG] switch_core_state_machine.c:314
> (sofia/internal/sip:1001 at 192.168.10.192:41080) Running State Change
> CS_INIT
> 2010-02-09 21:49:32.906075 [DEBUG] switch_core_state_machine.c:338
> (sofia/internal/sip:1001 at 192.168.10.192:41080) State INIT
> 2010-02-09 21:49:32.906075 [DEBUG] mod_sofia.c:83 sofia/internal/
> sip:1001 at 192.168.10.192:41080 SOFIA INIT
> 2010-02-09 21:49:32.907184 [DEBUG] mod_sofia.c:111 (sofia/internal/
> sip:1001 at 192.168.10.192:41080) State Change CS_INIT -> CS_ROUTING
> 2010-02-09 21:49:32.907184 [DEBUG] switch_core_session.c:999 Send signal
> sofia/internal/sip:1001 at 192.168.10.192:41080 [BREAK]
> 2010-02-09 21:49:32.907184 [DEBUG] sofia.c:3727 Channel sofia/internal/
> sip:1001 at 192.168.10.192:41080 entering state [calling][0]
> 2010-02-09 21:49:32.907184 [DEBUG] switch_core_state_machine.c:338
> (sofia/internal/sip:1001 at 192.168.10.192:41080) State INIT going to sleep
> 2010-02-09 21:49:32.907184 [DEBUG] switch_core_state_machine.c:314
> (sofia/internal/sip:1001 at 192.168.10.192:41080) Running State Change
> CS_ROUTING
> 2010-02-09 21:49:32.907184 [DEBUG] switch_core_state_machine.c:341
> (sofia/internal/sip:1001 at 192.168.10.192:41080) State ROUTING
> 2010-02-09 21:49:32.907184 [DEBUG] mod_sofia.c:132 sofia/internal/
> sip:1001 at 192.168.10.192:41080 SOFIA ROUTING
> 2010-02-09 21:49:32.907184 [DEBUG] switch_ivr_originate.c:66
> (sofia/internal/sip:1001 at 192.168.10.192:41080) State Change CS_ROUTING ->
> CS_CONSUME_MEDIA
> 2010-02-09 21:49:32.907184 [DEBUG] switch_core_session.c:999 Send signal
> sofia/internal/sip:1001 at 192.168.10.192:41080 [BREAK]
> 2010-02-09 21:49:32.907184 [DEBUG] switch_core_state_machine.c:341
> (sofia/internal/sip:1001 at 192.168.10.192:41080) State ROUTING going to
> sleep
> 2010-02-09 21:49:32.907184 [DEBUG] switch_core_state_machine.c:314
> (sofia/internal/sip:1001 at 192.168.10.192:41080) Running State Change
> CS_CONSUME_MEDIA
> 2010-02-09 21:49:33.023071 [DEBUG] switch_core_state_machine.c:360
> (sofia/internal/sip:1001 at 192.168.10.192:41080) State CONSUME_MEDIA
> 2010-02-09 21:49:33.023071 [DEBUG] switch_core_state_machine.c:360
> (sofia/internal/sip:1001 at 192.168.10.192:41080) State CONSUME_MEDIA going
> to sleep
> 2010-02-09 21:49:33.329319 [DEBUG] sofia.c:3727 Channel sofia/internal/
> sip:1001 at 192.168.10.192:41080 entering state [terminated][488]
> 2010-02-09 21:49:33.329319 [NOTICE] sofia.c:4331 Hangup sofia/internal/
> sip:1001 at 192.168.10.192:41080 [CS_CONSUME_MEDIA]
> [INCOMPATIBLE_DESTINATION]
> 2010-02-09 21:49:33.329319 [DEBUG] switch_channel.c:1912 Send signal
> sofia/internal/sip:1001 at 192.168.10.192:41080 [KILL]
> 2010-02-09 21:49:33.329319 [DEBUG] switch_core_session.c:999 Send signal
> sofia/internal/sip:1001 at 192.168.10.192:41080 [BREAK]
> 2010-02-09 21:49:33.329319 [DEBUG] switch_core_state_machine.c:459
> sofia/internal/sip:1001 at 192.168.10.192:41080 thread mismatch skipping
> state handler.
> 2010-02-09 21:49:33.329319 [DEBUG] switch_core_state_machine.c:314
> (sofia/internal/sip:1001 at 192.168.10.192:41080) Running State Change
> CS_HANGUP
> 2010-02-09 21:49:33.330075 [DEBUG] switch_core_state_machine.c:488
> (sofia/internal/sip:1001 at 192.168.10.192:41080) State HANGUP
> 2010-02-09 21:49:33.330075 [DEBUG] mod_sofia.c:352 sofia/internal/
> sip:1001 at 192.168.10.192:41080 Overriding SIP cause 488 with 488 from the
> other leg
> 2010-02-09 21:49:33.330075 [DEBUG] mod_sofia.c:358 Channel sofia/internal/
> sip:1001 at 192.168.10.192:41080 hanging up, cause: INCOMPATIBLE_DESTINATION
> 2010-02-09 21:49:33.330075 [DEBUG] switch_core_state_machine.c:46
> sofia/internal/sip:1001 at 192.168.10.192:41080 Standard HANGUP, cause:
> INCOMPATIBLE_DESTINATION
> 2010-02-09 21:49:33.330075 [DEBUG] switch_core_state_machine.c:488
> (sofia/internal/sip:1001 at 192.168.10.192:41080) State HANGUP going to sleep
> 2010-02-09 21:49:33.330075 [DEBUG] switch_core_state_machine.c:333
> (sofia/internal/sip:1001 at 192.168.10.192:41080) State Change CS_HANGUP ->
> CS_REPORTING
> 2010-02-09 21:49:33.330075 [DEBUG] switch_core_session.c:999 Send signal
> sofia/internal/sip:1001 at 192.168.10.192:41080 [BREAK]
> 2010-02-09 21:49:33.330075 [DEBUG] switch_core_state_machine.c:314
> (sofia/internal/sip:1001 at 192.168.10.192:41080) Running State Change
> CS_REPORTING
> 2010-02-09 21:49:33.330075 [DEBUG] switch_core_state_machine.c:579
> (sofia/internal/sip:1001 at 192.168.10.192:41080) State REPORTING
> 2010-02-09 21:49:33.330075 [DEBUG] switch_core_state_machine.c:53
> sofia/internal/sip:1001 at 192.168.10.192:41080 Standard REPORTING, cause:
> INCOMPATIBLE_DESTINATION
> 2010-02-09 21:49:33.330075 [DEBUG] switch_core_state_machine.c:579
> (sofia/internal/sip:1001 at 192.168.10.192:41080) State REPORTING going to
> sleep
> 2010-02-09 21:49:33.330075 [DEBUG] switch_core_state_machine.c:327
> (sofia/internal/sip:1001 at 192.168.10.192:41080) State Change CS_REPORTING
> -> CS_DESTROY
> 2010-02-09 21:49:33.330075 [DEBUG] switch_core_session.c:999 Send signal
> sofia/internal/sip:1001 at 192.168.10.192:41080 [BREAK]
> 2010-02-09 21:49:33.330075 [DEBUG] switch_core_session.c:1136 Session 2
> (sofia/internal/sip:1001 at 192.168.10.192:41080) Locked, Waiting on external
> entities
> 2010-02-09 21:49:33.331072 [DEBUG] switch_ivr_originate.c:3009 Originate
> Resulted in Error Cause: 88 [INCOMPATIBLE_DESTINATION]
> 2010-02-09 21:49:33.331072 [NOTICE] switch_core_session.c:1154 Session 2
> (sofia/internal/sip:1001 at 192.168.10.192:41080) Ended
> 2010-02-09 21:49:33.331072 [NOTICE] switch_core_session.c:1156 Close
> Channel sofia/internal/sip:1001 at 192.168.10.192:41080 [CS_DESTROY]
> 2010-02-09 21:49:33.331072 [DEBUG] switch_core_state_machine.c:423
> (sofia/internal/sip:1001 at 192.168.10.192:41080) Running State Change
> CS_DESTROY
> 2010-02-09 21:49:33.331072 [DEBUG] switch_core_state_machine.c:434
> (sofia/internal/sip:1001 at 192.168.10.192:41080) State DESTROY
> 2010-02-09 21:49:33.331072 [DEBUG] mod_sofia.c:293 sofia/internal/
> sip:1001 at 192.168.10.192:41080 SOFIA DESTROY
> 2010-02-09 21:49:33.331072 [DEBUG] switch_core_state_machine.c:60
> sofia/internal/sip:1001 at 192.168.10.192:41080 Standard DESTROY
> 2010-02-09 21:49:33.331072 [DEBUG] switch_core_state_machine.c:434
> (sofia/internal/sip:1001 at 192.168.10.192:41080) State DESTROY going to
> sleep
> 2010-02-09 21:49:33.331072 [ERR] switch_ivr_originate.c:2249 Cannot create
> outgoing channel of type [user] cause: [INCOMPATIBLE_DESTINATION]
> 2010-02-09 21:49:33.331072 [DEBUG] switch_ivr_originate.c:3009 Originate
> Resulted in Error Cause: 88 [INCOMPATIBLE_DESTINATION]
> 2010-02-09 21:49:33.331072 [INFO] mod_dptools.c:2294 Originate Failed.
> Cause: INCOMPATIBLE_DESTINATION
> EXECUTE sofia/internal/60002 at 192.168.10.30 answer()
> 2010-02-09 21:49:33.332075 [DEBUG] mod_dptools.c:658 sofia/internal/
> 60002 at 192.168.10.30 receive message [ANSWER]
> 2010-02-09 21:49:33.332075 [DEBUG] sofia_glue.c:2381 AUDIO RTP
> [sofia/internal/60002 at 192.168.10.30] 192.168.10.30 port 27634 ->
> 192.168.10.131 port 25592 codec: 9 ms: 20
> 2010-02-09 21:49:33.332075 [DEBUG] switch_rtp.c:1167 Starting timer [soft]
> 160 bytes per 20ms
> 2010-02-09 21:49:33.333080 [DEBUG] mod_sofia.c:571 Local SDP
> sofia/internal/60002 at 192.168.10.30:
> v=0
> o=FreeSWITCH 1265704739 1265704740 IN IP4 192.168.10.30
> s=FreeSWITCH
> c=IN IP4 192.168.10.30
> t=0 0
> m=audio 27634 RTP/AVP 9 101
> a=rtpmap:9 G722/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
>
> 2010-02-09 21:49:33.333080 [DEBUG] sofia.c:3727 Channel sofia/internal/
> 60002 at 192.168.10.30 entering state [completed][200]
> 2010-02-09 21:49:33.333080 [DEBUG] switch_core_session.c:645 Send signal
> sofia/internal/60002 at 192.168.10.30 [BREAK]
> 2010-02-09 21:49:33.333080 [NOTICE] mod_dptools.c:658 Channel
> [sofia/internal/60002 at 192.168.10.30] has been answered
> 2010-02-09 21:49:33.333080 [DEBUG] switch_channel.c:182 sofia/internal/
> 60002 at 192.168.10.30 receive message [AUDIO_SYNC]
> EXECUTE sofia/internal/60002 at 192.168.10.30 sleep(1000)
> 2010-02-09 21:49:33.334081 [DEBUG] switch_channel.c:182 sofia/internal/
> 60002 at 192.168.10.30 receive message [AUDIO_SYNC]
> 2010-02-09 21:49:33.395069 [DEBUG] switch_rtp.c:2004 Correct ip/port
> confirmed.
> 2010-02-09 21:49:33.455069 [DEBUG] sofia.c:3727 Channel sofia/internal/
> 60002 at 192.168.10.30 entering state [ready][200]
> EXECUTE sofia/internal/60002 at 192.168.10.30 voicemail(default 192.168.10.30
> 1001)
> 2010-02-09 21:49:34.335070 [DEBUG] mod_voicemail.c:730 [default] rwlock
> 2010-02-09 21:49:34.370083 [DEBUG] switch_channel.c:182 sofia/internal/
> 60002 at 192.168.10.30 receive message [AUDIO_SYNC]
> 2010-02-09 21:49:34.475070 [DEBUG] switch_ivr_play_say.c:118 No language
> specified - Using [en]
> 2010-02-09 21:49:34.475070 [DEBUG] switch_ivr_play_say.c:273 Handle
> play-file:[voicemail/vm-person.wav] (en:en)
> 2010-02-09 21:49:34.523070 [WARNING] switch_core_file.c:177 Sample rate
> doesn't match
> 2010-02-09 21:49:34.524073 [DEBUG] switch_ivr_play_say.c:1154 Codec
> Activated L16 at 16000hz 1 channels 20ms
> 2010-02-09 21:49:34.526072 [DEBUG] switch_core_io.c:652 sofia/internal/
> 60002 at 192.168.10.30 receive message [TRANSCODING_NECESSARY]
> 2010-02-09 21:49:35.875063 [DEBUG] switch_ivr_play_say.c:1446 done playing
> file
> 2010-02-09 21:49:35.995062 [DEBUG] switch_ivr_play_say.c:273 Handle
> say:[1001] (en:en)
> 2010-02-09 21:49:36.027062 [WARNING] switch_core_file.c:177 Sample rate
> doesn't match
> 2010-02-09 21:49:36.027062 [DEBUG] switch_ivr_play_say.c:1154 Codec
> Activated L16 at 16000hz 1 channels 20ms
> 2010-02-09 21:49:36.028065 [DEBUG] switch_core_io.c:652 sofia/internal/
> 60002 at 192.168.10.30 receive message [TRANSCODING_NECESSARY]
> 2010-02-09 21:49:36.455072 [DEBUG] switch_ivr_play_say.c:1446 done playing
> file
> 2010-02-09 21:49:36.460061 [WARNING] switch_core_file.c:177 Sample rate
> doesn't match
> 2010-02-09 21:49:36.460061 [DEBUG] switch_ivr_play_say.c:1154 Codec
> Activated L16 at 16000hz 1 channels 20ms
> 2010-02-09 21:49:36.461074 [DEBUG] switch_core_io.c:652 sofia/internal/
> 60002 at 192.168.10.30 receive message [TRANSCODING_NECESSARY]
> 2010-02-09 21:49:36.995060 [DEBUG] switch_ivr_play_say.c:1446 done playing
> file
> 2010-02-09 21:49:36.995060 [WARNING] switch_core_file.c:177 Sample rate
> doesn't match
> 2010-02-09 21:49:36.995060 [DEBUG] switch_ivr_play_say.c:1154 Codec
> Activated L16 at 16000hz 1 channels 20ms
> 2010-02-09 21:49:36.995060 [DEBUG] switch_core_io.c:652 sofia/internal/
> 60002 at 192.168.10.30 receive message [TRANSCODING_NECESSARY]
> 2010-02-09 21:49:37.535061 [DEBUG] switch_ivr_play_say.c:1446 done playing
> file
> 2010-02-09 21:49:37.535061 [WARNING] switch_core_file.c:177 Sample rate
> doesn't match
> 2010-02-09 21:49:37.535061 [DEBUG] switch_ivr_play_say.c:1154 Codec
> Activated L16 at 16000hz 1 channels 20ms
> 2010-02-09 21:49:37.535061 [DEBUG] switch_core_io.c:652 sofia/internal/
> 60002 at 192.168.10.30 receive message [TRANSCODING_NECESSARY]
> 2010-02-09 21:49:37.995060 [DEBUG] switch_ivr_play_say.c:1446 done playing
> file
> 2010-02-09 21:49:38.095060 [DEBUG] switch_ivr_play_say.c:273 Handle
> play-file:[voicemail/vm-not_available.wav] (en:en)
> 2010-02-09 21:49:38.118060 [WARNING] switch_core_file.c:177 Sample rate
> doesn't match
> 2010-02-09 21:49:38.119063 [DEBUG] switch_ivr_play_say.c:1154 Codec
> Activated L16 at 16000hz 1 channels 20ms
> 2010-02-09 21:49:38.120062 [DEBUG] switch_core_io.c:652 sofia/internal/
> 60002 at 192.168.10.30 receive message [TRANSCODING_NECESSARY]
> 2010-02-09 21:49:39.075057 [DEBUG] switch_ivr_play_say.c:1446 done playing
> file
> 2010-02-09 21:49:39.175057 [DEBUG] switch_ivr_play_say.c:118 No language
> specified - Using [en]
> 2010-02-09 21:49:39.177086 [DEBUG] switch_ivr_play_say.c:273 Handle
> play-file:[voicemail/vm-record_message.wav] (en:en)
> 2010-02-09 21:49:39.181967 [WARNING] switch_core_file.c:177 Sample rate
> doesn't match
> 2010-02-09 21:49:39.181967 [DEBUG] switch_ivr_play_say.c:1154 Codec
> Activated L16 at 16000hz 1 channels 20ms
> 2010-02-09 21:49:39.192068 [DEBUG] switch_core_io.c:652 sofia/internal/
> 60002 at 192.168.10.30 receive message [TRANSCODING_NECESSARY]
> 2010-02-09 21:49:40.999151 [NOTICE] sofia.c:329 Hangup sofia/internal/
> 60002 at 192.168.10.30 [CS_EXECUTE] [NORMAL_CLEARING]
> 2010-02-09 21:49:40.999151 [DEBUG] switch_channel.c:1912 Send signal
> sofia/internal/60002 at 192.168.10.30 [KILL]
> 2010-02-09 21:49:40.999151 [DEBUG] switch_core_session.c:999 Send signal
> sofia/internal/60002 at 192.168.10.30 [BREAK]
> 2010-02-09 21:49:40.999151 [DEBUG] switch_core_state_machine.c:459
> sofia/internal/60002 at 192.168.10.30 thread mismatch skipping state handler.
> 2010-02-09 21:49:41.015049 [DEBUG] switch_ivr_play_say.c:1446 done playing
> file
> 2010-02-09 21:49:41.114049 [DEBUG] switch_core_state_machine.c:348
> (sofia/internal/60002 at 192.168.10.30) State EXECUTE going to sleep
> 2010-02-09 21:49:41.114049 [DEBUG] switch_core_state_machine.c:314
> (sofia/internal/60002 at 192.168.10.30) Running State Change CS_HANGUP
> 2010-02-09 21:49:41.114049 [DEBUG] switch_core_state_machine.c:488
> (sofia/internal/60002 at 192.168.10.30) State HANGUP
> 2010-02-09 21:49:41.114049 [DEBUG] mod_sofia.c:352 sofia/internal/
> 60002 at 192.168.10.30 Overriding SIP cause 480 with 488 from the other leg
> 2010-02-09 21:49:41.114049 [DEBUG] mod_sofia.c:358 Channel sofia/internal/
> 60002 at 192.168.10.30 hanging up, cause: NORMAL_CLEARING
> 2010-02-09 21:49:41.114049 [DEBUG] switch_core_state_machine.c:46
> sofia/internal/60002 at 192.168.10.30 Standard HANGUP, cause: NORMAL_CLEARING
> 2010-02-09 21:49:41.114049 [DEBUG] switch_core_state_machine.c:488
> (sofia/internal/60002 at 192.168.10.30) State HANGUP going to sleep
> 2010-02-09 21:49:41.114049 [DEBUG] switch_core_state_machine.c:333
> (sofia/internal/60002 at 192.168.10.30) State Change CS_HANGUP ->
> CS_REPORTING
> 2010-02-09 21:49:41.114049 [DEBUG] switch_core_session.c:999 Send signal
> sofia/internal/60002 at 192.168.10.30 [BREAK]
> 2010-02-09 21:49:41.114049 [DEBUG] switch_core_state_machine.c:314
> (sofia/internal/60002 at 192.168.10.30) Running State Change CS_REPORTING
> 2010-02-09 21:49:41.114049 [DEBUG] switch_core_state_machine.c:579
> (sofia/internal/60002 at 192.168.10.30) State REPORTING
> 2010-02-09 21:49:41.129049 [DEBUG] switch_core_state_machine.c:53
> sofia/internal/60002 at 192.168.10.30 Standard REPORTING, cause:
> NORMAL_CLEARING
> 2010-02-09 21:49:41.129049 [DEBUG] switch_core_state_machine.c:579
> (sofia/internal/60002 at 192.168.10.30) State REPORTING going to sleep
> 2010-02-09 21:49:41.129049 [DEBUG] switch_core_state_machine.c:327
> (sofia/internal/60002 at 192.168.10.30) State Change CS_REPORTING ->
> CS_DESTROY
> 2010-02-09 21:49:41.129049 [DEBUG] switch_core_session.c:999 Send signal
> sofia/internal/60002 at 192.168.10.30 [BREAK]
> 2010-02-09 21:49:41.129049 [DEBUG] switch_core_session.c:1136 Session 1
> (sofia/internal/60002 at 192.168.10.30) Locked, Waiting on external entities
> 2010-02-09 21:49:41.129049 [NOTICE] switch_core_session.c:1154 Session 1
> (sofia/internal/60002 at 192.168.10.30) Ended
> 2010-02-09 21:49:41.129049 [NOTICE] switch_core_session.c:1156 Close
> Channel sofia/internal/60002 at 192.168.10.30 [CS_DESTROY]
> 2010-02-09 21:49:41.129049 [DEBUG] switch_core_state_machine.c:423
> (sofia/internal/60002 at 192.168.10.30) Running State Change CS_DESTROY
> 2010-02-09 21:49:41.129049 [DEBUG] switch_core_state_machine.c:434
> (sofia/internal/60002 at 192.168.10.30) State DESTROY
> 2010-02-09 21:49:41.129049 [DEBUG] mod_sofia.c:293 sofia/internal/
> 60002 at 192.168.10.30 SOFIA DESTROY
> 2010-02-09 21:49:41.129049 [DEBUG] switch_core_state_machine.c:60
> sofia/internal/60002 at 192.168.10.30 Standard DESTROY
> 2010-02-09 21:49:41.129049 [DEBUG] switch_core_state_machine.c:434
> (sofia/internal/60002 at 192.168.10.30) State DESTROY going to sleep
>
>
>
>
>
>
>
> On 9 February 2010 21:18, Brian West <brian at freeswitch.org> wrote:
>
>> What you're saying makes little or no sense to me even on 1.0.4, Can you
>> pastebin your logs?
>>
>> /b
>>
>> On Feb 9, 2010, at 3:15 PM, Bruce Hopkins wrote:
>>
>> > OK, many thanks for the extremely swift response Brian.
>> >
>> > I will try to get up and running as soon as I can with 1.0.5 and see if
>> the issue goes away.
>> >
>> > thanks again
>> > Bruce
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100209/e2ab1b54/attachment-0002.html 

From brian at freeswitch.org  Tue Feb  9 14:08:28 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 9 Feb 2010 16:08:28 -0600
Subject: [Freeswitch-users] g.722 --> SPEEX/16000 or SPEEX/32000
	transcoding issue
In-Reply-To: <d03a10e01002091404i5e7aefaam7235bef68828d42b@mail.gmail.com>
References: <d03a10e01002091203qde4caabi1a7565929e60918f@mail.gmail.com>
	<03A2F6D0-754A-4067-A106-9D91B0D68A92@freeswitch.org>
	<d03a10e01002091315n5fa9242eoe2a86c4edee1501d@mail.gmail.com>
	<F167991C-151B-4D40-BF27-80A4C3A44D1E@freeswitch.org>
	<d03a10e01002091355n1934627t3902e0d8af5c7dc3@mail.gmail.com>
	<d03a10e01002091404i5e7aefaam7235bef68828d42b@mail.gmail.com>
Message-ID: <A97C0347-C9C5-4AFE-854E-10468A5E7711@freeswitch.org>

Nope we don't look at that... that isn't the problem.  The issue is you haven't allowed SPEEX at 16000h or SPEEX at 32000h

You can't just allow SPEEX... it can't figure out what you mean by just SPEEX.  See previous email.

/b

On Feb 9, 2010, at 4:04 PM, Bruce Hopkins wrote:

> g.722 might not look like it from the SDP which announces g722/8000 - but this is a historical error in the RFC.
> 
> Bruce




From brian at freeswitch.org  Tue Feb  9 14:10:33 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 9 Feb 2010 16:10:33 -0600
Subject: [Freeswitch-users] g.722 --> SPEEX/16000 or SPEEX/32000
	transcoding issue
In-Reply-To: <d03a10e01002091404i5e7aefaam7235bef68828d42b@mail.gmail.com>
References: <d03a10e01002091203qde4caabi1a7565929e60918f@mail.gmail.com>
	<03A2F6D0-754A-4067-A106-9D91B0D68A92@freeswitch.org>
	<d03a10e01002091315n5fa9242eoe2a86c4edee1501d@mail.gmail.com>
	<F167991C-151B-4D40-BF27-80A4C3A44D1E@freeswitch.org>
	<d03a10e01002091355n1934627t3902e0d8af5c7dc3@mail.gmail.com>
	<d03a10e01002091404i5e7aefaam7235bef68828d42b@mail.gmail.com>
Message-ID: <DBD23DC8-C9F3-432E-90E0-E4132D1C5C52@freeswitch.org>

This means you're playing a wav file that doesn't match so it has to resample it... Please open up vars.xml and read thru the codec docs there in the XML directly.  It will explain the codecs and how to allow them in various ptimes and rates as you need to do in your case.  Its got nothing to do with G722/8000 or anything to do with the file you're playing (which isn't matching the channel sample rate so it has to resample so at the very least it works.

Run the "file" util on your wav file and see what its real rate is.

/b

On Feb 9, 2010, at 4:04 PM, Bruce Hopkins wrote:

> 2010-02-09 21:49:36.995060 [WARNING] switch_core_file.c:177 Sample rate doesn't match




From jbrucehopkins at gmail.com  Tue Feb  9 14:13:06 2010
From: jbrucehopkins at gmail.com (Bruce Hopkins)
Date: Tue, 9 Feb 2010 22:13:06 +0000
Subject: [Freeswitch-users] g.722 --> SPEEX/16000 or SPEEX/32000
	transcoding issue
In-Reply-To: <51A80F87-F2B9-477E-8D14-99C40E24413D@freeswitch.org>
References: <d03a10e01002091203qde4caabi1a7565929e60918f@mail.gmail.com>
	<03A2F6D0-754A-4067-A106-9D91B0D68A92@freeswitch.org>
	<d03a10e01002091315n5fa9242eoe2a86c4edee1501d@mail.gmail.com>
	<F167991C-151B-4D40-BF27-80A4C3A44D1E@freeswitch.org>
	<d03a10e01002091355n1934627t3902e0d8af5c7dc3@mail.gmail.com>
	<51A80F87-F2B9-477E-8D14-99C40E24413D@freeswitch.org>
Message-ID: <d03a10e01002091413see99fe3ke52e0d3f91714216@mail.gmail.com>

Aha - I will try this now.  At the moment I only have SPEEX without anything
like @8000h or 16000h.

Thanks in advance, as I am sure you have spotted my noob error !

Bruce

On 9 February 2010 22:00, Brian West <brian at freeswitch.org> wrote:

> You need to allow SPEEX at 8000h,SPEEX at 16000h,SPEEX at 32000h
>
> /b
>
> On Feb 9, 2010, at 3:55 PM, Bruce Hopkins wrote:
>
> > Willdo,
> >
> > To clarify in brief though, the scenario which occurs and causes the call
> to fail is:
> >
> > SIP client  1 (g.722 enabled only ) -----> INVITE (with SDP offer: Media
> Attribute (a): rtpmap:9 g722/8000 ) -->FreeSWITCH
> >
> > ---> INVITE (with SDP offer including a bunch of codecs including rtpmap:
> rtpmap:98 SPEEX/8000 but crucially not including SPEEX/16000 or SPEEX/32000)
> >
> > ---> SIP client 2 (with only SPEEX/16000 or SPEEX/32000 enabled).
> >
> > The second SIP client does not get offered a codec it can accept, so SIP
> client 1 is sent a method 488 "Not Acceptable Here" message and the calling
> party gets directed to the voicemail for the other SIP client.
> >
> > By contrast, there is no problem calling SPEEX/32000 --> SPEEX/32000 or
> calling SPEEX/16000 --> SPEEX/16000.
> >
> > there is also no problem calling SPEEX/32000 --> g.722/8000.
> >
> > I am wondering if the problem is that FreeSWITCH is interpreting g.722 as
> being a narrowband (8kHz sample rate) codec, due to the historic anomaly of
> it presenting g722/8000 in the SDP even though it in fact uses 16kHz
> sampling, and for that reason not wanting to offer a 16kHz sample rate codec
> to the second SIP client?
> >
> > I suggest this as I also found trying to call alaw --> SPEEX/16000 does
> not work, for example.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100209/57a0d542/attachment-0002.html 

From lon at kickasspixels.com  Tue Feb  9 14:17:30 2010
From: lon at kickasspixels.com (Lon Baker)
Date: Tue, 9 Feb 2010 14:17:30 -0800
Subject: [Freeswitch-users] T.38 Fax mode?
Message-ID: <5d3e0dc61002091417l32526087je432ee465f88ef46@mail.gmail.com>

Hi,

Is 1.0.5 going to support T.38 fax mode? It looks like its partially
implemented.

The comment leads me to think its not

/*
		   Here goes the T.38 SpanDSP initializing functions
		   T.38 will require a big effort as it needs a different approach
		   but the pieces are already in place
		*/

Lon



From jbrucehopkins at gmail.com  Tue Feb  9 14:26:24 2010
From: jbrucehopkins at gmail.com (Bruce Hopkins)
Date: Tue, 9 Feb 2010 22:26:24 +0000
Subject: [Freeswitch-users] g.722 --> SPEEX/16000 or SPEEX/32000
	transcoding issue
In-Reply-To: <d03a10e01002091413see99fe3ke52e0d3f91714216@mail.gmail.com>
References: <d03a10e01002091203qde4caabi1a7565929e60918f@mail.gmail.com>
	<03A2F6D0-754A-4067-A106-9D91B0D68A92@freeswitch.org>
	<d03a10e01002091315n5fa9242eoe2a86c4edee1501d@mail.gmail.com>
	<F167991C-151B-4D40-BF27-80A4C3A44D1E@freeswitch.org>
	<d03a10e01002091355n1934627t3902e0d8af5c7dc3@mail.gmail.com>
	<51A80F87-F2B9-477E-8D14-99C40E24413D@freeswitch.org>
	<d03a10e01002091413see99fe3ke52e0d3f91714216@mail.gmail.com>
Message-ID: <d03a10e01002091426q391ac6f9nb14cbf678e3fd840@mail.gmail.com>

Hi again Brian,

I can confirm that having followed your advice, it now works perfectly.

Many, many thanks for your extraordinarily quick and comprehensive help with
this.

Best wishes
Bruce

On 9 February 2010 22:13, Bruce Hopkins <jbrucehopkins at gmail.com> wrote:

> Aha - I will try this now.  At the moment I only have SPEEX without
> anything like @8000h or 16000h.
>
> Thanks in advance, as I am sure you have spotted my noob error !
>
> Bruce
>
>
> On 9 February 2010 22:00, Brian West <brian at freeswitch.org> wrote:
>
>> You need to allow SPEEX at 8000h,SPEEX at 16000h,SPEEX at 32000h
>>
>> /b
>>
>> On Feb 9, 2010, at 3:55 PM, Bruce Hopkins wrote:
>>
>> > Willdo,
>> >
>> > To clarify in brief though, the scenario which occurs and causes the
>> call to fail is:
>> >
>> > SIP client  1 (g.722 enabled only ) -----> INVITE (with SDP offer: Media
>> Attribute (a): rtpmap:9 g722/8000 ) -->FreeSWITCH
>> >
>> > ---> INVITE (with SDP offer including a bunch of codecs including
>> rtpmap: rtpmap:98 SPEEX/8000 but crucially not including SPEEX/16000 or
>> SPEEX/32000)
>> >
>> > ---> SIP client 2 (with only SPEEX/16000 or SPEEX/32000 enabled).
>> >
>> > The second SIP client does not get offered a codec it can accept, so SIP
>> client 1 is sent a method 488 "Not Acceptable Here" message and the calling
>> party gets directed to the voicemail for the other SIP client.
>> >
>> > By contrast, there is no problem calling SPEEX/32000 --> SPEEX/32000 or
>> calling SPEEX/16000 --> SPEEX/16000.
>> >
>> > there is also no problem calling SPEEX/32000 --> g.722/8000.
>> >
>> > I am wondering if the problem is that FreeSWITCH is interpreting g.722
>> as being a narrowband (8kHz sample rate) codec, due to the historic anomaly
>> of it presenting g722/8000 in the SDP even though it in fact uses 16kHz
>> sampling, and for that reason not wanting to offer a 16kHz sample rate codec
>> to the second SIP client?
>> >
>> > I suggest this as I also found trying to call alaw --> SPEEX/16000 does
>> not work, for example.
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100209/f5d948e2/attachment-0002.html 

From sergey.kobzar at mail.ru  Tue Feb  9 14:44:22 2010
From: sergey.kobzar at mail.ru (Sergey Kobzar)
Date: Wed, 10 Feb 2010 00:44:22 +0200
Subject: [Freeswitch-users] Video conferencing
In-Reply-To: <D9657862-D8A0-45C8-98E2-42814BE8D6A1@jerris.com>
References: <57499143.20100209185702@mail.ru>
	<D9657862-D8A0-45C8-98E2-42814BE8D6A1@jerris.com>
Message-ID: <1899501996.20100210004422@mail.ru>

Which softphone do you use?

Do you know any other alternatives which works better?


Tuesday, February 9, 2010, 11:16:13 PM, Michael wrote:

> Our video conference features "work" but the functionality is
> pretty limited.  We don;t have iframe detection and can not do any
> video transcoding, just video follow audio support.  This code needs some work for sure.

> Mike

> On Feb 9, 2010, at 11:57 AM, Sergey Kobzar wrote:

>> Does anybody have a success story of implementing video conferencing?
>> 
>> I've spend some time with Goole and found that Asterisk has many
>> limitations, hardware solutions are quite expensive. I played with FS
>> without luck.


> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




-- 
Sergey




From rupa at rupa.com  Tue Feb  9 14:55:05 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Tue, 9 Feb 2010 16:55:05 -0600
Subject: [Freeswitch-users] UPnP Timeout
In-Reply-To: <89dbfdc31002090652h884bc2av82d53c6958fdc409@mail.gmail.com>
References: <89dbfdc31002090652h884bc2av82d53c6958fdc409@mail.gmail.com>
Message-ID: <b39273531002091455meb77162o2baefc3866efaed9@mail.gmail.com>

That is a different keep-alive.  I'm specifically talking about the
keep-alive packet that we get via upnp multicast.  Whenever we receive one
from the gateway we republish the nat mappings to.. um... keep them alive.
:)

On Tue, Feb 9, 2010 at 8:52 AM, Kim Culhan <w8hdkim at gmail.com> wrote:

> On Mon, February 8, 2010 11:21 pm, Troy Anderson wrote:
> > I have been using pfSense (1.2.3) and FS.  FS nicely uses UPnP to poke
> > holes in the firewall, but it seems that the holes close after a while.
>  I
> > cannot find any documentation in FS nor in pfSense as to what the timeout
> > is.  Is there a setting in FS to do some kind of keep-alive thing with
> > UPnP to keep, e.g. 5060, open?  Or is it already doing that and pfSense
> is
> > the issue?
>
> FS has provisions for keep-alive, see the bottom of the page for ping
> time value:
>
> http://wiki.freeswitch.org/wiki/SIP_Provider_Examples
>
> To watch the pf firewall hole timing you can install pftop from
> FreeBSD ports/sysutils
> which displays the filter states 'and more'.
>
> -kim
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
-Rupa
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100209/3ea876cd/attachment-0002.html 

From msc at freeswitch.org  Tue Feb  9 14:55:11 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 9 Feb 2010 14:55:11 -0800
Subject: [Freeswitch-users] Passing channel variables to mod_fifo and
	xml_cdr
In-Reply-To: <004301caa608$534747d0$f9d5d770$@net>
References: <004301caa608$534747d0$f9d5d770$@net>
Message-ID: <87f2f3b91002091455v6af079e1ie28ed6891d5ed628@mail.gmail.com>

On Thu, Feb 4, 2010 at 6:09 PM, Adam Ford <lists at redbonez.net> wrote:

>  When sending a call through mod_fifo I seem to be losing my custom
> channel variables that were assigned during prior processing of the call.
> In my example, I am trying to assign a unique identifier at the time the
> call enters my FreeSWITCH system in order to more easily tie the xml_cdr
> logs together.  This works great, until a call is processed through
> mod_fifo, which drops my custom channel variable in the calls that it
> generates.  Is it likely that I have something wrong with my config? Or does
> mod_fifo not support the passing of custom channel variables?
>
>
>
> The overall problem I am trying to solve is that mod_fifo generates a
> separate a-leg for every time it rings an agent.  If the agent answers, the
> a-leg log gets tied to the associated b-leg log with the uuids and I am able
> to see the entire call in xml_cdr.  However, if the agent rejects the call
> or doesn?t answer, the a-leg is abandoned with seemingly no association back
> to a b-leg log.  Anyone have a better suggestion for tying all these xml_cdr
> logs together for a full picture of the call?
>

Just curious - are you looking at this from the caller's perspective or the
agent's perspective? An unanswered/rejected call from FIFO to an agent
doesn't tell you very much. However, if you're trying to gather statistics
on an individual agent then I could see why you'd want to know how many FIFO
calls they failed to answer. As far as the "new" A leg not being tied back
to a B leg - Mike J is 100% correct: A FIFO call to an agent has absolutely
no correlation to any caller waiting in queue. (I suppose the only exception
to this rule would be if there was only one caller in the queue when the
FIFO called out to the agents.)

Like Mike said, FIFO is not ACD. FIFO is "get the caller to a human as
efficiently and quickly as possible." ACD is more of "connect the longest
waiting caller to the longest waiting agent, with possible exceptions for
skills, etc." Check out Andrew Thompson's SpiceCSM for a possible solution
to your scenario.

http://www.opencsm.org/wiki/index.php/SpiceCSM_Community_Edition

-MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100209/a3382a29/attachment-0002.html 

From mike at jerris.com  Tue Feb  9 15:03:43 2010
From: mike at jerris.com (Michael Jerris)
Date: Tue, 9 Feb 2010 18:03:43 -0500
Subject: [Freeswitch-users] Way to pass max-members from dialplan into
	conference?
In-Reply-To: <C7136E29C3FD472BB367FF412E282BED@greyhawk.tonecommander.com>
References: <C7136E29C3FD472BB367FF412E282BED@greyhawk.tonecommander.com>
Message-ID: <57F0B3AE-BB20-4F27-9CF5-7A9AE77E9738@jerris.com>

We don 't have this right now where you can set a var, but you could add it with one line around line 5265, just set confierence->max_members there based on the result of switch_channel_get_variable.  This would let you set max_members when you first create a conference.  If you would like it to be re-set with any caller, you could do it later down.

Mike

On Feb 9, 2010, at 4:14 PM, Robert Hadley wrote:

> I have setting max-members=10 in conference.conf.xml working.  However, is there are way to pass in the max-members=10 from the dialplan/default.xml to mod_conference?  I tried using action application=?set? data=?max-members=10? but it didn?t work.  Also tried action application=?export? data=?max-members=10? but it didn?t work either.
>  
> From default.xml:
>     <extension name="nb_conferences">
>       <condition field="destination_number" expression="^(30\d{2})$">
>        <action application="answer"/>
> <!-- tried set and export here to no avail -->
>        <action application="conference" data="$1-${domain_name}@default"/>
>  
> Thanks,
> Robert
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100209/5f87f89a/attachment-0002.html 

From mike at jerris.com  Tue Feb  9 15:04:02 2010
From: mike at jerris.com (Michael Jerris)
Date: Tue, 9 Feb 2010 18:04:02 -0500
Subject: [Freeswitch-users] T.38 Fax mode?
In-Reply-To: <5d3e0dc61002091417l32526087je432ee465f88ef46@mail.gmail.com>
References: <5d3e0dc61002091417l32526087je432ee465f88ef46@mail.gmail.com>
Message-ID: <34A64C2E-88A5-48C2-818D-85242321BCA4@jerris.com>

no

On Feb 9, 2010, at 5:17 PM, Lon Baker wrote:

> Hi,
> 
> Is 1.0.5 going to support T.38 fax mode? It looks like its partially
> implemented.
> 
> The comment leads me to think its not
> 
> /*
> 		   Here goes the T.38 SpanDSP initializing functions
> 		   T.38 will require a big effort as it needs a different approach
> 		   but the pieces are already in place
> 		*/




From mike at jerris.com  Tue Feb  9 15:05:03 2010
From: mike at jerris.com (Michael Jerris)
Date: Tue, 9 Feb 2010 18:05:03 -0500
Subject: [Freeswitch-users] Video conferencing
In-Reply-To: <1899501996.20100210004422@mail.ru>
References: <57499143.20100209185702@mail.ru>
	<D9657862-D8A0-45C8-98E2-42814BE8D6A1@jerris.com>
	<1899501996.20100210004422@mail.ru>
Message-ID: <A50D4526-42A7-44B2-99E6-6876303D3358@jerris.com>

they all suck, I have used hard devices and several different softphones, none of which I found usable or acceptable.

Mike

On Feb 9, 2010, at 5:44 PM, Sergey Kobzar wrote:

> Which softphone do you use?
> 
> Do you know any other alternatives which works better?
> 
> 
> Tuesday, February 9, 2010, 11:16:13 PM, Michael wrote:
> 
>> Our video conference features "work" but the functionality is
>> pretty limited.  We don;t have iframe detection and can not do any
>> video transcoding, just video follow audio support.  This code needs some work for sure.
> 
>> Mike




From mike at jerris.com  Tue Feb  9 15:07:20 2010
From: mike at jerris.com (Michael Jerris)
Date: Tue, 9 Feb 2010 18:07:20 -0500
Subject: [Freeswitch-users] Where are new sounds_dir and recordings_dir
	variables set?
In-Reply-To: <28788FA27BE74B9FA6BFFE52BED11B60@greyhawk.tonecommander.com>
References: <28788FA27BE74B9FA6BFFE52BED11B60@greyhawk.tonecommander.com>
Message-ID: <17F2ACB0-0344-4EC1-BB1F-21F3BF4C0892@jerris.com>

These are set at configure time based on the prefix.  When FreeSWITCH starts, it sets global vars (before the config pre-processing) that should work just like the old hard-coded values and how base_dir works with /sounds added to it

Mike

On Feb 9, 2010, at 2:28 PM, Robert Hadley wrote:

>  
> The XML conf files have been recently modified to replace ?$${base_dir}/sounds? with $${sounds_dir}.  The same replacement was done for $${base_dir}/recordings and $${recordings_dir}.  In vars.xml, the X-PRE-PROCESS cmd to setting the old variable sound_prefix was removed but set commands were not added for the new variables sounds_dir and recordings_dir.  How is FS finding determining the value of $${sounds_dir} and $${recordings_dir} in the XML files?
>  
> I ask because I have 2 cloned FS servers where one can find sound files in the IVR when an invalid extension is entered, but the other does not and is using the wrong path to search for sound files.
>  
> Server 2 Fails:
> 2010-02-09 11:11:16.735745 [DEBUG] switch_ivr_menu.c:381 digits '9999'
> 2010-02-09 11:11:16.735745 [DEBUG] switch_ivr_menu.c:475 action regex [9999] [/^((5[34][8901][0-9])||(10[0-5][0-9])||(30\d{2}))$/] [0]
> 2010-02-09 11:11:16.735745 [DEBUG] switch_ivr_menu.c:565 IVR menu 'teo_ivr' caught invalid input '9999'
> 2010-02-09 11:11:16.735745 [ERR] mod_sndfile.c:194 Error Opening File [/opt/teoswitch/sounds/ivr/ivr-that_was_an_invalid_entry.wav] [System error : No such file or directory.]
>  
> Server 1 Works:
> 2010-02-09 11:17:25.085660 [DEBUG] switch_ivr_menu.c:378 digits '9999'
> 2010-02-09 11:17:25.085660 [DEBUG] switch_ivr_menu.c:472 action regex [9999] [/^((5[34][8901][0-9])||(10[0-5][0-9])||(30\d{2}))$/] [0]
> 2010-02-09 11:17:25.085660 [DEBUG] switch_ivr_menu.c:562 IVR menu 'teo_ivr' caught invalid input '9999'
> 2010-02-09 11:17:25.085660 [DEBUG] switch_ivr_play_say.c:1162 Codec Activated L16 at 8000hz 1 channels 20ms
> 2010-02-09 11:17:25.085660 [DEBUG] switch_core_io.c:652 sofia/internal/1045 at 192.168.72.141:5060 receive message [TRANSCODING_NECESSARY]
> 2010-02-09 11:17:26.765470 [DEBUG] switch_ivr_play_say.c:1454 done playing file
>  
> I have compared the conf folders and they are nearly identical.  The only difference I know of is Server 2 was originally compiled with mod_flite enabled but it is not loaded at runtime.  Any suggestions why one server can find sound files but the other looks in the wrong path?
>  
> Thanks,
> Robert
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100209/d7eb530a/attachment-0002.html 

From msc at freeswitch.org  Tue Feb  9 15:09:16 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 9 Feb 2010 15:09:16 -0800
Subject: [Freeswitch-users] Way to pass max-members from dialplan into
	conference?
In-Reply-To: <C7136E29C3FD472BB367FF412E282BED@greyhawk.tonecommander.com>
References: <C7136E29C3FD472BB367FF412E282BED@greyhawk.tonecommander.com>
Message-ID: <87f2f3b91002091509w56eacdedt6a291d9b98838521@mail.gmail.com>

On Tue, Feb 9, 2010 at 1:14 PM, Robert Hadley <robert.hadley at teotech.com>wrote:

>  I have setting max-members=10 in conference.conf.xml working.  However,
> is there are way to pass in the max-members=10 from the dialplan/default.xml
> to mod_conference?  I tried using action application=?set?
> data=?max-members=10? but it didn?t work.  Also tried action
> application=?export? data=?max-members=10? but it didn?t work either.
>
>
>
> From default.xml:
>
>     <extension name="nb_conferences">
>
>       <condition field="destination_number" expression="^(30\d{2})$">
>
>        <action application="answer"/>
>
> <!-- tried set and export here to no avail -->
>
>        <action application="conference" data="$1-${domain_name}@default"/>
>
>
>
> Thanks,
>
> Robert
>

Robert,

I checked with Brian and also took a look inside mod_conference.c. I didn't
see any way that you could override the conference params that are contained
in the conference profile. So you'll either need to make a new profile or
join the big leagues and start trying out mod_xml_curl. :)

-MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100209/80c1f79f/attachment-0002.html 

From msc at freeswitch.org  Tue Feb  9 15:12:45 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 9 Feb 2010 15:12:45 -0800
Subject: [Freeswitch-users] UPnP Timeout
In-Reply-To: <b39273531002091455meb77162o2baefc3866efaed9@mail.gmail.com>
References: <89dbfdc31002090652h884bc2av82d53c6958fdc409@mail.gmail.com>
	<b39273531002091455meb77162o2baefc3866efaed9@mail.gmail.com>
Message-ID: <87f2f3b91002091512n6b66c233xe2436376c347e992@mail.gmail.com>

On Tue, Feb 9, 2010 at 2:55 PM, Rupa Schomaker <rupa at rupa.com> wrote:

> That is a different keep-alive.  I'm specifically talking about the
> keep-alive packet that we get via upnp multicast.  Whenever we receive one
> from the gateway we republish the nat mappings to.. um... keep them alive.
> :)
>
>
Ah.. I see: keep them alive. And all this time I thought it was to keep the
NAT mappings from dying! :P
-MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100209/566e46a5/attachment-0002.html 

From mike at jerris.com  Tue Feb  9 15:17:34 2010
From: mike at jerris.com (Michael Jerris)
Date: Tue, 9 Feb 2010 18:17:34 -0500
Subject: [Freeswitch-users] Bridging sessions from two separate lua
	scripts; uuid_bridge? intercept?
In-Reply-To: <eb7797d71002080737k3f8fa5c9m8bf3ef2fcf42881e@mail.gmail.com>
References: <eb7797d71002071831w27301c5dva767b82b38c2ce6f@mail.gmail.com>
	<C23AA196-E851-4D5B-B1B5-2C2F32F6BB68@freeswitch.org>
	<eb7797d71002071929pb53e1adue2e91e6bdb598251@mail.gmail.com>
	<eb7797d71002080737k3f8fa5c9m8bf3ef2fcf42881e@mail.gmail.com>
Message-ID: <D753EDBD-6ED7-43C2-B5F5-A899E3C3D3E2@jerris.com>

1.4?  how does the future look, report back?

http://files-sync.freeswitch.org/windows_installer/freepbx_svn.exe

I think this has latest FreeSWITCH in it to, Carlos, can you confirm that?

Mike

On Feb 8, 2010, at 10:37 AM, Adam Wilt wrote:

> One other thing I should mention. I'm running FreeSWITCH version 1.4 (build 14460) in Windows. 
> Brian suggested I upgrade to the build in the http://files-sync.freeswitch.org/windows_installer/  folder, but it turned out to be the exact same build I already had. I'd love to try upgrade to 1.5 in case this problem has been fixed already.
> 
> 
> On Sun, Feb 7, 2010 at 10:29 PM, Adam Wilt <wiltingtree at gmail.com> wrote:
> Thanks Michael for the reply.
> Here's the pastebin link: http://pastebin.freeswitch.org/12084
> 
> 
> On Sun, Feb 7, 2010 at 9:50 PM, Michael S Collins <msc at freeswitch.org> wrote:
> Pastebin a debug log so we can see what is happening when the script runs. 
> 
> -MC
> 
> Sent from my iPhone
> 
> On Feb 7, 2010, at 8:31 PM, Adam Wilt <wiltingtree at gmail.com> wrote:
> 
>> Hi. I have two sessions running in two separate Lua scripts, and I want to bridge them so that the bridged call is being controlled by the first (a-leg) script.
>> If I simply use uuid_bridge, I get no error but the calls don't bridge.
>> I've tried intercept, but I don't understand how it should be used; nothing I try seems to work.
>> Here's what I have:
>> 
>> function bridge_calls(session,api,b_leg_uuid, call_len)
>>   session:setAutoHangup(false)
>>   session:execute("sched_hangup","+" .. tostring(call_len) .. " " .. tostring(session.uuid))
>>   session:execute("set","continue_on_fail=true")
>>   api:executeString("intercept -bleg " .. tostring(b_leg_uuid))
>>   api:executeString("uuid_bridge " .. tostring(session.uuid) .. " " .. tostring(b_leg_uuid))
>> end
>> 
>> I'd really appreciate any help.
>> 
>> Thanks,
>> Adam
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100209/9a147cf7/attachment-0002.html 

From sergey.kobzar at mail.ru  Tue Feb  9 15:31:45 2010
From: sergey.kobzar at mail.ru (Sergey Kobzar)
Date: Wed, 10 Feb 2010 01:31:45 +0200
Subject: [Freeswitch-users] Video conferencing
In-Reply-To: <A50D4526-42A7-44B2-99E6-6876303D3358@jerris.com>
References: <57499143.20100209185702@mail.ru>
	<D9657862-D8A0-45C8-98E2-42814BE8D6A1@jerris.com>
	<1899501996.20100210004422@mail.ru>
	<A50D4526-42A7-44B2-99E6-6876303D3358@jerris.com>
Message-ID: <794626665.20100210013145@mail.ru>

I've found only this: http://code.google.com/p/openmeetings/

But didn't try yet. Also it is not exactly what I want.


Wednesday, February 10, 2010, 1:05:03 AM, Michael wrote:

> they all suck, I have used hard devices and several different
> softphones, none of which I found usable or acceptable.

> Mike

> On Feb 9, 2010, at 5:44 PM, Sergey Kobzar wrote:

>> Which softphone do you use?
>> 
>> Do you know any other alternatives which works better?
>> 
>> 
>> Tuesday, February 9, 2010, 11:16:13 PM, Michael wrote:
>> 
>>> Our video conference features "work" but the functionality is
>>> pretty limited.  We don;t have iframe detection and can not do any
>>> video transcoding, just video follow audio support.  This code needs some work for sure.
>> 
>>> Mike


> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




-- 
Sergey




From kristian.kielhofner at gmail.com  Tue Feb  9 15:36:10 2010
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Tue, 9 Feb 2010 18:36:10 -0500
Subject: [Freeswitch-users] T.38 Fax mode?
In-Reply-To: <5d3e0dc61002091417l32526087je432ee465f88ef46@mail.gmail.com>
References: <5d3e0dc61002091417l32526087je432ee465f88ef46@mail.gmail.com>
Message-ID: <2d9149cd1002091536p266f8b13x74224a3fb46f1af4@mail.gmail.com>

Pass-through with proxy media/bypass media works well...

On Tue, Feb 9, 2010 at 5:17 PM, Lon Baker <lon at kickasspixels.com> wrote:
> Hi,
>
> Is 1.0.5 going to support T.38 fax mode? It looks like its partially
> implemented.
>
> The comment leads me to think its not
>
> /*
> ? ? ? ? ? ? ? ? ? Here goes the T.38 SpanDSP initializing functions
> ? ? ? ? ? ? ? ? ? T.38 will require a big effort as it needs a different approach
> ? ? ? ? ? ? ? ? ? but the pieces are already in place
> ? ? ? ? ? ? ? ?*/
>
> Lon
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com



From anthony.minessale at gmail.com  Tue Feb  9 15:57:58 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 9 Feb 2010 17:57:58 -0600
Subject: [Freeswitch-users] Presence PUBLISH Not Updating After
	Softphone OffLine Then Available
In-Reply-To: <45963EB5-DFE5-43FA-AFCC-857E1E719E2D@jerris.com>
References: <31FEE9072F104AE18B26812978FCA341@greyhawk.tonecommander.com>
	<45963EB5-DFE5-43FA-AFCC-857E1E719E2D@jerris.com>
Message-ID: <191c3a031002091557jfc9b486lb832ea538883bce9@mail.gmail.com>

he means update to trunk first then try it again obviously.


On Tue, Feb 9, 2010 at 3:10 PM, Michael Jerris <mike at jerris.com> wrote:

> Try this again, I think I saw changes go in for this issue.
>
> Mike
>
> On Feb 5, 2010, at 2:38 PM, Jerry Richards wrote:
>
> > I found a scenario where presence status is not distributed to
> subscribers.
> > This is using the latest changes (as of Feb 03, 2010).  The scenario
> > follows:
> >
> > 1) Register two Bria softphones (A and B), which each have the other as a
> > contact.
> > 2) Set softphone B's presence status to 'Appear Offline'.
> > 3) Softphone A correctly reflects contact B is offline.
> > 4) Set softphone B's presence status to 'Available'.
> > 5) ******* There is no change to contact B's status at softphone A
> *******
> >
> > I posted a log at http://pastebin.freeswitch.org/12054.  At line 773,
> there
> > is an error when FS is processing the PUBLISH from softphone B:
> >
> > 773.2010-02-05 10:29:21.254221 [ERR] sofia_presence.c:674 DUMP PRESENCE
> SQL:
> >
> > I did notice that after about 30 minutes, softphone B's status gets
> > reflected at softphone A.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100209/dd8ca2b5/attachment-0002.html 

From pjintheusa at gmail.com  Tue Feb  9 16:01:09 2010
From: pjintheusa at gmail.com (Phillip Jones)
Date: Tue, 9 Feb 2010 19:01:09 -0500
Subject: [Freeswitch-users] Last call: buy the devs dinner!
In-Reply-To: <4B71885F.5090908@gmx.net>
References: <87f2f3b91002090735g1c6c69eby5274f6dfa9127fc5@mail.gmail.com>
	<4B71885F.5090908@gmx.net>
Message-ID: <367751821002091601u5c4a275bs6f2894a087b3ffcf@mail.gmail.com>

>> Thanks for the great work you all have done so far.

I second that.

On Tue, Feb 9, 2010 at 11:07 AM, Peter P GMX <Prometheus001 at gmx.net> wrote:

> Hello Michael,
>
> just hit the paypal button. Enjoy your dinner! I think it's not just
> dinner, it will be also 50% work I think (discussing about issues and
> new features etc.) which brings additional benefits to the copmmunity.
> Thanks for the great work you all have done so far.
>
> Best regards
> Peter
>
> Michael Collins schrieb:
> > Hey all,
> >
> > Thanks so much for the donations that have come in already! We
> > appreciate your generosity. The dev team really wants to release 1.0.5
> > but they're kinda hungry! :) Please hit the PayPal button on the main
> > freeswitch.org <http://freeswitch.org> page to drop a few dollars in
> > the hat. Also, keep in mind that we have the "extended family" of
> > developers all here so it's not just Tony, Mike, and Brian. Let's all
> > pitch in and have a great dinner for them.
> >
> > Thanks!
> > -Michael
> > ------------------------------------------------------------------------
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100209/ce5f6b89/attachment-0002.html 

From peder at networkoblivion.com  Tue Feb  9 16:26:01 2010
From: peder at networkoblivion.com (Peder)
Date: Tue, 9 Feb 2010 18:26:01 -0600
Subject: [Freeswitch-users] T.38 Fax mode?
In-Reply-To: <2d9149cd1002091536p266f8b13x74224a3fb46f1af4@mail.gmail.com>
References: <5d3e0dc61002091417l32526087je432ee465f88ef46@mail.gmail.com>
	<2d9149cd1002091536p266f8b13x74224a3fb46f1af4@mail.gmail.com>
Message-ID: <008201caa9e7$9fccf770$df66e650$@com>

Do you find any difference in success rate between using proxy instead of
bypass for T.38?  I wouldn't think it would really matter, I am just
curious.

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Kristian
Kielhofner
Sent: Tuesday, February 09, 2010 5:36 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] T.38 Fax mode?

Pass-through with proxy media/bypass media works well...

On Tue, Feb 9, 2010 at 5:17 PM, Lon Baker <lon at kickasspixels.com> wrote:
> Hi,
>
> Is 1.0.5 going to support T.38 fax mode? It looks like its partially
> implemented.
>
> The comment leads me to think its not
>
> /*
> ? ? ? ? ? ? ? ? ? Here goes the T.38 SpanDSP initializing functions
> ? ? ? ? ? ? ? ? ? T.38 will require a big effort as it needs a different
approach
> ? ? ? ? ? ? ? ? ? but the pieces are already in place
> ? ? ? ? ? ? ? ?*/
>
> Lon
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




From kristian.kielhofner at gmail.com  Tue Feb  9 17:21:30 2010
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Tue, 9 Feb 2010 20:21:30 -0500
Subject: [Freeswitch-users] T.38 Fax mode?
In-Reply-To: <008201caa9e7$9fccf770$df66e650$@com>
References: <5d3e0dc61002091417l32526087je432ee465f88ef46@mail.gmail.com>
	<2d9149cd1002091536p266f8b13x74224a3fb46f1af4@mail.gmail.com>
	<008201caa9e7$9fccf770$df66e650$@com>
Message-ID: <2d9149cd1002091721o4f100970k568acef52644f823@mail.gmail.com>

We almost always use proxy media because our T.38 endpoints are behind NAT.

On Tue, Feb 9, 2010 at 7:26 PM, Peder <peder at networkoblivion.com> wrote:
> Do you find any difference in success rate between using proxy instead of
> bypass for T.38? ?I wouldn't think it would really matter, I am just
> curious.
>

-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com



From jingwei.yang at gmail.com  Tue Feb  9 18:19:21 2010
From: jingwei.yang at gmail.com (Jingwei Yang)
Date: Wed, 10 Feb 2010 10:19:21 +0800
Subject: [Freeswitch-users]  How to record the call upon successful bridge
Message-ID: <13529f9d1002091819y2b363f50vc314d1b6197503b2@mail.gmail.com>

Hi,

I'm using uuid_bridge to bridge two calls. May I know how to start recording
only when the bridge succeeds?

Thanks,
-Jingwei
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100210/3aafdd07/attachment-0002.html 

From frank at carmickle.com  Tue Feb  9 18:46:39 2010
From: frank at carmickle.com (Frank Carmickle)
Date: Tue, 9 Feb 2010 21:46:39 -0500
Subject: [Freeswitch-users] ACL question and js error
In-Reply-To: <cfe871e3305675257b8c74ed4fb5a11b.squirrel@www.webcontracts.co.uk>
References: <ea05d45dca9be2d979fb386f960bc077.squirrel@www.webcontracts.co.uk>
	<20100207145907.GF31942@base.carmickle.com>
	<cfe871e3305675257b8c74ed4fb5a11b.squirrel@www.webcontracts.co.uk>
Message-ID: <20100210024638.GN31942@base.carmickle.com>

On Tue, Feb 09, Matthew Law wrote:
> On Sun, February 7, 2010 2:59 pm, Frank Carmickle wrote:
> > On Sun, Feb 07, Matthew Law wrote:
> >> After some more experiments I have a working replacement for the
> >> asterisk
> >> box we were using before, which is great.
> >>
> >> I had problems getting incoming calls to work. Changing the entry in
> >> acl.conf.xml from:
> >>
> >> <list name="domains" default="deny">
> >>    <node type="allow" domain="$${domain}"/>
> >> </list>
> >>
> >> to:
> >>
> >> <list name="domains" default="allow">
> >>    <node type="allow" domain="$${domain}"/>
> >> </list>
> >>
> >> and reloading xml works but this gets reverted every time FS starts up.
> >> I've scanned the wiki docs and can't see anything pertaining to that.
> >> Why/where is this happening and how do I make it the default?  Actually,
> >> the question should probably be is it sensible to do that? - the box is
> >> out on the internet and I really only want to take incoming calls from
> >> voiptalk.org, but I can't find a list of IPs on their site which I could
> >> create an acl from...
> >
> > This is what gateway definitions are for in sofia.
> 
> I'm still struggling with this.  How where do I tell sofia to allow
> incoming connections from this gateway?
> 
> Here's my sip_profiles/external/voiptalk.org.xml with the sensitive stuff
> removed:
> 
> <include>
>         <gateway name="voiptalk">
>                 <param name="username" value="XXXXXXX" />
>                 <param name="realm" value="voiptalk.org" />
>                 <param name="from-user" value="XXXXXXX"/>
>                 <param name="from-domain" value="voiptalk.org"/>
>                 <param name="password" value="XXXXXXX" />
>                 <param name="extension" value="voiptalk-incoming"/>
>                 <param name="proxy" value="voiptalk.org"/>
>                 <param name="register-proxy" value="voiptalk.org"/>
>                 <param name="expire-seconds" value="600"/>
>                 <param name="register" value="true"/>
>                 <param name="ping" value="25"/>
>         </gateway>
> </include>
> 
> Do I need to add something to this file or maybe sofia.conf.xml to allow
> connections from this domain?  Most everything else is working now, just
> banging my head on this.

After doing a 
sofia profile $profile rescan reloadxml

it still doesn't work?  Are you sure it isn't hitting the dialplan and failing?  I have never used

<param name="extension" value="voiptalk-incoming"/>

I usually leave it commented and then match on the  destination in the dialplan.

<condition field="destination_number" expression="^(4124134655|4128484655|19734226137)$">

HTH
--FC



From andrew at hijacked.us  Tue Feb  9 18:56:07 2010
From: andrew at hijacked.us (Andrew Thompson)
Date: Tue, 9 Feb 2010 21:56:07 -0500
Subject: [Freeswitch-users] Passing channel variables to mod_fifo and
	xml_cdr
In-Reply-To: <87f2f3b91002091455v6af079e1ie28ed6891d5ed628@mail.gmail.com>
References: <004301caa608$534747d0$f9d5d770$@net>
	<87f2f3b91002091455v6af079e1ie28ed6891d5ed628@mail.gmail.com>
Message-ID: <20100210025607.GE27785@hijacked.us>

On Tue, Feb 09, 2010 at 02:55:11PM -0800, Michael Collins wrote:
> On Thu, Feb 4, 2010 at 6:09 PM, Adam Ford <lists at redbonez.net> wrote:
> 
> >  When sending a call through mod_fifo I seem to be losing my custom
> > channel variables that were assigned during prior processing of the call.
> > In my example, I am trying to assign a unique identifier at the time the
> > call enters my FreeSWITCH system in order to more easily tie the xml_cdr
> > logs together.  This works great, until a call is processed through
> > mod_fifo, which drops my custom channel variable in the calls that it
> > generates.  Is it likely that I have something wrong with my config? Or does
> > mod_fifo not support the passing of custom channel variables?
> >
> >
> >
> > The overall problem I am trying to solve is that mod_fifo generates a
> > separate a-leg for every time it rings an agent.  If the agent answers, the
> > a-leg log gets tied to the associated b-leg log with the uuids and I am able
> > to see the entire call in xml_cdr.  However, if the agent rejects the call
> > or doesn?t answer, the a-leg is abandoned with seemingly no association back
> > to a b-leg log.  Anyone have a better suggestion for tying all these xml_cdr
> > logs together for a full picture of the call?
> >
> 
> Just curious - are you looking at this from the caller's perspective or the
> agent's perspective? An unanswered/rejected call from FIFO to an agent
> doesn't tell you very much. However, if you're trying to gather statistics
> on an individual agent then I could see why you'd want to know how many FIFO
> calls they failed to answer. As far as the "new" A leg not being tied back
> to a B leg - Mike J is 100% correct: A FIFO call to an agent has absolutely
> no correlation to any caller waiting in queue. (I suppose the only exception
> to this rule would be if there was only one caller in the queue when the
> FIFO called out to the agents.)
> 
> Like Mike said, FIFO is not ACD. FIFO is "get the caller to a human as
> efficiently and quickly as possible." ACD is more of "connect the longest
> waiting caller to the longest waiting agent, with possible exceptions for
> skills, etc." Check out Andrew Thompson's SpiceCSM for a possible solution
> to your scenario.
> 
> http://www.opencsm.org/wiki/index.php/SpiceCSM_Community_Edition
>

Actually http://github.com/Vagabond/OpenACD is the place to go now
(although its pretty short on documentation right now). I actually
deployed it last friday and its been running fairly well since then.
After I get a few remaining TODOs out of the way I'll cut a 1.0 RC1 and
do a real announcement (and write some documentation).

On a related note, OpenACD tracks agent ringouts and ring cancels so you
can see if an agent is being lazy and bouncing calls.

Andrew



From spiritonly at gmail.com  Tue Feb  9 19:14:46 2010
From: spiritonly at gmail.com (=?UTF-8?B?5p2o5rGf6aqF?=)
Date: Wed, 10 Feb 2010 11:14:46 +0800
Subject: [Freeswitch-users] How to exchange media when I developed new
	endpoint module?
Message-ID: <93b0f8ce1002091914q581f12fvd1de01e2181f0f3a@mail.gmail.com>

Hi,
    I am developping a new endpoint module, now I can make an inbound call
and execute IVR.
When I make an outbound call and bridge the inbound leg and outbound leg, I
receive remote alerting and pickup remote phone but there isn't
any voice exchange.
    So how to exchange media next?
----------------------------------------------------------------------
gtalk:    spiritonly at gmail.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100210/f743acb0/attachment-0002.html 

From jmesquita at freeswitch.org  Tue Feb  9 19:32:28 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Wed, 10 Feb 2010 01:32:28 -0200
Subject: [Freeswitch-users] How to exchange media when I developed new
	endpoint module?
In-Reply-To: <93b0f8ce1002091914q581f12fvd1de01e2181f0f3a@mail.gmail.com>
References: <93b0f8ce1002091914q581f12fvd1de01e2181f0f3a@mail.gmail.com>
Message-ID: <ebbc2d551002091932u52dbf1b6jfecf2dfdb43b8a7c@mail.gmail.com>

You should look at read_frame and write_frame implementations of other
endpoint modules.

This should pretty much tell you how things work...

Jo?o Mesquita


On Wed, Feb 10, 2010 at 1:14 AM, ??? <spiritonly at gmail.com> wrote:

> Hi,
>     I am developping a new endpoint module, now I can make an inbound call
> and execute IVR.
> When I make an outbound call and bridge the inbound leg and outbound leg, I
> receive remote alerting and pickup remote phone but there isn't
> any voice exchange.
>     So how to exchange media next?
> ----------------------------------------------------------------------
> gtalk:    spiritonly at gmail.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100210/86caf75f/attachment-0002.html 

From brian at freeswitch.org  Tue Feb  9 19:44:26 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 9 Feb 2010 21:44:26 -0600
Subject: [Freeswitch-users] How to exchange media when I developed new
	endpoint module?
In-Reply-To: <ebbc2d551002091932u52dbf1b6jfecf2dfdb43b8a7c@mail.gmail.com>
References: <93b0f8ce1002091914q581f12fvd1de01e2181f0f3a@mail.gmail.com>
	<ebbc2d551002091932u52dbf1b6jfecf2dfdb43b8a7c@mail.gmail.com>
Message-ID: <4021C303-1AC3-48F2-A17B-63C02C1C54D4@freeswitch.org>

But the bigger question is what protocol are you doing that you have to create your own endpoint module?  

/b

On Feb 9, 2010, at 9:32 PM, Jo?o Mesquita wrote:

> You should look at read_frame and write_frame implementations of other endpoint modules.
> 
> This should pretty much tell you how things work...
> 
> Jo?o Mesquita




From intralanman at freeswitch.org  Tue Feb  9 20:00:22 2010
From: intralanman at freeswitch.org (Raymond Chandler)
Date: Tue, 9 Feb 2010 23:00:22 -0500
Subject: [Freeswitch-users] How to record the call upon successful bridge
In-Reply-To: <13529f9d1002091819y2b363f50vc314d1b6197503b2@mail.gmail.com>
References: <13529f9d1002091819y2b363f50vc314d1b6197503b2@mail.gmail.com>
Message-ID: <E66ACC04-AFFA-4052-8707-C583ACEDDF7B@freeswitch.org>


On Feb 9, 2010, at 9:19 PM, Jingwei Yang wrote:

> Hi,
> 
> I'm using uuid_bridge to bridge two calls. May I know how to start recording only when the bridge succeeds?
> 

try setting api_hangup_hook to session record

-Ray


From intralanman at freeswitch.org  Tue Feb  9 20:01:03 2010
From: intralanman at freeswitch.org (Raymond Chandler)
Date: Tue, 9 Feb 2010 23:01:03 -0500
Subject: [Freeswitch-users] How to record the call upon successful bridge
In-Reply-To: <13529f9d1002091819y2b363f50vc314d1b6197503b2@mail.gmail.com>
References: <13529f9d1002091819y2b363f50vc314d1b6197503b2@mail.gmail.com>
Message-ID: <257401E3-B32F-4699-96F6-415BA2ADDE88@freeswitch.org>

errr... .execute_on_answer might be better ;-)
-Ray 

On Feb 9, 2010, at 9:19 PM, Jingwei Yang wrote:

> Hi,
> 
> I'm using uuid_bridge to bridge two calls. May I know how to start recording only when the bridge succeeds?
> 
> Thanks,
> -Jingwei
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From mike at jerris.com  Tue Feb  9 20:09:26 2010
From: mike at jerris.com (Michael Jerris)
Date: Tue, 9 Feb 2010 23:09:26 -0500
Subject: [Freeswitch-users] Announcement: FreeSWITCH Add Broadsoft SCA
	Support
In-Reply-To: <767B677E-0D19-43A9-89DE-D96EC4A715EC@freeswitch.org>
References: <87f2f3b91001111742k691966cfs82515db77c13d4d4@mail.gmail.com>
	<d361ca7b1002082110h4e3c60d0p88412f753444ec3d@mail.gmail.com>
	<016d01caa98e$f6df25f0$e49d71d0$@com>
	<d361ca7b1002090817h4aa3c13dka4d4db8b4adc7606@mail.gmail.com>
	<1C0A1BEE-3480-4A33-916F-52067AF5B9C9@freeswitch.org>
	<d361ca7b1002090838j6c42dcd2of8affb5f281900fd@mail.gmail.com>
	<767B677E-0D19-43A9-89DE-D96EC4A715EC@freeswitch.org>
Message-ID: <8D062B6E-4691-4BA6-B038-0EC6E8642DDA@jerris.com>

will it work with multi-domain if you have the aliases for all the domains?

Mike

On Feb 9, 2010, at 12:02 PM, Brian West wrote:

>     <!--<param name="dbname" value="share_presence"/>-->
>     <!--<param name="presence-hosts" value="$${domain}"/>-->
> 
> On Feb 9, 2010, at 10:38 AM, Yehavi Bourvine wrote:
> 
>> What do you mean by "bonding them"?
>>  
>> 2010/2/9 Brian West <brian at freeswitch.org>
>> It will work across profiles if you bond them.  :P

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100209/3deca515/attachment-0002.html 

From mike at jerris.com  Tue Feb  9 20:13:59 2010
From: mike at jerris.com (Michael Jerris)
Date: Tue, 9 Feb 2010 23:13:59 -0500
Subject: [Freeswitch-users] Any Known Dual-Registration Issue?
In-Reply-To: <035001caa9c8$2dca81c0$895f8540$@com>
References: <5AD33F8B1F0848CF9C3A314A60039366@greyhawk.tonecommander.com>
	<035001caa9c8$2dca81c0$895f8540$@com>
Message-ID: <E54E66C0-20C4-4472-B746-29CE498837E8@jerris.com>

As a note, this can cause issues too if you have multiple legitimate registrations from the same ip and port, beware unless you understand the consequences.

Mike

On Feb 9, 2010, at 3:40 PM, Peder wrote:

> What kind of phones?  If you have multiple registartion, this can happen
> sometimes if you reboot a phone.  Crappy phones, like Grandstream, don't
> un-register when you reboot and then when they come back up, they register
> again and thus two registrations until the lifetime of the registration ends
> and it gets flushed.  Changing the multiple-registration to contact can help
> as I believe that uses port and source IP as part of the registration info:
> 
> <param name="multiple-registrations" value="contact"/>




From pablosaro at gmail.com  Tue Feb  9 20:23:35 2010
From: pablosaro at gmail.com (Pablo Hernan Saro)
Date: Wed, 10 Feb 2010 01:23:35 -0300
Subject: [Freeswitch-users] Question regarding testing IVRs
Message-ID: <247f8101002092023t3f30a600o1a9f43771c879e61@mail.gmail.com>

Hi there,

I was wondering if anyone knows about an automated tool for testing IVR
systems... Or should I use SIPP for this purpose?
I will really appreciate your inputs.
Regards

Pablo
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100210/e28d8555/attachment-0002.html 

From brian at freeswitch.org  Tue Feb  9 20:30:06 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 9 Feb 2010 22:30:06 -0600
Subject: [Freeswitch-users] Announcement: FreeSWITCH Add Broadsoft SCA
	Support
In-Reply-To: <8D062B6E-4691-4BA6-B038-0EC6E8642DDA@jerris.com>
References: <87f2f3b91001111742k691966cfs82515db77c13d4d4@mail.gmail.com>
	<d361ca7b1002082110h4e3c60d0p88412f753444ec3d@mail.gmail.com>
	<016d01caa98e$f6df25f0$e49d71d0$@com>
	<d361ca7b1002090817h4aa3c13dka4d4db8b4adc7606@mail.gmail.com>
	<1C0A1BEE-3480-4A33-916F-52067AF5B9C9@freeswitch.org>
	<d361ca7b1002090838j6c42dcd2of8affb5f281900fd@mail.gmail.com>
	<767B677E-0D19-43A9-89DE-D96EC4A715EC@freeswitch.org>
	<8D062B6E-4691-4BA6-B038-0EC6E8642DDA@jerris.com>
Message-ID: <9D328485-D4C3-4B14-808A-CB63BBF06DC1@freeswitch.org>

In theory it should.  I haven't tested that but it should work the same.

/b

On Feb 9, 2010, at 10:09 PM, Michael Jerris wrote:

> will it work with multi-domain if you have the aliases for all the domains?
> 
> Mike




From brian at freeswitch.org  Tue Feb  9 20:30:52 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 9 Feb 2010 22:30:52 -0600
Subject: [Freeswitch-users] Question regarding testing IVRs
In-Reply-To: <247f8101002092023t3f30a600o1a9f43771c879e61@mail.gmail.com>
References: <247f8101002092023t3f30a600o1a9f43771c879e61@mail.gmail.com>
Message-ID: <857B3E7B-0ECA-4B6D-A72E-C0258BCFF24E@freeswitch.org>

Nope.  Unless you know exactly what you're doing its a waste of time.

/b

On Feb 9, 2010, at 10:23 PM, Pablo Hernan Saro wrote:

> Hi there,
> 
> I was wondering if anyone knows about an automated tool for testing IVR systems... Or should I use SIPP for this purpose?
> I will really appreciate your inputs.
> Regards
> 
> Pablo




From mike at jerris.com  Tue Feb  9 20:56:46 2010
From: mike at jerris.com (Michael Jerris)
Date: Tue, 9 Feb 2010 23:56:46 -0500
Subject: [Freeswitch-users] Way to pass max-members from dialplan into
	conference?
In-Reply-To: <87f2f3b91002091509w56eacdedt6a291d9b98838521@mail.gmail.com>
References: <C7136E29C3FD472BB367FF412E282BED@greyhawk.tonecommander.com>
	<87f2f3b91002091509w56eacdedt6a291d9b98838521@mail.gmail.com>
Message-ID: <8965B012-D22D-4155-B76B-CC026DA9D884@jerris.com>

or:

svn commit -m"mod_conference add conference_max_members channel variable that can be set on the first channel calling a conference to override the profiles max-members param"
Sending        mod_conference/mod_conference.c
Transmitting file data .
Committed revision 16597.

compile tested, for your building pleasure.

Mike

On Feb 9, 2010, at 6:09 PM, Michael Collins wrote:

> 
> 
> On Tue, Feb 9, 2010 at 1:14 PM, Robert Hadley <robert.hadley at teotech.com> wrote:
> I have setting max-members=10 in conference.conf.xml working.  However, is there are way to pass in the max-members=10 from the dialplan/default.xml to mod_conference?  I tried using action application=?set? data=?max-members=10? but it didn?t work.  Also tried action application=?export? data=?max-members=10? but it didn?t work either.
> 
>  
> From default.xml:
> 
>     <extension name="nb_conferences">
> 
>       <condition field="destination_number" expression="^(30\d{2})$">
> 
>        <action application="answer"/>
> 
> <!-- tried set and export here to no avail -->
> 
>        <action application="conference" data="$1-${domain_name}@default"/>
> 
>  
> Thanks,
> 
> Robert
> 
> 
> Robert,
> 
> I checked with Brian and also took a look inside mod_conference.c. I didn't see any way that you could override the conference params that are contained in the conference profile. So you'll either need to make a new profile or join the big leagues and start trying out mod_xml_curl. :)
> 
> -MC

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100209/23a0b9f3/attachment-0002.html 

From nagalenoj at gmail.com  Tue Feb  9 21:57:30 2010
From: nagalenoj at gmail.com (Nagalenoj H.)
Date: Wed, 10 Feb 2010 11:27:30 +0530
Subject: [Freeswitch-users] Play music to A leg.
In-Reply-To: <87f2f3b91002090737l4efa8en29bde9457fef7ea5@mail.gmail.com>
References: <fe67a3341002090519r2cc6e8edm7702b8d012fc2be3@mail.gmail.com>
	<87f2f3b91002090737l4efa8en29bde9457fef7ea5@mail.gmail.com>
Message-ID: <fe67a3341002092157x7484f3d2id6678510bbec66cc@mail.gmail.com>

Because, I want to get some digits before bridging the legs. I've tried
group_confirm_key, but it accepts only one digit, I need multiple digits, so
I can't use.
I've also tried group_confirm_file, but when I do originate for multiple
extensions, I want this script to work based on the answered extension.

So, I've originated and processed the events to do my job.

How do I play some music to A leg?

On Tue, Feb 9, 2010 at 9:07 PM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Tue, Feb 9, 2010 at 5:19 AM, Nagalenoj H. <nagalenoj at gmail.com> wrote:
>
>> Dear friends,
>>   In event socket, I'm originating a call to a number from A leg and till
>> the person answers the call, I would want to play some music to the A leg,
>> till I bridge these A leg and originated call.
>>
>>   I don't want to use bridge, in which I could use ringback.
>>
> You don't want to use bridge because... why?
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Regards,
Nagalenoj H.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100210/23fd3b23/attachment-0002.html 

From rm at callrica.co.za  Tue Feb  9 23:31:22 2010
From: rm at callrica.co.za (Roly Maz)
Date: Wed, 10 Feb 2010 09:31:22 +0200
Subject: [Freeswitch-users] Can't access event socket from 0.0.0.0
Message-ID: <00f501caaa23$26068820$72139860$@co.za>

 

I am trying to execute commands over telnet from my XP box  to my Debian FS
box. I have modified the event_socket.conf.xml so that the Listen IP is
0.0.0.0, as per wiki

 

When i try telnet i get a Connect Failed. 

 

If i run netstat on the FS box it shows 127.0.0.1 is listening on port 8021?
It also shows my FS static IP 10.0.18.244 is listening on 5060, which is
correct.

 

I am using the default conf on Freeswitch version 1.0.trunk 16590M  Am i
missing a setting?

 

Any pointers would be much appreciated

 

Roly

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100210/92540339/attachment-0002.html 

From tim at novion.ru  Wed Feb 10 00:35:06 2010
From: tim at novion.ru (Timur Valishev)
Date: Wed, 10 Feb 2010 11:35:06 +0300
Subject: [Freeswitch-users] No reINVITE when bridging two sessions from
	JavaScript with bypass_media_after_bridge=true
In-Reply-To: <154214DA-5781-4FC1-81EA-B41392E6ACF6@jerris.com>
References: <8e9d67561002041147q6c135546u1f7e1887529a916d@mail.gmail.com>
	<3D739D7F-F5C6-4266-B04E-9413DCFDA6AF@freeswitch.org>
	<8e9d67561002050018q24e3a806kc8807b55268e759b@mail.gmail.com>
	<B66289FF-C2A9-4AC4-A56C-66E51B4726F4@freeswitch.org>
	<8e9d67561002051202qeaf1551v8fdd29565f8c02fd@mail.gmail.com>
	<154214DA-5781-4FC1-81EA-B41392E6ACF6@jerris.com>
Message-ID: <8e9d67561002100035n2d14c74dj83f784713c59d542@mail.gmail.com>

Dear Mike,

I'm trying to build a kind of complicated callback. I will be happy if
you suggest alternative way to do it!

Scenario is:

0. Get command over the socket to initiate connection, get A-number,
B-number, route preference, Caller ID option (incognito/normal)
1. Call billing stored procedure to determine maximum call duration
2. Reply over the socket (or better through database?) that connection
is in progrees (to display it on the user GUI)
3. Initiate connection to A-number
4. Upon connection to A, say welcome, start calling B, play ringback tone
5. Upon dialling B, call billing stored procedure to report that
session state changed and report the status to user GUI (over the
socket or through the database?)
5.1 If there was error during connection - speak the reason to end
user (e.g. "Number busy" or "Timeout expired" etc.)
6. Join peers in bypass media mode
6.1 Wait for various commands over the socket - e.g. transfer the
call, put on hold, join to conference etc. User will have GUI for that
operations.
7. If B hangs up, call billing stored procedure to finalize session
and calculate the cost of the call. Cost is to be calculated only by
B-leg length. Speak to A the cost of the call, say thanks and hang up.
7.1 If A hangs up, just call billing and terminate.

Best regards,
Timur Valishev

2010/2/10 Michael Jerris <mike at jerris.com>:
> controlling multiple calls in a script like this is tricky, you need to use
> the first session to create the second one. ?Why are you not just doing an
> originate to do all of this not even in a js file? ?What exactly are you
> trying to accomplish
> Mike
> On Feb 5, 2010, at 3:02 PM, Timur Valishev wrote:
>
> I think we are on the right way) still does not work, but there is hope)
> First of all, this script does not produce any reinvite either (even if
> replace?bypass_media to?bypass_media_after_bridge, or set?bypass_media only
> on one channel):
> <<<<<<<<<<<<<<<<<<<<<<<<<<<<<
> session = new
> Session("{bypass_media=true,ignore_early_media=true,hangup_after_bridge=true}user/1001");
> session2 = new
> Session("{bypass_media=true,ignore_early_media=true}?user/1001");
> bridge(session, session2);
>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>
> BUT! if I run the following script:
> <<<<<<<<<<<<<<<<<<<<<<<<<<<<<
> session = new
> Session("{bypass_media=true,ignore_early_media=true,hangup_after_bridge=true}?user/1001");
> session2 = new
> Session("{bypass_media=true,ignore_early_media=true}user/1001");
>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>
> And then manually type in the console
> uuid_media off <uuid of my session>
> - then I get the reINVITE!
> BUT! When I try to write it to the script:
> <<<<<<<<<<<<<<<<<<<<<<<<<<<<<
> session = new
> Session("{bypass_media=true,ignore_early_media=true,hangup_after_bridge=true}sofia/external/timwork at novion.ru");
> session2 = new
> Session("{bypass_media=true,ignore_early_media=true}sofia/external/timwork at novion.ru");
> bridge(session, session2);
> apiExecute('uuid_media off '+session.uuid); // <-- this line is not
> executed, because bridge hangs up untill BYE
>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>
> the last line is not executed, because bridge hangs up untill BYE
> Then I've tried to do like this:
> <<<<<<<<<<<<<<<<<<<<<<<<<<<<<
> session = new
> Session("{bypass_media=true,ignore_early_media=true,hangup_after_bridge=true}user/1001");
> session2 = new
> Session("{bypass_media=true,ignore_early_media=true}user/1001");
> session.setAutoHangup(false)
> session2.setAutoHangup(false)
> apiExecute("uuid_bridge "+session.uuid+" "+session2.uuid);
> apiExecute('uuid_media off '+session.uuid);
>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>
> But sessions do not get bridged -( Even if I insert session.ready() after
> each call.
> Any ideas on how to call the functions correctly to get the reINVITE?
> Best regards,
> Timur Valishev
> 2010/2/5 Brian West <brian at freeswitch.org>
>>
>> set it inside each of the {} for each session you create its not set after
>> the fact the call is up already... ?you're setting it too late.
>> you an also issue uuid_media off <uuid>
>> /b
>> On Feb 5, 2010, at 2:18 AM, Timur Valishev wrote:
>>
>> I've?modified my script to make sure: <<<<<<<<<<<<<<<<<<<<<<<<<<<<,
>> session = new Session(
>>
>> "{ignore_early_media=true,hangup_after_bridge=true}sofia/external/timwork at novion.ru"
>> );
>> session2 = new Session(
>> "{ignore_early_media=true}sofia/external/timwork at novion.ru"
>> );
>> session.setVariable('bypass_media', 'true');
>> session2.setVariable('bypass_media', 'true');
>> bridge(session, session2);
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From jingwei.yang at gmail.com  Wed Feb 10 00:39:41 2010
From: jingwei.yang at gmail.com (Jingwei Yang)
Date: Wed, 10 Feb 2010 16:39:41 +0800
Subject: [Freeswitch-users] How to record the call upon successful bridge
In-Reply-To: <257401E3-B32F-4699-96F6-415BA2ADDE88@freeswitch.org>
References: <13529f9d1002091819y2b363f50vc314d1b6197503b2@mail.gmail.com>
	<257401E3-B32F-4699-96F6-415BA2ADDE88@freeswitch.org>
Message-ID: <13529f9d1002100039q43775563r49b11136ddda0a0@mail.gmail.com>

Hi Ray,

Thanks a lot for the replies. Allow me to elaborate a little bit about my
situation.

1. client A calls in and parks at Fifo myq.
2. FS connets Agent B to an extension (via originate
skypiax/ANY/jingwei.yang 33333)
3. uuid_bridge client A and agent B

I believe the spice I can add is in the extension 33333. Here's how I define
it.

    <extension name="agent_hold">
      <condition field="destination_number" expression="^33333$">
        <action application="set" data="*execute_on_answer*=record_session
/tmp/33333.wav"/>
        <action application="answer"/>
        <action application="sleep" data="2000"/>
        <action application="info"/>
        <action application="playback"
data="ivr/ivr-hold_connect_call.wav"/>
      </condition>
    </extension>

But the wav file didn't get generated at all. Please advise whether the
above is in correct usage.

I've also tried bridge_pre_execute_bleg_app and bridge_pre_execute_bleg_data

        <action application="set"
data="bridge_pre_execute_bleg_app=record_session"/>
        <action application="set"
data="bridge_pre_execute_bleg_data=/tmp/33333.wav"/>

The audio file didn't appear either.

The only successful method is by using record_session directly like this:

        <action application="record_session" data="/tmp/33333.wav"/>

However, in this form, the wav file includes the waiting music, which is not
ideal.

Thanks and best regards,
-Jingwei

On Wed, Feb 10, 2010 at 12:01 PM, Raymond Chandler <
intralanman at freeswitch.org> wrote:

> errr... .execute_on_answer might be better ;-)
> -Ray
>
> On Feb 9, 2010, at 9:19 PM, Jingwei Yang wrote:
>
> > Hi,
> >
> > I'm using uuid_bridge to bridge two calls. May I know how to start
> recording only when the bridge succeeds?
> >
> > Thanks,
> > -Jingwei
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100210/1493da48/attachment-0002.html 

From xanlich at gmail.com  Wed Feb 10 01:17:30 2010
From: xanlich at gmail.com (Chia-Yen Wu)
Date: Wed, 10 Feb 2010 17:17:30 +0800
Subject: [Freeswitch-users] Lua script hangup detect
Message-ID: <314dc3f81002100117h6b1bc49cna989624a1ae4e0e2@mail.gmail.com>

Hello,
i tried to use Lua script to replace xml macro in dialplan,
but I found out that Lua wont terminate if client hangup,
,so the session is still on but client is already hangup,
is there a way to avoid this ?
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100210/39f27e2f/attachment-0002.html 

From kond at nstel.ru  Wed Feb 10 01:40:14 2010
From: kond at nstel.ru (Nikolay Kondratyev)
Date: Wed, 10 Feb 2010 12:40:14 +0300
Subject: [Freeswitch-users] can't see sip messages in the log file
Message-ID: <20100210094014.C746F11F81@mail.nstel.ru>

Hi all,

I have a little problem with FreeSWITCH Version 1.0.5-20100209-0400
(16587M).

I'd like to see SIP messages in the log file. and I tried 

sofia profile internal siptrace on

sofia loglevel all 9

console loglevel 9

but alas i can only see what SDP is used..

The configuration is default with the exception for some new local
extensions and mod_h323 compiled in.

 Am I missing something?

 

Can anybody please advise how to include sip messages into the log file?

(I thought that "sofia profile internal siptrace on" should be enough, but
alas..)

 

Thanks in advance,

Nikolay.

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100210/4c31d3cf/attachment-0002.html 

From jason at jasonjgw.net  Wed Feb 10 02:02:43 2010
From: jason at jasonjgw.net (Jason White)
Date: Wed, 10 Feb 2010 21:02:43 +1100
Subject: [Freeswitch-users] can't see sip messages in the log file
In-Reply-To: <20100210094014.C746F11F81@mail.nstel.ru>
References: <20100210094014.C746F11F81@mail.nstel.ru>
Message-ID: <20100210100243.GA16435@jdc.jasonjgw.net>

Nikolay Kondratyev <kond at nstel.ru> wrote:
> Can anybody please advise how to include sip messages into the log file?

<param name="sip-trace" value="yes"/>
in the SIP profile you want to trace, then
sofia profile profile-name restart reloadxml
or restarting FreeSWITCH should do it.




From rm at callrica.co.za  Wed Feb 10 02:12:27 2010
From: rm at callrica.co.za (Roly Maz)
Date: Wed, 10 Feb 2010 12:12:27 +0200
Subject: [Freeswitch-users] Can't access event socket from 0.0.0.0
In-Reply-To: <00f501caaa23$26068820$72139860$@co.za>
References: <00f501caaa23$26068820$72139860$@co.za>
Message-ID: <012601caaa39$a79ca840$f6d5f8c0$@co.za>

I have figured this out...newbie Linux error -I was making my changes in the
usr/src/freeswitch folder - i should have been using the
usr/local/freeswitch folder.  

 

Duh!

 

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Roly Maz
Sent: 10 February 2010 09:31 AM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Can't access event socket from 0.0.0.0

 

 

I am trying to execute commands over telnet from my XP box  to my Debian FS
box. I have modified the event_socket.conf.xml so that the Listen IP is
0.0.0.0, as per wiki

 

When i try telnet i get a Connect Failed. 

 

If i run netstat on the FS box it shows 127.0.0.1 is listening on port 8021?
It also shows my FS static IP 10.0.18.244 is listening on 5060, which is
correct.

 

I am using the default conf on Freeswitch version 1.0.trunk 16590M  Am i
missing a setting?

 

Any pointers would be much appreciated

 

Roly

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100210/779f53e4/attachment-0002.html 

From jingwei.yang at gmail.com  Wed Feb 10 02:38:18 2010
From: jingwei.yang at gmail.com (Jingwei Yang)
Date: Wed, 10 Feb 2010 18:38:18 +0800
Subject: [Freeswitch-users] Can't access event socket from 0.0.0.0
In-Reply-To: <012601caaa39$a79ca840$f6d5f8c0$@co.za>
References: <00f501caaa23$26068820$72139860$@co.za>
	<012601caaa39$a79ca840$f6d5f8c0$@co.za>
Message-ID: <13529f9d1002100238p55bd97c2mb5b64f3a6d3c9521@mail.gmail.com>

Totally understandable. I've made the same mistake before :)

On Wed, Feb 10, 2010 at 6:12 PM, Roly Maz <rm at callrica.co.za> wrote:

>  I have figured this out...newbie Linux error ?I was making my changes in
> the usr/src/freeswitch folder ? i should have been using the
> usr/local/freeswitch folder.
>
>
>
> Duh!
>
>
>
>
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Roly Maz
> *Sent:* 10 February 2010 09:31 AM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* [Freeswitch-users] Can't access event socket from 0.0.0.0
>
>
>
>
>
> I am trying to execute commands over telnet from my XP box  to my Debian FS
> box. I have modified the event_socket.conf.xml so that the Listen IP is
> 0.0.0.0, as per wiki
>
>
>
> When i try telnet i get a Connect Failed.
>
>
>
> If i run netstat on the FS box it shows 127.0.0.1 is listening on port
> 8021? It also shows my FS static IP 10.0.18.244 is listening on 5060, which
> is correct.
>
>
>
> I am using the default conf on Freeswitch version 1.0.trunk 16590M  Am i
> missing a setting?
>
>
>
> Any pointers would be much appreciated
>
>
>
> Roly
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100210/985b5365/attachment-0002.html 

From vmaruani at interwise.com  Wed Feb 10 02:44:24 2010
From: vmaruani at interwise.com (Victor Maruani)
Date: Wed, 10 Feb 2010 12:44:24 +0200
Subject: [Freeswitch-users] Bypass-media and REFER method
In-Reply-To: <B816EF85C732094BA2016B2112F3DC2B0402C777@isr-brass.interwise.com>
References: <B816EF85C732094BA2016B2112F3DC2B0402C777@isr-brass.interwise.com>
Message-ID: <B816EF85C732094BA2016B2112F3DC2B040C5CA6@isr-brass.interwise.com>

Hi,

 

I can't have a blind transfer work properly if I use bypass-media=true.

 

My first message may have been unclear, here I added excerpt from the
dialplan:

 

                <!--- ext MyIVR -->

                 <extension name="Extension MyIVR" continue="false">

                   <condition field="destination_number"
expression="^337$">

                                   <action application="set"
data="bypass_media=true"/>

                                   <action application="bridge"
data="sofia/internal/337 at 10.10.5.51"/>

                   </condition>

                 </extension>

 

 

                <!--- ext MyServ -->

                 <extension name="Extension MyServ " continue="false">

                   <condition field="destination_number"
expression="^3341$">

                                                <action
application="set" data="bypass_media=true"/>

                                   <action application="bridge"
data="sofia/internal/3341 at 10.10.5.48"/>

                   </condition>

                 </extension>   

 

The connection to MyIVR works. Then it sends Refer-to (3341 at ...) which
fails as I described it in the previous mail.

 

I would like to know if the feature has been validated and if I'm
missing something in the configuration.

 

Any help would be very appreciated.

 

Thanks!

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Victor Maruani
Sent: Sunday, February 07, 2010 5:01 PM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Bypass-media and REFER method

 

Hi,

 

I'm trying to do a POC using FS, the goal is to have FS handle REFERs
containing proprietary data. 

I want to have some logic on top of FS and also use the fail over
mechanism.

in short, I have something like this:

(third party) A side  --- FS ---- B side (IVR server)

 

the IVR the sends a REFER to FS. I don't want A to deal with it. 

now say B refers to C, it would be considered as a "group" C1, C2 ... to
which I want FS to failover.

only when one has answered should A be updated (REINVITE) and B notified
and disconnected.

if all fails I would expect B to be notified of the failure and proceed
as I wish without "losing" A.

 

from what I've read FS should be OK for the job but I have a couple
issues:

 

1 ) I have some issues getting FS handle a REFER while in bypass-media
mode.

(I tried with the release and some revisions including latest)

first when I bridge A and B everything is fine and media is bypassed.

When B sends REFER to C:

-           FS immediately NOTIFY B of success and send a reinvite to A
with SDP containing its own media IP/port.

-          then it does INVITE C with A's SDP.

-          B gets disconnected. A is not updated with C's sdp.

so at this point A sends RTP to FS and C sends RTP to A. ...

 

I basically have one extension for B: (set bypass-media and bridge to B)

and another extension to C which does the same actions.

what do you think I do wrong? 

 

 

2 ) how can I catch the REFER and set variables from it? (like ref-by or
ref-to)

in the dial plan I do catch the INVITE sent to C, but how to do it with
the REFER itself?

 

 

thanks for your help!

 

 

Best Regards,

Victor.

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100210/15e25966/attachment-0002.html 

From kond at nstel.ru  Wed Feb 10 02:49:05 2010
From: kond at nstel.ru (Nikolay Kondratyev)
Date: Wed, 10 Feb 2010 13:49:05 +0300
Subject: [Freeswitch-users] h323 - sip call is not working
Message-ID: <20100210104911.A632A11F49@mail.nstel.ru>

Hi all,

I compiled FreeSWITCH Version 1.0.5-20100209-0400 (16587M) with mod_h323.

When I call from h323 (Avaya IPOffice) to local fs extention (x-lite) I hear
ring back, but when x-lite picks up, he hears silence, while IPOffice user
continues to hear ringback.

The log is at the http://pastebin.freeswitch.org/12091

My configuration is almost default, several local extentions added, and
h323.conf from http://wiki.freeswitch.org/wiki/Mod_h323

5840 - user at IPOffice

2853 - x-lite registered at FS

IPOffice ip address: 172.23.14.2

FS ip address 172.23.22.49

 

Can anybody please advise how to solve that?

Is it a configuration or a software problem?

 

Thanks in advance,

Nikolay.

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100210/c4517f38/attachment-0002.html 

From nazim.agabekov at gmail.com  Wed Feb 10 02:58:29 2010
From: nazim.agabekov at gmail.com (Nazim Agabekov)
Date: Wed, 10 Feb 2010 14:58:29 +0400
Subject: [Freeswitch-users] Lua script hangup detect
In-Reply-To: <314dc3f81002100117h6b1bc49cna989624a1ae4e0e2@mail.gmail.com>
References: <314dc3f81002100117h6b1bc49cna989624a1ae4e0e2@mail.gmail.com>
Message-ID: <4B729155.7010708@gmail.com>

Hello,
Can you pastebin your script?

http://pastebin.freeswitch.org

On 02/10/2010 01:17 PM, Chia-Yen Wu wrote:
> Hello, 
> i tried to use Lua script to replace xml macro in dialplan,
> but I found out that Lua wont terminate if client hangup,
> ,so the session is still on but client is already hangup,
> is there a way to avoid this ?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100210/1f577801/attachment-0002.html 

From kond at nstel.ru  Wed Feb 10 03:00:08 2010
From: kond at nstel.ru (Nikolay Kondratyev)
Date: Wed, 10 Feb 2010 14:00:08 +0300
Subject: [Freeswitch-users] can't see sip messages in the log file
In-Reply-To: <20100210100243.GA16435@jdc.jasonjgw.net>
Message-ID: <20100210110008.25BFD12292@mail.nstel.ru>

Jason, thanks for the reply.
Isn't "sofia profile internal siptrace on" a command line equivalent of
<param name="sip-trace" value="yes"/>?

Any way I tried it, but with the same result.
I still don't see SIP.

Thanks and regards,
Nikolay.

> > Can anybody please advise how to include sip messages into the log file?
> 
> <param name="sip-trace" value="yes"/>
> in the SIP profile you want to trace, then
> sofia profile profile-name restart reloadxml
> or restarting FreeSWITCH should do it.




From lakindia89 at gmail.com  Wed Feb 10 03:07:31 2010
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Wed, 10 Feb 2010 16:37:31 +0530
Subject: [Freeswitch-users] How to kill multiple UUIDs.
Message-ID: <7d79b3931002100307h4d370a22l6f04cafa5bf5806d@mail.gmail.com>

Hi all,

My situation is
    A called to 1005 -- Which executes an ESL program.
    Now from the program I will made the parallel call using "api originate
[origination_uuid=uuid1]user/1006,[origination_uuid2]user/1001 &park()".
    UUID's are obtained from create_uuid.

    I'll then wait for the api to return, to check whether the call is
answered or rejected by the other end.
    But while I'm waiting, if A hangup the call, I just want to kill the
calls that are originated by my program.
    So I taught of using api_hang_up_hook and I set that variable to
uuid_kill uuid1 uuid2.
    But it only killed the uuid1.

Is there any other ways to kill multiple uuid's??
please help?
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100210/e374f564/attachment-0002.html 

From kond at nstel.ru  Wed Feb 10 03:08:01 2010
From: kond at nstel.ru (Nikolay Kondratyev)
Date: Wed, 10 Feb 2010 14:08:01 +0300
Subject: [Freeswitch-users] h323 - sip call is not working
In-Reply-To: <20100210104911.A632A11F49@mail.nstel.ru>
Message-ID: <20100210110812.01E6512112@mail.nstel.ru>

I forgot to add, that from tcpdump trace (and from log) one can see that FS
does not send answer to IPOffice.

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Nikolay
Kondratyev
Sent: Wednesday, February 10, 2010 1:49 PM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] h323 - sip call is not working

 

Hi all,

I compiled FreeSWITCH Version 1.0.5-20100209-0400 (16587M) with mod_h323.

When I call from h323 (Avaya IPOffice) to local fs extention (x-lite) I hear
ring back, but when x-lite picks up, he hears silence, while IPOffice user
continues to hear ringback.

The log is at the http://pastebin.freeswitch.org/12091

My configuration is almost default, several local extentions added, and
h323.conf from http://wiki.freeswitch.org/wiki/Mod_h323

5840 - user at IPOffice

2853 - x-lite registered at FS

IPOffice ip address: 172.23.14.2

FS ip address 172.23.22.49

 

Can anybody please advise how to solve that?

Is it a configuration or a software problem?

 

Thanks in advance,

Nikolay.

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100210/bdcf3ad6/attachment-0002.html 

From nazim.agabekov at gmail.com  Wed Feb 10 03:28:19 2010
From: nazim.agabekov at gmail.com (Nazim Agabekov)
Date: Wed, 10 Feb 2010 15:28:19 +0400
Subject: [Freeswitch-users] Lua script hangup detect
In-Reply-To: <314dc3f81002100117h6b1bc49cna989624a1ae4e0e2@mail.gmail.com>
References: <314dc3f81002100117h6b1bc49cna989624a1ae4e0e2@mail.gmail.com>
Message-ID: <4B729853.1090206@gmail.com>

Generally you could avoid this by checking session status. Below is an
example.

function selectAge (langId)

   session:flushDigits()

   digits = ""
   while ("" == digits) do

      if not session:ready() then return nil end

      digits = session:playAndGetDigits(2, 3, 1, 5000, "#",
              wav_base .. langId .. "/" .. age_prompt_wav,
              wav_base .. langId .. "/" .. age_incorrect_wav,
              "\\d+");

      log("info", "Got dtmf: ".. digits .."\n");
      if "" ~= digits then
         if (age_max < tonumber(digits) or age_min > tonumber(digits) ) then
             log("info", "Incorrect age: ".. digits ..".. retrying\n");
             session:streamFile (wav_base .. langId .. "/" ..
age_incorrect_wav)
             digits = ""
         end
      end

   end

   log("info", "Got age dtmf: ".. digits .."\n");
   return tonumber(digits)
end



On 02/10/2010 01:17 PM, Chia-Yen Wu wrote:
> Hello, 
> i tried to use Lua script to replace xml macro in dialplan,
> but I found out that Lua wont terminate if client hangup,
> ,so the session is still on but client is already hangup,
> is there a way to avoid this ?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100210/a8705c42/attachment-0002.html 

From brian at freeswitch.org  Wed Feb 10 06:24:45 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 10 Feb 2010 08:24:45 -0600
Subject: [Freeswitch-users] Lua script hangup detect
In-Reply-To: <4B729853.1090206@gmail.com>
References: <314dc3f81002100117h6b1bc49cna989624a1ae4e0e2@mail.gmail.com>
	<4B729853.1090206@gmail.com>
Message-ID: <3302DFBC-8D82-460C-BC9A-11D309E50012@freeswitch.org>

Just wrap it in a while(session:ready() == true)

/b



On Feb 10, 2010, at 5:28 AM, Nazim Agabekov wrote:

> session:ready()




From kond at nstel.ru  Wed Feb 10 06:25:04 2010
From: kond at nstel.ru (Nikolay Kondratyev)
Date: Wed, 10 Feb 2010 17:25:04 +0300
Subject: [Freeswitch-users] can't see sip messages in the log file
In-Reply-To: <20100210110008.25BFD12292@mail.nstel.ru>
Message-ID: <20100210142456.2DA9A11F4A@mail.nstel.ru>

After some experiments i clarified this question.
SIP messages go into the freeswitch log when:
<param name="sip-trace" value="yes"/> ("sofia profile profile-name siptrace
on" is cli equivalent for this parameter)
AND
Sofia tracelevel is set to info (sofia tracelevel info). 

The default sofia tracelevel is 'console'. That's why I did not see sip
messages in the log after turning on just "siptrace".

Thanks and regards,
Nikolay.

> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-
> users-bounces at lists.freeswitch.org] On Behalf Of Nikolay Kondratyev
> Sent: Wednesday, February 10, 2010 2:00 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] can't see sip messages in the log file
> 
> Jason, thanks for the reply.
> Isn't "sofia profile internal siptrace on" a command line equivalent of
> <param name="sip-trace" value="yes"/>?
> 
> Any way I tried it, but with the same result.
> I still don't see SIP.
> 
> Thanks and regards,
> Nikolay.
> 
> > > Can anybody please advise how to include sip messages into the log
> file?
> >
> > <param name="sip-trace" value="yes"/>
> > in the SIP profile you want to trace, then
> > sofia profile profile-name restart reloadxml
> > or restarting FreeSWITCH should do it.
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From anthony.minessale at gmail.com  Wed Feb 10 06:45:36 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 10 Feb 2010 08:45:36 -0600
Subject: [Freeswitch-users] Bypass-media and REFER method
In-Reply-To: <B816EF85C732094BA2016B2112F3DC2B040C5CA6@isr-brass.interwise.com>
References: <B816EF85C732094BA2016B2112F3DC2B0402C777@isr-brass.interwise.com>
	<B816EF85C732094BA2016B2112F3DC2B040C5CA6@isr-brass.interwise.com>
Message-ID: <191c3a031002100645n5d827949xdefd24143f10759@mail.gmail.com>

update to latest trunk and reproduce your problem with full debug enabled.

sofia profile internal siptrace on
console loglevel debug


On Wed, Feb 10, 2010 at 4:44 AM, Victor Maruani <vmaruani at interwise.com>wrote:

>  Hi,
>
>
>
> I can't have a blind transfer work properly if I use bypass-media=true.
>
>
>
> My first message may have been unclear, here I added excerpt from the
> dialplan:
>
>
>
>                 <!--- ext MyIVR -->
>
>                  <extension name="Extension MyIVR" continue="false">
>
>                    <condition field="destination_number"
> expression="^337$">
>
>                                    <action application="set"
> data="bypass_media=true"/>
>
>                                    <action application="bridge"
> data="sofia/internal/337 at 10.10.5.51"/>
>
>                    </condition>
>
>                  </extension>
>
>
>
>
>
>                 <!--- ext MyServ -->
>
>                  <extension name="Extension MyServ " continue="false">
>
>                    <condition field="destination_number"
> expression="^3341$">
>
>                                                 <action application="set"
> data="bypass_media=true"/>
>
>                                    <action application="bridge"
> data="sofia/internal/3341 at 10.10.5.48"/>
>
>                    </condition>
>
>                  </extension>
>
>
>
> The connection to MyIVR works. Then it sends Refer-to (3341 at ...) which
> fails as I described it in the previous mail.
>
>
>
> I would like to know if the feature has been validated and if I'm missing
> something in the configuration.
>
>
>
> Any help would be very appreciated.
>
>
>
> Thanks!
>
>
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Victor
> Maruani
> *Sent:* Sunday, February 07, 2010 5:01 PM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* [Freeswitch-users] Bypass-media and REFER method
>
>
>
> Hi,
>
>
>
> I'm trying to do a POC using FS, the goal is to have FS handle REFERs
> containing proprietary data.
>
> I want to have some logic on top of FS and also use the fail over
> mechanism.
>
> in short, I have something like this:
>
> (third party) A side  --- FS ---- B side (IVR server)
>
>
>
> the IVR the sends a REFER to FS. I don't want A to deal with it.
>
> now say B refers to C, it would be considered as a "group" C1, C2 ... to
> which I want FS to failover.
>
> only when one has answered should A be updated (REINVITE) and B notified
> and disconnected.
>
> if all fails I would expect B to be notified of the failure and proceed as
> I wish without "losing" A.
>
>
>
> from what I've read FS should be OK for the job but I have a couple issues:
>
>
>
> 1 ) I have some issues getting FS handle a REFER while in bypass-media
> mode.
>
> (I tried with the release and some revisions including latest)
>
> first when I bridge A and B everything is fine and media is bypassed.
>
> When B sends REFER to C:
>
> -           FS immediately NOTIFY B of success and send a reinvite to A
> with SDP containing its own media IP/port.
>
> -          then it does INVITE C with A's SDP.
>
> -          B gets disconnected. A is not updated with C's sdp.
>
> so at this point A sends RTP to FS and C sends RTP to A. ?
>
>
>
> I basically have one extension for B: (set bypass-media and bridge to B)
>
> and another extension to C which does the same actions.
>
> what do you think I do wrong?
>
>
>
>
>
> 2 ) how can I catch the REFER and set variables from it? (like ref-by or
> ref-to)
>
> in the dial plan I do catch the INVITE sent to C, but how to do it with the
> REFER itself?
>
>
>
>
>
> thanks for your help!
>
>
>
>
>
> Best Regards,
>
> Victor.
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100210/a0487c3e/attachment-0002.html 

From rupa at rupa.com  Wed Feb 10 06:50:00 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Wed, 10 Feb 2010 08:50:00 -0600
Subject: [Freeswitch-users] can't see sip messages in the log file
In-Reply-To: <20100210142456.2DA9A11F4A@mail.nstel.ru>
References: <20100210110008.25BFD12292@mail.nstel.ru>
	<20100210142456.2DA9A11F4A@mail.nstel.ru>
Message-ID: <b39273531002100650j2ff8652cu733256c88e5b7aea@mail.gmail.com>

Thanks.  I've added that to the sofia help/completion.

On Wed, Feb 10, 2010 at 8:25 AM, Nikolay Kondratyev <kond at nstel.ru> wrote:

> After some experiments i clarified this question.
> SIP messages go into the freeswitch log when:
> <param name="sip-trace" value="yes"/> ("sofia profile profile-name siptrace
> on" is cli equivalent for this parameter)
> AND
> Sofia tracelevel is set to info (sofia tracelevel info).
>
> The default sofia tracelevel is 'console'. That's why I did not see sip
> messages in the log after turning on just "siptrace".
>
> Thanks and regards,
> Nikolay.
>
> > -----Original Message-----
> > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-
> > users-bounces at lists.freeswitch.org] On Behalf Of Nikolay Kondratyev
> > Sent: Wednesday, February 10, 2010 2:00 PM
> > To: freeswitch-users at lists.freeswitch.org
> > Subject: Re: [Freeswitch-users] can't see sip messages in the log file
> >
> > Jason, thanks for the reply.
> > Isn't "sofia profile internal siptrace on" a command line equivalent of
> > <param name="sip-trace" value="yes"/>?
> >
> > Any way I tried it, but with the same result.
> > I still don't see SIP.
> >
> > Thanks and regards,
> > Nikolay.
> >
> > > > Can anybody please advise how to include sip messages into the log
> > file?
> > >
> > > <param name="sip-trace" value="yes"/>
> > > in the SIP profile you want to trace, then
> > > sofia profile profile-name restart reloadxml
> > > or restarting FreeSWITCH should do it.
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
-Rupa
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100210/553704c1/attachment-0002.html 

From msc at freeswitch.org  Wed Feb 10 07:02:02 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 10 Feb 2010 07:02:02 -0800
Subject: [Freeswitch-users] Play music to A leg.
In-Reply-To: <fe67a3341002092157x7484f3d2id6678510bbec66cc@mail.gmail.com>
References: <fe67a3341002090519r2cc6e8edm7702b8d012fc2be3@mail.gmail.com>
	<87f2f3b91002090737l4efa8en29bde9457fef7ea5@mail.gmail.com>
	<fe67a3341002092157x7484f3d2id6678510bbec66cc@mail.gmail.com>
Message-ID: <87f2f3b91002100702u7581589ua866d2492a008b23@mail.gmail.com>

On Tue, Feb 9, 2010 at 9:57 PM, Nagalenoj H. <nagalenoj at gmail.com> wrote:

> Because, I want to get some digits before bridging the legs. I've tried
> group_confirm_key, but it accepts only one digit, I need multiple digits, so
> I can't use.
> I've also tried group_confirm_file, but when I do originate for multiple
> extensions, I want this script to work based on the answered extension.
>
> So, I've originated and processed the events to do my job.
>
> How do I play some music to A leg?
>
> I might be missing something, but couldn't you just park the call ("A leg")
until you connect to the other party ("B leg") and then uuid_bridge at
whatever point you want?
-MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100210/0936b48b/attachment-0002.html 

From msc at freeswitch.org  Wed Feb 10 07:09:44 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 10 Feb 2010 07:09:44 -0800
Subject: [Freeswitch-users] How to record the call upon successful bridge
In-Reply-To: <13529f9d1002100039q43775563r49b11136ddda0a0@mail.gmail.com>
References: <13529f9d1002091819y2b363f50vc314d1b6197503b2@mail.gmail.com>
	<257401E3-B32F-4699-96F6-415BA2ADDE88@freeswitch.org>
	<13529f9d1002100039q43775563r49b11136ddda0a0@mail.gmail.com>
Message-ID: <87f2f3b91002100709m32d975b5rf16823e2b02bd8c2@mail.gmail.com>

Make sure that you are on the latest SVN trunk and retest. Pastebin the
complete debug log from start to finish of the call.
-MC

On Wed, Feb 10, 2010 at 12:39 AM, Jingwei Yang <jingwei.yang at gmail.com>wrote:

> Hi Ray,
>
> Thanks a lot for the replies. Allow me to elaborate a little bit about my
> situation.
>
> 1. client A calls in and parks at Fifo myq.
> 2. FS connets Agent B to an extension (via originate
> skypiax/ANY/jingwei.yang 33333)
> 3. uuid_bridge client A and agent B
>
> I believe the spice I can add is in the extension 33333. Here's how I
> define it.
>
>     <extension name="agent_hold">
>       <condition field="destination_number" expression="^33333$">
>         <action application="set" data="*execute_on_answer*=record_session
> /tmp/33333.wav"/>
>         <action application="answer"/>
>         <action application="sleep" data="2000"/>
>         <action application="info"/>
>         <action application="playback"
> data="ivr/ivr-hold_connect_call.wav"/>
>       </condition>
>     </extension>
>
> But the wav file didn't get generated at all. Please advise whether the
> above is in correct usage.
>
> I've also tried bridge_pre_execute_bleg_app and
> bridge_pre_execute_bleg_data
>
>         <action application="set"
> data="bridge_pre_execute_bleg_app=record_session"/>
>         <action application="set"
> data="bridge_pre_execute_bleg_data=/tmp/33333.wav"/>
>
> The audio file didn't appear either.
>
> The only successful method is by using record_session directly like this:
>
>         <action application="record_session" data="/tmp/33333.wav"/>
>
> However, in this form, the wav file includes the waiting music, which is
> not ideal.
>
> Thanks and best regards,
> -Jingwei
>
>
> On Wed, Feb 10, 2010 at 12:01 PM, Raymond Chandler <
> intralanman at freeswitch.org> wrote:
>
>> errr... .execute_on_answer might be better ;-)
>> -Ray
>>
>> On Feb 9, 2010, at 9:19 PM, Jingwei Yang wrote:
>>
>> > Hi,
>> >
>> > I'm using uuid_bridge to bridge two calls. May I know how to start
>> recording only when the bridge succeeds?
>> >
>> > Thanks,
>> > -Jingwei
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100210/e8d54d30/attachment-0002.html 

From mike at jerris.com  Wed Feb 10 07:14:22 2010
From: mike at jerris.com (Michael Jerris)
Date: Wed, 10 Feb 2010 10:14:22 -0500
Subject: [Freeswitch-users] No reINVITE when bridging two sessions from
	JavaScript with bypass_media_after_bridge=true
In-Reply-To: <8e9d67561002100035n2d14c74dj83f784713c59d542@mail.gmail.com>
References: <8e9d67561002041147q6c135546u1f7e1887529a916d@mail.gmail.com>
	<3D739D7F-F5C6-4266-B04E-9413DCFDA6AF@freeswitch.org>
	<8e9d67561002050018q24e3a806kc8807b55268e759b@mail.gmail.com>
	<B66289FF-C2A9-4AC4-A56C-66E51B4726F4@freeswitch.org>
	<8e9d67561002051202qeaf1551v8fdd29565f8c02fd@mail.gmail.com>
	<154214DA-5781-4FC1-81EA-B41392E6ACF6@jerris.com>
	<8e9d67561002100035n2d14c74dj83f784713c59d542@mail.gmail.com>
Message-ID: <E789278F-9A2F-462A-823D-8361F327E684@jerris.com>

I would suggest using scripts in between the bridge attempts when you need to do logic, but to do most of this in dialplan where you can.  so through 3, you can handle however you were, then drip A back to dialplan to bridge the call (continue_on_fail, hangup_after_bridge=false), then after bridge, call another script if you need to do more advanced logic.


On Feb 10, 2010, at 3:35 AM, Timur Valishev wrote:

> Dear Mike,
> 
> I'm trying to build a kind of complicated callback. I will be happy if
> you suggest alternative way to do it!
> 
> Scenario is:
> 
> 0. Get command over the socket to initiate connection, get A-number,
> B-number, route preference, Caller ID option (incognito/normal)
> 1. Call billing stored procedure to determine maximum call duration
> 2. Reply over the socket (or better through database?) that connection
> is in progrees (to display it on the user GUI)
> 3. Initiate connection to A-number
> 4. Upon connection to A, say welcome, start calling B, play ringback tone
> 5. Upon dialling B, call billing stored procedure to report that
> session state changed and report the status to user GUI (over the
> socket or through the database?)
> 5.1 If there was error during connection - speak the reason to end
> user (e.g. "Number busy" or "Timeout expired" etc.)
> 6. Join peers in bypass media mode
> 6.1 Wait for various commands over the socket - e.g. transfer the
> call, put on hold, join to conference etc. User will have GUI for that
> operations.
> 7. If B hangs up, call billing stored procedure to finalize session
> and calculate the cost of the call. Cost is to be calculated only by
> B-leg length. Speak to A the cost of the call, say thanks and hang up.
> 7.1 If A hangs up, just call billing and terminate.
> 
> Best regards,
> Timur Valishev
> 
> 2010/2/10 Michael Jerris <mike at jerris.com>:
>> controlling multiple calls in a script like this is tricky, you need to use
>> the first session to create the second one.  Why are you not just doing an
>> originate to do all of this not even in a js file?  What exactly are you
>> trying to accomplish
>> Mike




From mike at jerris.com  Wed Feb 10 07:16:09 2010
From: mike at jerris.com (Michael Jerris)
Date: Wed, 10 Feb 2010 10:16:09 -0500
Subject: [Freeswitch-users] How to kill multiple UUIDs.
In-Reply-To: <7d79b3931002100307h4d370a22l6f04cafa5bf5806d@mail.gmail.com>
References: <7d79b3931002100307h4d370a22l6f04cafa5bf5806d@mail.gmail.com>
Message-ID: <A35CD010-E86F-4B7E-A28B-2A9000B8E009@jerris.com>

you can api hangup hook to call

lua multi_kill.lua uuid1 uuid2 uuid?.

and then write the trivial lua script for that.

Mike

On Feb 10, 2010, at 6:07 AM, lakshmanan ganapathy wrote:

> Hi all,
> 
> My situation is
>     A called to 1005 -- Which executes an ESL program.
>     Now from the program I will made the parallel call using "api originate [origination_uuid=uuid1]user/1006,[origination_uuid2]user/1001 &park()".
>     UUID's are obtained from create_uuid.
> 
>     I'll then wait for the api to return, to check whether the call is answered or rejected by the other end.
>     But while I'm waiting, if A hangup the call, I just want to kill the calls that are originated by my program.
>     So I taught of using api_hang_up_hook and I set that variable to uuid_kill uuid1 uuid2.
>     But it only killed the uuid1.
> 
> Is there any other ways to kill multiple uuid's??
> please help?




From mike at jerris.com  Wed Feb 10 07:28:34 2010
From: mike at jerris.com (Michael Jerris)
Date: Wed, 10 Feb 2010 10:28:34 -0500
Subject: [Freeswitch-users] UPnP Timeout
In-Reply-To: <b39273531002091455meb77162o2baefc3866efaed9@mail.gmail.com>
References: <89dbfdc31002090652h884bc2av82d53c6958fdc409@mail.gmail.com>
	<b39273531002091455meb77162o2baefc3866efaed9@mail.gmail.com>
Message-ID: <4BB6DD8A-995D-4A36-AE92-1C15E5F70876@jerris.com>

Rupa-

	Can you offer up a tcpdump command he could run on the box that would catch all of this for later diagnosis if it happens again?

On Feb 9, 2010, at 5:55 PM, Rupa Schomaker wrote:

> That is a different keep-alive.  I'm specifically talking about the keep-alive packet that we get via upnp multicast.  Whenever we receive one from the gateway we republish the nat mappings to.. um... keep them alive. :)
> 
> On Tue, Feb 9, 2010 at 8:52 AM, Kim Culhan <w8hdkim at gmail.com> wrote:
> On Mon, February 8, 2010 11:21 pm, Troy Anderson wrote:
> > I have been using pfSense (1.2.3) and FS.  FS nicely uses UPnP to poke
> > holes in the firewall, but it seems that the holes close after a while.  I
> > cannot find any documentation in FS nor in pfSense as to what the timeout
> > is.  Is there a setting in FS to do some kind of keep-alive thing with
> > UPnP to keep, e.g. 5060, open?  Or is it already doing that and pfSense is
> > the issue?
> 
> FS has provisions for keep-alive, see the bottom of the page for ping
> time value:
> 
> http://wiki.freeswitch.org/wiki/SIP_Provider_Examples
> 
> To watch the pf firewall hole timing you can install pftop from
> FreeBSD ports/sysutils
> which displays the filter states 'and more'.
> 
> -kim
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> -- 
> -Rupa
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100210/f96cffe0/attachment-0002.html 

From rupa at rupa.com  Wed Feb 10 07:53:44 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Wed, 10 Feb 2010 09:53:44 -0600
Subject: [Freeswitch-users] UPnP Timeout
In-Reply-To: <4BB6DD8A-995D-4A36-AE92-1C15E5F70876@jerris.com>
References: <89dbfdc31002090652h884bc2av82d53c6958fdc409@mail.gmail.com>
	<b39273531002091455meb77162o2baefc3866efaed9@mail.gmail.com>
	<4BB6DD8A-995D-4A36-AE92-1C15E5F70876@jerris.com>
Message-ID: <b39273531002100753n77132cbai8e4a61ada8736179@mail.gmail.com>

sure,

for upnp:

tcpdump -w trace.pcap 'host 239.255.255.250'

for nat-pmp:

tcpdump -w trace.pcap 'host 224.0.0.1'

I updated the wiki in the troubleshooting section.

On Wed, Feb 10, 2010 at 9:28 AM, Michael Jerris <mike at jerris.com> wrote:

> Rupa-
>
> Can you offer up a tcpdump command he could run on the box that would catch
> all of this for later diagnosis if it happens again?
>
> On Feb 9, 2010, at 5:55 PM, Rupa Schomaker wrote:
>
> That is a different keep-alive.  I'm specifically talking about the
> keep-alive packet that we get via upnp multicast.  Whenever we receive one
> from the gateway we republish the nat mappings to.. um... keep them alive.
> :)
>
> On Tue, Feb 9, 2010 at 8:52 AM, Kim Culhan <w8hdkim at gmail.com> wrote:
>
>> On Mon, February 8, 2010 11:21 pm, Troy Anderson wrote:
>> > I have been using pfSense (1.2.3) and FS.  FS nicely uses UPnP to poke
>> > holes in the firewall, but it seems that the holes close after a while.
>>  I
>> > cannot find any documentation in FS nor in pfSense as to what the
>> timeout
>> > is.  Is there a setting in FS to do some kind of keep-alive thing with
>> > UPnP to keep, e.g. 5060, open?  Or is it already doing that and pfSense
>> is
>> > the issue?
>>
>> FS has provisions for keep-alive, see the bottom of the page for ping
>> time value:
>>
>> http://wiki.freeswitch.org/wiki/SIP_Provider_Examples
>>
>> To watch the pf firewall hole timing you can install pftop from
>> FreeBSD ports/sysutils
>> which displays the filter states 'and more'.
>>
>> -kim
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> -Rupa
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100210/80ceb8ac/attachment-0002.html 

From anthony.minessale at gmail.com  Wed Feb 10 07:59:43 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 10 Feb 2010 09:59:43 -0600
Subject: [Freeswitch-users] How to kill multiple UUIDs.
In-Reply-To: <A35CD010-E86F-4B7E-A28B-2A9000B8E009@jerris.com>
References: <7d79b3931002100307h4d370a22l6f04cafa5bf5806d@mail.gmail.com>
	<A35CD010-E86F-4B7E-A28B-2A9000B8E009@jerris.com>
Message-ID: <191c3a031002100759v2fef0f48na5afccf46bb08311@mail.gmail.com>

or you can set a common var like foo=bar on all the chans and do

hupall normal_clearing foo bar


On Wed, Feb 10, 2010 at 9:16 AM, Michael Jerris <mike at jerris.com> wrote:

> you can api hangup hook to call
>
> lua multi_kill.lua uuid1 uuid2 uuid?.
>
> and then write the trivial lua script for that.
>
> Mike
>
> On Feb 10, 2010, at 6:07 AM, lakshmanan ganapathy wrote:
>
> > Hi all,
> >
> > My situation is
> >     A called to 1005 -- Which executes an ESL program.
> >     Now from the program I will made the parallel call using "api
> originate [origination_uuid=uuid1]user/1006,[origination_uuid2]user/1001
> &park()".
> >     UUID's are obtained from create_uuid.
> >
> >     I'll then wait for the api to return, to check whether the call is
> answered or rejected by the other end.
> >     But while I'm waiting, if A hangup the call, I just want to kill the
> calls that are originated by my program.
> >     So I taught of using api_hang_up_hook and I set that variable to
> uuid_kill uuid1 uuid2.
> >     But it only killed the uuid1.
> >
> > Is there any other ways to kill multiple uuid's??
> > please help?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100210/efde6758/attachment-0002.html 

From xanlich at gmail.com  Wed Feb 10 08:00:39 2010
From: xanlich at gmail.com (Chia-Yen Wu)
Date: Thu, 11 Feb 2010 00:00:39 +0800
Subject: [Freeswitch-users] Lua script hangup detect
In-Reply-To: <4B729155.7010708@gmail.com>
References: <314dc3f81002100117h6b1bc49cna989624a1ae4e0e2@mail.gmail.com>
	<4B729155.7010708@gmail.com>
Message-ID: <314dc3f81002100800o41fd6561w394d5e83e69d3d80@mail.gmail.com>

thx for reply, but shouldnt Lua script terminate when client hangs up?

maybe it will stuck in a few situation before it reach to session:status
check point.

like GetDigit or something else? (sorry, I dont have FS right now, need test
it tomorrow)

2010/2/10 Nazim Agabekov <nazim.agabekov at gmail.com>

>  Hello,
> Can you pastebin your script?
>
> http://pastebin.freeswitch.org
>
>
> On 02/10/2010 01:17 PM, Chia-Yen Wu wrote:
>
> Hello,
>  i tried to use Lua script to replace xml macro in dialplan,
>  but I found out that Lua wont terminate if client hangup,
> ,so the session is still on but client is already hangup,
> is there a way to avoid this ?
>
>
> _______________________________________________
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/0e329d98/attachment-0002.html 

From jerry.richards at teotech.com  Wed Feb 10 08:13:26 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Wed, 10 Feb 2010 08:13:26 -0800
Subject: [Freeswitch-users] Any Known Dual-Registration Issue?
In-Reply-To: <035001caa9c8$2dca81c0$895f8540$@com>
References: <5AD33F8B1F0848CF9C3A314A60039366@greyhawk.tonecommander.com>
	<035001caa9c8$2dca81c0$895f8540$@com>
Message-ID: <3017421995484996A379AADB394D48CB@greyhawk.tonecommander.com>

I am using Bria Pro softphones, but it is possible that the PC was
disconnected before a restart.  Any phone does not necessarily send
unregister, for example if there is a power outage or it is disconnected
from the network.  Anyway, I'll just try setting multiple-registrations to
contact as you suggest.

Thanks,
Jerry
 

-----Original Message-----
From: Peder [mailto:peder at networkoblivion.com] 
Sent: Tuesday, February 09, 2010 12:41 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Any Known Dual-Registration Issue?

What kind of phones?  If you have multiple registartion, this can happen
sometimes if you reboot a phone.  Crappy phones, like Grandstream, don't
un-register when you reboot and then when they come back up, they register
again and thus two registrations until the lifetime of the registration ends
and it gets flushed.  Changing the multiple-registration to contact can help
as I believe that uses port and source IP as part of the registration info:

<param name="multiple-registrations" value="contact"/>



-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jerry
Richards
Sent: Tuesday, February 09, 2010 2:27 PM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Any Known Dual-Registration Issue?

I've noticed that sometimes my phones end up with two registrations with two
Call-IDs at Freeswitch.  Is there any known bug that would cause this?  I've
seen it on different phone models, so I'm thinking there is some timing
issue with Freeswitch.

Best Regards,
Jerry


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org







From nazim.agabekov at gmail.com  Wed Feb 10 08:34:01 2010
From: nazim.agabekov at gmail.com (Nazim Agabekov)
Date: Wed, 10 Feb 2010 20:34:01 +0400
Subject: [Freeswitch-users] Lua script hangup detect
In-Reply-To: <314dc3f81002100800o41fd6561w394d5e83e69d3d80@mail.gmail.com>
References: <314dc3f81002100117h6b1bc49cna989624a1ae4e0e2@mail.gmail.com>	<4B729155.7010708@gmail.com>
	<314dc3f81002100800o41fd6561w394d5e83e69d3d80@mail.gmail.com>
Message-ID: <4B72DFF9.4040402@gmail.com>

Lua script is not terminating immediately on hangup.
This behavior allows user to finalize the script nicely (free
dynamically allocated resources, update logs, e.t.c)
> maybe it will stuck in a few situation before it reach to
> session:status check point.
>
> like GetDigit or something else? (sorry, I dont have FS right now,
> need test it tomorrow)
I've never encountered such a problem. Usually GetDigit-like functions
have timeout parameter, so they don't block forever.
Just check the session status often and it will work like a charm ;


On 02/10/2010 08:00 PM, Chia-Yen Wu wrote:
> thx for reply, but shouldnt Lua script terminate when client hangs up?
>
> maybe it will stuck in a few situation before it reach to
> session:status check point.
>
> like GetDigit or something else? (sorry, I dont have FS right now,
> need test it tomorrow)
>
> 2010/2/10 Nazim Agabekov <nazim.agabekov at gmail.com
> <mailto:nazim.agabekov at gmail.com>>
>
>     Hello,
>     Can you pastebin your script?
>
>     http://pastebin.freeswitch.org
>
>
>     On 02/10/2010 01:17 PM, Chia-Yen Wu wrote:
>>     Hello, 
>>     i tried to use Lua script to replace xml macro in dialplan,
>>     but I found out that Lua wont terminate if client hangup,
>>     ,so the session is still on but client is already hangup,
>>     is there a way to avoid this ?
>>
>>
>>     _______________________________________________
>>     FreeSWITCH-users mailing list
>>     FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     http://www.freeswitch.org
>>       
>
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100210/d0c8e3eb/attachment-0002.html 

From vmaruani at interwise.com  Wed Feb 10 09:01:47 2010
From: vmaruani at interwise.com (Victor Maruani)
Date: Wed, 10 Feb 2010 19:01:47 +0200
Subject: [Freeswitch-users] Bypass-media and REFER method
In-Reply-To: <191c3a031002100645n5d827949xdefd24143f10759@mail.gmail.com>
References: <B816EF85C732094BA2016B2112F3DC2B0402C777@isr-brass.interwise.com><B816EF85C732094BA2016B2112F3DC2B040C5CA6@isr-brass.interwise.com>
	<191c3a031002100645n5d827949xdefd24143f10759@mail.gmail.com>
Message-ID: <B816EF85C732094BA2016B2112F3DC2B040C6061@isr-brass.interwise.com>

Hi,

 

Logs are on pb 12099

I hope this helps.

Reproduced with revision 16599.

 

A-side (10.10.5.19) is an x-lite registered with extension 1002

B (.5.51) refers to C (.5.48) none are registered.

 

Please refer to previous emails for details of dialplan and what I try
to do...

Let me know if you need more info 

 

Thanks!

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Anthony Minessale
Sent: Wednesday, February 10, 2010 4:46 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Bypass-media and REFER method

 

update to latest trunk and reproduce your problem with full debug
enabled.

sofia profile internal siptrace on
console loglevel debug



On Wed, Feb 10, 2010 at 4:44 AM, Victor Maruani <vmaruani at interwise.com>
wrote:

Hi,

 

I can't have a blind transfer work properly if I use bypass-media=true.

 

My first message may have been unclear, here I added excerpt from the
dialplan:

 

                <!--- ext MyIVR -->

                 <extension name="Extension MyIVR" continue="false">

                   <condition field="destination_number"
expression="^337$">

                                   <action application="set"
data="bypass_media=true"/>

                                   <action application="bridge"
data="sofia/internal/337 at 10.10.5.51"/>

                   </condition>

                 </extension>

 

 

                <!--- ext MyServ -->

                 <extension name="Extension MyServ " continue="false">

                   <condition field="destination_number"
expression="^3341$">

                                                <action
application="set" data="bypass_media=true"/>

                                   <action application="bridge"
data="sofia/internal/3341 at 10.10.5.48"/>

                   </condition>

                 </extension>   

 

The connection to MyIVR works. Then it sends Refer-to (3341 at ...) which
fails as I described it in the previous mail.

 

I would like to know if the feature has been validated and if I'm
missing something in the configuration.

 

Any help would be very appreciated.

 

Thanks!

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Victor Maruani
Sent: Sunday, February 07, 2010 5:01 PM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Bypass-media and REFER method

 

Hi,

 

I'm trying to do a POC using FS, the goal is to have FS handle REFERs
containing proprietary data. 

I want to have some logic on top of FS and also use the fail over
mechanism.

in short, I have something like this:

(third party) A side  --- FS ---- B side (IVR server)

 

the IVR the sends a REFER to FS. I don't want A to deal with it. 

now say B refers to C, it would be considered as a "group" C1, C2 ... to
which I want FS to failover.

only when one has answered should A be updated (REINVITE) and B notified
and disconnected.

if all fails I would expect B to be notified of the failure and proceed
as I wish without "losing" A.

 

from what I've read FS should be OK for the job but I have a couple
issues:

 

1 ) I have some issues getting FS handle a REFER while in bypass-media
mode.

(I tried with the release and some revisions including latest)

first when I bridge A and B everything is fine and media is bypassed.

When B sends REFER to C:

-           FS immediately NOTIFY B of success and send a reinvite to A
with SDP containing its own media IP/port.

-          then it does INVITE C with A's SDP.

-          B gets disconnected. A is not updated with C's sdp.

so at this point A sends RTP to FS and C sends RTP to A. ...

 

I basically have one extension for B: (set bypass-media and bridge to B)

and another extension to C which does the same actions.

what do you think I do wrong? 

 

 

2 ) how can I catch the REFER and set variables from it? (like ref-by or
ref-to)

in the dial plan I do catch the INVITE sent to C, but how to do it with
the REFER itself?

 

 

thanks for your help!

 

 

Best Regards,

Victor.

 


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
<mailto:MSN%3Aanthony_minessale at hotmail.com> 
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
<mailto:PAYPAL%3Aanthony.minessale at gmail.com> 
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:+19193869900

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100210/6ccad925/attachment-0002.html 

From red.rain.seven at gmail.com  Wed Feb 10 09:16:51 2010
From: red.rain.seven at gmail.com (Henry Huang)
Date: Thu, 11 Feb 2010 01:16:51 +0800
Subject: [Freeswitch-users] How to kill multiple UUIDs.
In-Reply-To: <191c3a031002100759v2fef0f48na5afccf46bb08311@mail.gmail.com>
References: <7d79b3931002100307h4d370a22l6f04cafa5bf5806d@mail.gmail.com>
	<A35CD010-E86F-4B7E-A28B-2A9000B8E009@jerris.com>
	<191c3a031002100759v2fef0f48na5afccf46bb08311@mail.gmail.com>
Message-ID: <59ad9ca11002100916l6982c88r22e1b45af55b80a1@mail.gmail.com>

is there an example of setting common var for multiple channels?
or just do it normally like set var=foo before all the bridged call I would
like to hang up at the same time?
and the var name can be anything I want?


Henry

On Wed, Feb 10, 2010 at 11:59 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> or you can set a common var like foo=bar on all the chans and do
>
> hupall normal_clearing foo bar
>
>
>
> On Wed, Feb 10, 2010 at 9:16 AM, Michael Jerris <mike at jerris.com> wrote:
>
>> you can api hangup hook to call
>>
>> lua multi_kill.lua uuid1 uuid2 uuid?.
>>
>> and then write the trivial lua script for that.
>>
>> Mike
>>
>> On Feb 10, 2010, at 6:07 AM, lakshmanan ganapathy wrote:
>>
>> > Hi all,
>> >
>> > My situation is
>> >     A called to 1005 -- Which executes an ESL program.
>> >     Now from the program I will made the parallel call using "api
>> originate [origination_uuid=uuid1]user/1006,[origination_uuid2]user/1001
>> &park()".
>> >     UUID's are obtained from create_uuid.
>> >
>> >     I'll then wait for the api to return, to check whether the call is
>> answered or rejected by the other end.
>> >     But while I'm waiting, if A hangup the call, I just want to kill the
>> calls that are originated by my program.
>> >     So I taught of using api_hang_up_hook and I set that variable to
>> uuid_kill uuid1 uuid2.
>> >     But it only killed the uuid1.
>> >
>> > Is there any other ways to kill multiple uuid's??
>> > please help?
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/620ba28a/attachment-0002.html 

From nicolas at medularis.com  Wed Feb 10 09:32:08 2010
From: nicolas at medularis.com (Nicolas Brenner)
Date: Wed, 10 Feb 2010 14:32:08 -0300
Subject: [Freeswitch-users] Lua script hangup detect
In-Reply-To: <4B72DFF9.4040402@gmail.com>
References: <314dc3f81002100117h6b1bc49cna989624a1ae4e0e2@mail.gmail.com>
	<4B729155.7010708@gmail.com>
	<314dc3f81002100800o41fd6561w394d5e83e69d3d80@mail.gmail.com>
	<4B72DFF9.4040402@gmail.com>
Message-ID: <1b46b4e81002100932x7af486ekd4b05f4750cace20@mail.gmail.com>

On Wed, Feb 10, 2010 at 1:34 PM, Nazim Agabekov <nazim.agabekov at gmail.com>wrote:

>  Lua script is not terminating immediately on hangup.
>


When does it terminate then? Will the script terminate when it finishes
running or does it need some special instruction?



> This behavior allows user to finalize the script nicely (free dynamically
> allocated resources, update logs, e.t.c)
>
>  maybe it will stuck in a few situation before it reach to session:status
> check point.
>
>  like GetDigit or something else? (sorry, I dont have FS right now, need
> test it tomorrow)
>
> I've never encountered such a problem. Usually GetDigit-like functions have
> timeout parameter, so they don't block forever.
> Just check the session status often and it will work like a charm ;
>
>
>
> On 02/10/2010 08:00 PM, Chia-Yen Wu wrote:
>
> thx for reply, but shouldnt Lua script terminate when client hangs up?
>
>  maybe it will stuck in a few situation before it reach to session:status
> check point.
>
>  like GetDigit or something else? (sorry, I dont have FS right now, need
> test it tomorrow)
>
> 2010/2/10 Nazim Agabekov <nazim.agabekov at gmail.com>
>
>> Hello,
>> Can you pastebin your script?
>>
>> http://pastebin.freeswitch.org
>>
>>
>> On 02/10/2010 01:17 PM, Chia-Yen Wu wrote:
>>
>>  Hello,
>>  i tried to use Lua script to replace xml macro in dialplan,
>>  but I found out that Lua wont terminate if client hangup,
>> ,so the session is still on but client is already hangup,
>> is there a way to avoid this ?
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100210/40e6ff7e/attachment-0002.html 

From nazim.agabekov at gmail.com  Wed Feb 10 09:53:44 2010
From: nazim.agabekov at gmail.com (Nazim Agabekov)
Date: Wed, 10 Feb 2010 21:53:44 +0400
Subject: [Freeswitch-users] Lua script hangup detect
In-Reply-To: <1b46b4e81002100932x7af486ekd4b05f4750cace20@mail.gmail.com>
References: <314dc3f81002100117h6b1bc49cna989624a1ae4e0e2@mail.gmail.com>	<4B729155.7010708@gmail.com>	<314dc3f81002100800o41fd6561w394d5e83e69d3d80@mail.gmail.com>	<4B72DFF9.4040402@gmail.com>
	<1b46b4e81002100932x7af486ekd4b05f4750cace20@mail.gmail.com>
Message-ID: <4B72F2A8.4070503@gmail.com>

I think it continues to run until it finishes. I'll check it tomorrow on
my test system.

On 02/10/2010 09:32 PM, Nicolas Brenner wrote:
>
> On Wed, Feb 10, 2010 at 1:34 PM, Nazim Agabekov
> <nazim.agabekov at gmail.com <mailto:nazim.agabekov at gmail.com>> wrote:
>
>     Lua script is not terminating immediately on hangup.
>
>
>
> When does it terminate then? Will the script terminate when it
> finishes running or does it need some special instruction?
>
>  
>
>     This behavior allows user to finalize the script nicely (free
>     dynamically allocated resources, update logs, e.t.c)
>
>>     maybe it will stuck in a few situation before it reach to
>>     session:status check point.
>>
>>     like GetDigit or something else? (sorry, I dont have FS right
>>     now, need test it tomorrow)
>     I've never encountered such a problem. Usually GetDigit-like
>     functions have timeout parameter, so they don't block forever.
>     Just check the session status often and it will work like a charm ;
>
>
>
>     On 02/10/2010 08:00 PM, Chia-Yen Wu wrote:
>>     thx for reply, but shouldnt Lua script terminate when client
>>     hangs up?
>>
>>     maybe it will stuck in a few situation before it reach to
>>     session:status check point.
>>
>>     like GetDigit or something else? (sorry, I dont have FS right
>>     now, need test it tomorrow)
>>
>>     2010/2/10 Nazim Agabekov <nazim.agabekov at gmail.com
>>     <mailto:nazim.agabekov at gmail.com>>
>>
>>         Hello,
>>         Can you pastebin your script?
>>
>>         http://pastebin.freeswitch.org
>>
>>
>>         On 02/10/2010 01:17 PM, Chia-Yen Wu wrote:
>>>         Hello, 
>>>         i tried to use Lua script to replace xml macro in dialplan,
>>>         but I found out that Lua wont terminate if client hangup,
>>>         ,so the session is still on but client is already hangup,
>>>         is there a way to avoid this ?
>>>
>>>
>>>         _______________________________________________
>>>         FreeSWITCH-users mailing list
>>>         FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>         http://www.freeswitch.org
>>>           
>>
>>
>>         _______________________________________________
>>         FreeSWITCH-users mailing list
>>         FreeSWITCH-users at lists.freeswitch.org
>>         <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>         http://www.freeswitch.org
>>
>>
>>
>>     _______________________________________________
>>     FreeSWITCH-users mailing list
>>     FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     http://www.freeswitch.org
>>       
>
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100210/03aa76a0/attachment-0002.html 

From Prometheus001 at gmx.net  Wed Feb 10 10:44:23 2010
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Wed, 10 Feb 2010 19:44:23 +0100
Subject: [Freeswitch-users] Skypiax latency
Message-ID: <4B72FE87.4000401@gmx.net>

Hello,

I have a problem with latency and mod_skypiax

Skype=>SIP is always fine (~0.3sec)
SIP => Skype is always bad (~2-4 sec)
I would expect that latency in both directions should be the same.

Anybody has discovered this before and has a solution?

The scenario is as follows:

SIP-UA => FreeswitchServer=>Freeswitch/SkypIaxServer=>Skype
Both freeswitch servers are in the same LAN, so latency should be low.

Best regards
Peter



From robert.hadley at teotech.com  Wed Feb 10 10:50:15 2010
From: robert.hadley at teotech.com (Robert Hadley)
Date: Wed, 10 Feb 2010 10:50:15 -0800
Subject: [Freeswitch-users] demo_ivr cannot find sound files via relative
	paths
Message-ID: <78030B9E01DD48F8B024A97035F96282@greyhawk.tonecommander.com>

Hi,

 

It appears a recent change (possibly the new sounds_dir variable or the new
ivr_menu folder?) may have broken relative sound file paths in the IVR.  I
built a today's trunk version and installed to the default location. Using
the default conf files the demo_ivr cannot find files based on the relative
paths specified in ivr_menus/demo_ivr.xml.

 

[root at TEO-UCM-T2 conf]# cat ivr_menus/demo_ivr.xml

<include>

  <!-- demo IVR setup -->

  <!-- demo IVR, Main Menu -->

  <menu name="demo_ivr"

      greet-long="phrase:demo_ivr_main_menu"

      greet-short="phrase:demo_ivr_main_menu_short"

      invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"

      exit-sound="voicemail/vm-goodbye.wav"

 

 

2010-02-10 10:32:15.300337 [DEBUG] switch_ivr_menu.c:414 Executing IVR menu
demo_ivr

2010-02-10 10:32:15.300337 [DEBUG] switch_ivr_play_say.c:118 No language
specified - Using [en]

2010-02-10 10:32:15.300337 [DEBUG] switch_ivr_play_say.c:273 Handle
play-file:[ivr/ivr-welcome_to_freeswitch.wav] (en:en)

2010-02-10 10:32:15.300337 [DEBUG] switch_ivr_play_say.c:1158 Codec
Activated L16 at 8000hz 1 channels 20ms

2010-02-10 10:32:17.922147 [DEBUG] switch_ivr_play_say.c:1450 done playing
file

2010-02-10 10:32:18.042149 [DEBUG] switch_ivr_play_say.c:273 Handle
play-file:[ivr/ivr-this_ivr_will_let_you_test_features.wav] (en:en)

2010-02-10 10:32:18.042149 [DEBUG] switch_ivr_play_say.c:1158 Codec
Activated L16 at 8000hz 1 channels 20ms

2010-02-10 10:32:19.962158 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF 2:800

2010-02-10 10:32:19.962158 [DEBUG] switch_ivr_play_say.c:1450 done playing
file

2010-02-10 10:32:20.082156 [DEBUG] switch_ivr_menu.c:329 waiting for 3/4
digits t/o 2000

2010-02-10 10:32:20.120617 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF 2:400

2010-02-10 10:32:20.442158 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF 2:800

2010-02-10 10:32:20.682162 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF 2:800

2010-02-10 10:32:20.682162 [DEBUG] switch_ivr_menu.c:376 digits '2222'

2010-02-10 10:32:20.682162 [DEBUG] switch_ivr_menu.c:470 action regex [2222]
[/^(10[01][0-9])$/] [0]

2010-02-10 10:32:20.682162 [DEBUG] switch_ivr_menu.c:560 IVR menu 'demo_ivr'
caught invalid input '2222'

2010-02-10 10:32:20.682162 [ERR] mod_sndfile.c:194 Error Opening File
[/usr/local/freeswitch/sounds/ivr/ivr-that_was_an_invalid_entry.wav] [System
error : No such file or directory.]

2010-02-10 10:32:21.701505 [DEBUG] switch_ivr_play_say.c:118 No language
specified - Using [en]

2010-02-10 10:32:21.701505 [DEBUG] switch_ivr_play_say.c:273 Handle
play-file:[silence_stream://1000] (en:en)

2010-02-10 10:32:21.701505 [DEBUG] switch_ivr_play_say.c:1158 Codec
Activated L16 at 8000hz 1 channels 20ms

2010-02-10 10:32:22.700352 [DEBUG] switch_ivr_play_say.c:1450 done playing
file

 

Regards,

Robert

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100210/4e17a0df/attachment-0002.html 

From woof at iwoof.org  Wed Feb 10 11:11:32 2010
From: woof at iwoof.org (Andy Spitzer)
Date: Wed, 10 Feb 2010 14:11:32 -0500
Subject: [Freeswitch-users] Question regarding testing IVRs
In-Reply-To: <247f8101002092023t3f30a600o1a9f43771c879e61@mail.gmail.com>
References: <247f8101002092023t3f30a600o1a9f43771c879e61@mail.gmail.com>
Message-ID: <op.u7xd9iyog0dksq@knights-macbook.fuzzy>

Woof!

On Tue, 09 Feb 2010 23:23:35 -0500, Pablo Hernan Saro  
<pablosaro at gmail.com> wrote:

> I was wondering if anyone knows about an automated tool for testing IVR  
> systems

There are the expensive systems out there you can spend $$$ on (e.g.  
Empirix Hammer), or you can just code up some scripts that drive a second  
instance of FreeSWITCH to place calls, dial digits, etc.  I prefer the  
latter, especially for simple load testing.  If you instrument your IVR so  
that these scripts can externally determine what state each call is in  
(often just by polling the log files), then this can be an excellent tool  
for more complicated testing scenarios as well.

But beware...writing a decent IVR test script (for any platform, be it  
home brew or commercial), can end up being about an order of magnitude  
more complicated than the building the original system under test.  This  
is because while an IVR system can just wait for something to happen or  
timeout, the test system should track response times, statistics, vary  
it's input but still be within certain ranges, query the same database the  
IVR is using to know about account numbers, state, etc.

For many cases, just automating simple calls to generate load, and then  
having a person call in and exercise the complicated bits of the IVR while  
under load will get you pretty far.

--Woof!



From tculjaga at gmail.com  Wed Feb 10 11:13:51 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Wed, 10 Feb 2010 20:13:51 +0100
Subject: [Freeswitch-users] h323 - sip call is not working
In-Reply-To: <20100210104911.A632A11F49@mail.nstel.ru>
References: <20100210104911.A632A11F49@mail.nstel.ru>
Message-ID: <65d96fc81002101113p44fe6f43oba72c3a2e13785cc@mail.gmail.com>

On Wed, Feb 10, 2010 at 11:49 AM, Nikolay Kondratyev <kond at nstel.ru> wrote:

>  Hi all,
>
> I compiled FreeSWITCH Version 1.0.5-20100209-0400 (16587M) with mod_h323.
>
> When I call from h323 (Avaya IPOffice) to local fs extention (x-lite) I
> hear ring back, but when x-lite picks up, he hears silence, while IPOffice
> user continues to hear ringback.
>
> The log is at the http://pastebin.freeswitch.org/12091
>
> My configuration is almost default, several local extentions added, and
> h323.conf from http://wiki.freeswitch.org/wiki/Mod_h323
>
> 5840 ? user at IPOffice
>
> 2853 ? x-lite registered at FS
>
> IPOffice ip address: 172.23.14.2
>
> FS ip address 172.23.22.49
>
>
>
> Can anybody please advise how to solve that?
>
> Is it a configuration or a software problem?
>
>
>

Please can you send me the tcpdump as well (not filtered). IPOffice i known
to have a "broken" H323 stack.
did you try to play with tunneling and h245 in setup settings as well ?

It looks like there is some h245 negotiation still pending but cant see that
from the logs.


> Thanks in advance,
>
> Nikolay.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100210/eec71392/attachment-0002.html 

From anthony.minessale at gmail.com  Wed Feb 10 11:29:26 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 10 Feb 2010 13:29:26 -0600
Subject: [Freeswitch-users] How to kill multiple UUIDs.
In-Reply-To: <59ad9ca11002100916l6982c88r22e1b45af55b80a1@mail.gmail.com>
References: <7d79b3931002100307h4d370a22l6f04cafa5bf5806d@mail.gmail.com>
	<A35CD010-E86F-4B7E-A28B-2A9000B8E009@jerris.com>
	<191c3a031002100759v2fef0f48na5afccf46bb08311@mail.gmail.com>
	<59ad9ca11002100916l6982c88r22e1b45af55b80a1@mail.gmail.com>
Message-ID: <191c3a031002101129h1103c2e7l6100b2d41807fddf@mail.gmail.com>

if you put a var in the leading {} it will get set on all channels

{foo=bar}sofia/internal/test at server.com


On Wed, Feb 10, 2010 at 11:16 AM, Henry Huang <red.rain.seven at gmail.com>wrote:

> is there an example of setting common var for multiple channels?
> or just do it normally like set var=foo before all the bridged call I would
> like to hang up at the same time?
> and the var name can be anything I want?
>
>
> Henry
>
>
> On Wed, Feb 10, 2010 at 11:59 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> or you can set a common var like foo=bar on all the chans and do
>>
>> hupall normal_clearing foo bar
>>
>>
>>
>> On Wed, Feb 10, 2010 at 9:16 AM, Michael Jerris <mike at jerris.com> wrote:
>>
>>> you can api hangup hook to call
>>>
>>> lua multi_kill.lua uuid1 uuid2 uuid?.
>>>
>>> and then write the trivial lua script for that.
>>>
>>> Mike
>>>
>>> On Feb 10, 2010, at 6:07 AM, lakshmanan ganapathy wrote:
>>>
>>> > Hi all,
>>> >
>>> > My situation is
>>> >     A called to 1005 -- Which executes an ESL program.
>>> >     Now from the program I will made the parallel call using "api
>>> originate [origination_uuid=uuid1]user/1006,[origination_uuid2]user/1001
>>> &park()".
>>> >     UUID's are obtained from create_uuid.
>>> >
>>> >     I'll then wait for the api to return, to check whether the call is
>>> answered or rejected by the other end.
>>> >     But while I'm waiting, if A hangup the call, I just want to kill
>>> the calls that are originated by my program.
>>> >     So I taught of using api_hang_up_hook and I set that variable to
>>> uuid_kill uuid1 uuid2.
>>> >     But it only killed the uuid1.
>>> >
>>> > Is there any other ways to kill multiple uuid's??
>>> > please help?
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100210/97461df1/attachment-0002.html 

From peter.olsson at visionutveckling.se  Wed Feb 10 11:42:55 2010
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Wed, 10 Feb 2010 20:42:55 +0100
Subject: [Freeswitch-users] h323 - sip call is not working
In-Reply-To: <65d96fc81002101113p44fe6f43oba72c3a2e13785cc@mail.gmail.com>
References: <20100210104911.A632A11F49@mail.nstel.ru>,
	<65d96fc81002101113p44fe6f43oba72c3a2e13785cc@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C5576CE4923@cooper>

I've been running both h323 and SIP between FS and Avaya IPO for some time. No problems at all. :)

But make sure to enable h323 fast start and disable "direct media path" in the IPO, if I remember correctly these where the only two parameters that made any real difference for me.

But I do recommenf to use SIP, since it's much better supported by FS.

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Tihomir Culjaga [tculjaga at gmail.com]
Skickat: den 10 februari 2010 20:13
Till: freeswitch-users at lists.freeswitch.org
?mne: Re: [Freeswitch-users] h323 - sip call is not working

On Wed, Feb 10, 2010 at 11:49 AM, Nikolay Kondratyev <kond at nstel.ru<mailto:kond at nstel.ru>> wrote:
Hi all,
I compiled FreeSWITCH Version 1.0.5-20100209-0400 (16587M) with mod_h323.
When I call from h323 (Avaya IPOffice) to local fs extention (x-lite) I hear ring back, but when x-lite picks up, he hears silence, while IPOffice user continues to hear ringback.
The log is at the http://pastebin.freeswitch.org/12091
My configuration is almost default, several local extentions added, and h323.conf from http://wiki.freeswitch.org/wiki/Mod_h323
5840 ? user at IPOffice
2853 ? x-lite registered at FS
IPOffice ip address: 172.23.14.2
FS ip address 172.23.22.49

Can anybody please advise how to solve that?
Is it a configuration or a software problem?


Please can you send me the tcpdump as well (not filtered). IPOffice i known to have a "broken" H323 stack.
did you try to play with tunneling and h245 in setup settings as well ?

It looks like there is some h245 negotiation still pending but cant see that from the logs.

Thanks in advance,
Nikolay.

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


!DSPAM:4b73075e32931909716500!



From tculjaga at gmail.com  Wed Feb 10 12:27:28 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Wed, 10 Feb 2010 21:27:28 +0100
Subject: [Freeswitch-users] h323 - sip call is not working
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C5576CE4923@cooper>
References: <20100210104911.A632A11F49@mail.nstel.ru>
	<65d96fc81002101113p44fe6f43oba72c3a2e13785cc@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C5576CE4923@cooper>
Message-ID: <65d96fc81002101227n3febe3bdgb0c36d767fa5be8e@mail.gmail.com>

On Wed, Feb 10, 2010 at 8:42 PM, Peter Olsson <
peter.olsson at visionutveckling.se> wrote:

> I've been running both h323 and SIP between FS and Avaya IPO for some time.
> No problems at all. :)
>
> But make sure to enable h323 fast start and disable "direct media path" in
> the IPO, if I remember correctly these where the only two parameters that
> made any real difference for me.
>
> But I do recommenf to use SIP, since it's much better supported by FS.
>
> /Peter
>
>
Peter, what H323plus version are you using ?

did you noticed q931 release cause is not mapped H323 => SIP correctly ?
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100210/191ba117/attachment-0002.html 

From matt at webcontracts.co.uk  Wed Feb 10 15:03:48 2010
From: matt at webcontracts.co.uk (Matthew Law)
Date: Wed, 10 Feb 2010 23:03:48 -0000
Subject: [Freeswitch-users] ACL question and js error
In-Reply-To: <20100210024638.GN31942@base.carmickle.com>
References: <ea05d45dca9be2d979fb386f960bc077.squirrel@www.webcontracts.co.uk>
	<20100207145907.GF31942@base.carmickle.com>
	<cfe871e3305675257b8c74ed4fb5a11b.squirrel@www.webcontracts.co.uk>
	<20100210024638.GN31942@base.carmickle.com>
Message-ID: <2112b95ba7e53b541a9f5aad82b77f96.squirrel@www.webcontracts.co.uk>


On Wed, February 10, 2010 2:46 am, Frank Carmickle wrote:
> After doing a
> sofia profile $profile rescan reloadxml
>
> it still doesn't work?  Are you sure it isn't hitting the dialplan and
> failing?  I have never used
>
> <param name="extension" value="voiptalk-incoming"/>
>
> I usually leave it commented and then match on the  destination in the
> dialplan.
>
> <condition field="destination_number"
> expression="^(4124134655|4128484655|19734226137)$">
>
> HTH
> --FC

It was working (there was another issue but that is now fixed too).  I was
not reloading the sofia config properly.  You live and learn.

Thanks,

Matt.




From ederwander at gmail.com  Wed Feb 10 15:43:04 2010
From: ederwander at gmail.com (Eder Souza)
Date: Wed, 10 Feb 2010 21:43:04 -0200
Subject: [Freeswitch-users] Pause during dialing ?
Message-ID: <622bedea1002101543h1c58dd05nb1f81c8d751a600f@mail.gmail.com>

Hi list!!

How i can make calls with Pause in string outgoing ??


here examples in asterisk

exten => _1201.,1,Dial(SIP/eder at eder,60,TtrD(ww891w${EXTEN:4}))

OR

exten => _8X.,1,Dial(Zap/g1/ww0w${EXTEN:1})


In Asterisk exist the flag "w", but im FS how make this ??


thx

Eng Eder de Souza
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100210/973caa7b/attachment-0002.html 

From mayamatakeshi at gmail.com  Wed Feb 10 15:45:09 2010
From: mayamatakeshi at gmail.com (mayamatakeshi)
Date: Thu, 11 Feb 2010 08:45:09 +0900
Subject: [Freeswitch-users] How to kill multiple UUIDs.
In-Reply-To: <191c3a031002101129h1103c2e7l6100b2d41807fddf@mail.gmail.com>
References: <7d79b3931002100307h4d370a22l6f04cafa5bf5806d@mail.gmail.com>
	<A35CD010-E86F-4B7E-A28B-2A9000B8E009@jerris.com>
	<191c3a031002100759v2fef0f48na5afccf46bb08311@mail.gmail.com>
	<59ad9ca11002100916l6982c88r22e1b45af55b80a1@mail.gmail.com>
	<191c3a031002101129h1103c2e7l6100b2d41807fddf@mail.gmail.com>
Message-ID: <15b9404e1002101545g421280adl93fb7ec036d05e9b@mail.gmail.com>

On Thu, Feb 11, 2010 at 4:29 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> if you put a var in the leading {} it will get set on all channels
>
> {foo=bar}sofia/internal/test at server.com


I can see that when I use command bridge, CHANNEL_ORIGINATE shows up with
variable Other-Leg-Unique-ID set to the UID of the channel executing bridge.
To be consistent, could I use this same name when calling originate to
create LegB for a parked LegA? Would it clash with anything?


>
>
> On Wed, Feb 10, 2010 at 11:16 AM, Henry Huang <red.rain.seven at gmail.com>wrote:
>
>> is there an example of setting common var for multiple channels?
>> or just do it normally like set var=foo before all the bridged call I
>> would like to hang up at the same time?
>> and the var name can be anything I want?
>>
>>
>> Henry
>>
>>
>> On Wed, Feb 10, 2010 at 11:59 PM, Anthony Minessale <
>> anthony.minessale at gmail.com> wrote:
>>
>>> or you can set a common var like foo=bar on all the chans and do
>>>
>>> hupall normal_clearing foo bar
>>>
>>>
>>>
>>> On Wed, Feb 10, 2010 at 9:16 AM, Michael Jerris <mike at jerris.com> wrote:
>>>
>>>> you can api hangup hook to call
>>>>
>>>> lua multi_kill.lua uuid1 uuid2 uuid?.
>>>>
>>>> and then write the trivial lua script for that.
>>>>
>>>> Mike
>>>>
>>>> On Feb 10, 2010, at 6:07 AM, lakshmanan ganapathy wrote:
>>>>
>>>> > Hi all,
>>>> >
>>>> > My situation is
>>>> >     A called to 1005 -- Which executes an ESL program.
>>>> >     Now from the program I will made the parallel call using "api
>>>> originate [origination_uuid=uuid1]user/1006,[origination_uuid2]user/1001
>>>> &park()".
>>>> >     UUID's are obtained from create_uuid.
>>>> >
>>>> >     I'll then wait for the api to return, to check whether the call is
>>>> answered or rejected by the other end.
>>>> >     But while I'm waiting, if A hangup the call, I just want to kill
>>>> the calls that are originated by my program.
>>>> >     So I taught of using api_hang_up_hook and I set that variable to
>>>> uuid_kill uuid1 uuid2.
>>>> >     But it only killed the uuid1.
>>>> >
>>>> > Is there any other ways to kill multiple uuid's??
>>>> > please help?
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>> iax:guest at conference.freeswitch.org/888
>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> pstn:+19193869900
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/31ebe7f6/attachment-0002.html 

From mayamatakeshi at gmail.com  Wed Feb 10 15:46:56 2010
From: mayamatakeshi at gmail.com (mayamatakeshi)
Date: Thu, 11 Feb 2010 08:46:56 +0900
Subject: [Freeswitch-users] How to kill multiple UUIDs.
In-Reply-To: <15b9404e1002101545g421280adl93fb7ec036d05e9b@mail.gmail.com>
References: <7d79b3931002100307h4d370a22l6f04cafa5bf5806d@mail.gmail.com>
	<A35CD010-E86F-4B7E-A28B-2A9000B8E009@jerris.com>
	<191c3a031002100759v2fef0f48na5afccf46bb08311@mail.gmail.com>
	<59ad9ca11002100916l6982c88r22e1b45af55b80a1@mail.gmail.com>
	<191c3a031002101129h1103c2e7l6100b2d41807fddf@mail.gmail.com>
	<15b9404e1002101545g421280adl93fb7ec036d05e9b@mail.gmail.com>
Message-ID: <15b9404e1002101546r66ffe4b6ye5842ac5a6590f3a@mail.gmail.com>

On Thu, Feb 11, 2010 at 8:45 AM, mayamatakeshi <mayamatakeshi at gmail.com>wrote:

>
>
> On Thu, Feb 11, 2010 at 4:29 AM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> if you put a var in the leading {} it will get set on all channels
>>
>> {foo=bar}sofia/internal/test at server.com
>
>
> I can see that when I use command bridge, CHANNEL_ORIGINATE shows up with
> variable Other-Leg-Unique-ID set to the UID of the channel executing bridge.
> To be consistent, could I use this same name when calling originate to
> create LegB for a parked LegA? Would it clash with anything?
>

Ah, nevermind. As soon as I hit the send button I realized the variable will
be named:
variable_Other-Leg-Unique-ID.


>
>
>>
>>
>> On Wed, Feb 10, 2010 at 11:16 AM, Henry Huang <red.rain.seven at gmail.com>wrote:
>>
>>> is there an example of setting common var for multiple channels?
>>> or just do it normally like set var=foo before all the bridged call I
>>> would like to hang up at the same time?
>>> and the var name can be anything I want?
>>>
>>>
>>> Henry
>>>
>>>
>>> On Wed, Feb 10, 2010 at 11:59 PM, Anthony Minessale <
>>> anthony.minessale at gmail.com> wrote:
>>>
>>>> or you can set a common var like foo=bar on all the chans and do
>>>>
>>>> hupall normal_clearing foo bar
>>>>
>>>>
>>>>
>>>> On Wed, Feb 10, 2010 at 9:16 AM, Michael Jerris <mike at jerris.com>wrote:
>>>>
>>>>> you can api hangup hook to call
>>>>>
>>>>> lua multi_kill.lua uuid1 uuid2 uuid?.
>>>>>
>>>>> and then write the trivial lua script for that.
>>>>>
>>>>> Mike
>>>>>
>>>>> On Feb 10, 2010, at 6:07 AM, lakshmanan ganapathy wrote:
>>>>>
>>>>> > Hi all,
>>>>> >
>>>>> > My situation is
>>>>> >     A called to 1005 -- Which executes an ESL program.
>>>>> >     Now from the program I will made the parallel call using "api
>>>>> originate [origination_uuid=uuid1]user/1006,[origination_uuid2]user/1001
>>>>> &park()".
>>>>> >     UUID's are obtained from create_uuid.
>>>>> >
>>>>> >     I'll then wait for the api to return, to check whether the call
>>>>> is answered or rejected by the other end.
>>>>> >     But while I'm waiting, if A hangup the call, I just want to kill
>>>>> the calls that are originated by my program.
>>>>> >     So I taught of using api_hang_up_hook and I set that variable to
>>>>> uuid_kill uuid1 uuid2.
>>>>> >     But it only killed the uuid1.
>>>>> >
>>>>> > Is there any other ways to kill multiple uuid's??
>>>>> > please help?
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> Anthony Minessale II
>>>>
>>>> FreeSWITCH http://www.freeswitch.org/
>>>> ClueCon http://www.cluecon.com/
>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>
>>>> AIM: anthm
>>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>> IRC: irc.freenode.net #freeswitch
>>>>
>>>> FreeSWITCH Developer Conference
>>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>>> iax:guest at conference.freeswitch.org/888
>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>> pstn:+19193869900
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/7fa156b0/attachment-0002.html 

From mike at jerris.com  Wed Feb 10 17:13:18 2010
From: mike at jerris.com (Michael Jerris)
Date: Wed, 10 Feb 2010 20:13:18 -0500
Subject: [Freeswitch-users] Any Known Dual-Registration Issue?
In-Reply-To: <3017421995484996A379AADB394D48CB@greyhawk.tonecommander.com>
References: <5AD33F8B1F0848CF9C3A314A60039366@greyhawk.tonecommander.com>
	<035001caa9c8$2dca81c0$895f8540$@com>
	<3017421995484996A379AADB394D48CB@greyhawk.tonecommander.com>
Message-ID: <6F52CA14-07F0-41D3-B137-8623A9F09675@jerris.com>

This is correct, so registrations have a timeout.  If your timeout is long, and you allow multiple registrations, it will keep trying to call the bad registration until timeout.  Our middle-ground setting is to have a setting that specifies only one reg per ip/port.  Anything beyond that is just not possible with the protocol, at least when using udp.  If you are using tcp for the registrations it *might* clear that down properly in these cases.  So you have 3 ways to address this.  Only allow 1 registration per user, only allow 1 registration per contact, allow all the registrations and expect this issue until timeout (set at the length you can tollerate), and a possible 4th of tcp may address this.

Mike


On Feb 10, 2010, at 11:13 AM, Jerry Richards wrote:

> I am using Bria Pro softphones, but it is possible that the PC was
> disconnected before a restart.  Any phone does not necessarily send
> unregister, for example if there is a power outage or it is disconnected
> from the network.  Anyway, I'll just try setting multiple-registrations to
> contact as you suggest.
> 
> Thanks,
> Jerry
> 




From mcampbellsmith at gmail.com  Wed Feb 10 18:24:23 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Thu, 11 Feb 2010 13:24:23 +1100
Subject: [Freeswitch-users] Forbidden using UDP, works with TCP/TLS
Message-ID: <33c87fa31002101824h21e3bf7lb13b68e8d4fd36a2@mail.gmail.com>

Hi!

I had a user registered using TLS transport. That was working fine but
I want to change the ATA over to use UDP instead.

All I thought I should have to do was to change the transport and
ports used to register in the ATA (SPA3102).  However, when I do this,
FS responds with Forbidden.

When I change the settings back to use TCP or TLS, registration is successful.

What would cause FS to respond with forbidden?  I do not change the
username/password fields in either case.

Thanks



From jingwei.yang at gmail.com  Wed Feb 10 19:44:14 2010
From: jingwei.yang at gmail.com (Jingwei Yang)
Date: Thu, 11 Feb 2010 11:44:14 +0800
Subject: [Freeswitch-users] How to record the call upon successful bridge
In-Reply-To: <87f2f3b91002100709m32d975b5rf16823e2b02bd8c2@mail.gmail.com>
References: <13529f9d1002091819y2b363f50vc314d1b6197503b2@mail.gmail.com>
	<257401E3-B32F-4699-96F6-415BA2ADDE88@freeswitch.org>
	<13529f9d1002100039q43775563r49b11136ddda0a0@mail.gmail.com>
	<87f2f3b91002100709m32d975b5rf16823e2b02bd8c2@mail.gmail.com>
Message-ID: <13529f9d1002101944s5a7b9454ya11209763e2cc62@mail.gmail.com>

Hi Michael,

Thanks for the reply. I've upgraded FS to the latest revision 16600 and here
are the log details.

Using execute_on_answer: http://pastebin.freeswitch.org/12109
Using bridge_pre_execute_bleg_app and bridge_pre_execute_bleg_data:
http://pastebin.freeswitch.org/12110

Thanks and best regards,
-Jingwei

On Wed, Feb 10, 2010 at 11:09 PM, Michael Collins <msc at freeswitch.org>wrote:

> Make sure that you are on the latest SVN trunk and retest. Pastebin the
> complete debug log from start to finish of the call.
> -MC
>
>
> On Wed, Feb 10, 2010 at 12:39 AM, Jingwei Yang <jingwei.yang at gmail.com>wrote:
>
>> Hi Ray,
>>
>> Thanks a lot for the replies. Allow me to elaborate a little bit about my
>> situation.
>>
>> 1. client A calls in and parks at Fifo myq.
>> 2. FS connets Agent B to an extension (via originate
>> skypiax/ANY/jingwei.yang 33333)
>> 3. uuid_bridge client A and agent B
>>
>> I believe the spice I can add is in the extension 33333. Here's how I
>> define it.
>>
>>     <extension name="agent_hold">
>>       <condition field="destination_number" expression="^33333$">
>>         <action application="set" data="*execute_on_answer*=record_session
>> /tmp/33333.wav"/>
>>         <action application="answer"/>
>>         <action application="sleep" data="2000"/>
>>         <action application="info"/>
>>         <action application="playback"
>> data="ivr/ivr-hold_connect_call.wav"/>
>>       </condition>
>>     </extension>
>>
>> But the wav file didn't get generated at all. Please advise whether the
>> above is in correct usage.
>>
>> I've also tried bridge_pre_execute_bleg_app and
>> bridge_pre_execute_bleg_data
>>
>>         <action application="set"
>> data="bridge_pre_execute_bleg_app=record_session"/>
>>         <action application="set"
>> data="bridge_pre_execute_bleg_data=/tmp/33333.wav"/>
>>
>> The audio file didn't appear either.
>>
>> The only successful method is by using record_session directly like this:
>>
>>         <action application="record_session" data="/tmp/33333.wav"/>
>>
>> However, in this form, the wav file includes the waiting music, which is
>> not ideal.
>>
>> Thanks and best regards,
>>  -Jingwei
>>
>>
>> On Wed, Feb 10, 2010 at 12:01 PM, Raymond Chandler <
>> intralanman at freeswitch.org> wrote:
>>
>>> errr... .execute_on_answer might be better ;-)
>>> -Ray
>>>
>>> On Feb 9, 2010, at 9:19 PM, Jingwei Yang wrote:
>>>
>>> > Hi,
>>> >
>>> > I'm using uuid_bridge to bridge two calls. May I know how to start
>>> recording only when the bridge succeeds?
>>> >
>>> > Thanks,
>>> > -Jingwei
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/7f276b7b/attachment-0002.html 

From msc at freeswitch.org  Wed Feb 10 20:57:49 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 10 Feb 2010 20:57:49 -0800
Subject: [Freeswitch-users] demo_ivr cannot find sound files via
	relative paths
In-Reply-To: <78030B9E01DD48F8B024A97035F96282@greyhawk.tonecommander.com>
References: <78030B9E01DD48F8B024A97035F96282@greyhawk.tonecommander.com>
Message-ID: <87f2f3b91002102057t7b039221kf3975553414880cf@mail.gmail.com>

If I read this log correctly it failed to find the "invalid entry" file but
it did find the phrases just fine. Can you confirm the presence of this
file:
/usr/local/freeswitch/sounds/en/us/callie/ivr/8000/ivr-that_was_an_invalid_entry.wav

(It looks like this call is at 8kHz so that's where I'm assuming FS is
looking to find the sound file...)

-MC

On Wed, Feb 10, 2010 at 10:50 AM, Robert Hadley
<robert.hadley at teotech.com>wrote:

>  Hi,
>
>
>
> It appears a recent change (possibly the new sounds_dir variable or the new
> ivr_menu folder?) may have broken relative sound file paths in the IVR.  I
> built a today?s trunk version and installed to the default location. Using
> the default conf files the demo_ivr cannot find files based on the relative
> paths specified in ivr_menus/demo_ivr.xml.
>
>
>
> [root at TEO-UCM-T2 conf]# cat ivr_menus/demo_ivr.xml
>
> <include>
>
>   <!-- demo IVR setup -->
>
>   <!-- demo IVR, Main Menu -->
>
>   <menu name="demo_ivr"
>
>       greet-long="phrase:demo_ivr_main_menu"
>
>       greet-short="phrase:demo_ivr_main_menu_short"
>
>       invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"
>
>       exit-sound="voicemail/vm-goodbye.wav"
>
>
>
>
>
> 2010-02-10 10:32:15.300337 [DEBUG] switch_ivr_menu.c:414 Executing IVR menu
> demo_ivr
>
> 2010-02-10 10:32:15.300337 [DEBUG] switch_ivr_play_say.c:118 No language
> specified - Using [en]
>
> 2010-02-10 10:32:15.300337 [DEBUG] switch_ivr_play_say.c:273 Handle
> play-file:[ivr/ivr-welcome_to_freeswitch.wav] (en:en)
>
> 2010-02-10 10:32:15.300337 [DEBUG] switch_ivr_play_say.c:1158 Codec
> Activated L16 at 8000hz 1 channels 20ms
>
> 2010-02-10 10:32:17.922147 [DEBUG] switch_ivr_play_say.c:1450 done playing
> file
>
> 2010-02-10 10:32:18.042149 [DEBUG] switch_ivr_play_say.c:273 Handle
> play-file:[ivr/ivr-this_ivr_will_let_you_test_features.wav] (en:en)
>
> 2010-02-10 10:32:18.042149 [DEBUG] switch_ivr_play_say.c:1158 Codec
> Activated L16 at 8000hz 1 channels 20ms
>
> 2010-02-10 10:32:19.962158 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF 2:800
>
> 2010-02-10 10:32:19.962158 [DEBUG] switch_ivr_play_say.c:1450 done playing
> file
>
> 2010-02-10 10:32:20.082156 [DEBUG] switch_ivr_menu.c:329 waiting for 3/4
> digits t/o 2000
>
> 2010-02-10 10:32:20.120617 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF 2:400
>
> 2010-02-10 10:32:20.442158 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF 2:800
>
> 2010-02-10 10:32:20.682162 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF 2:800
>
> 2010-02-10 10:32:20.682162 [DEBUG] switch_ivr_menu.c:376 digits '2222'
>
> 2010-02-10 10:32:20.682162 [DEBUG] switch_ivr_menu.c:470 action regex
> [2222] [/^(10[01][0-9])$/] [0]
>
> 2010-02-10 10:32:20.682162 [DEBUG] switch_ivr_menu.c:560 IVR menu
> 'demo_ivr' caught invalid input '2222'
>
> 2010-02-10 10:32:20.682162 [ERR] mod_sndfile.c:194 Error Opening File
> [/usr/local/freeswitch/sounds/ivr/ivr-that_was_an_invalid_entry.wav] [System
> error : No such file or directory.]
>
> 2010-02-10 10:32:21.701505 [DEBUG] switch_ivr_play_say.c:118 No language
> specified - Using [en]
>
> 2010-02-10 10:32:21.701505 [DEBUG] switch_ivr_play_say.c:273 Handle
> play-file:[silence_stream://1000] (en:en)
>
> 2010-02-10 10:32:21.701505 [DEBUG] switch_ivr_play_say.c:1158 Codec
> Activated L16 at 8000hz 1 channels 20ms
>
> 2010-02-10 10:32:22.700352 [DEBUG] switch_ivr_play_say.c:1450 done playing
> file
>
>
>
> Regards,
>
> Robert
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100210/61638b2e/attachment-0002.html 

From msc at freeswitch.org  Wed Feb 10 21:18:42 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 10 Feb 2010 21:18:42 -0800
Subject: [Freeswitch-users] Interesting article on OSS telephony
Message-ID: <87f2f3b91002102118x27df10c4qc7fc8d1fad2b7b92@mail.gmail.com>

Check out the article on TMCnet.com, linked-to here:
http://www.freeswitch.org/node/238

-Michael
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100210/e3c9608b/attachment-0002.html 

From mike at jerris.com  Wed Feb 10 21:21:06 2010
From: mike at jerris.com (Michael Jerris)
Date: Thu, 11 Feb 2010 00:21:06 -0500
Subject: [Freeswitch-users] Lua script hangup detect
In-Reply-To: <4B72F2A8.4070503@gmail.com>
References: <314dc3f81002100117h6b1bc49cna989624a1ae4e0e2@mail.gmail.com>	<4B729155.7010708@gmail.com>	<314dc3f81002100800o41fd6561w394d5e83e69d3d80@mail.gmail.com>	<4B72DFF9.4040402@gmail.com>
	<1b46b4e81002100932x7af486ekd4b05f4750cace20@mail.gmail.com>
	<4B72F2A8.4070503@gmail.com>
Message-ID: <4CA5432B-5F6E-438E-B6EF-3277847E7922@jerris.com>

lua runs until you finish your script.  You need to exit when you are supposed to.  We have session::ready() for you to test this.

Mike

On Feb 10, 2010, at 12:53 PM, Nazim Agabekov wrote:

> I think it continues to run until it finishes. I'll check it tomorrow on my test system.
> 
> On 02/10/2010 09:32 PM, Nicolas Brenner wrote:
>> 
>> 
>> On Wed, Feb 10, 2010 at 1:34 PM, Nazim Agabekov <nazim.agabekov at gmail.com> wrote:
>> Lua script is not terminating immediately on hangup.
>> 
>> 
>> When does it terminate then? Will the script terminate when it finishes running or does it need some special instruction?
>> 
>>  
>> This behavior allows user to finalize the script nicely (free dynamically allocated resources, update logs, e.t.c)
>> 
>>> maybe it will stuck in a few situation before it reach to session:status check point.
>>> 
>>> like GetDigit or something else? (sorry, I dont have FS right now, need test it tomorrow)
>> I've never encountered such a problem. Usually GetDigit-like functions have timeout parameter, so they don't block forever.
>> Just check the session status often and it will work like a charm ;
>> 
>> 
>> 
>> On 02/10/2010 08:00 PM, Chia-Yen Wu wrote:
>>> 
>>> thx for reply, but shouldnt Lua script terminate when client hangs up?
>>> 
>>> maybe it will stuck in a few situation before it reach to session:status check point.
>>> 
>>> like GetDigit or something else? (sorry, I dont have FS right now, need test it tomorrow)
>>> 
>>> 2010/2/10 Nazim Agabekov <nazim.agabekov at gmail.com>
>>> Hello,
>>> Can you pastebin your script?
>>> 
>>> http://pastebin.freeswitch.org
>>> 
>>> 
>>> On 02/10/2010 01:17 PM, Chia-Yen Wu wrote:
>>>> Hello, 
>>>> i tried to use Lua script to replace xml macro in dialplan,
>>>> but I found out that Lua wont terminate if client hangup,
>>>> ,so the session is still on but client is already hangup,
>>>> is there a way to avoid this ?
>>>> 
>>>> 
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>   
>>> 
>>> 
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>> 
>>> 
>>> 
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>   
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>   
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/13253628/attachment-0002.html 

From paul at apcl.us  Wed Feb 10 18:12:11 2010
From: paul at apcl.us (Paul Levin)
Date: Wed, 10 Feb 2010 21:12:11 -0500
Subject: [Freeswitch-users] is there a command to get a list of registered
	accounts?
Message-ID: <4B73677B.6020406@apcl.us>

An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100210/18b7c23c/attachment-0002.html 

From paul at apcl.us  Wed Feb 10 18:15:34 2010
From: paul at apcl.us (Paul Levin)
Date: Wed, 10 Feb 2010 21:15:34 -0500
Subject: [Freeswitch-users] how to set From address for voicemail emails?
Message-ID: <4B736846.1040908@apcl.us>

An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100210/ba1e3c35/attachment-0002.html 

From mike at jerris.com  Wed Feb 10 21:25:08 2010
From: mike at jerris.com (Michael Jerris)
Date: Thu, 11 Feb 2010 00:25:08 -0500
Subject: [Freeswitch-users] Forbidden using UDP, works with TCP/TLS
In-Reply-To: <33c87fa31002101824h21e3bf7lb13b68e8d4fd36a2@mail.gmail.com>
References: <33c87fa31002101824h21e3bf7lb13b68e8d4fd36a2@mail.gmail.com>
Message-ID: <2EF27EFC-1634-46A3-B438-2EE99BEF58F1@jerris.com>

Look at the FreeSWITCH debug logs and compare the differences, they should tell you why.

Mike

On Feb 10, 2010, at 9:24 PM, Mark Campbell-Smith wrote:

> Hi!
> 
> I had a user registered using TLS transport. That was working fine but
> I want to change the ATA over to use UDP instead.
> 
> All I thought I should have to do was to change the transport and
> ports used to register in the ATA (SPA3102).  However, when I do this,
> FS responds with Forbidden.
> 
> When I change the settings back to use TCP or TLS, registration is successful.
> 
> What would cause FS to respond with forbidden?  I do not change the
> username/password fields in either case.




From mcampbellsmith at gmail.com  Wed Feb 10 21:29:31 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Thu, 11 Feb 2010 16:29:31 +1100
Subject: [Freeswitch-users] is there a command to get a list of
	registered accounts?
In-Reply-To: <4B73677B.6020406@apcl.us>
References: <4B73677B.6020406@apcl.us>
Message-ID: <33c87fa31002102129r5d692dc5pfb414b80b47be221@mail.gmail.com>

sofia status profile xxx (where xxx is usually internal or external)

On Thu, Feb 11, 2010 at 1:12 PM, Paul Levin <paul at apcl.us> wrote:
> At the FreeSwitch console is there a command I can enter to get a list of
> all sip accounts that are currently registered?
> ??? Thanks,
> ??? Paul
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From mike at jerris.com  Wed Feb 10 21:33:39 2010
From: mike at jerris.com (Michael Jerris)
Date: Thu, 11 Feb 2010 00:33:39 -0500
Subject: [Freeswitch-users] demo_ivr cannot find sound files via
	relative paths
In-Reply-To: <87f2f3b91002102057t7b039221kf3975553414880cf@mail.gmail.com>
References: <78030B9E01DD48F8B024A97035F96282@greyhawk.tonecommander.com>
	<87f2f3b91002102057t7b039221kf3975553414880cf@mail.gmail.com>
Message-ID: <E4BC77C3-8271-4F30-B6F8-6822E4747BC3@jerris.com>

did you make any changes to the default configs?  what is in your vars.xml related to sounds?  the relative paths were always relative to that dir, did you make any changes to the sounds prefix ?

Mike

On Feb 10, 2010, at 11:57 PM, Michael Collins wrote:

> If I read this log correctly it failed to find the "invalid entry" file but it did find the phrases just fine. Can you confirm the presence of this file:
> /usr/local/freeswitch/sounds/en/us/callie/ivr/8000/ivr-that_was_an_invalid_entry.wav
> 
> (It looks like this call is at 8kHz so that's where I'm assuming FS is looking to find the sound file...)
> 
> -MC
> 
> On Wed, Feb 10, 2010 at 10:50 AM, Robert Hadley <robert.hadley at teotech.com> wrote:
> Hi,
> 
>  
> It appears a recent change (possibly the new sounds_dir variable or the new ivr_menu folder?) may have broken relative sound file paths in the IVR.  I built a today?s trunk version and installed to the default location. Using the default conf files the demo_ivr cannot find files based on the relative paths specified in ivr_menus/demo_ivr.xml.
> 
>  
> [root at TEO-UCM-T2 conf]# cat ivr_menus/demo_ivr.xml
> 
> <include>
> 
>   <!-- demo IVR setup -->
> 
>   <!-- demo IVR, Main Menu -->
> 
>   <menu name="demo_ivr"
> 
>       greet-long="phrase:demo_ivr_main_menu"
> 
>       greet-short="phrase:demo_ivr_main_menu_short"
> 
>       invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"
> 
>       exit-sound="voicemail/vm-goodbye.wav"
> 
>  
>  
> 2010-02-10 10:32:15.300337 [DEBUG] switch_ivr_menu.c:414 Executing IVR menu demo_ivr
> 
> 2010-02-10 10:32:15.300337 [DEBUG] switch_ivr_play_say.c:118 No language specified - Using [en]
> 
> 2010-02-10 10:32:15.300337 [DEBUG] switch_ivr_play_say.c:273 Handle play-file:[ivr/ivr-welcome_to_freeswitch.wav] (en:en)
> 
> 2010-02-10 10:32:15.300337 [DEBUG] switch_ivr_play_say.c:1158 Codec Activated L16 at 8000hz 1 channels 20ms
> 
> 2010-02-10 10:32:17.922147 [DEBUG] switch_ivr_play_say.c:1450 done playing file
> 
> 2010-02-10 10:32:18.042149 [DEBUG] switch_ivr_play_say.c:273 Handle play-file:[ivr/ivr-this_ivr_will_let_you_test_features.wav] (en:en)
> 
> 2010-02-10 10:32:18.042149 [DEBUG] switch_ivr_play_say.c:1158 Codec Activated L16 at 8000hz 1 channels 20ms
> 
> 2010-02-10 10:32:19.962158 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF 2:800
> 
> 2010-02-10 10:32:19.962158 [DEBUG] switch_ivr_play_say.c:1450 done playing file
> 
> 2010-02-10 10:32:20.082156 [DEBUG] switch_ivr_menu.c:329 waiting for 3/4 digits t/o 2000
> 
> 2010-02-10 10:32:20.120617 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF 2:400
> 
> 2010-02-10 10:32:20.442158 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF 2:800
> 
> 2010-02-10 10:32:20.682162 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF 2:800
> 
> 2010-02-10 10:32:20.682162 [DEBUG] switch_ivr_menu.c:376 digits '2222'
> 
> 2010-02-10 10:32:20.682162 [DEBUG] switch_ivr_menu.c:470 action regex [2222] [/^(10[01][0-9])$/] [0]
> 
> 2010-02-10 10:32:20.682162 [DEBUG] switch_ivr_menu.c:560 IVR menu 'demo_ivr' caught invalid input '2222'
> 
> 2010-02-10 10:32:20.682162 [ERR] mod_sndfile.c:194 Error Opening File [/usr/local/freeswitch/sounds/ivr/ivr-that_was_an_invalid_entry.wav] [System error : No such file or directory.]
> 
> 2010-02-10 10:32:21.701505 [DEBUG] switch_ivr_play_say.c:118 No language specified - Using [en]
> 
> 2010-02-10 10:32:21.701505 [DEBUG] switch_ivr_play_say.c:273 Handle play-file:[silence_stream://1000] (en:en)
> 
> 2010-02-10 10:32:21.701505 [DEBUG] switch_ivr_play_say.c:1158 Codec Activated L16 at 8000hz 1 channels 20ms
> 
> 2010-02-10 10:32:22.700352 [DEBUG] switch_ivr_play_say.c:1450 done playing file
> 
>  
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/0a233797/attachment-0002.html 

From mike at jerris.com  Wed Feb 10 21:34:18 2010
From: mike at jerris.com (Michael Jerris)
Date: Thu, 11 Feb 2010 00:34:18 -0500
Subject: [Freeswitch-users] is there a command to get a list of
	registered accounts?
In-Reply-To: <4B73677B.6020406@apcl.us>
References: <4B73677B.6020406@apcl.us>
Message-ID: <1FD15E7C-1942-4C3A-9D64-36CAB61C49E5@jerris.com>

sofia status profile <profile_name>

On Feb 10, 2010, at 9:12 PM, Paul Levin wrote:
> At the FreeSwitch console is there a command I can enter to get a list of all sip accounts that are currently registered?

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/15c013d3/attachment-0002.html 

From mike at jerris.com  Wed Feb 10 21:36:47 2010
From: mike at jerris.com (Michael Jerris)
Date: Thu, 11 Feb 2010 00:36:47 -0500
Subject: [Freeswitch-users] how to set From address for voicemail emails?
In-Reply-To: <4B736846.1040908@apcl.us>
References: <4B736846.1040908@apcl.us>
Message-ID: <7471DDD5-0A73-4F18-AF90-BA8534EBC9E7@jerris.com>

This is the difference between what is sent to the mail server in the mime content, and what is passed as MAIL FROM:  to the smtp server.  The latter is controlled by that param, the former is in the template.

Mike

On Feb 10, 2010, at 9:15 PM, Paul Levin wrote:

> I am running FreeSwitch on Windows.  I have msmtp setup and voice mail emails are being sent. 
> 
> I have msmtp configured to set a "From" address of me at mydomain.com, but when FreeSwitch sends an email with a voice mail message from Alice, the From address of the email is Alice at sipServerDomain.com.  According to the Mod voicemail document (http://wiki.freeswitch.org/wiki/Mod_voicemail) the email_email-from parameter should control this, but I tried setting it in conf\autoload_configs\directory.conf.xml (as per that document) and also in Bob.xml (the account getting the voicemail).  Neither place changed the value being used.  
> 
> How do I get this changed?

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/1b75d3e5/attachment-0002.html 

From mailinglist at fribert.dk  Wed Feb 10 21:45:45 2010
From: mailinglist at fribert.dk (mailinglist)
Date: Thu, 11 Feb 2010 06:45:45 +0100
Subject: [Freeswitch-users] Need help setting up a feature
Message-ID: <4B73A799020000E100000470@mail.fribert.dk>

Sorry for the repost, but the previous thread just died :-)
 
I'm trying to get the possibility of transfering an incoming call from one extension to another, and give the possibility of turning it into a conference.
I don't have a 'transfer' button.
I do have an 'R' button on the Siemens handsets, and a 'Flash' button on the Sipura. The 'Flash' button gives me a new dialtone, gives the caller MOH, and then I can dial the new extension, and transfer the call, but not create a conference.
But the Siemens handset does not have a 'flash', and pressing the R doesn't do anything.
 
It might be two different features 'transfer' and 'conference'...

But I thought that using the bind_meta_app would accomplish both.
 
It's on an incoming call from the outside.
So the situation:
The Public folder has an entry that matches the dialed number, and does a transfer to 8202.
Then the dialplan matches the 8202 with a group, and the phone rings.
Somebody picks it up, finds out that it needs to be transferred to another extension, or transferred to a conference with a second extension.
How do I construct that?
 
 
 
Best regards
Fribse
 
/Running Freeswitch on pfSense running on a VMWare ESXi 4.0
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/89c5612d/attachment-0002.html 

From ustcorporation at yahoo.com  Wed Feb 10 23:30:19 2010
From: ustcorporation at yahoo.com (teldev)
Date: Wed, 10 Feb 2010 23:30:19 -0800 (PST)
Subject: [Freeswitch-users] Seeking Advice on SIP Phones like Aastra
In-Reply-To: <9DFA265F-8632-466B-8910-1E2F9D956908@jerris.com>
References: <9DFA265F-8632-466B-8910-1E2F9D956908@jerris.com>
Message-ID: <1265873419114-4553224.post@n2.nabble.com>


Hello Mike,

"Can you post for all to see some idea of how these applications work,
lanagages used, some samples so we can see if we can get some interest in
this?"

"I am not sure the phone gui has much to do with FreeSWITCH at all, other
than pulling a little data from the databases (again, ODBC highly
recommended)" 

-->  The idea would be to create a phone GUI on Aastra 6739i that would
enable the touchscreen to do something like what is shown in these demos:

http://www.youtube.com/watch?v=xZDHibW1gUs
http://www.youtube.com/watch?v=Y58TPNCOTZI

Perform FreeSWITCH actions: answer calls, place calls, setup conferences/add
participants, etc.
Display FreeSWITCH states: voice mail waiting count, extensions that are
online/offline, call log, etc. 
Introduce new actions: dial By name directory with pictures, visual setup of
conferences, etc.  

Since the Aastra 6739i, SNOM 870, and some others have built-in XML browsers
it seems like they could be programmed to interact well with FreeSWITCH and
various third party systems such as online feeds like weather reports, news,
stock prices.  Aastra is taking the IPhone approach of letting developers
create the apps.  They even have a similar slogan "There's an App for That"
versus Apple's "There's an app for just about anything". 

http://www.ucstrategies.com/unified-communications-expert-views/aastra-phone-theres-an-app-for-that.aspx

I don't know to what extent FreeSWITCH interacts with this phone out of the
box, it reportedly works well with Asterisk though.  I'm sure FS will likely
understand certain button presses like hold, mute, speaker, transfer, etc. 
I will be ordering one within a few days. 

A few quesions for the FS community:

Does this project have merit and fulfill a need?  (I have yet to use FS as a
PBX)

Can anyone shed some light on what features on the phone will work and what
I've mentioned will require custom development?

Any ideas on how to architect an integration that would leverage what FS
already does?  Currently our IVR work uses Javascript/SpiderMonkey to call
our own web service for database reads/writes.  For this project, we'd need
to interact more directly with FS and learn more about it's internals.  

teldev

-- 
View this message in context: http://n2.nabble.com/Re-Seeking-Advice-on-SIP-Phones-like-Aastra-tp4543930p4553224.html
Sent from the freeswitch-users mailing list archive at Nabble.com.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100210/cb7e6d7b/attachment-0002.html 

From nazim.agabekov at gmail.com  Wed Feb 10 23:45:02 2010
From: nazim.agabekov at gmail.com (Nazim Agabekov)
Date: Thu, 11 Feb 2010 11:45:02 +0400
Subject: [Freeswitch-users] Lua script hangup detect
In-Reply-To: <4CA5432B-5F6E-438E-B6EF-3277847E7922@jerris.com>
References: <314dc3f81002100117h6b1bc49cna989624a1ae4e0e2@mail.gmail.com>	<4B729155.7010708@gmail.com>	<314dc3f81002100800o41fd6561w394d5e83e69d3d80@mail.gmail.com>	<4B72DFF9.4040402@gmail.com>	<1b46b4e81002100932x7af486ekd4b05f4750cace20@mail.gmail.com>	<4B72F2A8.4070503@gmail.com>
	<4CA5432B-5F6E-438E-B6EF-3277847E7922@jerris.com>
Message-ID: <4B73B57E.9040901@gmail.com>

Thanks Mike! Mod_Lua is great. I had experience with a lot of
proprietary IVR systems in the past.
FreeSWITCH and Mod_lua + luasql beats them all. Functionality is really
impressive.

On 02/11/2010 09:21 AM, Michael Jerris wrote:
> lua runs until you finish your script.  You need to exit when you are
> supposed to.  We have session::ready() for you to test this.
>
> Mike
>
> On Feb 10, 2010, at 12:53 PM, Nazim Agabekov wrote:
>
>> I think it continues to run until it finishes. I'll check it tomorrow
>> on my test system.
>>
>> On 02/10/2010 09:32 PM, Nicolas Brenner wrote:
>>>
>>> On Wed, Feb 10, 2010 at 1:34 PM, Nazim Agabekov
>>> <nazim.agabekov at gmail.com <mailto:nazim.agabekov at gmail.com>> wrote:
>>>
>>>     Lua script is not terminating immediately on hangup.
>>>
>>>
>>>
>>> When does it terminate then? Will the script terminate when it
>>> finishes running or does it need some special instruction?
>>>
>>>  
>>>
>>>     This behavior allows user to finalize the script nicely (free
>>>     dynamically allocated resources, update logs, e.t.c)
>>>
>>>>     maybe it will stuck in a few situation before it reach to
>>>>     session:status check point.
>>>>
>>>>     like GetDigit or something else? (sorry, I dont have FS right
>>>>     now, need test it tomorrow)
>>>     I've never encountered such a problem. Usually GetDigit-like
>>>     functions have timeout parameter, so they don't block forever.
>>>     Just check the session status often and it will work like a charm ;
>>>
>>>
>>>
>>>     On 02/10/2010 08:00 PM, Chia-Yen Wu wrote:
>>>>     thx for reply, but shouldnt Lua script terminate when client
>>>>     hangs up?
>>>>
>>>>     maybe it will stuck in a few situation before it reach to
>>>>     session:status check point.
>>>>
>>>>     like GetDigit or something else? (sorry, I dont have FS right
>>>>     now, need test it tomorrow)
>>>>
>>>>     2010/2/10 Nazim Agabekov <nazim.agabekov at gmail.com
>>>>     <mailto:nazim.agabekov at gmail.com>>
>>>>
>>>>         Hello,
>>>>         Can you pastebin your script?
>>>>
>>>>         http://pastebin.freeswitch.org
>>>>         <http://pastebin.freeswitch.org/>
>>>>
>>>>
>>>>         On 02/10/2010 01:17 PM, Chia-Yen Wu wrote:
>>>>>         Hello, 
>>>>>         i tried to use Lua script to replace xml macro in dialplan,
>>>>>         but I found out that Lua wont terminate if client hangup,
>>>>>         ,so the session is still on but client is already hangup,
>>>>>         is there a way to avoid this ?
>>>>>
>>>>>
>>>>>         _______________________________________________
>>>>>         FreeSWITCH-users mailing list
>>>>>         FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>         http://www.freeswitch.org <http://www.freeswitch.org/>
>>>>>           
>>>>
>>>>
>>>>         _______________________________________________
>>>>         FreeSWITCH-users mailing list
>>>>         FreeSWITCH-users at lists.freeswitch.org
>>>>         <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>         http://www.freeswitch.org <http://www.freeswitch.org/>
>>>>
>>>>
>>>>
>>>>     _______________________________________________
>>>>     FreeSWITCH-users mailing list
>>>>     FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>     http://www.freeswitch.org <http://www.freeswitch.org/>
>>>>       
>>>
>>>
>>>     _______________________________________________
>>>     FreeSWITCH-users mailing list
>>>     FreeSWITCH-users at lists.freeswitch.org
>>>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>     http://www.freeswitch.org <http://www.freeswitch.org/>
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>   
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/32984796/attachment-0002.html 

From peter.olsson at visionutveckling.se  Wed Feb 10 23:50:31 2010
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Thu, 11 Feb 2010 08:50:31 +0100
Subject: [Freeswitch-users] h323 - sip call is not working
In-Reply-To: <65d96fc81002101227n3febe3bdgb0c36d767fa5be8e@mail.gmail.com>
References: <20100210104911.A632A11F49@mail.nstel.ru>
	<65d96fc81002101113p44fe6f43oba72c3a2e13785cc@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C5576CE4923@cooper>
	<65d96fc81002101227n3febe3bdgb0c36d767fa5be8e@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C5577127F67@cooper>

I'm using the opal module, since mod_h323 is not available in Windows yet. We're not so dependant of all the correct release codes etc, we use it mostly for ASR and voicemail applications, but I do know there have been discussions about this on the mailing list. My recommendation is to use SIP to the IPO instead (mostly beacuse SIP is much better supported by FS), we're using that more and more.

/Peter

Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Tihomir Culjaga
Skickat: den 10 februari 2010 21:27
Till: freeswitch-users at lists.freeswitch.org
?mne: Re: [Freeswitch-users] h323 - sip call is not working


On Wed, Feb 10, 2010 at 8:42 PM, Peter Olsson <peter.olsson at visionutveckling.se<mailto:peter.olsson at visionutveckling.se>> wrote:
I've been running both h323 and SIP between FS and Avaya IPO for some time. No problems at all. :)

But make sure to enable h323 fast start and disable "direct media path" in the IPO, if I remember correctly these where the only two parameters that made any real difference for me.

But I do recommenf to use SIP, since it's much better supported by FS.

/Peter

Peter, what H323plus version are you using ?

did you noticed q931 release cause is not mapped H323 => SIP correctly ?

!DSPAM:4b73196232931389031035!
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/905e4fcb/attachment-0002.html 

From ustcorporation at yahoo.com  Thu Feb 11 00:01:47 2010
From: ustcorporation at yahoo.com (teldev)
Date: Thu, 11 Feb 2010 00:01:47 -0800 (PST)
Subject: [Freeswitch-users] Seeking Advice on SIP Phones like Aastra
In-Reply-To: <9DFA265F-8632-466B-8910-1E2F9D956908@jerris.com>
References: <9DFA265F-8632-466B-8910-1E2F9D956908@jerris.com>
Message-ID: <1265875307968-4553322.post@n2.nabble.com>


Hello Mike,

Last post lost all formatting when I used Nabble, this is readable now.

"Can you post for all to see some idea of how these applications work,
lanagages used, some samples so we can see if we can get some interest in
this?"

"I am not sure the phone gui has much to do with FreeSWITCH at all, other
than pulling a little data from the databases (again, ODBC highly
recommended)" 

-->  The idea would be to create a phone GUI on Aastra 6739i that would
enable the touchscreen to do something like what is shown in these demos:

http://www.youtube.com/watch?v=xZDHibW1gUs
http://www.youtube.com/watch?v=Y58TPNCOTZI

Perform FreeSWITCH actions: answer calls, place calls, setup conferences/add
participants, etc.
Display FreeSWITCH states: voice mail waiting count, extensions that are
online/offline, call log, etc. 
Introduce new actions: dial By name directory with pictures, visual setup of
conferences, etc.  

Since the Aastra 6739i, SNOM 870, and some others have built-in XML browsers
it seems like they could be programmed to interact well with FreeSWITCH and
various third party systems such as online feeds like weather reports, news,
stock prices.  Aastra is taking the IPhone approach of letting developers
create the apps.  They even have a similar slogan "There's an App for That"
versus Apple's "There's an app for just about anything". 

http://www.ucstrategies.com/unified-communications-expert-views/aastra-phone-theres-an-app-for-that.aspx

I don't know to what extent FreeSWITCH interacts with this phone out of the
box, it reportedly works well with Asterisk though.  I'm sure FS will likely
understand certain button presses like hold, mute, speaker, transfer, etc. 
I will be ordering one within a few days. 

A few quesions for the FS community:

Does this project have merit and fulfill a need?  (I have yet to use FS as a
PBX)

Can anyone shed some light on what features on the phone will work and what
I've mentioned will require custom development?

Any ideas on how to architect an integration that would leverage what FS
already does?  Currently our IVR work uses Javascript/SpiderMonkey to call
our own web service for database reads/writes.  For this project, we'd need
to interact more directly with FS and learn more about it's internals.  
-- 
View this message in context: http://n2.nabble.com/Re-Seeking-Advice-on-SIP-Phones-like-Aastra-tp4543930p4553322.html
Sent from the freeswitch-users mailing list archive at Nabble.com.



From kond at nstel.ru  Thu Feb 11 00:04:31 2010
From: kond at nstel.ru (Nikolay Kondratyev)
Date: Thu, 11 Feb 2010 11:04:31 +0300
Subject: [Freeswitch-users] h323 - sip call is not working
In-Reply-To: <65d96fc81002101113p44fe6f43oba72c3a2e13785cc@mail.gmail.com>
Message-ID: <20100211080431.DD68E11F34@mail.nstel.ru>

Tihomir,

I've just sent the tcpdump to your address (because I think mail list will
not accept 300K attachement).

Thanks and regards,

Nikolay.

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Tihomir
Culjaga
Sent: Wednesday, February 10, 2010 10:14 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] h323 - sip call is not working

 

 

On Wed, Feb 10, 2010 at 11:49 AM, Nikolay Kondratyev <kond at nstel.ru> wrote:

Hi all,

I compiled FreeSWITCH Version 1.0.5-20100209-0400 (16587M) with mod_h323.

When I call from h323 (Avaya IPOffice) to local fs extention (x-lite) I hear
ring back, but when x-lite picks up, he hears silence, while IPOffice user
continues to hear ringback.

The log is at the http://pastebin.freeswitch.org/12091

My configuration is almost default, several local extentions added, and
h323.conf from http://wiki.freeswitch.org/wiki/Mod_h323

5840 - user at IPOffice

2853 - x-lite registered at FS

IPOffice ip address: 172.23.14.2

FS ip address 172.23.22.49

 

Can anybody please advise how to solve that?

Is it a configuration or a software problem?

 


Please can you send me the tcpdump as well (not filtered). IPOffice i known
to have a "broken" H323 stack.
did you try to play with tunneling and h245 in setup settings as well ?

It looks like there is some h245 negotiation still pending but cant see that
from the logs.
 

Thanks in advance,

Nikolay.


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/4ac225cd/attachment-0002.html 

From kond at nstel.ru  Thu Feb 11 00:08:18 2010
From: kond at nstel.ru (Nikolay Kondratyev)
Date: Thu, 11 Feb 2010 11:08:18 +0300
Subject: [Freeswitch-users] can't see sip messages in the log file
In-Reply-To: <b39273531002100650j2ff8652cu733256c88e5b7aea@mail.gmail.com>
Message-ID: <20100211080818.D469D1210F@mail.nstel.ru>

What is the version number where "sofia help" will show tracelevel? 

Thanks and regards,

Nikolay.

 

Thanks.  I've added that to the sofia help/completion.

On Wed, Feb 10, 2010 at 8:25 AM, Nikolay Kondratyev <kond at nstel.ru> wrote:

After some experiments i clarified this question.
SIP messages go into the freeswitch log when:
<param name="sip-trace" value="yes"/> ("sofia profile profile-name siptrace
on" is cli equivalent for this parameter)
AND
Sofia tracelevel is set to info (sofia tracelevel info).

The default sofia tracelevel is 'console'. That's why I did not see sip
messages in the log after turning on just "siptrace".

Thanks and regards,
Nikolay.


> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-
> users-bounces at lists.freeswitch.org] On Behalf Of Nikolay Kondratyev
> Sent: Wednesday, February 10, 2010 2:00 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] can't see sip messages in the log file
>
> Jason, thanks for the reply.
> Isn't "sofia profile internal siptrace on" a command line equivalent of
> <param name="sip-trace" value="yes"/>?
>
> Any way I tried it, but with the same result.
> I still don't see SIP.
>
> Thanks and regards,
> Nikolay.
>
> > > Can anybody please advise how to include sip messages into the log
> file?
> >
> > <param name="sip-trace" value="yes"/>
> > in the SIP profile you want to trace, then
> > sofia profile profile-name restart reloadxml
> > or restarting FreeSWITCH should do it.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




-- 
-Rupa

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/a32a0fe5/attachment-0002.html 

From kond at nstel.ru  Thu Feb 11 00:18:17 2010
From: kond at nstel.ru (Nikolay Kondratyev)
Date: Thu, 11 Feb 2010 11:18:17 +0300
Subject: [Freeswitch-users] h323 - sip call is not working
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C5576CE4923@cooper>
Message-ID: <20100211081817.BAB611200B@mail.nstel.ru>

> But I do recommenf to use SIP, since it's much better supported by FS.
> 
> /Peter
But SIP is poorly supported by IPO.
Thanks and regards,
Nikolay.




From mike at jerris.com  Thu Feb 11 00:36:16 2010
From: mike at jerris.com (Michael Jerris)
Date: Thu, 11 Feb 2010 03:36:16 -0500
Subject: [Freeswitch-users] Lua script hangup detect
In-Reply-To: <4B73B57E.9040901@gmail.com>
References: <314dc3f81002100117h6b1bc49cna989624a1ae4e0e2@mail.gmail.com>	<4B729155.7010708@gmail.com>	<314dc3f81002100800o41fd6561w394d5e83e69d3d80@mail.gmail.com>	<4B72DFF9.4040402@gmail.com>	<1b46b4e81002100932x7af486ekd4b05f4750cace20@mail.gmail.com>	<4B72F2A8.4070503@gmail.com>
	<4CA5432B-5F6E-438E-B6EF-3277847E7922@jerris.com>
	<4B73B57E.9040901@gmail.com>
Message-ID: <D8FFF1FB-7B2D-49BC-940C-445EBF838BE3@jerris.com>

luasql, now with free memory leaks, while supplies last (don't use mysql with lua? or really ever)

Mike

On Feb 11, 2010, at 2:45 AM, Nazim Agabekov wrote:

> Thanks Mike! Mod_Lua is great. I had experience with a lot of proprietary IVR systems in the past.
> FreeSWITCH and Mod_lua + luasql beats them all. Functionality is really impressive.
> 
> On 02/11/2010 09:21 AM, Michael Jerris wrote:
>> 
>> lua runs until you finish your script.  You need to exit when you are supposed to.  We have session::ready() for you to test this.
>> 
>> Mike
>> 
>> On Feb 10, 2010, at 12:53 PM, Nazim Agabekov wrote:
>> 
>>> I think it continues to run until it finishes. I'll check it tomorrow on my test system.
>>> 




From mike at jerris.com  Thu Feb 11 00:36:55 2010
From: mike at jerris.com (Michael Jerris)
Date: Thu, 11 Feb 2010 03:36:55 -0500
Subject: [Freeswitch-users] Seeking Advice on SIP Phones like Aastra
In-Reply-To: <1265873419114-4553224.post@n2.nabble.com>
References: <9DFA265F-8632-466B-8910-1E2F9D956908@jerris.com>
	<1265873419114-4553224.post@n2.nabble.com>
Message-ID: <F1451E5C-CE20-49C9-878D-B3B4ABD3BD78@jerris.com>

parse error, line too long

On Feb 11, 2010, at 2:30 AM, teldev wrote:

> Hello Mike, "Can you post for all to see some idea of how these applications work, lanagages used, some samples so we can see if we can get some interest in this?" "I am not sure the phone gui has much to do with FreeSWITCH at all, other than pulling a little data from the databases (again, ODBC highly recommended)" --> The idea would be to create a phone GUI on Aastra 6739i that would enable the touchscreen to do something like what is shown in these demos: http://www.youtube.com/watch?v=xZDHibW1gUs http://www.youtube.com/watch?v=Y58TPNCOTZI Perform FreeSWITCH actions: answer calls, place calls, setup conferences/add participants, etc. Display FreeSWITCH states: voice mail waiting count, extensions that are online/offline, call log, etc. Introduce new actions: dial By name directory with pictures, visual setup of conferences, etc. Since the Aastra 6739i, SNOM 870, and some others have built-in XML browsers it seems like they could be programmed to interact well with FreeSWITCH and various third party systems such as online feeds like weather reports, news, stock prices. Aastra is taking the IPhone approach of letting developers create the apps. They even have a similar slogan "There's an App for That" versus Apple's "There's an app for just about anything". http://www.ucstrategies.com/unified-communications-expert-views/aastra-phone-theres-an-app-for-that.aspx I don't know to what extent FreeSWITCH interacts with this phone out of the box, it reportedly works well with Asterisk though. I'm sure FS will likely understand certain button presses like hold, mute, speaker, transfer, etc. I will be ordering one within a few days. A few quesions for the FS community: Does this project have merit and fulfill a need? (I have yet to use FS as a PBX) Can anyone shed some light on what features on the phone will work and what I've mentioned will require custom development? Any ideas on how to architect an integration that would leverage what FS already does? Currently our IVR work uses Javascript/SpiderMonkey to call our own web service for database reads/writes. For this project, we'd need to interact more directly with FS and learn more about it's internals. teldev 
> View this message in context: Re: Seeking Advice on SIP Phones like Aastra
> Sent from the freeswitch-users mailing list archive at Nabble.com.
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/d19684fb/attachment-0002.html 

From mike at jerris.com  Thu Feb 11 00:43:18 2010
From: mike at jerris.com (Michael Jerris)
Date: Thu, 11 Feb 2010 03:43:18 -0500
Subject: [Freeswitch-users] Seeking Advice on SIP Phones like Aastra
In-Reply-To: <1265875307968-4553322.post@n2.nabble.com>
References: <9DFA265F-8632-466B-8910-1E2F9D956908@jerris.com>
	<1265875307968-4553322.post@n2.nabble.com>
Message-ID: <5AFFA858-A491-467E-9AEA-F9F9BBEAD2AD@jerris.com>


On Feb 11, 2010, at 3:01 AM, teldev wrote:

> 
> Hello Mike,
> 
> Last post lost all formatting when I used Nabble, this is readable now.
> 
> "Can you post for all to see some idea of how these applications work,
> lanagages used, some samples so we can see if we can get some interest in
> this?"
> 
> "I am not sure the phone gui has much to do with FreeSWITCH at all, other
> than pulling a little data from the databases (again, ODBC highly
> recommended)" 
> 
> -->  The idea would be to create a phone GUI on Aastra 6739i that would
> enable the touchscreen to do something like what is shown in these demos:
> 
> http://www.youtube.com/watch?v=xZDHibW1gUs
> http://www.youtube.com/watch?v=Y58TPNCOTZI
> 
> Perform FreeSWITCH actions: answer calls, place calls, setup conferences/add
> participants, etc.
> Display FreeSWITCH states: voice mail waiting count, extensions that are
> online/offline, call log, etc. 
> Introduce new actions: dial By name directory with pictures, visual setup of
> conferences, etc.  
> 
> Since the Aastra 6739i, SNOM 870, and some others have built-in XML browsers
> it seems like they could be programmed to interact well with FreeSWITCH and
> various third party systems such as online feeds like weather reports, news,
> stock prices.  Aastra is taking the IPhone approach of letting developers
> create the apps.  They even have a similar slogan "There's an App for That"
> versus Apple's "There's an app for just about anything". 

Slogan ripoff FAIL

> 
> http://www.ucstrategies.com/unified-communications-expert-views/aastra-phone-theres-an-app-for-that.aspx
> 
> I don't know to what extent FreeSWITCH interacts with this phone out of the
> box, it reportedly works well with Asterisk though.  I'm sure FS will likely
> understand certain button presses like hold, mute, speaker, transfer, etc. 
> I will be ordering one within a few days. 
> 
> A few quesions for the FS community:
> 
> Does this project have merit and fulfill a need?  (I have yet to use FS as a
> PBX)
> 

Sure.

> Can anyone shed some light on what features on the phone will work and what
> I've mentioned will require custom development?

Thats a hard question, short answer, all the normal sip stuff, and a bunch of the abnormal stuff too.

> Any ideas on how to architect an integration that would leverage what FS
> already does?  Currently our IVR work uses Javascript/SpiderMonkey to call
> our own web service for database reads/writes.  For this project, we'd need
> to interact more directly with FS and learn more about it's internals.

I am not sure this is totally true, its probably pretty trivial database work for everything you need.

Mike







From mike at jerris.com  Thu Feb 11 00:45:37 2010
From: mike at jerris.com (Michael Jerris)
Date: Thu, 11 Feb 2010 03:45:37 -0500
Subject: [Freeswitch-users] can't see sip messages in the log file
In-Reply-To: <20100211080818.D469D1210F@mail.nstel.ru>
References: <20100211080818.D469D1210F@mail.nstel.ru>
Message-ID: <8A8A10A3-19CE-4C41-8EF2-D0630B9AE7B4@jerris.com>


Revision
16599
Author
rupa
Date
2010-02-10 08:49:32 -0600 (Wed, 10 Feb 2010)
Log Message

document tracelevel, add completion for tracelevel
Modified Paths

freeswitch/trunk/src/mod/endpoints/mod_sofia/mod_sofia.c

On Feb 11, 2010, at 3:08 AM, Nikolay Kondratyev wrote:

> What is the version number where ?sofia help? will show tracelevel?
> Thanks and regards,
> Nikolay.
>  
> Thanks.  I've added that to the sofia help/completion.
> 
> On Wed, Feb 10, 2010 at 8:25 AM, Nikolay Kondratyev <kond at nstel.ru> wrote:
> After some experiments i clarified this question.
> SIP messages go into the freeswitch log when:
> <param name="sip-trace" value="yes"/> ("sofia profile profile-name siptrace
> on" is cli equivalent for this parameter)
> AND
> Sofia tracelevel is set to info (sofia tracelevel info).
> 
> The default sofia tracelevel is 'console'. That's why I did not see sip
> messages in the log after turning on just "siptrace".
> 
> Thanks and regards,
> Nikolay.
> 
> > -----Original Message-----
> > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-
> > users-bounces at lists.freeswitch.org] On Behalf Of Nikolay Kondratyev
> > Sent: Wednesday, February 10, 2010 2:00 PM
> > To: freeswitch-users at lists.freeswitch.org
> > Subject: Re: [Freeswitch-users] can't see sip messages in the log file
> >
> > Jason, thanks for the reply.
> > Isn't "sofia profile internal siptrace on" a command line equivalent of
> > <param name="sip-trace" value="yes"/>?
> >
> > Any way I tried it, but with the same result.
> > I still don't see SIP.
> >
> > Thanks and regards,
> > Nikolay.
> >
> > > > Can anybody please advise how to include sip messages into the log
> > file?
> > >
> > > <param name="sip-trace" value="yes"/>
> > > in the SIP profile you want to trace, then
> > > sofia profile profile-name restart reloadxml
> > > or restarting FreeSWITCH should do it.
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> -- 
> -Rupa
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/8c89cd95/attachment-0002.html 

From nazim.agabekov at gmail.com  Thu Feb 11 00:56:32 2010
From: nazim.agabekov at gmail.com (Nazim Agabekov)
Date: Thu, 11 Feb 2010 12:56:32 +0400
Subject: [Freeswitch-users] Lua script hangup detect
In-Reply-To: <D8FFF1FB-7B2D-49BC-940C-445EBF838BE3@jerris.com>
References: <314dc3f81002100117h6b1bc49cna989624a1ae4e0e2@mail.gmail.com>	<4B729155.7010708@gmail.com>	<314dc3f81002100800o41fd6561w394d5e83e69d3d80@mail.gmail.com>	<4B72DFF9.4040402@gmail.com>	<1b46b4e81002100932x7af486ekd4b05f4750cace20@mail.gmail.com>	<4B72F2A8.4070503@gmail.com>	<4CA5432B-5F6E-438E-B6EF-3277847E7922@jerris.com>	<4B73B57E.9040901@gmail.com>
	<D8FFF1FB-7B2D-49BC-940C-445EBF838BE3@jerris.com>
Message-ID: <4B73C640.20700@gmail.com>

I'm using luasql with ODBC MySQL driver in production. I've never tried
to use luasql with "native" mysql driver, but ODBC one works great.

On 02/11/2010 12:36 PM, Michael Jerris wrote:
> luasql, now with free memory leaks, while supplies last (don't use mysql with lua? or really ever)
>
> Mike
>
> On Feb 11, 2010, at 2:45 AM, Nazim Agabekov wrote:
>
>   
>> Thanks Mike! Mod_Lua is great. I had experience with a lot of proprietary IVR systems in the past.
>> FreeSWITCH and Mod_lua + luasql beats them all. Functionality is really impressive.
>>
>> On 02/11/2010 09:21 AM, Michael Jerris wrote:
>>     
>>> lua runs until you finish your script.  You need to exit when you are supposed to.  We have session::ready() for you to test this.
>>>
>>> Mike
>>>
>>> On Feb 10, 2010, at 12:53 PM, Nazim Agabekov wrote:
>>>
>>>       
>>>> I think it continues to run until it finishes. I'll check it tomorrow on my test system.
>>>>
>>>>         
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   




From max.bridgewater at gmail.com  Thu Feb 11 01:03:59 2010
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Thu, 11 Feb 2010 01:03:59 -0800
Subject: [Freeswitch-users] Freeswitch and G729
Message-ID: <c55d46951002110103y458b4b15r33df3e403d5eeef2@mail.gmail.com>

Hi,

Some quick questions related to the upcomming Freeswitch G729 support:

1) When can it be tried?
2) Does it support lower bit rate extensions such as D, F, H, I, C?

Thanks,
Max.



From jingwei.yang at gmail.com  Thu Feb 11 01:30:44 2010
From: jingwei.yang at gmail.com (Jingwei Yang)
Date: Thu, 11 Feb 2010 17:30:44 +0800
Subject: [Freeswitch-users] Is it possible to specify music on hold with
	uuid_hold?
Message-ID: <13529f9d1002110130v335d8177m74f791577b558ca@mail.gmail.com>

Hello,

Is it possible to specify a customized music file when the caller is put on
hold by uuid_hold?

Maybe something like this? uuid_hold {music_on_hold=/tmp/moh.wav}
<caller_uuid>

Regards,
-Jingwei
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/cf729028/attachment-0002.html 

From mcampbellsmith at gmail.com  Thu Feb 11 01:31:08 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Thu, 11 Feb 2010 20:31:08 +1100
Subject: [Freeswitch-users] Forbidden using UDP, works with TCP/TLS
In-Reply-To: <2EF27EFC-1634-46A3-B438-2EE99BEF58F1@jerris.com>
References: <33c87fa31002101824h21e3bf7lb13b68e8d4fd36a2@mail.gmail.com>
	<2EF27EFC-1634-46A3-B438-2EE99BEF58F1@jerris.com>
Message-ID: <33c87fa31002110131w3cf6ad00n802669856085a879@mail.gmail.com>

I don't see debug logs (fsctl debug_level 9 or loglevel 9) during the
registration process.

All I see is the sip messages when the sip trace is activated (403 Forbidden)

Is there other debugging that I can enable?

On Thu, Feb 11, 2010 at 4:25 PM, Michael Jerris <mike at jerris.com> wrote:
> Look at the FreeSWITCH debug logs and compare the differences, they should tell you why.
>
> Mike
>
> On Feb 10, 2010, at 9:24 PM, Mark Campbell-Smith wrote:
>
>> Hi!
>>
>> I had a user registered using TLS transport. That was working fine but
>> I want to change the ATA over to use UDP instead.
>>
>> All I thought I should have to do was to change the transport and
>> ports used to register in the ATA (SPA3102). ?However, when I do this,
>> FS responds with Forbidden.
>>
>> When I change the settings back to use TCP or TLS, registration is successful.
>>
>> What would cause FS to respond with forbidden? ?I do not change the
>> username/password fields in either case.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From mike at jerris.com  Thu Feb 11 01:57:55 2010
From: mike at jerris.com (Michael Jerris)
Date: Thu, 11 Feb 2010 04:57:55 -0500
Subject: [Freeswitch-users] Forbidden using UDP, works with TCP/TLS
In-Reply-To: <33c87fa31002110131w3cf6ad00n802669856085a879@mail.gmail.com>
References: <33c87fa31002101824h21e3bf7lb13b68e8d4fd36a2@mail.gmail.com>
	<2EF27EFC-1634-46A3-B438-2EE99BEF58F1@jerris.com>
	<33c87fa31002110131w3cf6ad00n802669856085a879@mail.gmail.com>
Message-ID: <CA21231C-108B-4445-A3B9-BF8D94FFE898@jerris.com>

you can crank up the sofia loglevel as well

Mike

On Feb 11, 2010, at 4:31 AM, Mark Campbell-Smith wrote:

> I don't see debug logs (fsctl debug_level 9 or loglevel 9) during the
> registration process.
> 
> All I see is the sip messages when the sip trace is activated (403 Forbidden)
> 
> Is there other debugging that I can enable?
> 
> On Thu, Feb 11, 2010 at 4:25 PM, Michael Jerris <mike at jerris.com> wrote:
>> Look at the FreeSWITCH debug logs and compare the differences, they should tell you why.
>> 
>> Mike
>> 
>> On Feb 10, 2010, at 9:24 PM, Mark Campbell-Smith wrote:
>> 
>>> Hi!
>>> 
>>> I had a user registered using TLS transport. That was working fine but
>>> I want to change the ATA over to use UDP instead.
>>> 
>>> All I thought I should have to do was to change the transport and
>>> ports used to register in the ATA (SPA3102).  However, when I do this,
>>> FS responds with Forbidden.
>>> 
>>> When I change the settings back to use TCP or TLS, registration is successful.
>>> 
>>> What would cause FS to respond with forbidden?  I do not change the
>>> username/password fields in either case.
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From nagalenoj at gmail.com  Thu Feb 11 02:11:49 2010
From: nagalenoj at gmail.com (Nagalenoj H.)
Date: Thu, 11 Feb 2010 15:41:49 +0530
Subject: [Freeswitch-users] Play music to A leg.
In-Reply-To: <87f2f3b91002100702u7581589ua866d2492a008b23@mail.gmail.com>
References: <fe67a3341002090519r2cc6e8edm7702b8d012fc2be3@mail.gmail.com>
	<87f2f3b91002090737l4efa8en29bde9457fef7ea5@mail.gmail.com>
	<fe67a3341002092157x7484f3d2id6678510bbec66cc@mail.gmail.com>
	<87f2f3b91002100702u7581589ua866d2492a008b23@mail.gmail.com>
Message-ID: <fe67a3341002110211w60390e1cm2d12fadd1e802778@mail.gmail.com>

But My scenario is,
   After I get the call from X.
   I answer the call in some scenarios and won't answer the call. So, this
leg can either be answered or unanswered.
   I originate a call to another number.
   After getting some digits from this originated leg.
   I do uuid_bridge of these 2 legs.

I want to play some file[ringback] to leg A before bridging to B.

On Wed, Feb 10, 2010 at 8:32 PM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Tue, Feb 9, 2010 at 9:57 PM, Nagalenoj H. <nagalenoj at gmail.com> wrote:
>
>> Because, I want to get some digits before bridging the legs. I've tried
>> group_confirm_key, but it accepts only one digit, I need multiple digits, so
>> I can't use.
>> I've also tried group_confirm_file, but when I do originate for multiple
>> extensions, I want this script to work based on the answered extension.
>>
>> So, I've originated and processed the events to do my job.
>>
>> How do I play some music to A leg?
>>
>> I might be missing something, but couldn't you just park the call ("A
> leg") until you connect to the other party ("B leg") and then uuid_bridge at
> whatever point you want?
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Regards,
Nagalenoj H.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/04bf0402/attachment-0002.html 

From mcampbellsmith at gmail.com  Thu Feb 11 02:13:53 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Thu, 11 Feb 2010 21:13:53 +1100
Subject: [Freeswitch-users] Forbidden using UDP, works with TCP/TLS
In-Reply-To: <CA21231C-108B-4445-A3B9-BF8D94FFE898@jerris.com>
References: <33c87fa31002101824h21e3bf7lb13b68e8d4fd36a2@mail.gmail.com>
	<2EF27EFC-1634-46A3-B438-2EE99BEF58F1@jerris.com>
	<33c87fa31002110131w3cf6ad00n802669856085a879@mail.gmail.com>
	<CA21231C-108B-4445-A3B9-BF8D94FFE898@jerris.com>
Message-ID: <33c87fa31002110213q1de0eebfsb950d8725ea9aba@mail.gmail.com>

ah thats true... The trace is not too readable to me, but may give
some insight to someone that can read the sofia logs....


recv 752 bytes from udp/[121.xxx.xxx.xxx]:5060 at 10:10:37.803288:
   ------------------------------------------------------------------------
   REGISTER sip:mydns.dyndns.org SIP/2.0
   Via: SIP/2.0/UDP 121.xxx.xxx.xxx:5060;branch=z9hG4bK-9052c91f
   From: 2000 <sip:2000 at mydns.dyndns.org>;tag=7a9dbbbfa691136do0
   To: 2000 <sip:2000 at mydns.dyndns.org>
   Call-ID: 610db38-dd3b511f at 121.xxx.xxx.xxx
   CSeq: 32330 REGISTER
   Contact: 2000 <sip:2000 at 121.xxx.xxx.xxx:5060>;expires=900
   Authorization: Digest username="2000", realm="mydns.dyndns.org",
nonce="b3298cfe-16f5-11df-9734-e77ba7a22ac3",
uri="sip:mydns.dyndns.org",
response="724fca542ce08d3f12b9ba1043bebb0c", algorithm=MD5,
qop="1225e2f1"
   Max-Forwards: 70
   User-Agent: Linksys/SPA3102-5.1.10(GW)
   Supported: x-sipura
   Supported: replaces
   Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
   Content-Length: 0


   ------------------------------------------------------------------------
tport_deliver(0x97cde80): msg 0x98297e8 (752 bytes) from
udp/121.xxx.xxx.xxx:5060/sip next=(nil)
nta: received REGISTER sip:mydns.dyndns.org SIP/2.0 (CSeq 32330)
nta: REGISTER (32330) going to a default leg
nua: nua_stack_process_request: entering
nua: nh_create: entering
nua: nh_create_handle: entering
nua: nua_stack_set_params: entering
soa_clone(static::0x97cc698, 0x9794808, 0x981cb70) called
soa_set_params(static::0x9758ba8, ...) called
nua(0x981cb70): event i_register 100 Trying
nua: nua_application_event: entering
nua: nua_respond: entering
nua(0x981cb70): sent signal r_respond
nua: nua_handle_destroy: entering
nua(0x981cb70): sent signal r_destroy
nua: nua_handle_magic: entering
nua: nua_handle_destroy: entering
nua(0x981cb70): recv signal r_respond 403 Forbidden
nua: nua_stack_set_params: entering
soa_set_params(static::0x9758ba8, ...) called
tport_tsend(0x97cde80) tpn = UDP/121.xxx.xxx.xxx:5060
tport_resolve addrinfo = 121.xxx.xxx.xxx:5060
tport_by_addrinfo(0x97cde80): not found by name UDP/121.xxx.xxx.xxx:5060
tport_vsend(0x97cde80): 495 bytes of 495 to udp/121.xxx.xxx.xxx:5060
tport_vsend returned 495
send 495 bytes to udp/[121.xxx.xxx.xxx]:5060 at 10:10:37.812955:
   ------------------------------------------------------------------------
   SIP/2.0 403 Forbidden

On Thu, Feb 11, 2010 at 8:57 PM, Michael Jerris <mike at jerris.com> wrote:
> you can crank up the sofia loglevel as well
>
> Mike
>
> On Feb 11, 2010, at 4:31 AM, Mark Campbell-Smith wrote:
>
>> I don't see debug logs (fsctl debug_level 9 or loglevel 9) during the
>> registration process.
>>
>> All I see is the sip messages when the sip trace is activated (403 Forbidden)
>>
>> Is there other debugging that I can enable?
>>
>> On Thu, Feb 11, 2010 at 4:25 PM, Michael Jerris <mike at jerris.com> wrote:
>>> Look at the FreeSWITCH debug logs and compare the differences, they should tell you why.
>>>
>>> Mike
>>>
>>> On Feb 10, 2010, at 9:24 PM, Mark Campbell-Smith wrote:
>>>
>>>> Hi!
>>>>
>>>> I had a user registered using TLS transport. That was working fine but
>>>> I want to change the ATA over to use UDP instead.
>>>>
>>>> All I thought I should have to do was to change the transport and
>>>> ports used to register in the ATA (SPA3102). ?However, when I do this,
>>>> FS responds with Forbidden.
>>>>
>>>> When I change the settings back to use TCP or TLS, registration is successful.
>>>>
>>>> What would cause FS to respond with forbidden? ?I do not change the
>>>> username/password fields in either case.
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From jingwei.yang at gmail.com  Thu Feb 11 02:33:10 2010
From: jingwei.yang at gmail.com (Jingwei Yang)
Date: Thu, 11 Feb 2010 18:33:10 +0800
Subject: [Freeswitch-users]  Is it possible to repeat music in playback
Message-ID: <13529f9d1002110233g57c8a1a5j3cdd38e22fa6a13d@mail.gmail.com>

Hello,

I've defined a very simple dialplan like the one below and when the caller
is connected to this plan, I hope to keep the call alive and repeat the
music set by playback. How am I able to achieve this?


    <extension name="agent_hold">
      <condition field="destination_number" expression="^33333$">
        <action application="answer"/>
        <action application="playback" data="customized/test.wav"/>
      </condition>
    </extension>


Thanks,
-Jingwei
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/d911bda1/attachment-0002.html 

From tculjaga at gmail.com  Thu Feb 11 03:34:22 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Thu, 11 Feb 2010 12:34:22 +0100
Subject: [Freeswitch-users] h323 - sip call is not working
In-Reply-To: <20100211081817.BAB611200B@mail.nstel.ru>
References: <549CFEF87AEDE841A38E9D15EAB4C04C5576CE4923@cooper>
	<20100211081817.BAB611200B@mail.nstel.ru>
Message-ID: <65d96fc81002110334o1ec01bddp521528cab618acfc@mail.gmail.com>

Nikolay, you are sending slow start with tunneling=true ?!?!

It is not gong to work :)

Please can you set fast start instead?


Your call failed because there was no mediaControll channel negotiated at
all... actually the call had to be aborted because wrong signaling .. but
anyhow.

Please on your IPO use FastStart with h245Tunneling=true... also, same
settings on FS side as well (exclude h245 in setup as well).




Frame 13 (277 bytes on wire, 277 bytes captured)
Ethernet II, Src: AlliedTe_22:9b:4a (00:00:cd:22:9b:4a), Dst:
Vmware_67:33:a7 (00:0c:29:67:33:a7)
Internet Protocol, Src: 172.23.14.2 (172.23.14.2), Dst: 172.23.22.49
(172.23.22.49)
Transmission Control Protocol, Src Port: oirtgsvc (4141), Dst Port:
h323hostcall (1720), Seq: 1, Ack: 1, Len: 223
TPKT, Version: 3, Length: 223
Q.931
    Protocol discriminator: Q.931
    Call reference value length: 2
    Call reference flag: Message sent from originating side
    Call reference value: 0012
    Message type: SETUP (0x05)
    Bearer capability
    Display  'Gornak Alexandr>2853'
    Calling party number: '5840'
    Called party number: '2853'
    User-user
H.225.0 CS
    H323-UserInformation
        h323-uu-pdu
            h323-message-body: setup (0)
                setup
            h4501SupplementaryService: 1 item
           * 1... .... h245Tunneling: True*







On Thu, Feb 11, 2010 at 9:18 AM, Nikolay Kondratyev <kond at nstel.ru> wrote:

> > But I do recommenf to use SIP, since it's much better supported by FS.
> >
> > /Peter
> But SIP is poorly supported by IPO.
> Thanks and regards,
> Nikolay.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/03009b45/attachment-0002.html 

From dist.lists at gmail.com  Thu Feb 11 04:08:19 2010
From: dist.lists at gmail.com (Hristo Trendev)
Date: Thu, 11 Feb 2010 14:08:19 +0200
Subject: [Freeswitch-users] Skypiax latency
In-Reply-To: <4B72FE87.4000401@gmx.net>
References: <4B72FE87.4000401@gmx.net>
Message-ID: <2a73afe1002110408r7f72cabdwf6a02ba27a314065@mail.gmail.com>

Hi Peter,
Take a look at http://jira.freeswitch.org/browse/MODSKYPIAX-29 and
http://jira.freeswitch.org/browse/MODLANG-130. I haven't tested the
trick in MODLANG-130 with skypiax yet, but if you use script to handle
the calls this may help.

BR,
Hristo


On 2/10/10, Peter P GMX <Prometheus001 at gmx.net> wrote:
> Hello,
>
> I have a problem with latency and mod_skypiax
>
> Skype=>SIP is always fine (~0.3sec)
> SIP => Skype is always bad (~2-4 sec)
> I would expect that latency in both directions should be the same.
>
> Anybody has discovered this before and has a solution?
>
> The scenario is as follows:
>
> SIP-UA => FreeswitchServer=>Freeswitch/SkypIaxServer=>Skype
> Both freeswitch servers are in the same LAN, so latency should be low.
>
> Best regards
> Peter
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From paul at apcl.us  Thu Feb 11 05:10:53 2010
From: paul at apcl.us (Paul Levin)
Date: Thu, 11 Feb 2010 08:10:53 -0500
Subject: [Freeswitch-users] how to set From address for voicemail emails?
In-Reply-To: <7471DDD5-0A73-4F18-AF90-BA8534EBC9E7@jerris.com>
References: <4B736846.1040908@apcl.us>
	<7471DDD5-0A73-4F18-AF90-BA8534EBC9E7@jerris.com>
Message-ID: <4B7401DD.6050408@apcl.us>

An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/283fbabf/attachment-0002.html 

From brian at freeswitch.org  Thu Feb 11 05:22:03 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 11 Feb 2010 07:22:03 -0600
Subject: [Freeswitch-users] Freeswitch and G729
In-Reply-To: <c55d46951002110103y458b4b15r33df3e403d5eeef2@mail.gmail.com>
References: <c55d46951002110103y458b4b15r33df3e403d5eeef2@mail.gmail.com>
Message-ID: <0737B9F4-25A7-4044-BC50-97388844C739@freeswitch.org>


On Feb 11, 2010, at 3:03 AM, Max Bridgewater wrote:

> Hi,
> 
> Some quick questions related to the upcomming Freeswitch G729 support:
> 
> 1) When can it be tried?

SOON!

> 2) Does it support lower bit rate extensions such as D, F, H, I, C?

A and B.

> 
> Thanks,
> Max.




From brian at freeswitch.org  Thu Feb 11 05:22:31 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 11 Feb 2010 07:22:31 -0600
Subject: [Freeswitch-users] Is it possible to specify music on hold with
	uuid_hold?
In-Reply-To: <13529f9d1002110130v335d8177m74f791577b558ca@mail.gmail.com>
References: <13529f9d1002110130v335d8177m74f791577b558ca@mail.gmail.com>
Message-ID: <8FF07098-581B-47FB-A039-63A58041E5BF@freeswitch.org>

uuid_hold isn't for that.  It sends the hold indication to the far end... not the near end.

/b

On Feb 11, 2010, at 3:30 AM, Jingwei Yang wrote:

> Hello,
> 
> Is it possible to specify a customized music file when the caller is put on hold by uuid_hold?
> 
> Maybe something like this? uuid_hold {music_on_hold=/tmp/moh.wav} <caller_uuid>
> 
> Regards,
> -Jingwei




From brian at freeswitch.org  Thu Feb 11 05:26:21 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 11 Feb 2010 07:26:21 -0600
Subject: [Freeswitch-users] Is it possible to repeat music in playback
In-Reply-To: <13529f9d1002110233g57c8a1a5j3cdd38e22fa6a13d@mail.gmail.com>
References: <13529f9d1002110233g57c8a1a5j3cdd38e22fa6a13d@mail.gmail.com>
Message-ID: <2FBDB302-2B54-40CC-B694-332854219774@freeswitch.org>

Why not just use Fifo to hold them?  Or Park the agent and send the session a message to play music?  You then have options to define loop count.

http://wiki.freeswitch.org/wiki/Event_Socket#execute

/b

On Feb 11, 2010, at 4:33 AM, Jingwei Yang wrote:

> Hello,
> 
> I've defined a very simple dialplan like the one below and when the caller is connected to this plan, I hope to keep the call alive and repeat the music set by playback. How am I able to achieve this?
> 
> 
>     <extension name="agent_hold">
>       <condition field="destination_number" expression="^33333$">
>         <action application="answer"/>
>         <action application="playback" data="customized/test.wav"/>
>       </condition>
>     </extension>
> 
> 
> Thanks,
> -Jingwei




From brian at freeswitch.org  Thu Feb 11 05:26:59 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 11 Feb 2010 07:26:59 -0600
Subject: [Freeswitch-users] Lua script hangup detect
In-Reply-To: <4B73C640.20700@gmail.com>
References: <314dc3f81002100117h6b1bc49cna989624a1ae4e0e2@mail.gmail.com>	<4B729155.7010708@gmail.com>	<314dc3f81002100800o41fd6561w394d5e83e69d3d80@mail.gmail.com>	<4B72DFF9.4040402@gmail.com>	<1b46b4e81002100932x7af486ekd4b05f4750cace20@mail.gmail.com>	<4B72F2A8.4070503@gmail.com>	<4CA5432B-5F6E-438E-B6EF-3277847E7922@jerris.com>	<4B73B57E.9040901@gmail.com>
	<D8FFF1FB-7B2D-49BC-940C-445EBF838BE3@jerris.com>
	<4B73C640.20700@gmail.com>
Message-ID: <FFD1A126-5CC0-4D02-87CE-B3EBA553FC4A@freeswitch.org>

Yes ODBC doesn't seem to leak... while the native one leaks like a sieve.

/b

On Feb 11, 2010, at 2:56 AM, Nazim Agabekov wrote:

> I'm using luasql with ODBC MySQL driver in production. I've never tried
> to use luasql with "native" mysql driver, but ODBC one works great.




From rupa at rupa.com  Thu Feb 11 05:28:22 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Thu, 11 Feb 2010 07:28:22 -0600
Subject: [Freeswitch-users] Need help setting up a feature
In-Reply-To: <4B73A799020000E100000470@mail.fribert.dk>
References: <4B73A799020000E100000470@mail.fribert.dk>
Message-ID: <b39273531002110528p42218d07q95e97485139315f1@mail.gmail.com>

My Siemens A580 has options for controlling the R key.  It seems that you
can either have it setup for transfer or as a hook flash.  Default is as a
transfer key.

I haven't succeeded in getting it to work for transfer and it is wayyyyy
down low on my list of things to do with the phone.

>From the web UI:

 Call Transfer   Use the R key to initiate call transfer with the SIP Refer
method.: Yes    No  Transfer Call by On-Hook: Yes    No  Derive target
address: from SIP URL    from SIP contact header  Find target addr.
automatically: Yes    No
  Hook Flash (R-key)     R key settings are disabled because the R key is
being used for call transfer.

On Wed, Feb 10, 2010 at 11:45 PM, mailinglist <mailinglist at fribert.dk>wrote:

>  Sorry for the repost, but the previous thread just died :-)
>
> I'm trying to get the possibility of transfering an incoming call from one
> extension to another, and give the possibility of turning it into a
> conference.
> I don't have a 'transfer' button.
> I do have an 'R' button on the Siemens handsets, and a 'Flash' button on
> the Sipura. The 'Flash' button gives me a new dialtone, gives the caller
> MOH, and then I can dial the new extension, and transfer the call, but not
> create a conference.
> But the Siemens handset does not have a 'flash', and pressing the R doesn't
> do anything.
>
> It might be two different features 'transfer' and 'conference'...
>
> But I thought that using the bind_meta_app would accomplish both.
>
> It's on an incoming call from the outside.
> So the situation:
> The Public folder has an entry that matches the dialed number, and does a
> transfer to 8202.
> Then the dialplan matches the 8202 with a group, and the phone rings.
> Somebody picks it up, finds out that it needs to be transferred to another
> extension, or transferred to a conference with a second extension.
> How do I construct that?
>
>
>
> Best regards
> Fribse
>
> /Running Freeswitch on pfSense running on a VMWare ESXi 4.0
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/28955317/attachment-0002.html 

From moizchinoy at gmail.com  Thu Feb 11 05:32:55 2010
From: moizchinoy at gmail.com (Moiz Chinoy)
Date: Thu, 11 Feb 2010 17:32:55 +0400
Subject: [Freeswitch-users] Is it necessary to call hangup....
Message-ID: <29b888f81002110532o3136d376g655c4b7a6dcbf598@mail.gmail.com>

Hello,

Is it necessary to call hangup in dialplan after say playing a file
through playback application.

-- 
Regards,
Moiz Chinoy.



From brian at freeswitch.org  Thu Feb 11 05:36:53 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 11 Feb 2010 07:36:53 -0600
Subject: [Freeswitch-users] Is it necessary to call hangup....
In-Reply-To: <29b888f81002110532o3136d376g655c4b7a6dcbf598@mail.gmail.com>
References: <29b888f81002110532o3136d376g655c4b7a6dcbf598@mail.gmail.com>
Message-ID: <7F8C140A-01BD-45A9-B71A-D3036BE10B3E@freeswitch.org>

If the dialplan runs out of stuff to do it'll hang up on you anyway.

/b

On Feb 11, 2010, at 7:32 AM, Moiz Chinoy wrote:

> Hello,
> 
> Is it necessary to call hangup in dialplan after say playing a file
> through playback application.
> 
> -- 
> Regards,
> Moiz Chinoy.




From wessels147 at gmail.com  Thu Feb 11 05:40:31 2010
From: wessels147 at gmail.com (wessels)
Date: Thu, 11 Feb 2010 14:40:31 +0100
Subject: [Freeswitch-users] multiple isdn msn on an extension with a single
	sip account
Message-ID: <a58c5fe71002110540q4e8eabb9qd8c57fc765d69bca@mail.gmail.com>

Hi,

I'm evaluating the replacement of an ISDN pbx. One of the things I
can't figure out is a suitable replacement for the msn numbers on a
phone with a single sip account.
The current ISDN phones can be programmed to recognize up to eight
different msn numbers. The user of the phone can see and hear which
msn number was called.
So I've got a BRI or PRI , with multiple msn, feed that into
freeswitch and simple extensions like the linksys spa922. I want the
extension to ring on more than one msn but I want to be able to see
which msn was originally called in the display. Can this be setup
using freeswitch and a simple phone??? Any directions would be most
helpful.

Thank you,

Wessel s



From jingwei.yang at gmail.com  Thu Feb 11 05:54:42 2010
From: jingwei.yang at gmail.com (Jingwei Yang)
Date: Thu, 11 Feb 2010 21:54:42 +0800
Subject: [Freeswitch-users] Is it possible to specify music on hold with
	uuid_hold?
In-Reply-To: <8FF07098-581B-47FB-A039-63A58041E5BF@freeswitch.org>
References: <13529f9d1002110130v335d8177m74f791577b558ca@mail.gmail.com>
	<8FF07098-581B-47FB-A039-63A58041E5BF@freeswitch.org>
Message-ID: <13529f9d1002110554u65431499hc3fa877252023577@mail.gmail.com>

Sorry Brian, I don't quite understand your answer. What is the far end and
what is the near end? In my case, I bridge client A to agent B. While
they're talking, I use uuid_hold to put client A on hold. Then A hears the
default music. After a while, uuid_hold off A and the conversation between A
and B resumes. uuid_hold looks perfect for my situation except I'm not able
to change the default music.

Regards,
-Jingwei

On Thu, Feb 11, 2010 at 9:22 PM, Brian West <brian at freeswitch.org> wrote:

> uuid_hold isn't for that.  It sends the hold indication to the far end...
> not the near end.
>
> /b
>
> On Feb 11, 2010, at 3:30 AM, Jingwei Yang wrote:
>
> > Hello,
> >
> > Is it possible to specify a customized music file when the caller is put
> on hold by uuid_hold?
> >
> > Maybe something like this? uuid_hold {music_on_hold=/tmp/moh.wav}
> <caller_uuid>
> >
> > Regards,
> > -Jingwei
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/79efae51/attachment-0002.html 

From brian at freeswitch.org  Thu Feb 11 05:59:53 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 11 Feb 2010 07:59:53 -0600
Subject: [Freeswitch-users] Is it possible to specify music on hold with
	uuid_hold?
In-Reply-To: <13529f9d1002110554u65431499hc3fa877252023577@mail.gmail.com>
References: <13529f9d1002110130v335d8177m74f791577b558ca@mail.gmail.com>
	<8FF07098-581B-47FB-A039-63A58041E5BF@freeswitch.org>
	<13529f9d1002110554u65431499hc3fa877252023577@mail.gmail.com>
Message-ID: <E37048AD-5A46-42D5-B724-6C2344AD8A2B@freeswitch.org>

uuid_setvar the variable hold_music on the opposite UUID you're holding... uuid_hold isn't doing exactly what you think it is.  ;)

/b

On Feb 11, 2010, at 7:54 AM, Jingwei Yang wrote:

> Sorry Brian, I don't quite understand your answer. What is the far end and what is the near end? In my case, I bridge client A to agent B. While they're talking, I use uuid_hold to put client A on hold. Then A hears the default music. After a while, uuid_hold off A and the conversation between A and B resumes. uuid_hold looks perfect for my situation except I'm not able to change the default music.
> 
> Regards,
> -Jingwei




From anthony.minessale at gmail.com  Thu Feb 11 06:39:15 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 11 Feb 2010 08:39:15 -0600
Subject: [Freeswitch-users] Is it possible to repeat music in playback
In-Reply-To: <2FBDB302-2B54-40CC-B694-332854219774@freeswitch.org>
References: <13529f9d1002110233g57c8a1a5j3cdd38e22fa6a13d@mail.gmail.com>
	<2FBDB302-2B54-40CC-B694-332854219774@freeswitch.org>
Message-ID: <191c3a031002110639h3cf14efdvf7715b973fe3caed@mail.gmail.com>

or try endless_playback app


On Thu, Feb 11, 2010 at 7:26 AM, Brian West <brian at freeswitch.org> wrote:

> Why not just use Fifo to hold them?  Or Park the agent and send the session
> a message to play music?  You then have options to define loop count.
>
> http://wiki.freeswitch.org/wiki/Event_Socket#execute
>
> /b
>
> On Feb 11, 2010, at 4:33 AM, Jingwei Yang wrote:
>
> > Hello,
> >
> > I've defined a very simple dialplan like the one below and when the
> caller is connected to this plan, I hope to keep the call alive and repeat
> the music set by playback. How am I able to achieve this?
> >
> >
> >     <extension name="agent_hold">
> >       <condition field="destination_number" expression="^33333$">
> >         <action application="answer"/>
> >         <action application="playback" data="customized/test.wav"/>
> >       </condition>
> >     </extension>
> >
> >
> > Thanks,
> > -Jingwei
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/e7642b87/attachment-0002.html 

From kond at nstel.ru  Thu Feb 11 07:18:18 2010
From: kond at nstel.ru (Nikolay Kondratyev)
Date: Thu, 11 Feb 2010 18:18:18 +0300
Subject: [Freeswitch-users] h323 - sip call is not working
In-Reply-To: <65d96fc81002110334o1ec01bddp521528cab618acfc@mail.gmail.com>
Message-ID: <20100211151819.95FC711F60@mail.nstel.ru>

Tihomir,

Thanks for help.

I enabled fast start on IPO and I can hear voice now. But the ringback tone
and voice appears to be wheezy, but I will investigate that tomorrow.

Thanks again.

Nikolay.

 

 

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Tihomir
Culjaga
Sent: Thursday, February 11, 2010 2:34 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] h323 - sip call is not working

 

Nikolay, you are sending slow start with tunneling=true ?!?!

It is not gong to work :)

Please can you set fast start instead?


Your call failed because there was no mediaControll channel negotiated at
all... actually the call had to be aborted because wrong signaling .. but
anyhow.

Please on your IPO use FastStart with h245Tunneling=true... also, same
settings on FS side as well (exclude h245 in setup as well).




Frame 13 (277 bytes on wire, 277 bytes captured)
Ethernet II, Src: AlliedTe_22:9b:4a (00:00:cd:22:9b:4a), Dst:
Vmware_67:33:a7 (00:0c:29:67:33:a7)
Internet Protocol, Src: 172.23.14.2 (172.23.14.2), Dst: 172.23.22.49
(172.23.22.49)
Transmission Control Protocol, Src Port: oirtgsvc (4141), Dst Port:
h323hostcall (1720), Seq: 1, Ack: 1, Len: 223
TPKT, Version: 3, Length: 223
Q.931
    Protocol discriminator: Q.931
    Call reference value length: 2
    Call reference flag: Message sent from originating side
    Call reference value: 0012
    Message type: SETUP (0x05)
    Bearer capability
    Display  'Gornak Alexandr>2853'
    Calling party number: '5840'
    Called party number: '2853'
    User-user
H.225.0 CS
    H323-UserInformation
        h323-uu-pdu
            h323-message-body: setup (0)
                setup
            h4501SupplementaryService: 1 item
            1... .... h245Tunneling: True








On Thu, Feb 11, 2010 at 9:18 AM, Nikolay Kondratyev <kond at nstel.ru> wrote:

> But I do recommenf to use SIP, since it's much better supported by FS.
>
> /Peter

But SIP is poorly supported by IPO.
Thanks and regards,

Nikolay.


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org

http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/fc3f9420/attachment-0002.html 

From ederwander at gmail.com  Thu Feb 11 07:24:03 2010
From: ederwander at gmail.com (Eder Souza)
Date: Thu, 11 Feb 2010 13:24:03 -0200
Subject: [Freeswitch-users] call die after 30 seconds in lua script
Message-ID: <622bedea1002110724p4757a5a4r42a9da7330f10498@mail.gmail.com>

Hi list

im testing the script welcome.lua from
http://wiki.freeswitch.org/wiki/Lua_Welcome_IVR_Example

but when do a transfer my call drop in 30 seconds just in script lua!!

callig one ramal (ex: 12345) direct in x-lite the call works !!

When make a session:execute("transfer","12345") in the script lua the call
drop after 30 seconds!!


why ??


Att,


Eng Eder de Souza
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/1bd4d037/attachment-0002.html 

From msc at freeswitch.org  Thu Feb 11 07:54:03 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 11 Feb 2010 07:54:03 -0800
Subject: [Freeswitch-users] Play music to A leg.
In-Reply-To: <fe67a3341002110211w60390e1cm2d12fadd1e802778@mail.gmail.com>
References: <fe67a3341002090519r2cc6e8edm7702b8d012fc2be3@mail.gmail.com>
	<87f2f3b91002090737l4efa8en29bde9457fef7ea5@mail.gmail.com>
	<fe67a3341002092157x7484f3d2id6678510bbec66cc@mail.gmail.com>
	<87f2f3b91002100702u7581589ua866d2492a008b23@mail.gmail.com>
	<fe67a3341002110211w60390e1cm2d12fadd1e802778@mail.gmail.com>
Message-ID: <87f2f3b91002110754laf9d01dxa94a36bc1b8c83df@mail.gmail.com>

Hehe, this is getting more and more complicated. You may want to consider
using the event socket and have your call control be done from a more 3rd
party-ish perspective. If you've got all these different scenarios it might
be better to let an external script do all the work.

http://wiki.freeswitch.org/wiki/Event_Socket

-MC

On Thu, Feb 11, 2010 at 2:11 AM, Nagalenoj H. <nagalenoj at gmail.com> wrote:

> But My scenario is,
>    After I get the call from X.
>    I answer the call in some scenarios and won't answer the call. So, this
> leg can either be answered or unanswered.
>    I originate a call to another number.
>    After getting some digits from this originated leg.
>    I do uuid_bridge of these 2 legs.
>
> I want to play some file[ringback] to leg A before bridging to B.
>
> On Wed, Feb 10, 2010 at 8:32 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>>
>>
>> On Tue, Feb 9, 2010 at 9:57 PM, Nagalenoj H. <nagalenoj at gmail.com> wrote:
>>
>>> Because, I want to get some digits before bridging the legs. I've tried
>>> group_confirm_key, but it accepts only one digit, I need multiple digits, so
>>> I can't use.
>>> I've also tried group_confirm_file, but when I do originate for multiple
>>> extensions, I want this script to work based on the answered extension.
>>>
>>> So, I've originated and processed the events to do my job.
>>>
>>> How do I play some music to A leg?
>>>
>>> I might be missing something, but couldn't you just park the call ("A
>> leg") until you connect to the other party ("B leg") and then uuid_bridge at
>> whatever point you want?
>> -MC
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Regards,
> Nagalenoj H.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/06c6dac0/attachment-0002.html 

From msc at freeswitch.org  Thu Feb 11 08:03:24 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 11 Feb 2010 08:03:24 -0800
Subject: [Freeswitch-users] call die after 30 seconds in lua script
In-Reply-To: <622bedea1002110724p4757a5a4r42a9da7330f10498@mail.gmail.com>
References: <622bedea1002110724p4757a5a4r42a9da7330f10498@mail.gmail.com>
Message-ID: <87f2f3b91002110803u52b84246qfa529177a0e37e44@mail.gmail.com>

On Thu, Feb 11, 2010 at 7:24 AM, Eder Souza <ederwander at gmail.com> wrote:

> Hi list
>
> im testing the script welcome.lua from
> http://wiki.freeswitch.org/wiki/Lua_Welcome_IVR_Example
>
> but when do a transfer my call drop in 30 seconds just in script lua!!
>
> callig one ramal (ex: 12345) direct in x-lite the call works !!
>
> When make a session:execute("transfer","12345") in the script lua the call
> drop after 30 seconds!!
>
>
> why ??
>
>
What does the FS console debug log say right before the disconnect?
-MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/9c664943/attachment-0002.html 

From anthony.minessale at gmail.com  Thu Feb 11 08:04:34 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 11 Feb 2010 10:04:34 -0600
Subject: [Freeswitch-users] Play music to A leg.
In-Reply-To: <87f2f3b91002110754laf9d01dxa94a36bc1b8c83df@mail.gmail.com>
References: <fe67a3341002090519r2cc6e8edm7702b8d012fc2be3@mail.gmail.com>
	<87f2f3b91002090737l4efa8en29bde9457fef7ea5@mail.gmail.com>
	<fe67a3341002092157x7484f3d2id6678510bbec66cc@mail.gmail.com>
	<87f2f3b91002100702u7581589ua866d2492a008b23@mail.gmail.com>
	<fe67a3341002110211w60390e1cm2d12fadd1e802778@mail.gmail.com>
	<87f2f3b91002110754laf9d01dxa94a36bc1b8c83df@mail.gmail.com>
Message-ID: <191c3a031002110804m4c18e7e0y317f44272b42c8b4@mail.gmail.com>

group_confirm_key in execute mode can execute a lua script instead that can
read as many digits as you want and parse the results.


On Thu, Feb 11, 2010 at 9:54 AM, Michael Collins <msc at freeswitch.org> wrote:

> Hehe, this is getting more and more complicated. You may want to consider
> using the event socket and have your call control be done from a more 3rd
> party-ish perspective. If you've got all these different scenarios it might
> be better to let an external script do all the work.
>
> http://wiki.freeswitch.org/wiki/Event_Socket
>
> -MC
>
>
> On Thu, Feb 11, 2010 at 2:11 AM, Nagalenoj H. <nagalenoj at gmail.com> wrote:
>
>> But My scenario is,
>>    After I get the call from X.
>>    I answer the call in some scenarios and won't answer the call. So, this
>> leg can either be answered or unanswered.
>>    I originate a call to another number.
>>    After getting some digits from this originated leg.
>>    I do uuid_bridge of these 2 legs.
>>
>> I want to play some file[ringback] to leg A before bridging to B.
>>
>> On Wed, Feb 10, 2010 at 8:32 PM, Michael Collins <msc at freeswitch.org>wrote:
>>
>>>
>>>
>>> On Tue, Feb 9, 2010 at 9:57 PM, Nagalenoj H. <nagalenoj at gmail.com>wrote:
>>>
>>>> Because, I want to get some digits before bridging the legs. I've tried
>>>> group_confirm_key, but it accepts only one digit, I need multiple digits, so
>>>> I can't use.
>>>> I've also tried group_confirm_file, but when I do originate for multiple
>>>> extensions, I want this script to work based on the answered extension.
>>>>
>>>> So, I've originated and processed the events to do my job.
>>>>
>>>> How do I play some music to A leg?
>>>>
>>>> I might be missing something, but couldn't you just park the call ("A
>>> leg") until you connect to the other party ("B leg") and then uuid_bridge at
>>> whatever point you want?
>>> -MC
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Regards,
>> Nagalenoj H.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/64c2458f/attachment-0002.html 

From ederwander at gmail.com  Thu Feb 11 08:37:06 2010
From: ederwander at gmail.com (Eder Souza)
Date: Thu, 11 Feb 2010 14:37:06 -0200
Subject: [Freeswitch-users] call die after 30 seconds in lua script
In-Reply-To: <87f2f3b91002110803u52b84246qfa529177a0e37e44@mail.gmail.com>
References: <622bedea1002110724p4757a5a4r42a9da7330f10498@mail.gmail.com>
	<87f2f3b91002110803u52b84246qfa529177a0e37e44@mail.gmail.com>
Message-ID: <622bedea1002110837h43bf7ab2k756ba703663fe667@mail.gmail.com>

Hi Michael

resolved i think lol lol

2010-02-11 13:02:12.566609 [NOTICE] sofia.c:322 Hangup
sofia/internal/eder at ip [CS_EXECUTE] [NORMAL_CLEARING]
2010-02-11 13:02:12.566609 [DEBUG] switch_channel.c:1683 Send signal
sofia/internal/eder at ip [KILL]
2010-02-11 13:02:12.566609 [DEBUG] switch_core_session.c:932 Send signal
sofia/internal/eder at ip [BREAK]
2010-02-11 13:02:12.572884 [DEBUG] switch_ivr_play_say.c:1391 done playing
file
2010-02-11 13:02:12.572884 [DEBUG] switch_core_state_machine.c:490 (
sofia/internal/eder at ip) State EXECUTE going to sleep
2010-02-11 13:02:12.572884 [DEBUG] switch_core_state_machine.c:397 (
sofia/internal/eder at ip) Running State Change CS_HANGUP
2010-02-11 13:02:12.572884 [DEBUG] switch_core_state_machine.c:433 (
sofia/internal/eder at ip) State HANGUP
2010-02-11 13:02:12.572884 [DEBUG] mod_sofia.c:338 Channel
sofia/internal/eder at ip hanging up, cause: NORMAL_CLEARING
2010-02-11 13:02:12.572884 [DEBUG] switch_core_state_machine.c:46
sofia/internal/eder at ip Standard HANGUP, cause: NORMAL_CLEARING
2010-02-11 13:02:12.572884 [DEBUG] switch_core_state_machine.c:433 (
sofia/internal/eder at ip) State HANGUP going to sleep
2010-02-11 13:02:12.572884 [DEBUG] switch_core_state_machine.c:475 (
sofia/internal/eder at ip) State Change CS_HANGUP -> CS_REPORTING
2010-02-11 13:02:12.572884 [DEBUG] switch_core_session.c:932 Send signal
sofia/internal/eder at ip [BREAK]
2010-02-11 13:02:12.572884 [DEBUG] switch_core_state_machine.c:397 (
sofia/internal/eder at ip) Running State Change CS_REPORTING
2010-02-11 13:02:12.572884 [DEBUG] switch_core_state_machine.c:607 (
sofia/internal/eder at ip) State REPORTING
2010-02-11 13:02:12.572884 [DEBUG] switch_core_state_machine.c:53
sofia/internal/eder at ip Standard REPORTING, cause: NORMAL_CLEARING
2010-02-11 13:02:12.572884 [DEBUG] switch_core_state_machine.c:607 (
sofia/internal/eder at ip) State REPORTING going to sleep
2010-02-11 13:02:12.572884 [DEBUG] switch_core_state_machine.c:410 (
sofia/internal/eder at ip) State Change CS_REPORTING -> CS_DESTROY
2010-02-11 13:02:12.572884 [DEBUG] switch_core_session.c:1066 Session 55 (
sofia/internal/eder at ip) Locked, Waiting on external entities
2010-02-11 13:02:12.572884 [NOTICE] switch_core_session.c:1084 Session 55 (
sofia/internal/eder at ip) Ended
2010-02-11 13:02:12.572884 [NOTICE] switch_core_session.c:1086 Close Channel
sofia/internal/eder at ip [CS_DESTROY]
2010-02-11 13:02:12.572884 [DEBUG] switch_core_state_machine.c:559 (
sofia/internal/eder at ip) State DESTROY
2010-02-11 13:02:12.572884 [DEBUG] mod_sofia.c:255
sofia/internal/eder at ipSOFIA DESTROY
2010-02-11 13:02:12.572884 [DEBUG] switch_core_state_machine.c:60
sofia/internal/eder at ip Standard DESTROY
2010-02-11 13:02:12.572884 [DEBUG] switch_core_state_machine.c:559 (
sofia/internal/eder at ip) State DESTROY going to sleep

im note this error when i transfer my calls to hold_music (MOH) extension!!

when set "answer" the call drop after 30 seconds removing the flag "answer"
my transfer dont die see:

in my lua script i remove "session:answer();"
here

--session:answer(); --"""comented line work now""
session:setAutoHangup(false)
digito = "hua"
digito = session:playAndGetDigits(1, 1, 1, 10000, "#",
"/usr/local/freeswitch/sounds/en/us/callie/conference/8000/conf-pin.wav",
"", "\\d+");
        freeswitch.consoleLog("info", "Got dtmf: ".. digito .."\n");
     end
     if (digito == "5") then
         freeswitch.consoleLog("info", "tecla digitada: ".. digito .."\n");
         session:execute("transfer","9000");

.
.
.


in my /freeswitch/conf/dialplan/default.xml i remove "<action
application="answer"/>" see line for answer coment

<extension name="teste_eder">
      <condition field="destination_number" expression="^9000$">
          <!-- <action application="answer"/> """"comented line work
now"""-->
          <action application="playback" data="$${hold_music}"/>
      </condition>
    </extension>




Att,


Eng Eder de Souza

On Thu, Feb 11, 2010 at 2:03 PM, Michael Collins <msc at freeswitch.org> wrote:

>
>
>   On Thu, Feb 11, 2010 at 7:24 AM, Eder Souza <ederwander at gmail.com>wrote:
>
>> Hi list
>>
>> im testing the script welcome.lua from
>> http://wiki.freeswitch.org/wiki/Lua_Welcome_IVR_Example
>>
>> but when do a transfer my call drop in 30 seconds just in script lua!!
>>
>> callig one ramal (ex: 12345) direct in x-lite the call works !!
>>
>> When make a session:execute("transfer","12345") in the script lua the call
>> drop after 30 seconds!!
>>
>>
>> why ??
>>
>>
> What does the FS console debug log say right before the disconnect?
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/a0feec0c/attachment-0002.html 

From jmesquita at freeswitch.org  Thu Feb 11 08:49:58 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Thu, 11 Feb 2010 14:49:58 -0200
Subject: [Freeswitch-users] Is it necessary to call hangup....
In-Reply-To: <7F8C140A-01BD-45A9-B71A-D3036BE10B3E@freeswitch.org>
References: <29b888f81002110532o3136d376g655c4b7a6dcbf598@mail.gmail.com>
	<7F8C140A-01BD-45A9-B71A-D3036BE10B3E@freeswitch.org>
Message-ID: <ebbc2d551002110849r11f6e586p4e8d90274233ac8a@mail.gmail.com>

Just a side question that relates to your answer Brian. When we run out of
stuff to do on the dialplan, we return 404 (if the leg is SIP of course)?

JM


On Thu, Feb 11, 2010 at 11:36 AM, Brian West <brian at freeswitch.org> wrote:

> If the dialplan runs out of stuff to do it'll hang up on you anyway.
>
> /b
>
> On Feb 11, 2010, at 7:32 AM, Moiz Chinoy wrote:
>
> > Hello,
> >
> > Is it necessary to call hangup in dialplan after say playing a file
> > through playback application.
> >
> > --
> > Regards,
> > Moiz Chinoy.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/199886ae/attachment-0002.html 

From ivdreg at gmail.com  Thu Feb 11 08:51:36 2010
From: ivdreg at gmail.com (ivdreg ivdreg)
Date: Thu, 11 Feb 2010 18:51:36 +0200
Subject: [Freeswitch-users] Help on: park_timeout variable
Message-ID: <b437a0e1002110851v274b8cbah54d920a5e59a809b@mail.gmail.com>

Hi All,

After updating to current SVN from 1.0.4 I have a problem when caller party
hangs up a call. I have 3 seconds timeout before B leg disconnects. I think
that this is caused by  code in switch_ivr_bridge.c in function static
switch_status_t audio_bridge_on_exchange_media(switch_core_session_t
*session)
......

        if (switch_channel_get_state(channel) == CS_EXCHANGE_MEDIA) {
                switch_channel_set_variable(channel, "park_timeout", "3");
                switch_channel_set_state(channel, CS_PARK);
        }
......

This happens even if I set park_after_bridge=false variable.
Is anybody has this problem ?

Thanks
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/80155a37/attachment-0002.html 

From robert.hadley at teotech.com  Thu Feb 11 08:53:12 2010
From: robert.hadley at teotech.com (Robert Hadley)
Date: Thu, 11 Feb 2010 08:53:12 -0800
Subject: [Freeswitch-users] demo_ivr cannot find sound files viarelative
	paths
In-Reply-To: <E4BC77C3-8271-4F30-B6F8-6822E4747BC3@jerris.com>
References: <78030B9E01DD48F8B024A97035F96282@greyhawk.tonecommander.com><87f2f3b91002102057t7b039221kf3975553414880cf@mail.gmail.com>
	<E4BC77C3-8271-4F30-B6F8-6822E4747BC3@jerris.com>
Message-ID: <54726F0FE98C44398AFD34C3685CC47A@greyhawk.tonecommander.com>

That was it.  Thanks Mike.

 

This test build was trunk and default configs.  The problem is in the trunk
vars.xml this line has been removed.  I added this line back to vars.xml and
the demo_ivr started working again.

 

  <X-PRE-PROCESS cmd="set"
data="sound_prefix=$${base_dir}/sounds/en/us/callie"/>

 

The IVR source must still require sound_prefix to be set (and does not use
the new sounds_dir variable).

 

Thanks again,

Robert

 

  _____  

From: Michael Jerris [mailto:mike at jerris.com] 
Sent: Wednesday, February 10, 2010 9:34 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] demo_ivr cannot find sound files viarelative
paths

 

did you make any changes to the default configs?  what is in your vars.xml
related to sounds?  the relative paths were always relative to that dir, did
you make any changes to the sounds prefix ?

 

Mike

 

On Feb 10, 2010, at 11:57 PM, Michael Collins wrote:





If I read this log correctly it failed to find the "invalid entry" file but
it did find the phrases just fine. Can you confirm the presence of this
file:
/usr/local/freeswitch/sounds/en/us/callie/ivr/8000/ivr-that_was_an_invalid_e
ntry.wav

(It looks like this call is at 8kHz so that's where I'm assuming FS is
looking to find the sound file...)

-MC

On Wed, Feb 10, 2010 at 10:50 AM, Robert Hadley <robert.hadley at teotech.com>
wrote:



Hi,

 

It appears a recent change (possibly the new sounds_dir variable or the new
ivr_menu folder?) may have broken relative sound file paths in the IVR.  I
built a today's trunk version and installed to the default location. Using
the default conf files the demo_ivr cannot find files based on the relative
paths specified in ivr_menus/demo_ivr.xml.

 

[root at TEO-UCM-T2 conf]# cat ivr_menus/demo_ivr.xml

<include>

  <!-- demo IVR setup -->

  <!-- demo IVR, Main Menu -->

  <menu name="demo_ivr"

      greet-long="phrase:demo_ivr_main_menu"

      greet-short="phrase:demo_ivr_main_menu_short"

      invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"

      exit-sound="voicemail/vm-goodbye.wav"

 

 

2010-02-10 10:32:15.300337 [DEBUG] switch_ivr_menu.c:414 Executing IVR menu
demo_ivr

2010-02-10 10:32:15.300337 [DEBUG] switch_ivr_play_say.c:118 No language
specified - Using [en]

2010-02-10 10:32:15.300337 [DEBUG] switch_ivr_play_say.c:273 Handle
play-file:[ivr/ivr-welcome_to_freeswitch.wav] (en:en)

2010-02-10 10:32:15.300337 [DEBUG] switch_ivr_play_say.c:1158 Codec
Activated L16 at 8000hz 1 channels 20ms

2010-02-10 10:32:17.922147 [DEBUG] switch_ivr_play_say.c:1450 done playing
file

2010-02-10 10:32:18.042149 [DEBUG] switch_ivr_play_say.c:273 Handle
play-file:[ivr/ivr-this_ivr_will_let_you_test_features.wav] (en:en)

2010-02-10 10:32:18.042149 [DEBUG] switch_ivr_play_say.c:1158 Codec
Activated L16 at 8000hz 1 channels 20ms

2010-02-10 10:32:19.962158 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF 2:800

2010-02-10 10:32:19.962158 [DEBUG] switch_ivr_play_say.c:1450 done playing
file

2010-02-10 10:32:20.082156 [DEBUG] switch_ivr_menu.c:329 waiting for 3/4
digits t/o 2000

2010-02-10 10:32:20.120617 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF 2:400

2010-02-10 10:32:20.442158 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF 2:800

2010-02-10 10:32:20.682162 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF 2:800

2010-02-10 10:32:20.682162 [DEBUG] switch_ivr_menu.c:376 digits '2222'

2010-02-10 10:32:20.682162 [DEBUG] switch_ivr_menu.c:470 action regex [2222]
[/^(10[01][0-9])$/] [0]

2010-02-10 10:32:20.682162 [DEBUG] switch_ivr_menu.c:560 IVR menu 'demo_ivr'
caught invalid input '2222'

2010-02-10 10:32:20.682162 [ERR] mod_sndfile.c:194 Error Opening File
[/usr/local/freeswitch/sounds/ivr/ivr-that_was_an_invalid_entry.wav] [System
error : No such file or directory.]

2010-02-10 10:32:21.701505 [DEBUG] switch_ivr_play_say.c:118 No language
specified - Using [en]

2010-02-10 10:32:21.701505 [DEBUG] switch_ivr_play_say.c:273 Handle
play-file:[silence_stream://1000] (en:en)

2010-02-10 10:32:21.701505 [DEBUG] switch_ivr_play_say.c:1158 Codec
Activated L16 at 8000hz 1 channels 20ms

2010-02-10 10:32:22.700352 [DEBUG] switch_ivr_play_say.c:1450 done playing
file

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/b1616c63/attachment-0002.html 

From tculjaga at gmail.com  Thu Feb 11 10:13:50 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Thu, 11 Feb 2010 19:13:50 +0100
Subject: [Freeswitch-users] h323 - sip call is not working
In-Reply-To: <20100211151819.95FC711F60@mail.nstel.ru>
References: <65d96fc81002110334o1ec01bddp521528cab618acfc@mail.gmail.com>
	<20100211151819.95FC711F60@mail.nstel.ru>
Message-ID: <65d96fc81002111013r38d938b0t3ffc1a49b7ff5b92@mail.gmail.com>

On Thu, Feb 11, 2010 at 4:18 PM, Nikolay Kondratyev <kond at nstel.ru> wrote:

>  Tihomir,
>
> Thanks for help.
>
> I enabled fast start on IPO and I can hear voice now. But the ringback tone
> and voice appears to be wheezy, but I will investigate that tomorrow.
>

set framing time for your codec to 30ms in IPO, also play with  PI in
alerting... set it to 2.



> Thanks again.
>
> Nikolay.
>
>
>
>
>
>
>   ------------------------------
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Tihomir
> Culjaga
> *Sent:* Thursday, February 11, 2010 2:34 PM
>
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] h323 - sip call is not working
>
>
>
> Nikolay, you are sending slow start with tunneling=true ?!?!
>
>
> It is not gong to work :)
>
> Please can you set fast start instead?
>
>
> Your call failed because there was no mediaControll channel negotiated at
> all... actually the call had to be aborted because wrong signaling .. but
> anyhow.
>
> Please on your IPO use FastStart with h245Tunneling=true... also, same
> settings on FS side as well (exclude h245 in setup as well).
>
>
>
>
> Frame 13 (277 bytes on wire, 277 bytes captured)
> Ethernet II, Src: AlliedTe_22:9b:4a (00:00:cd:22:9b:4a), Dst:
> Vmware_67:33:a7 (00:0c:29:67:33:a7)
> Internet Protocol, Src: 172.23.14.2 (172.23.14.2), Dst: 172.23.22.49
> (172.23.22.49)
> Transmission Control Protocol, Src Port: oirtgsvc (4141), Dst Port:
> h323hostcall (1720), Seq: 1, Ack: 1, Len: 223
> TPKT, Version: 3, Length: 223
> Q.931
>     Protocol discriminator: Q.931
>     Call reference value length: 2
>     Call reference flag: Message sent from originating side
>     Call reference value: 0012
>     Message type: SETUP (0x05)
>     Bearer capability
>     Display  'Gornak Alexandr>2853'
>     Calling party number: '5840'
>     Called party number: '2853'
>     User-user
> H.225.0 CS
>     H323-UserInformation
>         h323-uu-pdu
>             h323-message-body: setup (0)
>                 setup
>             h4501SupplementaryService: 1 item
>            * 1... .... h245Tunneling: True*
>
>
>
>
>
>
>  On Thu, Feb 11, 2010 at 9:18 AM, Nikolay Kondratyev <kond at nstel.ru>
> wrote:
>
> > But I do recommenf to use SIP, since it's much better supported by FS.
> >
> > /Peter
>
> But SIP is poorly supported by IPO.
> Thanks and regards,
>
> Nikolay.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/bc426167/attachment-0002.html 

From paul at apcl.us  Thu Feb 11 10:43:09 2010
From: paul at apcl.us (Paul Levin)
Date: Thu, 11 Feb 2010 13:43:09 -0500
Subject: [Freeswitch-users] is there a command to get a list
 of	registered accounts?
In-Reply-To: <33c87fa31002102129r5d692dc5pfb414b80b47be221@mail.gmail.com>
References: <4B73677B.6020406@apcl.us>
	<33c87fa31002102129r5d692dc5pfb414b80b47be221@mail.gmail.com>
Message-ID: <4B744FBD.1090700@apcl.us>

An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/db3b0238/attachment-0002.html 

From troy at tlainvestments.com  Thu Feb 11 11:11:18 2010
From: troy at tlainvestments.com (Troy Anderson)
Date: Thu, 11 Feb 2010 12:11:18 -0700
Subject: [Freeswitch-users] Calls being parked on DTMF
Message-ID: <4663A209-CA71-4389-8ED9-C574CD939711@tlainvestments.com>

This is a strange one.  I make a call using an FXO analog line (mod_openzap).  During the call, I dial 200 and FS parks the call.  This is a Sangoma card using wanpipe.  Is there some kind of setting in there where it interprets DTMF?

Is there a way to see what OpenZAP is writing "ending bridge by request from write function"?

Thanks for any help on this!
-Troy

Here's where it happens (phone number is redacted).

2010-02-11 11:49:48.455727 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF 2:1680
2010-02-11 11:49:48.466015 [DEBUG] zap_io.c:1870 1:1 GENERATE DTMF [2]
2010-02-11 11:49:48.895733 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF 0:1280
2010-02-11 11:49:48.905726 [DEBUG] zap_io.c:1870 1:1 GENERATE DTMF [0]
2010-02-11 11:49:49.455734 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF 0:1360
2010-02-11 11:49:49.465771 [DEBUG] zap_io.c:1870 1:1 GENERATE DTMF [0]
2010-02-11 11:49:49.706363 [DEBUG] switch_ivr_bridge.c:469 OpenZAP/1:1/602xxxxxxx ending bridge by request from write function
2010-02-11 11:49:49.706363 [DEBUG] switch_core_session.c:638 Send signal sofia/internal/105 at 10.0.1.202 [BREAK]
2010-02-11 11:49:49.706363 [DEBUG] switch_ivr_bridge.c:562 BRIDGE THREAD DONE [sofia/internal/105 at 10.0.1.202]
2010-02-11 11:49:49.706363 [DEBUG] switch_ivr_bridge.c:582 Send signal OpenZAP/1:1/602xxxxxxx [BREAK]
2010-02-11 11:49:49.716359 [DEBUG] switch_core_session.c:638 Send signal OpenZAP/1:1/602xxxxxxx [BREAK]
2010-02-11 11:49:49.716359 [DEBUG] switch_ivr_bridge.c:562 BRIDGE THREAD DONE [OpenZAP/1:1/602xxxxxxx]
2010-02-11 11:49:49.716359 [DEBUG] switch_ivr_bridge.c:582 Send signal sofia/internal/105 at 10.0.1.202 [BREAK]
2010-02-11 11:49:49.716359 [DEBUG] switch_ivr_bridge.c:640 (OpenZAP/1:1/602xxxxxxx) State Change CS_EXCHANGE_MEDIA -> CS_PARK
2010-02-11 11:49:49.716359 [DEBUG] switch_core_session.c:1018 Send signal OpenZAP/1:1/602xxxxxxx [BREAK]
2010-02-11 11:49:49.716359 [DEBUG] switch_core_state_machine.c:351 (OpenZAP/1:1/602xxxxxxx) State EXCHANGE_MEDIA going to sleep
2010-02-11 11:49:49.716359 [DEBUG] switch_core_state_machine.c:314 (OpenZAP/1:1/602xxxxxxx) Running State Change CS_PARK
2010-02-11 11:49:49.716359 [DEBUG] switch_core_state_machine.c:357 (OpenZAP/1:1/602xxxxxxx) State PARK
2010-02-11 11:49:49.716359 [DEBUG] switch_core_state_machine.c:206 OpenZAP/1:1/602xxxxxxx Standard PARK






From anthony.minessale at gmail.com  Thu Feb 11 11:42:46 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 11 Feb 2010 13:42:46 -0600
Subject: [Freeswitch-users] Help on: park_timeout variable
In-Reply-To: <b437a0e1002110851v274b8cbah54d920a5e59a809b@mail.gmail.com>
References: <b437a0e1002110851v274b8cbah54d920a5e59a809b@mail.gmail.com>
Message-ID: <191c3a031002111142n625ca0d4n641378242cd69b87@mail.gmail.com>

This is what happens in a b leg, it only happens when you transfer a call.
This is by design to give the other phone a chance to kill the leg.  This is
not really a problem persae.


On Thu, Feb 11, 2010 at 10:51 AM, ivdreg ivdreg <ivdreg at gmail.com> wrote:

> Hi All,
>
> After updating to current SVN from 1.0.4 I have a problem when caller party
> hangs up a call. I have 3 seconds timeout before B leg disconnects. I think
> that this is caused by  code in switch_ivr_bridge.c in function static
> switch_status_t audio_bridge_on_exchange_media(switch_core_session_t
> *session)
> ......
>
>         if (switch_channel_get_state(channel) == CS_EXCHANGE_MEDIA) {
>                 switch_channel_set_variable(channel, "park_timeout", "3");
>                 switch_channel_set_state(channel, CS_PARK);
>         }
> ......
>
> This happens even if I set park_after_bridge=false variable.
> Is anybody has this problem ?
>
> Thanks
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/f3731dff/attachment-0002.html 

From anthony.minessale at gmail.com  Thu Feb 11 11:52:49 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 11 Feb 2010 13:52:49 -0600
Subject: [Freeswitch-users] Calls being parked on DTMF
In-Reply-To: <4663A209-CA71-4389-8ED9-C574CD939711@tlainvestments.com>
References: <4663A209-CA71-4389-8ED9-C574CD939711@tlainvestments.com>
Message-ID: <191c3a031002111152r1a56d46dl905d9c9c63e598d9@mail.gmail.com>

try lastest


On Thu, Feb 11, 2010 at 1:11 PM, Troy Anderson <troy at tlainvestments.com>wrote:

> This is a strange one.  I make a call using an FXO analog line
> (mod_openzap).  During the call, I dial 200 and FS parks the call.  This is
> a Sangoma card using wanpipe.  Is there some kind of setting in there where
> it interprets DTMF?
>
> Is there a way to see what OpenZAP is writing "ending bridge by request
> from write function"?
>
> Thanks for any help on this!
> -Troy
>
> Here's where it happens (phone number is redacted).
>
> 2010-02-11 11:49:48.455727 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF 2:1680
> 2010-02-11 11:49:48.466015 [DEBUG] zap_io.c:1870 1:1 GENERATE DTMF [2]
> 2010-02-11 11:49:48.895733 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF 0:1280
> 2010-02-11 11:49:48.905726 [DEBUG] zap_io.c:1870 1:1 GENERATE DTMF [0]
> 2010-02-11 11:49:49.455734 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF 0:1360
> 2010-02-11 11:49:49.465771 [DEBUG] zap_io.c:1870 1:1 GENERATE DTMF [0]
> 2010-02-11 11:49:49.706363 [DEBUG] switch_ivr_bridge.c:469
> OpenZAP/1:1/602xxxxxxx ending bridge by request from write function
> 2010-02-11 11:49:49.706363 [DEBUG] switch_core_session.c:638 Send signal
> sofia/internal/105 at 10.0.1.202 [BREAK]
> 2010-02-11 11:49:49.706363 [DEBUG] switch_ivr_bridge.c:562 BRIDGE THREAD
> DONE [sofia/internal/105 at 10.0.1.202]
> 2010-02-11 11:49:49.706363 [DEBUG] switch_ivr_bridge.c:582 Send signal
> OpenZAP/1:1/602xxxxxxx [BREAK]
> 2010-02-11 11:49:49.716359 [DEBUG] switch_core_session.c:638 Send signal
> OpenZAP/1:1/602xxxxxxx [BREAK]
> 2010-02-11 11:49:49.716359 [DEBUG] switch_ivr_bridge.c:562 BRIDGE THREAD
> DONE [OpenZAP/1:1/602xxxxxxx]
> 2010-02-11 11:49:49.716359 [DEBUG] switch_ivr_bridge.c:582 Send signal
> sofia/internal/105 at 10.0.1.202 [BREAK]
> 2010-02-11 11:49:49.716359 [DEBUG] switch_ivr_bridge.c:640
> (OpenZAP/1:1/602xxxxxxx) State Change CS_EXCHANGE_MEDIA -> CS_PARK
> 2010-02-11 11:49:49.716359 [DEBUG] switch_core_session.c:1018 Send signal
> OpenZAP/1:1/602xxxxxxx [BREAK]
> 2010-02-11 11:49:49.716359 [DEBUG] switch_core_state_machine.c:351
> (OpenZAP/1:1/602xxxxxxx) State EXCHANGE_MEDIA going to sleep
> 2010-02-11 11:49:49.716359 [DEBUG] switch_core_state_machine.c:314
> (OpenZAP/1:1/602xxxxxxx) Running State Change CS_PARK
> 2010-02-11 11:49:49.716359 [DEBUG] switch_core_state_machine.c:357
> (OpenZAP/1:1/602xxxxxxx) State PARK
> 2010-02-11 11:49:49.716359 [DEBUG] switch_core_state_machine.c:206
> OpenZAP/1:1/602xxxxxxx Standard PARK
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/ebafb4b4/attachment-0002.html 

From anthony.minessale at gmail.com  Thu Feb 11 11:53:06 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 11 Feb 2010 13:53:06 -0600
Subject: [Freeswitch-users] Calls being parked on DTMF
In-Reply-To: <191c3a031002111152r1a56d46dl905d9c9c63e598d9@mail.gmail.com>
References: <4663A209-CA71-4389-8ED9-C574CD939711@tlainvestments.com>
	<191c3a031002111152r1a56d46dl905d9c9c63e598d9@mail.gmail.com>
Message-ID: <191c3a031002111153h2f98f657n9fd4e71f47846710@mail.gmail.com>

err latest

On Thu, Feb 11, 2010 at 1:52 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> try lastest
>
>
>
> On Thu, Feb 11, 2010 at 1:11 PM, Troy Anderson <troy at tlainvestments.com>wrote:
>
>> This is a strange one.  I make a call using an FXO analog line
>> (mod_openzap).  During the call, I dial 200 and FS parks the call.  This is
>> a Sangoma card using wanpipe.  Is there some kind of setting in there where
>> it interprets DTMF?
>>
>> Is there a way to see what OpenZAP is writing "ending bridge by request
>> from write function"?
>>
>> Thanks for any help on this!
>> -Troy
>>
>> Here's where it happens (phone number is redacted).
>>
>> 2010-02-11 11:49:48.455727 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF 2:1680
>> 2010-02-11 11:49:48.466015 [DEBUG] zap_io.c:1870 1:1 GENERATE DTMF [2]
>> 2010-02-11 11:49:48.895733 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF 0:1280
>> 2010-02-11 11:49:48.905726 [DEBUG] zap_io.c:1870 1:1 GENERATE DTMF [0]
>> 2010-02-11 11:49:49.455734 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF 0:1360
>> 2010-02-11 11:49:49.465771 [DEBUG] zap_io.c:1870 1:1 GENERATE DTMF [0]
>> 2010-02-11 11:49:49.706363 [DEBUG] switch_ivr_bridge.c:469
>> OpenZAP/1:1/602xxxxxxx ending bridge by request from write function
>> 2010-02-11 11:49:49.706363 [DEBUG] switch_core_session.c:638 Send signal
>> sofia/internal/105 at 10.0.1.202 [BREAK]
>> 2010-02-11 11:49:49.706363 [DEBUG] switch_ivr_bridge.c:562 BRIDGE THREAD
>> DONE [sofia/internal/105 at 10.0.1.202]
>> 2010-02-11 11:49:49.706363 [DEBUG] switch_ivr_bridge.c:582 Send signal
>> OpenZAP/1:1/602xxxxxxx [BREAK]
>> 2010-02-11 11:49:49.716359 [DEBUG] switch_core_session.c:638 Send signal
>> OpenZAP/1:1/602xxxxxxx [BREAK]
>> 2010-02-11 11:49:49.716359 [DEBUG] switch_ivr_bridge.c:562 BRIDGE THREAD
>> DONE [OpenZAP/1:1/602xxxxxxx]
>> 2010-02-11 11:49:49.716359 [DEBUG] switch_ivr_bridge.c:582 Send signal
>> sofia/internal/105 at 10.0.1.202 [BREAK]
>> 2010-02-11 11:49:49.716359 [DEBUG] switch_ivr_bridge.c:640
>> (OpenZAP/1:1/602xxxxxxx) State Change CS_EXCHANGE_MEDIA -> CS_PARK
>> 2010-02-11 11:49:49.716359 [DEBUG] switch_core_session.c:1018 Send signal
>> OpenZAP/1:1/602xxxxxxx [BREAK]
>> 2010-02-11 11:49:49.716359 [DEBUG] switch_core_state_machine.c:351
>> (OpenZAP/1:1/602xxxxxxx) State EXCHANGE_MEDIA going to sleep
>> 2010-02-11 11:49:49.716359 [DEBUG] switch_core_state_machine.c:314
>> (OpenZAP/1:1/602xxxxxxx) Running State Change CS_PARK
>> 2010-02-11 11:49:49.716359 [DEBUG] switch_core_state_machine.c:357
>> (OpenZAP/1:1/602xxxxxxx) State PARK
>> 2010-02-11 11:49:49.716359 [DEBUG] switch_core_state_machine.c:206
>> OpenZAP/1:1/602xxxxxxx Standard PARK
>>
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/4e7f6f0a/attachment-0002.html 

From anthony.minessale at gmail.com  Thu Feb 11 11:53:42 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 11 Feb 2010 13:53:42 -0600
Subject: [Freeswitch-users] Help on: park_timeout variable
In-Reply-To: <191c3a031002111142n625ca0d4n641378242cd69b87@mail.gmail.com>
References: <b437a0e1002110851v274b8cbah54d920a5e59a809b@mail.gmail.com>
	<191c3a031002111142n625ca0d4n641378242cd69b87@mail.gmail.com>
Message-ID: <191c3a031002111153n655103b4ub46d67f9094d6f57@mail.gmail.com>

actually, i see a small buglet there, try trunk.


On Thu, Feb 11, 2010 at 1:42 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> This is what happens in a b leg, it only happens when you transfer a call.
> This is by design to give the other phone a chance to kill the leg.  This is
> not really a problem persae.
>
>
> On Thu, Feb 11, 2010 at 10:51 AM, ivdreg ivdreg <ivdreg at gmail.com> wrote:
>
>> Hi All,
>>
>> After updating to current SVN from 1.0.4 I have a problem when caller
>> party hangs up a call. I have 3 seconds timeout before B leg disconnects. I
>> think that this is caused by  code in switch_ivr_bridge.c in function static
>> switch_status_t audio_bridge_on_exchange_media(switch_core_session_t
>> *session)
>> ......
>>
>>         if (switch_channel_get_state(channel) == CS_EXCHANGE_MEDIA) {
>>                 switch_channel_set_variable(channel, "park_timeout", "3");
>>                 switch_channel_set_state(channel, CS_PARK);
>>         }
>> ......
>>
>> This happens even if I set park_after_bridge=false variable.
>> Is anybody has this problem ?
>>
>> Thanks
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/5dd3c584/attachment-0002.html 

From Prometheus001 at gmx.net  Thu Feb 11 11:53:34 2010
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Thu, 11 Feb 2010 20:53:34 +0100
Subject: [Freeswitch-users] Skypiax latency
In-Reply-To: <2a73afe1002110408r7f72cabdwf6a02ba27a314065@mail.gmail.com>
References: <4B72FE87.4000401@gmx.net>
	<2a73afe1002110408r7f72cabdwf6a02ba27a314065@mail.gmail.com>
Message-ID: <4B74603E.3060208@gmx.net>

Hello Hristo,

I am using xm-curl, so xml is processed. But your hints really helped me
to tie the problem down.

As described, my call scenario is as follows:
SIP-UA => FreeswitchServer=>Freeswitch/SkypIaxServer=>Skype

When I call from Skype to SIP and

    * let the SIP phone ringing for 1 sec, I receive a ~1,5 sec delay
      from SIP to Skype
    * let the SIP phone ringing for 2 sec, I receive a ~2,5 sec delay
      from SIP to Skype
    * let the SIP phone ringing for 3 sec, I receive a ~3,5 sec delay
      from SIP to Skype

So - the longer the SIP phone rings - the longer is the delay.

I also tested the other direction:
If I call from SIP to Skype the behaviour is exactly the same: The
longer the Skype phone rings - the longer is the delay from SIP to Skype.

This may be somehow related to MODSKYPIAX-29, but is a different
scenario. Should I open a new Jira for this or should I attach my
findings to MODSKYPIAX-29?

Best regards
Peter



Hristo Trendev schrieb:
> Hi Peter,
> Take a look at http://jira.freeswitch.org/browse/MODSKYPIAX-29 and
> http://jira.freeswitch.org/browse/MODLANG-130. I haven't tested the
> trick in MODLANG-130 with skypiax yet, but if you use script to handle
> the calls this may help.
>
> BR,
> Hristo
>
>
> On 2/10/10, Peter P GMX <Prometheus001 at gmx.net> wrote:
>   
>> Hello,
>>
>> I have a problem with latency and mod_skypiax
>>
>> Skype=>SIP is always fine (~0.3sec)
>> SIP => Skype is always bad (~2-4 sec)
>> I would expect that latency in both directions should be the same.
>>
>> Anybody has discovered this before and has a solution?
>>
>> The scenario is as follows:
>>
>> SIP-UA => FreeswitchServer=>Freeswitch/SkypIaxServer=>Skype
>> Both freeswitch servers are in the same LAN, so latency should be low.
>>
>> Best regards
>> Peter
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>     
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>   



From gmaruzz at celliax.org  Thu Feb 11 11:58:40 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Thu, 11 Feb 2010 20:58:40 +0100
Subject: [Freeswitch-users] Skypiax latency
In-Reply-To: <4B74603E.3060208@gmx.net>
References: <4B72FE87.4000401@gmx.net>
	<2a73afe1002110408r7f72cabdwf6a02ba27a314065@mail.gmail.com> 
	<4B74603E.3060208@gmx.net>
Message-ID: <7b197bef1002111158w6eeda53as77451e93ff1f6e04@mail.gmail.com>

pick that phone!

just jokin

open a new Jira

-giovanni

On Thu, Feb 11, 2010 at 8:53 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
> Hello Hristo,
>
> I am using xm-curl, so xml is processed. But your hints really helped me
> to tie the problem down.
>
> As described, my call scenario is as follows:
> SIP-UA => FreeswitchServer=>Freeswitch/SkypIaxServer=>Skype
>
> When I call from Skype to SIP and
>
> ? ?* let the SIP phone ringing for 1 sec, I receive a ~1,5 sec delay
> ? ? ?from SIP to Skype
> ? ?* let the SIP phone ringing for 2 sec, I receive a ~2,5 sec delay
> ? ? ?from SIP to Skype
> ? ?* let the SIP phone ringing for 3 sec, I receive a ~3,5 sec delay
> ? ? ?from SIP to Skype
>
> So - the longer the SIP phone rings - the longer is the delay.
>
> I also tested the other direction:
> If I call from SIP to Skype the behaviour is exactly the same: The
> longer the Skype phone rings - the longer is the delay from SIP to Skype.
>
> This may be somehow related to MODSKYPIAX-29, but is a different
> scenario. Should I open a new Jira for this or should I attach my
> findings to MODSKYPIAX-29?
>
> Best regards
> Peter
>
>
>
> Hristo Trendev schrieb:
>> Hi Peter,
>> Take a look at http://jira.freeswitch.org/browse/MODSKYPIAX-29 and
>> http://jira.freeswitch.org/browse/MODLANG-130. I haven't tested the
>> trick in MODLANG-130 with skypiax yet, but if you use script to handle
>> the calls this may help.
>>
>> BR,
>> Hristo
>>
>>
>> On 2/10/10, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>
>>> Hello,
>>>
>>> I have a problem with latency and mod_skypiax
>>>
>>> Skype=>SIP is always fine (~0.3sec)
>>> SIP => Skype is always bad (~2-4 sec)
>>> I would expect that latency in both directions should be the same.
>>>
>>> Anybody has discovered this before and has a solution?
>>>
>>> The scenario is as follows:
>>>
>>> SIP-UA => FreeswitchServer=>Freeswitch/SkypIaxServer=>Skype
>>> Both freeswitch servers are in the same LAN, so latency should be low.
>>>
>>> Best regards
>>> Peter
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618



From maxim.tsvetov at gmail.com  Thu Feb 11 12:09:09 2010
From: maxim.tsvetov at gmail.com (Maxim Tsvetov)
Date: Thu, 11 Feb 2010 23:09:09 +0300
Subject: [Freeswitch-users] DTMF problem
Message-ID: <89c9bbf81002111209v6e2f36b3x86a628c5e1390599@mail.gmail.com>

Hello everybody

Please help!
I'm trying to setup connection between Cisco 2811 and FS (Win2003) using
SIP.
Everything working correctly and I can make calls both ways. The only
problem is
when I'm calling from PSTN to FS over Cisco. It doesn't recognize DTMF.

I use G711 a-law (PCMA) codec and inband DTMF.

Regards,
Maxim
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/f37ad059/attachment-0002.html 

From brian at freeswitch.org  Thu Feb 11 12:22:35 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 11 Feb 2010 14:22:35 -0600
Subject: [Freeswitch-users] DTMF problem
In-Reply-To: <89c9bbf81002111209v6e2f36b3x86a628c5e1390599@mail.gmail.com>
References: <89c9bbf81002111209v6e2f36b3x86a628c5e1390599@mail.gmail.com>
Message-ID: <F960CC45-C9D2-4150-A108-C0E8B230284D@freeswitch.org>

Add 'dtmf-relay rtp-nte' and 'dtmf-interworkingrtp-nte' to your voice peer.

/b

On Feb 11, 2010, at 2:09 PM, Maxim Tsvetov wrote:

> Hello everybody
> 
> Please help!
> I'm trying to setup connection between Cisco 2811 and FS (Win2003) using SIP.
> Everything working correctly and I can make calls both ways. The only problem is
> when I'm calling from PSTN to FS over Cisco. It doesn't recognize DTMF.
> 
> I use G711 a-law (PCMA) codec and inband DTMF.
> 
> Regards,
> Maxim
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From troy at tlainvestments.com  Thu Feb 11 12:35:09 2010
From: troy at tlainvestments.com (Troy Anderson)
Date: Thu, 11 Feb 2010 13:35:09 -0700
Subject: [Freeswitch-users] Calls being parked on DTMF
In-Reply-To: <191c3a031002111153h2f98f657n9fd4e71f47846710@mail.gmail.com>
References: <4663A209-CA71-4389-8ED9-C574CD939711@tlainvestments.com>
	<191c3a031002111152r1a56d46dl905d9c9c63e598d9@mail.gmail.com>
	<191c3a031002111153h2f98f657n9fd4e71f47846710@mail.gmail.com>
Message-ID: <D97A70D0-9196-41CB-9430-C6EAA2213D13@tlainvestments.com>

The test was from 16605 (from this morning). I just tried it with latest (16608) and the issue persists.

Thanks! 

-Troy

 
Troy Anderson
President, Lead Software Architect
e troy at chronostelecom.com
o 480.522.2115
c 602.327.1729

On Feb 11, 2010, at 12:53 PM, Anthony Minessale wrote:

> err latest
> 
> On Thu, Feb 11, 2010 at 1:52 PM, Anthony Minessale <anthony.minessale at gmail.com> wrote:
> try lastest
> 
> 
> 
> On Thu, Feb 11, 2010 at 1:11 PM, Troy Anderson <troy at tlainvestments.com> wrote:
> This is a strange one.  I make a call using an FXO analog line (mod_openzap).  During the call, I dial 200 and FS parks the call.  This is a Sangoma card using wanpipe.  Is there some kind of setting in there where it interprets DTMF?
> 
> Is there a way to see what OpenZAP is writing "ending bridge by request from write function"?
> 
> Thanks for any help on this!
> -Troy
> 
> Here's where it happens (phone number is redacted).
> 
> 2010-02-11 11:49:48.455727 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF 2:1680
> 2010-02-11 11:49:48.466015 [DEBUG] zap_io.c:1870 1:1 GENERATE DTMF [2]
> 2010-02-11 11:49:48.895733 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF 0:1280
> 2010-02-11 11:49:48.905726 [DEBUG] zap_io.c:1870 1:1 GENERATE DTMF [0]
> 2010-02-11 11:49:49.455734 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF 0:1360
> 2010-02-11 11:49:49.465771 [DEBUG] zap_io.c:1870 1:1 GENERATE DTMF [0]
> 2010-02-11 11:49:49.706363 [DEBUG] switch_ivr_bridge.c:469 OpenZAP/1:1/602xxxxxxx ending bridge by request from write function
> 2010-02-11 11:49:49.706363 [DEBUG] switch_core_session.c:638 Send signal sofia/internal/105 at 10.0.1.202 [BREAK]
> 2010-02-11 11:49:49.706363 [DEBUG] switch_ivr_bridge.c:562 BRIDGE THREAD DONE [sofia/internal/105 at 10.0.1.202]
> 2010-02-11 11:49:49.706363 [DEBUG] switch_ivr_bridge.c:582 Send signal OpenZAP/1:1/602xxxxxxx [BREAK]
> 2010-02-11 11:49:49.716359 [DEBUG] switch_core_session.c:638 Send signal OpenZAP/1:1/602xxxxxxx [BREAK]
> 2010-02-11 11:49:49.716359 [DEBUG] switch_ivr_bridge.c:562 BRIDGE THREAD DONE [OpenZAP/1:1/602xxxxxxx]
> 2010-02-11 11:49:49.716359 [DEBUG] switch_ivr_bridge.c:582 Send signal sofia/internal/105 at 10.0.1.202 [BREAK]
> 2010-02-11 11:49:49.716359 [DEBUG] switch_ivr_bridge.c:640 (OpenZAP/1:1/602xxxxxxx) State Change CS_EXCHANGE_MEDIA -> CS_PARK
> 2010-02-11 11:49:49.716359 [DEBUG] switch_core_session.c:1018 Send signal OpenZAP/1:1/602xxxxxxx [BREAK]
> 2010-02-11 11:49:49.716359 [DEBUG] switch_core_state_machine.c:351 (OpenZAP/1:1/602xxxxxxx) State EXCHANGE_MEDIA going to sleep
> 2010-02-11 11:49:49.716359 [DEBUG] switch_core_state_machine.c:314 (OpenZAP/1:1/602xxxxxxx) Running State Change CS_PARK
> 2010-02-11 11:49:49.716359 [DEBUG] switch_core_state_machine.c:357 (OpenZAP/1:1/602xxxxxxx) State PARK
> 2010-02-11 11:49:49.716359 [DEBUG] switch_core_state_machine.c:206 OpenZAP/1:1/602xxxxxxx Standard PARK
> 
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/4d60cfcb/attachment-0002.html 

From anthony.minessale at gmail.com  Thu Feb 11 12:40:14 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 11 Feb 2010 14:40:14 -0600
Subject: [Freeswitch-users] Is it necessary to call hangup....
In-Reply-To: <ebbc2d551002110849r11f6e586p4e8d90274233ac8a@mail.gmail.com>
References: <29b888f81002110532o3136d376g655c4b7a6dcbf598@mail.gmail.com>
	<7F8C140A-01BD-45A9-B71A-D3036BE10B3E@freeswitch.org>
	<ebbc2d551002110849r11f6e586p4e8d90274233ac8a@mail.gmail.com>
Message-ID: <191c3a031002111240o55a7c36ib25531198e9845a8@mail.gmail.com>

Only returns 404 if it did nothing in dp

On Feb 11, 2010 10:57 AM, "Jo?o Mesquita" <jmesquita at freeswitch.org> wrote:

Just a side question that relates to your answer Brian. When we run out of
stuff to do on the dialplan, we return 404 (if the leg is SIP of course)?

JM




On Thu, Feb 11, 2010 at 11:36 AM, Brian West <brian at freeswitch.org> wrote:
>
> If the dialplan ru...

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/9bcb2c1c/attachment-0002.html 

From mike at jerris.com  Thu Feb 11 12:42:25 2010
From: mike at jerris.com (Michael Jerris)
Date: Thu, 11 Feb 2010 14:42:25 -0600
Subject: [Freeswitch-users] demo_ivr cannot find sound files viarelative
	paths
In-Reply-To: <54726F0FE98C44398AFD34C3685CC47A@greyhawk.tonecommander.com>
References: <78030B9E01DD48F8B024A97035F96282@greyhawk.tonecommander.com><87f2f3b91002102057t7b039221kf3975553414880cf@mail.gmail.com>
	<E4BC77C3-8271-4F30-B6F8-6822E4747BC3@jerris.com>
	<54726F0FE98C44398AFD34C3685CC47A@greyhawk.tonecommander.com>
Message-ID: <438A7FC8-D68A-4D5D-AB1B-404F4C9A8AA9@jerris.com>

Just looked back at this and I am completely wrong.  Sounds prefix  
used to be set correctly in vars.xml and I remove that and set it to  
just the sounds dir in the core.  This is incorrect.  I think we will  
make the core set a default to the sounds dir and allow it to be  
overridden in vars.xml and restore that line in the default config.   
Expect a patch on this shortly.

Mike

On 2010-02-11, at 10:53 AM, "Robertre Hadley"  
<robert.hadley at teotech.com> wrote:

> That was it.  Thanks Mike.
>
> This test build was trunk and default configs.  The problem is in  
> the trunk vars.xml this line has been removed.  I added this line  
> back to vars.xml and the demo_ivr started working again.
>
>   <X-PRE-PROCESS cmd="set" data="sound_prefix=$${base_dir}/sounds/en/ 
> us/callie"/>
>
> The IVR source must still require sound_prefix to be set (and does  
> not use the new sounds_dir variable).
>
> Thanks again,
> Robert
>
> From: Michael Jerris [mailto:mike at jerris.com]
> Sent: Wednesday, February 10, 2010 9:34 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] demo_ivr cannot find sound files  
> viarelative paths
>
> did you make any changes to the default configs?  what is in your  
> vars.xml related to sounds?  the relative paths were always relative  
> to that dir, did you make any changes to the sounds prefix ?
>
> Mike
>
> On Feb 10, 2010, at 11:57 PM, Michael Collins wrote:
>
>
> If I read this log correctly it failed to find the "invalid entry"  
> file but it did find the phrases just fine. Can you confirm the  
> presence of this file:
> /usr/local/freeswitch/sounds/en/us/callie/ivr/8000/ivr- 
> that_was_an_invalid_entry.wav
>
> (It looks like this call is at 8kHz so that's where I'm assuming FS  
> is looking to find the sound file...)
>
> -MC
>
> On Wed, Feb 10, 2010 at 10:50 AM, Robert  Hadley <robert.hadley at teotech.com 
> > wrote:
>
> Hi,
>
> It appears a recent change (possibly the new sounds_dir variable or  
> the new ivr_menu folder?) may have broken relative sound file paths  
> in the IVR.  I built a today?s trunk version and installed to the de 
> fault location. Using the default conf files the demo_ivr cannot fin 
> d files based on the relative paths specified in ivr_menus/demo_ivr. 
> xml.
>
> [root at TEO-UCM-T2 conf]# cat ivr_menus/demo_ivr.xml
> <include>
>   <!-- demo IVR setup -->
>   <!-- demo IVR, Main Menu -->
>   <menu name="demo_ivr"
>       greet-long="phrase:demo_ivr_main_menu"
>       greet-short="phrase:demo_ivr_main_menu_short"
>       invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"
>       exit-sound="voicemail/vm-goodbye.wav"
>
>
> 2010-02-10 10:32:15.300337 [DEBUG] switch_ivr_menu.c:414 Executing  
> IVR menu demo_ivr
> 2010-02-10 10:32:15.300337 [DEBUG] switch_ivr_play_say.c:118 No  
> language specified - Using [en]
> 2010-02-10 10:32:15.300337 [DEBUG] switch_ivr_play_say.c:273 Handle  
> play-file:[ivr/ivr-welcome_to_freeswitch.wav] (en:en)
> 2010-02-10 10:32:15.300337 [DEBUG] switch_ivr_play_say.c:1158 Codec  
> Activated L16 at 8000hz 1 channels 20ms
> 2010-02-10 10:32:17.922147 [DEBUG] switch_ivr_play_say.c:1450 done  
> playing file
> 2010-02-10 10:32:18.042149 [DEBUG] switch_ivr_play_say.c:273 Handle  
> play-file:[ivr/ivr-this_ivr_will_let_you_test_features.wav] (en:en)
> 2010-02-10 10:32:18.042149 [DEBUG] switch_ivr_play_say.c:1158 Codec  
> Activated L16 at 8000hz 1 channels 20ms
> 2010-02-10 10:32:19.962158 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF  
> 2:800
> 2010-02-10 10:32:19.962158 [DEBUG] switch_ivr_play_say.c:1450 done  
> playing file
> 2010-02-10 10:32:20.082156 [DEBUG] switch_ivr_menu.c:329 waiting for  
> 3/4 digits t/o 2000
> 2010-02-10 10:32:20.120617 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF  
> 2:400
> 2010-02-10 10:32:20.442158 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF  
> 2:800
> 2010-02-10 10:32:20.682162 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF  
> 2:800
> 2010-02-10 10:32:20.682162 [DEBUG] switch_ivr_menu.c:376 digits '2222'
> 2010-02-10 10:32:20.682162 [DEBUG] switch_ivr_menu.c:470 action  
> regex [2222] [/^(10[01][0-9])$/] [0]
> 2010-02-10 10:32:20.682162 [DEBUG] switch_ivr_menu.c:560 IVR menu  
> 'demo_ivr' caught invalid input '2222'
> 2010-02-10 10:32:20.682162 [ERR] mod_sndfile.c:194 Error Opening  
> File [/usr/local/freeswitch/sounds/ivr/ivr- 
> that_was_an_invalid_entry.wav] [System error : No such file or  
> directory.]
> 2010-02-10 10:32:21.701505 [DEBUG] switch_ivr_play_say.c:118 No  
> language specified - Using [en]
> 2010-02-10 10:32:21.701505 [DEBUG] switch_ivr_play_say.c:273 Handle  
> play-file:[silence_stream://1000] (en:en)
> 2010-02-10 10:32:21.701505 [DEBUG] switch_ivr_play_say.c:1158 Codec  
> Activated L16 at 8000hz 1 channels 20ms
> 2010-02-10 10:32:22.700352 [DEBUG] switch_ivr_play_say.c:1450 done  
> playing file
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/8c69271b/attachment-0002.html 

From anthony.minessale at gmail.com  Thu Feb 11 12:46:46 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 11 Feb 2010 14:46:46 -0600
Subject: [Freeswitch-users] Calls being parked on DTMF
In-Reply-To: <D97A70D0-9196-41CB-9430-C6EAA2213D13@tlainvestments.com>
References: <4663A209-CA71-4389-8ED9-C574CD939711@tlainvestments.com>
	<191c3a031002111152r1a56d46dl905d9c9c63e598d9@mail.gmail.com>
	<191c3a031002111153h2f98f657n9fd4e71f47846710@mail.gmail.com>
	<D97A70D0-9196-41CB-9430-C6EAA2213D13@tlainvestments.com>
Message-ID: <191c3a031002111246h1ca4e0bo18549e150a29836e@mail.gmail.com>

I am skeptical,

Did you actually rebuild and restart FS after updating.

get a console trace with debug level please.

console loglevel debug



On Thu, Feb 11, 2010 at 2:35 PM, Troy Anderson <troy at tlainvestments.com>wrote:

> The test was from 16605 (from this morning). I just tried it with latest
> (16608) and the issue persists.
>
> Thanks!
>
> -Troy
>
>    <http://www.chronostelecom.com>
>
> Troy Anderson
>
> President, Lead Software Architect
>
> e troy at chronostelecom.com
>
> o 480.522.2115
>
> c 602.327.1729
>
> On Feb 11, 2010, at 12:53 PM, Anthony Minessale wrote:
>
> err latest
>
> On Thu, Feb 11, 2010 at 1:52 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> try lastest
>>
>>
>>
>> On Thu, Feb 11, 2010 at 1:11 PM, Troy Anderson <troy at tlainvestments.com>wrote:
>>
>>> This is a strange one.  I make a call using an FXO analog line
>>> (mod_openzap).  During the call, I dial 200 and FS parks the call.  This is
>>> a Sangoma card using wanpipe.  Is there some kind of setting in there where
>>> it interprets DTMF?
>>>
>>> Is there a way to see what OpenZAP is writing "ending bridge by request
>>> from write function"?
>>>
>>> Thanks for any help on this!
>>> -Troy
>>>
>>> Here's where it happens (phone number is redacted).
>>>
>>> 2010-02-11 11:49:48.455727 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF 2:1680
>>> 2010-02-11 11:49:48.466015 [DEBUG] zap_io.c:1870 1:1 GENERATE DTMF [2]
>>> 2010-02-11 11:49:48.895733 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF 0:1280
>>> 2010-02-11 11:49:48.905726 [DEBUG] zap_io.c:1870 1:1 GENERATE DTMF [0]
>>> 2010-02-11 11:49:49.455734 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF 0:1360
>>> 2010-02-11 11:49:49.465771 [DEBUG] zap_io.c:1870 1:1 GENERATE DTMF [0]
>>> 2010-02-11 11:49:49.706363 [DEBUG] switch_ivr_bridge.c:469
>>> OpenZAP/1:1/602xxxxxxx ending bridge by request from write function
>>> 2010-02-11 11:49:49.706363 [DEBUG] switch_core_session.c:638 Send signal
>>> sofia/internal/105 at 10.0.1.202 [BREAK]
>>> 2010-02-11 11:49:49.706363 [DEBUG] switch_ivr_bridge.c:562 BRIDGE THREAD
>>> DONE [sofia/internal/105 at 10.0.1.202]
>>> 2010-02-11 11:49:49.706363 [DEBUG] switch_ivr_bridge.c:582 Send signal
>>> OpenZAP/1:1/602xxxxxxx [BREAK]
>>> 2010-02-11 11:49:49.716359 [DEBUG] switch_core_session.c:638 Send signal
>>> OpenZAP/1:1/602xxxxxxx [BREAK]
>>> 2010-02-11 11:49:49.716359 [DEBUG] switch_ivr_bridge.c:562 BRIDGE THREAD
>>> DONE [OpenZAP/1:1/602xxxxxxx]
>>> 2010-02-11 11:49:49.716359 [DEBUG] switch_ivr_bridge.c:582 Send signal
>>> sofia/internal/105 at 10.0.1.202 [BREAK]
>>> 2010-02-11 11:49:49.716359 [DEBUG] switch_ivr_bridge.c:640
>>> (OpenZAP/1:1/602xxxxxxx) State Change CS_EXCHANGE_MEDIA -> CS_PARK
>>> 2010-02-11 11:49:49.716359 [DEBUG] switch_core_session.c:1018 Send signal
>>> OpenZAP/1:1/602xxxxxxx [BREAK]
>>> 2010-02-11 11:49:49.716359 [DEBUG] switch_core_state_machine.c:351
>>> (OpenZAP/1:1/602xxxxxxx) State EXCHANGE_MEDIA going to sleep
>>> 2010-02-11 11:49:49.716359 [DEBUG] switch_core_state_machine.c:314
>>> (OpenZAP/1:1/602xxxxxxx) Running State Change CS_PARK
>>> 2010-02-11 11:49:49.716359 [DEBUG] switch_core_state_machine.c:357
>>> (OpenZAP/1:1/602xxxxxxx) State PARK
>>> 2010-02-11 11:49:49.716359 [DEBUG] switch_core_state_machine.c:206
>>> OpenZAP/1:1/602xxxxxxx Standard PARK
>>>
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/d84a1eaf/attachment-0002.html 

From anthony.minessale at gmail.com  Thu Feb 11 12:49:57 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 11 Feb 2010 14:49:57 -0600
Subject: [Freeswitch-users] Lua script hangup detect
In-Reply-To: <FFD1A126-5CC0-4D02-87CE-B3EBA553FC4A@freeswitch.org>
References: <314dc3f81002100117h6b1bc49cna989624a1ae4e0e2@mail.gmail.com>
	<314dc3f81002100800o41fd6561w394d5e83e69d3d80@mail.gmail.com>
	<4B72DFF9.4040402@gmail.com>
	<1b46b4e81002100932x7af486ekd4b05f4750cace20@mail.gmail.com>
	<4B72F2A8.4070503@gmail.com>
	<4CA5432B-5F6E-438E-B6EF-3277847E7922@jerris.com>
	<4B73B57E.9040901@gmail.com>
	<D8FFF1FB-7B2D-49BC-940C-445EBF838BE3@jerris.com>
	<4B73C640.20700@gmail.com>
	<FFD1A126-5CC0-4D02-87CE-B3EBA553FC4A@freeswitch.org>
Message-ID: <191c3a031002111249sd655896s5da75e61736b238d@mail.gmail.com>

The script cannot end if you are stuck in a while loop that never exits.
The same thing is true in any script in any language.


On Thu, Feb 11, 2010 at 7:26 AM, Brian West <brian at freeswitch.org> wrote:

> Yes ODBC doesn't seem to leak... while the native one leaks like a sieve.
>
> /b
>
> On Feb 11, 2010, at 2:56 AM, Nazim Agabekov wrote:
>
> > I'm using luasql with ODBC MySQL driver in production. I've never tried
> > to use luasql with "native" mysql driver, but ODBC one works great.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/242fd618/attachment-0002.html 

From maxim.tsvetov at gmail.com  Thu Feb 11 13:12:33 2010
From: maxim.tsvetov at gmail.com (maxim.tsvetov)
Date: Thu, 11 Feb 2010 13:12:33 -0800 (PST)
Subject: [Freeswitch-users] DTMF problem
In-Reply-To: <F960CC45-C9D2-4150-A108-C0E8B230284D@freeswitch.org>
References: <89c9bbf81002111209v6e2f36b3x86a628c5e1390599@mail.gmail.com>
	<F960CC45-C9D2-4150-A108-C0E8B230284D@freeswitch.org>
Message-ID: <1265922753047-4557446.post@n2.nabble.com>


I already added "dtmf-relay rtp-nte" and this doesn't work.

Also I don't have "dtmf-interworking rtp-nte" command in Cisco.
-- 
View this message in context: http://n2.nabble.com/DTMF-problem-tp4557122p4557446.html
Sent from the freeswitch-users mailing list archive at Nabble.com.



From troy at tlainvestments.com  Thu Feb 11 13:14:48 2010
From: troy at tlainvestments.com (Troy Anderson)
Date: Thu, 11 Feb 2010 14:14:48 -0700
Subject: [Freeswitch-users] Calls being parked on DTMF
In-Reply-To: <4C603407-C72A-4D93-9451-771A8634BD54@tlainvestments.com>
References: <4663A209-CA71-4389-8ED9-C574CD939711@tlainvestments.com>
	<191c3a031002111152r1a56d46dl905d9c9c63e598d9@mail.gmail.com>
	<191c3a031002111153h2f98f657n9fd4e71f47846710@mail.gmail.com>
	<D97A70D0-9196-41CB-9430-C6EAA2213D13@tlainvestments.com>
	<191c3a031002111246h1ca4e0bo18549e150a29836e@mail.gmail.com>
	<4C603407-C72A-4D93-9451-771A8634BD54@tlainvestments.com>
Message-ID: <87de771b1002111314n4ecc3681m20cd3cbaa5f77def@mail.gmail.com>

I did a fs stop, make, make install, fs start. ?As we speak, I'm doing
a make sure, etc. And will report back.

-Troy

On Thursday, February 11, 2010, Troy Anderson <admin at tlainvestments.com> wrote:
> I did a fs stop, make, make install, fs start. ?As we speak, I'm doing a make sure, etc. And will report back.
> -Troy
> --?I'm probably driving while typing this, so pardon the typos! --
> On Feb 11, 2010, at 1:46 PM, Anthony Minessale <anthony.minessale at gmail.com> wrote:
>
> I am skeptical,
>
> Did you actually rebuild and restart FS after updating.
>
> get a console trace with debug level please.
>
> console loglevel debug
>
>
>
> On Thu, Feb 11, 2010 at 2:35 PM, Troy Anderson <troy at tlainvestments.com> wrote:
> The test was from 16605 (from this morning). I just tried it with latest (16608) and the issue persists.
>
> Thanks!
> -Troy
>
>
>
> 	
> 		??<http://www.chronostelecom.com>
> 	
> 	
> 		Troy Anderson
> 		President, Lead Software Architect
> 		e troy at chronostelecom.com
> 		o 480.522.2115
> 		c 602.327.1729
> 		
> 	
>
>
>
>
> On Feb 11, 2010, at 12:53 PM, Anthony Minessale wrote:
> err latest
>
> On Thu, Feb 11, 2010 at 1:52 PM, Anthony Minessale <anthony.minessale at gmail.com> wrote:
>
> try lastest
>
>
> On Thu, Feb 11, 2010 at 1:11 PM, Troy Anderson <troy at tlainvestments.com> wrote:
>
> This is a strange one. ?I make a call using an FXO analog line (mod_openzap). ?During the call, I dial 200 and FS parks the call. ?This is a Sangoma card using wanpipe. ?Is there some kind of setting in there where it interprets DTMF?
>
> Is there a way to see what OpenZAP is writing "ending bridge by request from write function"?
>
> Thanks for any help on this!
> -Troy
>
> Here's where it happens (phone number is redacted).
>
> 2010-02-11 11:49:48.455727 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF 2:1680
> 2010-02-11 11:49:48.466015 [DEBUG] zap_io.c:1870 1:1 GENERATE DTMF [2]
> 2010-02-11 11:49:48.895733 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF 0:1280
> 2010-02-11 11:49:48.905726 [DEBUG] zap_io.c:1870 1:1 GENERATE DTMF [0]
> 2010-02-11 11:49:49.455734 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF 0:1360
> 2010-02-11 11:49:49.465771 [DEBUG] zap_io.c:1870 1:1 GENERATE DTMF [0]
> 2010-02-11 11:49:49.706363 [DEBUG] switch_ivr_bridge.c:469 OpenZAP/1:1/602xxxxxxx ending bridge by request from write function
> 2010-02-11 11:49:49.706363 [DEBUG] switch_core_session.c:638 Send signal sofia/internal/105 at 10.0.1.202 [BREAK]
> 2010-02-11 11:49:49.706363 [DEBUG] switch_ivr_bridge.c:562 BRIDGE THREAD DONE [sofia/internal/105 at 10.0.1.202]
> 2010-02-11 11:49:49.706363 [DEBUG] switch_ivr_bridge.c:582 Send signal OpenZAP/1:1/602xxxxxxx [BREAK]
> 2010-02-11 11:49:49.716359 [DEBUG] switch_core_session.c:638 Send signal OpenZAP/1:1/602xxxxxxx [BREAK]
> 2010-02-11 11:49:49.716359 [DEBUG] switch_ivr_bridge.c:562 BRIDGE THREAD DONE [OpenZAP/1:1/602xxxxxxx]
> 2010-02-11 11:49:49.716359 [DEBUG] switch_ivr_bridge.c:582 Send signal sofia/internal/105 at 10.0.1.202 [BREAK]
> 2010-02-11 11:49:49.716359 [DEBUG] switch_ivr_bridge.c:640 (OpenZAP/1:1/602xxxxxxx) State Change CS_EXCHANGE_MEDIA -> CS_PARK
> 2010-02-11 11:49:49.716359 [DEBUG] switch_core_session.c:1018 Send signal OpenZAP/1:1/602xxxxxxx [BREAK]
> 2010-02-11 11:49:49.716359 [DEBUG] switch_core_state_machine.c:351 (OpenZ

-- 
Troy Anderson
Chronos Consulting
6501 E. Greenway Pkwy #103/422
Scottsdale, AZ 85254
(480) 922-5380 (office)
(602) 327-1729 (cell)

--



From brian at freeswitch.org  Thu Feb 11 13:19:44 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 11 Feb 2010 15:19:44 -0600
Subject: [Freeswitch-users] Calls being parked on DTMF
In-Reply-To: <87de771b1002111314n4ecc3681m20cd3cbaa5f77def@mail.gmail.com>
References: <4663A209-CA71-4389-8ED9-C574CD939711@tlainvestments.com>
	<191c3a031002111152r1a56d46dl905d9c9c63e598d9@mail.gmail.com>
	<191c3a031002111153h2f98f657n9fd4e71f47846710@mail.gmail.com>
	<D97A70D0-9196-41CB-9430-C6EAA2213D13@tlainvestments.com>
	<191c3a031002111246h1ca4e0bo18549e150a29836e@mail.gmail.com>
	<4C603407-C72A-4D93-9451-771A8634BD54@tlainvestments.com>
	<87de771b1002111314n4ecc3681m20cd3cbaa5f77def@mail.gmail.com>
Message-ID: <530A2EC6-3585-418A-B375-720E973F68A0@freeswitch.org>

"make current" is the best thing you can do.

/b

On Feb 11, 2010, at 3:14 PM, Troy Anderson wrote:

> I did a fs stop, make, make install, fs start.  As we speak, I'm doing
> a make sure, etc. And will report back.
> 
> -Troy




From mike at van.lammeren.net  Thu Feb 11 13:51:20 2010
From: mike at van.lammeren.net (Mike van Lammeren)
Date: Thu, 11 Feb 2010 16:51:20 -0500
Subject: [Freeswitch-users] Lua script hangup detect
In-Reply-To: <FFD1A126-5CC0-4D02-87CE-B3EBA553FC4A@freeswitch.org>
References: <314dc3f81002100117h6b1bc49cna989624a1ae4e0e2@mail.gmail.com>
	<314dc3f81002100800o41fd6561w394d5e83e69d3d80@mail.gmail.com>
	<4B72DFF9.4040402@gmail.com>
	<1b46b4e81002100932x7af486ekd4b05f4750cace20@mail.gmail.com>
	<4B72F2A8.4070503@gmail.com>
	<4CA5432B-5F6E-438E-B6EF-3277847E7922@jerris.com>
	<4B73B57E.9040901@gmail.com>
	<D8FFF1FB-7B2D-49BC-940C-445EBF838BE3@jerris.com>
	<4B73C640.20700@gmail.com>
	<FFD1A126-5CC0-4D02-87CE-B3EBA553FC4A@freeswitch.org>
Message-ID: <5d2828f1002111351s5f0cdff2odb6b35fa9be9eb32@mail.gmail.com>

D'oh!

I am currently working on a project that uses Lua and the native MySQL
driver, so as soon as I read this comment, I decided that I had better do a
bit of research.

I wrote a Lua test script that makes 10 queries against a MySQL database,
then ran it repeatedly.

My results show that "leaks like a sieve" is quite correct. To me, it
doesn't look like any memory is released, ever. After only 5000 queries or
so, the memory allocated for FreeSWITCH balloons from 15 Mb to 50 Mb, and
never goes back down.

Thanks, Brian, for the heads up!

Mike van Lammeren


On Thu, Feb 11, 2010 at 8:26 AM, Brian West <brian at freeswitch.org> wrote:

> Yes ODBC doesn't seem to leak... while the native one leaks like a sieve.
>
> /b
>
> On Feb 11, 2010, at 2:56 AM, Nazim Agabekov wrote:
>
> > I'm using luasql with ODBC MySQL driver in production. I've never tried
> > to use luasql with "native" mysql driver, but ODBC one works great.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/a78bb120/attachment-0002.html 

From mike at jerris.com  Thu Feb 11 14:22:46 2010
From: mike at jerris.com (Michael Jerris)
Date: Thu, 11 Feb 2010 17:22:46 -0500
Subject: [Freeswitch-users] Forbidden using UDP, works with TCP/TLS
In-Reply-To: <33c87fa31002110213q1de0eebfsb950d8725ea9aba@mail.gmail.com>
References: <33c87fa31002101824h21e3bf7lb13b68e8d4fd36a2@mail.gmail.com>
	<2EF27EFC-1634-46A3-B438-2EE99BEF58F1@jerris.com>
	<33c87fa31002110131w3cf6ad00n802669856085a879@mail.gmail.com>
	<CA21231C-108B-4445-A3B9-BF8D94FFE898@jerris.com>
	<33c87fa31002110213q1de0eebfsb950d8725ea9aba@mail.gmail.com>
Message-ID: <0A905834-850F-4D1F-AEC5-D3A87CE4141B@jerris.com>

how is this different from the working one?

Mike

On Feb 11, 2010, at 5:13 AM, Mark Campbell-Smith wrote:

> ah thats true... The trace is not too readable to me, but may give
> some insight to someone that can read the sofia logs....
> 
> 
> recv 752 bytes from udp/[121.xxx.xxx.xxx]:5060 at 10:10:37.803288:
>   ------------------------------------------------------------------------
>   REGISTER sip:mydns.dyndns.org SIP/2.0
>   Via: SIP/2.0/UDP 121.xxx.xxx.xxx:5060;branch=z9hG4bK-9052c91f
>   From: 2000 <sip:2000 at mydns.dyndns.org>;tag=7a9dbbbfa691136do0
>   To: 2000 <sip:2000 at mydns.dyndns.org>
>   Call-ID: 610db38-dd3b511f at 121.xxx.xxx.xxx
>   CSeq: 32330 REGISTER
>   Contact: 2000 <sip:2000 at 121.xxx.xxx.xxx:5060>;expires=900
>   Authorization: Digest username="2000", realm="mydns.dyndns.org",
> nonce="b3298cfe-16f5-11df-9734-e77ba7a22ac3",
> uri="sip:mydns.dyndns.org",
> response="724fca542ce08d3f12b9ba1043bebb0c", algorithm=MD5,
> qop="1225e2f1"
>   Max-Forwards: 70
>   User-Agent: Linksys/SPA3102-5.1.10(GW)
>   Supported: x-sipura
>   Supported: replaces
>   Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
>   Content-Length: 0
> 
> 
>   ------------------------------------------------------------------------
> tport_deliver(0x97cde80): msg 0x98297e8 (752 bytes) from
> udp/121.xxx.xxx.xxx:5060/sip next=(nil)
> nta: received REGISTER sip:mydns.dyndns.org SIP/2.0 (CSeq 32330)
> nta: REGISTER (32330) going to a default leg
> nua: nua_stack_process_request: entering
> nua: nh_create: entering
> nua: nh_create_handle: entering
> nua: nua_stack_set_params: entering
> soa_clone(static::0x97cc698, 0x9794808, 0x981cb70) called
> soa_set_params(static::0x9758ba8, ...) called
> nua(0x981cb70): event i_register 100 Trying
> nua: nua_application_event: entering
> nua: nua_respond: entering
> nua(0x981cb70): sent signal r_respond
> nua: nua_handle_destroy: entering
> nua(0x981cb70): sent signal r_destroy
> nua: nua_handle_magic: entering
> nua: nua_handle_destroy: entering
> nua(0x981cb70): recv signal r_respond 403 Forbidden
> nua: nua_stack_set_params: entering
> soa_set_params(static::0x9758ba8, ...) called
> tport_tsend(0x97cde80) tpn = UDP/121.xxx.xxx.xxx:5060
> tport_resolve addrinfo = 121.xxx.xxx.xxx:5060
> tport_by_addrinfo(0x97cde80): not found by name UDP/121.xxx.xxx.xxx:5060
> tport_vsend(0x97cde80): 495 bytes of 495 to udp/121.xxx.xxx.xxx:5060
> tport_vsend returned 495
> send 495 bytes to udp/[121.xxx.xxx.xxx]:5060 at 10:10:37.812955:
>   ------------------------------------------------------------------------
>   SIP/2.0 403 Forbidden
> 
> On Thu, Feb 11, 2010 at 8:57 PM, Michael Jerris <mike at jerris.com> wrote:
>> you can crank up the sofia loglevel as well
>> 
>> Mike
>> 
>> On Feb 11, 2010, at 4:31 AM, Mark Campbell-Smith wrote:
>> 
>>> I don't see debug logs (fsctl debug_level 9 or loglevel 9) during the
>>> registration process.
>>> 
>>> All I see is the sip messages when the sip trace is activated (403 Forbidden)
>>> 
>>> Is there other debugging that I can enable?
>>> 
>>> On Thu, Feb 11, 2010 at 4:25 PM, Michael Jerris <mike at jerris.com> wrote:
>>>> Look at the FreeSWITCH debug logs and compare the differences, they should tell you why.
>>>> 
>>>> Mike




From mike at jerris.com  Thu Feb 11 14:34:30 2010
From: mike at jerris.com (Michael Jerris)
Date: Thu, 11 Feb 2010 17:34:30 -0500
Subject: [Freeswitch-users] how to set From address for voicemail emails?
In-Reply-To: <4B7401DD.6050408@apcl.us>
References: <4B736846.1040908@apcl.us>
	<7471DDD5-0A73-4F18-AF90-BA8534EBC9E7@jerris.com>
	<4B7401DD.6050408@apcl.us>
Message-ID: <E9393F2E-4EBC-410D-8E8A-A3D947DABE26@jerris.com>

what your looking for is in the template files in voicemail, they are in the conf directory with a tpl file extension

http://svn.freeswitch.org/svn/freeswitch/trunk/conf/voicemail.tpl

Mike


On Feb 11, 2010, at 8:10 AM, Paul Levin wrote:

> Mike,
>     Thank you very much for the reply.  But you are talking way over my head.  Can you please tell me very specifically what file(s) and value(s) I need to set/change in order to set the From address on voice mail emails?
>     Thanks,
>     Paul
> 
> 
> Michael Jerris wrote:
>> 
>> This is the difference between what is sent to the mail server in the mime content, and what is passed as MAIL FROM:  to the smtp server.  The latter is controlled by that param, the former is in the template.
>> 
>> Mike
>> 
>> On Feb 10, 2010, at 9:15 PM, Paul Levin wrote:
>> 
>>> I am running FreeSwitch on Windows.  I have msmtp setup and voice mail emails are being sent. 
>>> 
>>> I have msmtp configured to set a "From" address of me at mydomain.com, but when FreeSwitch sends an email with a voice mail message from Alice, the From address of the email is Alice at sipServerDomain.com.  According to the Mod voicemail document (http://wiki.freeswitch.org/wiki/Mod_voicemail) the email_email-from parameter should control this, but I tried setting it in conf\autoload_configs\directory.conf.xml (as per that document) and also in Bob.xml (the account getting the voicemail).  Neither place changed the value being used.  
>>> 
>>> How do I get this changed?
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>   
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/8701a210/attachment-0002.html 

From paul at apcl.us  Thu Feb 11 14:42:08 2010
From: paul at apcl.us (Paul Levin)
Date: Thu, 11 Feb 2010 17:42:08 -0500
Subject: [Freeswitch-users] how to set From address for voicemail emails?
	(repost)
Message-ID: <4B7487C0.2010200@apcl.us>

An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/cf871ff2/attachment-0002.html 

From joel.sisko at iconverged.com  Thu Feb 11 14:48:50 2010
From: joel.sisko at iconverged.com (joel.sisko at iconverged.com)
Date: Thu, 11 Feb 2010 16:48:50 -0600 (CST)
Subject: [Freeswitch-users] Mod_Conference capacity....
In-Reply-To: <268119402.962161265928492648.JavaMail.root@mail-2.01.com>
Message-ID: <715035894.962251265928530648.JavaMail.root@mail-2.01.com>




Mod_Conference 

Group, I am looking for anyone?s input (based on your own experience) on how many individual three party conference?s and the biggest single conference (total number of listeners with just 1 speaker) FreeSwitch can handle using the mod_conference application? Just looking for rough numbers if I were to use a dual processor quad core system with 12GB of RAM. 

I understand that transcoding and other factors create limits but I am just looking for same raw numbers that I should be able to obtain if the moon and stars were to align correctly. 

Thanks for the input in advance. 

Joel
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/e40795a0/attachment-0002.html 

From paul at apcl.us  Thu Feb 11 14:48:19 2010
From: paul at apcl.us (Paul Levin)
Date: Thu, 11 Feb 2010 17:48:19 -0500
Subject: [Freeswitch-users] how to set From address for voicemail emails?
In-Reply-To: <E9393F2E-4EBC-410D-8E8A-A3D947DABE26@jerris.com>
References: <4B736846.1040908@apcl.us>
	<7471DDD5-0A73-4F18-AF90-BA8534EBC9E7@jerris.com>	<4B7401DD.6050408@apcl.us>
	<E9393F2E-4EBC-410D-8E8A-A3D947DABE26@jerris.com>
Message-ID: <4B748933.7000100@apcl.us>

An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/dc8a1f02/attachment-0002.html 

From paul at apcl.us  Thu Feb 11 14:48:55 2010
From: paul at apcl.us (Paul Levin)
Date: Thu, 11 Feb 2010 17:48:55 -0500
Subject: [Freeswitch-users] how to set From address for voicemail
 emails? (repost)
In-Reply-To: <4B7487C0.2010200@apcl.us>
References: <4B7487C0.2010200@apcl.us>
Message-ID: <4B748957.5070206@apcl.us>

An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/95b70ce3/attachment-0002.html 

From anthony.minessale at gmail.com  Thu Feb 11 15:01:52 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 11 Feb 2010 17:01:52 -0600
Subject: [Freeswitch-users] Bypass-media and REFER method
In-Reply-To: <B816EF85C732094BA2016B2112F3DC2B040C6061@isr-brass.interwise.com>
References: <B816EF85C732094BA2016B2112F3DC2B0402C777@isr-brass.interwise.com>
	<B816EF85C732094BA2016B2112F3DC2B040C5CA6@isr-brass.interwise.com>
	<191c3a031002100645n5d827949xdefd24143f10759@mail.gmail.com>
	<B816EF85C732094BA2016B2112F3DC2B040C6061@isr-brass.interwise.com>
Message-ID: <191c3a031002111501y7e5917b8g4faae4b8752e6fc2@mail.gmail.com>

Where do you think it's failing?
that log shows it get refer, go back to dp, invite to 3341 and then the
other end hangs up at some point.


On Wed, Feb 10, 2010 at 11:01 AM, Victor Maruani <vmaruani at interwise.com>wrote:

>  Hi,
>
>
>
> Logs are on pb 12099
>
> I hope this helps.
>
> Reproduced with revision 16599.
>
>
>
> A-side (10.10.5.19) is an x-lite registered with extension 1002
>
> B (.5.51) refers to C (.5.48) none are registered.
>
>
>
> Please refer to previous emails for details of dialplan and what I try to
> do?
>
> Let me know if you need more info
>
>
>
> Thanks!
>
>
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony
> Minessale
> *Sent:* Wednesday, February 10, 2010 4:46 PM
>
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Bypass-media and REFER method
>
>
>
> update to latest trunk and reproduce your problem with full debug enabled.
>
> sofia profile internal siptrace on
> console loglevel debug
>
>  On Wed, Feb 10, 2010 at 4:44 AM, Victor Maruani <vmaruani at interwise.com>
> wrote:
>
> Hi,
>
>
>
> I can't have a blind transfer work properly if I use bypass-media=true.
>
>
>
> My first message may have been unclear, here I added excerpt from the
> dialplan:
>
>
>
>                 <!--- ext MyIVR -->
>
>                  <extension name="Extension MyIVR" continue="false">
>
>                    <condition field="destination_number"
> expression="^337$">
>
>                                    <action application="set"
> data="bypass_media=true"/>
>
>                                    <action application="bridge"
> data="sofia/internal/337 at 10.10.5.51"/>
>
>                    </condition>
>
>                  </extension>
>
>
>
>
>
>                 <!--- ext MyServ -->
>
>                  <extension name="Extension MyServ " continue="false">
>
>                    <condition field="destination_number"
> expression="^3341$">
>
>                                                 <action application="set"
> data="bypass_media=true"/>
>
>                                    <action application="bridge"
> data="sofia/internal/3341 at 10.10.5.48"/>
>
>                    </condition>
>
>                  </extension>
>
>
>
> The connection to MyIVR works. Then it sends Refer-to (3341 at ...) which
> fails as I described it in the previous mail.
>
>
>
> I would like to know if the feature has been validated and if I'm missing
> something in the configuration.
>
>
>
> Any help would be very appreciated.
>
>
>
> Thanks!
>
>
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Victor
> Maruani
> *Sent:* Sunday, February 07, 2010 5:01 PM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* [Freeswitch-users] Bypass-media and REFER method
>
>
>
> Hi,
>
>
>
> I'm trying to do a POC using FS, the goal is to have FS handle REFERs
> containing proprietary data.
>
> I want to have some logic on top of FS and also use the fail over
> mechanism.
>
> in short, I have something like this:
>
> (third party) A side  --- FS ---- B side (IVR server)
>
>
>
> the IVR the sends a REFER to FS. I don't want A to deal with it.
>
> now say B refers to C, it would be considered as a "group" C1, C2 ... to
> which I want FS to failover.
>
> only when one has answered should A be updated (REINVITE) and B notified
> and disconnected.
>
> if all fails I would expect B to be notified of the failure and proceed as
> I wish without "losing" A.
>
>
>
> from what I've read FS should be OK for the job but I have a couple issues:
>
>
>
> 1 ) I have some issues getting FS handle a REFER while in bypass-media
> mode.
>
> (I tried with the release and some revisions including latest)
>
> first when I bridge A and B everything is fine and media is bypassed.
>
> When B sends REFER to C:
>
> -           FS immediately NOTIFY B of success and send a reinvite to A
> with SDP containing its own media IP/port.
>
> -          then it does INVITE C with A's SDP.
>
> -          B gets disconnected. A is not updated with C's sdp.
>
> so at this point A sends RTP to FS and C sends RTP to A. ?
>
>
>
> I basically have one extension for B: (set bypass-media and bridge to B)
>
> and another extension to C which does the same actions.
>
> what do you think I do wrong?
>
>
>
>
>
> 2 ) how can I catch the REFER and set variables from it? (like ref-by or
> ref-to)
>
> in the dial plan I do catch the INVITE sent to C, but how to do it with the
> REFER itself?
>
>
>
>
>
> thanks for your help!
>
>
>
>
>
> Best Regards,
>
> Victor.
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100211/0e7b5e70/attachment-0002.html 

From gavin.henry at gmail.com  Thu Feb 11 15:22:49 2010
From: gavin.henry at gmail.com (Gavin Henry)
Date: Thu, 11 Feb 2010 23:22:49 +0000
Subject: [Freeswitch-users] Mod_Conference capacity....
In-Reply-To: <715035894.962251265928530648.JavaMail.root@mail-2.01.com>
References: <268119402.962161265928492648.JavaMail.root@mail-2.01.com>
	<715035894.962251265928530648.JavaMail.root@mail-2.01.com>
Message-ID: <13ca621c1002111522v7fe55a62w84043ed986155b28@mail.gmail.com>

Why don't use script a test or use sipp and then dial in yourself to listen?

Cheers.

On 11/02/2010, joel.sisko at iconverged.com <joel.sisko at iconverged.com> wrote:
>
>
>
> Mod_Conference
>
> Group, I am looking for anyone?s input (based on your own experience) on how
> many individual three party conference?s and the biggest single conference
> (total number of listeners with just 1 speaker) FreeSwitch can handle using
> the mod_conference application? Just looking for rough numbers if I were to
> use a dual processor quad core system with 12GB of RAM.
>
> I understand that transcoding and other factors create limits but I am just
> looking for same raw numbers that I should be able to obtain if the moon and
> stars were to align correctly.
>
> Thanks for the input in advance.
>
> Joel

-- 
Sent from my mobile device

http://www.suretecsystems.com/services/openldap/
http://www.suretectelecom.com



From brian at freeswitch.org  Thu Feb 11 15:47:46 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 11 Feb 2010 17:47:46 -0600
Subject: [Freeswitch-users] Mod_Conference capacity....
In-Reply-To: <13ca621c1002111522v7fe55a62w84043ed986155b28@mail.gmail.com>
References: <268119402.962161265928492648.JavaMail.root@mail-2.01.com>
	<715035894.962251265928530648.JavaMail.root@mail-2.01.com>
	<13ca621c1002111522v7fe55a62w84043ed986155b28@mail.gmail.com>
Message-ID: <E7571BC2-135B-4743-B941-1A84968A4395@freeswitch.org>

Not an optimal test scenario unless you know wtf you're doing!

/b

On Feb 11, 2010, at 5:22 PM, Gavin Henry wrote:

> Why don't use script a test or use sipp and then dial in yourself to listen?
> 
> Cheers.




From joel.sisko at iconverged.com  Thu Feb 11 16:05:58 2010
From: joel.sisko at iconverged.com (Joel Sisko)
Date: Thu, 11 Feb 2010 18:05:58 -0600 (CST)
Subject: [Freeswitch-users] Mod_Conference capacity....
In-Reply-To: <895275167.980621265932897183.JavaMail.root@mail-2.01.com>
Message-ID: <530719564.981411265933158149.JavaMail.root@mail-2.01.com>

Brian/Gavin thanks for the input. But I agree with Brian, if were that easy I would have done it prior to the post.

Just looking to find out what some of the communities success has been to see if this is a path we should go down for a conference solution platform.

Joel
----- Original Message -----
From: "Brian West" <brian at freeswitch.org>
To: freeswitch-users at lists.freeswitch.org
Sent: Thursday, February 11, 2010 3:47:46 PM GMT -08:00 US/Canada Pacific
Subject: Re: [Freeswitch-users] Mod_Conference capacity....

Not an optimal test scenario unless you know wtf you're doing!

/b

On Feb 11, 2010, at 5:22 PM, Gavin Henry wrote:

> Why don't use script a test or use sipp and then dial in yourself to listen?
> 
> Cheers.


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



From ustcorporation at yahoo.com  Thu Feb 11 19:12:07 2010
From: ustcorporation at yahoo.com (teldev)
Date: Thu, 11 Feb 2010 19:12:07 -0800 (PST)
Subject: [Freeswitch-users] Problem installing latest Wanpipe for Sangoma
 A104DE under Centos 5.3 32-bit
Message-ID: <1265944327228-4559052.post@n2.nabble.com>


On the step "Compiling API Development Utilities ...Failed" received message
"Error:  Failed to compile WANPIPE API Tools  !!!"




-- 
View this message in context: http://n2.nabble.com/Problem-installing-latest-Wanpipe-for-Sangoma-A104DE-under-Centos-5-3-32-bit-tp4559052p4559052.html
Sent from the freeswitch-users mailing list archive at Nabble.com.



From brian at freeswitch.org  Thu Feb 11 19:17:43 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 11 Feb 2010 21:17:43 -0600
Subject: [Freeswitch-users] Problem installing latest Wanpipe for
	Sangoma A104DE under Centos 5.3 32-bit
In-Reply-To: <1265944327228-4559052.post@n2.nabble.com>
References: <1265944327228-4559052.post@n2.nabble.com>
Message-ID: <93BAA7CC-0930-4D87-88D7-D0D9A3C2EAE5@freeswitch.org>

I would recommend you contact Sangoma for assistance.

/b

On Feb 11, 2010, at 9:12 PM, teldev wrote:

> 
> On the step "Compiling API Development Utilities ...Failed" received message
> "Error:  Failed to compile WANPIPE API Tools  !!!"
> 




From lakindia89 at gmail.com  Thu Feb 11 20:18:25 2010
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Fri, 12 Feb 2010 09:48:25 +0530
Subject: [Freeswitch-users] Play music to A leg.
In-Reply-To: <191c3a031002110804m4c18e7e0y317f44272b42c8b4@mail.gmail.com>
References: <fe67a3341002090519r2cc6e8edm7702b8d012fc2be3@mail.gmail.com>
	<87f2f3b91002090737l4efa8en29bde9457fef7ea5@mail.gmail.com>
	<fe67a3341002092157x7484f3d2id6678510bbec66cc@mail.gmail.com>
	<87f2f3b91002100702u7581589ua866d2492a008b23@mail.gmail.com>
	<fe67a3341002110211w60390e1cm2d12fadd1e802778@mail.gmail.com>
	<87f2f3b91002110754laf9d01dxa94a36bc1b8c83df@mail.gmail.com>
	<191c3a031002110804m4c18e7e0y317f44272b42c8b4@mail.gmail.com>
Message-ID: <7d79b3931002112018y2512954cuca83ce21ff3406fe@mail.gmail.com>

Dear Antony,
In bridge if we are making parallel calls, then group_confirm_key in execute
mode will execute for all the extensions, and whomsoever finishes the script
first, will be bridged.

But I think nagalenoj need to execute the script for the extension which
answers the call first, not for all the extension.!!!.

>From nanalenoj's post
            " but when I do originate for multiple extensions, I want this
script to work based on the answered extension."


On Thu, Feb 11, 2010 at 9:34 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> group_confirm_key in execute mode can execute a lua script instead that can
> read as many digits as you want and parse the results.
>
>
>
> On Thu, Feb 11, 2010 at 9:54 AM, Michael Collins <msc at freeswitch.org>wrote:
>
>> Hehe, this is getting more and more complicated. You may want to consider
>> using the event socket and have your call control be done from a more 3rd
>> party-ish perspective. If you've got all these different scenarios it might
>> be better to let an external script do all the work.
>>
>> http://wiki.freeswitch.org/wiki/Event_Socket
>>
>> -MC
>>
>>
>> On Thu, Feb 11, 2010 at 2:11 AM, Nagalenoj H. <nagalenoj at gmail.com>wrote:
>>
>>> But My scenario is,
>>>    After I get the call from X.
>>>    I answer the call in some scenarios and won't answer the call. So,
>>> this leg can either be answered or unanswered.
>>>    I originate a call to another number.
>>>    After getting some digits from this originated leg.
>>>    I do uuid_bridge of these 2 legs.
>>>
>>> I want to play some file[ringback] to leg A before bridging to B.
>>>
>>> On Wed, Feb 10, 2010 at 8:32 PM, Michael Collins <msc at freeswitch.org>wrote:
>>>
>>>>
>>>>
>>>> On Tue, Feb 9, 2010 at 9:57 PM, Nagalenoj H. <nagalenoj at gmail.com>wrote:
>>>>
>>>>> Because, I want to get some digits before bridging the legs. I've tried
>>>>> group_confirm_key, but it accepts only one digit, I need multiple digits, so
>>>>> I can't use.
>>>>> I've also tried group_confirm_file, but when I do originate for
>>>>> multiple extensions, I want this script to work based on the answered
>>>>> extension.
>>>>>
>>>>> So, I've originated and processed the events to do my job.
>>>>>
>>>>> How do I play some music to A leg?
>>>>>
>>>>> I might be missing something, but couldn't you just park the call ("A
>>>> leg") until you connect to the other party ("B leg") and then uuid_bridge at
>>>> whatever point you want?
>>>> -MC
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> Regards,
>>> Nagalenoj H.
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100212/3116590a/attachment-0002.html 

From lakindia89 at gmail.com  Thu Feb 11 20:21:57 2010
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Fri, 12 Feb 2010 09:51:57 +0530
Subject: [Freeswitch-users] How to kill multiple UUIDs.
In-Reply-To: <191c3a031002100759v2fef0f48na5afccf46bb08311@mail.gmail.com>
References: <7d79b3931002100307h4d370a22l6f04cafa5bf5806d@mail.gmail.com>
	<A35CD010-E86F-4B7E-A28B-2A9000B8E009@jerris.com>
	<191c3a031002100759v2fef0f48na5afccf46bb08311@mail.gmail.com>
Message-ID: <7d79b3931002112021y21ad551do50f3913352aa1855@mail.gmail.com>

Thanks antony, that works.

Thank you all very much!!

On Wed, Feb 10, 2010 at 9:29 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> or you can set a common var like foo=bar on all the chans and do
>
> hupall normal_clearing foo bar
>
>
>
> On Wed, Feb 10, 2010 at 9:16 AM, Michael Jerris <mike at jerris.com> wrote:
>
>> you can api hangup hook to call
>>
>> lua multi_kill.lua uuid1 uuid2 uuid?.
>>
>> and then write the trivial lua script for that.
>>
>> Mike
>>
>> On Feb 10, 2010, at 6:07 AM, lakshmanan ganapathy wrote:
>>
>> > Hi all,
>> >
>> > My situation is
>> >     A called to 1005 -- Which executes an ESL program.
>> >     Now from the program I will made the parallel call using "api
>> originate [origination_uuid=uuid1]user/1006,[origination_uuid2]user/1001
>> &park()".
>> >     UUID's are obtained from create_uuid.
>> >
>> >     I'll then wait for the api to return, to check whether the call is
>> answered or rejected by the other end.
>> >     But while I'm waiting, if A hangup the call, I just want to kill the
>> calls that are originated by my program.
>> >     So I taught of using api_hang_up_hook and I set that variable to
>> uuid_kill uuid1 uuid2.
>> >     But it only killed the uuid1.
>> >
>> > Is there any other ways to kill multiple uuid's??
>> > please help?
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100212/10476717/attachment-0002.html 

From jingwei.yang at gmail.com  Thu Feb 11 21:56:55 2010
From: jingwei.yang at gmail.com (Jingwei Yang)
Date: Fri, 12 Feb 2010 13:56:55 +0800
Subject: [Freeswitch-users] Is it possible to repeat music in playback
In-Reply-To: <191c3a031002110639h3cf14efdvf7715b973fe3caed@mail.gmail.com>
References: <13529f9d1002110233g57c8a1a5j3cdd38e22fa6a13d@mail.gmail.com>
	<2FBDB302-2B54-40CC-B694-332854219774@freeswitch.org>
	<191c3a031002110639h3cf14efdvf7715b973fe3caed@mail.gmail.com>
Message-ID: <13529f9d1002112156u11603033u698236c86e0d9abb@mail.gmail.com>

Thanks Anthony, exactly what I want.

On Thu, Feb 11, 2010 at 10:39 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> or try endless_playback app
>
>
>
> On Thu, Feb 11, 2010 at 7:26 AM, Brian West <brian at freeswitch.org> wrote:
>
>> Why not just use Fifo to hold them?  Or Park the agent and send the
>> session a message to play music?  You then have options to define loop
>> count.
>>
>> http://wiki.freeswitch.org/wiki/Event_Socket#execute
>>
>> /b
>>
>> On Feb 11, 2010, at 4:33 AM, Jingwei Yang wrote:
>>
>> > Hello,
>> >
>> > I've defined a very simple dialplan like the one below and when the
>> caller is connected to this plan, I hope to keep the call alive and repeat
>> the music set by playback. How am I able to achieve this?
>> >
>> >
>> >     <extension name="agent_hold">
>> >       <condition field="destination_number" expression="^33333$">
>> >         <action application="answer"/>
>> >         <action application="playback" data="customized/test.wav"/>
>> >       </condition>
>> >     </extension>
>> >
>> >
>> > Thanks,
>> > -Jingwei
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100212/f027575c/attachment-0002.html 

From jingwei.yang at gmail.com  Fri Feb 12 00:44:56 2010
From: jingwei.yang at gmail.com (Jingwei Yang)
Date: Fri, 12 Feb 2010 16:44:56 +0800
Subject: [Freeswitch-users] Is it possible to specify music on hold with
	uuid_hold?
In-Reply-To: <E37048AD-5A46-42D5-B724-6C2344AD8A2B@freeswitch.org>
References: <13529f9d1002110130v335d8177m74f791577b558ca@mail.gmail.com>
	<8FF07098-581B-47FB-A039-63A58041E5BF@freeswitch.org>
	<13529f9d1002110554u65431499hc3fa877252023577@mail.gmail.com>
	<E37048AD-5A46-42D5-B724-6C2344AD8A2B@freeswitch.org>
Message-ID: <13529f9d1002120044j712590ch11456b6b98c86d72@mail.gmail.com>

Thanks Brian. I think I understand what you mean about holding the far end
now. When client A and agent B are in a call and I try to uuid_hold B,
client A is in fact put on hold. This looks a bit strange but the good thing
is I'm able to let A hear the customized hold music by defining it in the
dialplan that B is connected to before bridged to A. Thanks!

On Thu, Feb 11, 2010 at 9:59 PM, Brian West <brian at freeswitch.org> wrote:

> uuid_setvar the variable hold_music on the opposite UUID you're holding...
> uuid_hold isn't doing exactly what you think it is.  ;)
>
> /b
>
> On Feb 11, 2010, at 7:54 AM, Jingwei Yang wrote:
>
> > Sorry Brian, I don't quite understand your answer. What is the far end
> and what is the near end? In my case, I bridge client A to agent B. While
> they're talking, I use uuid_hold to put client A on hold. Then A hears the
> default music. After a while, uuid_hold off A and the conversation between A
> and B resumes. uuid_hold looks perfect for my situation except I'm not able
> to change the default music.
> >
> > Regards,
> > -Jingwei
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100212/f828733c/attachment-0002.html 

From codecomplete at free.fr  Fri Feb 12 02:17:47 2010
From: codecomplete at free.fr (Fred-145)
Date: Fri, 12 Feb 2010 11:17:47 +0100
Subject: [Freeswitch-users] Driving peripherals through Freeswitch?
References: <2srvm5945qgcno44oetn9ngii0u3aed73p@4ax.com>
	<2srvm5945qgcno44oetn9ngii0u3aed73p-e09XROE/p8c@public.gmane.org>
	<20F1477A-E98A-4BC9-904D-CB313D8E7B4C@jerris.com>
Message-ID: <slaan5p2j1b0njesrn6u6eokdvno4dns0i@4ax.com>

On Tue, 9 Feb 2010 16:12:38 -0500, Michael Jerris
<mike at jerris.com> wrote:
>The possibilities are limitless, but requires someone to code a module to interface, or external scripts using the system api or some socket based application.

Thanks for the feedback. I'll see what I can find.




From ivdreg at gmail.com  Fri Feb 12 02:20:54 2010
From: ivdreg at gmail.com (ivdreg ivdreg)
Date: Fri, 12 Feb 2010 12:20:54 +0200
Subject: [Freeswitch-users] Help on: park_timeout variable
In-Reply-To: <191c3a031002111153n655103b4ub46d67f9094d6f57@mail.gmail.com>
References: <b437a0e1002110851v274b8cbah54d920a5e59a809b@mail.gmail.com>
	<191c3a031002111142n625ca0d4n641378242cd69b87@mail.gmail.com>
	<191c3a031002111153n655103b4ub46d67f9094d6f57@mail.gmail.com>
Message-ID: <b437a0e1002120220i2b4a9a04jdafdd0ae09102bf@mail.gmail.com>

Thanks Anthony. Everything is fine now.

2010/2/11 Anthony Minessale <anthony.minessale at gmail.com>

> actually, i see a small buglet there, try trunk.
>
>
>
> On Thu, Feb 11, 2010 at 1:42 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> This is what happens in a b leg, it only happens when you transfer a
>> call.  This is by design to give the other phone a chance to kill the leg.
>> This is not really a problem persae.
>>
>>
>> On Thu, Feb 11, 2010 at 10:51 AM, ivdreg ivdreg <ivdreg at gmail.com> wrote:
>>
>>> Hi All,
>>>
>>> After updating to current SVN from 1.0.4 I have a problem when caller
>>> party hangs up a call. I have 3 seconds timeout before B leg disconnects. I
>>> think that this is caused by  code in switch_ivr_bridge.c in function static
>>> switch_status_t audio_bridge_on_exchange_media(switch_core_session_t
>>> *session)
>>> ......
>>>
>>>         if (switch_channel_get_state(channel) == CS_EXCHANGE_MEDIA) {
>>>                 switch_channel_set_variable(channel, "park_timeout",
>>> "3");
>>>                 switch_channel_set_state(channel, CS_PARK);
>>>         }
>>> ......
>>>
>>> This happens even if I set park_after_bridge=false variable.
>>> Is anybody has this problem ?
>>>
>>> Thanks
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100212/710985fd/attachment-0002.html 

From codecomplete at free.fr  Fri Feb 12 02:20:25 2010
From: codecomplete at free.fr (Fred-145)
Date: Fri, 12 Feb 2010 11:20:25 +0100
Subject: [Freeswitch-users] FS based Softphone?
References: <e90351e71002081039y7c7723a6x7181d9fdd208bdde@mail.gmail.com>
Message-ID: <tpaan5t7uldeqk8ud10k4pja31hi2jv7tt@4ax.com>

On Mon, 8 Feb 2010 10:39:46 -0800, Christian Jensen
<christian at officepools.com> wrote:
> If not, what is the best softphone for use with
> a mostly windows but some ubuntu and mac environment?

Looks like ZoIPer is the only softphone available for the three OS's

http://www.zoiper.com/




From lakindia89 at gmail.com  Fri Feb 12 03:23:32 2010
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Fri, 12 Feb 2010 16:53:32 +0530
Subject: [Freeswitch-users] Is it possible to repeat music in playback
In-Reply-To: <191c3a031002110639h3cf14efdvf7715b973fe3caed@mail.gmail.com>
References: <13529f9d1002110233g57c8a1a5j3cdd38e22fa6a13d@mail.gmail.com>
	<2FBDB302-2B54-40CC-B694-332854219774@freeswitch.org>
	<191c3a031002110639h3cf14efdvf7715b973fe3caed@mail.gmail.com>
Message-ID: <7d79b3931002120323x7a2520fcr1610af69ba4ca51@mail.gmail.com>

Hi antony,
Is there any way to stop the endless_playback??
I tried with break. But it didn't worked!!


On Thu, Feb 11, 2010 at 8:09 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> or try endless_playback app
>
>
>
> On Thu, Feb 11, 2010 at 7:26 AM, Brian West <brian at freeswitch.org> wrote:
>
>> Why not just use Fifo to hold them?  Or Park the agent and send the
>> session a message to play music?  You then have options to define loop
>> count.
>>
>> http://wiki.freeswitch.org/wiki/Event_Socket#execute
>>
>> /b
>>
>> On Feb 11, 2010, at 4:33 AM, Jingwei Yang wrote:
>>
>> > Hello,
>> >
>> > I've defined a very simple dialplan like the one below and when the
>> caller is connected to this plan, I hope to keep the call alive and repeat
>> the music set by playback. How am I able to achieve this?
>> >
>> >
>> >     <extension name="agent_hold">
>> >       <condition field="destination_number" expression="^33333$">
>> >         <action application="answer"/>
>> >         <action application="playback" data="customized/test.wav"/>
>> >       </condition>
>> >     </extension>
>> >
>> >
>> > Thanks,
>> > -Jingwei
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100212/9346b79b/attachment-0002.html 

From kond at nstel.ru  Fri Feb 12 03:24:27 2010
From: kond at nstel.ru (Nikolay Kondratyev)
Date: Fri, 12 Feb 2010 14:24:27 +0300
Subject: [Freeswitch-users] h323 - sip call is not working
In-Reply-To: <65d96fc81002111013r38d938b0t3ffc1a49b7ff5b92@mail.gmail.com>
Message-ID: <20100212112427.0FC1F11F9E@mail.nstel.ru>

Tihomir, 

Thanks for the reply.

 

I was quite surprised to see 30ms packetization time. and first thought it's
a typo.

But now I see (if I'm not mistaken) that FS really sends open logical
channel with Alaw:30ms when opening logical channel during fast start (see
attached file).

This parameter is "display only" on my IPO. 

Is it possible to change it on the FS side? I think it should be somewhere
in mod_h323 configuration. But I did not found any mod_h323 configuration
parameters except ip addr. and port.

 

By the way, I know that one can use different packetization times for the
same codec, but I've never heard, that somebody really uses 30 ms for
G711Alaw. Always 20ms.

 

Thanks ans regards,

Nikolay.

 

 

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Tihomir
Culjaga
Sent: Thursday, February 11, 2010 9:14 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] h323 - sip call is not working

 

 

On Thu, Feb 11, 2010 at 4:18 PM, Nikolay Kondratyev <kond at nstel.ru> wrote:

Tihomir,

Thanks for help.

I enabled fast start on IPO and I can hear voice now. But the ringback tone
and voice appears to be wheezy, but I will investigate that tomorrow.


set framing time for your codec to 30ms in IPO, also play with  PI in
alerting... set it to 2.

 

Thanks again.

Nikolay.

 

 

 


  _____  


From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Tihomir
Culjaga
Sent: Thursday, February 11, 2010 2:34 PM


To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] h323 - sip call is not working

 

Nikolay, you are sending slow start with tunneling=true ?!?!



It is not gong to work :)

Please can you set fast start instead?


Your call failed because there was no mediaControll channel negotiated at
all... actually the call had to be aborted because wrong signaling .. but
anyhow.

Please on your IPO use FastStart with h245Tunneling=true... also, same
settings on FS side as well (exclude h245 in setup as well).




Frame 13 (277 bytes on wire, 277 bytes captured)
Ethernet II, Src: AlliedTe_22:9b:4a (00:00:cd:22:9b:4a), Dst:
Vmware_67:33:a7 (00:0c:29:67:33:a7)
Internet Protocol, Src: 172.23.14.2 (172.23.14.2), Dst: 172.23.22.49
(172.23.22.49)
Transmission Control Protocol, Src Port: oirtgsvc (4141), Dst Port:
h323hostcall (1720), Seq: 1, Ack: 1, Len: 223
TPKT, Version: 3, Length: 223
Q.931
    Protocol discriminator: Q.931
    Call reference value length: 2
    Call reference flag: Message sent from originating side
    Call reference value: 0012
    Message type: SETUP (0x05)
    Bearer capability
    Display  'Gornak Alexandr>2853'
    Calling party number: '5840'
    Called party number: '2853'
    User-user
H.225.0 CS
    H323-UserInformation
        h323-uu-pdu
            h323-message-body: setup (0)
                setup
            h4501SupplementaryService: 1 item
            1... .... h245Tunneling: True







On Thu, Feb 11, 2010 at 9:18 AM, Nikolay Kondratyev <kond at nstel.ru> wrote:

> But I do recommenf to use SIP, since it's much better supported by FS.
>
> /Peter

But SIP is poorly supported by IPO.
Thanks and regards,

Nikolay.


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org

http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

 


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100212/403d779e/attachment-0002.html 
-------------- next part --------------
An embedded and charset-unspecified text was scrubbed...
Name: alaw30ms.txt
Url: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100212/403d779e/attachment-0002.txt 

From vetali100 at gmail.com  Fri Feb 12 01:33:27 2010
From: vetali100 at gmail.com (Vitalii Colosov)
Date: Fri, 12 Feb 2010 11:33:27 +0200
Subject: [Freeswitch-users] Other party does not hear voice when connecting
	with X-Lite
Message-ID: <a858e7691002120133q5e4c806fx309517e4fdec45c5@mail.gmail.com>

Hi,

I have installed FS default configuration and setup an external gateway for
international calls.
When I call to international phone using YATE windows client, both parties
able to hear voice... erevything is OK.

But when I connect using X-Lite, it connects but other party cannot hear me.
I can hear him well...

Could you please hint in which direction should I look?

Thank you,
Vitalii Colosov
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100212/5ac6d9ec/attachment-0002.html 

From bottleman at icf.org.ru  Fri Feb 12 04:07:39 2010
From: bottleman at icf.org.ru (Georgiewskiy Yuriy)
Date: Fri, 12 Feb 2010 15:07:39 +0300 (MSK)
Subject: [Freeswitch-users] h323 - sip call is not working
In-Reply-To: <20100212112427.0FC1F11F9E@mail.nstel.ru>
References: <20100212112427.0FC1F11F9E@mail.nstel.ru>
Message-ID: <alpine.DEB.2.00.1002121506130.27140@srv.icf.org.ru>

On 2010-02-12 14:24 +0300, Nikolay Kondratyev wrote freeswitch-users at lists....:

NK>Tihomir, 
NK>
NK>Thanks for the reply.
NK>
NK> 
NK>
NK>I was quite surprised to see 30ms packetization time. and first thought it's
NK>a typo.
NK>
NK>But now I see (if I'm not mistaken) that FS really sends open logical
NK>channel with Alaw:30ms when opening logical channel during fast start (see
NK>attached file).
NK>
NK>This parameter is "display only" on my IPO. 
NK>
NK>Is it possible to change it on the FS side? I think it should be somewhere
NK>in mod_h323 configuration. But I did not found any mod_h323 configuration
NK>parameters except ip addr. and port.

It's not implemented at this time.

NK>By the way, I know that one can use different packetization times for the
NK>same codec, but I've never heard, that somebody really uses 30 ms for
NK>G711Alaw. Always 20ms.
NK>
NK> 
NK>
NK>Thanks ans regards,
NK>
NK>Nikolay.
NK>
NK> 
NK>
NK> 
NK>
NK> 
NK>
NK>  _____  
NK>
NK>From: freeswitch-users-bounces at lists.freeswitch.org
NK>[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Tihomir
NK>Culjaga
NK>Sent: Thursday, February 11, 2010 9:14 PM
NK>To: freeswitch-users at lists.freeswitch.org
NK>Subject: Re: [Freeswitch-users] h323 - sip call is not working
NK>
NK> 
NK>
NK> 
NK>
NK>On Thu, Feb 11, 2010 at 4:18 PM, Nikolay Kondratyev <kond at nstel.ru> wrote:
NK>
NK>Tihomir,
NK>
NK>Thanks for help.
NK>
NK>I enabled fast start on IPO and I can hear voice now. But the ringback tone
NK>and voice appears to be wheezy, but I will investigate that tomorrow.
NK>
NK>
NK>set framing time for your codec to 30ms in IPO, also play with  PI in
NK>alerting... set it to 2.
NK>
NK> 
NK>
NK>Thanks again.
NK>
NK>Nikolay.
NK>
NK> 
NK>
NK> 
NK>
NK> 
NK>
NK>
NK>  _____  
NK>
NK>
NK>From: freeswitch-users-bounces at lists.freeswitch.org
NK>[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Tihomir
NK>Culjaga
NK>Sent: Thursday, February 11, 2010 2:34 PM
NK>
NK>
NK>To: freeswitch-users at lists.freeswitch.org
NK>Subject: Re: [Freeswitch-users] h323 - sip call is not working
NK>
NK> 
NK>
NK>Nikolay, you are sending slow start with tunneling=true ?!?!
NK>
NK>
NK>
NK>It is not gong to work :)
NK>
NK>Please can you set fast start instead?
NK>
NK>
NK>Your call failed because there was no mediaControll channel negotiated at
NK>all... actually the call had to be aborted because wrong signaling .. but
NK>anyhow.
NK>
NK>Please on your IPO use FastStart with h245Tunneling=true... also, same
NK>settings on FS side as well (exclude h245 in setup as well).
NK>
NK>
NK>
NK>
NK>Frame 13 (277 bytes on wire, 277 bytes captured)
NK>Ethernet II, Src: AlliedTe_22:9b:4a (00:00:cd:22:9b:4a), Dst:
NK>Vmware_67:33:a7 (00:0c:29:67:33:a7)
NK>Internet Protocol, Src: 172.23.14.2 (172.23.14.2), Dst: 172.23.22.49
NK>(172.23.22.49)
NK>Transmission Control Protocol, Src Port: oirtgsvc (4141), Dst Port:
NK>h323hostcall (1720), Seq: 1, Ack: 1, Len: 223
NK>TPKT, Version: 3, Length: 223
NK>Q.931
NK>    Protocol discriminator: Q.931
NK>    Call reference value length: 2
NK>    Call reference flag: Message sent from originating side
NK>    Call reference value: 0012
NK>    Message type: SETUP (0x05)
NK>    Bearer capability
NK>    Display  'Gornak Alexandr>2853'
NK>    Calling party number: '5840'
NK>    Called party number: '2853'
NK>    User-user
NK>H.225.0 CS
NK>    H323-UserInformation
NK>        h323-uu-pdu
NK>            h323-message-body: setup (0)
NK>                setup
NK>            h4501SupplementaryService: 1 item
NK>            1... .... h245Tunneling: True
NK>
NK>
NK>
NK>
NK>
NK>
NK>
NK>On Thu, Feb 11, 2010 at 9:18 AM, Nikolay Kondratyev <kond at nstel.ru> wrote:
NK>
NK>> But I do recommenf to use SIP, since it's much better supported by FS.
NK>>
NK>> /Peter
NK>
NK>But SIP is poorly supported by IPO.
NK>Thanks and regards,
NK>
NK>Nikolay.
NK>
NK>
NK>_______________________________________________
NK>FreeSWITCH-users mailing list
NK>FreeSWITCH-users at lists.freeswitch.org
NK>
NK>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
NK>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
NK>http://www.freeswitch.org
NK>
NK> 
NK>
NK>
NK>_______________________________________________
NK>FreeSWITCH-users mailing list
NK>FreeSWITCH-users at lists.freeswitch.org
NK>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
NK>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
NK>http://www.freeswitch.org
NK>
NK> 
NK>
NK>

C ?????????                       With Best Regards
???????????? ????.                Georgiewskiy Yuriy
+7 4872 711666                    +7 4872 711666
???? +7 4872 711143               fax +7 4872 711143
???????? ??? "?? ?? ??????"       IT Service Ltd
http://nkoort.ru                  http://nkoort.ru
JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
YG129-RIPE                        YG129-RIPE

From tculjaga at gmail.com  Fri Feb 12 04:47:08 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Fri, 12 Feb 2010 13:47:08 +0100
Subject: [Freeswitch-users] h323 - sip call is not working
In-Reply-To: <alpine.DEB.2.00.1002121506130.27140@srv.icf.org.ru>
References: <20100212112427.0FC1F11F9E@mail.nstel.ru>
	<alpine.DEB.2.00.1002121506130.27140@srv.icf.org.ru>
Message-ID: <65d96fc81002120447j2010d4fbha771f30b0fc42e6b@mail.gmail.com>

.

> NK>
> NK>This parameter is "display only" on my IPO.
> NK>
> NK>Is it possible to change it on the FS side? I think it should be
> somewhere
>

It is deep in H323plus... The problem is that IPO should stick to the
protocol and adjust the framing size accordingly... Anyhow, yes this is one
of the issues that makes trigger a different call flow on the remote side.
In my scenario, when cisco PGW receives a setup with different framing size,
it triggers a different behaviour.... PGW starts to negotiate framing using
Facility OLC.

On Avaya S8700 the behavior is clean ... it accepts it.

Anyhow, i'm looking for a way to make this config available in the
h323.conf.xml


T.


> NK>in mod_h323 configuration. But I did not found any mod_h323
> configuration
> NK>parameters except ip addr. and port.
>
> It's not implemented at this time.
>
> NK>By the way, I know that one can use different packetization times for
> the
> NK>same codec, but I've never heard, that somebody really uses 30 ms for
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100212/661c9d2a/attachment-0002.html 

From brian at freeswitch.org  Fri Feb 12 06:29:06 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 12 Feb 2010 08:29:06 -0600
Subject: [Freeswitch-users] h323 - sip call is not working
In-Reply-To: <20100212112427.0FC1F11F9E@mail.nstel.ru>
References: <20100212112427.0FC1F11F9E@mail.nstel.ru>
Message-ID: <26C4E111-8329-48F8-A8DA-081B851A9514@freeswitch.org>

This is a rather broad assumption.  I have seen 40ms, 60ms and even 80ms in the wild.  It all depends on what you want to do.  It lowers overhead and increases efficiency on the wire.

/b

On Feb 12, 2010, at 5:24 AM, Nikolay Kondratyev wrote:

> By the way, I know that one can use different packetization times for the same codec, but I?ve never heard, that somebody really uses 30 ms for G711Alaw. Always 20ms.
>  

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100212/d60a20d1/attachment-0002.html 

From brian at freeswitch.org  Fri Feb 12 06:30:28 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 12 Feb 2010 08:30:28 -0600
Subject: [Freeswitch-users] Other party does not hear voice when
	connecting with X-Lite
In-Reply-To: <a858e7691002120133q5e4c806fx309517e4fdec45c5@mail.gmail.com>
References: <a858e7691002120133q5e4c806fx309517e4fdec45c5@mail.gmail.com>
Message-ID: <2F81A680-6A5C-4E3F-9590-FFF6AC4EA80D@freeswitch.org>

is x-lite behind nat with the freeswitch box?  If so you'll need to disable the discover global IP so that it doesn't try to hair pin thru your NAT router.... Most nat routers won't work correctly trying to do that.

/b

On Feb 12, 2010, at 3:33 AM, Vitalii Colosov wrote:

>  
> But when I connect using X-Lite, it connects but other party cannot hear me.
> I can hear him well...




From brian at freeswitch.org  Fri Feb 12 06:31:17 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 12 Feb 2010 08:31:17 -0600
Subject: [Freeswitch-users] FS based Softphone?
In-Reply-To: <tpaan5t7uldeqk8ud10k4pja31hi2jv7tt@4ax.com>
References: <e90351e71002081039y7c7723a6x7181d9fdd208bdde@mail.gmail.com>
	<tpaan5t7uldeqk8ud10k4pja31hi2jv7tt@4ax.com>
Message-ID: <1941DDEC-0B9D-4A9E-87E5-12E70178A0B0@freeswitch.org>

LIES LIES LIES... FSComm runs on all three OS's heck even FreeSWITCH on the command line runs on all three.  :P

/b
PS: Zoiper doesn't do 32kHz or 48kHz voip.

On Feb 12, 2010, at 4:20 AM, Fred-145 wrote:

> Looks like ZoIPer is the only softphone available for the three OS's
> 
> http://www.zoiper.com/

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100212/48e7ec85/attachment-0002.html 

From Prometheus001 at gmx.net  Fri Feb 12 06:37:51 2010
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Fri, 12 Feb 2010 15:37:51 +0100
Subject: [Freeswitch-users] FS based Softphone?
In-Reply-To: <tpaan5t7uldeqk8ud10k4pja31hi2jv7tt@4ax.com>
References: <e90351e71002081039y7c7723a6x7181d9fdd208bdde@mail.gmail.com>
	<tpaan5t7uldeqk8ud10k4pja31hi2jv7tt@4ax.com>
Message-ID: <4B7567BF.3010609@gmx.net>

I even got TLS/SRTP working with Zoiper Bizz.

Best regards
Peter

Fred-145 schrieb:
> On Mon, 8 Feb 2010 10:39:46 -0800, Christian Jensen
> <christian at officepools.com> wrote:
>   
>> If not, what is the best softphone for use with
>> a mostly windows but some ubuntu and mac environment?
>>     
>
> Looks like ZoIPer is the only softphone available for the three OS's
>
> http://www.zoiper.com/
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>   



From edpimentl at gmail.com  Fri Feb 12 06:49:45 2010
From: edpimentl at gmail.com (EdPimentl)
Date: Fri, 12 Feb 2010 09:49:45 -0500
Subject: [Freeswitch-users] FS based Softphone?
In-Reply-To: <4B7567BF.3010609@gmx.net>
References: <e90351e71002081039y7c7723a6x7181d9fdd208bdde@mail.gmail.com> 
	<tpaan5t7uldeqk8ud10k4pja31hi2jv7tt@4ax.com> <4B7567BF.3010609@gmx.net>
Message-ID: <9dc4a1671002120649p45940f7bi85b041740a4fbe1@mail.gmail.com>

Why not take the positive approach and if there are features, you want
FScomm to support, then develop it, or put up a bounty for it.
This would be a more productive and welcome alternative.

-E
http://vCardCloud.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100212/f6edb375/attachment-0002.html 

From brian at freeswitch.org  Fri Feb 12 06:55:00 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 12 Feb 2010 08:55:00 -0600
Subject: [Freeswitch-users] FS based Softphone?
In-Reply-To: <9dc4a1671002120649p45940f7bi85b041740a4fbe1@mail.gmail.com>
References: <e90351e71002081039y7c7723a6x7181d9fdd208bdde@mail.gmail.com>
	<tpaan5t7uldeqk8ud10k4pja31hi2jv7tt@4ax.com>
	<4B7567BF.3010609@gmx.net>
	<9dc4a1671002120649p45940f7bi85b041740a4fbe1@mail.gmail.com>
Message-ID: <0C4E0EED-F33C-4F25-897E-7FCFB8FF35C7@freeswitch.org>

Excellent idea.  ;) BTW it does everything FreeSWITCH can do already.  :P

/b

On Feb 12, 2010, at 8:49 AM, EdPimentl wrote:

> Why not take the positive approach and if there are features, you want FScomm to support, then develop it, or put up a bounty for it.
> This would be a more productive and welcome alternative.
> 
> -E
> http://vCardCloud.com

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100212/1d1d203e/attachment-0002.html 

From jmesquita at freeswitch.org  Fri Feb 12 07:10:45 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Fri, 12 Feb 2010 13:10:45 -0200
Subject: [Freeswitch-users] FS based Softphone?
In-Reply-To: <0C4E0EED-F33C-4F25-897E-7FCFB8FF35C7@freeswitch.org>
References: <e90351e71002081039y7c7723a6x7181d9fdd208bdde@mail.gmail.com>
	<tpaan5t7uldeqk8ud10k4pja31hi2jv7tt@4ax.com>
	<4B7567BF.3010609@gmx.net>
	<9dc4a1671002120649p45940f7bi85b041740a4fbe1@mail.gmail.com>
	<0C4E0EED-F33C-4F25-897E-7FCFB8FF35C7@freeswitch.org>
Message-ID: <ebbc2d551002120710l57ac8dfer83b731dd9ea7c747@mail.gmail.com>

And some extra work and goodies will be available this weekend.


Jo?o Mesquita
FSComm Developer


On Fri, Feb 12, 2010 at 12:55 PM, Brian West <brian at freeswitch.org> wrote:

> Excellent idea.  ;) BTW it does everything FreeSWITCH can do already.  :P
>
> /b
>
> On Feb 12, 2010, at 8:49 AM, EdPimentl wrote:
>
> Why not take the positive approach and if there are features, you want
> FScomm to support, then develop it, or put up a bounty for it.
> This would be a more productive and welcome alternative.
>
> -E
> http://vCardCloud.com
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100212/f1c07c40/attachment-0002.html 

From woodydickson at gmail.com  Fri Feb 12 07:32:30 2010
From: woodydickson at gmail.com (Woody Dickson)
Date: Fri, 12 Feb 2010 07:32:30 -0800
Subject: [Freeswitch-users] transmission status of channel
Message-ID: <a44688a1002120732v7d95a6f9x2d4bd7ed68de59eb@mail.gmail.com>

Hi,

Is there anyway to obtain the %packet lost, latency, and jitter info
for each channel?

Any idea how to obtain those information?

thx,
Woody



From joel.sisko at iconverged.com  Fri Feb 12 08:33:45 2010
From: joel.sisko at iconverged.com (Joel Sisko)
Date: Fri, 12 Feb 2010 10:33:45 -0600 (CST)
Subject: [Freeswitch-users] Mod_Conference capacity....
In-Reply-To: <530719564.981411265933158149.JavaMail.root@mail-2.01.com>
Message-ID: <1804026806.1068641265992425257.JavaMail.root@mail-2.01.com>

Brian/Gavin do you know of any resource/link that can give an indication on what we could be expected of conference capacity? We are trying to determine a few different platforms we can possible use for conferencing and doing a little preliminary homework.

Kindest regards,

Joel


----- Original Message -----
From: "Joel Sisko" <joel.sisko at iconverged.com>
To: freeswitch-users at lists.freeswitch.org
Sent: Thursday, February 11, 2010 4:05:58 PM GMT -08:00 US/Canada Pacific
Subject: Re: [Freeswitch-users] Mod_Conference capacity....

Brian/Gavin thanks for the input. But I agree with Brian, if were that easy I would have done it prior to the post.

Just looking to find out what some of the communities success has been to see if this is a path we should go down for a conference solution platform.

Joel
----- Original Message -----
From: "Brian West" <brian at freeswitch.org>
To: freeswitch-users at lists.freeswitch.org
Sent: Thursday, February 11, 2010 3:47:46 PM GMT -08:00 US/Canada Pacific
Subject: Re: [Freeswitch-users] Mod_Conference capacity....

Not an optimal test scenario unless you know wtf you're doing!

/b

On Feb 11, 2010, at 5:22 PM, Gavin Henry wrote:

> Why don't use script a test or use sipp and then dial in yourself to listen?
> 
> Cheers.


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



From anthony.minessale at gmail.com  Fri Feb 12 08:56:31 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 12 Feb 2010 10:56:31 -0600
Subject: [Freeswitch-users] Mod_Conference capacity....
In-Reply-To: <1804026806.1068641265992425257.JavaMail.root@mail-2.01.com>
References: <530719564.981411265933158149.JavaMail.root@mail-2.01.com>
	<1804026806.1068641265992425257.JavaMail.root@mail-2.01.com>
Message-ID: <191c3a031002120856q93e52cep14ee96f1e7ed6d1@mail.gmail.com>

consider commercial support from consulting at freeswitch.org


On Fri, Feb 12, 2010 at 10:33 AM, Joel Sisko <joel.sisko at iconverged.com>wrote:

> Brian/Gavin do you know of any resource/link that can give an indication on
> what we could be expected of conference capacity? We are trying to determine
> a few different platforms we can possible use for conferencing and doing a
> little preliminary homework.
>
> Kindest regards,
>
> Joel
>
>
> ----- Original Message -----
> From: "Joel Sisko" <joel.sisko at iconverged.com>
> To: freeswitch-users at lists.freeswitch.org
> Sent: Thursday, February 11, 2010 4:05:58 PM GMT -08:00 US/Canada Pacific
> Subject: Re: [Freeswitch-users] Mod_Conference capacity....
>
> Brian/Gavin thanks for the input. But I agree with Brian, if were that easy
> I would have done it prior to the post.
>
> Just looking to find out what some of the communities success has been to
> see if this is a path we should go down for a conference solution platform.
>
> Joel
> ----- Original Message -----
> From: "Brian West" <brian at freeswitch.org>
> To: freeswitch-users at lists.freeswitch.org
> Sent: Thursday, February 11, 2010 3:47:46 PM GMT -08:00 US/Canada Pacific
> Subject: Re: [Freeswitch-users] Mod_Conference capacity....
>
> Not an optimal test scenario unless you know wtf you're doing!
>
> /b
>
> On Feb 11, 2010, at 5:22 PM, Gavin Henry wrote:
>
> > Why don't use script a test or use sipp and then dial in yourself to
> listen?
> >
> > Cheers.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100212/8f065cfd/attachment-0002.html 

From anthony.minessale at gmail.com  Fri Feb 12 09:07:03 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 12 Feb 2010 11:07:03 -0600
Subject: [Freeswitch-users] Play music to A leg.
In-Reply-To: <7d79b3931002112018y2512954cuca83ce21ff3406fe@mail.gmail.com>
References: <fe67a3341002090519r2cc6e8edm7702b8d012fc2be3@mail.gmail.com>
	<87f2f3b91002090737l4efa8en29bde9457fef7ea5@mail.gmail.com>
	<fe67a3341002092157x7484f3d2id6678510bbec66cc@mail.gmail.com>
	<87f2f3b91002100702u7581589ua866d2492a008b23@mail.gmail.com>
	<fe67a3341002110211w60390e1cm2d12fadd1e802778@mail.gmail.com>
	<87f2f3b91002110754laf9d01dxa94a36bc1b8c83df@mail.gmail.com>
	<191c3a031002110804m4c18e7e0y317f44272b42c8b4@mail.gmail.com>
	<7d79b3931002112018y2512954cuca83ce21ff3406fe@mail.gmail.com>
Message-ID: <191c3a031002120907l28fbdf2dgab5df7dd1b5a2f76@mail.gmail.com>

the script executes for everyone and gives them a chance to dial multiple
digits to test for, this is what he asked for, instead of 1 digit dial
multiple digits.  you set the correct string as a variable on the channel
and everybody runs the script and whoever dials the right digits wins the
rest will be hungup on.


On Thu, Feb 11, 2010 at 10:18 PM, lakshmanan ganapathy <lakindia89 at gmail.com
> wrote:

> Dear Antony,
> In bridge if we are making parallel calls, then group_confirm_key in
> execute mode will execute for all the extensions, and whomsoever finishes
> the script first, will be bridged.
>
> But I think nagalenoj need to execute the script for the extension which
> answers the call first, not for all the extension.!!!.
>
> From nanalenoj's post
>
>             " but when I do originate for multiple extensions, I want this
> script to work based on the answered extension."
>
>
> On Thu, Feb 11, 2010 at 9:34 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> group_confirm_key in execute mode can execute a lua script instead that
>> can read as many digits as you want and parse the results.
>>
>>
>>
>> On Thu, Feb 11, 2010 at 9:54 AM, Michael Collins <msc at freeswitch.org>wrote:
>>
>>> Hehe, this is getting more and more complicated. You may want to consider
>>> using the event socket and have your call control be done from a more 3rd
>>> party-ish perspective. If you've got all these different scenarios it might
>>> be better to let an external script do all the work.
>>>
>>> http://wiki.freeswitch.org/wiki/Event_Socket
>>>
>>> -MC
>>>
>>>
>>> On Thu, Feb 11, 2010 at 2:11 AM, Nagalenoj H. <nagalenoj at gmail.com>wrote:
>>>
>>>> But My scenario is,
>>>>    After I get the call from X.
>>>>    I answer the call in some scenarios and won't answer the call. So,
>>>> this leg can either be answered or unanswered.
>>>>    I originate a call to another number.
>>>>    After getting some digits from this originated leg.
>>>>    I do uuid_bridge of these 2 legs.
>>>>
>>>> I want to play some file[ringback] to leg A before bridging to B.
>>>>
>>>> On Wed, Feb 10, 2010 at 8:32 PM, Michael Collins <msc at freeswitch.org>wrote:
>>>>
>>>>>
>>>>>
>>>>> On Tue, Feb 9, 2010 at 9:57 PM, Nagalenoj H. <nagalenoj at gmail.com>wrote:
>>>>>
>>>>>> Because, I want to get some digits before bridging the legs. I've
>>>>>> tried group_confirm_key, but it accepts only one digit, I need multiple
>>>>>> digits, so I can't use.
>>>>>> I've also tried group_confirm_file, but when I do originate for
>>>>>> multiple extensions, I want this script to work based on the answered
>>>>>> extension.
>>>>>>
>>>>>> So, I've originated and processed the events to do my job.
>>>>>>
>>>>>> How do I play some music to A leg?
>>>>>>
>>>>>> I might be missing something, but couldn't you just park the call ("A
>>>>> leg") until you connect to the other party ("B leg") and then uuid_bridge at
>>>>> whatever point you want?
>>>>> -MC
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>>
>>>> --
>>>> Regards,
>>>> Nagalenoj H.
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100212/7f207adc/attachment-0002.html 

From msc at freeswitch.org  Fri Feb 12 09:22:01 2010
From: msc at freeswitch.org (Michael S Collins)
Date: Fri, 12 Feb 2010 10:22:01 -0700
Subject: [Freeswitch-users] transmission status of channel
In-Reply-To: <a44688a1002120732v7d95a6f9x2d4bd7ed68de59eb@mail.gmail.com>
References: <a44688a1002120732v7d95a6f9x2d4bd7ed68de59eb@mail.gmail.com>
Message-ID: <3028D6AC-46F2-4291-BAA8-E49C9347917B@freeswitch.org>

Check in the XML cdrs. I'm on a plan right now so I can't easily point  
you to a specific wiki page. :)

-MC

Sent from my iPhone

On Feb 12, 2010, at 8:32 AM, Woody Dickson <woodydickson at gmail.com>  
wrote:

> Hi,
>
> Is there anyway to obtain the %packet lost, latency, and jitter info
> for each channel?
>
> Any idea how to obtain those information?
>
> thx,
> Woody
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org



From joel.sisko at iconverged.com  Fri Feb 12 09:30:45 2010
From: joel.sisko at iconverged.com (Joel Sisko)
Date: Fri, 12 Feb 2010 11:30:45 -0600 (CST)
Subject: [Freeswitch-users] Mod_Conference capacity....
In-Reply-To: <191c3a031002120856q93e52cep14ee96f1e7ed6d1@mail.gmail.com>
Message-ID: <324899354.1084631265995845519.JavaMail.root@mail-2.01.com>

So if I understand correctly that there are no numbers to be considered from anyone at this point in regards to capacity as a general guideline? 

Kindest regards, 

Joel 
----- Original Message ----- 
From: "Anthony Minessale" <anthony.minessale at gmail.com> 
To: freeswitch-users at lists.freeswitch.org 
Sent: Friday, February 12, 2010 8:56:31 AM GMT -08:00 US/Canada Pacific 
Subject: Re: [Freeswitch-users] Mod_Conference capacity.... 

consider commercial support from consulting at freeswitch.org 



On Fri, Feb 12, 2010 at 10:33 AM, Joel Sisko < joel.sisko at iconverged.com > wrote: 


Brian/Gavin do you know of any resource/link that can give an indication on what we could be expected of conference capacity? We are trying to determine a few different platforms we can possible use for conferencing and doing a little preliminary homework. 

Kindest regards, 

Joel 



----- Original Message ----- 
From: "Joel Sisko" < joel.sisko at iconverged.com > 
To: freeswitch-users at lists.freeswitch.org 



Sent: Thursday, February 11, 2010 4:05:58 PM GMT -08:00 US/Canada Pacific 
Subject: Re: [Freeswitch-users] Mod_Conference capacity.... 

Brian/Gavin thanks for the input. But I agree with Brian, if were that easy I would have done it prior to the post. 

Just looking to find out what some of the communities success has been to see if this is a path we should go down for a conference solution platform. 

Joel 
----- Original Message ----- 
From: "Brian West" < brian at freeswitch.org > 
To: freeswitch-users at lists.freeswitch.org 
Sent: Thursday, February 11, 2010 3:47:46 PM GMT -08:00 US/Canada Pacific 
Subject: Re: [Freeswitch-users] Mod_Conference capacity.... 

Not an optimal test scenario unless you know wtf you're doing! 

/b 

On Feb 11, 2010, at 5:22 PM, Gavin Henry wrote: 

> Why don't use script a test or use sipp and then dial in yourself to listen? 
> 
> Cheers. 


_______________________________________________ 
FreeSWITCH-users mailing list 
FreeSWITCH-users at lists.freeswitch.org 
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users 
UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users 
http://www.freeswitch.org 

_______________________________________________ 
FreeSWITCH-users mailing list 
FreeSWITCH-users at lists.freeswitch.org 
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users 
UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users 
http://www.freeswitch.org 

_______________________________________________ 
FreeSWITCH-users mailing list 
FreeSWITCH-users at lists.freeswitch.org 
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users 
UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users 
http://www.freeswitch.org 



-- 
Anthony Minessale II 

FreeSWITCH http://www.freeswitch.org/ 
ClueCon http://www.cluecon.com/ 
Twitter: http://twitter.com/FreeSWITCH_wire 

AIM: anthm 
MSN:anthony_minessale at hotmail.com 
GTALK/JABBER/ PAYPAL:anthony.minessale at gmail.com 
IRC: irc.freenode.net #freeswitch 

FreeSWITCH Developer Conference 
sip:888 at conference.freeswitch.org 
iax:guest at conference.freeswitch.org/888 
googletalk:conf+888 at conference.freeswitch.org 
pstn:+19193869900 

_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org 
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100212/f499aaff/attachment-0002.html 

From anthony.minessale at gmail.com  Fri Feb 12 09:44:24 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 12 Feb 2010 11:44:24 -0600
Subject: [Freeswitch-users] Mod_Conference capacity....
In-Reply-To: <324899354.1084631265995845519.JavaMail.root@mail-2.01.com>
References: <191c3a031002120856q93e52cep14ee96f1e7ed6d1@mail.gmail.com>
	<324899354.1084631265995845519.JavaMail.root@mail-2.01.com>
Message-ID: <191c3a031002120944u346a3f7cq779f5c9e0ae31522@mail.gmail.com>

Producing benchmark numbers for an opensource project is a cardinal sin.
I can safely tell you that it is "many" and very competitive with anything
else you will encounter as long as you use a modern multi-core 64bit
machine.

As I said there is commercial support available which is customary for
anyone needing assistance setting up a company.


On Fri, Feb 12, 2010 at 11:30 AM, Joel Sisko <joel.sisko at iconverged.com>wrote:

> So if I understand correctly that there are no numbers to be considered
> from anyone at this point in regards to capacity as a general guideline?
>
>
> Kindest regards,
>
> Joel
> ----- Original Message -----
> From: "Anthony Minessale" <anthony.minessale at gmail.com>
> To: freeswitch-users at lists.freeswitch.org
> Sent: Friday, February 12, 2010 8:56:31 AM GMT -08:00 US/Canada Pacific
> Subject: Re: [Freeswitch-users] Mod_Conference capacity....
>
> consider commercial support from consulting at freeswitch.org
>
>
> On Fri, Feb 12, 2010 at 10:33 AM, Joel Sisko <joel.sisko at iconverged.com>wrote:
>
>> Brian/Gavin do you know of any resource/link that can give an indication
>> on what we could be expected of conference capacity? We are trying to
>> determine a few different platforms we can possible use for conferencing and
>> doing a little preliminary homework.
>>
>> Kindest regards,
>>
>> Joel
>>
>>
>> ----- Original Message -----
>> From: "Joel Sisko" <joel.sisko at iconverged.com>
>> To: freeswitch-users at lists.freeswitch.org
>> Sent: Thursday, February 11, 2010 4:05:58 PM GMT -08:00 US/Canada Pacific
>> Subject: Re: [Freeswitch-users] Mod_Conference capacity....
>>
>> Brian/Gavin thanks for the input. But I agree with Brian, if were that
>> easy I would have done it prior to the post.
>>
>> Just looking to find out what some of the communities success has been to
>> see if this is a path we should go down for a conference solution platform.
>>
>> Joel
>> ----- Original Message -----
>> From: "Brian West" <brian at freeswitch.org>
>> To: freeswitch-users at lists.freeswitch.org
>> Sent: Thursday, February 11, 2010 3:47:46 PM GMT -08:00 US/Canada Pacific
>> Subject: Re: [Freeswitch-users] Mod_Conference capacity....
>>
>> Not an optimal test scenario unless you know wtf you're doing!
>>
>> /b
>>
>> On Feb 11, 2010, at 5:22 PM, Gavin Henry wrote:
>>
>> > Why don't use script a test or use sipp and then dial in yourself to
>> listen?
>> >
>> > Cheers.
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________ FreeSWITCH-users mailing
> list FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100212/a8baa250/attachment-0002.html 

From max.bridgewater at gmail.com  Fri Feb 12 09:51:30 2010
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Fri, 12 Feb 2010 09:51:30 -0800
Subject: [Freeswitch-users] Choppy connection in one direction
Message-ID: <c55d46951002120951o324c16bwd8b0da5c02b30721@mail.gmail.com>

Hi Gents,

What can be wrong with my settings? Pleas help. I have a simple
Portech VoIP-GSM gateway in an African country where i try to
terminate calls. My setup is:

Skype<->Skypiax<->Portech Gateway<->GSM Phone

I have this setup in place right now. Calling from Europe, people can
understand me down there without a glitch. But I can barely hear what
they say.

I replaced the setup with: Skype  <->Skypiax<-> Skype. No change in result.

Then I tried with the Howler G729 codec:
Setup 1:  Xlite <-> Freeswitch <->Portech <->GSM Phone: No change in result.
Setup 2:  GSM Phone <-> Voip.MS <-> Freeswitch <->Portech <->GSM
Phone: No change in result.

In all these cases, the communication is perfect in one direction but
very choppy in the orther.
The only thing that works is direct Skype to Skype. And it works
perfectly, suggesting that the connection is probably not the issue.
The speed i could measure is 140kbs/30kbs.

And assuming the the internet connection is the issue, i would have
hoped that if Skype<->Skype works, then Skype<->Skypiax<->Skype would
also work.

Any idea?

Max.



From rupa at rupa.com  Fri Feb 12 09:52:13 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Fri, 12 Feb 2010 11:52:13 -0600
Subject: [Freeswitch-users] transmission status of channel
In-Reply-To: <3028D6AC-46F2-4291-BAA8-E49C9347917B@freeswitch.org>
References: <a44688a1002120732v7d95a6f9x2d4bd7ed68de59eb@mail.gmail.com>
	<3028D6AC-46F2-4291-BAA8-E49C9347917B@freeswitch.org>
Message-ID: <b39273531002120952u698ab5d1x799696b052ced9a9@mail.gmail.com>

I'm pretty sure that info doesn't exist.  Don't we need RTCP (plus
infrastructure for measuring) for this?

On Fri, Feb 12, 2010 at 11:22 AM, Michael S Collins <msc at freeswitch.org>wrote:

> Check in the XML cdrs. I'm on a plan right now so I can't easily point
> you to a specific wiki page. :)
>
> -MC
>
> Sent from my iPhone
>
> On Feb 12, 2010, at 8:32 AM, Woody Dickson <woodydickson at gmail.com>
> wrote:
>
> > Hi,
> >
> > Is there anyway to obtain the %packet lost, latency, and jitter info
> > for each channel?
> >
> > Any idea how to obtain those information?
> >
> > thx,
> > Woody
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> > users
> > http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
-Rupa
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100212/ce583762/attachment-0002.html 

From tim at novion.ru  Fri Feb 12 10:10:23 2010
From: tim at novion.ru (Timur Valishev)
Date: Fri, 12 Feb 2010 21:10:23 +0300
Subject: [Freeswitch-users] transmission status of channel
In-Reply-To: <b39273531002120952u698ab5d1x799696b052ced9a9@mail.gmail.com>
References: <a44688a1002120732v7d95a6f9x2d4bd7ed68de59eb@mail.gmail.com>
	<3028D6AC-46F2-4291-BAA8-E49C9347917B@freeswitch.org>
	<b39273531002120952u698ab5d1x799696b052ced9a9@mail.gmail.com>
Message-ID: <8e9d67561002121010p37b16aceyb06cda0420e1aa99@mail.gmail.com>

It would be very nice if FS pass RTCP information to channel vars...

Best regards,
Timur Valishev

2010/2/12 Rupa Schomaker <rupa at rupa.com>:
> I'm pretty sure that info doesn't exist. ?Don't we need RTCP (plus
> infrastructure for measuring) for this?
>
> On Fri, Feb 12, 2010 at 11:22 AM, Michael S Collins <msc at freeswitch.org>
> wrote:
>>
>> Check in the XML cdrs. I'm on a plan right now so I can't easily point
>> you to a specific wiki page. :)
>>
>> -MC
>>
>> Sent from my iPhone
>>
>> On Feb 12, 2010, at 8:32 AM, Woody Dickson <woodydickson at gmail.com>
>> wrote:
>>
>> > Hi,
>> >
>> > Is there anyway to obtain the %packet lost, latency, and jitter info
>> > for each channel?
>> >
>> > Any idea how to obtain those information?
>> >
>> > thx,
>> > Woody
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>> > users
>> > http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From joel.sisko at iconverged.com  Fri Feb 12 10:25:25 2010
From: joel.sisko at iconverged.com (joel.sisko at iconverged.com)
Date: Fri, 12 Feb 2010 12:25:25 -0600 (CST)
Subject: [Freeswitch-users] Mod_Conference capacity....
In-Reply-To: <892897350.1100311265999066927.JavaMail.root@mail-2.01.com>
Message-ID: <1235523542.1100531265999125823.JavaMail.root@mail-2.01.com>


Anthony thanks for the input. I am not looking for benchmark and understand the reasons why it does not make sense to do so since application usage and hardware will effect that benchmark. 

Looking for some successes by the group that they can share that would lead me to believe that using FreeSwitch would be worth the time and money to invest over our current conference solution. As an example, I can state that I have seen Yate used with 200 people in a single conference (all G711) work flawlessly, so does 200 count as "many" from your perspective or will FreeSwitch do that in its sleep? 

Thanks for the help. 

Joel 


----- Original Message ----- 
From: "Anthony Minessale" <anthony.minessale at gmail.com> 
To: freeswitch-users at lists.freeswitch.org 
Sent: Friday, February 12, 2010 9:44:24 AM GMT -08:00 US/Canada Pacific 
Subject: Re: [Freeswitch-users] Mod_Conference capacity.... 

Producing benchmark numbers for an opensource project is a cardinal sin. 
I can safely tell you that it is "many" and very competitive with anything else you will encounter as long as you use a modern multi-core 64bit machine. 

As I said there is commercial support available which is customary for anyone needing assistance setting up a company. 



On Fri, Feb 12, 2010 at 11:30 AM, Joel Sisko < joel.sisko at iconverged.com > wrote: 




So if I understand correctly that there are no numbers to be considered from anyone at this point in regards to capacity as a general guideline? 


Kindest regards, 

Joel 
----- Original Message ----- 

From: "Anthony Minessale" < anthony.minessale at gmail.com > 
To: freeswitch-users at lists.freeswitch.org 



Sent: Friday, February 12, 2010 8:56:31 AM GMT -08:00 US/Canada Pacific 
Subject: Re: [Freeswitch-users] Mod_Conference capacity.... 

consider commercial support from consulting at freeswitch.org 



On Fri, Feb 12, 2010 at 10:33 AM, Joel Sisko < joel.sisko at iconverged.com > wrote: 


Brian/Gavin do you know of any resource/link that can give an indication on what we could be expected of conference capacity? We are trying to determine a few different platforms we can possible use for conferencing and doing a little preliminary homework. 

Kindest regards, 

Joel 



----- Original Message ----- 
From: "Joel Sisko" < joel.sisko at iconverged.com > 
To: freeswitch-users at lists.freeswitch.org 



Sent: Thursday, February 11, 2010 4:05:58 PM GMT -08:00 US/Canada Pacific 
Subject: Re: [Freeswitch-users] Mod_Conference capacity.... 

Brian/Gavin thanks for the input. But I agree with Brian, if were that easy I would have done it prior to the post. 

Just looking to find out what some of the communities success has been to see if this is a path we should go down for a conference solution platform. 

Joel 
----- Original Message ----- 
From: "Brian West" < brian at freeswitch.org > 
To: freeswitch-users at lists.freeswitch.org 
Sent: Thursday, February 11, 2010 3:47:46 PM GMT -08:00 US/Canada Pacific 
Subject: Re: [Freeswitch-users] Mod_Conference capacity.... 

Not an optimal test scenario unless you know wtf you're doing! 

/b 

On Feb 11, 2010, at 5:22 PM, Gavin Henry wrote: 

> Why don't use script a test or use sipp and then dial in yourself to listen? 
> 
> Cheers. 


_______________________________________________ 
FreeSWITCH-users mailing list 
FreeSWITCH-users at lists.freeswitch.org 
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users 
UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users 
http://www.freeswitch.org 

_______________________________________________ 
FreeSWITCH-users mailing list 
FreeSWITCH-users at lists.freeswitch.org 
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users 
UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users 
http://www.freeswitch.org 

_______________________________________________ 
FreeSWITCH-users mailing list 
FreeSWITCH-users at lists.freeswitch.org 
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users 
UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users 
http://www.freeswitch.org 



-- 
Anthony Minessale II 

FreeSWITCH http://www.freeswitch.org/ 
ClueCon http://www.cluecon.com/ 
Twitter: http://twitter.com/FreeSWITCH_wire 

AIM: anthm 
MSN:anthony_minessale at hotmail.com 
GTALK/JABBER/ PAYPAL:anthony.minessale at gmail.com 
IRC: irc.freenode.net #freeswitch 

FreeSWITCH Developer Conference 
sip:888 at conference.freeswitch.org 
iax:guest at conference.freeswitch.org/888 
googletalk:conf+888 at conference.freeswitch.org 
pstn:+19193869900 

_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org 
_______________________________________________ 
FreeSWITCH-users mailing list 
FreeSWITCH-users at lists.freeswitch.org 
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users 
UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users 
http://www.freeswitch.org 




-- 
Anthony Minessale II 

FreeSWITCH http://www.freeswitch.org/ 
ClueCon http://www.cluecon.com/ 
Twitter: http://twitter.com/FreeSWITCH_wire 

AIM: anthm 
MSN:anthony_minessale at hotmail.com 
GTALK/JABBER/ PAYPAL:anthony.minessale at gmail.com 
IRC: irc.freenode.net #freeswitch 

FreeSWITCH Developer Conference 
sip:888 at conference.freeswitch.org 
iax:guest at conference.freeswitch.org/888 
googletalk:conf+888 at conference.freeswitch.org 
pstn:+19193869900 

_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org 
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100212/57a30aad/attachment-0002.html 

From gmaruzz at celliax.org  Fri Feb 12 11:10:06 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Fri, 12 Feb 2010 20:10:06 +0100
Subject: [Freeswitch-users] Choppy connection in one direction
In-Reply-To: <c55d46951002120951o324c16bwd8b0da5c02b30721@mail.gmail.com>
References: <c55d46951002120951o324c16bwd8b0da5c02b30721@mail.gmail.com>
Message-ID: <7b197bef1002121110v31bcbf0bmaf46c3b54cd6645a@mail.gmail.com>

it do not seems to be a problem of connection, seems a problem of
audio levels, I mean volumes...

Have I understood correctly?

-gm

On Fri, Feb 12, 2010 at 6:51 PM, Max Bridgewater
<max.bridgewater at gmail.com> wrote:
> Hi Gents,
>
> What can be wrong with my settings? Pleas help. I have a simple
> Portech VoIP-GSM gateway in an African country where i try to
> terminate calls. My setup is:
>
> Skype<->Skypiax<->Portech Gateway<->GSM Phone
>
> I have this setup in place right now. Calling from Europe, people can
> understand me down there without a glitch. But I can barely hear what
> they say.
>
> I replaced the setup with: Skype ?<->Skypiax<-> Skype. No change in result.
>
> Then I tried with the Howler G729 codec:
> Setup 1: ?Xlite <-> Freeswitch <->Portech <->GSM Phone: No change in result.
> Setup 2: ?GSM Phone <-> Voip.MS <-> Freeswitch <->Portech <->GSM
> Phone: No change in result.
>
> In all these cases, the communication is perfect in one direction but
> very choppy in the orther.
> The only thing that works is direct Skype to Skype. And it works
> perfectly, suggesting that the connection is probably not the issue.
> The speed i could measure is 140kbs/30kbs.
>
> And assuming the the internet connection is the issue, i would have
> hoped that if Skype<->Skype works, then Skype<->Skypiax<->Skype would
> also work.
>
> Any idea?
>
> Max.
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618



From max.bridgewater at gmail.com  Fri Feb 12 11:17:39 2010
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Fri, 12 Feb 2010 11:17:39 -0800
Subject: [Freeswitch-users] Choppy connection in one direction
In-Reply-To: <7b197bef1002121110v31bcbf0bmaf46c3b54cd6645a@mail.gmail.com>
References: <c55d46951002120951o324c16bwd8b0da5c02b30721@mail.gmail.com>
	<7b197bef1002121110v31bcbf0bmaf46c3b54cd6645a@mail.gmail.com>
Message-ID: <c55d46951002121117m67e24aa2y37c393161d318a93@mail.gmail.com>

Well, i can hear the other side but not understand. The sound is
complely choppy; sounding more like an alien trying to talk to me!

On Fri, Feb 12, 2010 at 11:10 AM, Giovanni Maruzzelli
<gmaruzz at celliax.org> wrote:
> it do not seems to be a problem of connection, seems a problem of
> audio levels, I mean volumes...
>
> Have I understood correctly?
>
> -gm
>
> On Fri, Feb 12, 2010 at 6:51 PM, Max Bridgewater
> <max.bridgewater at gmail.com> wrote:
>> Hi Gents,
>>
>> What can be wrong with my settings? Pleas help. I have a simple
>> Portech VoIP-GSM gateway in an African country where i try to
>> terminate calls. My setup is:
>>
>> Skype<->Skypiax<->Portech Gateway<->GSM Phone
>>
>> I have this setup in place right now. Calling from Europe, people can
>> understand me down there without a glitch. But I can barely hear what
>> they say.
>>
>> I replaced the setup with: Skype ?<->Skypiax<-> Skype. No change in result.
>>
>> Then I tried with the Howler G729 codec:
>> Setup 1: ?Xlite <-> Freeswitch <->Portech <->GSM Phone: No change in result.
>> Setup 2: ?GSM Phone <-> Voip.MS <-> Freeswitch <->Portech <->GSM
>> Phone: No change in result.
>>
>> In all these cases, the communication is perfect in one direction but
>> very choppy in the orther.
>> The only thing that works is direct Skype to Skype. And it works
>> perfectly, suggesting that the connection is probably not the issue.
>> The speed i could measure is 140kbs/30kbs.
>>
>> And assuming the the internet connection is the issue, i would have
>> hoped that if Skype<->Skype works, then Skype<->Skypiax<->Skype would
>> also work.
>>
>> Any idea?
>>
>> Max.
>>
>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>



From sos at sokhapkin.dyndns.org  Fri Feb 12 11:22:18 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Fri, 12 Feb 2010 14:22:18 -0500
Subject: [Freeswitch-users] bypass_media bug?
Message-ID: <201002121422.18544.sos@sokhapkin.dyndns.org>

Simple dialplan:

    <extension name="103">
      <condition field="destination_number" expression="^103$">
        <action application="set" data="continue_on_fail=true"/>
        <action application="set" data="bypass_media=true"/>
        <action application="bridge" data="sofia/cwu/103 at 192.168.1.254"/>
        <action application="bridge" data="sofia/cwu/104 at 192.168.1.254"/>
      </condition>
    </extension>

103 at 192.168.1.254 returns 183 early media and then "480 temporary 
unavailable", 104 at 192.168.1.254 answers the call (echo test).

When 104 answers, Freeswitch returns to caller in SDP media port from "183", 
but not media port from 104's "200 OK"

Is it a bug or expected behavior? If expected - is there a variable to control 
the behavior? Everything works OK if I replace bypass_media=true with 
bypass_media_after_bridge=true, but sending reinvites is not acceptable to 
me.



From gmaruzz at celliax.org  Fri Feb 12 11:40:07 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Fri, 12 Feb 2010 20:40:07 +0100
Subject: [Freeswitch-users] Choppy connection in one direction
In-Reply-To: <c55d46951002121117m67e24aa2y37c393161d318a93@mail.gmail.com>
References: <c55d46951002120951o324c16bwd8b0da5c02b30721@mail.gmail.com> 
	<7b197bef1002121110v31bcbf0bmaf46c3b54cd6645a@mail.gmail.com> 
	<c55d46951002121117m67e24aa2y37c393161d318a93@mail.gmail.com>
Message-ID: <7b197bef1002121140n1814c09fr70293e46ada06175@mail.gmail.com>

Please, don't talk to aliens!

And you have this issue in both cases, when you use skypiax and when
you don't use it?

I mean, you have this issue any times you put FS in the middle, huh?




On Fri, Feb 12, 2010 at 8:17 PM, Max Bridgewater
<max.bridgewater at gmail.com> wrote:
> Well, i can hear the other side but not understand. The sound is
> complely choppy; sounding more like an alien trying to talk to me!
>
> On Fri, Feb 12, 2010 at 11:10 AM, Giovanni Maruzzelli
> <gmaruzz at celliax.org> wrote:
>> it do not seems to be a problem of connection, seems a problem of
>> audio levels, I mean volumes...
>>
>> Have I understood correctly?
>>
>> -gm
>>
>> On Fri, Feb 12, 2010 at 6:51 PM, Max Bridgewater
>> <max.bridgewater at gmail.com> wrote:
>>> Hi Gents,
>>>
>>> What can be wrong with my settings? Pleas help. I have a simple
>>> Portech VoIP-GSM gateway in an African country where i try to
>>> terminate calls. My setup is:
>>>
>>> Skype<->Skypiax<->Portech Gateway<->GSM Phone
>>>
>>> I have this setup in place right now. Calling from Europe, people can
>>> understand me down there without a glitch. But I can barely hear what
>>> they say.
>>>
>>> I replaced the setup with: Skype ?<->Skypiax<-> Skype. No change in result.
>>>
>>> Then I tried with the Howler G729 codec:
>>> Setup 1: ?Xlite <-> Freeswitch <->Portech <->GSM Phone: No change in result.
>>> Setup 2: ?GSM Phone <-> Voip.MS <-> Freeswitch <->Portech <->GSM
>>> Phone: No change in result.
>>>
>>> In all these cases, the communication is perfect in one direction but
>>> very choppy in the orther.
>>> The only thing that works is direct Skype to Skype. And it works
>>> perfectly, suggesting that the connection is probably not the issue.
>>> The speed i could measure is 140kbs/30kbs.
>>>
>>> And assuming the the internet connection is the issue, i would have
>>> hoped that if Skype<->Skype works, then Skype<->Skypiax<->Skype would
>>> also work.
>>>
>>> Any idea?
>>>
>>> Max.
>>>
>>
>>
>>
>> --
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> Cell : +39-347-2665618
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618



From max.bridgewater at gmail.com  Fri Feb 12 12:01:54 2010
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Fri, 12 Feb 2010 12:01:54 -0800
Subject: [Freeswitch-users] Choppy connection in one direction
In-Reply-To: <7b197bef1002121140n1814c09fr70293e46ada06175@mail.gmail.com>
References: <c55d46951002120951o324c16bwd8b0da5c02b30721@mail.gmail.com>
	<7b197bef1002121110v31bcbf0bmaf46c3b54cd6645a@mail.gmail.com>
	<c55d46951002121117m67e24aa2y37c393161d318a93@mail.gmail.com>
	<7b197bef1002121140n1814c09fr70293e46ada06175@mail.gmail.com>
Message-ID: <c55d46951002121201x4a23565fxa44ec15e39c9a969@mail.gmail.com>

On Fri, Feb 12, 2010 at 11:40 AM, Giovanni Maruzzelli
<gmaruzz at celliax.org> wrote:
> Please, don't talk to aliens!
Why not? I could understand what they say, i would be at least curious
to see them.

>
> I mean, you have this issue any times you put FS in the middle, huh?

Yeah, it seems this appears only when FS is in the mix. Direct
connection to the Portech gateway is possible, but without G729 (or
something similar) on my softphones, it definitely sounds crappy.

Max.



From chrisg.lists at gmail.com  Fri Feb 12 12:14:26 2010
From: chrisg.lists at gmail.com (Chris Graham)
Date: Fri, 12 Feb 2010 22:14:26 +0200
Subject: [Freeswitch-users] Mod_IAX
Message-ID: <b8bb96081002121214y1b9ba3a3x6bcab6c53bce30c9@mail.gmail.com>

Hi All,

I have only recently gotten into freeswitch being a asterisk guy for 5 years
give or take. I love the freeswitch XML way of doing things. My question is
why was mod_iax dropped? Is opal the replacement? On the Wiki is says its
beta?

Thanks for the clarity.
Chris G.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100212/d4cfb18e/attachment-0002.html 

From anthony.minessale at gmail.com  Fri Feb 12 12:37:00 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 12 Feb 2010 14:37:00 -0600
Subject: [Freeswitch-users] Mod_IAX
In-Reply-To: <b8bb96081002121214y1b9ba3a3x6bcab6c53bce30c9@mail.gmail.com>
References: <b8bb96081002121214y1b9ba3a3x6bcab6c53bce30c9@mail.gmail.com>
Message-ID: <191c3a031002121237y5de4af7fxd971164c79e29d20@mail.gmail.com>

mod_iax fell into disuse and nobody volunteered when a call to maintain it
was put on the community.
The short answer is that the IAX2 protocol changed somewhere over time in a
way that made the IAX library we were using crash unexpectedly.  We don't
have the resources to debug the 3rd party code in that IAX lib so we dropped
it to protect FreeSWITCH from unwanted crashes.


On Fri, Feb 12, 2010 at 2:14 PM, Chris Graham <chrisg.lists at gmail.com>wrote:

> Hi All,
>
> I have only recently gotten into freeswitch being a asterisk guy for 5
> years give or take. I love the freeswitch XML way of doing things. My
> question is why was mod_iax dropped? Is opal the replacement? On the Wiki is
> says its beta?
>
> Thanks for the clarity.
> Chris G.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100212/a29d8e32/attachment-0002.html 

From peder at networkoblivion.com  Fri Feb 12 12:40:59 2010
From: peder at networkoblivion.com (Peder)
Date: Fri, 12 Feb 2010 14:40:59 -0600
Subject: [Freeswitch-users] Mod_IAX
In-Reply-To: <b8bb96081002121214y1b9ba3a3x6bcab6c53bce30c9@mail.gmail.com>
References: <b8bb96081002121214y1b9ba3a3x6bcab6c53bce30c9@mail.gmail.com>
Message-ID: <08dd01caac23$af4b9740$0de2c5c0$@com>

Lack of support.  Nobody wanted to take over development of it.

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Chris
Graham
Sent: Friday, February 12, 2010 2:14 PM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Mod_IAX

 

Hi All,

I have only recently gotten into freeswitch being a asterisk guy for 5 years
give or take. I love the freeswitch XML way of doing things. My question is
why was mod_iax dropped? Is opal the replacement? On the Wiki is says its
beta?

Thanks for the clarity.
Chris G.

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100212/d2061c32/attachment-0002.html 

From tculjaga at gmail.com  Fri Feb 12 12:46:11 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Fri, 12 Feb 2010 21:46:11 +0100
Subject: [Freeswitch-users] h323 - sip call is not working
In-Reply-To: <26C4E111-8329-48F8-A8DA-081B851A9514@freeswitch.org>
References: <20100212112427.0FC1F11F9E@mail.nstel.ru>
	<26C4E111-8329-48F8-A8DA-081B851A9514@freeswitch.org>
Message-ID: <65d96fc81002121246r48e867abp3c11f7f72a0ee906@mail.gmail.com>

On Fri, Feb 12, 2010 at 3:29 PM, Brian West <brian at freeswitch.org> wrote:

> This is a rather broad assumption.  I have seen 40ms, 60ms and even 80ms in
> the wild.  It all depends on what you want to do.  It lowers overhead and
> increases efficiency on the wire.
>
> /b
>
> On Feb 12, 2010, at 5:24 AM, Nikolay Kondratyev wrote:
>
> By the way, I know that one can use different packetization times for the
> same codec, but I?ve never heard, that somebody really uses 30 ms for
> G711Alaw. Always 20ms.
>
>
>
everything above 60 ms is a nonsense ... and ugly :) It screws your voice
quality not even thinking VBD (voice band data) over that line :).

Anyhow, Nikolay, your problem is broken IPO h323 stack and the know avaya
"flexibility" when interoping with other vendor equipments. Here IPO is
unable to negotiate a different framing size than the default and sadly this
is the core of the problem.

Please, can you send me two tcpdump captures of calls between IPO and FS:

   1. a capture with fast start & h245tunneling=true
   2. a captire with fast start & h245tunelling=true + h245inSetup


I just want to be sure of something.

T.



>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100212/82cdafaa/attachment-0002.html 

From tculjaga at gmail.com  Fri Feb 12 14:07:59 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Fri, 12 Feb 2010 23:07:59 +0100
Subject: [Freeswitch-users] Mod_Conference capacity....
In-Reply-To: <1235523542.1100531265999125823.JavaMail.root@mail-2.01.com>
References: <892897350.1100311265999066927.JavaMail.root@mail-2.01.com>
	<1235523542.1100531265999125823.JavaMail.root@mail-2.01.com>
Message-ID: <65d96fc81002121407l5f4cbd3cw8af46337a13520c2@mail.gmail.com>

On Fri, Feb 12, 2010 at 7:25 PM, <joel.sisko at iconverged.com> wrote:

> Anthony thanks for the input. I am not looking for benchmark and understand
> the reasons why it does not make sense to do so since application usage and
> hardware will effect that benchmark.
>
> Looking for some successes by the group that they can share that would lead
> me to believe that using FreeSwitch would be worth the time and money to
> invest over our current conference solution. As an example, I can state that
> I have seen Yate used with 200 people in a single conference (all G711) work
> flawlessly, so does 200 count as "many" from your perspective or will
> FreeSwitch do that in its sleep?
>
>
Again, what is your problem with SIPP ?

I can say i used it wery well and reached incredible numbers ... I didn't
try conference but i did other things:

1. calls to FS that answered the call and played some prompts (wav -
transcoding needed) - 200 CPS and 2000 sim calls (limited!)  - a quad core
64bit proc (unfortunately AMD but :P)
2. calls to FS acting as an AS returning 302 messages - 480 CPS including
LDAP BD lookup in the background - a dualcore 64bit proc intel

There is not rule of thumb that can tell you what performance can you get.
It realy depends of the application you are running, the dialplan you are
using,  what HW (RAM, CPU, HDD/ramDISK...) are you running FS on...



The only way you kan know all of this is hit the limit of your platform
yourself (SIPP is an excelent tool for that, just play with it for a day or
two).... or pay someone to make this test for you within your enviorment.

Chears!
T.




> Thanks for the help.
>
>
> Joel
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100212/fd09cf01/attachment-0002.html 

From gavin.henry at gmail.com  Fri Feb 12 14:25:19 2010
From: gavin.henry at gmail.com (Gavin Henry)
Date: Fri, 12 Feb 2010 22:25:19 +0000
Subject: [Freeswitch-users] Mod_Conference capacity....
In-Reply-To: <E7571BC2-135B-4743-B941-1A84968A4395@freeswitch.org>
References: <268119402.962161265928492648.JavaMail.root@mail-2.01.com>
	<715035894.962251265928530648.JavaMail.root@mail-2.01.com>
	<13ca621c1002111522v7fe55a62w84043ed986155b28@mail.gmail.com>
	<E7571BC2-135B-4743-B941-1A84968A4395@freeswitch.org>
Message-ID: <13ca621c1002121425s79d5e814h651aa7ce674e242a@mail.gmail.com>

Exactly, but he could setup his expected users via sipp and increase
participants whilst he is dialled in and wait until the audio gets
bad. A rough and ready test but give you some figures.

There's plenty of info if you search on the asterisks lists etc. Just
a thought, and yes you need to know wtf you're doing :-)

On 11/02/2010, Brian West <brian at freeswitch.org> wrote:
> Not an optimal test scenario unless you know wtf you're doing!
>
> /b
>
> On Feb 11, 2010, at 5:22 PM, Gavin Henry wrote:
>
>> Why don't use script a test or use sipp and then dial in yourself to
>> listen?
>>
>> Cheers.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>

-- 
Sent from my mobile device

http://www.suretecsystems.com/services/openldap/
http://www.suretectelecom.com



From gavin.henry at gmail.com  Fri Feb 12 14:36:56 2010
From: gavin.henry at gmail.com (Gavin Henry)
Date: Fri, 12 Feb 2010 22:36:56 +0000
Subject: [Freeswitch-users] Mod_Conference capacity....
In-Reply-To: <191c3a031002120944u346a3f7cq779f5c9e0ae31522@mail.gmail.com>
References: <191c3a031002120856q93e52cep14ee96f1e7ed6d1@mail.gmail.com>
	<324899354.1084631265995845519.JavaMail.root@mail-2.01.com>
	<191c3a031002120944u346a3f7cq779f5c9e0ae31522@mail.gmail.com>
Message-ID: <13ca621c1002121436n9b44dcexffeaafe998d4c88a@mail.gmail.com>

Hi,

I think this is unfair to automatically point a user to commercial
support. Its like holding your users hostage until they pay for info.

I'm willing to write a page on the wiki with info on the recommended
tools to use etc. with links to the commercial support resource list
if they can't be bothered to do the setup themselves or want the
experts paid for guidance.

It just reads like a bit of an insult. Have a question? Well you must
be stupid and need to pay for help immediately.

He may have even written the page for the docs team if he was
encouraged first with a few pointers on how to do the tests and
written them up with results for others to find with a big YMMV
warning.

Just my thoughts from experiences as the Doc team lead for the
OpenLDAP project and dealing with lots of users asking questions too.

But some users never read docs and there a very good amount of high
quality pages in the wiki. Others, if helped, come back and
contribute, but you may have lost this one.

Gav.

On 12/02/2010, Anthony Minessale <anthony.minessale at gmail.com> wrote:
> Producing benchmark numbers for an opensource project is a cardinal sin.
> I can safely tell you that it is "many" and very competitive with anything
> else you will encounter as long as you use a modern multi-core 64bit
> machine.
>
> As I said there is commercial support available which is customary for
> anyone needing assistance setting up a company.
>
>
> On Fri, Feb 12, 2010 at 11:30 AM, Joel Sisko
> <joel.sisko at iconverged.com>wrote:
>
>> So if I understand correctly that there are no numbers to be considered
>> from anyone at this point in regards to capacity as a general guideline?
>>
>>
>> Kindest regards,
>>
>> Joel
>> ----- Original Message -----
>> From: "Anthony Minessale" <anthony.minessale at gmail.com>
>> To: freeswitch-users at lists.freeswitch.org
>> Sent: Friday, February 12, 2010 8:56:31 AM GMT -08:00 US/Canada Pacific
>> Subject: Re: [Freeswitch-users] Mod_Conference capacity....
>>
>> consider commercial support from consulting at freeswitch.org
>>
>>
>> On Fri, Feb 12, 2010 at 10:33 AM, Joel Sisko
>> <joel.sisko at iconverged.com>wrote:
>>
>>> Brian/Gavin do you know of any resource/link that can give an indication
>>> on what we could be expected of conference capacity? We are trying to
>>> determine a few different platforms we can possible use for conferencing
>>> and
>>> doing a little preliminary homework.
>>>
>>> Kindest regards,
>>>
>>> Joel
>>>
>>>
>>> ----- Original Message -----
>>> From: "Joel Sisko" <joel.sisko at iconverged.com>
>>> To: freeswitch-users at lists.freeswitch.org
>>> Sent: Thursday, February 11, 2010 4:05:58 PM GMT -08:00 US/Canada Pacific
>>> Subject: Re: [Freeswitch-users] Mod_Conference capacity....
>>>
>>> Brian/Gavin thanks for the input. But I agree with Brian, if were that
>>> easy I would have done it prior to the post.
>>>
>>> Just looking to find out what some of the communities success has been to
>>> see if this is a path we should go down for a conference solution
>>> platform.
>>>
>>> Joel
>>> ----- Original Message -----
>>> From: "Brian West" <brian at freeswitch.org>
>>> To: freeswitch-users at lists.freeswitch.org
>>> Sent: Thursday, February 11, 2010 3:47:46 PM GMT -08:00 US/Canada Pacific
>>> Subject: Re: [Freeswitch-users] Mod_Conference capacity....
>>>
>>> Not an optimal test scenario unless you know wtf you're doing!
>>>
>>> /b
>>>
>>> On Feb 11, 2010, at 5:22 PM, Gavin Henry wrote:
>>>
>>> > Why don't use script a test or use sipp and then dial in yourself to
>>> listen?
>>> >
>>> > Cheers.
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>> _______________________________________________ FreeSWITCH-users mailing
>> list FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>

-- 
Sent from my mobile device

http://www.suretecsystems.com/services/openldap/
http://www.suretectelecom.com



From joseph.puchalski at personalcyberspace.com  Fri Feb 12 15:35:04 2010
From: joseph.puchalski at personalcyberspace.com (Joseph Puchalski)
Date: Fri, 12 Feb 2010 23:35:04 +0000
Subject: [Freeswitch-users] How to set outbound caller id info for multiple
	users/extensions
Message-ID: <093DD565390C1E4FB15D7B383E86BB05AF05AB@Goose.personalcyberspace.net>

I'm having problems setting different outbound caller id info for different extensions/users.

I've set up a small system with two active users. I set up my users by copying and modifying existing entries from the dialplan files that come with freeSWITCH

Here are my two extensions:

<include>
  <user id="5859">
    <params>
      <param name="password" value="xxxxxx"/>
      <param name="vm-password" value="xxxxxx"/>
    </params>
    <variables>
      <variable name="toll_allow" value="domestic,international,local"/>
      <variable name="accountcode" value="5859"/>
      <variable name="user_context" value="default"/>
      <variable name="effective_caller_id_name" value="5859"/>
      <variable name="effective_caller_id_number" value="5859"/>
      <variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/>
      <variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/>
      <variable name="callgroup" value="techsupport"/>
    </variables>
  </user>
</include>

<include>
  <user id="5515">
    <params>
      <param name="password" value="xxxxxxx"/>
      <param name="vm-password" value="xxxxxxx"/>
    </params>
    <variables>
      <variable name="toll_allow" value="domestic,international,local"/>
      <variable name="accountcode" value="5515"/>
      <variable name="user_context" value="default"/>
      <variable name="effective_caller_id_name" value="5515"/>
      <variable name="effective_caller_id_number" value="5515"/>
      <variable name="outbound_caller_id_name" value="Jane Doe"/>
      <variable name="outbound_caller_id_number" value="5558395515"/>
      <variable name="callgroup" value="techsupport"/>
    </variables>
  </user>
</include>

These extensions are in files named 5859.xml and 5515.xml respectively.

I'm using a SIP trunk from Vitelity (in and out) with two DIDs corresponding to the extensions above.

Inbound and outbound calling work as needed with one exception: Calls originating from user/extension 5515 go out with the caller ID of extension/user 5859.

Extension 5859 was the first that I created.

Where should I be setting the outbound caller id number for my second extension? I've been trying to track this down in the available documentation but have been unable to do so.

I apologize ahead of time if this is answered somewhere obvious that I've missed.

Thanks for any help.

Joe (FreeSWITCH newbie)
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100212/c836abc9/attachment-0002.html 

From mcampbellsmith at gmail.com  Fri Feb 12 16:29:25 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Sat, 13 Feb 2010 11:29:25 +1100
Subject: [Freeswitch-users] Forbidden using UDP, works with TCP/TLS
In-Reply-To: <0A905834-850F-4D1F-AEC5-D3A87CE4141B@jerris.com>
References: <33c87fa31002101824h21e3bf7lb13b68e8d4fd36a2@mail.gmail.com>
	<2EF27EFC-1634-46A3-B438-2EE99BEF58F1@jerris.com>
	<33c87fa31002110131w3cf6ad00n802669856085a879@mail.gmail.com>
	<CA21231C-108B-4445-A3B9-BF8D94FFE898@jerris.com>
	<33c87fa31002110213q1de0eebfsb950d8725ea9aba@mail.gmail.com>
	<0A905834-850F-4D1F-AEC5-D3A87CE4141B@jerris.com>
Message-ID: <33c87fa31002121629k74f26807i131841aa4252461b@mail.gmail.com>

This is the working one with TLS... does this shed any light on why
this user can not register using UDP?

   ------------------------------------------------------------------------
   REGISTER sip:mydns.dyndns.org:442 SIP/2.0
   Via: SIP/2.0/TLS 121.xxx.xxx.xxx:10371;branch=z9hG4bK-9465a0f9;rport
   From: 2000 <sip:2000 at mydns.dyndns.org:442>;tag=34ac954c7d2abf51o0
   To: 2000 <sip:2000 at mydns.dyndns.org:442>
   Call-ID: 8657c383-fea70d0a at 10.0.0.1
   CSeq: 31055 REGISTER
   Max-Forwards: 70
   Authorization: Digest
username="2000",realm="mydns.dyndns.org",nonce="5000f49a-1836-11df-985e-e77ba7a22ac3",uri="sip:mydns.dyndns.org:442",algorithm=MD5,response="2bad1d5fadbb0465b0a513352db0292b",qop=auth,nc=00000001,cnonce="6951ee10"
   Contact: 2000 <sip:2000 at 121.xxx.xxx.xxx:10371;transport=tls>;expires=1800
   User-Agent: Linksys/SPA3102-5.1.10(GW)
   Content-Length: 0
   Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
   Supported: x-sipura, replaces

   ------------------------------------------------------------------------
tport_deliver(0xb6d07cd8): msg 0xb6da3cc0 (795 bytes) from
tls/121.xxx.xxx.xxx:10371/sips next=(nil)
nta: received REGISTER sip:mydns.dyndns.org:442 SIP/2.0 (CSeq 31055)
nta: REGISTER (31055) going to a default leg
nua: nua_stack_process_request: entering
nua: nh_create: entering
nua: nh_create_handle: entering
nua: nua_stack_set_params: entering
soa_clone(static::0x97cc698, 0x9794808, 0xb6d83680) called
soa_set_params(static::0xb6d77020, ...) called
nua(0xb6d83680): event i_register 100 Trying
nua: nua_application_event: entering
nua: nua_respond: entering
nua(0xb6d83680): sent signal r_respond
nua: nua_handle_destroy: entering
nua(0xb6d83680): sent signal r_destroy
nua: nua_handle_magic: entering
nua: nua_handle_destroy: entering
tport(0xb6d07cd8): reset timer
nua(0xb6d83680): recv signal r_respond 200 OK
nua: nua_stack_set_params: entering
soa_set_params(static::0xb6d77020, ...) called
tport_tsend(0xb6d07cd8) tpn = TLS/121.xxx.xxx.xxx:10371
tport_tls_writevec: vec 0xb6d2eb80 0xb6da4270 92 (92)
tport_tls_writevec: vec 0xb6d2eb80 0xb6ded80a 76 (76)
tport_tls_writevec: vec 0xb6d2eb80 0xb6da42cc 69 (69)
tport_tls_writevec: vec 0xb6d2eb80 0xb6ded889 59 (59)
tport_tls_writevec: vec 0xb6d2eb80 0xb6da4311 329 (329)
tport_vsend(0xb6d07cd8): 625 bytes of 625 to tls/121.xxx.xxx.xxx:10371
tport_vsend returned 625
send 625 bytes to tls/[121.xxx.xxx.xxx]:10371 at 00:25:39.795356:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK

On Fri, Feb 12, 2010 at 9:22 AM, Michael Jerris <mike at jerris.com> wrote:
> how is this different from the working one?
>
> Mike
>
> On Feb 11, 2010, at 5:13 AM, Mark Campbell-Smith wrote:
>
>> ah thats true... The trace is not too readable to me, but may give
>> some insight to someone that can read the sofia logs....
>>
>>
>> recv 752 bytes from udp/[121.xxx.xxx.xxx]:5060 at 10:10:37.803288:
>> ? ------------------------------------------------------------------------
>> ? REGISTER sip:mydns.dyndns.org SIP/2.0
>> ? Via: SIP/2.0/UDP 121.xxx.xxx.xxx:5060;branch=z9hG4bK-9052c91f
>> ? From: 2000 <sip:2000 at mydns.dyndns.org>;tag=7a9dbbbfa691136do0
>> ? To: 2000 <sip:2000 at mydns.dyndns.org>
>> ? Call-ID: 610db38-dd3b511f at 121.xxx.xxx.xxx
>> ? CSeq: 32330 REGISTER
>> ? Contact: 2000 <sip:2000 at 121.xxx.xxx.xxx:5060>;expires=900
>> ? Authorization: Digest username="2000", realm="mydns.dyndns.org",
>> nonce="b3298cfe-16f5-11df-9734-e77ba7a22ac3",
>> uri="sip:mydns.dyndns.org",
>> response="724fca542ce08d3f12b9ba1043bebb0c", algorithm=MD5,
>> qop="1225e2f1"
>> ? Max-Forwards: 70
>> ? User-Agent: Linksys/SPA3102-5.1.10(GW)
>> ? Supported: x-sipura
>> ? Supported: replaces
>> ? Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
>> ? Content-Length: 0
>>
>>
>> ? ------------------------------------------------------------------------
>> tport_deliver(0x97cde80): msg 0x98297e8 (752 bytes) from
>> udp/121.xxx.xxx.xxx:5060/sip next=(nil)
>> nta: received REGISTER sip:mydns.dyndns.org SIP/2.0 (CSeq 32330)
>> nta: REGISTER (32330) going to a default leg
>> nua: nua_stack_process_request: entering
>> nua: nh_create: entering
>> nua: nh_create_handle: entering
>> nua: nua_stack_set_params: entering
>> soa_clone(static::0x97cc698, 0x9794808, 0x981cb70) called
>> soa_set_params(static::0x9758ba8, ...) called
>> nua(0x981cb70): event i_register 100 Trying
>> nua: nua_application_event: entering
>> nua: nua_respond: entering
>> nua(0x981cb70): sent signal r_respond
>> nua: nua_handle_destroy: entering
>> nua(0x981cb70): sent signal r_destroy
>> nua: nua_handle_magic: entering
>> nua: nua_handle_destroy: entering
>> nua(0x981cb70): recv signal r_respond 403 Forbidden
>> nua: nua_stack_set_params: entering
>> soa_set_params(static::0x9758ba8, ...) called
>> tport_tsend(0x97cde80) tpn = UDP/121.xxx.xxx.xxx:5060
>> tport_resolve addrinfo = 121.xxx.xxx.xxx:5060
>> tport_by_addrinfo(0x97cde80): not found by name UDP/121.xxx.xxx.xxx:5060
>> tport_vsend(0x97cde80): 495 bytes of 495 to udp/121.xxx.xxx.xxx:5060
>> tport_vsend returned 495
>> send 495 bytes to udp/[121.xxx.xxx.xxx]:5060 at 10:10:37.812955:
>> ? ------------------------------------------------------------------------
>> ? SIP/2.0 403 Forbidden
>>
>> On Thu, Feb 11, 2010 at 8:57 PM, Michael Jerris <mike at jerris.com> wrote:
>>> you can crank up the sofia loglevel as well
>>>
>>> Mike
>>>
>>> On Feb 11, 2010, at 4:31 AM, Mark Campbell-Smith wrote:
>>>
>>>> I don't see debug logs (fsctl debug_level 9 or loglevel 9) during the
>>>> registration process.
>>>>
>>>> All I see is the sip messages when the sip trace is activated (403 Forbidden)
>>>>
>>>> Is there other debugging that I can enable?
>>>>
>>>> On Thu, Feb 11, 2010 at 4:25 PM, Michael Jerris <mike at jerris.com> wrote:
>>>>> Look at the FreeSWITCH debug logs and compare the differences, they should tell you why.
>>>>>
>>>>> Mike
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From wiltingtree at gmail.com  Fri Feb 12 19:04:01 2010
From: wiltingtree at gmail.com (Adam Wilt)
Date: Fri, 12 Feb 2010 22:04:01 -0500
Subject: [Freeswitch-users] Bridging sessions from two separate lua
	scripts; uuid_bridge? intercept?
In-Reply-To: <D753EDBD-6ED7-43C2-B5F5-A899E3C3D3E2@jerris.com>
References: <eb7797d71002071831w27301c5dva767b82b38c2ce6f@mail.gmail.com>
	<C23AA196-E851-4D5B-B1B5-2C2F32F6BB68@freeswitch.org>
	<eb7797d71002071929pb53e1adue2e91e6bdb598251@mail.gmail.com>
	<eb7797d71002080737k3f8fa5c9m8bf3ef2fcf42881e@mail.gmail.com>
	<D753EDBD-6ED7-43C2-B5F5-A899E3C3D3E2@jerris.com>
Message-ID: <eb7797d71002121904v5bdfac87w2684c461ca1770cc@mail.gmail.com>

Thanks again for the help Michael.

I'm now upgraded to version 1.5, but I'm still getting the same
problem. When I try to bridge sessions from two separate lua scripts,
both sessions hang up on me. I think maybe I don't understand how
"intercept" works.
Anyway, I posted the debug trace here:

                      http://pastebin.freeswitch.org/12121

And I also put together a small example which exhibits the problem.
The first script is started by an inbound call and starts the second
script. The second script places an outbound call and tries to bridge
the two sessions together:

Inbound script:  http://pastebin.freeswitch.org/12122
Outbound script: http://pastebin.freeswitch.org/12123

Thanks,
Adam




On 2/9/10, Michael Jerris <mike at jerris.com> wrote:
> 1.4?  how does the future look, report back?
>
> http://files-sync.freeswitch.org/windows_installer/freepbx_svn.exe
>
> I think this has latest FreeSWITCH in it to, Carlos, can you confirm that?
>
> Mike
>
> On Feb 8, 2010, at 10:37 AM, Adam Wilt wrote:
>
>> One other thing I should mention. I'm running FreeSWITCH version 1.4
>> (build 14460) in Windows.
>> Brian suggested I upgrade to the build in the
>> http://files-sync.freeswitch.org/windows_installer/  folder, but it turned
>> out to be the exact same build I already had. I'd love to try upgrade to
>> 1.5 in case this problem has been fixed already.
>>
>>
>> On Sun, Feb 7, 2010 at 10:29 PM, Adam Wilt <wiltingtree at gmail.com> wrote:
>> Thanks Michael for the reply.
>> Here's the pastebin link: http://pastebin.freeswitch.org/12084
>>
>>
>> On Sun, Feb 7, 2010 at 9:50 PM, Michael S Collins <msc at freeswitch.org>
>> wrote:
>> Pastebin a debug log so we can see what is happening when the script runs.
>>
>>
>> -MC
>>
>> Sent from my iPhone
>>
>> On Feb 7, 2010, at 8:31 PM, Adam Wilt <wiltingtree at gmail.com> wrote:
>>
>>> Hi. I have two sessions running in two separate Lua scripts, and I want
>>> to bridge them so that the bridged call is being controlled by the first
>>> (a-leg) script.
>>> If I simply use uuid_bridge, I get no error but the calls don't bridge.
>>> I've tried intercept, but I don't understand how it should be used;
>>> nothing I try seems to work.
>>> Here's what I have:
>>>
>>> function bridge_calls(session,api,b_leg_uuid, call_len)
>>>   session:setAutoHangup(false)
>>>   session:execute("sched_hangup","+" .. tostring(call_len) .. " " ..
>>> tostring(session.uuid))
>>>   session:execute("set","continue_on_fail=true")
>>>   api:executeString("intercept -bleg " .. tostring(b_leg_uuid))
>>>   api:executeString("uuid_bridge " .. tostring(session.uuid) .. " " ..
>>> tostring(b_leg_uuid))
>>> end
>>>
>>> I'd really appreciate any help.
>>>
>>> Thanks,
>>> Adam
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>



From jason at jasonjgw.net  Fri Feb 12 19:08:12 2010
From: jason at jasonjgw.net (Jason White)
Date: Sat, 13 Feb 2010 14:08:12 +1100
Subject: [Freeswitch-users] FreeSWITCH Debian packages - build issues
Message-ID: <20100213030812.GA19108@jdc.jasonjgw.net>

Has anyone successfully built the Debian packages recently from the source
repository?

The problem I'm experiencing is that openzap is specified to be built, but it
is never actually compiled. Consequently, the packages can't be created (the
process fails due to the missing mod_openzap.so file).

I don't need openzap; I can easily comment it out, but I also think the
supplied package files should work as is.

first step: confirm whether my experience under Debian Sid is shared by others
using different versions of Debian or Ubuntu.




From woodydickson at gmail.com  Fri Feb 12 23:14:48 2010
From: woodydickson at gmail.com (Woody Dickson)
Date: Fri, 12 Feb 2010 23:14:48 -0800
Subject: [Freeswitch-users] transmission status of channel
In-Reply-To: <8e9d67561002121010p37b16aceyb06cda0420e1aa99@mail.gmail.com>
References: <a44688a1002120732v7d95a6f9x2d4bd7ed68de59eb@mail.gmail.com>
	<3028D6AC-46F2-4291-BAA8-E49C9347917B@freeswitch.org>
	<b39273531002120952u698ab5d1x799696b052ced9a9@mail.gmail.com>
	<8e9d67561002121010p37b16aceyb06cda0420e1aa99@mail.gmail.com>
Message-ID: <a44688a1002122314t4060f29ehce920077ee90704b@mail.gmail.com>

Does anyone know where to find those RTCP info from the core rtp stack?

On Fri, Feb 12, 2010 at 10:10 AM, Timur Valishev <tim at novion.ru> wrote:
> It would be very nice if FS pass RTCP information to channel vars...
>
> Best regards,
> Timur Valishev
>
> 2010/2/12 Rupa Schomaker <rupa at rupa.com>:
>> I'm pretty sure that info doesn't exist. ?Don't we need RTCP (plus
>> infrastructure for measuring) for this?
>>
>> On Fri, Feb 12, 2010 at 11:22 AM, Michael S Collins <msc at freeswitch.org>
>> wrote:
>>>
>>> Check in the XML cdrs. I'm on a plan right now so I can't easily point
>>> you to a specific wiki page. :)
>>>
>>> -MC
>>>
>>> Sent from my iPhone
>>>
>>> On Feb 12, 2010, at 8:32 AM, Woody Dickson <woodydickson at gmail.com>
>>> wrote:
>>>
>>> > Hi,
>>> >
>>> > Is there anyway to obtain the %packet lost, latency, and jitter info
>>> > for each channel?
>>> >
>>> > Any idea how to obtain those information?
>>> >
>>> > thx,
>>> > Woody
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>>> > users
>>> > http://www.freeswitch.org
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>>
>> --
>> -Rupa
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From mailinglist at fribert.dk  Sat Feb 13 00:54:30 2010
From: mailinglist at fribert.dk (mailinglist)
Date: Sat, 13 Feb 2010 09:54:30 +0100
Subject: [Freeswitch-users] Svar: Re:  Need help setting up a feature
In-Reply-To: <b39273531002110528p42218d07q95e97485139315f1@mail.gmail.com>
References: <4B73A799020000E100000470@mail.fribert.dk>
	<b39273531002110528p42218d07q95e97485139315f1@mail.gmail.com>
Message-ID: <4B7676D6020000E100000481@mail.fribert.dk>

Hi Rupa
 
I've got similar settings here, but I can't make them work, nothing happens when I press R:
 
The settings are
DTMP over VoIP connections: Send settings [ ]Auto   [X] Audio  [ ] RFC 2833 [X] SIP Info
 
Call Transfer
Use the R key to initiate call     ( ) Yes   (X) No
transfer with the SIP Refer Method
 
Transfer Call by On-Hook        (X) Yes   ( ) No
 
Derive Target address             ( ) from SIP URL   (X)   from SIP Contact Header
 
Find Target address automatically    ( ) Yes   (X) No
 
Hook Flash  (R-key)
 
Application Type:   dtmf-relay
Application Signal: 16
 
That's all the settings, I had to set 'send settings' from auto to audio+SIP Info otherwise the R-Flash settings were disabled.

 
But all in all, I think a *1 or something during the call would be a better method, because I should be able to make that work on al types of phones, right?
 
 
Best regards
Fribse
 
/Running Freeswitch on pfSense running on a VMWare ESXi 4.0

>>> 11-02-2010 kl. 14:28 skrev Rupa Schomaker <rupa at rupa.com> i meddelelsen <b39273531002110528p42218d07q95e97485139315f1 at mail.gmail.com>:

My Siemens A580 has options for controlling the R key. It seems that you can either have it setup for transfer or as a hook flash. Default is as a transfer key.

I haven't succeeded in getting it to work for transfer and it is wayyyyy down low on my list of things to do with the phone.

>From the web UI:


Call Transfer
Use the R key to initiate call transfer with the SIP Refer method.:Yes No 
Transfer Call by On-Hook:Yes No 
Derive target address:from SIP URL from SIP contact header 
Find target addr. automatically:Yes No 

Hook Flash (R-key)
R key settings are disabled because the R key is being used for call transfer. 

On Wed, Feb 10, 2010 at 11:45 PM, mailinglist <mailinglist at fribert.dk> wrote:


Sorry for the repost, but the previous thread just died :-)
I'm trying to get the possibility of transfering an incoming call from one extension to another, and give the possibility of turning it into a conference.
I don't have a 'transfer' button.
I do have an 'R' button on the Siemens handsets, and a 'Flash' button on the Sipura. The 'Flash' button gives me a new dialtone, gives the caller MOH, and then I can dial the new extension, and transfer the call, but not create a conference.
But the Siemens handset does not have a 'flash', and pressing the R doesn't do anything.
It might be two different features 'transfer' and 'conference'...

But I thought that using the bind_meta_app would accomplish both.
It's on an incoming call from the outside.
So the situation:
The Public folder has an entry that matches the dialed number, and does a transfer to 8202.
Then the dialplan matches the 8202 with a group, and the phone rings.
Somebody picks it up, finds out that it needs to be transferred to another extension, or transferred to a conference with a second extension.
How do I construct that?
Best regards
Fribse
/Running Freeswitch on pfSense running on a VMWare ESXi 4.0

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org





-- 
-Rupa
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100213/2df30d3b/attachment-0002.html 

From errotan at gmail.com  Sat Feb 13 00:57:11 2010
From: errotan at gmail.com (=?iso-8859-1?q?Pusk=E1s_Zsolt?=)
Date: Sat, 13 Feb 2010 09:57:11 +0100
Subject: [Freeswitch-users] FreeSWITCH Debian packages - build issues
In-Reply-To: <20100213030812.GA19108@jdc.jasonjgw.net>
References: <20100213030812.GA19108@jdc.jasonjgw.net>
Message-ID: <201002130957.11633.errotan@gmail.com>

2010. febru?r 13. 04.08.12 Jason White d?tummal ezt ?rta:
> Has anyone successfully built the Debian packages recently from the source
> repository?
> 
> The problem I'm experiencing is that openzap is specified to be built, but
>  it is never actually compiled. Consequently, the packages can't be created
>  (the process fails due to the missing mod_openzap.so file).
> 
> I don't need openzap; I can easily comment it out, but I also think the
> supplied package files should work as is.
> 
> first step: confirm whether my experience under Debian Sid is shared by
>  others using different versions of Debian or Ubuntu.
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 

Openzap compiles without errors on Debian "testing" amd64 for me.
Svn version: 16625 



From jason at jasonjgw.net  Sat Feb 13 01:09:07 2010
From: jason at jasonjgw.net (Jason White)
Date: Sat, 13 Feb 2010 20:09:07 +1100
Subject: [Freeswitch-users] FreeSWITCH Debian packages - build issues
In-Reply-To: <201002130957.11633.errotan@gmail.com>
References: <20100213030812.GA19108@jdc.jasonjgw.net>
	<201002130957.11633.errotan@gmail.com>
Message-ID: <20100213090907.GA29452@jdc.jasonjgw.net>

Pusk?s Zsolt <errotan at gmail.com> wrote:
 
> Openzap compiles without errors on Debian "testing" amd64 for me.
> Svn version: 16625 

Thanks. Was that with the package build?

It seems I have local modifications here that I'd forgotten about. I'll
investigate further.




From errotan at gmail.com  Sat Feb 13 02:09:35 2010
From: errotan at gmail.com (=?iso-8859-1?q?Pusk=E1s_Zsolt?=)
Date: Sat, 13 Feb 2010 11:09:35 +0100
Subject: [Freeswitch-users] FreeSWITCH Debian packages - build issues
In-Reply-To: <20100213090907.GA29452@jdc.jasonjgw.net>
References: <20100213030812.GA19108@jdc.jasonjgw.net>
	<201002130957.11633.errotan@gmail.com>
	<20100213090907.GA29452@jdc.jasonjgw.net>
Message-ID: <201002131109.35877.errotan@gmail.com>

2010. febru?r 13. 10.09.07 Jason White d?tummal ezt ?rta:
> Pusk?s Zsolt <errotan at gmail.com> wrote:
> > Openzap compiles without errors on Debian "testing" amd64 for me.
> > Svn version: 16625
> 
> Thanks. Was that with the package build?
> 
> It seems I have local modifications here that I'd forgotten about. I'll
> investigate further.
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 

I just compiled fs using defaults (just uncommented the openzap line in 
modues.conf). I don't know how to build a package for that but I can try if 
you got some instructions how to do that for testing purposes.



From vetali100 at gmail.com  Sat Feb 13 02:47:51 2010
From: vetali100 at gmail.com (Vitalii Colosov)
Date: Sat, 13 Feb 2010 12:47:51 +0200
Subject: [Freeswitch-users] Other party does not hear voice when
	connecting with X-Lite
In-Reply-To: <2F81A680-6A5C-4E3F-9590-FFF6AC4EA80D@freeswitch.org>
References: <a858e7691002120133q5e4c806fx309517e4fdec45c5@mail.gmail.com>
	<2F81A680-6A5C-4E3F-9590-FFF6AC4EA80D@freeswitch.org>
Message-ID: <a858e7691002130247s439177acg23d6a41de3a9d21@mail.gmail.com>

Thanks for the advice!

Yes, x-lite and FS are behind the nat.

I changed x-lite settings from "Discover global IP" to "Use Local IP
adress", but this did not solve the problem.

I would like to add that when x-lite is configured directly to international
gateway sip proxy, it is working fine.
Only when it is connected via FS, voice is missing...

There is definetely something wrong with FS configuration...

Do you know if I can enable some debug level that will provide me some
useful information about voice part...?

Thank you!

Vitalie



2010/2/12 Brian West <brian at freeswitch.org>

> is x-lite behind nat with the freeswitch box?  If so you'll need to disable
> the discover global IP so that it doesn't try to hair pin thru your NAT
> router.... Most nat routers won't work correctly trying to do that.
>
> /b
>
> On Feb 12, 2010, at 3:33 AM, Vitalii Colosov wrote:
>
> >
> > But when I connect using X-Lite, it connects but other party cannot hear
> me.
> > I can hear him well...
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100213/41cf968c/attachment-0002.html 

From tculjaga at gmail.com  Sat Feb 13 03:27:11 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Sat, 13 Feb 2010 12:27:11 +0100
Subject: [Freeswitch-users] Mod_Conference capacity....
In-Reply-To: <13ca621c1002121436n9b44dcexffeaafe998d4c88a@mail.gmail.com>
References: <191c3a031002120856q93e52cep14ee96f1e7ed6d1@mail.gmail.com>
	<324899354.1084631265995845519.JavaMail.root@mail-2.01.com>
	<191c3a031002120944u346a3f7cq779f5c9e0ae31522@mail.gmail.com>
	<13ca621c1002121436n9b44dcexffeaafe998d4c88a@mail.gmail.com>
Message-ID: <65d96fc81002130327q66d8227ao29ae46f8c7ee556a@mail.gmail.com>

On Fri, Feb 12, 2010 at 11:36 PM, Gavin Henry <gavin.henry at gmail.com> wrote:

> Hi,
>
> I think this is unfair to automatically point a user to commercial
> support. Its like holding your users hostage until they pay for info.
>
>
well, the point is that every application is different and nobody can say
what the performance for this or that HW exactly is. This is not like
holding a hostage it is more like "we need to play on your existing platform
run some tests so we can come out with some real benchmark"... this is what
i can read from Anthony's e-mail.


I'm willing to write a page on the wiki with info on the recommended
> tools to use etc. with links to the commercial support resource list
> if they can't be bothered to do the setup themselves or want the
> experts paid for guidance.
>
>

yes, SIPP together with nmon and wireshark and Adobe Audition are the right
tools (at least thats what i'm using...) for such benchmarking.


   - SIPP to generate traffic load,
   - nmon to get some real stats on the machine
   - wireshark to quickly check the jitter on your test call and extract the
   voice stream (sniffing has to be done on a mirrored switch port)
   - adobe autition to perform voice quality check on the extracted voice
   stream




> It just reads like a bit of an insult. Have a question? Well you must
> be stupid and need to pay for help immediately.
>
>
dont think so, "you must learn how to use all these tools"... if you don't
want to do it, there is another option but it is not free :(



> He may have even written the page for the docs team if he was
> encouraged first with a few pointers on how to do the tests and
> written them up with results for others to find with a big YMMV
> warning.
>

well ... i think the time is always the issue + nobody likes documenting
things :(


>
> Just my thoughts from experiences as the Doc team lead for the
> OpenLDAP project and dealing with lots of users asking questions too.
>
> But some users never read docs and there a very good amount of high
> quality pages in the wiki. Others, if helped, come back and
> contribute, but you may have lost this one.
>
>
I cannot say anything here except you are right and i agree what you are
saying about documenting things ...



> Gav.
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100213/ea836fdf/attachment-0002.html 

From mcampbellsmith at gmail.com  Sat Feb 13 03:43:41 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Sat, 13 Feb 2010 22:43:41 +1100
Subject: [Freeswitch-users] Forbidden using UDP, works with TCP/TLS
In-Reply-To: <33c87fa31002121629k74f26807i131841aa4252461b@mail.gmail.com>
References: <33c87fa31002101824h21e3bf7lb13b68e8d4fd36a2@mail.gmail.com>
	<2EF27EFC-1634-46A3-B438-2EE99BEF58F1@jerris.com>
	<33c87fa31002110131w3cf6ad00n802669856085a879@mail.gmail.com>
	<CA21231C-108B-4445-A3B9-BF8D94FFE898@jerris.com>
	<33c87fa31002110213q1de0eebfsb950d8725ea9aba@mail.gmail.com>
	<0A905834-850F-4D1F-AEC5-D3A87CE4141B@jerris.com>
	<33c87fa31002121629k74f26807i131841aa4252461b@mail.gmail.com>
Message-ID: <33c87fa31002130343k7a01e9abt9c33360399f246f8@mail.gmail.com>

More testing. The device registers successfully to my SIP provider
directly using UDP - why would FS be rejecting the registration
request?



On Sat, Feb 13, 2010 at 11:29 AM, Mark Campbell-Smith
<mcampbellsmith at gmail.com> wrote:
> This is the working one with TLS... does this shed any light on why
> this user can not register using UDP?
>
> ? ------------------------------------------------------------------------
> ? REGISTER sip:mydns.dyndns.org:442 SIP/2.0
> ? Via: SIP/2.0/TLS 121.xxx.xxx.xxx:10371;branch=z9hG4bK-9465a0f9;rport
> ? From: 2000 <sip:2000 at mydns.dyndns.org:442>;tag=34ac954c7d2abf51o0
> ? To: 2000 <sip:2000 at mydns.dyndns.org:442>
> ? Call-ID: 8657c383-fea70d0a at 10.0.0.1
> ? CSeq: 31055 REGISTER
> ? Max-Forwards: 70
> ? Authorization: Digest
> username="2000",realm="mydns.dyndns.org",nonce="5000f49a-1836-11df-985e-e77ba7a22ac3",uri="sip:mydns.dyndns.org:442",algorithm=MD5,response="2bad1d5fadbb0465b0a513352db0292b",qop=auth,nc=00000001,cnonce="6951ee10"
> ? Contact: 2000 <sip:2000 at 121.xxx.xxx.xxx:10371;transport=tls>;expires=1800
> ? User-Agent: Linksys/SPA3102-5.1.10(GW)
> ? Content-Length: 0
> ? Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
> ? Supported: x-sipura, replaces
>
> ? ------------------------------------------------------------------------
> tport_deliver(0xb6d07cd8): msg 0xb6da3cc0 (795 bytes) from
> tls/121.xxx.xxx.xxx:10371/sips next=(nil)
> nta: received REGISTER sip:mydns.dyndns.org:442 SIP/2.0 (CSeq 31055)
> nta: REGISTER (31055) going to a default leg
> nua: nua_stack_process_request: entering
> nua: nh_create: entering
> nua: nh_create_handle: entering
> nua: nua_stack_set_params: entering
> soa_clone(static::0x97cc698, 0x9794808, 0xb6d83680) called
> soa_set_params(static::0xb6d77020, ...) called
> nua(0xb6d83680): event i_register 100 Trying
> nua: nua_application_event: entering
> nua: nua_respond: entering
> nua(0xb6d83680): sent signal r_respond
> nua: nua_handle_destroy: entering
> nua(0xb6d83680): sent signal r_destroy
> nua: nua_handle_magic: entering
> nua: nua_handle_destroy: entering
> tport(0xb6d07cd8): reset timer
> nua(0xb6d83680): recv signal r_respond 200 OK
> nua: nua_stack_set_params: entering
> soa_set_params(static::0xb6d77020, ...) called
> tport_tsend(0xb6d07cd8) tpn = TLS/121.xxx.xxx.xxx:10371
> tport_tls_writevec: vec 0xb6d2eb80 0xb6da4270 92 (92)
> tport_tls_writevec: vec 0xb6d2eb80 0xb6ded80a 76 (76)
> tport_tls_writevec: vec 0xb6d2eb80 0xb6da42cc 69 (69)
> tport_tls_writevec: vec 0xb6d2eb80 0xb6ded889 59 (59)
> tport_tls_writevec: vec 0xb6d2eb80 0xb6da4311 329 (329)
> tport_vsend(0xb6d07cd8): 625 bytes of 625 to tls/121.xxx.xxx.xxx:10371
> tport_vsend returned 625
> send 625 bytes to tls/[121.xxx.xxx.xxx]:10371 at 00:25:39.795356:
> ? ------------------------------------------------------------------------
> ? SIP/2.0 200 OK
>
> On Fri, Feb 12, 2010 at 9:22 AM, Michael Jerris <mike at jerris.com> wrote:
>> how is this different from the working one?
>>
>> Mike
>>
>> On Feb 11, 2010, at 5:13 AM, Mark Campbell-Smith wrote:
>>
>>> ah thats true... The trace is not too readable to me, but may give
>>> some insight to someone that can read the sofia logs....
>>>
>>>
>>> recv 752 bytes from udp/[121.xxx.xxx.xxx]:5060 at 10:10:37.803288:
>>> ? ------------------------------------------------------------------------
>>> ? REGISTER sip:mydns.dyndns.org SIP/2.0
>>> ? Via: SIP/2.0/UDP 121.xxx.xxx.xxx:5060;branch=z9hG4bK-9052c91f
>>> ? From: 2000 <sip:2000 at mydns.dyndns.org>;tag=7a9dbbbfa691136do0
>>> ? To: 2000 <sip:2000 at mydns.dyndns.org>
>>> ? Call-ID: 610db38-dd3b511f at 121.xxx.xxx.xxx
>>> ? CSeq: 32330 REGISTER
>>> ? Contact: 2000 <sip:2000 at 121.xxx.xxx.xxx:5060>;expires=900
>>> ? Authorization: Digest username="2000", realm="mydns.dyndns.org",
>>> nonce="b3298cfe-16f5-11df-9734-e77ba7a22ac3",
>>> uri="sip:mydns.dyndns.org",
>>> response="724fca542ce08d3f12b9ba1043bebb0c", algorithm=MD5,
>>> qop="1225e2f1"
>>> ? Max-Forwards: 70
>>> ? User-Agent: Linksys/SPA3102-5.1.10(GW)
>>> ? Supported: x-sipura
>>> ? Supported: replaces
>>> ? Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
>>> ? Content-Length: 0
>>>
>>>
>>> ? ------------------------------------------------------------------------
>>> tport_deliver(0x97cde80): msg 0x98297e8 (752 bytes) from
>>> udp/121.xxx.xxx.xxx:5060/sip next=(nil)
>>> nta: received REGISTER sip:mydns.dyndns.org SIP/2.0 (CSeq 32330)
>>> nta: REGISTER (32330) going to a default leg
>>> nua: nua_stack_process_request: entering
>>> nua: nh_create: entering
>>> nua: nh_create_handle: entering
>>> nua: nua_stack_set_params: entering
>>> soa_clone(static::0x97cc698, 0x9794808, 0x981cb70) called
>>> soa_set_params(static::0x9758ba8, ...) called
>>> nua(0x981cb70): event i_register 100 Trying
>>> nua: nua_application_event: entering
>>> nua: nua_respond: entering
>>> nua(0x981cb70): sent signal r_respond
>>> nua: nua_handle_destroy: entering
>>> nua(0x981cb70): sent signal r_destroy
>>> nua: nua_handle_magic: entering
>>> nua: nua_handle_destroy: entering
>>> nua(0x981cb70): recv signal r_respond 403 Forbidden
>>> nua: nua_stack_set_params: entering
>>> soa_set_params(static::0x9758ba8, ...) called
>>> tport_tsend(0x97cde80) tpn = UDP/121.xxx.xxx.xxx:5060
>>> tport_resolve addrinfo = 121.xxx.xxx.xxx:5060
>>> tport_by_addrinfo(0x97cde80): not found by name UDP/121.xxx.xxx.xxx:5060
>>> tport_vsend(0x97cde80): 495 bytes of 495 to udp/121.xxx.xxx.xxx:5060
>>> tport_vsend returned 495
>>> send 495 bytes to udp/[121.xxx.xxx.xxx]:5060 at 10:10:37.812955:
>>> ? ------------------------------------------------------------------------
>>> ? SIP/2.0 403 Forbidden
>>>
>>> On Thu, Feb 11, 2010 at 8:57 PM, Michael Jerris <mike at jerris.com> wrote:
>>>> you can crank up the sofia loglevel as well
>>>>
>>>> Mike
>>>>
>>>> On Feb 11, 2010, at 4:31 AM, Mark Campbell-Smith wrote:
>>>>
>>>>> I don't see debug logs (fsctl debug_level 9 or loglevel 9) during the
>>>>> registration process.
>>>>>
>>>>> All I see is the sip messages when the sip trace is activated (403 Forbidden)
>>>>>
>>>>> Is there other debugging that I can enable?
>>>>>
>>>>> On Thu, Feb 11, 2010 at 4:25 PM, Michael Jerris <mike at jerris.com> wrote:
>>>>>> Look at the FreeSWITCH debug logs and compare the differences, they should tell you why.
>>>>>>
>>>>>> Mike
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>



From vetali100 at gmail.com  Sat Feb 13 03:46:12 2010
From: vetali100 at gmail.com (Vitalii Colosov)
Date: Sat, 13 Feb 2010 13:46:12 +0200
Subject: [Freeswitch-users] Other party does not hear voice when
	connecting with X-Lite
In-Reply-To: <a858e7691002130247s439177acg23d6a41de3a9d21@mail.gmail.com>
References: <a858e7691002120133q5e4c806fx309517e4fdec45c5@mail.gmail.com>
	<2F81A680-6A5C-4E3F-9590-FFF6AC4EA80D@freeswitch.org>
	<a858e7691002130247s439177acg23d6a41de3a9d21@mail.gmail.com>
Message-ID: <a858e7691002130346y2018746o2a71a084a3fd12ee@mail.gmail.com>

I found the following differences when using YATE and X-Lite clients in FS
log:

YATE client (working)
[DEBUG] sofia_glue.c:2528 AUDIO RTP [sofia/internal/
1000 at sip.myprovider123.com] 10.244.47.100 port 22894 ->
<<<my.global.ip.adress>>> port 20080 codec: 0 ms: 30

X-Lite client (not working)
[DEBUG] sofia_glue.c:2528 AUDIO RTP [sofia/internal/
1000 at sip.myprovider123.com] 10.244.47.100 port 23952 -> 192.168.2.10
(<<<my.LOCAL.ip.adress behind the router>>>) port 46370 codec: 0 ms: 20


It looks like FS is trying to send MEDIA part to my local ip adress instead
of global...And Yate sends correct, to global IP...

Any ideas?

Thank

2010/2/13 Vitalii Colosov <vetali100 at gmail.com>

> Thanks for the advice!
>
> Yes, x-lite and FS are behind the nat.
>
> I changed x-lite settings from "Discover global IP" to "Use Local IP
> adress", but this did not solve the problem.
>
> I would like to add that when x-lite is configured directly to
> international gateway sip proxy, it is working fine.
> Only when it is connected via FS, voice is missing...
>
>  There is definetely something wrong with FS configuration...
>
> Do you know if I can enable some debug level that will provide me some
> useful information about voice part...?
>
> Thank you!
>
> Vitalie
>
>
>
> 2010/2/12 Brian West <brian at freeswitch.org>
>
> is x-lite behind nat with the freeswitch box?  If so you'll need to disable
>> the discover global IP so that it doesn't try to hair pin thru your NAT
>> router.... Most nat routers won't work correctly trying to do that.
>>
>> /b
>>
>> On Feb 12, 2010, at 3:33 AM, Vitalii Colosov wrote:
>>
>> >
>> > But when I connect using X-Lite, it connects but other party cannot hear
>> me.
>> > I can hear him well...
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100213/c608ff02/attachment-0002.html 

From rupa at rupa.com  Sat Feb 13 06:23:19 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Sat, 13 Feb 2010 08:23:19 -0600
Subject: [Freeswitch-users] Svar: Re: Need help setting up a feature
In-Reply-To: <4B7676D6020000E100000481@mail.fribert.dk>
References: <4B73A799020000E100000470@mail.fribert.dk>
	<b39273531002110528p42218d07q95e97485139315f1@mail.gmail.com>
	<4B7676D6020000E100000481@mail.fribert.dk>
Message-ID: <b39273531002130623s4f4a8617r6b4ad0f3ebad1836@mail.gmail.com>

Ok, how to transfer a call on my siemens a580.  It is more complex than it
needs to.

While call is up with partya, press Menu and then select External call.
Place call to party b, talk to party b and tell them you are transfering
Press the flash (R) key to initiate the transfer.

This works for me with the settings I have down below.  With the settings
you have listed in your paste, pressing the flash key will just send a INFO
packet with the key set to "16".  This is probably not what you want.

On Sat, Feb 13, 2010 at 2:54 AM, mailinglist <mailinglist at fribert.dk> wrote:

>  Hi Rupa
>
> I've got similar settings here, but I can't make them work, nothing happens
> when I press R:
>
> The settings are
> DTMP over VoIP connections: Send settings [ ]Auto   [X] Audio  [ ] RFC 2833
> [X] SIP Info
>
> Call Transfer
> Use the R key to initiate call     ( ) Yes   (X) No
> transfer with the SIP Refer Method
>
> Transfer Call by On-Hook        (X) Yes   ( ) No
>
> Derive Target address             ( ) from SIP URL   (X)   from SIP Contact
> Header
>
> Find Target address automatically    ( ) Yes   (X) No
>
> Hook Flash  (R-key)
>
> Application Type:   dtmf-relay
> Application Signal: 16
>
> That's all the settings, I had to set 'send settings' from auto to
> audio+SIP Info otherwise the R-Flash settings were disabled.
>
>
> But all in all, I think a *1 or something during the call would be a better
> method, because I should be able to make that work on al types of phones,
> right?
>
>
>  Best regards
> Fribse
>
> /Running Freeswitch on pfSense running on a VMWare ESXi 4.0
>
> >>> 11-02-2010 kl. 14:28 skrev Rupa Schomaker <rupa at rupa.com> i
> meddelelsen <b39273531002110528p42218d07q95e97485139315f1 at mail.gmail.com>:
>   My Siemens A580 has options for controlling the R key. It seems that you
> can either have it setup for transfer or as a hook flash. Default is as a
> transfer key.
>
> I haven't succeeded in getting it to work for transfer and it is wayyyyy
> down low on my list of things to do with the phone.
>
> From the web UI:
>
>    Call Transfer  Use the R key to initiate call transfer with the SIP
> Refer method.: Yes No  Transfer Call by On-Hook: Yes No  Derive target
> address: from SIP URL from SIP contact header  Find target addr.
> automatically: Yes No
>   Hook Flash (R-key)  R key settings are disabled because the R key is
> being used for call transfer.
>
> On Wed, Feb 10, 2010 at 11:45 PM, mailinglist <mailinglist at fribert.dk>wrote:
>
>>  Sorry for the repost, but the previous thread just died :-)
>>  I'm trying to get the possibility of transfering an incoming call from
>> one extension to another, and give the possibility of turning it into a
>> conference.
>> I don't have a 'transfer' button.
>> I do have an 'R' button on the Siemens handsets, and a 'Flash' button on
>> the Sipura. The 'Flash' button gives me a new dialtone, gives the caller
>> MOH, and then I can dial the new extension, and transfer the call, but not
>> create a conference.
>> But the Siemens handset does not have a 'flash', and pressing the R
>> doesn't do anything.
>>  It might be two different features 'transfer' and 'conference'...
>>
>> But I thought that using the bind_meta_app would accomplish both.
>>  It's on an incoming call from the outside.
>> So the situation:
>> The Public folder has an entry that matches the dialed number, and does a
>> transfer to 8202.
>> Then the dialplan matches the 8202 with a group, and the phone rings.
>> Somebody picks it up, finds out that it needs to be transferred to another
>> extension, or transferred to a conference with a second extension.
>> How do I construct that?
>>   Best regards
>> Fribse
>>  /Running Freeswitch on pfSense running on a VMWare ESXi 4.0
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100213/8a8d5e2f/attachment-0002.html 

From anthony.minessale at gmail.com  Sat Feb 13 07:07:33 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 13 Feb 2010 09:07:33 -0600
Subject: [Freeswitch-users] How to set outbound caller id info for
	multiple users/extensions
In-Reply-To: <093DD565390C1E4FB15D7B383E86BB05AF05AB@Goose.personalcyberspace.net>
References: <093DD565390C1E4FB15D7B383E86BB05AF05AB@Goose.personalcyberspace.net>
Message-ID: <191c3a031002130707v5f1d5ee0lb9e791e9a0aab956@mail.gmail.com>

It should be covered on the wiki http://wiki.freeswitch.org

On Feb 12, 2010 6:23 PM, "Joseph Puchalski" <
joseph.puchalski at personalcyberspace.com> wrote:

 I'm having problems setting different outbound caller id info for different
extensions/users.



I've set up a small system with two active users. I set up my users by
copying and modifying existing entries from the dialplan files that come
with freeSWITCH



Here are my two extensions:



<include>

  <user id="5859">

    <params>

      <param name="password" value="xxxxxx"/>

      <param name="vm-password" value="xxxxxx"/>

    </params>

    <variables>

      <variable name="toll_allow" value="domestic,international,local"/>

      <variable name="accountcode" value="5859"/>

      <variable name="user_context" value="default"/>

      <variable name="effective_caller_id_name" value="5859"/>

      <variable name="effective_caller_id_number" value="5859"/>

      <variable name="outbound_caller_id_name"
value="$${outbound_caller_name}"/>

      <variable name="outbound_caller_id_number"
value="$${outbound_caller_id}"/>

      <variable name="callgroup" value="techsupport"/>

    </variables>

  </user>

</include>



<include>

  <user id="5515">

    <params>

      <param name="password" value="xxxxxxx"/>

      <param name="vm-password" value="xxxxxxx"/>

    </params>

    <variables>

      <variable name="toll_allow" value="domestic,international,local"/>

      <variable name="accountcode" value="5515"/>

      <variable name="user_context" value="default"/>

      <variable name="effective_caller_id_name" value="5515"/>

      <variable name="effective_caller_id_number" value="5515"/>

      <variable name="outbound_caller_id_name" value="Jane Doe"/>

      <variable name="outbound_caller_id_number" value="5558395515"/>

      <variable name="callgroup" value="techsupport"/>

    </variables>

  </user>

</include>



These extensions are in files named 5859.xml and 5515.xml respectively.



I'm using a SIP trunk from Vitelity (in and out) with two DIDs corresponding
to the extensions above.



Inbound and outbound calling work as needed with one exception: Calls
originating from user/extension 5515 go out with the caller ID of
extension/user 5859.



Extension 5859 was the first that I created.



Where should I be setting the outbound caller id number for my second
extension? I've been trying to track this down in the available
documentation but have been unable to do so.



I apologize ahead of time if this is answered somewhere obvious that I've
missed.



Thanks for any help.



Joe (FreeSWITCH newbie)

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100213/d7e7160a/attachment-0002.html 

From brian at freeswitch.org  Sat Feb 13 07:08:53 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 13 Feb 2010 09:08:53 -0600
Subject: [Freeswitch-users] Forbidden using UDP, works with TCP/TLS
In-Reply-To: <33c87fa31002130343k7a01e9abt9c33360399f246f8@mail.gmail.com>
References: <33c87fa31002101824h21e3bf7lb13b68e8d4fd36a2@mail.gmail.com>
	<2EF27EFC-1634-46A3-B438-2EE99BEF58F1@jerris.com>
	<33c87fa31002110131w3cf6ad00n802669856085a879@mail.gmail.com>
	<CA21231C-108B-4445-A3B9-BF8D94FFE898@jerris.com>
	<33c87fa31002110213q1de0eebfsb950d8725ea9aba@mail.gmail.com>
	<0A905834-850F-4D1F-AEC5-D3A87CE4141B@jerris.com>
	<33c87fa31002121629k74f26807i131841aa4252461b@mail.gmail.com>
	<33c87fa31002130343k7a01e9abt9c33360399f246f8@mail.gmail.com>
Message-ID: <C37AB6F3-F712-41B6-9D6D-7A8AE3C6CABB@freeswitch.org>

Can't tell since you keep cutting the lines required to figure this out... FULL console log with FULL sip trace.  You always cut out what I wanna see.  Just get a pcap.

/b

On Feb 13, 2010, at 5:43 AM, Mark Campbell-Smith wrote:

> More testing. The device registers successfully to my SIP provider
> directly using UDP - why would FS be rejecting the registration
> request?




From brian at freeswitch.org  Sat Feb 13 07:09:35 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 13 Feb 2010 09:09:35 -0600
Subject: [Freeswitch-users] Other party does not hear voice when
	connecting with X-Lite
In-Reply-To: <a858e7691002130346y2018746o2a71a084a3fd12ee@mail.gmail.com>
References: <a858e7691002120133q5e4c806fx309517e4fdec45c5@mail.gmail.com>
	<2F81A680-6A5C-4E3F-9590-FFF6AC4EA80D@freeswitch.org>
	<a858e7691002130247s439177acg23d6a41de3a9d21@mail.gmail.com>
	<a858e7691002130346y2018746o2a71a084a3fd12ee@mail.gmail.com>
Message-ID: <D160C09B-1623-4A0A-BB14-1A1CC04A9276@freeswitch.org>

That should work either way then...are you trying to do this all on the same machine?

/b

On Feb 13, 2010, at 5:46 AM, Vitalii Colosov wrote:

> It looks like FS is trying to send MEDIA part to my local ip adress instead of global...And Yate sends correct, to global IP...
> 
> Any ideas?
> 
> Thank 




From brian at freeswitch.org  Sat Feb 13 07:11:05 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 13 Feb 2010 09:11:05 -0600
Subject: [Freeswitch-users] transmission status of channel
In-Reply-To: <a44688a1002122314t4060f29ehce920077ee90704b@mail.gmail.com>
References: <a44688a1002120732v7d95a6f9x2d4bd7ed68de59eb@mail.gmail.com>
	<3028D6AC-46F2-4291-BAA8-E49C9347917B@freeswitch.org>
	<b39273531002120952u698ab5d1x799696b052ced9a9@mail.gmail.com>
	<8e9d67561002121010p37b16aceyb06cda0420e1aa99@mail.gmail.com>
	<a44688a1002122314t4060f29ehce920077ee90704b@mail.gmail.com>
Message-ID: <2FA9500D-5990-4B8E-8DBF-0966DEDC42EA@freeswitch.org>

FreeSWITCH doesn't support RTCP yet.

/b

On Feb 13, 2010, at 1:14 AM, Woody Dickson wrote:

> Does anyone know where to find those RTCP info from the core rtp stack?
> 
> On Fri, Feb 12, 2010 at 10:10 AM, Timur Valishev <tim at novion.ru> wrote:
>> It would be very nice if FS pass RTCP information to channel vars...
>> 
>> Best regards,
>> Timur Valishev

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100213/6e5a8d36/attachment-0002.html 

From brian at freeswitch.org  Sat Feb 13 07:14:26 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 13 Feb 2010 09:14:26 -0600
Subject: [Freeswitch-users] Bridging sessions from two separate lua
	scripts; uuid_bridge? intercept?
In-Reply-To: <eb7797d71002121904v5bdfac87w2684c461ca1770cc@mail.gmail.com>
References: <eb7797d71002071831w27301c5dva767b82b38c2ce6f@mail.gmail.com>
	<C23AA196-E851-4D5B-B1B5-2C2F32F6BB68@freeswitch.org>
	<eb7797d71002071929pb53e1adue2e91e6bdb598251@mail.gmail.com>
	<eb7797d71002080737k3f8fa5c9m8bf3ef2fcf42881e@mail.gmail.com>
	<D753EDBD-6ED7-43C2-B5F5-A899E3C3D3E2@jerris.com>
	<eb7797d71002121904v5bdfac87w2684c461ca1770cc@mail.gmail.com>
Message-ID: <BD307995-B481-4F2A-B072-21AEA34BBEAC@freeswitch.org>

http://wiki.freeswitch.org/wiki/Mod_lua#session:setAutoHangup

But I don't get why you're doing this in such a convoluted way... I'm still not clear why you're going at the task in this manner.  Can you clearly outline what exactly you're trying to accomplish?

/b

On Feb 12, 2010, at 9:04 PM, Adam Wilt wrote:

> Thanks again for the help Michael.
> 
> I'm now upgraded to version 1.5, but I'm still getting the same
> problem. When I try to bridge sessions from two separate lua scripts,
> both sessions hang up on me. I think maybe I don't understand how
> "intercept" works.
> Anyway, I posted the debug trace here:
> 
>                      http://pastebin.freeswitch.org/12121
> 
> And I also put together a small example which exhibits the problem.
> The first script is started by an inbound call and starts the second
> script. The second script places an outbound call and tries to bridge
> the two sessions together:
> 
> Inbound script:  http://pastebin.freeswitch.org/12122
> Outbound script: http://pastebin.freeswitch.org/12123
> 
> Thanks,
> Adam

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100213/31a38f36/attachment-0002.html 

From anthony.minessale at gmail.com  Sat Feb 13 07:14:40 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 13 Feb 2010 09:14:40 -0600
Subject: [Freeswitch-users] Bridging sessions from two separate lua
	scripts; uuid_bridge? intercept?
In-Reply-To: <eb7797d71002121904v5bdfac87w2684c461ca1770cc@mail.gmail.com>
References: <eb7797d71002071831w27301c5dva767b82b38c2ce6f@mail.gmail.com>
	<C23AA196-E851-4D5B-B1B5-2C2F32F6BB68@freeswitch.org>
	<eb7797d71002071929pb53e1adue2e91e6bdb598251@mail.gmail.com>
	<eb7797d71002080737k3f8fa5c9m8bf3ef2fcf42881e@mail.gmail.com>
	<D753EDBD-6ED7-43C2-B5F5-A899E3C3D3E2@jerris.com>
	<eb7797d71002121904v5bdfac87w2684c461ca1770cc@mail.gmail.com>
Message-ID: <191c3a031002130714x2d876657p9c90d768d28a4496@mail.gmail.com>

One or the other of intercept or uuid_bridge not both.

Either uuid_bridge uuid1 uuid2 or
Execute intercept on session1 with uuid2
Or uuid_transfer to intercept:uuid2 inline

When you uuid_bridge you must exit the script.

When you intercept the call will block until the bridge is over unless its
bypass media otherwise exit the script

If either session was created inside the script use session:setAutoHangup(0)
on them first.

On Feb 12, 2010 9:11 PM, "Adam Wilt" <wiltingtree at gmail.com> wrote:

Thanks again for the help Michael.

I'm now upgraded to version 1.5, but I'm still getting the same
problem. When I try to bridge sessions from two separate lua scripts,
both sessions hang up on me. I think maybe I don't understand how
"intercept" works.
Anyway, I posted the debug trace here:

                     http://pastebin.freeswitch.org/12121

And I also put together a small example which exhibits the problem.
The first script is started by an inbound call and starts the second
script. The second script places an outbound call and tries to bridge
the two sessions together:

Inbound script:  http://pastebin.freeswitch.org/12122
Outbound script: http://pastebin.freeswitch.org/12123

Thanks,
Adam





On 2/9/10, Michael Jerris <mike at jerris.com> wrote:
> 1.4? how does the future look, report back...
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100213/74e4f824/attachment-0002.html 

From anthony.minessale at gmail.com  Sat Feb 13 07:15:30 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 13 Feb 2010 09:15:30 -0600
Subject: [Freeswitch-users] transmission status of channel
In-Reply-To: <a44688a1002122314t4060f29ehce920077ee90704b@mail.gmail.com>
References: <a44688a1002120732v7d95a6f9x2d4bd7ed68de59eb@mail.gmail.com>
	<3028D6AC-46F2-4291-BAA8-E49C9347917B@freeswitch.org>
	<b39273531002120952u698ab5d1x799696b052ced9a9@mail.gmail.com>
	<8e9d67561002121010p37b16aceyb06cda0420e1aa99@mail.gmail.com>
	<a44688a1002122314t4060f29ehce920077ee90704b@mail.gmail.com>
Message-ID: <191c3a031002130715w46e2dc7eo284314498a4b53fa@mail.gmail.com>

Yes in the patch that would be written when an appropriate bounty is
collected.

On Feb 13, 2010 1:22 AM, "Woody Dickson" <woodydickson at gmail.com> wrote:

Does anyone know where to find those RTCP info from the core rtp stack?


On Fri, Feb 12, 2010 at 10:10 AM, Timur Valishev <tim at novion.ru> wrote:
> It would be very nice if ...
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100213/0cbef68c/attachment-0002.html 

From brian at freeswitch.org  Sat Feb 13 07:17:01 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 13 Feb 2010 09:17:01 -0600
Subject: [Freeswitch-users] How to set outbound caller id info for
	multiple users/extensions
In-Reply-To: <191c3a031002130707v5f1d5ee0lb9e791e9a0aab956@mail.gmail.com>
References: <093DD565390C1E4FB15D7B383E86BB05AF05AB@Goose.personalcyberspace.net>
	<191c3a031002130707v5f1d5ee0lb9e791e9a0aab956@mail.gmail.com>
Message-ID: <3FF62C88-2423-43F0-B8A3-C64EF4BC80AC@freeswitch.org>

I also have to point out their is no such official variable for "outbound_caller_id_name" or "outbound_caller_id_number", Those are just made up variables I used in the default config.

01_example.com.xml

You'll notice I use these lines.

      <action application="set" data="effective_caller_id_number=${outbound_caller_id_number}"/>
      <action application="set" data="effective_caller_id_name=${outbound_caller_id_name}"/>

Its just a way to set the users default outbound caller ID .

/b


On Feb 13, 2010, at 9:07 AM, Anthony Minessale wrote:

> It should be covered on the wiki http://wiki.freeswitch.org
> 
> 
>> On Feb 12, 2010 6:23 PM, "Joseph Puchalski" <joseph.puchalski at personalcyberspace.com> wrote:
>> 
>> I'm having problems setting different outbound caller id info for different extensions/users.
>> 
>>  
>> I've set up a small system with two active users. I set up my users by copying and modifying existing entries from the dialplan files that come with freeSWITCH
>> 
>>  
>> Here are my two extensions:
>> 
>>  
>> <include>
>> 
>>   <user id="5859">
>> 
>>     <params>
>> 
>>       <param name="password" value="xxxxxx"/>
>> 
>>       <param name="vm-password" value="xxxxxx"/>
>> 
>>     </params>
>> 
>>     <variables>
>> 
>>       <variable name="toll_allow" value="domestic,international,local"/>
>> 
>>       <variable name="accountcode" value="5859"/>
>> 
>>       <variable name="user_context" value="default"/>
>> 
>>       <variable name="effective_caller_id_name" value="5859"/>
>> 
>>       <variable name="effective_caller_id_number" value="5859"/>
>> 
>>       <variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/>
>> 
>>       <variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/>
>> 
>>       <variable name="callgroup" value="techsupport"/>
>> 
>>     </variables>
>> 
>>   </user>
>> 
>> </include>
>> 
>>  
>> <include>
>> 
>>   <user id="5515">
>> 
>>     <params>
>> 
>>       <param name="password" value="xxxxxxx"/>
>> 
>>       <param name="vm-password" value="xxxxxxx"/>
>> 
>>     </params>
>> 
>>     <variables>
>> 
>>       <variable name="toll_allow" value="domestic,international,local"/>
>> 
>>       <variable name="accountcode" value="5515"/>
>> 
>>       <variable name="user_context" value="default"/>
>> 
>>       <variable name="effective_caller_id_name" value="5515"/>
>> 
>>       <variable name="effective_caller_id_number" value="5515"/>
>> 
>>       <variable name="outbound_caller_id_name" value="Jane Doe"/>
>> 
>>       <variable name="outbound_caller_id_number" value="5558395515"/>
>> 
>>       <variable name="callgroup" value="techsupport"/>
>> 
>>     </variables>
>> 
>>   </user>
>> 
>> </include>
>> 
>>  
>> These extensions are in files named 5859.xml and 5515.xml respectively.
>> 
>>  
>> I'm using a SIP trunk from Vitelity (in and out) with two DIDs corresponding to the extensions above.
>> 
>>  
>> Inbound and outbound calling work as needed with one exception: Calls originating from user/extension 5515 go out with the caller ID of extension/user 5859.
>> 
>>  
>> Extension 5859 was the first that I created.
>> 
>>  
>> Where should I be setting the outbound caller id number for my second extension? I've been trying to track this down in the available documentation but have been unable to do so.
>> 
>>  
>> I apologize ahead of time if this is answered somewhere obvious that I've missed.
>> 
>>  
>> Thanks for any help.
>> 
>>  
>> Joe (FreeSWITCH newbie)
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100213/52f12681/attachment-0002.html 

From anthony.minessale at gmail.com  Sat Feb 13 07:22:38 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 13 Feb 2010 09:22:38 -0600
Subject: [Freeswitch-users] Mod_Conference capacity....
In-Reply-To: <65d96fc81002130327q66d8227ao29ae46f8c7ee556a@mail.gmail.com>
References: <191c3a031002120856q93e52cep14ee96f1e7ed6d1@mail.gmail.com>
	<324899354.1084631265995845519.JavaMail.root@mail-2.01.com>
	<191c3a031002120944u346a3f7cq779f5c9e0ae31522@mail.gmail.com>
	<13ca621c1002121436n9b44dcexffeaafe998d4c88a@mail.gmail.com>
	<65d96fc81002130327q66d8227ao29ae46f8c7ee556a@mail.gmail.com>
Message-ID: <191c3a031002130722v5c282c00s69f0b51b91e42520@mail.gmail.com>

Look through the archives of this list and count the answers to questions I
provide daily for the last several years.  It will be a clear pattern.

People persistantly asking about load testing and benchmark numbers will
always get the same response: we only support it commerically.

We learned from experience the dangers of entertaining such questioning and
its our policy to not do so over our free public forum.

On Feb 13, 2010 5:34 AM, "Tihomir Culjaga" <tculjaga at gmail.com> wrote:



On Fri, Feb 12, 2010 at 11:36 PM, Gavin Henry <gavin.henry at gmail.com> wrote:
>
> Hi,
>
> I think thi...

well, the point is that every application is different and nobody can say
what the performance for this or that HW exactly is. This is not like
holding a hostage it is more like "we need to play on your existing platform
run some tests so we can come out with some real benchmark"... this is what
i can read from Anthony's e-mail.


> I'm willing to write a page on the wiki with info on the recommended
> tools to use etc. with link...


yes, SIPP together with nmon and wireshark and Adobe Audition are the right
tools (at least thats what i'm using...) for such benchmarking.


   - SIPP to generate traffic load,
   - nmon to get some real stats on the machine
   - wireshark to quickly check the jitter on your test call and extract the
   voice stream (sniffing has to be done on a mirrored switch port)
   - adobe autition to perform voice quality check on the extracted voice
   stream





>
> It just reads like a bit of an insult. Have a question? Well you must
> be stupid and need to p...

dont think so, "you must learn how to use all these tools"... if you don't
want to do it, there is another option but it is not free :(




>
> He may have even written the page for the docs team if he was
> encouraged first with a few poi...

well ... i think the time is always the issue + nobody likes documenting
things :(



>
>
> Just my thoughts from experiences as the Doc team lead for the
> OpenLDAP project and dealing...

I cannot say anything here except you are right and i agree what you are
saying about documenting things ...



> Gav.
>
>

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100213/281f69d6/attachment-0002.html 

From scottferri09 at gmail.com  Sat Feb 13 07:52:32 2010
From: scottferri09 at gmail.com (Scott Fernandez)
Date: Sat, 13 Feb 2010 21:22:32 +0530
Subject: [Freeswitch-users] Establishing a Call from .Net based
	application
In-Reply-To: <367751821002010737o4ca83476j62c96cb60c8a456a@mail.gmail.com>
References: <874941.17255.qm@web33502.mail.mud.yahoo.com>
	<df97b9c51001310545n54195cadr9faa3db46917ca9e@mail.gmail.com>
	<367751821002010737o4ca83476j62c96cb60c8a456a@mail.gmail.com>
Message-ID: <df97b9c51002130752h48aae119ja80797ab8a3f5b2d@mail.gmail.com>

Hi,

I've done the following things in order for my .NET application to talk to
FS.

1. Per the link http://wiki.freeswitch.org/wiki/Mod_managed , I've enabled
the Mod_managed module in the mentioned location in FS on CentOS 5.2.

2. There is a configuration settings required to Map the "DLL" to ".so"
object in CentOS.
Now, the question is which DLL and .so file to be made available and where?

3. Do we need to include the AppDemo.class in .NET C# classes that we have
built? If so. how do we initiate the call and get the status of that
particular call with the help of AppDemo.class?. Can I have any specific
code for this?

All I need is to initiate a call from .NET application and then it should
talk to mod_managed module and establish a call. Secondly, I need to know
the status of the call such as Ringing, Active, Hangup etc.

How do we achieve this with mod_managed?. Any sample coding that we can get
for this scenario?

Thanks for your help as always :)

Regards,
Scott


On Mon, Feb 1, 2010 at 9:07 PM, Phillip Jones <pjintheusa at gmail.com> wrote:

> As mentioned http://wiki.freeswitch.org/wiki/Mod_managed should give you
> every thing you need to get mod_managed set up.
>
> Then in the source take a look at demo.csx and particularity AppDemo class.
>
> That should get you started.
>
>
>
> On Sun, Jan 31, 2010 at 8:45 AM, Scott Fernandez <scottferri09 at gmail.com>wrote:
>
>> Hi,
>>
>> Thx for the information. Can I have some detailed steps to configure
>> mod_managed class call control and how do we write the API commands in .Net
>> applications?
>>
>> In addition, how do we get the current STATE of the call when I use
>> webapi?. Because it is required for me to route the call to the user upon it
>> is answered or disconnect  it.
>>
>> Thanks,
>> Scott
>>
>> On Wed, Jan 20, 2010 at 8:47 PM, Diego Toro <dftoro at yahoo.com> wrote:
>>
>>> Hi, the answer is yes, you can to use mod_managed wich offer C# managed
>>> class to call control http://wiki.freeswitch.org/wiki/Mod_managed. Or
>>> using managed ESL (libs/esl/managed) which offer C# managed class to receive
>>> and send events and commands to FreeSwitch.
>>>
>>> Diego Toro
>>> http://lacarretade.blogspot.com/
>>>
>>>
>>> --- On Wed, 1/20/10, Scott Fernandez <scottferri09 at gmail.com> wrote:
>>>
>>> > From: Scott Fernandez <scottferri09 at gmail.com>
>>> > Subject: Re: [Freeswitch-users] Establishing a Call from .Net based
>>> application
>>> > To: freeswitch-users at lists.freeswitch.org
>>> > Date: Wednesday, January 20, 2010, 2:17 AM
>>>  > Thanks Dome. Will try it out and get back to
>>> > you if I come across any issues.
>>> >
>>> > Regards,
>>> > Scott.
>>> >
>>> > On Wed, Jan 20, 2010 at 11:02 AM,
>>> > Dome Charoenyost <dome at tel.co.th>
>>> > wrote:
>>> >
>>> > Please try http://wiki.freeswitch.org/wiki/Webapi
>>> >
>>> >
>>> > you can create class and map to webapi.
>>> >
>>> >
>>> >
>>> > Dome C.
>>> >
>>> >
>>> >
>>> > 2010/1/19 Scott Fernandez <scottferri09 at gmail.com>:
>>> >
>>> > > Hi,
>>> >
>>> > >
>>> >
>>> > > Is there any API modules available for me to initiate
>>> > a call from .Net based
>>> >
>>> > > application?.
>>> >
>>> > >
>>> >
>>> > > The idea is to include the API modules if any with the
>>> > .NET base classes so
>>> >
>>> > > that the API commands will be made available on it. I
>>> > know it is doable when
>>> >
>>> > > I use socket programming in .NET in which Telnet
>>> > session is created.
>>> >
>>> > > However, this would potentially hamper the performance
>>> > of the application
>>> >
>>> > > because of multiple sessions that will be created for
>>> > each call.
>>> >
>>> > >
>>> >
>>> > > Other than that, Is there any Freeswitch API modules
>>> > (like plug-ins)
>>> >
>>> > > available in order to include it into the .Net classes
>>> > and start building
>>> >
>>> > > the customized application?
>>> >
>>> > >
>>> >
>>> > > Any help from any one is highly appreciated.
>>> >
>>> > >
>>> >
>>> > > Thanks,
>>> >
>>> > > Scott
>>> >
>>> > >
>>> >
>>> > >
>>> > _______________________________________________
>>> >
>>> > > FreeSWITCH-users mailing list
>>> >
>>> > > FreeSWITCH-users at lists.freeswitch.org
>>> >
>>> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >
>>> > > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >
>>> > > http://www.freeswitch.org
>>> >
>>> > >
>>> >
>>> > >
>>> >
>>> >
>>> >
>>> > _______________________________________________
>>> >
>>> > FreeSWITCH-users mailing list
>>> >
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> >
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >
>>> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >
>>> > http://www.freeswitch.org
>>> >
>>> >
>>> >
>>> >
>>> > -----Inline Attachment Follows-----
>>>  >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>>
>>>
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100213/77549daa/attachment-0002.html 

From dftoro at yahoo.com  Sat Feb 13 09:05:59 2010
From: dftoro at yahoo.com (Diego Toro)
Date: Sat, 13 Feb 2010 09:05:59 -0800 (PST)
Subject: [Freeswitch-users] Establishing a Call from .Net based
	application
In-Reply-To: <df97b9c51002130752h48aae119ja80797ab8a3f5b2d@mail.gmail.com>
Message-ID: <494815.23786.qm@web33508.mail.mud.yahoo.com>


> 1. Per the link http://wiki.freeswitch.org/wiki/Mod_managed
>  , I've enabled the Mod_managed module in the mentioned
> location in FS on CentOS 5.2.

mod_managed is supported on CentOs with Mono.


3. Do we need to include the AppDemo.class in .NET C#
> classes that we have built? If so. how do we initiate the
> call and get the status of that particular call with the
> help of AppDemo.class?. Can I have any specific code for
> this?

AppDemo is only a example about how to use mod_managed like application.

a. You should implement IAppPlugin interface so FreeSwitch brings call control to your C# class through mod_managed.

Simple Example:

using FreeSWITCH;
using FreeSWITCH.Native;

namespace BITS.Ivr.Bp.Server 
{
 public class CIV_BPFSProcess : FreeSWITCH.IAppPlugin
{
  public void Run(AppContext context)
  {
   //answer call
   context.Session.Answer();
   //sleep 2 seconds
   context.Session.sleep(2000, 1);
   //hangup call
   context.Session.Hangup("NORMAL_CLEARING");
  }
 }
}

b. Add next extension to a dialplan

<include>
  <extension name="test">
    <condition field="destination_number" expression="^445(\d+)$" >
    <action application="managed" data="CCU_BPFSProcess"/>
 </condition>	
  </extension>
</include>

c. Call for example to 445100 from a softphone


Diego Toro 
http://lacarretade.blogspot.com/


--- On Sat, 2/13/10, Scott Fernandez <scottferri09 at gmail.com> wrote:

> From: Scott Fernandez <scottferri09 at gmail.com>
> Subject: Re: [Freeswitch-users] Establishing a Call from .Net based application
> To: freeswitch-users at lists.freeswitch.org
> Date: Saturday, February 13, 2010, 10:52 AM
> Hi,
> 
> I've done the following things in order for my .NET
> application to talk to FS.
> 
> 1. Per the link http://wiki.freeswitch.org/wiki/Mod_managed
>  , I've enabled the Mod_managed module in the mentioned
> location in FS on CentOS 5.2.
> 
> 2. There is a configuration settings required to Map the
> "DLL" to ".so" object in CentOS.
> Now, the question is which DLL and .so file to be made
> available and where?
> 
> 
> 3. Do we need to include the AppDemo.class in .NET C#
> classes that we have built? If so. how do we initiate the
> call and get the status of that particular call with the
> help of AppDemo.class?. Can I have any specific code for
> this?
> 
> 
> All I need is to initiate a call from .NET application and
> then it should talk to mod_managed module and establish a
> call. Secondly, I need to know the status of the call such
> as Ringing, Active, Hangup etc. 
> 
> How do we achieve this with mod_managed?. Any sample coding
> that we can get for this scenario?
> 
> 
> Thanks for your help as always :)
> 
> Regards,
> Scott
> 
> 
> On Mon, Feb 1, 2010 at 9:07 PM,
> Phillip Jones <pjintheusa at gmail.com>
> wrote:
> 
> As mentioned http://wiki.freeswitch.org/wiki/Mod_managed
> should give you every thing you need to get mod_managed set
> up. 
> 
> 
> Then in the source take a look at demo.csx and
> particularity AppDemo class.
> 
> 
> That should get you started.
> 
> 
> On Sun, Jan 31, 2010 at 8:45 AM,
> Scott Fernandez <scottferri09 at gmail.com>
> wrote:
> 
> 
> Hi,
> ?
> Thx for the information. Can I have some detailed
> steps to configure mod_managed class call control and how do
> we write the API commands in .Net applications?
> ?
> In addition, how do we get the current STATE of the
> call when I use webapi?. Because it is required for me to
> route the call to the user upon it is answered or
> disconnect? it.
> ?
> Thanks,
> Scott
> 
> 
> On Wed, Jan 20, 2010 at 8:47 PM,
> Diego Toro <dftoro at yahoo.com>
> wrote:
> 
> Hi, the answer is yes, you can to
> use mod_managed wich offer C# managed class to call control
> http://wiki.freeswitch.org/wiki/Mod_managed.
> Or using managed ESL (libs/esl/managed) which offer C#
> managed class to receive and send events and commands to
> FreeSwitch.
> 
> 
> 
> 
> Diego Toro
> http://lacarretade.blogspot.com/
> 
> 
> --- On Wed, 1/20/10, Scott Fernandez <scottferri09 at gmail.com>
> wrote:
> 
> 
> 
> 
> > From: Scott Fernandez <scottferri09 at gmail.com>
> 
> > Subject: Re: [Freeswitch-users] Establishing a
> Call from .Net based application
> > To: freeswitch-users at lists.freeswitch.org
> 
> 
> > Date: Wednesday, January 20, 2010, 2:17 AM
> 
> 
> 
> 
> > Thanks Dome. Will try it out and get back to
> > you if I come across any issues.
> >
> > Regards,
> > Scott.
> >
> > On Wed, Jan 20, 2010 at 11:02 AM,
> > Dome Charoenyost <dome at tel.co.th>
> 
> 
> 
> > wrote:
> >
> > Please try http://wiki.freeswitch.org/wiki/Webapi
> >
> >
> > you can create class and map to webapi.
> >
> 
> 
> 
> >
> >
> > Dome C.
> >
> >
> >
> > 2010/1/19 Scott Fernandez <scottferri09 at gmail.com>:
> >
> > > Hi,
> >
> > >
> 
> 
> >
> > > Is there any API modules available for me to
> initiate
> 
> > a call from .Net based
> >
> > > application?.
> >
> > >
> >
> > > The idea is to include the API modules if any
> with the
> > .NET base classes so
> >
> > > that the API commands will be made available on
> it. I
> 
> 
> 
> > know it is doable when
> >
> > > I use socket programming in .NET in which Telnet
> > session is created.
> >
> > > However, this would potentially hamper the
> performance
> > of the application
> 
> 
> 
> >
> > > because of multiple sessions that will be created
> for
> > each call.
> >
> > >
> >
> > > Other than that, Is there any Freeswitch API
> modules
> > (like plug-ins)
> >
> 
> 
> 
> > > available in order to include it into the .Net
> classes
> > and start building
> >
> > > the customized application?
> >
> > >
> >
> > > Any help from any one is highly appreciated.
> 
> 
> 
> >
> > >
> >
> > > Thanks,
> >
> > > Scott
> >
> > >
> >
> > >
> > _______________________________________________
> >
> > > FreeSWITCH-users mailing list
> 
> 
> 
> >
> > > FreeSWITCH-users at lists.freeswitch.org
> >
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> 
> 
> 
> >
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >
> > > http://www.freeswitch.org
> 
> 
> 
> >
> > >
> >
> > >
> >
> >
> >
> > _______________________________________________
> >
> > FreeSWITCH-users mailing list
> >
> > FreeSWITCH-users at lists.freeswitch.org
> 
> 
> 
> >
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 
> 
> 
> >
> > http://www.freeswitch.org
> >
> >
> >
> >
> > -----Inline Attachment Follows-----
> 
> 
> 
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> 
> 
> 
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 
> 
> 
> > http://www.freeswitch.org
> >
> 
> 
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> 
> 
> 
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 
> 
> 
> http://www.freeswitch.org
> 
> 
> 
> _______________________________________________
> 
> FreeSWITCH-users mailing list
> 
> FreeSWITCH-users at lists.freeswitch.org
> 
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> 
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 
> http://www.freeswitch.org
> 
> 
> 
> 
> 
> _______________________________________________
> 
> FreeSWITCH-users mailing list
> 
> FreeSWITCH-users at lists.freeswitch.org
> 
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> 
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 
> http://www.freeswitch.org
> 
> 
> 
> 
> 
> -----Inline Attachment Follows-----
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 


      



From wiltingtree at gmail.com  Sat Feb 13 10:18:17 2010
From: wiltingtree at gmail.com (Adam Wilt)
Date: Sat, 13 Feb 2010 13:18:17 -0500
Subject: [Freeswitch-users] Bridging sessions from two separate lua
	scripts; uuid_bridge? intercept?
In-Reply-To: <191c3a031002130714x2d876657p9c90d768d28a4496@mail.gmail.com>
References: <eb7797d71002071831w27301c5dva767b82b38c2ce6f@mail.gmail.com>
	<C23AA196-E851-4D5B-B1B5-2C2F32F6BB68@freeswitch.org>
	<eb7797d71002071929pb53e1adue2e91e6bdb598251@mail.gmail.com>
	<eb7797d71002080737k3f8fa5c9m8bf3ef2fcf42881e@mail.gmail.com>
	<D753EDBD-6ED7-43C2-B5F5-A899E3C3D3E2@jerris.com>
	<eb7797d71002121904v5bdfac87w2684c461ca1770cc@mail.gmail.com>
	<191c3a031002130714x2d876657p9c90d768d28a4496@mail.gmail.com>
Message-ID: <eb7797d71002131018r56e86adnc054a3eaa2c05a96@mail.gmail.com>

Thanks Anthony and Brian.

setAutoHangup was specified in the script calling intercept, but not the
other script. When I add it to the other script, neither side hangs up, but
the call is not bridged. Neither party can hear the other party. From your
description, intercept sounds like what I want to do.

I made sure bypass media was off. Does the session being intercepted have to
be in a certain state for the intercept to work? In my example, a prompt is
being played during the time it is intercepted, but I tried having it sleep
instead with the same result.

The reason I'm doing it this way is because both parties have to go through
a bunch of different states before they are allowed to speak to each other.
I tried controlling both legs from the same script previously, but sometimes
one session would block waiting for the other session.

Thanks,
Adam


On Sat, Feb 13, 2010 at 10:14 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> One or the other of intercept or uuid_bridge not both.
>
> Either uuid_bridge uuid1 uuid2 or
> Execute intercept on session1 with uuid2
> Or uuid_transfer to intercept:uuid2 inline
>
> When you uuid_bridge you must exit the script.
>
> When you intercept the call will block until the bridge is over unless its
> bypass media otherwise exit the script
>
> If either session was created inside the script use
> session:setAutoHangup(0) on them first.
>
> On Feb 12, 2010 9:11 PM, "Adam Wilt" <wiltingtree at gmail.com> wrote:
>
> Thanks again for the help Michael.
>
> I'm now upgraded to version 1.5, but I'm still getting the same
> problem. When I try to bridge sessions from two separate lua scripts,
> both sessions hang up on me. I think maybe I don't understand how
> "intercept" works.
> Anyway, I posted the debug trace here:
>
>                      http://pastebin.freeswitch.org/12121
>
> And I also put together a small example which exhibits the problem.
> The first script is started by an inbound call and starts the second
> script. The second script places an outbound call and tries to bridge
> the two sessions together:
>
> Inbound script:  http://pastebin.freeswitch.org/12122
> Outbound script: http://pastebin.freeswitch.org/12123
>
> Thanks,
> Adam
>
>
>
>
>
> On 2/9/10, Michael Jerris <mike at jerris.com> wrote:
> > 1.4? how does the future look, report back...
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100213/6dcabba2/attachment-0002.html 

From gavin.henry at gmail.com  Sat Feb 13 11:20:21 2010
From: gavin.henry at gmail.com (Gavin Henry)
Date: Sat, 13 Feb 2010 19:20:21 +0000
Subject: [Freeswitch-users] Mod_Conference capacity....
In-Reply-To: <191c3a031002130722v5c282c00s69f0b51b91e42520@mail.gmail.com>
References: <191c3a031002120856q93e52cep14ee96f1e7ed6d1@mail.gmail.com>
	<324899354.1084631265995845519.JavaMail.root@mail-2.01.com>
	<191c3a031002120944u346a3f7cq779f5c9e0ae31522@mail.gmail.com>
	<13ca621c1002121436n9b44dcexffeaafe998d4c88a@mail.gmail.com>
	<65d96fc81002130327q66d8227ao29ae46f8c7ee556a@mail.gmail.com>
	<191c3a031002130722v5c282c00s69f0b51b91e42520@mail.gmail.com>
Message-ID: <13ca621c1002131120u482f6420xea1f7cad534cbe1a@mail.gmail.com>

On 13 February 2010 15:22, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> Look through the archives of this list and count the answers to questions I
> provide daily for the last several years.? It will be a clear pattern.
>
> People persistantly asking about load testing and benchmark numbers will
> always get the same response: we only support it commerically.
>
> We learned from experience the dangers of entertaining such questioning and
> its our policy to not do so over our free public forum.

OK, thanks for the info. If this is now the case these three links
should be updated to reflect the
FreeSWITCH projects stand point:

http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations
http://wiki.freeswitch.org/wiki/Long_term_testing_chevymanjosh
http://wiki.freeswitch.org/wiki/FreeSwitch_FAQ#Q:_How_many_concurrent_calls_can_it_support.3F__Any_benchmarks.3F

or a new FAQ entry created etc.

Thanks,

Gavin.


-- 
http://www.suretecsystems.com/services/openldap/
http://www.suretectelecom.com



From wiltingtree at gmail.com  Sat Feb 13 11:32:41 2010
From: wiltingtree at gmail.com (Adam Wilt)
Date: Sat, 13 Feb 2010 14:32:41 -0500
Subject: [Freeswitch-users] Bridging sessions from two separate lua
	scripts; uuid_bridge? intercept?
In-Reply-To: <eb7797d71002131018r56e86adnc054a3eaa2c05a96@mail.gmail.com>
References: <eb7797d71002071831w27301c5dva767b82b38c2ce6f@mail.gmail.com>
	<C23AA196-E851-4D5B-B1B5-2C2F32F6BB68@freeswitch.org>
	<eb7797d71002071929pb53e1adue2e91e6bdb598251@mail.gmail.com>
	<eb7797d71002080737k3f8fa5c9m8bf3ef2fcf42881e@mail.gmail.com>
	<D753EDBD-6ED7-43C2-B5F5-A899E3C3D3E2@jerris.com>
	<eb7797d71002121904v5bdfac87w2684c461ca1770cc@mail.gmail.com>
	<191c3a031002130714x2d876657p9c90d768d28a4496@mail.gmail.com>
	<eb7797d71002131018r56e86adnc054a3eaa2c05a96@mail.gmail.com>
Message-ID: <eb7797d71002131132x2da4d2a0g54b8b8faf19520be@mail.gmail.com>

Here's a new debug log which shows what happens using only intercept and
having setAutoHangup(false) on both sides:

           http://pastebin.freeswitch.org/12124

<http://pastebin.freeswitch.org/12124>And here's my updated bridge function:

function bridge_calls(session,api,b_leg_uuid, call_len)
  freeswitch.consoleLog("info","A leg - Bridging " .. tostring(b_leg_uuid)
.. " with " .. tostring(session.uuid) .. "\n")
  session:setAutoHangup(false)
  session:execute("set","continue_on_fail=true")
  api:executeString("uuid_media " .. tostring(b_leg_uuid))
  api:executeString("uuid_media " .. tostring(session.uuid))
  api:executeString("intercept -bleg " .. tostring(b_leg_uuid))
end


Thanks,
Adam


On Sat, Feb 13, 2010 at 1:18 PM, Adam Wilt <wiltingtree at gmail.com> wrote:

> Thanks Anthony and Brian.
>
> setAutoHangup was specified in the script calling intercept, but not the
> other script. When I add it to the other script, neither side hangs up, but
> the call is not bridged. Neither party can hear the other party. From your
> description, intercept sounds like what I want to do.
>
> I made sure bypass media was off. Does the session being intercepted have
> to be in a certain state for the intercept to work? In my example, a prompt
> is being played during the time it is intercepted, but I tried having it
> sleep instead with the same result.
>
> The reason I'm doing it this way is because both parties have to go through
> a bunch of different states before they are allowed to speak to each other.
> I tried controlling both legs from the same script previously, but sometimes
> one session would block waiting for the other session.
>
> Thanks,
> Adam
>
>
> On Sat, Feb 13, 2010 at 10:14 AM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> One or the other of intercept or uuid_bridge not both.
>>
>> Either uuid_bridge uuid1 uuid2 or
>> Execute intercept on session1 with uuid2
>> Or uuid_transfer to intercept:uuid2 inline
>>
>> When you uuid_bridge you must exit the script.
>>
>> When you intercept the call will block until the bridge is over unless its
>> bypass media otherwise exit the script
>>
>> If either session was created inside the script use
>> session:setAutoHangup(0) on them first.
>>
>> On Feb 12, 2010 9:11 PM, "Adam Wilt" <wiltingtree at gmail.com> wrote:
>>
>> Thanks again for the help Michael.
>>
>> I'm now upgraded to version 1.5, but I'm still getting the same
>> problem. When I try to bridge sessions from two separate lua scripts,
>> both sessions hang up on me. I think maybe I don't understand how
>> "intercept" works.
>> Anyway, I posted the debug trace here:
>>
>>                      http://pastebin.freeswitch.org/12121
>>
>> And I also put together a small example which exhibits the problem.
>> The first script is started by an inbound call and starts the second
>> script. The second script places an outbound call and tries to bridge
>> the two sessions together:
>>
>> Inbound script:  http://pastebin.freeswitch.org/12122
>> Outbound script: http://pastebin.freeswitch.org/12123
>>
>> Thanks,
>> Adam
>>
>>
>>
>>
>>
>> On 2/9/10, Michael Jerris <mike at jerris.com> wrote:
>> > 1.4? how does the future look, report back...
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100213/04625cbe/attachment-0002.html 

From mgg at giagnocavo.net  Sat Feb 13 11:45:03 2010
From: mgg at giagnocavo.net (Michael Giagnocavo)
Date: Sat, 13 Feb 2010 14:45:03 -0500
Subject: [Freeswitch-users] Establishing a Call from .Net
	based	application
In-Reply-To: <df97b9c51002130752h48aae119ja80797ab8a3f5b2d@mail.gmail.com>
References: <874941.17255.qm@web33502.mail.mud.yahoo.com>
	<df97b9c51001310545n54195cadr9faa3db46917ca9e@mail.gmail.com>
	<367751821002010737o4ca83476j62c96cb60c8a456a@mail.gmail.com>
	<df97b9c51002130752h48aae119ja80797ab8a3f5b2d@mail.gmail.com>
Message-ID: <6E8D2069C08AA84A83D336E996AE4C67032C9D5346@mse17be1.mse17.exchange.ms>


2. There is a configuration settings required to Map the "DLL" to ".so" object in CentOS.
Now, the question is which DLL and .so file to be made available and where?

"
If you are experiencing NullReferenceExceptions with any plugin run through the dialplan, make sure you have included the appropriate entry in your dllmap<http://mono-project.com/Config_DllMap> configuration:
 <dllmap dll="mod_managed.dll" target="mod_managed.so" os="!windows"/>
"
mod_managed.so will be in your freeswitch mod directory.

All I need is to initiate a call from .NET application and then it should talk to mod_managed module and establish a call. Secondly, I need to know the status of the call such as Ringing, Active, Hangup etc.

To initiate a call, try ManagedSession.Originate.
-Michael
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100213/44ac713a/attachment-0002.html 

From brian at freeswitch.org  Sat Feb 13 11:54:10 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 13 Feb 2010 13:54:10 -0600
Subject: [Freeswitch-users] Mod_Conference capacity....
In-Reply-To: <13ca621c1002131120u482f6420xea1f7cad534cbe1a@mail.gmail.com>
References: <191c3a031002120856q93e52cep14ee96f1e7ed6d1@mail.gmail.com>
	<324899354.1084631265995845519.JavaMail.root@mail-2.01.com>
	<191c3a031002120944u346a3f7cq779f5c9e0ae31522@mail.gmail.com>
	<13ca621c1002121436n9b44dcexffeaafe998d4c88a@mail.gmail.com>
	<65d96fc81002130327q66d8227ao29ae46f8c7ee556a@mail.gmail.com>
	<191c3a031002130722v5c282c00s69f0b51b91e42520@mail.gmail.com>
	<13ca621c1002131120u482f6420xea1f7cad534cbe1a@mail.gmail.com>
Message-ID: <E2A03ABF-69BF-4F85-9BB9-7CEAA7738E64@freeswitch.org>

This is the FAQ entry.  It applies to EVERYTHING related to load testing.

/b

On Feb 13, 2010, at 1:20 PM, Gavin Henry wrote:

> http://wiki.freeswitch.org/wiki/FreeSwitch_FAQ#Q:_How_many_concurrent_calls_can_it_support.3F__Any_benchmarks.3F

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100213/b8d90379/attachment-0002.html 

From mbsip at gazeta.pl  Sat Feb 13 12:03:07 2010
From: mbsip at gazeta.pl (mbsip)
Date: Sat, 13 Feb 2010 21:03:07 +0100
Subject: [Freeswitch-users]  mod_python fetching data from mysql for VM
Message-ID: <28f27f5d1002131203j132c6230sa260358611a16247@mail.gmail.com>

Hello,

I am trying to use mod_python to fetch data from Mysql db (through
ODBC) and execute voicemail application.
Below a part of my script:

db=MySQLdb.connect("localhost","root","","test")
       Cursor=db.cursor()
       sql = "SELECT email FROM VM WHERE called_num=%s" % the_dest
       Cursor.execute(sql)
       while (1):
               Results = Cursor.fetchone()
               if Results == None:
                       break
               consoleLog("debug", "Found email " + Results[0] +"\n")
               the_recipient = Results[0]
       db.close()

Now i have email address corresponding with called number. The
question is how to use it for voicemail application?
So it also means how to omit all /directory/default/....xml, where
there are all VM parameters set and use fetched data.

       session.answer()
       session.execute("voicemail", "default ${domain} " + the_dest)

Is this possible or should I start all VM app in python from the scratch?

Thanks,
Maciej



From anthony.minessale at gmail.com  Sat Feb 13 12:39:48 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 13 Feb 2010 14:39:48 -0600
Subject: [Freeswitch-users] Bridging sessions from two separate lua
	scripts; uuid_bridge? intercept?
In-Reply-To: <eb7797d71002131132x2da4d2a0g54b8b8faf19520be@mail.gmail.com>
References: <eb7797d71002071831w27301c5dva767b82b38c2ce6f@mail.gmail.com>
	<C23AA196-E851-4D5B-B1B5-2C2F32F6BB68@freeswitch.org>
	<eb7797d71002071929pb53e1adue2e91e6bdb598251@mail.gmail.com>
	<eb7797d71002080737k3f8fa5c9m8bf3ef2fcf42881e@mail.gmail.com>
	<D753EDBD-6ED7-43C2-B5F5-A899E3C3D3E2@jerris.com>
	<eb7797d71002121904v5bdfac87w2684c461ca1770cc@mail.gmail.com>
	<191c3a031002130714x2d876657p9c90d768d28a4496@mail.gmail.com>
	<eb7797d71002131018r56e86adnc054a3eaa2c05a96@mail.gmail.com>
	<eb7797d71002131132x2da4d2a0g54b8b8faf19520be@mail.gmail.com>
Message-ID: <191c3a031002131239v8d0a8e2xd42725a4682b54b2@mail.gmail.com>

Sigh

On Feb 13, 2010 1:38 PM, "Adam Wilt" <wiltingtree at gmail.com> wrote:

Here's a new debug log which shows what happens using only intercept and
having setAutoHangup(false) on both sides:

           http://pastebin.freeswitch.org/12124

<http://pastebin.freeswitch.org/12124>And here's my updated bridge function:

> function bridge_calls(session,api,b_leg_uuid, call_len)
  freeswitch.consoleLog("info","A leg - Bridging " .. tostring(b_leg_uuid)
.. " with " .. tostring(session.uuid) .. "\n")
  session:setAutoHangup(false)


>   session:execute("set","continue_on_fail=true")
  api:executeString("uuid_media " .. tostring(b_leg_uuid))
  api:executeString("uuid_media " .. tostring(session.uuid))


>   api:executeString("intercept -bleg " .. tostring(b_leg_uuid))
end


Thanks,
Adam




On Sat, Feb 13, 2010 at 1:18 PM, Adam Wilt <wiltingtree at gmail.com> wrote:
>
> Thanks Anthony and ...

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100213/093ad23b/attachment-0002.html 

From sos at sokhapkin.dyndns.org  Sat Feb 13 12:52:12 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Sat, 13 Feb 2010 15:52:12 -0500
Subject: [Freeswitch-users] bypass_media bug?
In-Reply-To: <201002121422.18544.sos@sokhapkin.dyndns.org>
References: <201002121422.18544.sos@sokhapkin.dyndns.org>
Message-ID: <201002131552.12289.sos@sokhapkin.dyndns.org>

More precisely, FS returns to A leg caller SDP from first 183, but not from 
final 200 OK.

On Friday 12 February 2010, Sergey Okhapkin wrote:
> Simple dialplan:
>
>     <extension name="103">
>       <condition field="destination_number" expression="^103$">
>         <action application="set" data="continue_on_fail=true"/>
>         <action application="set" data="bypass_media=true"/>
>         <action application="bridge" data="sofia/cwu/103 at 192.168.1.254"/>
>         <action application="bridge" data="sofia/cwu/104 at 192.168.1.254"/>
>       </condition>
>     </extension>
>
> 103 at 192.168.1.254 returns 183 early media and then "480 temporary
> unavailable", 104 at 192.168.1.254 answers the call (echo test).
>
> When 104 answers, Freeswitch returns to caller in SDP media port from
> "183", but not media port from 104's "200 OK"
>
> Is it a bug or expected behavior? If expected - is there a variable to
> control the behavior? Everything works OK if I replace bypass_media=true
> with bypass_media_after_bridge=true, but sending reinvites is not
> acceptable to me.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org





From paul at apcl.us  Sat Feb 13 12:59:05 2010
From: paul at apcl.us (Paul Levin)
Date: Sat, 13 Feb 2010 15:59:05 -0500
Subject: [Freeswitch-users] can vm settings go in
	conf\directory\default\default.xml?
Message-ID: <4B771299.8000001@apcl.us>

An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100213/46ebc760/attachment-0002.html 

From wiltingtree at gmail.com  Sat Feb 13 13:03:19 2010
From: wiltingtree at gmail.com (Adam Wilt)
Date: Sat, 13 Feb 2010 16:03:19 -0500
Subject: [Freeswitch-users] Bridging sessions from two separate lua
	scripts; uuid_bridge? intercept?
In-Reply-To: <191c3a031002131239v8d0a8e2xd42725a4682b54b2@mail.gmail.com>
References: <eb7797d71002071831w27301c5dva767b82b38c2ce6f@mail.gmail.com>
	<C23AA196-E851-4D5B-B1B5-2C2F32F6BB68@freeswitch.org>
	<eb7797d71002071929pb53e1adue2e91e6bdb598251@mail.gmail.com>
	<eb7797d71002080737k3f8fa5c9m8bf3ef2fcf42881e@mail.gmail.com>
	<D753EDBD-6ED7-43C2-B5F5-A899E3C3D3E2@jerris.com>
	<eb7797d71002121904v5bdfac87w2684c461ca1770cc@mail.gmail.com>
	<191c3a031002130714x2d876657p9c90d768d28a4496@mail.gmail.com>
	<eb7797d71002131018r56e86adnc054a3eaa2c05a96@mail.gmail.com>
	<eb7797d71002131132x2da4d2a0g54b8b8faf19520be@mail.gmail.com>
	<191c3a031002131239v8d0a8e2xd42725a4682b54b2@mail.gmail.com>
Message-ID: <eb7797d71002131303q19b657dv91b7ae08ab5b34cf@mail.gmail.com>

Does "sigh" mean that it's a problem with FreeSWITCH, or perhaps that I did
something stupid?



On Sat, Feb 13, 2010 at 3:39 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> Sigh
>
> On Feb 13, 2010 1:38 PM, "Adam Wilt" <wiltingtree at gmail.com> wrote:
>
> Here's a new debug log which shows what happens using only intercept and
> having setAutoHangup(false) on both sides:
>
>            http://pastebin.freeswitch.org/12124
>
> <http://pastebin.freeswitch.org/12124>And here's my updated bridge
> function:
>
> > function bridge_calls(session,api,b_leg_uuid, call_len)
>   freeswitch.consoleLog("info","A leg - Bridging " .. tostring(b_leg_uuid)
> .. " with " .. tostring(session.uuid) .. "\n")
>   session:setAutoHangup(false)
>
>
> >   session:execute("set","continue_on_fail=true")
>   api:executeString("uuid_media " .. tostring(b_leg_uuid))
>   api:executeString("uuid_media " .. tostring(session.uuid))
>
>
> >   api:executeString("intercept -bleg " .. tostring(b_leg_uuid))
> end
>
>
> Thanks,
> Adam
>
>
>
>
> On Sat, Feb 13, 2010 at 1:18 PM, Adam Wilt <wiltingtree at gmail.com> wrote:
> >
> > Thanks Anthony and ...
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100213/7c50c0b2/attachment-0002.html 

From mike at jerris.com  Sat Feb 13 13:21:04 2010
From: mike at jerris.com (Michael Jerris)
Date: Sat, 13 Feb 2010 16:21:04 -0500
Subject: [Freeswitch-users] mod_python fetching data from mysql for VM
In-Reply-To: <28f27f5d1002131203j132c6230sa260358611a16247@mail.gmail.com>
References: <28f27f5d1002131203j132c6230sa260358611a16247@mail.gmail.com>
Message-ID: <8E4B95E9-3914-4FEE-9C8C-9A5B4C5E2AB2@jerris.com>

Can you describe what your trying to accomplish, I don't understand what the goal is.  What feature are you looking for that does not already exist in mod_voiceamil.

Mike

On Feb 13, 2010, at 3:03 PM, mbsip wrote:

> Hello,
> 
> I am trying to use mod_python to fetch data from Mysql db (through
> ODBC) and execute voicemail application.
> Below a part of my script:
> 
> db=MySQLdb.connect("localhost","root","","test")
>       Cursor=db.cursor()
>       sql = "SELECT email FROM VM WHERE called_num=%s" % the_dest
>       Cursor.execute(sql)
>       while (1):
>               Results = Cursor.fetchone()
>               if Results == None:
>                       break
>               consoleLog("debug", "Found email " + Results[0] +"\n")
>               the_recipient = Results[0]
>       db.close()
> 
> Now i have email address corresponding with called number. The
> question is how to use it for voicemail application?
> So it also means how to omit all /directory/default/....xml, where
> there are all VM parameters set and use fetched data.
> 
>       session.answer()
>       session.execute("voicemail", "default ${domain} " + the_dest)
> 
> Is this possible or should I start all VM app in python from the scratch?




From mbsip at gazeta.pl  Sat Feb 13 13:23:02 2010
From: mbsip at gazeta.pl (mbsip)
Date: Sat, 13 Feb 2010 22:23:02 +0100
Subject: [Freeswitch-users] can vm settings go in
	conf\directory\default\default.xml?
In-Reply-To: <4B771299.8000001@apcl.us>
References: <4B771299.8000001@apcl.us>
Message-ID: <28f27f5d1002131323n86f03e9wdb60b784559b291f@mail.gmail.com>

Have you tried doing the same with
/usr/local/freeswitch/conf/directory/default.xml ?

Maciej

2010/2/13 Paul Levin <paul at apcl.us>:
> I'm running FS on Windows (in case that matters here).
>
> In conf\directory\default\Bob.xml I have the settings:
>
> ??? ? <param name="vm-email-all-messages" value="true"/>
> ??? ? <param name="vm-keep-local-after-email" value="false"/>
> ??? ? <param name="vm-attach-file" value="true"/>
>
> in addition to other vm- setting that are specific to Bob.? When a voice
> mail is left for Bob, an email is sent to the configured email address.? It
> is working well.? When the email is sent, I can see in the console the
> lines:
>
> 2010-02-10 15:41:36.949484 [DEBUG]
> switch_utils.c:633 Emailed file [C:\WINDOWS\TEMP\mail.12658416960810]
> to [bob at domain.com]
>
> 2010-02-10 15:41:36.949484 [DEBUG]
> mod_voicemail.c:2541 Sending message to bob at domain.com
>
> I then move those 3 <param> lines into the default\default.xml file.? Now
> when a voice mail is left for Bob, an email is not sent and those debug
> lines do not appear on the console.
>
> I don't mind keeping those 3 lines in each user file, but I'm expecting to
> have about 10,000 users and its kinda silly to repeat those lines in each
> user's file.? Can't they go in the default.xml file (and have it work)?
>
> ??? Thanks,
> ??? Paul
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From gavin.henry at gmail.com  Sat Feb 13 13:26:05 2010
From: gavin.henry at gmail.com (Gavin Henry)
Date: Sat, 13 Feb 2010 21:26:05 +0000
Subject: [Freeswitch-users] Mod_Conference capacity....
In-Reply-To: <E2A03ABF-69BF-4F85-9BB9-7CEAA7738E64@freeswitch.org>
References: <191c3a031002120856q93e52cep14ee96f1e7ed6d1@mail.gmail.com>
	<324899354.1084631265995845519.JavaMail.root@mail-2.01.com>
	<191c3a031002120944u346a3f7cq779f5c9e0ae31522@mail.gmail.com>
	<13ca621c1002121436n9b44dcexffeaafe998d4c88a@mail.gmail.com>
	<65d96fc81002130327q66d8227ao29ae46f8c7ee556a@mail.gmail.com>
	<191c3a031002130722v5c282c00s69f0b51b91e42520@mail.gmail.com>
	<13ca621c1002131120u482f6420xea1f7cad534cbe1a@mail.gmail.com>
	<E2A03ABF-69BF-4F85-9BB9-7CEAA7738E64@freeswitch.org>
Message-ID: <13ca621c1002131326p38eb73bcs444f57b26fdfe381@mail.gmail.com>

On 13 February 2010 19:54, Brian West <brian at freeswitch.org> wrote:
> This is the FAQ entry. ?It applies to EVERYTHING related to load testing.
> /b
> On Feb 13, 2010, at 1:20 PM, Gavin Henry wrote:
>
> http://wiki.freeswitch.org/wiki/FreeSwitch_FAQ#Q:_How_many_concurrent_calls_can_it_support.3F__Any_benchmarks.3F
>

Hi Brian,

OK, then It should clearly say what Anthony said or something like it:

"Please do not ask this question on the mailing lists as you will
always get the same official response from the FreeSWITCH project; "we
only perform benchmarking and confirm these results per FreeSWITCH
deployment, as each deployment will result in varying figures.
Commercial support is available from the project for this task. The
project has learned from experience the dangers of entertaining such
questions and its policy is to not do so over the free public forum."

Thanks.



-- 
http://www.suretecsystems.com/services/openldap/
http://www.suretectelecom.com



From mcampbellsmith at gmail.com  Sat Feb 13 13:36:14 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Sun, 14 Feb 2010 08:36:14 +1100
Subject: [Freeswitch-users] Forbidden using UDP, works with TCP/TLS
In-Reply-To: <C37AB6F3-F712-41B6-9D6D-7A8AE3C6CABB@freeswitch.org>
References: <33c87fa31002101824h21e3bf7lb13b68e8d4fd36a2@mail.gmail.com>
	<2EF27EFC-1634-46A3-B438-2EE99BEF58F1@jerris.com>
	<33c87fa31002110131w3cf6ad00n802669856085a879@mail.gmail.com>
	<CA21231C-108B-4445-A3B9-BF8D94FFE898@jerris.com>
	<33c87fa31002110213q1de0eebfsb950d8725ea9aba@mail.gmail.com>
	<0A905834-850F-4D1F-AEC5-D3A87CE4141B@jerris.com>
	<33c87fa31002121629k74f26807i131841aa4252461b@mail.gmail.com>
	<33c87fa31002130343k7a01e9abt9c33360399f246f8@mail.gmail.com>
	<C37AB6F3-F712-41B6-9D6D-7A8AE3C6CABB@freeswitch.org>
Message-ID: <33c87fa31002131336j7b05db7aof6ba6e3439aebc41@mail.gmail.com>

Thanks Brian.

The full log is pasted here http://pastebin.freeswitch.org/12133



On Sun, Feb 14, 2010 at 2:08 AM, Brian West <brian at freeswitch.org> wrote:
> Can't tell since you keep cutting the lines required to figure this out... FULL console log with FULL sip trace. ?You always cut out what I wanna see. ?Just get a pcap.
>
> /b
>
> On Feb 13, 2010, at 5:43 AM, Mark Campbell-Smith wrote:
>
>> More testing. The device registers successfully to my SIP provider
>> directly using UDP - why would FS be rejecting the registration
>> request?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From mbsip at gazeta.pl  Sat Feb 13 13:54:11 2010
From: mbsip at gazeta.pl (mbsip)
Date: Sat, 13 Feb 2010 22:54:11 +0100
Subject: [Freeswitch-users] mod_python fetching data from mysql for VM
In-Reply-To: <8E4B95E9-3914-4FEE-9C8C-9A5B4C5E2AB2@jerris.com>
References: <28f27f5d1002131203j132c6230sa260358611a16247@mail.gmail.com>
	<8E4B95E9-3914-4FEE-9C8C-9A5B4C5E2AB2@jerris.com>
Message-ID: <28f27f5d1002131354k23a0eaf5jc51b05106c68d719@mail.gmail.com>

Thx for prompt reply.

The main task is to be able to use Mysql db in conjunction with VM
(but not only voicemail_msgs, voicemail_prefs).

Lets imagine sb is calling 1000 and wants to record the message.
According to mod_voicemail settings message should be sent to some
email address.
But the information about user 1000 and his settings like email
address, passwd, quota should be fetched from Mysql db, not from
directory/default/1000.xml.
That's why I am using <action application="python" data="test"/> in my
dialplan to work with python script which in turn should do the magic.
The script should be able to gather all necessery data about user 1000
(like email address in shown example) and use them in VM.

So the problem is how to modify the script to force voicemail app to
use data from DB.
Currently  session.execute("voicemail", "default ${domain} " +
the_dest) is still using .xml files.

Thx,
Maciej.


2010/2/13 Michael Jerris <mike at jerris.com>:
> Can you describe what your trying to accomplish, I don't understand what the goal is. ?What feature are you looking for that does not already exist in mod_voiceamil.
>
> Mike
>
> On Feb 13, 2010, at 3:03 PM, mbsip wrote:
>
>> Hello,
>>
>> I am trying to use mod_python to fetch data from Mysql db (through
>> ODBC) and execute voicemail application.
>> Below a part of my script:
>>
>> db=MySQLdb.connect("localhost","root","","test")
>> ? ? ? Cursor=db.cursor()
>> ? ? ? sql = "SELECT email FROM VM WHERE called_num=%s" % the_dest
>> ? ? ? Cursor.execute(sql)
>> ? ? ? while (1):
>> ? ? ? ? ? ? ? Results = Cursor.fetchone()
>> ? ? ? ? ? ? ? if Results == None:
>> ? ? ? ? ? ? ? ? ? ? ? break
>> ? ? ? ? ? ? ? consoleLog("debug", "Found email " + Results[0] +"\n")
>> ? ? ? ? ? ? ? the_recipient = Results[0]
>> ? ? ? db.close()
>>
>> Now i have email address corresponding with called number. The
>> question is how to use it for voicemail application?
>> So it also means how to omit all /directory/default/....xml, where
>> there are all VM parameters set and use fetched data.
>>
>> ? ? ? session.answer()
>> ? ? ? session.execute("voicemail", "default ${domain} " + the_dest)
>>
>> Is this possible or should I start all VM app in python from the scratch?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From mbsip at gazeta.pl  Sat Feb 13 13:59:13 2010
From: mbsip at gazeta.pl (mbsip)
Date: Sat, 13 Feb 2010 22:59:13 +0100
Subject: [Freeswitch-users] mod_python fetching data from mysql for VM
In-Reply-To: <28f27f5d1002131354k23a0eaf5jc51b05106c68d719@mail.gmail.com>
References: <28f27f5d1002131203j132c6230sa260358611a16247@mail.gmail.com>
	<8E4B95E9-3914-4FEE-9C8C-9A5B4C5E2AB2@jerris.com>
	<28f27f5d1002131354k23a0eaf5jc51b05106c68d719@mail.gmail.com>
Message-ID: <28f27f5d1002131359h164e4077x326f646ff2367ad9@mail.gmail.com>

There is a lack of connection between fatched data and voicemail and I
dont know how to achieve it.

Thx,
Maciej.


2010/2/13 mbsip <mbsip at gazeta.pl>:
> Thx for prompt reply.
>
> The main task is to be able to use Mysql db in conjunction with VM
> (but not only voicemail_msgs, voicemail_prefs).
>
> Lets imagine sb is calling 1000 and wants to record the message.
> According to mod_voicemail settings message should be sent to some
> email address.
> But the information about user 1000 and his settings like email
> address, passwd, quota should be fetched from Mysql db, not from
> directory/default/1000.xml.
> That's why I am using <action application="python" data="test"/> in my
> dialplan to work with python script which in turn should do the magic.
> The script should be able to gather all necessery data about user 1000
> (like email address in shown example) and use them in VM.
>
> So the problem is how to modify the script to force voicemail app to
> use data from DB.
> Currently ?session.execute("voicemail", "default ${domain} " +
> the_dest) is still using .xml files.
>
> Thx,
> Maciej.
>
>
> 2010/2/13 Michael Jerris <mike at jerris.com>:
>> Can you describe what your trying to accomplish, I don't understand what the goal is. ?What feature are you looking for that does not already exist in mod_voiceamil.
>>
>> Mike
>>
>> On Feb 13, 2010, at 3:03 PM, mbsip wrote:
>>
>>> Hello,
>>>
>>> I am trying to use mod_python to fetch data from Mysql db (through
>>> ODBC) and execute voicemail application.
>>> Below a part of my script:
>>>
>>> db=MySQLdb.connect("localhost","root","","test")
>>> ? ? ? Cursor=db.cursor()
>>> ? ? ? sql = "SELECT email FROM VM WHERE called_num=%s" % the_dest
>>> ? ? ? Cursor.execute(sql)
>>> ? ? ? while (1):
>>> ? ? ? ? ? ? ? Results = Cursor.fetchone()
>>> ? ? ? ? ? ? ? if Results == None:
>>> ? ? ? ? ? ? ? ? ? ? ? break
>>> ? ? ? ? ? ? ? consoleLog("debug", "Found email " + Results[0] +"\n")
>>> ? ? ? ? ? ? ? the_recipient = Results[0]
>>> ? ? ? db.close()
>>>
>>> Now i have email address corresponding with called number. The
>>> question is how to use it for voicemail application?
>>> So it also means how to omit all /directory/default/....xml, where
>>> there are all VM parameters set and use fetched data.
>>>
>>> ? ? ? session.answer()
>>> ? ? ? session.execute("voicemail", "default ${domain} " + the_dest)
>>>
>>> Is this possible or should I start all VM app in python from the scratch?
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>



From jmesquita at freeswitch.org  Sat Feb 13 14:00:05 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Sat, 13 Feb 2010 19:00:05 -0300
Subject: [Freeswitch-users] mod_python fetching data from mysql for VM
In-Reply-To: <28f27f5d1002131203j132c6230sa260358611a16247@mail.gmail.com>
References: <28f27f5d1002131203j132c6230sa260358611a16247@mail.gmail.com>
Message-ID: <ebbc2d551002131400k7fbfca7fm4b72fe545ae37e22@mail.gmail.com>

Not sure what you are asking since you provided the answer yourself...

Yes it is possible and that is the way to do it.

JM


On Sat, Feb 13, 2010 at 5:03 PM, mbsip <mbsip at gazeta.pl> wrote:

> Hello,
>
> I am trying to use mod_python to fetch data from Mysql db (through
> ODBC) and execute voicemail application.
> Below a part of my script:
>
> db=MySQLdb.connect("localhost","root","","test")
>       Cursor=db.cursor()
>       sql = "SELECT email FROM VM WHERE called_num=%s" % the_dest
>       Cursor.execute(sql)
>       while (1):
>               Results = Cursor.fetchone()
>               if Results == None:
>                       break
>               consoleLog("debug", "Found email " + Results[0] +"\n")
>               the_recipient = Results[0]
>       db.close()
>
> Now i have email address corresponding with called number. The
> question is how to use it for voicemail application?
> So it also means how to omit all /directory/default/....xml, where
> there are all VM parameters set and use fetched data.
>
>       session.answer()
>       session.execute("voicemail", "default ${domain} " + the_dest)
>
> Is this possible or should I start all VM app in python from the scratch?
>
> Thanks,
> Maciej
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100213/8fc53e30/attachment-0002.html 

From jmesquita at freeswitch.org  Sat Feb 13 14:02:51 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Sat, 13 Feb 2010 19:02:51 -0300
Subject: [Freeswitch-users] Mod_Conference capacity....
In-Reply-To: <13ca621c1002131326p38eb73bcs444f57b26fdfe381@mail.gmail.com>
References: <191c3a031002120856q93e52cep14ee96f1e7ed6d1@mail.gmail.com>
	<324899354.1084631265995845519.JavaMail.root@mail-2.01.com>
	<191c3a031002120944u346a3f7cq779f5c9e0ae31522@mail.gmail.com>
	<13ca621c1002121436n9b44dcexffeaafe998d4c88a@mail.gmail.com>
	<65d96fc81002130327q66d8227ao29ae46f8c7ee556a@mail.gmail.com>
	<191c3a031002130722v5c282c00s69f0b51b91e42520@mail.gmail.com>
	<13ca621c1002131120u482f6420xea1f7cad534cbe1a@mail.gmail.com>
	<E2A03ABF-69BF-4F85-9BB9-7CEAA7738E64@freeswitch.org>
	<13ca621c1002131326p38eb73bcs444f57b26fdfe381@mail.gmail.com>
Message-ID: <ebbc2d551002131402o66f6d641p13e0c7dcef29c8be@mail.gmail.com>

The wiki is public, you know?

JM


On Sat, Feb 13, 2010 at 6:26 PM, Gavin Henry <gavin.henry at gmail.com> wrote:

> On 13 February 2010 19:54, Brian West <brian at freeswitch.org> wrote:
> > This is the FAQ entry.  It applies to EVERYTHING related to load testing.
> > /b
> > On Feb 13, 2010, at 1:20 PM, Gavin Henry wrote:
> >
> >
> http://wiki.freeswitch.org/wiki/FreeSwitch_FAQ#Q:_How_many_concurrent_calls_can_it_support.3F__Any_benchmarks.3F
> >
>
> Hi Brian,
>
> OK, then It should clearly say what Anthony said or something like it:
>
> "Please do not ask this question on the mailing lists as you will
> always get the same official response from the FreeSWITCH project; "we
> only perform benchmarking and confirm these results per FreeSWITCH
> deployment, as each deployment will result in varying figures.
> Commercial support is available from the project for this task. The
> project has learned from experience the dangers of entertaining such
> questions and its policy is to not do so over the free public forum."
>
> Thanks.
>
>
>
> --
> http://www.suretecsystems.com/services/openldap/
> http://www.suretectelecom.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100213/0d9a77ae/attachment-0002.html 

From peder at networkoblivion.com  Sat Feb 13 14:07:45 2010
From: peder at networkoblivion.com (Peder)
Date: Sat, 13 Feb 2010 16:07:45 -0600
Subject: [Freeswitch-users] Mod_Conference capacity....
In-Reply-To: <13ca621c1002131326p38eb73bcs444f57b26fdfe381@mail.gmail.com>
References: <191c3a031002120856q93e52cep14ee96f1e7ed6d1@mail.gmail.com>	<324899354.1084631265995845519.JavaMail.root@mail-2.01.com>	<191c3a031002120944u346a3f7cq779f5c9e0ae31522@mail.gmail.com>	<13ca621c1002121436n9b44dcexffeaafe998d4c88a@mail.gmail.com>	<65d96fc81002130327q66d8227ao29ae46f8c7ee556a@mail.gmail.com>	<191c3a031002130722v5c282c00s69f0b51b91e42520@mail.gmail.com>	<13ca621c1002131120u482f6420xea1f7cad534cbe1a@mail.gmail.com>	<E2A03ABF-69BF-4F85-9BB9-7CEAA7738E64@freeswitch.org>
	<13ca621c1002131326p38eb73bcs444f57b26fdfe381@mail.gmail.com>
Message-ID: <0ad201caacf8$f8c35c20$ea4a1460$@com>

Instead of complaining about this, why don't you benchmark it yourself on
your hardware and post your results on the wiki for others to see in the
future?  Anybody can edit and add info to the wiki.  

By the way, if you stay on the list for any length of time you will see that
about once a week, someone says "I want to do load testing, please tell me
how", or "I did a load test and didn't get x calls per second, why".  I'm
not even involved in the development of this project and it gets old seeing
that same question over and over again.  Really, the standard answer should
be "there is no software limit, it depends on your hardware".  If you want
to know the limits of your setup, test it and see what it is.  


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Gavin
Henry
Sent: Saturday, February 13, 2010 3:26 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Mod_Conference capacity....

On 13 February 2010 19:54, Brian West <brian at freeswitch.org> wrote:
> This is the FAQ entry. ?It applies to EVERYTHING related to load testing.
> /b
> On Feb 13, 2010, at 1:20 PM, Gavin Henry wrote:
>
>
http://wiki.freeswitch.org/wiki/FreeSwitch_FAQ#Q:_How_many_concurrent_calls_
can_it_support.3F__Any_benchmarks.3F
>

Hi Brian,

OK, then It should clearly say what Anthony said or something like it:

"Please do not ask this question on the mailing lists as you will
always get the same official response from the FreeSWITCH project; "we
only perform benchmarking and confirm these results per FreeSWITCH
deployment, as each deployment will result in varying figures.
Commercial support is available from the project for this task. The
project has learned from experience the dangers of entertaining such
questions and its policy is to not do so over the free public forum."

Thanks.



-- 
http://www.suretecsystems.com/services/openldap/
http://www.suretectelecom.com

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




From jmesquita at freeswitch.org  Sat Feb 13 14:20:00 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Sat, 13 Feb 2010 19:20:00 -0300
Subject: [Freeswitch-users] mod_python fetching data from mysql for VM
In-Reply-To: <28f27f5d1002131359h164e4077x326f646ff2367ad9@mail.gmail.com>
References: <28f27f5d1002131203j132c6230sa260358611a16247@mail.gmail.com>
	<8E4B95E9-3914-4FEE-9C8C-9A5B4C5E2AB2@jerris.com>
	<28f27f5d1002131354k23a0eaf5jc51b05106c68d719@mail.gmail.com>
	<28f27f5d1002131359h164e4077x326f646ff2367ad9@mail.gmail.com>
Message-ID: <ebbc2d551002131420v58cf95f2lfc9356f2377728b6@mail.gmail.com>

Maciej,

Take a look at the xml_hooks we have on mod_python. Might do the trick for
you.

http://svn.freeswitch.org/svn/freeswitch/trunk/src/mod/languages/mod_python/python_example.py

JM


On Sat, Feb 13, 2010 at 6:59 PM, mbsip <mbsip at gazeta.pl> wrote:

> There is a lack of connection between fatched data and voicemail and I
> dont know how to achieve it.
>
> Thx,
> Maciej.
>
>
> 2010/2/13 mbsip <mbsip at gazeta.pl>:
> > Thx for prompt reply.
> >
> > The main task is to be able to use Mysql db in conjunction with VM
> > (but not only voicemail_msgs, voicemail_prefs).
> >
> > Lets imagine sb is calling 1000 and wants to record the message.
> > According to mod_voicemail settings message should be sent to some
> > email address.
> > But the information about user 1000 and his settings like email
> > address, passwd, quota should be fetched from Mysql db, not from
> > directory/default/1000.xml.
> > That's why I am using <action application="python" data="test"/> in my
> > dialplan to work with python script which in turn should do the magic.
> > The script should be able to gather all necessery data about user 1000
> > (like email address in shown example) and use them in VM.
> >
> > So the problem is how to modify the script to force voicemail app to
> > use data from DB.
> > Currently  session.execute("voicemail", "default ${domain} " +
> > the_dest) is still using .xml files.
> >
> > Thx,
> > Maciej.
> >
> >
> > 2010/2/13 Michael Jerris <mike at jerris.com>:
> >> Can you describe what your trying to accomplish, I don't understand what
> the goal is.  What feature are you looking for that does not already exist
> in mod_voiceamil.
> >>
> >> Mike
> >>
> >> On Feb 13, 2010, at 3:03 PM, mbsip wrote:
> >>
> >>> Hello,
> >>>
> >>> I am trying to use mod_python to fetch data from Mysql db (through
> >>> ODBC) and execute voicemail application.
> >>> Below a part of my script:
> >>>
> >>> db=MySQLdb.connect("localhost","root","","test")
> >>>       Cursor=db.cursor()
> >>>       sql = "SELECT email FROM VM WHERE called_num=%s" % the_dest
> >>>       Cursor.execute(sql)
> >>>       while (1):
> >>>               Results = Cursor.fetchone()
> >>>               if Results == None:
> >>>                       break
> >>>               consoleLog("debug", "Found email " + Results[0] +"\n")
> >>>               the_recipient = Results[0]
> >>>       db.close()
> >>>
> >>> Now i have email address corresponding with called number. The
> >>> question is how to use it for voicemail application?
> >>> So it also means how to omit all /directory/default/....xml, where
> >>> there are all VM parameters set and use fetched data.
> >>>
> >>>       session.answer()
> >>>       session.execute("voicemail", "default ${domain} " + the_dest)
> >>>
> >>> Is this possible or should I start all VM app in python from the
> scratch?
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100213/6bcb289f/attachment-0002.html 

From gavin.henry at gmail.com  Sat Feb 13 14:33:40 2010
From: gavin.henry at gmail.com (Gavin Henry)
Date: Sat, 13 Feb 2010 22:33:40 +0000
Subject: [Freeswitch-users] Mod_Conference capacity....
In-Reply-To: <ebbc2d551002131402o66f6d641p13e0c7dcef29c8be@mail.gmail.com>
References: <191c3a031002120856q93e52cep14ee96f1e7ed6d1@mail.gmail.com>
	<324899354.1084631265995845519.JavaMail.root@mail-2.01.com>
	<191c3a031002120944u346a3f7cq779f5c9e0ae31522@mail.gmail.com>
	<13ca621c1002121436n9b44dcexffeaafe998d4c88a@mail.gmail.com>
	<65d96fc81002130327q66d8227ao29ae46f8c7ee556a@mail.gmail.com>
	<191c3a031002130722v5c282c00s69f0b51b91e42520@mail.gmail.com>
	<13ca621c1002131120u482f6420xea1f7cad534cbe1a@mail.gmail.com>
	<E2A03ABF-69BF-4F85-9BB9-7CEAA7738E64@freeswitch.org>
	<13ca621c1002131326p38eb73bcs444f57b26fdfe381@mail.gmail.com>
	<ebbc2d551002131402o66f6d641p13e0c7dcef29c8be@mail.gmail.com>
Message-ID: <13ca621c1002131433w7963879bqcab11c278387356d@mail.gmail.com>

2010/2/13 Jo?o Mesquita <jmesquita at freeswitch.org>:
> The wiki is public, you know?
>

Of course I know. I don't represent the project so couldn't obviously
add that so it was in anyway official. I'm not trying to be picky here
only make things clear in the docs for when the next person comes
along and the FreeSWITCH team can just answer an email with the link
to the FAQ on the wiki.

-- 
http://www.suretecsystems.com/services/openldap/
http://www.suretectelecom.com



From paul at apcl.us  Sat Feb 13 14:35:23 2010
From: paul at apcl.us (Paul Levin)
Date: Sat, 13 Feb 2010 17:35:23 -0500
Subject: [Freeswitch-users] can vm settings go in
 conf\directory\default\default.xml?
Message-ID: <4B77292B.6080207@apcl.us>

An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100213/6f607f72/attachment-0002.html 

From gavin.henry at gmail.com  Sat Feb 13 14:38:06 2010
From: gavin.henry at gmail.com (Gavin Henry)
Date: Sat, 13 Feb 2010 22:38:06 +0000
Subject: [Freeswitch-users] Mod_Conference capacity....
In-Reply-To: <0ad201caacf8$f8c35c20$ea4a1460$@com>
References: <191c3a031002120856q93e52cep14ee96f1e7ed6d1@mail.gmail.com>
	<324899354.1084631265995845519.JavaMail.root@mail-2.01.com>
	<191c3a031002120944u346a3f7cq779f5c9e0ae31522@mail.gmail.com>
	<13ca621c1002121436n9b44dcexffeaafe998d4c88a@mail.gmail.com>
	<65d96fc81002130327q66d8227ao29ae46f8c7ee556a@mail.gmail.com>
	<191c3a031002130722v5c282c00s69f0b51b91e42520@mail.gmail.com>
	<13ca621c1002131120u482f6420xea1f7cad534cbe1a@mail.gmail.com>
	<E2A03ABF-69BF-4F85-9BB9-7CEAA7738E64@freeswitch.org>
	<13ca621c1002131326p38eb73bcs444f57b26fdfe381@mail.gmail.com>
	<0ad201caacf8$f8c35c20$ea4a1460$@com>
Message-ID: <13ca621c1002131438t78864be4y2ef6e0f19b14f991@mail.gmail.com>

On 13 February 2010 22:07, Peder <peder at networkoblivion.com> wrote:
> Instead of complaining about this, why don't you benchmark it yourself on
> your hardware and post your results on the wiki for others to see in the
> future? ?Anybody can edit and add info to the wiki.
>
> By the way, if you stay on the list for any length of time you will see that
> about once a week, someone says "I want to do load testing, please tell me
> how", or "I did a load test and didn't get x calls per second, why". ?I'm
> not even involved in the development of this project and it gets old seeing
> that same question over and over again. ?Really, the standard answer should
> be "there is no software limit, it depends on your hardware". ?If you want
> to know the limits of your setup, test it and see what it is.

Did you read any of this thread at all? The OP asked about techniques
to benchmark his kit, and any rough
figures. The projects answer was to hire them for some commercial
support. So anything I post on the wiki is not
official. My suggestion is merely to put the official project stance
on any benchmark questions or post to the
wiki some tips on how to run your own tests with a link to getting
them verified by the team via the commercial
support.

In no way am I complaining, just trying to get this cleared up for the
next time someone asks. That's the whole
point of docs and why I enjoy writing them. Write them once and refer
people to them, fixing doc bugs along the
way.

Gavin.



-- 
http://www.suretecsystems.com/services/openldap/
http://www.suretectelecom.com



From mbsip at gazeta.pl  Sat Feb 13 14:38:17 2010
From: mbsip at gazeta.pl (mbsip)
Date: Sat, 13 Feb 2010 23:38:17 +0100
Subject: [Freeswitch-users] mod_python fetching data from mysql for VM
In-Reply-To: <ebbc2d551002131420v58cf95f2lfc9356f2377728b6@mail.gmail.com>
References: <28f27f5d1002131203j132c6230sa260358611a16247@mail.gmail.com>
	<8E4B95E9-3914-4FEE-9C8C-9A5B4C5E2AB2@jerris.com>
	<28f27f5d1002131354k23a0eaf5jc51b05106c68d719@mail.gmail.com>
	<28f27f5d1002131359h164e4077x326f646ff2367ad9@mail.gmail.com>
	<ebbc2d551002131420v58cf95f2lfc9356f2377728b6@mail.gmail.com>
Message-ID: <28f27f5d1002131438l58718bbat304407f49397c965@mail.gmail.com>

Jo?o,

Thanks for hint, because i don't know how the db fetched data could be
used with voicemail.
I am about to ready it carefully :P

Thanks,
Maciej

2010/2/13 Jo?o Mesquita <jmesquita at freeswitch.org>:
> Maciej,
>
> Take a look at the xml_hooks we have on mod_python. Might do the trick for
> you.
>
> http://svn.freeswitch.org/svn/freeswitch/trunk/src/mod/languages/mod_python/python_example.py
>
> JM
>
>
> On Sat, Feb 13, 2010 at 6:59 PM, mbsip <mbsip at gazeta.pl> wrote:
>>
>> There is a lack of connection between fatched data and voicemail and I
>> dont know how to achieve it.
>>
>> Thx,
>> Maciej.
>>
>>
>> 2010/2/13 mbsip <mbsip at gazeta.pl>:
>> > Thx for prompt reply.
>> >
>> > The main task is to be able to use Mysql db in conjunction with VM
>> > (but not only voicemail_msgs, voicemail_prefs).
>> >
>> > Lets imagine sb is calling 1000 and wants to record the message.
>> > According to mod_voicemail settings message should be sent to some
>> > email address.
>> > But the information about user 1000 and his settings like email
>> > address, passwd, quota should be fetched from Mysql db, not from
>> > directory/default/1000.xml.
>> > That's why I am using <action application="python" data="test"/> in my
>> > dialplan to work with python script which in turn should do the magic.
>> > The script should be able to gather all necessery data about user 1000
>> > (like email address in shown example) and use them in VM.
>> >
>> > So the problem is how to modify the script to force voicemail app to
>> > use data from DB.
>> > Currently ?session.execute("voicemail", "default ${domain} " +
>> > the_dest) is still using .xml files.
>> >
>> > Thx,
>> > Maciej.
>> >
>> >
>> > 2010/2/13 Michael Jerris <mike at jerris.com>:
>> >> Can you describe what your trying to accomplish, I don't understand
>> >> what the goal is. ?What feature are you looking for that does not already
>> >> exist in mod_voiceamil.
>> >>
>> >> Mike
>> >>
>> >> On Feb 13, 2010, at 3:03 PM, mbsip wrote:
>> >>
>> >>> Hello,
>> >>>
>> >>> I am trying to use mod_python to fetch data from Mysql db (through
>> >>> ODBC) and execute voicemail application.
>> >>> Below a part of my script:
>> >>>
>> >>> db=MySQLdb.connect("localhost","root","","test")
>> >>> ? ? ? Cursor=db.cursor()
>> >>> ? ? ? sql = "SELECT email FROM VM WHERE called_num=%s" % the_dest
>> >>> ? ? ? Cursor.execute(sql)
>> >>> ? ? ? while (1):
>> >>> ? ? ? ? ? ? ? Results = Cursor.fetchone()
>> >>> ? ? ? ? ? ? ? if Results == None:
>> >>> ? ? ? ? ? ? ? ? ? ? ? break
>> >>> ? ? ? ? ? ? ? consoleLog("debug", "Found email " + Results[0] +"\n")
>> >>> ? ? ? ? ? ? ? the_recipient = Results[0]
>> >>> ? ? ? db.close()
>> >>>
>> >>> Now i have email address corresponding with called number. The
>> >>> question is how to use it for voicemail application?
>> >>> So it also means how to omit all /directory/default/....xml, where
>> >>> there are all VM parameters set and use fetched data.
>> >>>
>> >>> ? ? ? session.answer()
>> >>> ? ? ? session.execute("voicemail", "default ${domain} " + the_dest)
>> >>>
>> >>> Is this possible or should I start all VM app in python from the
>> >>> scratch?
>> >>
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From dftoro at yahoo.com  Sat Feb 13 14:42:34 2010
From: dftoro at yahoo.com (Diego Toro)
Date: Sat, 13 Feb 2010 14:42:34 -0800 (PST)
Subject: [Freeswitch-users] signaling information SIP INFO
Message-ID: <650320.69699.qm@web33503.mail.mud.yahoo.com>

Hi,

I need send SIP INFO message where the body of the SIP message consists of signaling information to a gateway.

Example:

INFO sip:7007471000 at example.com SIP/2.0 
Via: SIP/2.0/UDP alice.uk.example.com:5060 
From: <sip:7007471234 at alice.uk.example.com>;tag=d3f423d 
To: <sip:7007471000 at example.com>;tag=8942 
Call-ID: 312352 at myphone 
CSeq: 5 INFO 
Content-Length: 24 
Content-Type: application/dtmf-relay 

Signal=16
Duration=160 

 
How I can do that with FreeSwitch events ?


Thank you

Diego Toro
http://lacarretade.blogspot.com/


      



From mbsip at gazeta.pl  Sat Feb 13 14:44:33 2010
From: mbsip at gazeta.pl (mbsip)
Date: Sat, 13 Feb 2010 23:44:33 +0100
Subject: [Freeswitch-users] can vm settings go in
	conf\directory\default\default.xml?
In-Reply-To: <4B77292B.6080207@apcl.us>
References: <4B77292B.6080207@apcl.us>
Message-ID: <28f27f5d1002131444h36df1ba7g7438dd685d8f4281@mail.gmail.com>

So now I am with You Paul. I have the same thoughs and problem :P

Thanks,
Maciej.

2010/2/13 Paul Levin <paul at apcl.us>:
> Thank you for the reply Maciej.
>
> Looks like I made an error in my first email, so let me repeat the problem.
> If I put the lines:
>
> <param name="vm-email-all-messages" value="true"/>
> <param name="vm-keep-local-after-email" value="false"/>
> <param name="vm-attach-file" value="true"/>
>
> into?? conf/directory/default/Bob.xml?? then emails for voice mail are sent.
>
> If I remove those three lines from Bob.xml and put them into
> conf/directory/default.xml?? then emails are not sent.
>
> I would have thought that putting those lines in
> conf/directory/default.xml? would remove the requirement to have them in
> Bob.xml.? No?
>
> ??? Thanks,
> ??? Paul
>
>
>
> From:
> mbsip <mbsip at gazeta.pl>
> Date:
> Sat, 13 Feb 2010 22:23:02 +0100
>
>> Have you tried doing the same with
>> /usr/local/freeswitch/conf/directory/default.xml ?
>
>> Maciej
>
> 2010/2/13 Paul Levin <paul at apcl.us>:
>
>> I'm running FS on Windows (in case that matters here).
>>
>> In conf\directory\default\Bob.xml I have the settings:
>>
>> ??? ? <param name="vm-email-all-messages" value="true"/>
>> ??? ? <param name="vm-keep-local-after-email" value="false"/>
>> ??? ? <param name="vm-attach-file" value="true"/>
>>
>> in addition to other vm- setting that are specific to Bob.? When a voice
>> mail is left for Bob, an email is sent to the configured email address.
>> It
>> is working well.? When the email is sent, I can see in the console the
>> lines:
>>
>> 2010-02-10 15:41:36.949484 [DEBUG]
>> switch_utils.c:633 Emailed file [C:\WINDOWS\TEMP\mail.12658416960810]
>> to [bob at domain.com]
>>
>> 2010-02-10 15:41:36.949484 [DEBUG]
>> mod_voicemail.c:2541 Sending message to bob at domain.com
>>
>> I then move those 3 <param> lines into the default\default.xml file.? Now
>> when a voice mail is left for Bob, an email is not sent and those debug
>> lines do not appear on the console.
>>
>> I don't mind keeping those 3 lines in each user file, but I'm expecting to
>> have about 10,000 users and its kinda silly to repeat those lines in each
>> user's file.? Can't they go in the default.xml file (and have it work)?
>>
>> ??? Thanks,
>> ??? Paul
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From anthony.minessale at gmail.com  Sat Feb 13 14:48:53 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 13 Feb 2010 16:48:53 -0600
Subject: [Freeswitch-users] Mod_Conference capacity....
In-Reply-To: <13ca621c1002131438t78864be4y2ef6e0f19b14f991@mail.gmail.com>
References: <191c3a031002120856q93e52cep14ee96f1e7ed6d1@mail.gmail.com>
	<191c3a031002120944u346a3f7cq779f5c9e0ae31522@mail.gmail.com>
	<13ca621c1002121436n9b44dcexffeaafe998d4c88a@mail.gmail.com>
	<65d96fc81002130327q66d8227ao29ae46f8c7ee556a@mail.gmail.com>
	<191c3a031002130722v5c282c00s69f0b51b91e42520@mail.gmail.com>
	<13ca621c1002131120u482f6420xea1f7cad534cbe1a@mail.gmail.com>
	<E2A03ABF-69BF-4F85-9BB9-7CEAA7738E64@freeswitch.org>
	<13ca621c1002131326p38eb73bcs444f57b26fdfe381@mail.gmail.com>
	<0ad201caacf8$f8c35c20$ea4a1460$@com>
	<13ca621c1002131438t78864be4y2ef6e0f19b14f991@mail.gmail.com>
Message-ID: <191c3a031002131448x59e42e22q6f931dd54834c0a7@mail.gmail.com>

Helping with docs, now there is a topic we all groove on.

On Feb 13, 2010 4:43 PM, "Gavin Henry" <gavin.henry at gmail.com> wrote:

On 13 February 2010 22:07, Peder <peder at networkoblivion.com> wrote:
> Instead of complaining about t...
Did you read any of this thread at all? The OP asked about techniques
to benchmark his kit, and any rough
figures. The projects answer was to hire them for some commercial
support. So anything I post on the wiki is not
official. My suggestion is merely to put the official project stance
on any benchmark questions or post to the
wiki some tips on how to run your own tests with a link to getting
them verified by the team via the commercial
support.

In no way am I complaining, just trying to get this cleared up for the
next time someone asks. That's the whole
point of docs and why I enjoy writing them. Write them once and refer
people to them, fixing doc bugs along the
way.

Gavin.




-- 
http://www.suretecsystems.com/services/openldap/
http://www.suretectelecom.com

_____________...
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100213/cc675829/attachment-0002.html 

From jason at jasonjgw.net  Sat Feb 13 14:53:20 2010
From: jason at jasonjgw.net (Jason White)
Date: Sun, 14 Feb 2010 09:53:20 +1100
Subject: [Freeswitch-users] FreeSWITCH Debian packages - build issues
In-Reply-To: <201002131109.35877.errotan@gmail.com>
References: <20100213030812.GA19108@jdc.jasonjgw.net>
	<201002130957.11633.errotan@gmail.com>
	<20100213090907.GA29452@jdc.jasonjgw.net>
	<201002131109.35877.errotan@gmail.com>
Message-ID: <20100213225320.GA4990@jdc.jasonjgw.net>

Pusk?s Zsolt <errotan at gmail.com> wrote:
 
> I just compiled fs using defaults (just uncommented the openzap line in 
> modues.conf). I don't know how to build a package for that but I can try if 
> you got some instructions how to do that for testing purposes.

The instructions are on the FreeSWITCH wiki for building Debian packages.

The problem appears not to be OpenZap; it's Memcache not finding libpthreads
during its ./configure step and failing at that point.




From rupa at rupa.com  Sat Feb 13 15:03:10 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Sat, 13 Feb 2010 17:03:10 -0600
Subject: [Freeswitch-users] FreeSWITCH Debian packages - build issues
In-Reply-To: <20100213225320.GA4990@jdc.jasonjgw.net>
References: <20100213030812.GA19108@jdc.jasonjgw.net>
	<201002130957.11633.errotan@gmail.com>
	<20100213090907.GA29452@jdc.jasonjgw.net>
	<201002131109.35877.errotan@gmail.com>
	<20100213225320.GA4990@jdc.jasonjgw.net>
Message-ID: <b39273531002131503q1fa89c27n1d7a0406967ed558@mail.gmail.com>

hmmmm... memcache builds for me but I don't do the deb builds, just
./configure (blah blah) && make.  Are we just missing a dependency or is it
configure missing something that is new/different in sid but not testing or
stable?

On Sat, Feb 13, 2010 at 4:53 PM, Jason White <jason at jasonjgw.net> wrote:

> Pusk?s Zsolt <errotan at gmail.com> wrote:
>
> > I just compiled fs using defaults (just uncommented the openzap line in
> > modues.conf). I don't know how to build a package for that but I can try
> if
> > you got some instructions how to do that for testing purposes.
>
> The instructions are on the FreeSWITCH wiki for building Debian packages.
>
> The problem appears not to be OpenZap; it's Memcache not finding
> libpthreads
> during its ./configure step and failing at that point.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
-Rupa
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100213/be19723e/attachment-0002.html 

From brian at freeswitch.org  Sat Feb 13 15:08:28 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 13 Feb 2010 17:08:28 -0600
Subject: [Freeswitch-users] Mod_Conference capacity....
In-Reply-To: <0ad201caacf8$f8c35c20$ea4a1460$@com>
References: <191c3a031002120856q93e52cep14ee96f1e7ed6d1@mail.gmail.com>	<324899354.1084631265995845519.JavaMail.root@mail-2.01.com>	<191c3a031002120944u346a3f7cq779f5c9e0ae31522@mail.gmail.com>	<13ca621c1002121436n9b44dcexffeaafe998d4c88a@mail.gmail.com>	<65d96fc81002130327q66d8227ao29ae46f8c7ee556a@mail.gmail.com>	<191c3a031002130722v5c282c00s69f0b51b91e42520@mail.gmail.com>	<13ca621c1002131120u482f6420xea1f7cad534cbe1a@mail.gmail.com>	<E2A03ABF-69BF-4F85-9BB9-7CEAA7738E64@freeswitch.org>
	<13ca621c1002131326p38eb73bcs444f57b26fdfe381@mail.gmail.com>
	<0ad201caacf8$f8c35c20$ea4a1460$@com>
Message-ID: <8ED2A793-9F68-4064-8593-6D142E099769@freeswitch.org>

We have install guides and yet people still have problems installing FreeSWITCH.  shrug!  More docs are a plus.

/b

On Feb 13, 2010, at 4:07 PM, Peder wrote:

> Instead of complaining about this, why don't you benchmark it yourself on
> your hardware and post your results on the wiki for others to see in the
> future?  Anybody can edit and add info to the wiki.  




From brian at freeswitch.org  Sat Feb 13 15:11:31 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 13 Feb 2010 17:11:31 -0600
Subject: [Freeswitch-users] Forbidden using UDP, works with TCP/TLS
In-Reply-To: <33c87fa31002131336j7b05db7aof6ba6e3439aebc41@mail.gmail.com>
References: <33c87fa31002101824h21e3bf7lb13b68e8d4fd36a2@mail.gmail.com>
	<2EF27EFC-1634-46A3-B438-2EE99BEF58F1@jerris.com>
	<33c87fa31002110131w3cf6ad00n802669856085a879@mail.gmail.com>
	<CA21231C-108B-4445-A3B9-BF8D94FFE898@jerris.com>
	<33c87fa31002110213q1de0eebfsb950d8725ea9aba@mail.gmail.com>
	<0A905834-850F-4D1F-AEC5-D3A87CE4141B@jerris.com>
	<33c87fa31002121629k74f26807i131841aa4252461b@mail.gmail.com>
	<33c87fa31002130343k7a01e9abt9c33360399f246f8@mail.gmail.com>
	<C37AB6F3-F712-41B6-9D6D-7A8AE3C6CABB@freeswitch.org>
	<33c87fa31002131336j7b05db7aof6ba6e3439aebc41@mail.gmail.com>
Message-ID: <9D702362-BB5F-4C0F-9EE7-DABA3B926C36@freeswitch.org>

You know you could have obscured the first part of the IP and not the LAST... kinda removes the ability to tell WHO sent what.  

From that log I guess your password is wrong.

/b

On Feb 13, 2010, at 3:36 PM, Mark Campbell-Smith wrote:

> Thanks Brian.
> 
> The full log is pasted here http://pastebin.freeswitch.org/12133
> 
> 
> 
> On Sun, Feb 14, 2010 at 2:08 AM, Brian West <brian at freeswitch.org> wrote:
>> Can't tell since you keep cutting the lines required to figure this out... FULL console log with FULL sip trace.  You always cut out what I wanna see.  Just get a pcap.
>> 
>> /b
>> 
>> On Feb 13, 2010, at 5:43 AM, Mark Campbell-Smith wrote:
>> 
>>> More testing. The device registers successfully to my SIP provider
>>> directly using UDP - why would FS be rejecting the registration
>>> request?
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From brian at freeswitch.org  Sat Feb 13 15:12:58 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 13 Feb 2010 17:12:58 -0600
Subject: [Freeswitch-users] signaling information SIP INFO
In-Reply-To: <650320.69699.qm@web33503.mail.mud.yahoo.com>
References: <650320.69699.qm@web33503.mail.mud.yahoo.com>
Message-ID: <B2999AB0-F1BF-462F-8891-BB3E9A641B4C@freeswitch.org>

Read "case SWITCH_EVENT_SEND_INFO:" in mod_sofia.c you'll see how.

/b

On Feb 13, 2010, at 4:42 PM, Diego Toro wrote:

> Hi,
> 
> I need send SIP INFO message where the body of the SIP message consists of signaling information to a gateway.
> 
> Example:
> 
> INFO sip:7007471000 at example.com SIP/2.0 
> Via: SIP/2.0/UDP alice.uk.example.com:5060 
> From: <sip:7007471234 at alice.uk.example.com>;tag=d3f423d 
> To: <sip:7007471000 at example.com>;tag=8942 
> Call-ID: 312352 at myphone 
> CSeq: 5 INFO 
> Content-Length: 24 
> Content-Type: application/dtmf-relay 
> 
> Signal=16
> Duration=160 
> 
> 
> How I can do that with FreeSwitch events ?
> 
> 
> Thank you
> 
> Diego Toro
> http://lacarretade.blogspot.com/
> 
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From brian at freeswitch.org  Sat Feb 13 15:14:07 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 13 Feb 2010 17:14:07 -0600
Subject: [Freeswitch-users] bypass_media bug?
In-Reply-To: <201002131552.12289.sos@sokhapkin.dyndns.org>
References: <201002121422.18544.sos@sokhapkin.dyndns.org>
	<201002131552.12289.sos@sokhapkin.dyndns.org>
Message-ID: <5DBF5E29-BAD3-4B8A-B94E-A0986C39551D@freeswitch.org>

Don't you have a jira on this already?  If not open it.. and put the details in... but I think you already have one.

/b

On Feb 13, 2010, at 2:52 PM, Sergey Okhapkin wrote:

> More precisely, FS returns to A leg caller SDP from first 183, but not from 
> final 200 OK.




From brian at freeswitch.org  Sat Feb 13 15:20:34 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 13 Feb 2010 17:20:34 -0600
Subject: [Freeswitch-users] Bridging sessions from two separate lua
	scripts; uuid_bridge? intercept?
In-Reply-To: <eb7797d71002131303q19b657dv91b7ae08ab5b34cf@mail.gmail.com>
References: <eb7797d71002071831w27301c5dva767b82b38c2ce6f@mail.gmail.com>
	<C23AA196-E851-4D5B-B1B5-2C2F32F6BB68@freeswitch.org>
	<eb7797d71002071929pb53e1adue2e91e6bdb598251@mail.gmail.com>
	<eb7797d71002080737k3f8fa5c9m8bf3ef2fcf42881e@mail.gmail.com>
	<D753EDBD-6ED7-43C2-B5F5-A899E3C3D3E2@jerris.com>
	<eb7797d71002121904v5bdfac87w2684c461ca1770cc@mail.gmail.com>
	<191c3a031002130714x2d876657p9c90d768d28a4496@mail.gmail.com>
	<eb7797d71002131018r56e86adnc054a3eaa2c05a96@mail.gmail.com>
	<eb7797d71002131132x2da4d2a0g54b8b8faf19520be@mail.gmail.com>
	<191c3a031002131239v8d0a8e2xd42725a4682b54b2@mail.gmail.com>
	<eb7797d71002131303q19b657dv91b7ae08ab5b34cf@mail.gmail.com>
Message-ID: <AEB40341-C9EB-4DA4-B139-916BC59472AF@freeswitch.org>

You need to stop trying to do everything manually.  You know FreeSWITCH does a lot of things for you so you can just work with the call but you seem to be trying WAY too hard to accomplish your task.  Let FreeSWITCH do the hard work for you... 

api:executeString("intercept -bleg " .. tostring(b_leg_uuid))

This line is 100% WRONG.

Intercept is an application NOT an API call.

Just do session:execute("intercept", "");, Those uuid_media calls are pointless just set the bypass_media_after_bridge variable if you want to go bypass after the bridge.  

I would give you my extended you're trying too hard speech but I think you get the picture.

/b

On Feb 13, 2010, at 3:03 PM, Adam Wilt wrote:

> Does "sigh" mean that it's a problem with FreeSWITCH, or perhaps that I did something stupid?




From lawwton at gmail.com  Sat Feb 13 15:24:13 2010
From: lawwton at gmail.com (Alfredo Quiroga-Villamil)
Date: Sat, 13 Feb 2010 18:24:13 -0500
Subject: [Freeswitch-users] Mod_Conference capacity....
In-Reply-To: <191c3a031002131448x59e42e22q6f931dd54834c0a7@mail.gmail.com>
References: <191c3a031002120856q93e52cep14ee96f1e7ed6d1@mail.gmail.com>
	<13ca621c1002121436n9b44dcexffeaafe998d4c88a@mail.gmail.com>
	<65d96fc81002130327q66d8227ao29ae46f8c7ee556a@mail.gmail.com>
	<191c3a031002130722v5c282c00s69f0b51b91e42520@mail.gmail.com>
	<13ca621c1002131120u482f6420xea1f7cad534cbe1a@mail.gmail.com>
	<E2A03ABF-69BF-4F85-9BB9-7CEAA7738E64@freeswitch.org>
	<13ca621c1002131326p38eb73bcs444f57b26fdfe381@mail.gmail.com>
	<0ad201caacf8$f8c35c20$ea4a1460$@com>
	<13ca621c1002131438t78864be4y2ef6e0f19b14f991@mail.gmail.com>
	<191c3a031002131448x59e42e22q6f931dd54834c0a7@mail.gmail.com>
Message-ID: <5fe6fa8f1002131524k61d18d9ej906b1c52c551065a@mail.gmail.com>

One thing that seems to help a lot is to have a "Success Stories"
page. A lot of the previous emails emphasize the point of benchmarking
FS and hitting it hard with a suite of tools available to do just
that. That all makes sense. The problem is that for a lot of
newcomers, benchmarking doesn't really make a lot of sense right away.
In more details ... the process would be something like this:

a) I need to use a system that's able to do certain things for a
conference for instance (happens to be my case). It needs to be
flexible and do a bunch of things.

b) FS seems like the best candidate out there "today".

c) Get a whatever server and install FS on it. Nice the thing runs, it
seems flexible, good performance, I can't believe my own eyes. Good
product.

d) Let me build an application around it now. Here is an ESL, WEB API,
cool. The sky is the limit.

e) Application is now built.

f) Let's benchmark ... Ok, let me now buy what I think the best server
will be and it'll cost $2000.00.

g) FS is amazing, I am getting these many concurrent calls, active
sessions, things are great for this X system that I bought. I will
scale up and get 5 more of the same type of server.

Now the question is, was that the right choice to make? Having access
to a place where others post their results help you make a better
determination on what kind of platforms to run your system on. Someone
with whom I've had a nice exchange of words in the past threw
something out there that was really helpful ... "Modern machine with
dual core, 64 bit for best results"; if my memory serves me right.
This is the kind of information that would be needed to then go out
and buy the server/s and then do the actual benchmarking and correctly
engineer/architect the platform. If I see that some of the success
stories are using Servers X with Foo CPU/s and bar MEM, then that
would be for me a good indication of the direction or areas to
investigate, go into.

If I then want top of the line tuning, excellent benchmarking and many
more things, yeah having the commercial support and having to pay for
it it's def. not a problem. After all a lot of work and hours go into
developing the product and we are getting it free. That part is
perfectly understood and supported as well as the development of some
more specific features.

Alfredo

On Sat, Feb 13, 2010 at 5:48 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> Helping with docs, now there is a topic we all groove on.
>
> On Feb 13, 2010 4:43 PM, "Gavin Henry" <gavin.henry at gmail.com> wrote:
>
> On 13 February 2010 22:07, Peder <peder at networkoblivion.com> wrote:
>> Instead of complaining about t...
>
> Did you read any of this thread at all? The OP asked about techniques
> to benchmark his kit, and any rough
> figures. The projects answer was to hire them for some commercial
> support. So anything I post on the wiki is not
> official. My suggestion is merely to put the official project stance
> on any benchmark questions or post to the
> wiki some tips on how to run your own tests with a link to getting
> them verified by the team via the commercial
> support.
>
> In no way am I complaining, just trying to get this cleared up for the
> next time someone asks. That's the whole
> point of docs and why I enjoy writing them. Write them once and refer
> people to them, fixing doc bugs along the
> way.
>
> Gavin.
>
>
>
> --
> http://www.suretecsystems.com/services/openldap/
> http://www.suretectelecom.com
>
> _____________...
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From ederwander at gmail.com  Sat Feb 13 15:37:36 2010
From: ederwander at gmail.com (Eder Souza)
Date: Sat, 13 Feb 2010 21:37:36 -0200
Subject: [Freeswitch-users] Send DTMF after call bridge
Message-ID: <622bedea1002131537l314bb5a4g9ed51bafd323c728@mail.gmail.com>

Hi

How i can send DTMF digits after call bridge ??

Example::

<action application="bridge" data="*openzap*/2/1/1831"/>


After my ramal 1831 answer i want send DTMF's how i can make this??


Att,


Eng Eder de Souza
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100213/2bfec53d/attachment-0002.html 

From gavin.henry at gmail.com  Sat Feb 13 15:45:45 2010
From: gavin.henry at gmail.com (Gavin Henry)
Date: Sat, 13 Feb 2010 23:45:45 +0000
Subject: [Freeswitch-users] Mod_Conference capacity....
In-Reply-To: <191c3a031002131448x59e42e22q6f931dd54834c0a7@mail.gmail.com>
References: <191c3a031002120856q93e52cep14ee96f1e7ed6d1@mail.gmail.com>
	<13ca621c1002121436n9b44dcexffeaafe998d4c88a@mail.gmail.com>
	<65d96fc81002130327q66d8227ao29ae46f8c7ee556a@mail.gmail.com>
	<191c3a031002130722v5c282c00s69f0b51b91e42520@mail.gmail.com>
	<13ca621c1002131120u482f6420xea1f7cad534cbe1a@mail.gmail.com>
	<E2A03ABF-69BF-4F85-9BB9-7CEAA7738E64@freeswitch.org>
	<13ca621c1002131326p38eb73bcs444f57b26fdfe381@mail.gmail.com>
	<0ad201caacf8$f8c35c20$ea4a1460$@com>
	<13ca621c1002131438t78864be4y2ef6e0f19b14f991@mail.gmail.com>
	<191c3a031002131448x59e42e22q6f931dd54834c0a7@mail.gmail.com>
Message-ID: <13ca621c1002131545r486c3aeaobea7716af03e0a2a@mail.gmail.com>

On 13 February 2010 22:48, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> Helping with docs, now there is a topic we all groove on.

Agreed! When I find something that isn't there I'll add it or improve it!

Gavin.

-- 
http://www.suretecsystems.com/services/openldap/
http://www.suretectelecom.com



From jmesquita at freeswitch.org  Sat Feb 13 15:49:05 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Sat, 13 Feb 2010 21:49:05 -0200
Subject: [Freeswitch-users] Send DTMF after call bridge
In-Reply-To: <622bedea1002131537l314bb5a4g9ed51bafd323c728@mail.gmail.com>
References: <622bedea1002131537l314bb5a4g9ed51bafd323c728@mail.gmail.com>
Message-ID: <ebbc2d551002131549w43313062w9a63ea20bdf3383a@mail.gmail.com>

Eder, acho que ? isso que vc precisa.

http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_queue_dtmf

Abra?os,
Jo?o Mesquita


On Sat, Feb 13, 2010 at 9:37 PM, Eder Souza <ederwander at gmail.com> wrote:

> Hi
>
> How i can send DTMF digits after call bridge ??
>
> Example::
>
> <action application="bridge" data="*openzap*/2/1/1831"/>
>
>
> After my ramal 1831 answer i want send DTMF's how i can make this??
>
>
> Att,
>
>
> Eng Eder de Souza
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100213/2c9b6dfe/attachment-0002.html 

From ederwander at gmail.com  Sat Feb 13 16:05:57 2010
From: ederwander at gmail.com (Eder Souza)
Date: Sat, 13 Feb 2010 22:05:57 -0200
Subject: [Freeswitch-users] Send DTMF after call bridge
In-Reply-To: <ebbc2d551002131549w43313062w9a63ea20bdf3383a@mail.gmail.com>
References: <622bedea1002131537l314bb5a4g9ed51bafd323c728@mail.gmail.com>
	<ebbc2d551002131549w43313062w9a63ea20bdf3383a@mail.gmail.com>
Message-ID: <622bedea1002131605p7c328ee1v993cbc38fd6da9dc@mail.gmail.com>

*Jo?o Thank you very much *

*brazilian ?  LOl LOl*

*
*

*Muito Obrigado irei testar *

*
*

*Eng Eder de Souza*


2010/2/13 Jo?o Mesquita <jmesquita at freeswitch.org>

> Eder, acho que ? isso que vc precisa.
>
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_queue_dtmf
>
> Abra?os,
> Jo?o Mesquita
>
>
> On Sat, Feb 13, 2010 at 9:37 PM, Eder Souza <ederwander at gmail.com> wrote:
>
>> Hi
>>
>> How i can send DTMF digits after call bridge ??
>>
>> Example::
>>
>> <action application="bridge" data="*openzap*/2/1/1831"/>
>>
>>
>> After my ramal 1831 answer i want send DTMF's how i can make this??
>>
>>
>> Att,
>>
>>
>> Eng Eder de Souza
>>
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100213/3c51a611/attachment-0002.html 

From gmaruzz at celliax.org  Sat Feb 13 19:35:01 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Sun, 14 Feb 2010 04:35:01 +0100
Subject: [Freeswitch-users] mod_skypiax (skype endpoint) new timing
Message-ID: <7b197bef1002131935j3fdf80c4w4050f10bea105d49@mail.gmail.com>

Hello FreeSWITCHers,

I've just committed on svn16640 new timing for mod_skypiax, and I
would like if you guys give it a test in the various use cases.

ciao for now,

-giovanni

-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618



From jason at jasonjgw.net  Sat Feb 13 20:20:33 2010
From: jason at jasonjgw.net (Jason White)
Date: Sun, 14 Feb 2010 15:20:33 +1100
Subject: [Freeswitch-users] FreeSWITCH Debian packages - build issues
In-Reply-To: <20100213225320.GA4990@jdc.jasonjgw.net>
References: <20100213030812.GA19108@jdc.jasonjgw.net>
	<201002130957.11633.errotan@gmail.com>
	<20100213090907.GA29452@jdc.jasonjgw.net>
	<201002131109.35877.errotan@gmail.com>
	<20100213225320.GA4990@jdc.jasonjgw.net>
Message-ID: <20100214042033.GA19822@jdc.jasonjgw.net>

Just to close this thread for now, FreeSWITCH builds correctly if I remove the
memcache module from the Debian package files.

Maybe when memcache in FreeSWITCH is updated to libmemcache 0.37 (which is in
Debian unstable currently) the autoconf problem, which I'm not inclined to
track down myself at the moement as I don't use memcache, will go away.




From lakindia89 at gmail.com  Sun Feb 14 01:29:04 2010
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Sun, 14 Feb 2010 14:59:04 +0530
Subject: [Freeswitch-users] Is it possible to repeat music in playback
In-Reply-To: <7d79b3931002120323x7a2520fcr1610af69ba4ca51@mail.gmail.com>
References: <13529f9d1002110233g57c8a1a5j3cdd38e22fa6a13d@mail.gmail.com>
	<2FBDB302-2B54-40CC-B694-332854219774@freeswitch.org>
	<191c3a031002110639h3cf14efdvf7715b973fe3caed@mail.gmail.com>
	<7d79b3931002120323x7a2520fcr1610af69ba4ca51@mail.gmail.com>
Message-ID: <7d79b3931002140129k6c9655c8o9a6956966bb22b70@mail.gmail.com>

Hi all,
Any update on this. How to stop an endless_playback???

On Fri, Feb 12, 2010 at 4:53 PM, lakshmanan ganapathy
<lakindia89 at gmail.com>wrote:

> Hi antony,
> Is there any way to stop the endless_playback??
> I tried with break. But it didn't worked!!
>
>
>
> On Thu, Feb 11, 2010 at 8:09 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> or try endless_playback app
>>
>>
>>
>> On Thu, Feb 11, 2010 at 7:26 AM, Brian West <brian at freeswitch.org> wrote:
>>
>>> Why not just use Fifo to hold them?  Or Park the agent and send the
>>> session a message to play music?  You then have options to define loop
>>> count.
>>>
>>> http://wiki.freeswitch.org/wiki/Event_Socket#execute
>>>
>>> /b
>>>
>>> On Feb 11, 2010, at 4:33 AM, Jingwei Yang wrote:
>>>
>>> > Hello,
>>> >
>>> > I've defined a very simple dialplan like the one below and when the
>>> caller is connected to this plan, I hope to keep the call alive and repeat
>>> the music set by playback. How am I able to achieve this?
>>> >
>>> >
>>> >     <extension name="agent_hold">
>>> >       <condition field="destination_number" expression="^33333$">
>>> >         <action application="answer"/>
>>> >         <action application="playback" data="customized/test.wav"/>
>>> >       </condition>
>>> >     </extension>
>>> >
>>> >
>>> > Thanks,
>>> > -Jingwei
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100214/04662e4a/attachment-0002.html 

From vmaruani at interwise.com  Sun Feb 14 01:33:29 2010
From: vmaruani at interwise.com (Victor Maruani)
Date: Sun, 14 Feb 2010 11:33:29 +0200
Subject: [Freeswitch-users] Bypass-media and REFER method
In-Reply-To: <191c3a031002111501y7e5917b8g4faae4b8752e6fc2@mail.gmail.com>
References: <B816EF85C732094BA2016B2112F3DC2B0402C777@isr-brass.interwise.com><B816EF85C732094BA2016B2112F3DC2B040C5CA6@isr-brass.interwise.com><191c3a031002100645n5d827949xdefd24143f10759@mail.gmail.com><B816EF85C732094BA2016B2112F3DC2B040C6061@isr-brass.interwise.com>
	<191c3a031002111501y7e5917b8g4faae4b8752e6fc2@mail.gmail.com>
Message-ID: <B816EF85C732094BA2016B2112F3DC2B040C693B@isr-brass.interwise.com>

Hi,

 

I would say it fails in 2 points:

First in the fact that a "NOTIFY 200 OK" (line 1070) is sent right after
FS gets the REFER.

 

Then in the REINVITE (line 1094) sent to A (10.10.5.19)  just after this
NOTIFY,

This REINVITE contains the SDP of the FS (10.10.5.92) causing the A side
to send media to FS. There will be no REINVITE with SDP of C
(10.10.5.48)

 

But as you say, just afterwards, the REFER action is actually done and C
is invited by FS with the SDP of A.

 

Conclusion :

1)

B is notified of success just after it sent the REFER and is
disconnected.

B may be notified of every step of the connection to C (100 trying...
200 OK) when these actually happen.

What if C is down? Can't FS notify a failure? (didn't test that.)

 

2)

'A' gets to send media to FS Because of a REINVITE which disconnect him
from B (we are in bypass media mode) . 

during the process of REFER, A should be still connected to B from a
media perspective.

The REINVITE is not done at the right time with the right params.

Here, a pseudo bridge (on way voice)  is established when C gets the
INVITE and is sending media to A.

 

A can hear C but C can't hear A after the REFER.

If C was down, A would be "lost" in FS...

 

I believe the correct behavior would be:

B sends REFER.

FS INVITE C 

C replies 100, 180... 200 and FS notifies B in accordance.

Once C has sent 200 OK with its SDP. B is disconnected and A is updated
(REINVITE) with C's SDP.

 

Please share your thoughts, I still don't know if it's a bug or if I
configured something wrong although I don't think so.

Hasn't anyone done that before?

 

Thank you.

 

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Anthony Minessale
Sent: Friday, February 12, 2010 1:02 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Bypass-media and REFER method

 

Where do you think it's failing?
that log shows it get refer, go back to dp, invite to 3341 and then the
other end hangs up at some point.



On Wed, Feb 10, 2010 at 11:01 AM, Victor Maruani
<vmaruani at interwise.com> wrote:

Hi,

 

Logs are on pb 12099

I hope this helps.

Reproduced with revision 16599.

 

A-side (10.10.5.19) is an x-lite registered with extension 1002

B (.5.51) refers to C (.5.48) none are registered.

 

Please refer to previous emails for details of dialplan and what I try
to do...

Let me know if you need more info 

 

Thanks!

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Anthony Minessale
Sent: Wednesday, February 10, 2010 4:46 PM


To: freeswitch-users at lists.freeswitch.org

Subject: Re: [Freeswitch-users] Bypass-media and REFER method

 

update to latest trunk and reproduce your problem with full debug
enabled.

sofia profile internal siptrace on
console loglevel debug

On Wed, Feb 10, 2010 at 4:44 AM, Victor Maruani <vmaruani at interwise.com>
wrote:

Hi,

 

I can't have a blind transfer work properly if I use bypass-media=true.

 

My first message may have been unclear, here I added excerpt from the
dialplan:

 

                <!--- ext MyIVR -->

                 <extension name="Extension MyIVR" continue="false">

                   <condition field="destination_number"
expression="^337$">

                                   <action application="set"
data="bypass_media=true"/>

                                   <action application="bridge"
data="sofia/internal/337 at 10.10.5.51"/>

                   </condition>

                 </extension>

 

 

                <!--- ext MyServ -->

                 <extension name="Extension MyServ " continue="false">

                   <condition field="destination_number"
expression="^3341$">

                                                <action
application="set" data="bypass_media=true"/>

                                   <action application="bridge"
data="sofia/internal/3341 at 10.10.5.48"/>

                   </condition>

                 </extension>   

 

The connection to MyIVR works. Then it sends Refer-to (3341 at ...) which
fails as I described it in the previous mail.

 

I would like to know if the feature has been validated and if I'm
missing something in the configuration.

 

Any help would be very appreciated.

 

Thanks!

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Victor Maruani
Sent: Sunday, February 07, 2010 5:01 PM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Bypass-media and REFER method

 

Hi,

 

I'm trying to do a POC using FS, the goal is to have FS handle REFERs
containing proprietary data. 

I want to have some logic on top of FS and also use the fail over
mechanism.

in short, I have something like this:

(third party) A side  --- FS ---- B side (IVR server)

 

the IVR the sends a REFER to FS. I don't want A to deal with it. 

now say B refers to C, it would be considered as a "group" C1, C2 ... to
which I want FS to failover.

only when one has answered should A be updated (REINVITE) and B notified
and disconnected.

if all fails I would expect B to be notified of the failure and proceed
as I wish without "losing" A.

 

from what I've read FS should be OK for the job but I have a couple
issues:

 

1 ) I have some issues getting FS handle a REFER while in bypass-media
mode.

(I tried with the release and some revisions including latest)

first when I bridge A and B everything is fine and media is bypassed.

When B sends REFER to C:

-           FS immediately NOTIFY B of success and send a reinvite to A
with SDP containing its own media IP/port.

-          then it does INVITE C with A's SDP.

-          B gets disconnected. A is not updated with C's sdp.

so at this point A sends RTP to FS and C sends RTP to A. ...

 

I basically have one extension for B: (set bypass-media and bridge to B)

and another extension to C which does the same actions.

what do you think I do wrong? 

 

 

2 ) how can I catch the REFER and set variables from it? (like ref-by or
ref-to)

in the dial plan I do catch the INVITE sent to C, but how to do it with
the REFER itself?

 

 

thanks for your help!

 

 

Best Regards,

Victor.

 


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
<mailto:MSN%3Aanthony_minessale at hotmail.com> 
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
<mailto:PAYPAL%3Aanthony.minessale at gmail.com> 
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:+19193869900


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
<mailto:MSN%3Aanthony_minessale at hotmail.com> 
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
<mailto:PAYPAL%3Aanthony.minessale at gmail.com> 
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:+19193869900

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100214/35b3de6d/attachment-0002.html 

From vetali100 at gmail.com  Sun Feb 14 02:06:34 2010
From: vetali100 at gmail.com (Vitalii Colosov)
Date: Sun, 14 Feb 2010 12:06:34 +0200
Subject: [Freeswitch-users] Other party does not hear voice when
	connecting with X-Lite
In-Reply-To: <D160C09B-1623-4A0A-BB14-1A1CC04A9276@freeswitch.org>
References: <a858e7691002120133q5e4c806fx309517e4fdec45c5@mail.gmail.com>
	<2F81A680-6A5C-4E3F-9590-FFF6AC4EA80D@freeswitch.org>
	<a858e7691002130247s439177acg23d6a41de3a9d21@mail.gmail.com>
	<a858e7691002130346y2018746o2a71a084a3fd12ee@mail.gmail.com>
	<D160C09B-1623-4A0A-BB14-1A1CC04A9276@freeswitch.org>
Message-ID: <a858e7691002140206q2292e055t3511f98ffa6a22fb@mail.gmail.com>

No, it is done on the different PCs...

Sorry, when I started the topic, I have described the problem how it is
visible from PC of my friend.
Then I tried to reproduce the same on my own PC, and you are right...I was
not able to hear anything as well, not only both party wasn't.
Also, from my PC I was NOT able to hear guitar on test number "9999".

This log reflects the call from my PC.
SIP header sent by XLITE was :

       INVITE sip:9999 at sip.voipsler.com <sip%3A9999 at sip.voipsler.com>SIP/2.0

       Via: SIP/2.0/UDP 192.168.2.10:8342<<<LOCAL IP>>>
;branch=z9hG4bK-d8754z-2f03fe469803ff0f-1---d8754z-;rport

       Contact: <sip:1000 at 192.168.2.10:8342<<<LOCAL IP>>>>




*Yesterday I put STUN server* at the XLITE settings and started to hear
guitar on "9999", and SIP header has changed.

       INVITE sip:1001 at sip.voipsler.com <sip%3A1001 at sip.voipsler.com>SIP/2.0

       Via: SIP/2.0/UDP 192.168.2.10:8216<<<LOCAL
IP>>>;branch=z9hG4bK-d8754z-1758da08c07c1e37-1---d8754z-;rport


       Contact: <sip:1000@<<<*my-global-ip-adress*>>>:8216>

BUT I am still NOT able to hear anything on my PC...

All ports are open on my PC and on FS server.


I tried to use the following option, but no luck: <param
name="aggressive-nat-detection" value="true"/>

http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#NATing_.5Bapply-nat-acl.2C_aggressive-nat-detection.5D


                           "This will enable NAT mode if the network IP/port
from which the request was received differs from the IP/Port combination in
the SIP Via: header, or if the Via: header contains the received parameter
(regardless of what it contains.) "


Do you know what else can I try?
Thank you,
Vitalii



2010/2/13 Brian West <brian at freeswitch.org>

> That should work either way then...are you trying to do this all on the
> same machine?
>
> /b
>
> On Feb 13, 2010, at 5:46 AM, Vitalii Colosov wrote:
>
> > It looks like FS is trying to send MEDIA part to my local ip adress
> instead of global...And Yate sends correct, to global IP...
> >
> > Any ideas?
> >
> > Thank
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100214/e95cf2ff/attachment-0002.html 

From mbsip at gazeta.pl  Sun Feb 14 06:52:03 2010
From: mbsip at gazeta.pl (mbsip)
Date: Sun, 14 Feb 2010 15:52:03 +0100
Subject: [Freeswitch-users] mod_python fetching data from mysql for VM
In-Reply-To: <28f27f5d1002131438l58718bbat304407f49397c965@mail.gmail.com>
References: <28f27f5d1002131203j132c6230sa260358611a16247@mail.gmail.com>
	<8E4B95E9-3914-4FEE-9C8C-9A5B4C5E2AB2@jerris.com>
	<28f27f5d1002131354k23a0eaf5jc51b05106c68d719@mail.gmail.com>
	<28f27f5d1002131359h164e4077x326f646ff2367ad9@mail.gmail.com>
	<ebbc2d551002131420v58cf95f2lfc9356f2377728b6@mail.gmail.com>
	<28f27f5d1002131438l58718bbat304407f49397c965@mail.gmail.com>
Message-ID: <28f27f5d1002140652q656ab6deh40109b223d4f2f4a@mail.gmail.com>

Hi.

Please correct me if my approach is okay.
1. in python.conf.xml
    <param name="xml-handler-script" value="testscript"/>
    <param name="xml-handler-bindings" value="directory"/>
2. in dialplan
    <extension name="public_did_11digits">
      <condition field="destination_number" expression="^(\d{11})$">
        <action application="answer"/>
        <action application="voicemail" data="default $${domain} $1"/>
      </condition>
    </extension>
3.testscript.py (as for now only static entries)
def xml_fetch(params):

        xml = '''
<?xml version="1.0" encoding="UTF-8" standalone="no"?>
<document type="freeswitch/xml">
  <section name="directory">
    <domain name="mydomainHERE">
      <params>
        <param name="dial-string"
value="{presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
      </params>
      <groups>
        <group name="default">
         <users>
          <user id="12345678901">
            <params>
               <param name="vm-mailto" value="email at aaa.com"/
            </params>
          </user>
        </group>
       </users>
      </groups>
    </domain>
  </section>
</document>
'''

        return xml


Unfortunately aforemetnioned configuration does not work at all and
produce following errors:
2010-02-14 17:31:16.878878 [DEBUG] sofia.c:4110 Channel
sofia/internal/100 at 10.10.10.10 entering state [completed][200]
2010-02-14 17:31:16.878878 [DEBUG] switch_core_session.c:638 Send
signal sofia/internal/100 at 10.10.10.10 [BREAK]
2010-02-14 17:31:16.878878 [NOTICE] mod_dptools.c:715 Channel
[sofia/internal/100 at 10.10.10.10] has been answered
EXECUTE sofia/internal/100 at 10.10.10.10 voicemail(default mydomainHERE
12345678901)
2010-02-14 17:31:16.888821 [DEBUG] mod_voicemail.c:728 [default] rwlock
2010-02-14 17:31:16.888821 [NOTICE] mod_python.c:118 Invoking py module: obadamy
2010-02-14 17:31:16.888821 [DEBUG] mod_python.c:188 Call python script
2010-02-14 17:31:16.888821 [DEBUG] mod_python.c:191 Finished calling
python script
2010-02-14 17:31:16.888821 [ERR] mod_python.c:200 Error calling python script
2010-02-14 17:31:16.888821 [WARNING] mod_voicemail.c:2923 Can't find
user [12345678901 at mydomainHERE]
2010-02-14 17:31:16.888821 [DEBUG] switch_ivr_play_say.c:118 No
language specified - Using [en]
2010-02-14 17:31:16.888821 [DEBUG] sofia.c:4110 Channel
sofia/internal/100 at 10.10.10.10 entering state [ready][200]
2010-02-14 17:31:16.888821 [DEBUG] switch_ivr_play_say.c:273 Handle
play-file:[voicemail/vm-goodbye.wav] (en:en)
2010-02-14 17:31:16.888821 [DEBUG] switch_ivr_play_say.c:1158 Codec
Activated L16 at 8000hz 1 channels 20ms

The same output with simplified python script.
def xml_fetch(params):

        xml = '''
<?xml version="1.0" encoding="UTF-8" standalone="no"?>
<document type="freeswitch/xml">
  <section name="directory">
    <domain name="62.29.162.90">
      <user id="48223891699">
        <params>
        </params>
      </user>
    </domain>
  </section>
</document>
'''

        return xml



As for now I have no idea how to solve this, but still digging.
Funny is that dialplan bindings work okay.

Any help pls.
Thx,
Maciej


2010/2/13 mbsip <mbsip at gazeta.pl>:
> Jo?o,
>
> Thanks for hint, because i don't know how the db fetched data could be
> used with voicemail.
> I am about to ready it carefully :P
>
> Thanks,
> Maciej
>
> 2010/2/13 Jo?o Mesquita <jmesquita at freeswitch.org>:
>> Maciej,
>>
>> Take a look at the xml_hooks we have on mod_python. Might do the trick for
>> you.
>>
>> http://svn.freeswitch.org/svn/freeswitch/trunk/src/mod/languages/mod_python/python_example.py
>>
>> JM
>>
>>
>> On Sat, Feb 13, 2010 at 6:59 PM, mbsip <mbsip at gazeta.pl> wrote:
>>>
>>> There is a lack of connection between fatched data and voicemail and I
>>> dont know how to achieve it.
>>>
>>> Thx,
>>> Maciej.
>>>
>>>
>>> 2010/2/13 mbsip <mbsip at gazeta.pl>:
>>> > Thx for prompt reply.
>>> >
>>> > The main task is to be able to use Mysql db in conjunction with VM
>>> > (but not only voicemail_msgs, voicemail_prefs).
>>> >
>>> > Lets imagine sb is calling 1000 and wants to record the message.
>>> > According to mod_voicemail settings message should be sent to some
>>> > email address.
>>> > But the information about user 1000 and his settings like email
>>> > address, passwd, quota should be fetched from Mysql db, not from
>>> > directory/default/1000.xml.
>>> > That's why I am using <action application="python" data="test"/> in my
>>> > dialplan to work with python script which in turn should do the magic.
>>> > The script should be able to gather all necessery data about user 1000
>>> > (like email address in shown example) and use them in VM.
>>> >
>>> > So the problem is how to modify the script to force voicemail app to
>>> > use data from DB.
>>> > Currently ?session.execute("voicemail", "default ${domain} " +
>>> > the_dest) is still using .xml files.
>>> >
>>> > Thx,
>>> > Maciej.
>>> >
>>> >
>>> > 2010/2/13 Michael Jerris <mike at jerris.com>:
>>> >> Can you describe what your trying to accomplish, I don't understand
>>> >> what the goal is. ?What feature are you looking for that does not already
>>> >> exist in mod_voiceamil.
>>> >>
>>> >> Mike
>>> >>
>>> >> On Feb 13, 2010, at 3:03 PM, mbsip wrote:
>>> >>
>>> >>> Hello,
>>> >>>
>>> >>> I am trying to use mod_python to fetch data from Mysql db (through
>>> >>> ODBC) and execute voicemail application.
>>> >>> Below a part of my script:
>>> >>>
>>> >>> db=MySQLdb.connect("localhost","root","","test")
>>> >>> ? ? ? Cursor=db.cursor()
>>> >>> ? ? ? sql = "SELECT email FROM VM WHERE called_num=%s" % the_dest
>>> >>> ? ? ? Cursor.execute(sql)
>>> >>> ? ? ? while (1):
>>> >>> ? ? ? ? ? ? ? Results = Cursor.fetchone()
>>> >>> ? ? ? ? ? ? ? if Results == None:
>>> >>> ? ? ? ? ? ? ? ? ? ? ? break
>>> >>> ? ? ? ? ? ? ? consoleLog("debug", "Found email " + Results[0] +"\n")
>>> >>> ? ? ? ? ? ? ? the_recipient = Results[0]
>>> >>> ? ? ? db.close()
>>> >>>
>>> >>> Now i have email address corresponding with called number. The
>>> >>> question is how to use it for voicemail application?
>>> >>> So it also means how to omit all /directory/default/....xml, where
>>> >>> there are all VM parameters set and use fetched data.
>>> >>>
>>> >>> ? ? ? session.answer()
>>> >>> ? ? ? session.execute("voicemail", "default ${domain} " + the_dest)
>>> >>>
>>> >>> Is this possible or should I start all VM app in python from the
>>> >>> scratch?
>>> >>
>>> >>
>>> >> _______________________________________________
>>> >> FreeSWITCH-users mailing list
>>> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>
>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> http://www.freeswitch.org
>>> >>
>>> >
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>



From errotan at gmail.com  Sun Feb 14 06:52:31 2010
From: errotan at gmail.com (=?iso-8859-1?q?Pusk=E1s_Zsolt?=)
Date: Sun, 14 Feb 2010 15:52:31 +0100
Subject: [Freeswitch-users] mod_skypiax (skype endpoint) new timing
In-Reply-To: <7b197bef1002131935j3fdf80c4w4050f10bea105d49@mail.gmail.com>
References: <7b197bef1002131935j3fdf80c4w4050f10bea105d49@mail.gmail.com>
Message-ID: <201002141552.31159.errotan@gmail.com>

2010. febru?r 14. 04.35.01 Giovanni Maruzzelli d?tummal ezt ?rta:
> Hello FreeSWITCHers,
> 
> I've just committed on svn16640 new timing for mod_skypiax, and I
> would like if you guys give it a test in the various use cases.
> 
> ciao for now,
> 
> -giovanni
> 

It is not really working:

2010-02-14 15:37:39.182686 [ERR] skypiax_protocol.c:196 rev 16640M[(nil)|37     
][ERRORA  196  ][gun_at_koli][-1, 0, 0] Skype got ERROR: |||ERROR 589 ALTER 
CALL: unable to alter input/output|||
2010-02-14 15:37:39.182686 [ERR] skypiax_protocol.c:198 rev 16640M[(nil)|37     
][ERRORA  198  ][gun_at_koli][-1, 0,16] skype_call now is DOWN
2010-02-14 15:37:39.182686 [DEBUG] mod_skypiax.c:1068 rev 16640M[(nil)|37     
][DEBUG_SKYPE  1068 ][gun_at_koli][-1, 1,16] skype call ended                                                                                                 
2010-02-14 15:37:39.182686 [DEBUG] mod_skypiax.c:1085 rev 16640M[(nil)|37     
][DEBUG_SKYPE  1085 ][gun_at_koli][-1, 1,16] no session


Also can't unload mod_skypiax because the channels are up:

freeswitch at internal> reload mod_skypiax
-ERR unloading module [Module in use.]

2010-02-14 15:40:13.570678 [WARNING] switch_loadable_module.c:1268 Module 
mod_skypiax is in use, cannot unload.
freeswitch at internal> show channels
uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,write_codec,write_rate,secure,hostname,presence_id,presence_data
4f302086-1976-11df-a1d8-47a9ffb132d9,inbound,2010-02-14 
15:36:16,1266158176,skypiax/gun_at_koli,CS_EXECUTE,Pusk?s 
Zsolt,errotan,,4219,echo,,XML,default,L16,16000,L16,16000,,pc,,
80007580-1976-11df-a1d9-47a9ffb132d9,inbound,2010-02-14 
15:37:38,1266158258,skypiax/gun_at_koli,CS_EXECUTE,Pusk?s 
Zsolt,errotan,,4219,echo,,XML,default,L16,16000,L16,16000,,pc,,

2 total.

Have to do a fsctl shutdown to kill those channels.

My home skype (errotan) client displays: Remote sound problem. when I try to 
call the other (gun_at_koli).



From anthony.minessale at gmail.com  Sun Feb 14 06:54:27 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sun, 14 Feb 2010 08:54:27 -0600
Subject: [Freeswitch-users] Bypass-media and REFER method
In-Reply-To: <191c3a031002140645s358688c8g68280267d5d650d3@mail.gmail.com>
References: <B816EF85C732094BA2016B2112F3DC2B0402C777@isr-brass.interwise.com>
	<B816EF85C732094BA2016B2112F3DC2B040C5CA6@isr-brass.interwise.com>
	<191c3a031002100645n5d827949xdefd24143f10759@mail.gmail.com>
	<B816EF85C732094BA2016B2112F3DC2B040C6061@isr-brass.interwise.com>
	<191c3a031002111501y7e5917b8g4faae4b8752e6fc2@mail.gmail.com>
	<B816EF85C732094BA2016B2112F3DC2B040C693B@isr-brass.interwise.com>
	<191c3a031002140645s358688c8g68280267d5d650d3@mail.gmail.com>
Message-ID: <191c3a031002140654x37b65f1alc45907b545249cb4@mail.gmail.com>

We don't support the series of 100,180,200 in the notifies that is typically
a pure sip pbx feature.
We are a b2b and protocol agnostic softswitch.
When you do refer to a sofia leg you are talking to that leg independantly.
We do not track the progress of what the channel does once you transfer it,
instead we send the channel back to the dialplan and accept the refer.  This
operation if sucessful will generate 202.
The resume media on hold and bypass after att xfer sofia options or
bypass_media_after_bridge var may be what you need if you want no media on
fs.
We have to make some sacrifices on sip madness to gain some general
flexibility and call volume. =/

On Feb 14, 2010 3:39 AM, "Victor Maruani" <vmaruani at interwise.com> wrote:

 Hi,



I would say it fails in 2 points:

First in the fact that a "NOTIFY 200 OK" (line 1070) is sent right after FS
gets the REFER.



Then in the REINVITE (line 1094) sent to A (10.10.5.19)  just after this
NOTIFY,

This REINVITE contains the SDP of the FS (10.10.5.92) causing the A side to
send media to FS. There will be no REINVITE with SDP of C (10.10.5.48)



But as you say, just afterwards, the REFER action is actually done and C is
invited by FS with the SDP of A.



Conclusion :

1)

B is notified of success just after it sent the REFER and is disconnected.

B may be notified of every step of the connection to C (100 trying? 200 OK)
when these actually happen.

What if C is down? Can't FS notify a failure? (didn't test that.)



2)

'A' gets to send media to FS Because of a REINVITE which disconnect him from
B (we are in bypass media mode) .

during the process of REFER, A should be still connected to B from a media
perspective.

The REINVITE is not done at the right time with the right params.

Here, a pseudo bridge (on way voice)  is established when C gets the INVITE
and is sending media to A.



A can hear C but C can't hear A after the REFER.

If C was down, A would be "lost" in FS?



I believe the correct behavior would be:

B sends REFER.

FS INVITE C

C replies 100, 180? 200 and FS notifies B in accordance.

Once C has sent 200 OK with its SDP. B is disconnected and A is updated
(REINVITE) with C's SDP.



Please share your thoughts, I still don't know if it's a bug or if I
configured something wrong although I don't think so.

Hasn't anyone done that before?



Thank you.







*From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony
Minessale
*Sent:* Friday, February 12, 2010 1:02 AM


To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Bypass-media and REFER me...





Where do you think it's failing?
that log shows it get refer, go back to dp, invite to 3341 and...

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100214/086a6d4c/attachment-0002.html 

From anthony.minessale at gmail.com  Sun Feb 14 07:04:09 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sun, 14 Feb 2010 09:04:09 -0600
Subject: [Freeswitch-users] Other party does not hear voice when
	connecting with X-Lite
In-Reply-To: <a858e7691002140206q2292e055t3511f98ffa6a22fb@mail.gmail.com>
References: <a858e7691002120133q5e4c806fx309517e4fdec45c5@mail.gmail.com>
	<2F81A680-6A5C-4E3F-9590-FFF6AC4EA80D@freeswitch.org>
	<a858e7691002130247s439177acg23d6a41de3a9d21@mail.gmail.com>
	<a858e7691002130346y2018746o2a71a084a3fd12ee@mail.gmail.com>
	<D160C09B-1623-4A0A-BB14-1A1CC04A9276@freeswitch.org>
	<a858e7691002140206q2292e055t3511f98ffa6a22fb@mail.gmail.com>
Message-ID: <191c3a031002140704g705bfc73rd8dd103f3d846062@mail.gmail.com>

You need to describe this again its too confusing now.
List each device, freeswitch, the phones and which ip and combo of addrs it
uses with the topology clearly stated.
Your attempt to simplify your explanation is actually making it harder to
follow.
Also consider a debug/sip trace as well.

Include
sofia status profile default.

Then capture a test call after entering these commands.

console loglevel debug.
sofa profile internal siptrace on

On Feb 14, 2010 4:12 AM, "Vitalii Colosov" <vetali100 at gmail.com> wrote:

No, it is done on the different PCs...

Sorry, when I started the topic, I have described the problem how it is
visible from PC of my friend.
Then I tried to reproduce the same on my own PC, and you are right...I was
not able to hear anything as well, not only both party wasn't.
Also, from my PC I was NOT able to hear guitar on test number "9999".

This log reflects the call from my PC.
SIP header sent by XLITE was :

       INVITE sip:9999 at sip.voipsler.com <sip%3A9999 at sip.voipsler.com>SIP/2.0

       Via: SIP/2.0/UDP 192.168.2.10:8342<<<LOCAL IP>>>
;branch=z9hG4bK-d8754z-2f03fe469803ff0f-1---d8754z-;rport

       Contact: <sip:1000 at 192.168.2.10:8342<<<LOCAL IP>>>>




*Yesterday I put STUN server* at the XLITE settings and started to hear
guitar on "9999", and SIP header has changed.

       INVITE sip:1001 at sip.voipsler.com <sip%3A1001 at sip.voipsler.com>SIP/2.0

       Via: SIP/2.0/UDP 192.168.2.10:8216<<<LOCAL
IP>>>;branch=z9hG4bK-d8754z-1758da08c07c1e37-1---d8754z-;rport


       Contact: <sip:1000@<<<*my-global-ip-adress*>>>:8216>

BUT I am still NOT able to hear anything on my PC...

All ports are open on my PC and on FS server.


I tried to use the following option, but no luck: <param
name="aggressive-nat-detection" value="true"/>

http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#NATing_.5Bapply-nat-acl.2C_aggressive-nat-detection.5D


                           "This will enable NAT mode if the network IP/port
from which the request was received differs from the IP/Port combination in
the SIP Via: header, or if the Via: header contains the received parameter
(regardless of what it contains.) "


Do you know what else can I try?
Thank you,
Vitalii



2010/2/13 Brian West <brian at freeswitch.org>


>
> That should work either way then...are you trying to do this all on the
same machine?
>
> /b
>
...


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100214/f7c82c8e/attachment-0002.html 

From anthony.minessale at gmail.com  Sun Feb 14 07:05:55 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sun, 14 Feb 2010 09:05:55 -0600
Subject: [Freeswitch-users] Is it possible to repeat music in playback
In-Reply-To: <7d79b3931002140129k6c9655c8o9a6956966bb22b70@mail.gmail.com>
References: <13529f9d1002110233g57c8a1a5j3cdd38e22fa6a13d@mail.gmail.com>
	<2FBDB302-2B54-40CC-B694-332854219774@freeswitch.org>
	<191c3a031002110639h3cf14efdvf7715b973fe3caed@mail.gmail.com>
	<7d79b3931002120323x7a2520fcr1610af69ba4ca51@mail.gmail.com>
	<7d79b3931002140129k6c9655c8o9a6956966bb22b70@mail.gmail.com>
Message-ID: <191c3a031002140705m525a8753h67ef2609ff51fe78@mail.gmail.com>

There is no way, that's why its called endless playback.
You could put in a feature request or a bounty and if it was deemed a
sensible req, it could be added.

On Feb 14, 2010 3:37 AM, "lakshmanan ganapathy" <lakindia89 at gmail.com>
wrote:

Hi all,
Any update on this. How to stop an endless_playback???



On Fri, Feb 12, 2010 at 4:53 PM, lakshmanan ganapathy <lakindia89 at gmail.com>
wrote:
>
> Hi antony,...

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100214/00d9ccdc/attachment-0002.html 

From errotan at gmail.com  Sun Feb 14 07:09:18 2010
From: errotan at gmail.com (=?iso-8859-1?q?Pusk=E1s_Zsolt?=)
Date: Sun, 14 Feb 2010 16:09:18 +0100
Subject: [Freeswitch-users] mod_skypiax (skype endpoint) new timing
In-Reply-To: <201002141552.31159.errotan@gmail.com>
References: <7b197bef1002131935j3fdf80c4w4050f10bea105d49@mail.gmail.com>
	<201002141552.31159.errotan@gmail.com>
Message-ID: <201002141609.18711.errotan@gmail.com>

2010. febru?r 14. 15.52.31 Pusk?s Zsolt d?tummal ezt ?rta:
> 2010. febru?r 14. 04.35.01 Giovanni Maruzzelli d?tummal ezt ?rta:
> > Hello FreeSWITCHers,
> >
> > I've just committed on svn16640 new timing for mod_skypiax, and I
> > would like if you guys give it a test in the various use cases.
> >
> > ciao for now,
> >
> > -giovanni
> 
> It is not really working:
> 
> 2010-02-14 15:37:39.182686 [ERR] skypiax_protocol.c:196 rev 16640M[(nil)|37
> ][ERRORA  196  ][gun_at_koli][-1, 0, 0] Skype got ERROR: |||ERROR 589 ALTER
> CALL: unable to alter input/output|||
> 2010-02-14 15:37:39.182686 [ERR] skypiax_protocol.c:198 rev 16640M[(nil)|37
> ][ERRORA  198  ][gun_at_koli][-1, 0,16] skype_call now is DOWN
> 2010-02-14 15:37:39.182686 [DEBUG] mod_skypiax.c:1068 rev 16640M[(nil)|37
> ][DEBUG_SKYPE  1068 ][gun_at_koli][-1, 1,16] skype call ended
> 2010-02-14 15:37:39.182686 [DEBUG] mod_skypiax.c:1085 rev 16640M[(nil)|37
> ][DEBUG_SKYPE  1085 ][gun_at_koli][-1, 1,16] no session
> 
> 
> Also can't unload mod_skypiax because the channels are up:
> 
> freeswitch at internal> reload mod_skypiax
> -ERR unloading module [Module in use.]
> 
> 2010-02-14 15:40:13.570678 [WARNING] switch_loadable_module.c:1268 Module
> mod_skypiax is in use, cannot unload.
> freeswitch at internal> show channels
> uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,de
> st,application,application_data,dialplan,context,read_codec,read_rate,write
> _codec,write_rate,secure,hostname,presence_id,presence_data
>  4f302086-1976-11df-a1d8-47a9ffb132d9,inbound,2010-02-14
> 15:36:16,1266158176,skypiax/gun_at_koli,CS_EXECUTE,Pusk?s
> Zsolt,errotan,,4219,echo,,XML,default,L16,16000,L16,16000,,pc,,
> 80007580-1976-11df-a1d9-47a9ffb132d9,inbound,2010-02-14
> 15:37:38,1266158258,skypiax/gun_at_koli,CS_EXECUTE,Pusk?s
> Zsolt,errotan,,4219,echo,,XML,default,L16,16000,L16,16000,,pc,,
> 
> 2 total.
> 
> Have to do a fsctl shutdown to kill those channels.
> 
> My home skype (errotan) client displays: Remote sound problem. when I try
>  to call the other (gun_at_koli).
> 

Ooops sorry the user running the skype client were not in the audio group 
therefore don't have access to snd_dummy.

Works now! Tested on Debian "Lenny" x86

:)



From mbsip at gazeta.pl  Sun Feb 14 08:01:47 2010
From: mbsip at gazeta.pl (mbsip)
Date: Sun, 14 Feb 2010 17:01:47 +0100
Subject: [Freeswitch-users] can vm settings go in
	conf\directory\default\default.xml?
In-Reply-To: <28f27f5d1002131444h36df1ba7g7438dd685d8f4281@mail.gmail.com>
References: <4B77292B.6080207@apcl.us>
	<28f27f5d1002131444h36df1ba7g7438dd685d8f4281@mail.gmail.com>
Message-ID: <28f27f5d1002140801r1e952d91l85b4b643350134e2@mail.gmail.com>

Paul,

Maybe you should think about providing dynamic directory information
by using DB.
>From my point of view, Its much easier to manage DB than .xml files.

Thx,
Maciej.

2010/2/13 mbsip <mbsip at gazeta.pl>:
> So now I am with You Paul. I have the same thoughs and problem :P
>
> Thanks,
> Maciej.
>
> 2010/2/13 Paul Levin <paul at apcl.us>:
>> Thank you for the reply Maciej.
>>
>> Looks like I made an error in my first email, so let me repeat the problem.
>> If I put the lines:
>>
>> <param name="vm-email-all-messages" value="true"/>
>> <param name="vm-keep-local-after-email" value="false"/>
>> <param name="vm-attach-file" value="true"/>
>>
>> into?? conf/directory/default/Bob.xml?? then emails for voice mail are sent.
>>
>> If I remove those three lines from Bob.xml and put them into
>> conf/directory/default.xml?? then emails are not sent.
>>
>> I would have thought that putting those lines in
>> conf/directory/default.xml? would remove the requirement to have them in
>> Bob.xml.? No?
>>
>> ??? Thanks,
>> ??? Paul
>>
>>
>>
>> From:
>> mbsip <mbsip at gazeta.pl>
>> Date:
>> Sat, 13 Feb 2010 22:23:02 +0100
>>
>>> Have you tried doing the same with
>>> /usr/local/freeswitch/conf/directory/default.xml ?
>>
>>> Maciej
>>
>> 2010/2/13 Paul Levin <paul at apcl.us>:
>>
>>> I'm running FS on Windows (in case that matters here).
>>>
>>> In conf\directory\default\Bob.xml I have the settings:
>>>
>>> ??? ? <param name="vm-email-all-messages" value="true"/>
>>> ??? ? <param name="vm-keep-local-after-email" value="false"/>
>>> ??? ? <param name="vm-attach-file" value="true"/>
>>>
>>> in addition to other vm- setting that are specific to Bob.? When a voice
>>> mail is left for Bob, an email is sent to the configured email address.
>>> It
>>> is working well.? When the email is sent, I can see in the console the
>>> lines:
>>>
>>> 2010-02-10 15:41:36.949484 [DEBUG]
>>> switch_utils.c:633 Emailed file [C:\WINDOWS\TEMP\mail.12658416960810]
>>> to [bob at domain.com]
>>>
>>> 2010-02-10 15:41:36.949484 [DEBUG]
>>> mod_voicemail.c:2541 Sending message to bob at domain.com
>>>
>>> I then move those 3 <param> lines into the default\default.xml file.? Now
>>> when a voice mail is left for Bob, an email is not sent and those debug
>>> lines do not appear on the console.
>>>
>>> I don't mind keeping those 3 lines in each user file, but I'm expecting to
>>> have about 10,000 users and its kinda silly to repeat those lines in each
>>> user's file.? Can't they go in the default.xml file (and have it work)?
>>>
>>> ??? Thanks,
>>> ??? Paul
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>



From bottleman at icf.org.ru  Sun Feb 14 08:09:43 2010
From: bottleman at icf.org.ru (Georgiewskiy Yuriy)
Date: Sun, 14 Feb 2010 19:09:43 +0300 (MSK)
Subject: [Freeswitch-users] FreeSWITCH Debian packages - build issues
In-Reply-To: <20100213030812.GA19108@jdc.jasonjgw.net>
References: <20100213030812.GA19108@jdc.jasonjgw.net>
Message-ID: <alpine.DEB.2.00.1002141906400.26475@srv.icf.org.ru>

On 2010-02-13 14:08 +1100, Jason White wrote Freeswitch-users:

As i see build on debian stable is broken, it's not build many modules, 
for example mod_say_xx, play, etc, actually there no error on build process, 
but resulted packages not usable becouse many modules not present in it.

JW>Has anyone successfully built the Debian packages recently from the source
JW>repository?
JW>
JW>The problem I'm experiencing is that openzap is specified to be built, but it
JW>is never actually compiled. Consequently, the packages can't be created (the
JW>process fails due to the missing mod_openzap.so file).
JW>
JW>I don't need openzap; I can easily comment it out, but I also think the
JW>supplied package files should work as is.
JW>
JW>first step: confirm whether my experience under Debian Sid is shared by others
JW>using different versions of Debian or Ubuntu.
JW>
JW>
JW>_______________________________________________
JW>FreeSWITCH-users mailing list
JW>FreeSWITCH-users at lists.freeswitch.org
JW>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
JW>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
JW>http://www.freeswitch.org
JW>

C ?????????                       With Best Regards
???????????? ????.                Georgiewskiy Yuriy
+7 4872 711666                    +7 4872 711666
???? +7 4872 711143               fax +7 4872 711143
???????? ??? "?? ?? ??????"       IT Service Ltd
http://nkoort.ru                  http://nkoort.ru
JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
YG129-RIPE                        YG129-RIPE

From gmaruzz at celliax.org  Sun Feb 14 08:40:24 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Sun, 14 Feb 2010 17:40:24 +0100
Subject: [Freeswitch-users] mod_skypiax (skype endpoint) new timing
In-Reply-To: <201002141609.18711.errotan@gmail.com>
References: <7b197bef1002131935j3fdf80c4w4050f10bea105d49@mail.gmail.com> 
	<201002141552.31159.errotan@gmail.com>
	<201002141609.18711.errotan@gmail.com>
Message-ID: <7b197bef1002140840i6dfbdaecu26c1ae7e1f1a0633@mail.gmail.com>

Thanks Pusk?s,

any problem of growing delay in sip->fs->skype or skype->fs->sip, or
conferences, or whatever?

Also, snd-dummy is now more compatible with kernels at 1000HZ (eg:
centOS), although I believe Lenny is 100HZ.

-giovanni

On Sun, Feb 14, 2010 at 4:09 PM, Pusk?s Zsolt <errotan at gmail.com> wrote:
> 2010. febru?r 14. 15.52.31 Pusk?s Zsolt d?tummal ezt ?rta:
>> 2010. febru?r 14. 04.35.01 Giovanni Maruzzelli d?tummal ezt ?rta:
>> > Hello FreeSWITCHers,
>> >
>> > I've just committed on svn16640 new timing for mod_skypiax, and I
>> > would like if you guys give it a test in the various use cases.
>> >
>> > ciao for now,
>> >
>> > -giovanni
>>
>> It is not really working:
>>
>> 2010-02-14 15:37:39.182686 [ERR] skypiax_protocol.c:196 rev 16640M[(nil)|37
>> ][ERRORA ?196 ?][gun_at_koli][-1, 0, 0] Skype got ERROR: |||ERROR 589 ALTER
>> CALL: unable to alter input/output|||
>> 2010-02-14 15:37:39.182686 [ERR] skypiax_protocol.c:198 rev 16640M[(nil)|37
>> ][ERRORA ?198 ?][gun_at_koli][-1, 0,16] skype_call now is DOWN
>> 2010-02-14 15:37:39.182686 [DEBUG] mod_skypiax.c:1068 rev 16640M[(nil)|37
>> ][DEBUG_SKYPE ?1068 ][gun_at_koli][-1, 1,16] skype call ended
>> 2010-02-14 15:37:39.182686 [DEBUG] mod_skypiax.c:1085 rev 16640M[(nil)|37
>> ][DEBUG_SKYPE ?1085 ][gun_at_koli][-1, 1,16] no session
>>
>>
>> Also can't unload mod_skypiax because the channels are up:
>>
>> freeswitch at internal> reload mod_skypiax
>> -ERR unloading module [Module in use.]
>>
>> 2010-02-14 15:40:13.570678 [WARNING] switch_loadable_module.c:1268 Module
>> mod_skypiax is in use, cannot unload.
>> freeswitch at internal> show channels
>> uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,de
>> st,application,application_data,dialplan,context,read_codec,read_rate,write
>> _codec,write_rate,secure,hostname,presence_id,presence_data
>> ?4f302086-1976-11df-a1d8-47a9ffb132d9,inbound,2010-02-14
>> 15:36:16,1266158176,skypiax/gun_at_koli,CS_EXECUTE,Pusk?s
>> Zsolt,errotan,,4219,echo,,XML,default,L16,16000,L16,16000,,pc,,
>> 80007580-1976-11df-a1d9-47a9ffb132d9,inbound,2010-02-14
>> 15:37:38,1266158258,skypiax/gun_at_koli,CS_EXECUTE,Pusk?s
>> Zsolt,errotan,,4219,echo,,XML,default,L16,16000,L16,16000,,pc,,
>>
>> 2 total.
>>
>> Have to do a fsctl shutdown to kill those channels.
>>
>> My home skype (errotan) client displays: Remote sound problem. when I try
>> ?to call the other (gun_at_koli).
>>
>
> Ooops sorry the user running the skype client were not in the audio group
> therefore don't have access to snd_dummy.
>
> Works now! Tested on Debian "Lenny" x86
>
> :)
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618



From gmaruzz at celliax.org  Sun Feb 14 08:42:57 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Sun, 14 Feb 2010 17:42:57 +0100
Subject: [Freeswitch-users] mod_skypiax (skype endpoint) new timing
In-Reply-To: <7b197bef1002140840i6dfbdaecu26c1ae7e1f1a0633@mail.gmail.com>
References: <7b197bef1002131935j3fdf80c4w4050f10bea105d49@mail.gmail.com> 
	<201002141552.31159.errotan@gmail.com>
	<201002141609.18711.errotan@gmail.com> 
	<7b197bef1002140840i6dfbdaecu26c1ae7e1f1a0633@mail.gmail.com>
Message-ID: <7b197bef1002140842i3be561a6sb0c7f1b3adb3ca0b@mail.gmail.com>

I mean, snd-dummy still working very well on kernels at 100HZ, but the
new one is supposed to work with 1000HZ kernels too.

But the most changes are in mod_skypiax own timing, that are supposed
to sole the various delay problems that now and then have surfaced.

Let me know, guys 'n gals.

-gm

On Sun, Feb 14, 2010 at 5:40 PM, Giovanni Maruzzelli
<gmaruzz at celliax.org> wrote:
> Thanks Pusk?s,
>
> any problem of growing delay in sip->fs->skype or skype->fs->sip, or
> conferences, or whatever?
>
> Also, snd-dummy is now more compatible with kernels at 1000HZ (eg:
> centOS), although I believe Lenny is 100HZ.
>
> -giovanni
>
> On Sun, Feb 14, 2010 at 4:09 PM, Pusk?s Zsolt <errotan at gmail.com> wrote:
>> 2010. febru?r 14. 15.52.31 Pusk?s Zsolt d?tummal ezt ?rta:
>>> 2010. febru?r 14. 04.35.01 Giovanni Maruzzelli d?tummal ezt ?rta:
>>> > Hello FreeSWITCHers,
>>> >
>>> > I've just committed on svn16640 new timing for mod_skypiax, and I
>>> > would like if you guys give it a test in the various use cases.
>>> >
>>> > ciao for now,
>>> >
>>> > -giovanni
>>>
>>> It is not really working:
>>>
>>> 2010-02-14 15:37:39.182686 [ERR] skypiax_protocol.c:196 rev 16640M[(nil)|37
>>> ][ERRORA ?196 ?][gun_at_koli][-1, 0, 0] Skype got ERROR: |||ERROR 589 ALTER
>>> CALL: unable to alter input/output|||
>>> 2010-02-14 15:37:39.182686 [ERR] skypiax_protocol.c:198 rev 16640M[(nil)|37
>>> ][ERRORA ?198 ?][gun_at_koli][-1, 0,16] skype_call now is DOWN
>>> 2010-02-14 15:37:39.182686 [DEBUG] mod_skypiax.c:1068 rev 16640M[(nil)|37
>>> ][DEBUG_SKYPE ?1068 ][gun_at_koli][-1, 1,16] skype call ended
>>> 2010-02-14 15:37:39.182686 [DEBUG] mod_skypiax.c:1085 rev 16640M[(nil)|37
>>> ][DEBUG_SKYPE ?1085 ][gun_at_koli][-1, 1,16] no session
>>>
>>>
>>> Also can't unload mod_skypiax because the channels are up:
>>>
>>> freeswitch at internal> reload mod_skypiax
>>> -ERR unloading module [Module in use.]
>>>
>>> 2010-02-14 15:40:13.570678 [WARNING] switch_loadable_module.c:1268 Module
>>> mod_skypiax is in use, cannot unload.
>>> freeswitch at internal> show channels
>>> uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,de
>>> st,application,application_data,dialplan,context,read_codec,read_rate,write
>>> _codec,write_rate,secure,hostname,presence_id,presence_data
>>> ?4f302086-1976-11df-a1d8-47a9ffb132d9,inbound,2010-02-14
>>> 15:36:16,1266158176,skypiax/gun_at_koli,CS_EXECUTE,Pusk?s
>>> Zsolt,errotan,,4219,echo,,XML,default,L16,16000,L16,16000,,pc,,
>>> 80007580-1976-11df-a1d9-47a9ffb132d9,inbound,2010-02-14
>>> 15:37:38,1266158258,skypiax/gun_at_koli,CS_EXECUTE,Pusk?s
>>> Zsolt,errotan,,4219,echo,,XML,default,L16,16000,L16,16000,,pc,,
>>>
>>> 2 total.
>>>
>>> Have to do a fsctl shutdown to kill those channels.
>>>
>>> My home skype (errotan) client displays: Remote sound problem. when I try
>>> ?to call the other (gun_at_koli).
>>>
>>
>> Ooops sorry the user running the skype client were not in the audio group
>> therefore don't have access to snd_dummy.
>>
>> Works now! Tested on Debian "Lenny" x86
>>
>> :)
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618



From paul at apcl.us  Sun Feb 14 09:14:50 2010
From: paul at apcl.us (Paul Levin)
Date: Sun, 14 Feb 2010 12:14:50 -0500
Subject: [Freeswitch-users] can vm settings go in
 conf\directory\default\default.xml?
Message-ID: <4B782F8A.6010402@apcl.us>

An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100214/9eff7240/attachment-0002.html 

From vmaruani at interwise.com  Sun Feb 14 09:34:06 2010
From: vmaruani at interwise.com (Victor Maruani)
Date: Sun, 14 Feb 2010 19:34:06 +0200
Subject: [Freeswitch-users] Bypass-media and REFER method
In-Reply-To: <191c3a031002140654x37b65f1alc45907b545249cb4@mail.gmail.com>
References: <B816EF85C732094BA2016B2112F3DC2B0402C777@isr-brass.interwise.com><B816EF85C732094BA2016B2112F3DC2B040C5CA6@isr-brass.interwise.com><191c3a031002100645n5d827949xdefd24143f10759@mail.gmail.com><B816EF85C732094BA2016B2112F3DC2B040C6061@isr-brass.interwise.com><191c3a031002111501y7e5917b8g4faae4b8752e6fc2@mail.gmail.com><B816EF85C732094BA2016B2112F3DC2B040C693B@isr-brass.interwise.com><191c3a031002140645s358688c8g68280267d5d650d3@mail.gmail.com>
	<191c3a031002140654x37b65f1alc45907b545249cb4@mail.gmail.com>
Message-ID: <B816EF85C732094BA2016B2112F3DC2B04130D8C@isr-brass.interwise.com>

Hi, 

 

I understand your point and the reason I want to bypass media has a lot
to do with load/call volume.

I don't mind the notify series being implemented or not, the notify OK
is enough if it succeeded indeed.

I think the issue here is the order of action. 

 

Instead of making a long speech and lose my point allow me to put it
like this:

What I see today is:

a-      REFER is received

b-      FS immediately sends 202 accepted ----- ok

c-       FS sends NOTIFY OK to B ---- wrong (not true yet)

d-      FS sends reINVITE to A  ------ wrong (why not after FS gets C's
sdp)

e-      B is disconnected ------- wrong (same as above , and if C is
down I lose the caller)

f-       FS INVITE C  ------ ok 

 

Without getting in sip madness, I would change the order to a,b, F ,
c,d,e 

This way only do I have the sofia leg talking to the B-leg independently
during the REFER in my view.

What do you think?

 

In the meantime I'll try playing with the config but so far I don't have
it work.

Thanks,

 

Regards,

Victor.

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Anthony Minessale
Sent: Sunday, February 14, 2010 4:54 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Bypass-media and REFER method

 

We don't support the series of 100,180,200 in the notifies that is
typically a pure sip pbx feature.
We are a b2b and protocol agnostic softswitch.
When you do refer to a sofia leg you are talking to that leg
independantly.
We do not track the progress of what the channel does once you transfer
it, instead we send the channel back to the dialplan and accept the
refer.  This operation if sucessful will generate 202.
The resume media on hold and bypass after att xfer sofia options or
bypass_media_after_bridge var may be what you need if you want no media
on fs.
We have to make some sacrifices on sip madness to gain some general
flexibility and call volume. =/

	On Feb 14, 2010 3:39 AM, "Victor Maruani"
<vmaruani at interwise.com> wrote:

	Hi,

	 

	I would say it fails in 2 points:

	First in the fact that a "NOTIFY 200 OK" (line 1070) is sent
right after FS gets the REFER.

	 

	Then in the REINVITE (line 1094) sent to A (10.10.5.19)  just
after this NOTIFY,

	This REINVITE contains the SDP of the FS (10.10.5.92) causing
the A side to send media to FS. There will be no REINVITE with SDP of C
(10.10.5.48)

	 

	But as you say, just afterwards, the REFER action is actually
done and C is invited by FS with the SDP of A.

	 

	Conclusion :

	1)

	B is notified of success just after it sent the REFER and is
disconnected.

	B may be notified of every step of the connection to C (100
trying... 200 OK) when these actually happen.

	What if C is down? Can't FS notify a failure? (didn't test
that.)

	 

	2)

	'A' gets to send media to FS Because of a REINVITE which
disconnect him from B (we are in bypass media mode) . 

	during the process of REFER, A should be still connected to B
from a media perspective.

	The REINVITE is not done at the right time with the right
params.

	Here, a pseudo bridge (on way voice)  is established when C gets
the INVITE and is sending media to A.

	 

	A can hear C but C can't hear A after the REFER.

	If C was down, A would be "lost" in FS...

	 

	I believe the correct behavior would be:

	B sends REFER.

	FS INVITE C 

	C replies 100, 180... 200 and FS notifies B in accordance.

	Once C has sent 200 OK with its SDP. B is disconnected and A is
updated (REINVITE) with C's SDP.

	 

	Please share your thoughts, I still don't know if it's a bug or
if I configured something wrong although I don't think so.

	Hasn't anyone done that before?

	 

	Thank you.

	 

	 

	 

	From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Anthony Minessale
	Sent: Friday, February 12, 2010 1:02 AM

	
	To: freeswitch-users at lists.freeswitch.org
	Subject: Re: [Freeswitch-users] Bypass-media and REFER me...

	
	
	 
	
	Where do you think it's failing?
	that log shows it get refer, go back to dp, invite to 3341
and...

	
	_______________________________________________
	FreeSWITCH-users mailing list
	FreeSWITCH-users at lists.freeswitch.org
	http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
	
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
	http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100214/1ab52697/attachment-0002.html 

From anthony.minessale at gmail.com  Sun Feb 14 10:37:53 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sun, 14 Feb 2010 12:37:53 -0600
Subject: [Freeswitch-users] Bypass-media and REFER method
In-Reply-To: <191c3a031002141037n2e8823ebic0e283aafd0dc002@mail.gmail.com>
References: <B816EF85C732094BA2016B2112F3DC2B0402C777@isr-brass.interwise.com>
	<B816EF85C732094BA2016B2112F3DC2B040C5CA6@isr-brass.interwise.com>
	<191c3a031002100645n5d827949xdefd24143f10759@mail.gmail.com>
	<B816EF85C732094BA2016B2112F3DC2B040C6061@isr-brass.interwise.com>
	<191c3a031002111501y7e5917b8g4faae4b8752e6fc2@mail.gmail.com>
	<B816EF85C732094BA2016B2112F3DC2B040C693B@isr-brass.interwise.com>
	<191c3a031002140645s358688c8g68280267d5d650d3@mail.gmail.com>
	<191c3a031002140654x37b65f1alc45907b545249cb4@mail.gmail.com>
	<B816EF85C732094BA2016B2112F3DC2B04130D8C@isr-brass.interwise.com>
	<191c3a031002141037n2e8823ebic0e283aafd0dc002@mail.gmail.com>
Message-ID: <191c3a031002141037h715fd09bsdb4fb085bfcc3d09@mail.gmail.com>

Right we can't do it
Its difficult to add

On Feb 14, 2010 11:39 AM, "Victor Maruani" <vmaruani at interwise.com> wrote:

 Hi,



I understand your point and the reason I want to bypass media has a lot to
do with load/call volume.

I don't mind the notify series being implemented or not, the notify OK is
enough if it succeeded indeed.

I think the issue here is the order of action.



Instead of making a long speech and lose my point allow me to put it like
this:

What I see today is:

a-      REFER is received

b-      FS immediately sends 202 accepted ----- ok

c-       FS sends NOTIFY OK to B ---- wrong (not true yet)

d-      FS sends reINVITE to A  ------ wrong (why not after FS gets C's sdp)

e-      B is disconnected ------- wrong (same as above , and if C is down I
lose the caller)

f-       FS INVITE C  ------ ok



Without getting in sip madness, I would change the order to a,b, F , c,d,e

This way only do I have the sofia leg talking to the B-leg independently
during the REFER in my view.

What do you think?



In the meantime I'll try playing with the config but so far I don't have it
work.

Thanks,



Regards,

Victor.





*From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony
Minessale
*Sent:* Sunday, February 14, 2010 4:54 PM


To: freeswitch-users at lists.freeswitch.org
*Subject:* Re: [Freeswitch-users] Bypass-media and REFER method





We don't support the series of 100,180,200 in the notifies that is typically
a pure sip pbx fea...

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100214/ea7fe0ff/attachment-0002.html 

From gkuri at ieee.org  Sun Feb 14 11:03:35 2010
From: gkuri at ieee.org (Gabriel Kuri)
Date: Sun, 14 Feb 2010 11:03:35 -0800
Subject: [Freeswitch-users] Broadsoft SCA w/ Cisco SPA-5xx series
Message-ID: <8b1c9cda1002141103te523643ob16db25e3bfa877f@mail.gmail.com>

I followed Brian's directions from one of the previous threads on
configuring the SPA-5xx series phones for Broadsoft SCA and set
manage-shared-appearance=true in the internal profile. SCA appears to
be working on outgoing calls between two phones, the line key starts
flashing red on the second phone when the first phone picks up the
receiver to make a call. However on incoming calls, both phones ring
(same extension), however when one of the phones picks up the line,
the second phone's line key doesn't flash red or show the first phone
on that incoming call. Any ideas? Does shared appearance only work on
outgoing phone calls?

Thanks,
Gabe



From mbsip at gazeta.pl  Sun Feb 14 12:52:54 2010
From: mbsip at gazeta.pl (mbsip)
Date: Sun, 14 Feb 2010 21:52:54 +0100
Subject: [Freeswitch-users] can vm settings go in
	conf\directory\default\default.xml?
In-Reply-To: <4B782F8A.6010402@apcl.us>
References: <4B782F8A.6010402@apcl.us>
Message-ID: <28f27f5d1002141252x2088d193k5e57fa0006957df7@mail.gmail.com>

Paul,

VM is using sqlite as a default configuration. You can use Mysql for
instance, but first you need to play around with ODBC -->
http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core

mod_lua, mod_python or mod_xml_curl can help you to use the same Mysql db to
provide dynamic directory/dialplan information.
For more information run though mentioned modules and take the one you are
good at.

Thx,
Maciej.


2010/2/14 Paul Levin <paul at apcl.us>

>  Maciej,
>     Interesting approach Maciej.  I'm really just a beginner user of FS.
> Can you point me towards documentation that specifically describes the DB
> and how to access it?  If it matters, I'm running this on Windows.
>     Thanks,
>     Paul
>
>
>    Subject:
> Re: [Freeswitch-users] can vm settings go in
> conf\directory\default\default.xml?
> From:
> mbsip <mbsip at gazeta.pl> <mbsip at gazeta.pl>
> Date:
> Sun, 14 Feb 2010 17:01:47 +0100
> To:
> freeswitch-users at lists.freeswitch.org
>
> Paul,
>
> Maybe you should think about providing dynamic directory information
> by using DB.
> >From my point of view, Its much easier to manage DB than .xml files.
>
> Thx,
> Maciej.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100214/88603c99/attachment-0002.html 

From errotan at gmail.com  Sun Feb 14 14:12:47 2010
From: errotan at gmail.com (=?iso-8859-1?q?Pusk=E1s_Zsolt?=)
Date: Sun, 14 Feb 2010 23:12:47 +0100
Subject: [Freeswitch-users] mod_skypiax (skype endpoint) new timing
In-Reply-To: <7b197bef1002140842i3be561a6sb0c7f1b3adb3ca0b@mail.gmail.com>
References: <7b197bef1002131935j3fdf80c4w4050f10bea105d49@mail.gmail.com>
	<7b197bef1002140840i6dfbdaecu26c1ae7e1f1a0633@mail.gmail.com>
	<7b197bef1002140842i3be561a6sb0c7f1b3adb3ca0b@mail.gmail.com>
Message-ID: <201002142312.47548.errotan@gmail.com>

2010. febru?r 14. 17.42.57 Giovanni Maruzzelli d?tummal ezt ?rta:
> I mean, snd-dummy still working very well on kernels at 100HZ, but the
> new one is supposed to work with 1000HZ kernels too.
> 
> But the most changes are in mod_skypiax own timing, that are supposed
> to sole the various delay problems that now and then have surfaced.
> 
> Let me know, guys 'n gals.
> 
> -gm
> 
> On Sun, Feb 14, 2010 at 5:40 PM, Giovanni Maruzzelli
> 
> <gmaruzz at celliax.org> wrote:
> > Thanks Pusk?s,
> >
> > any problem of growing delay in sip->fs->skype or skype->fs->sip, or
> > conferences, or whatever?
> >
> > Also, snd-dummy is now more compatible with kernels at 1000HZ (eg:
> > centOS), although I believe Lenny is 100HZ.
> >
> > -giovanni
> >
> > On Sun, Feb 14, 2010 at 4:09 PM, Pusk?s Zsolt <errotan at gmail.com> wrote:
> >> 2010. febru?r 14. 15.52.31 Pusk?s Zsolt d?tummal ezt ?rta:
> >>> 2010. febru?r 14. 04.35.01 Giovanni Maruzzelli d?tummal ezt ?rta:
> >>> > Hello FreeSWITCHers,
> >>> >
> >>> > I've just committed on svn16640 new timing for mod_skypiax, and I
> >>> > would like if you guys give it a test in the various use cases.
> >>> >
> >>> > ciao for now,
> >>> >
> >>> > -giovanni
> >>>
> >>> It is not really working:
> >>>
> >>> 2010-02-14 15:37:39.182686 [ERR] skypiax_protocol.c:196 rev
> >>> 16640M[(nil)|37 ][ERRORA  196  ][gun_at_koli][-1, 0, 0] Skype got
> >>> ERROR: |||ERROR 589 ALTER CALL: unable to alter input/output|||
> >>> 2010-02-14 15:37:39.182686 [ERR] skypiax_protocol.c:198 rev
> >>> 16640M[(nil)|37 ][ERRORA  198  ][gun_at_koli][-1, 0,16] skype_call now
> >>> is DOWN 2010-02-14 15:37:39.182686 [DEBUG] mod_skypiax.c:1068 rev
> >>> 16640M[(nil)|37 ][DEBUG_SKYPE  1068 ][gun_at_koli][-1, 1,16] skype call
> >>> ended
> >>> 2010-02-14 15:37:39.182686 [DEBUG] mod_skypiax.c:1085 rev
> >>> 16640M[(nil)|37 ][DEBUG_SKYPE  1085 ][gun_at_koli][-1, 1,16] no session
> >>>
> >>>
> >>> Also can't unload mod_skypiax because the channels are up:
> >>>
> >>> freeswitch at internal> reload mod_skypiax
> >>> -ERR unloading module [Module in use.]
> >>>
> >>> 2010-02-14 15:40:13.570678 [WARNING] switch_loadable_module.c:1268
> >>> Module mod_skypiax is in use, cannot unload.
> >>> freeswitch at internal> show channels
> >>> uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_add
> >>>r,de
> >>> st,application,application_data,dialplan,context,read_codec,read_rate,w
> >>>rite _codec,write_rate,secure,hostname,presence_id,presence_data
> >>>  4f302086-1976-11df-a1d8-47a9ffb132d9,inbound,2010-02-14
> >>> 15:36:16,1266158176,skypiax/gun_at_koli,CS_EXECUTE,Pusk?s
> >>> Zsolt,errotan,,4219,echo,,XML,default,L16,16000,L16,16000,,pc,,
> >>> 80007580-1976-11df-a1d9-47a9ffb132d9,inbound,2010-02-14
> >>> 15:37:38,1266158258,skypiax/gun_at_koli,CS_EXECUTE,Pusk?s
> >>> Zsolt,errotan,,4219,echo,,XML,default,L16,16000,L16,16000,,pc,,
> >>>
> >>> 2 total.
> >>>
> >>> Have to do a fsctl shutdown to kill those channels.
> >>>
> >>> My home skype (errotan) client displays: Remote sound problem. when I
> >>> try to call the other (gun_at_koli).
> >>
> >> Ooops sorry the user running the skype client were not in the audio
> >> group therefore don't have access to snd_dummy.
> >>
> >> Works now! Tested on Debian "Lenny" x86
> >>
> >> :)
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> > --
> > Sincerely,
> >
> > Giovanni Maruzzelli
> > Cell : +39-347-2665618
> 

My test was skype -> fs (echo && delay_echo) .
The echo call was 10 min, the delayed echo was 1 hour long.

Now i tried with skype -> fs -> sip.
I done a 15 minute call no delay issues.



From anthony.minessale at gmail.com  Sun Feb 14 14:48:50 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sun, 14 Feb 2010 16:48:50 -0600
Subject: [Freeswitch-users] Bypass-media and REFER method
In-Reply-To: <191c3a031002141037h715fd09bsdb4fb085bfcc3d09@mail.gmail.com>
References: <B816EF85C732094BA2016B2112F3DC2B0402C777@isr-brass.interwise.com>
	<191c3a031002100645n5d827949xdefd24143f10759@mail.gmail.com>
	<B816EF85C732094BA2016B2112F3DC2B040C6061@isr-brass.interwise.com>
	<191c3a031002111501y7e5917b8g4faae4b8752e6fc2@mail.gmail.com>
	<B816EF85C732094BA2016B2112F3DC2B040C693B@isr-brass.interwise.com>
	<191c3a031002140645s358688c8g68280267d5d650d3@mail.gmail.com>
	<191c3a031002140654x37b65f1alc45907b545249cb4@mail.gmail.com>
	<B816EF85C732094BA2016B2112F3DC2B04130D8C@isr-brass.interwise.com>
	<191c3a031002141037n2e8823ebic0e283aafd0dc002@mail.gmail.com>
	<191c3a031002141037h715fd09bsdb4fb085bfcc3d09@mail.gmail.com>
Message-ID: <191c3a031002141448w2749ab61r94b40ab37e83103e@mail.gmail.com>

Let me try to explain better now that I am not on my cell.

My original explanation already stated that the receipt of the refer on a
blind xfer is always answered with a 202 because the call will transfer to
the desired extension with success.  That does not mean that the subsequent
call that is going to take place when that leg hits the dialplan and tries
to make another outbound call will be successful as well.  There is no way
to predict that.  It could easily be a conference or moh or some other 1
legged call.  The 2nd leg in your description cannot exist unless the leg
you are transferring goes back to the dialplan.

The call you sent the REFER to already has moved on in the FS state machine
and there would be no way to go back.

This is what I was trying to explain when I said we have to sacrifice some
seemingly easy features from one perspective to gain all the other things we
can do in FreeSWITCH.

So its not 100% impossible to code but it currently does not exist and I
would be concerned trying to do it would blur the abstraction lines in the
code.  So the short answer is, no, we don't support what you are asking
about.




On Sun, Feb 14, 2010 at 12:37 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> Right we can't do it
> Its difficult to add
>
> On Feb 14, 2010 11:39 AM, "Victor Maruani" <vmaruani at interwise.com> wrote:
>
>  Hi,
>
>
>
> I understand your point and the reason I want to bypass media has a lot to
> do with load/call volume.
>
> I don't mind the notify series being implemented or not, the notify OK is
> enough if it succeeded indeed.
>
> I think the issue here is the order of action.
>
>
>
> Instead of making a long speech and lose my point allow me to put it like
> this:
>
> What I see today is:
>
> a-      REFER is received
>
> b-      FS immediately sends 202 accepted ----- ok
>
> c-       FS sends NOTIFY OK to B ---- wrong (not true yet)
>
> d-      FS sends reINVITE to A  ------ wrong (why not after FS gets C's
> sdp)
>
> e-      B is disconnected ------- wrong (same as above , and if C is down
> I lose the caller)
>
> f-       FS INVITE C  ------ ok
>
>
>
> Without getting in sip madness, I would change the order to a,b, F , c,d,e
>
> This way only do I have the sofia leg talking to the B-leg independently
> during the REFER in my view.
>
> What do you think?
>
>
>
> In the meantime I'll try playing with the config but so far I don't have it
> work.
>
> Thanks,
>
>
>
> Regards,
>
> Victor.
>
>
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony
> Minessale
> *Sent:* Sunday, February 14, 2010 4:54 PM
>
>
> To: freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Bypass-media and REFER method
>
>
>
>
>
> We don't support the series of 100,180,200 in the notifies that is
> typically a pure sip pbx fea...
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100214/c35b9aa9/attachment-0002.html 

From anthony.minessale at gmail.com  Sun Feb 14 15:42:14 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sun, 14 Feb 2010 17:42:14 -0600
Subject: [Freeswitch-users] can vm settings go in
	conf\directory\default\default.xml?
In-Reply-To: <191c3a031002141541n2358d92kf80e35134e1a43f9@mail.gmail.com>
References: <4B782F8A.6010402@apcl.us>
	<28f27f5d1002141252x2088d193k5e57fa0006957df7@mail.gmail.com>
	<191c3a031002141541n2358d92kf80e35134e1a43f9@mail.gmail.com>
Message-ID: <191c3a031002141542w50e92c9k4fc321f74b68f0a9@mail.gmail.com>

Vm users do not inherit params from the domain there is a patch coming to
fix this soon.

On Feb 14, 2010 2:59 PM, "mbsip" <mbsip at gazeta.pl> wrote:

Paul,

VM is using sqlite as a default configuration. You can use Mysql for
instance, but first you need to play around with ODBC -->
http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core

mod_lua, mod_python or mod_xml_curl can help you to use the same Mysql db to
provide dynamic directory/dialplan information.
For more information run though mentioned modules and take the one you are
good at.

Thx,
Maciej.


2010/2/14 Paul Levin <paul at apcl.us>

> >
> > Maciej,
> >     Interesting approach Maciej.  I'm really just a beginner user of FS.
> Can you poin...
>
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users...
>


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100214/d10afec7/attachment-0002.html 

From mcampbellsmith at gmail.com  Sun Feb 14 17:08:49 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Mon, 15 Feb 2010 12:08:49 +1100
Subject: [Freeswitch-users] Forbidden using UDP, works with TCP/TLS
In-Reply-To: <9D702362-BB5F-4C0F-9EE7-DABA3B926C36@freeswitch.org>
References: <33c87fa31002101824h21e3bf7lb13b68e8d4fd36a2@mail.gmail.com>
	<33c87fa31002110131w3cf6ad00n802669856085a879@mail.gmail.com>
	<CA21231C-108B-4445-A3B9-BF8D94FFE898@jerris.com>
	<33c87fa31002110213q1de0eebfsb950d8725ea9aba@mail.gmail.com>
	<0A905834-850F-4D1F-AEC5-D3A87CE4141B@jerris.com>
	<33c87fa31002121629k74f26807i131841aa4252461b@mail.gmail.com>
	<33c87fa31002130343k7a01e9abt9c33360399f246f8@mail.gmail.com>
	<C37AB6F3-F712-41B6-9D6D-7A8AE3C6CABB@freeswitch.org>
	<33c87fa31002131336j7b05db7aof6ba6e3439aebc41@mail.gmail.com>
	<9D702362-BB5F-4C0F-9EE7-DABA3B926C36@freeswitch.org>
Message-ID: <33c87fa31002141708s72cb7246h5372ecea40be615d@mail.gmail.com>

Hi,

The sip trace provided only contains 4 SIP messages.  Do you need the
IP's to decode the messages?

from udp/[121.xxx.xxx.xxx] is the SPA3102
from (udp/192.168.1.120:5060) is FS server

I can register the device to all my voip providers successfully using
UDP but it will not register to FS using UDP.
It can register using TLS and TCP.

Very confusing as to why that would be.


On Sun, Feb 14, 2010 at 10:11 AM, Brian West <brian at freeswitch.org> wrote:
> You know you could have obscured the first part of the IP and not the LAST... kinda removes the ability to tell WHO sent what.
>
> >From that log I guess your password is wrong.
>
> /b
>
> On Feb 13, 2010, at 3:36 PM, Mark Campbell-Smith wrote:
>
>> Thanks Brian.
>>
>> The full log is pasted here http://pastebin.freeswitch.org/12133
>>
>>
>>
>> On Sun, Feb 14, 2010 at 2:08 AM, Brian West <brian at freeswitch.org> wrote:
>>> Can't tell since you keep cutting the lines required to figure this out... FULL console log with FULL sip trace. ?You always cut out what I wanna see. ?Just get a pcap.
>>>
>>> /b
>>>
>>> On Feb 13, 2010, at 5:43 AM, Mark Campbell-Smith wrote:
>>>
>>>> More testing. The device registers successfully to my SIP provider
>>>> directly using UDP - why would FS be rejecting the registration
>>>> request?
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From jmesquita at freeswitch.org  Sun Feb 14 17:28:24 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Sun, 14 Feb 2010 22:28:24 -0300
Subject: [Freeswitch-users] play_and_get_digits + OutboundESL
Message-ID: <ebbc2d551002141728xa335781qecaa284d32544ae0@mail.gmail.com>

Gentleman,

I am trying to use TTS with play_and_get_digits with Outbound ESL with
little luck.

There seems to be some kind of parsing problem that I don't really know how
to solve.

Here is the command (my python script) that I am issuing:

http://pastebin.freeswitch.org/12147

Here is the FreeSWITCH log output:

http://pastebin.freeswitch.org/12146

And finally here is the ESL log:

http://pastebin.freeswitch.org/12145

It needs to be stated that the tts_engine and tts_voice channel vars have
been set before on the script.

If I use these commands the exact same way on the dialplan, it works. Is
this a bug or am I overlooking something really obvious?

Regards,
JM
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100214/758c3b36/attachment-0002.html 

From infos at madovsky.org  Sun Feb 14 09:58:43 2010
From: infos at madovsky.org (Madovsky)
Date: Sun, 14 Feb 2010 12:58:43 -0500
Subject: [Freeswitch-users] inbound outbound transparent proxy question
Message-ID: <059C32DA5C3647908A1499C8027464B4@MOBILEE1705>

Hi,

I'm new to freeswitch world and apologize if my question is not relevant.
I need to install freeswitch as inbound outbound transparent proxy (media only if possible, but not sure it's the right terms to use)
example :


Caller A (he can be anonymous with his own provider) connect with a softphone we provide in my network (nat and public)
so he sets the softphone as user callerA, domain iptel.org (which is not my domain) proxy 192.168.0.1 (which is my FS proxy).
So he calls an external user (who is unknown) like callerB at bobo.dot. So callerB receive the invite (until now he can see the caller user name but
with proxy ip subsituted from the domain (but I want the original caller domain kept intact) and the call is established.
But, for the contrary (callerB wants to call callerA), it doesn't work.

Is anyone have idea or example of this kind of configuration ?

Best Regards

Franck Chionna
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100214/8f0df933/attachment-0002.html 

From infos at madovsky.org  Sun Feb 14 11:33:19 2010
From: infos at madovsky.org (Madovsky)
Date: Sun, 14 Feb 2010 14:33:19 -0500
Subject: [Freeswitch-users] Fw: inbound outbound transparent proxy question
Message-ID: <650FD98CB36544129A1C325A0DBD95AB@MOBILEE1705>

Hi,

I'm new to freeswitch world and apologize if my question is not relevant.
I need to install freeswitch as inbound outbound transparent proxy (media only if possible, but not sure it's the right terms to use)
example :


Caller A (he can be anonymous with his own provider) connect with a softphone we provide in my network (nat and public)
so he sets the softphone as user callerA, domain iptel.org (which is not my domain) proxy 192.168.0.1 (which is my FS proxy).
So he calls an external user (who is unknown) like callerB at bobo.dot. So callerB receive the invite (until now he can see the caller user name but
with proxy ip subsituted from the domain (but I want the original caller domain kept intact) and the call is established.
But, for the contrary (callerB wants to call callerA), it doesn't work.

Is anyone have idea or example of this kind of configuration ?

Best Regards

Franck Chionna
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100214/c4c213a6/attachment-0002.html 

From brian at freeswitch.org  Sun Feb 14 19:58:15 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 14 Feb 2010 21:58:15 -0600
Subject: [Freeswitch-users] Broadsoft SCA w/ Cisco SPA-5xx series
In-Reply-To: <8b1c9cda1002141103te523643ob16db25e3bfa877f@mail.gmail.com>
References: <8b1c9cda1002141103te523643ob16db25e3bfa877f@mail.gmail.com>
Message-ID: <0D67132C-69D8-43AD-B7E6-C7545B544322@freeswitch.org>

Works fine here... is your box slow or something?

/b

On Feb 14, 2010, at 1:03 PM, Gabriel Kuri wrote:

> I followed Brian's directions from one of the previous threads on
> configuring the SPA-5xx series phones for Broadsoft SCA and set
> manage-shared-appearance=true in the internal profile. SCA appears to
> be working on outgoing calls between two phones, the line key starts
> flashing red on the second phone when the first phone picks up the
> receiver to make a call. However on incoming calls, both phones ring
> (same extension), however when one of the phones picks up the line,
> the second phone's line key doesn't flash red or show the first phone
> on that incoming call. Any ideas? Does shared appearance only work on
> outgoing phone calls?
> 
> Thanks,
> Gabe




From brian at freeswitch.org  Sun Feb 14 20:07:31 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 14 Feb 2010 22:07:31 -0600
Subject: [Freeswitch-users] Forbidden using UDP, works with TCP/TLS
In-Reply-To: <33c87fa31002141708s72cb7246h5372ecea40be615d@mail.gmail.com>
References: <33c87fa31002101824h21e3bf7lb13b68e8d4fd36a2@mail.gmail.com>
	<33c87fa31002110131w3cf6ad00n802669856085a879@mail.gmail.com>
	<CA21231C-108B-4445-A3B9-BF8D94FFE898@jerris.com>
	<33c87fa31002110213q1de0eebfsb950d8725ea9aba@mail.gmail.com>
	<0A905834-850F-4D1F-AEC5-D3A87CE4141B@jerris.com>
	<33c87fa31002121629k74f26807i131841aa4252461b@mail.gmail.com>
	<33c87fa31002130343k7a01e9abt9c33360399f246f8@mail.gmail.com>
	<C37AB6F3-F712-41B6-9D6D-7A8AE3C6CABB@freeswitch.org>
	<33c87fa31002131336j7b05db7aof6ba6e3439aebc41@mail.gmail.com>
	<9D702362-BB5F-4C0F-9EE7-DABA3B926C36@freeswitch.org>
	<33c87fa31002141708s72cb7246h5372ecea40be615d@mail.gmail.com>
Message-ID: <1888B1CA-D0A8-407C-9C44-B86E0F1925D6@freeswitch.org>

Is the device behind nat with your FreeSWITCH?  If so disable stun on the device.  If FS is 192.168.1.120 and your device is 121.x.x.x something then I suspect its doing a hair pin thru your router.  Your network is busted which is my final answer.  

/b

On Feb 14, 2010, at 7:08 PM, Mark Campbell-Smith wrote:

> Hi,
> 
> The sip trace provided only contains 4 SIP messages.  Do you need the
> IP's to decode the messages?
> 
> from udp/[121.xxx.xxx.xxx] is the SPA3102
> from (udp/192.168.1.120:5060) is FS server
> 
> I can register the device to all my voip providers successfully using
> UDP but it will not register to FS using UDP.
> It can register using TLS and TCP.
> 
> Very confusing as to why that would be.




From mcampbellsmith at gmail.com  Sun Feb 14 20:17:15 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Mon, 15 Feb 2010 15:17:15 +1100
Subject: [Freeswitch-users] Forbidden using UDP, works with TCP/TLS
In-Reply-To: <1888B1CA-D0A8-407C-9C44-B86E0F1925D6@freeswitch.org>
References: <33c87fa31002101824h21e3bf7lb13b68e8d4fd36a2@mail.gmail.com>
	<33c87fa31002110213q1de0eebfsb950d8725ea9aba@mail.gmail.com>
	<0A905834-850F-4D1F-AEC5-D3A87CE4141B@jerris.com>
	<33c87fa31002121629k74f26807i131841aa4252461b@mail.gmail.com>
	<33c87fa31002130343k7a01e9abt9c33360399f246f8@mail.gmail.com>
	<C37AB6F3-F712-41B6-9D6D-7A8AE3C6CABB@freeswitch.org>
	<33c87fa31002131336j7b05db7aof6ba6e3439aebc41@mail.gmail.com>
	<9D702362-BB5F-4C0F-9EE7-DABA3B926C36@freeswitch.org>
	<33c87fa31002141708s72cb7246h5372ecea40be615d@mail.gmail.com>
	<1888B1CA-D0A8-407C-9C44-B86E0F1925D6@freeswitch.org>
Message-ID: <33c87fa31002142017v49d8e7c5nf572d086ec89282b@mail.gmail.com>

FS and the ATA are on different networks.  FS is nat'd (192.168.1.120,
upnp enabled on the router) and the ATA is on the internet at another
location.

Any other ideas?

On Mon, Feb 15, 2010 at 3:07 PM, Brian West <brian at freeswitch.org> wrote:
> Is the device behind nat with your FreeSWITCH? ?If so disable stun on the device. ?If FS is 192.168.1.120 and your device is 121.x.x.x something then I suspect its doing a hair pin thru your router. ?Your network is busted which is my final answer.
>
> /b
>
> On Feb 14, 2010, at 7:08 PM, Mark Campbell-Smith wrote:
>
>> Hi,
>>
>> The sip trace provided only contains 4 SIP messages. ?Do you need the
>> IP's to decode the messages?
>>
>> from udp/[121.xxx.xxx.xxx] is the SPA3102
>> from (udp/192.168.1.120:5060) is FS server
>>
>> I can register the device to all my voip providers successfully using
>> UDP but it will not register to FS using UDP.
>> It can register using TLS and TCP.
>>
>> Very confusing as to why that would be.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From mcampbellsmith at gmail.com  Sun Feb 14 21:09:35 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Mon, 15 Feb 2010 16:09:35 +1100
Subject: [Freeswitch-users] Forbidden using UDP, works with TCP/TLS
In-Reply-To: <33c87fa31002142017v49d8e7c5nf572d086ec89282b@mail.gmail.com>
References: <33c87fa31002101824h21e3bf7lb13b68e8d4fd36a2@mail.gmail.com>
	<0A905834-850F-4D1F-AEC5-D3A87CE4141B@jerris.com>
	<33c87fa31002121629k74f26807i131841aa4252461b@mail.gmail.com>
	<33c87fa31002130343k7a01e9abt9c33360399f246f8@mail.gmail.com>
	<C37AB6F3-F712-41B6-9D6D-7A8AE3C6CABB@freeswitch.org>
	<33c87fa31002131336j7b05db7aof6ba6e3439aebc41@mail.gmail.com>
	<9D702362-BB5F-4C0F-9EE7-DABA3B926C36@freeswitch.org>
	<33c87fa31002141708s72cb7246h5372ecea40be615d@mail.gmail.com>
	<1888B1CA-D0A8-407C-9C44-B86E0F1925D6@freeswitch.org>
	<33c87fa31002142017v49d8e7c5nf572d086ec89282b@mail.gmail.com>
Message-ID: <33c87fa31002142109l5fcabed3ydba21b865ed6cbbb@mail.gmail.com>

A little more testing.    I noticed that the Authorization field
differs when TCP or UDP:

UDP (fails)
Digest username=\"2010\", realm=\"mydns.dyndns.org\",
nonce=\"e5f119c6-19e9-11df-bd09-773b7a755f78\",
uri=\"sip:mydns.dyndns.org:5060\",
response=\"e37be3e49c159d4f98e8bd04b36f2bd7\", algorithm=MD5,
qop=\"1fffcc9f\"

TCP (works)
Digest username=\"2010\",realm=\"mydns.dyndns.org\",nonce=\"5d9e75c2-19ea-11df-bd0b-773b7a755f78\",uri=\"sip:mydns.dyndns.org:5060\",algorithm=MD5,response=\"45ba55d3fbafcbf2bc2aa6418656ecc2\",qop=auth,nc=00000001,cnonce=\"3a650454\"

Is qop = 1fffcc9f valid in SIP?  Does a cnonce need to be included also?

FS sends qop=auth in the Unauthorized response.

Thanks



On Mon, Feb 15, 2010 at 3:17 PM, Mark Campbell-Smith
<mcampbellsmith at gmail.com> wrote:
> FS and the ATA are on different networks. ?FS is nat'd (192.168.1.120,
> upnp enabled on the router) and the ATA is on the internet at another
> location.
>
> Any other ideas?
>
> On Mon, Feb 15, 2010 at 3:07 PM, Brian West <brian at freeswitch.org> wrote:
>> Is the device behind nat with your FreeSWITCH? ?If so disable stun on the device. ?If FS is 192.168.1.120 and your device is 121.x.x.x something then I suspect its doing a hair pin thru your router. ?Your network is busted which is my final answer.
>>
>> /b
>>
>> On Feb 14, 2010, at 7:08 PM, Mark Campbell-Smith wrote:
>>
>>> Hi,
>>>
>>> The sip trace provided only contains 4 SIP messages. ?Do you need the
>>> IP's to decode the messages?
>>>
>>> from udp/[121.xxx.xxx.xxx] is the SPA3102
>>> from (udp/192.168.1.120:5060) is FS server
>>>
>>> I can register the device to all my voip providers successfully using
>>> UDP but it will not register to FS using UDP.
>>> It can register using TLS and TCP.
>>>
>>> Very confusing as to why that would be.
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>



From brian at freeswitch.org  Sun Feb 14 21:18:48 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 14 Feb 2010 23:18:48 -0600
Subject: [Freeswitch-users] Forbidden using UDP, works with TCP/TLS
In-Reply-To: <33c87fa31002142109l5fcabed3ydba21b865ed6cbbb@mail.gmail.com>
References: <33c87fa31002101824h21e3bf7lb13b68e8d4fd36a2@mail.gmail.com>
	<0A905834-850F-4D1F-AEC5-D3A87CE4141B@jerris.com>
	<33c87fa31002121629k74f26807i131841aa4252461b@mail.gmail.com>
	<33c87fa31002130343k7a01e9abt9c33360399f246f8@mail.gmail.com>
	<C37AB6F3-F712-41B6-9D6D-7A8AE3C6CABB@freeswitch.org>
	<33c87fa31002131336j7b05db7aof6ba6e3439aebc41@mail.gmail.com>
	<9D702362-BB5F-4C0F-9EE7-DABA3B926C36@freeswitch.org>
	<33c87fa31002141708s72cb7246h5372ecea40be615d@mail.gmail.com>
	<1888B1CA-D0A8-407C-9C44-B86E0F1925D6@freeswitch.org>
	<33c87fa31002142017v49d8e7c5nf572d086ec89282b@mail.gmail.com>
	<33c87fa31002142109l5fcabed3ydba21b865ed6cbbb@mail.gmail.com>
Message-ID: <308A282B-27A0-497B-B250-ED2EC02D0BD5@freeswitch.org>

Sounds like the device is BUSTED.  :P

/b

On Feb 14, 2010, at 11:09 PM, Mark Campbell-Smith wrote:

> A little more testing.    I noticed that the Authorization field
> differs when TCP or UDP:
> 
> UDP (fails)
> Digest username=\"2010\", realm=\"mydns.dyndns.org\",
> nonce=\"e5f119c6-19e9-11df-bd09-773b7a755f78\",
> uri=\"sip:mydns.dyndns.org:5060\",
> response=\"e37be3e49c159d4f98e8bd04b36f2bd7\", algorithm=MD5,
> qop=\"1fffcc9f\"
> 
> TCP (works)
> Digest username=\"2010\",realm=\"mydns.dyndns.org\",nonce=\"5d9e75c2-19ea-11df-bd0b-773b7a755f78\",uri=\"sip:mydns.dyndns.org:5060\",algorithm=MD5,response=\"45ba55d3fbafcbf2bc2aa6418656ecc2\",qop=auth,nc=00000001,cnonce=\"3a650454\"
> 
> Is qop = 1fffcc9f valid in SIP?  Does a cnonce need to be included also?
> 
> FS sends qop=auth in the Unauthorized response.
> 
> Thanks

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100214/ee387f1a/attachment-0002.html 

From nagalenoj at gmail.com  Sun Feb 14 21:31:54 2010
From: nagalenoj at gmail.com (Nagalenoj H.)
Date: Mon, 15 Feb 2010 11:01:54 +0530
Subject: [Freeswitch-users] Play music to A leg.
In-Reply-To: <191c3a031002120907l28fbdf2dgab5df7dd1b5a2f76@mail.gmail.com>
References: <fe67a3341002090519r2cc6e8edm7702b8d012fc2be3@mail.gmail.com>
	<87f2f3b91002090737l4efa8en29bde9457fef7ea5@mail.gmail.com>
	<fe67a3341002092157x7484f3d2id6678510bbec66cc@mail.gmail.com>
	<87f2f3b91002100702u7581589ua866d2492a008b23@mail.gmail.com>
	<fe67a3341002110211w60390e1cm2d12fadd1e802778@mail.gmail.com>
	<87f2f3b91002110754laf9d01dxa94a36bc1b8c83df@mail.gmail.com>
	<191c3a031002110804m4c18e7e0y317f44272b42c8b4@mail.gmail.com>
	<7d79b3931002112018y2512954cuca83ce21ff3406fe@mail.gmail.com>
	<191c3a031002120907l28fbdf2dgab5df7dd1b5a2f76@mail.gmail.com>
Message-ID: <fe67a3341002142131m28d7bc5ao8028c4ce8706f97c@mail.gmail.com>

Usually it works as follows,
bridge {group_confirm_key=exec,group_confirm_file=perl xx.pl
}user/1000,user/1001,user/1002

But, I want it like,
bridge [group_confirm_key=exec,group_confirm_file=perl
xx.pl]user/1000,[group_confirm_key=exec,group_confirm_file=perl
xx.pl]user/1001,user/1002

So, I don't want the script to be executed if 1002 answers the call.

Also, I need only one to answer the call. When the first person answers the
call, the other extensions have to stop ringing immediately.
But, this works as 'bridges with who completes the script first',

On Fri, Feb 12, 2010 at 10:37 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> the script executes for everyone and gives them a chance to dial multiple
> digits to test for, this is what he asked for, instead of 1 digit dial
> multiple digits.  you set the correct string as a variable on the channel
> and everybody runs the script and whoever dials the right digits wins the
> rest will be hungup on.
>
>
>
> On Thu, Feb 11, 2010 at 10:18 PM, lakshmanan ganapathy <
> lakindia89 at gmail.com> wrote:
>
>> Dear Antony,
>> In bridge if we are making parallel calls, then group_confirm_key in
>> execute mode will execute for all the extensions, and whomsoever finishes
>> the script first, will be bridged.
>>
>> But I think nagalenoj need to execute the script for the extension which
>> answers the call first, not for all the extension.!!!.
>>
>> From nanalenoj's post
>>
>>             " but when I do originate for multiple extensions, I want this
>> script to work based on the answered extension."
>>
>>
>> On Thu, Feb 11, 2010 at 9:34 PM, Anthony Minessale <
>> anthony.minessale at gmail.com> wrote:
>>
>>> group_confirm_key in execute mode can execute a lua script instead that
>>> can read as many digits as you want and parse the results.
>>>
>>>
>>>
>>> On Thu, Feb 11, 2010 at 9:54 AM, Michael Collins <msc at freeswitch.org>wrote:
>>>
>>>> Hehe, this is getting more and more complicated. You may want to
>>>> consider using the event socket and have your call control be done from a
>>>> more 3rd party-ish perspective. If you've got all these different scenarios
>>>> it might be better to let an external script do all the work.
>>>>
>>>> http://wiki.freeswitch.org/wiki/Event_Socket
>>>>
>>>> -MC
>>>>
>>>>
>>>> On Thu, Feb 11, 2010 at 2:11 AM, Nagalenoj H. <nagalenoj at gmail.com>wrote:
>>>>
>>>>> But My scenario is,
>>>>>    After I get the call from X.
>>>>>    I answer the call in some scenarios and won't answer the call. So,
>>>>> this leg can either be answered or unanswered.
>>>>>    I originate a call to another number.
>>>>>    After getting some digits from this originated leg.
>>>>>    I do uuid_bridge of these 2 legs.
>>>>>
>>>>> I want to play some file[ringback] to leg A before bridging to B.
>>>>>
>>>>> On Wed, Feb 10, 2010 at 8:32 PM, Michael Collins <msc at freeswitch.org>wrote:
>>>>>
>>>>>>
>>>>>>
>>>>>> On Tue, Feb 9, 2010 at 9:57 PM, Nagalenoj H. <nagalenoj at gmail.com>wrote:
>>>>>>
>>>>>>> Because, I want to get some digits before bridging the legs. I've
>>>>>>> tried group_confirm_key, but it accepts only one digit, I need multiple
>>>>>>> digits, so I can't use.
>>>>>>> I've also tried group_confirm_file, but when I do originate for
>>>>>>> multiple extensions, I want this script to work based on the answered
>>>>>>> extension.
>>>>>>>
>>>>>>> So, I've originated and processed the events to do my job.
>>>>>>>
>>>>>>> How do I play some music to A leg?
>>>>>>>
>>>>>>> I might be missing something, but couldn't you just park the call ("A
>>>>>> leg") until you connect to the other party ("B leg") and then uuid_bridge at
>>>>>> whatever point you want?
>>>>>> -MC
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> Regards,
>>>>> Nagalenoj H.
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>> iax:guest at conference.freeswitch.org/888
>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> pstn:+19193869900
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Regards,
Nagalenoj H.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100215/aae903bf/attachment-0002.html 

From gorand at donevtechconsulting.com  Sun Feb 14 20:47:14 2010
From: gorand at donevtechconsulting.com (Goran Donev)
Date: Sun, 14 Feb 2010 22:47:14 -0600
Subject: [Freeswitch-users] Version 1.5
In-Reply-To: <mailman.6373.1266207467.2495.freeswitch-users@lists.freeswitch.org>
References: <mailman.6373.1266207467.2495.freeswitch-users@lists.freeswitch.org>
Message-ID: <041101caadf9$f36d77e0$da4867a0$@com>

When is version 1.5 of free switch going to be released. The last update on
the website was on the week of Feb 8th. I still have not seen anything else.


Thanks




From jaybinks at gmail.com  Sun Feb 14 21:42:40 2010
From: jaybinks at gmail.com (jay binks)
Date: Mon, 15 Feb 2010 15:42:40 +1000
Subject: [Freeswitch-users] Version 1.5
In-Reply-To: <041101caadf9$f36d77e0$da4867a0$@com>
References: <mailman.6373.1266207467.2495.freeswitch-users@lists.freeswitch.org>
	<041101caadf9$f36d77e0$da4867a0$@com>
Message-ID: <ec2ce17e1002142142g1d71be24yf55132344b880514@mail.gmail.com>

did you contribute to the dinner ??
maybe thats why it hasnt been released yet ...

Jokes...

J

On Mon, Feb 15, 2010 at 2:47 PM, Goran Donev <gorand at donevtechconsulting.com
> wrote:

> When is version 1.5 of free switch going to be released. The last update on
> the website was on the week of Feb 8th. I still have not seen anything
> else.
>
>
> Thanks
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely

Jay
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100215/c6728b40/attachment-0002.html 

From gkuri at ieee.org  Sun Feb 14 21:50:58 2010
From: gkuri at ieee.org (Gabriel Kuri)
Date: Sun, 14 Feb 2010 21:50:58 -0800
Subject: [Freeswitch-users] Broadsoft SCA w/ Cisco SPA-5xx series
In-Reply-To: <0D67132C-69D8-43AD-B7E6-C7545B544322@freeswitch.org>
References: <8b1c9cda1002141103te523643ob16db25e3bfa877f@mail.gmail.com>
	<0D67132C-69D8-43AD-B7E6-C7545B544322@freeswitch.org>
Message-ID: <8b1c9cda1002142150i48e17045yac9596cb32ee6ee2@mail.gmail.com>

It's an Atom N330. Not sure why it doesn't work on incoming calls, are
there any other settings that need to be set on the phones other than
setting the lines to shared and server type to Broadsoft?

Not sure if this matters, since everything else seems to be working,
but the phones are behind one NAT and FreeSWITCH is behind a totally
different NAT. Everything else seems to be working, I don't have any
one way audio or other funny things going on that would point to NAT,
so I'm not sure NAT is the issue.

What should be sent to the phone, to light up the light after the
other phone is answered, a NOTIFY? I'm seeing a NOTIFY sent to the
other phone with Event: call-info, but the light isn't turning on to
indicate SCA.

Thanks,
Gabe





On Sun, Feb 14, 2010 at 7:58 PM, Brian West <brian at freeswitch.org> wrote:
> Works fine here... is your box slow or something?
>
> /b
>
> On Feb 14, 2010, at 1:03 PM, Gabriel Kuri wrote:
>
>> I followed Brian's directions from one of the previous threads on
>> configuring the SPA-5xx series phones for Broadsoft SCA and set
>> manage-shared-appearance=true in the internal profile. SCA appears to
>> be working on outgoing calls between two phones, the line key starts
>> flashing red on the second phone when the first phone picks up the
>> receiver to make a call. However on incoming calls, both phones ring
>> (same extension), however when one of the phones picks up the line,
>> the second phone's line key doesn't flash red or show the first phone
>> on that incoming call. Any ideas? Does shared appearance only work on
>> outgoing phone calls?
>>
>> Thanks,
>> Gabe
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From gkuri at ieee.org  Sun Feb 14 21:59:35 2010
From: gkuri at ieee.org (Gabriel Kuri)
Date: Sun, 14 Feb 2010 21:59:35 -0800
Subject: [Freeswitch-users] Broadsoft SCA w/ Cisco SPA-5xx series
In-Reply-To: <0D67132C-69D8-43AD-B7E6-C7545B544322@freeswitch.org>
References: <8b1c9cda1002141103te523643ob16db25e3bfa877f@mail.gmail.com>
	<0D67132C-69D8-43AD-B7E6-C7545B544322@freeswitch.org>
Message-ID: <8b1c9cda1002142159h5efcc765q9a41baa1bf694c8c@mail.gmail.com>

BTW, here's a copy of the NOTIFY (event call-info) sent to the other
phone after the first phone is answered, should this have a Call-Info
line with an "appearance-state=seized" to turn on the light on the
other phone?


NOTIFY sip:2551@<removed>:54446 SIP/2.0.
Via: SIP/2.0/UDP <removed>:9430;rport;branch=z9hG4bK71pN2cXgH851K.
Max-Forwards: 70.
From: <sip:2551 at mydomain.com>;tag=XeB6ZrKDevpHp.
To: <sip:2551 at mydomain.com>;tag=c2d34993aac6ea.
Call-ID: 34c34987-8b6fa786@<removed>.
CSeq: 126950830 NOTIFY.
Contact: <sip:2551@<removed>:9430>.
Expires: 3959.
Call-Info: <sip:mydomain.com>;appearance-index=*;appearance-state=idle.
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16633M.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, PRACK, NOTIFY, PUBLISH, SUBSCRIBE.
Supported: 100rel, timer, precondition, path, replaces.
Event: call-info.
Allow-Events: talk, presence, dialog, line-seize, call-info, sla,
include-session-description, presence.winfo, message-summary, refer.
Subscription-State: active;expires=3959.
Content-Length: 0.



On Sun, Feb 14, 2010 at 7:58 PM, Brian West <brian at freeswitch.org> wrote:
> Works fine here... is your box slow or something?
>
> /b
>
> On Feb 14, 2010, at 1:03 PM, Gabriel Kuri wrote:
>
>> I followed Brian's directions from one of the previous threads on
>> configuring the SPA-5xx series phones for Broadsoft SCA and set
>> manage-shared-appearance=true in the internal profile. SCA appears to
>> be working on outgoing calls between two phones, the line key starts
>> flashing red on the second phone when the first phone picks up the
>> receiver to make a call. However on incoming calls, both phones ring
>> (same extension), however when one of the phones picks up the line,
>> the second phone's line key doesn't flash red or show the first phone
>> on that incoming call. Any ideas? Does shared appearance only work on
>> outgoing phone calls?
>>
>> Thanks,
>> Gabe
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From mike at jerris.com  Sun Feb 14 22:19:32 2010
From: mike at jerris.com (Michael Jerris)
Date: Mon, 15 Feb 2010 01:19:32 -0500
Subject: [Freeswitch-users] Version 1.5
In-Reply-To: <ec2ce17e1002142142g1d71be24yf55132344b880514@mail.gmail.com>
References: <mailman.6373.1266207467.2495.freeswitch-users@lists.freeswitch.org>
	<041101caadf9$f36d77e0$da4867a0$@com>
	<ec2ce17e1002142142g1d71be24yf55132344b880514@mail.gmail.com>
Message-ID: <976016A5-C29F-4962-8779-F4DC257952F3@jerris.com>

I think 1.5 is probably quite a long ways off, 1.0.5 should be very soon now.

Mike

On Feb 15, 2010, at 12:42 AM, jay binks wrote:

> did you contribute to the dinner ??
> maybe thats why it hasnt been released yet ...
> 
> 
> On Mon, Feb 15, 2010 at 2:47 PM, Goran Donev <gorand at donevtechconsulting.com> wrote:
> When is version 1.5 of free switch going to be released. The last update on
> the website was on the week of Feb 8th. I still have not seen anything else.

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100215/0569e1b1/attachment-0002.html 

From jhonsonj at live.com  Sun Feb 14 22:18:52 2010
From: jhonsonj at live.com (John Jhonson)
Date: Mon, 15 Feb 2010 11:18:52 +0500
Subject: [Freeswitch-users] Looking Forward to know to create Partition in FS
Message-ID: <SNT126-W3516D0822131A1155DE0A5B84A0@phx.gbl>


Hi all,

I'm newbie in FS. I want to know how can I create/setup partitions in FS, like market well known VoIP Soft Switch products i.e. Nextone, VoIP Switch, Myra,etc can give partitions instead on buying whole product. Like I want to buy partition of 300 channels from any above mentioned products and add my carriers terminating gateways for whole sale scenario. 

Kindly advise/suggest me how can I do this setup in FS?



--

Regards,

John  


 		 	   		  
_________________________________________________________________
Hotmail: Free, trusted and rich email service.
https://signup.live.com/signup.aspx?id=60969
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100215/193390ba/attachment-0002.html 

From mike at jerris.com  Sun Feb 14 22:38:16 2010
From: mike at jerris.com (Michael Jerris)
Date: Mon, 15 Feb 2010 01:38:16 -0500
Subject: [Freeswitch-users] FreeSWITCH Debian packages - build issues
In-Reply-To: <20100214042033.GA19822@jdc.jasonjgw.net>
References: <20100213030812.GA19108@jdc.jasonjgw.net>
	<201002130957.11633.errotan@gmail.com>
	<20100213090907.GA29452@jdc.jasonjgw.net>
	<201002131109.35877.errotan@gmail.com>
	<20100213225320.GA4990@jdc.jasonjgw.net>
	<20100214042033.GA19822@jdc.jasonjgw.net>
Message-ID: <9CF8766D-2367-4275-ACCF-0F9C6ABC7A09@jerris.com>

Did anyone bother opening a bug on jira for this or are we going to just tag 1.0.5 without deb packages?

Mie

On Feb 13, 2010, at 11:20 PM, Jason White wrote:

> Just to close this thread for now, FreeSWITCH builds correctly if I remove the
> memcache module from the Debian package files.
> 
> Maybe when memcache in FreeSWITCH is updated to libmemcache 0.37 (which is in
> Debian unstable currently) the autoconf problem, which I'm not inclined to
> track down myself at the moement as I don't use memcache, will go away.
> 




From jason at jasonjgw.net  Sun Feb 14 22:46:16 2010
From: jason at jasonjgw.net (Jason White)
Date: Mon, 15 Feb 2010 17:46:16 +1100
Subject: [Freeswitch-users] FreeSWITCH Debian packages - build issues
In-Reply-To: <9CF8766D-2367-4275-ACCF-0F9C6ABC7A09@jerris.com>
References: <20100213030812.GA19108@jdc.jasonjgw.net>
	<201002130957.11633.errotan@gmail.com>
	<20100213090907.GA29452@jdc.jasonjgw.net>
	<201002131109.35877.errotan@gmail.com>
	<20100213225320.GA4990@jdc.jasonjgw.net>
	<20100214042033.GA19822@jdc.jasonjgw.net>
	<9CF8766D-2367-4275-ACCF-0F9C6ABC7A09@jerris.com>
Message-ID: <20100215064616.GA32700@jdc.jasonjgw.net>

Michael Jerris <mike at jerris.com> wrote:
> Did anyone bother opening a bug on jira for this or are we going to just tag
> 1.0.5 without deb packages?

Has anyone tried building these on Ubuntu 9.10 or Debian 5.0? I'm not in a
position to do so at the moment.




From jason at jasonjgw.net  Sun Feb 14 22:50:57 2010
From: jason at jasonjgw.net (Jason White)
Date: Mon, 15 Feb 2010 17:50:57 +1100
Subject: [Freeswitch-users] Version 1.5
In-Reply-To: <041101caadf9$f36d77e0$da4867a0$@com>
References: <mailman.6373.1266207467.2495.freeswitch-users@lists.freeswitch.org>
	<041101caadf9$f36d77e0$da4867a0$@com>
Message-ID: <20100215065057.GB32700@jdc.jasonjgw.net>

Goran Donev <gorand at donevtechconsulting.com> wrote:
> When is version 1.5 of free switch going to be released. The last update on
> the website was on the week of Feb 8th. I still have not seen anything else.

It will happen sooner if you help to find and fix the bugs, or if you
contribute funding to the development effort.




From mike at jerris.com  Sun Feb 14 22:52:01 2010
From: mike at jerris.com (Michael Jerris)
Date: Mon, 15 Feb 2010 01:52:01 -0500
Subject: [Freeswitch-users] Looking Forward to know to create Partition
	in FS
In-Reply-To: <SNT126-W3516D0822131A1155DE0A5B84A0@phx.gbl>
References: <SNT126-W3516D0822131A1155DE0A5B84A0@phx.gbl>
Message-ID: <8B6411F5-9ED2-4FF1-BDEF-5DA9CF49DDAB@jerris.com>

You can buy as many partitions from me as you like, I take paypal.

Mike

p.s., take a look on the wiki at sofia profiles and domains and mod limit, or just run multiple instances.



On Feb 15, 2010, at 1:18 AM, John Jhonson wrote:

> Hi all,
> 
> I'm newbie in FS. I want to know how can I create/setup partitions in FS, like market well known VoIP Soft Switch products i.e. Nextone, VoIP Switch, Myra,etc can give partitions instead on buying whole product. Like I want to buy partition of 300 channels from any above mentioned products and add my carriers terminating gateways for whole sale scenario. 
> 
> Kindly advise/suggest me how can I do this setup in FS?

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100215/5a8d480c/attachment-0002.html 

From dome at tel.co.th  Sun Feb 14 23:19:42 2010
From: dome at tel.co.th (Dome Charoenyost)
Date: Mon, 15 Feb 2010 14:19:42 +0700
Subject: [Freeswitch-users] Looking Forward to know to create Partition
	in FS
In-Reply-To: <SNT126-W3516D0822131A1155DE0A5B84A0@phx.gbl>
References: <SNT126-W3516D0822131A1155DE0A5B84A0@phx.gbl>
Message-ID: <8ccbff061002142319y4b2a6929m2ee6596654f1adc2@mail.gmail.com>

http://wiki.freeswitch.org/wiki/Multi-tenant

2010/2/15 John Jhonson <jhonsonj at live.com>:
> Hi all,
>
> I'm newbie in FS. I want to know how can I create/setup partitions in FS,
> like market well known VoIP Soft Switch products i.e. Nextone, VoIP Switch,
> Myra,etc can give partitions instead on buying whole product. Like I want to
> buy partition of 300 channels from any above mentioned products and add my
> carriers terminating gateways for whole sale scenario.
>
> Kindly advise/suggest me how can I do this setup in FS?
>
>
>
> --
>
> Regards,
>
> John
>
>
>
> ________________________________
> Hotmail: Free, trusted and rich email service. Get it now.
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From steve at justfone.com  Mon Feb 15 00:50:44 2010
From: steve at justfone.com (Steven Brown)
Date: Mon, 15 Feb 2010 08:50:44 +0000
Subject: [Freeswitch-users] Other party does not hear voice when
	connecting with X-Lite
Message-ID: <3e6d7b0c1002150050g3ccd23aas2e7f05f230bff4a7@mail.gmail.com>

I had the same problem with XLite / Freeswitch  a while back that I never
fully understood,  however the problem vanished when I disabled all codecs
on Xlite except G711 uLaw, as I say, no idea what was going on but this
might be worth trying.

Steve


Message: 1
Date: Sun, 14 Feb 2010 09:04:09 -0600
From: Anthony Minessale <anthony.minessale at gmail.com>
Subject: Re: [Freeswitch-users] Other party does not hear voice when
       connecting with X-Lite
To: freeswitch-users at lists.freeswitch.org
Message-ID:
       <191c3a031002140704g705bfc73rd8dd103f3d846062 at mail.gmail.com
>
Content-Type: text/plain; charset="iso-8859-1"

You need to describe this again its too confusing now.
List each device, freeswitch, the phones and which ip and combo of addrs it
uses with the topology clearly stated.
Your attempt to simplify your explanation is actually making it harder to
follow.
Also consider a debug/sip trace as well.

Include
sofia status profile default.

Then capture a test call after entering these commands.

console loglevel debug.
sofa profile internal siptrace on

On Feb 14, 2010 4:12 AM, "Vitalii Colosov" <vetali100 at gmail.com> wrote:

No, it is done on the different PCs...

Sorry, when I started the topic, I have described the problem how it is
visible from PC of my friend.
Then I tried to reproduce the same on my own PC, and you are right...I was
not able to hear anything as well, not only both party wasn't.
Also, from my PC I was NOT able to hear guitar on test number "9999".
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100215/37c7f74d/attachment-0002.html 

From kond at nstel.ru  Mon Feb 15 01:46:00 2010
From: kond at nstel.ru (Nikolay Kondratyev)
Date: Mon, 15 Feb 2010 12:46:00 +0300
Subject: [Freeswitch-users] h323 - sip call is not working
In-Reply-To: <65d96fc81002121246r48e867abp3c11f7f72a0ee906@mail.gmail.com>
Message-ID: <20100215094600.AED9312036@mail.nstel.ru>

Tihomir, 

I've just sent the trace to your gmail address.

Nikolay.

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Tihomir
Culjaga
Sent: Friday, February 12, 2010 11:46 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] h323 - sip call is not working

 

 

On Fri, Feb 12, 2010 at 3:29 PM, Brian West <brian at freeswitch.org> wrote:

This is a rather broad assumption.  I have seen 40ms, 60ms and even 80ms in
the wild.  It all depends on what you want to do.  It lowers overhead and
increases efficiency on the wire.

 

/b

 

On Feb 12, 2010, at 5:24 AM, Nikolay Kondratyev wrote:





By the way, I know that one can use different packetization times for the
same codec, but I've never heard, that somebody really uses 30 ms for
G711Alaw. Always 20ms.

 


everything above 60 ms is a nonsense ... and ugly :) It screws your voice
quality not even thinking VBD (voice band data) over that line :).

Anyhow, Nikolay, your problem is broken IPO h323 stack and the know avaya
"flexibility" when interoping with other vendor equipments. Here IPO is
unable to negotiate a different framing size than the default and sadly this
is the core of the problem.

Please, can you send me two tcpdump captures of calls between IPO and FS:

1.	a capture with fast start & h245tunneling=true
2.	a captire with fast start & h245tunelling=true + h245inSetup


I just want to be sure of something.

T.

 

 


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100215/5a399dac/attachment-0002.html 

From mbsip at gazeta.pl  Mon Feb 15 02:50:30 2010
From: mbsip at gazeta.pl (mbsip)
Date: Mon, 15 Feb 2010 11:50:30 +0100
Subject: [Freeswitch-users] mod_python fetching data from mysql for VM
In-Reply-To: <28f27f5d1002140652q656ab6deh40109b223d4f2f4a@mail.gmail.com>
References: <28f27f5d1002131203j132c6230sa260358611a16247@mail.gmail.com>
	<8E4B95E9-3914-4FEE-9C8C-9A5B4C5E2AB2@jerris.com>
	<28f27f5d1002131354k23a0eaf5jc51b05106c68d719@mail.gmail.com>
	<28f27f5d1002131359h164e4077x326f646ff2367ad9@mail.gmail.com>
	<ebbc2d551002131420v58cf95f2lfc9356f2377728b6@mail.gmail.com>
	<28f27f5d1002131438l58718bbat304407f49397c965@mail.gmail.com>
	<28f27f5d1002140652q656ab6deh40109b223d4f2f4a@mail.gmail.com>
Message-ID: <28f27f5d1002150250u7add2d0fq79b3a41803587f@mail.gmail.com>

Anybody could help with this?

Thx,
Maciej.


> Hi.
>
> Please correct me if my approach is okay.
> 1. in python.conf.xml
> ? ?<param name="xml-handler-script" value="testscript"/>
> ? ?<param name="xml-handler-bindings" value="directory"/>
> 2. in dialplan
> ? ?<extension name="public_did_11digits">
> ? ? ?<condition field="destination_number" expression="^(\d{11})$">
> ? ? ? ?<action application="answer"/>
> ? ? ? ?<action application="voicemail" data="default $${domain} $1"/>
> ? ? ?</condition>
> ? ?</extension>
> 3.testscript.py (as for now only static entries)
> def xml_fetch(params):
>
> ? ? ? ?xml = '''
> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
> <document type="freeswitch/xml">
> ?<section name="directory">
> ? ?<domain name="mydomainHERE">
> ? ? ?<params>
> ? ? ? ?<param name="dial-string"
> value="{presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
> ? ? ?</params>
> ? ? ?<groups>
> ? ? ? ?<group name="default">
> ? ? ? ? <users>
> ? ? ? ? ?<user id="12345678901">
> ? ? ? ? ? ?<params>
> ? ? ? ? ? ? ? <param name="vm-mailto" value="email at aaa.com"/
> ? ? ? ? ? ?</params>
> ? ? ? ? ?</user>
> ? ? ? ?</group>
> ? ? ? </users>
> ? ? ?</groups>
> ? ?</domain>
> ?</section>
> </document>
> '''
>
> ? ? ? ?return xml
>
>
> Unfortunately aforemetnioned configuration does not work at all and
> produce following errors:
> 2010-02-14 17:31:16.878878 [DEBUG] sofia.c:4110 Channel
> sofia/internal/100 at 10.10.10.10 entering state [completed][200]
> 2010-02-14 17:31:16.878878 [DEBUG] switch_core_session.c:638 Send
> signal sofia/internal/100 at 10.10.10.10 [BREAK]
> 2010-02-14 17:31:16.878878 [NOTICE] mod_dptools.c:715 Channel
> [sofia/internal/100 at 10.10.10.10] has been answered
> EXECUTE sofia/internal/100 at 10.10.10.10 voicemail(default mydomainHERE
> 12345678901)
> 2010-02-14 17:31:16.888821 [DEBUG] mod_voicemail.c:728 [default] rwlock
> 2010-02-14 17:31:16.888821 [NOTICE] mod_python.c:118 Invoking py module: obadamy
> 2010-02-14 17:31:16.888821 [DEBUG] mod_python.c:188 Call python script
> 2010-02-14 17:31:16.888821 [DEBUG] mod_python.c:191 Finished calling
> python script
> 2010-02-14 17:31:16.888821 [ERR] mod_python.c:200 Error calling python script
> 2010-02-14 17:31:16.888821 [WARNING] mod_voicemail.c:2923 Can't find
> user [12345678901 at mydomainHERE]
> 2010-02-14 17:31:16.888821 [DEBUG] switch_ivr_play_say.c:118 No
> language specified - Using [en]
> 2010-02-14 17:31:16.888821 [DEBUG] sofia.c:4110 Channel
> sofia/internal/100 at 10.10.10.10 entering state [ready][200]
> 2010-02-14 17:31:16.888821 [DEBUG] switch_ivr_play_say.c:273 Handle
> play-file:[voicemail/vm-goodbye.wav] (en:en)
> 2010-02-14 17:31:16.888821 [DEBUG] switch_ivr_play_say.c:1158 Codec
> Activated L16 at 8000hz 1 channels 20ms
>
> The same output with simplified python script.
> def xml_fetch(params):
>
> ? ? ? ?xml = '''
> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
> <document type="freeswitch/xml">
> ?<section name="directory">
> ? ?<domain name="62.29.162.90">
> ? ? ?<user id="48223891699">
> ? ? ? ?<params>
> ? ? ? ?</params>
> ? ? ?</user>
> ? ?</domain>
> ?</section>
> </document>
> '''
>
> ? ? ? ?return xml
>
>
>
> As for now I have no idea how to solve this, but still digging.
> Funny is that dialplan bindings work okay.
>
> Any help pls.
> Thx,
> Maciej
>
>
> 2010/2/13 mbsip <mbsip at gazeta.pl>:
>> Jo?o,
>>
>> Thanks for hint, because i don't know how the db fetched data could be
>> used with voicemail.
>> I am about to ready it carefully :P
>>
>> Thanks,
>> Maciej
>>
>> 2010/2/13 Jo?o Mesquita <jmesquita at freeswitch.org>:
>>> Maciej,
>>>
>>> Take a look at the xml_hooks we have on mod_python. Might do the trick for
>>> you.
>>>
>>> http://svn.freeswitch.org/svn/freeswitch/trunk/src/mod/languages/mod_python/python_example.py
>>>
>>> JM
>>>
>>>
>>> On Sat, Feb 13, 2010 at 6:59 PM, mbsip <mbsip at gazeta.pl> wrote:
>>>>
>>>> There is a lack of connection between fatched data and voicemail and I
>>>> dont know how to achieve it.
>>>>
>>>> Thx,
>>>> Maciej.
>>>>
>>>>
>>>> 2010/2/13 mbsip <mbsip at gazeta.pl>:
>>>> > Thx for prompt reply.
>>>> >
>>>> > The main task is to be able to use Mysql db in conjunction with VM
>>>> > (but not only voicemail_msgs, voicemail_prefs).
>>>> >
>>>> > Lets imagine sb is calling 1000 and wants to record the message.
>>>> > According to mod_voicemail settings message should be sent to some
>>>> > email address.
>>>> > But the information about user 1000 and his settings like email
>>>> > address, passwd, quota should be fetched from Mysql db, not from
>>>> > directory/default/1000.xml.
>>>> > That's why I am using <action application="python" data="test"/> in my
>>>> > dialplan to work with python script which in turn should do the magic.
>>>> > The script should be able to gather all necessery data about user 1000
>>>> > (like email address in shown example) and use them in VM.
>>>> >
>>>> > So the problem is how to modify the script to force voicemail app to
>>>> > use data from DB.
>>>> > Currently ?session.execute("voicemail", "default ${domain} " +
>>>> > the_dest) is still using .xml files.
>>>> >
>>>> > Thx,
>>>> > Maciej.
>>>> >
>>>> >
>>>> > 2010/2/13 Michael Jerris <mike at jerris.com>:
>>>> >> Can you describe what your trying to accomplish, I don't understand
>>>> >> what the goal is. ?What feature are you looking for that does not already
>>>> >> exist in mod_voiceamil.
>>>> >>
>>>> >> Mike
>>>> >>
>>>> >> On Feb 13, 2010, at 3:03 PM, mbsip wrote:
>>>> >>
>>>> >>> Hello,
>>>> >>>
>>>> >>> I am trying to use mod_python to fetch data from Mysql db (through
>>>> >>> ODBC) and execute voicemail application.
>>>> >>> Below a part of my script:
>>>> >>>
>>>> >>> db=MySQLdb.connect("localhost","root","","test")
>>>> >>> ? ? ? Cursor=db.cursor()
>>>> >>> ? ? ? sql = "SELECT email FROM VM WHERE called_num=%s" % the_dest
>>>> >>> ? ? ? Cursor.execute(sql)
>>>> >>> ? ? ? while (1):
>>>> >>> ? ? ? ? ? ? ? Results = Cursor.fetchone()
>>>> >>> ? ? ? ? ? ? ? if Results == None:
>>>> >>> ? ? ? ? ? ? ? ? ? ? ? break
>>>> >>> ? ? ? ? ? ? ? consoleLog("debug", "Found email " + Results[0] +"\n")
>>>> >>> ? ? ? ? ? ? ? the_recipient = Results[0]
>>>> >>> ? ? ? db.close()
>>>> >>>
>>>> >>> Now i have email address corresponding with called number. The
>>>> >>> question is how to use it for voicemail application?
>>>> >>> So it also means how to omit all /directory/default/....xml, where
>>>> >>> there are all VM parameters set and use fetched data.
>>>> >>>
>>>> >>> ? ? ? session.answer()
>>>> >>> ? ? ? session.execute("voicemail", "default ${domain} " + the_dest)
>>>> >>>
>>>> >>> Is this possible or should I start all VM app in python from the
>>>> >>> scratch?
>>>> >>
>>>> >>
>>>> >> _______________________________________________
>>>> >> FreeSWITCH-users mailing list
>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >>
>>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> >> http://www.freeswitch.org
>>>> >>
>>>> >
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>



From mbsip at gazeta.pl  Mon Feb 15 03:31:00 2010
From: mbsip at gazeta.pl (mbsip)
Date: Mon, 15 Feb 2010 12:31:00 +0100
Subject: [Freeswitch-users] mod_python fetching data from mysql for VM
In-Reply-To: <28f27f5d1002150250u7add2d0fq79b3a41803587f@mail.gmail.com>
References: <28f27f5d1002131203j132c6230sa260358611a16247@mail.gmail.com>
	<8E4B95E9-3914-4FEE-9C8C-9A5B4C5E2AB2@jerris.com>
	<28f27f5d1002131354k23a0eaf5jc51b05106c68d719@mail.gmail.com>
	<28f27f5d1002131359h164e4077x326f646ff2367ad9@mail.gmail.com>
	<ebbc2d551002131420v58cf95f2lfc9356f2377728b6@mail.gmail.com>
	<28f27f5d1002131438l58718bbat304407f49397c965@mail.gmail.com>
	<28f27f5d1002140652q656ab6deh40109b223d4f2f4a@mail.gmail.com>
	<28f27f5d1002150250u7add2d0fq79b3a41803587f@mail.gmail.com>
Message-ID: <28f27f5d1002150331k1001ac14o2e2bbbd27a96cbc7@mail.gmail.com>

Similiar script written in lua works okay.
Maybe there is sth wrong with providing dynamic directory information
via mod_python.

Thx,
Maciej

2010/2/15 mbsip <mbsip at gazeta.pl>:
> Anybody could help with this?
>
> Thx,
> Maciej.
>
>
>> Hi.
>>
>> Please correct me if my approach is okay.
>> 1. in python.conf.xml
>> ? ?<param name="xml-handler-script" value="testscript"/>
>> ? ?<param name="xml-handler-bindings" value="directory"/>
>> 2. in dialplan
>> ? ?<extension name="public_did_11digits">
>> ? ? ?<condition field="destination_number" expression="^(\d{11})$">
>> ? ? ? ?<action application="answer"/>
>> ? ? ? ?<action application="voicemail" data="default $${domain} $1"/>
>> ? ? ?</condition>
>> ? ?</extension>
>> 3.testscript.py (as for now only static entries)
>> def xml_fetch(params):
>>
>> ? ? ? ?xml = '''
>> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
>> <document type="freeswitch/xml">
>> ?<section name="directory">
>> ? ?<domain name="mydomainHERE">
>> ? ? ?<params>
>> ? ? ? ?<param name="dial-string"
>> value="{presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
>> ? ? ?</params>
>> ? ? ?<groups>
>> ? ? ? ?<group name="default">
>> ? ? ? ? <users>
>> ? ? ? ? ?<user id="12345678901">
>> ? ? ? ? ? ?<params>
>> ? ? ? ? ? ? ? <param name="vm-mailto" value="email at aaa.com"/
>> ? ? ? ? ? ?</params>
>> ? ? ? ? ?</user>
>> ? ? ? ?</group>
>> ? ? ? </users>
>> ? ? ?</groups>
>> ? ?</domain>
>> ?</section>
>> </document>
>> '''
>>
>> ? ? ? ?return xml
>>
>>
>> Unfortunately aforemetnioned configuration does not work at all and
>> produce following errors:
>> 2010-02-14 17:31:16.878878 [DEBUG] sofia.c:4110 Channel
>> sofia/internal/100 at 10.10.10.10 entering state [completed][200]
>> 2010-02-14 17:31:16.878878 [DEBUG] switch_core_session.c:638 Send
>> signal sofia/internal/100 at 10.10.10.10 [BREAK]
>> 2010-02-14 17:31:16.878878 [NOTICE] mod_dptools.c:715 Channel
>> [sofia/internal/100 at 10.10.10.10] has been answered
>> EXECUTE sofia/internal/100 at 10.10.10.10 voicemail(default mydomainHERE
>> 12345678901)
>> 2010-02-14 17:31:16.888821 [DEBUG] mod_voicemail.c:728 [default] rwlock
>> 2010-02-14 17:31:16.888821 [NOTICE] mod_python.c:118 Invoking py module: obadamy
>> 2010-02-14 17:31:16.888821 [DEBUG] mod_python.c:188 Call python script
>> 2010-02-14 17:31:16.888821 [DEBUG] mod_python.c:191 Finished calling
>> python script
>> 2010-02-14 17:31:16.888821 [ERR] mod_python.c:200 Error calling python script
>> 2010-02-14 17:31:16.888821 [WARNING] mod_voicemail.c:2923 Can't find
>> user [12345678901 at mydomainHERE]
>> 2010-02-14 17:31:16.888821 [DEBUG] switch_ivr_play_say.c:118 No
>> language specified - Using [en]
>> 2010-02-14 17:31:16.888821 [DEBUG] sofia.c:4110 Channel
>> sofia/internal/100 at 10.10.10.10 entering state [ready][200]
>> 2010-02-14 17:31:16.888821 [DEBUG] switch_ivr_play_say.c:273 Handle
>> play-file:[voicemail/vm-goodbye.wav] (en:en)
>> 2010-02-14 17:31:16.888821 [DEBUG] switch_ivr_play_say.c:1158 Codec
>> Activated L16 at 8000hz 1 channels 20ms
>>
>> The same output with simplified python script.
>> def xml_fetch(params):
>>
>> ? ? ? ?xml = '''
>> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
>> <document type="freeswitch/xml">
>> ?<section name="directory">
>> ? ?<domain name="62.29.162.90">
>> ? ? ?<user id="48223891699">
>> ? ? ? ?<params>
>> ? ? ? ?</params>
>> ? ? ?</user>
>> ? ?</domain>
>> ?</section>
>> </document>
>> '''
>>
>> ? ? ? ?return xml
>>
>>
>>
>> As for now I have no idea how to solve this, but still digging.
>> Funny is that dialplan bindings work okay.
>>
>> Any help pls.
>> Thx,
>> Maciej
>>
>>
>> 2010/2/13 mbsip <mbsip at gazeta.pl>:
>>> Jo?o,
>>>
>>> Thanks for hint, because i don't know how the db fetched data could be
>>> used with voicemail.
>>> I am about to ready it carefully :P
>>>
>>> Thanks,
>>> Maciej
>>>
>>> 2010/2/13 Jo?o Mesquita <jmesquita at freeswitch.org>:
>>>> Maciej,
>>>>
>>>> Take a look at the xml_hooks we have on mod_python. Might do the trick for
>>>> you.
>>>>
>>>> http://svn.freeswitch.org/svn/freeswitch/trunk/src/mod/languages/mod_python/python_example.py
>>>>
>>>> JM
>>>>
>>>>
>>>> On Sat, Feb 13, 2010 at 6:59 PM, mbsip <mbsip at gazeta.pl> wrote:
>>>>>
>>>>> There is a lack of connection between fatched data and voicemail and I
>>>>> dont know how to achieve it.
>>>>>
>>>>> Thx,
>>>>> Maciej.
>>>>>
>>>>>
>>>>> 2010/2/13 mbsip <mbsip at gazeta.pl>:
>>>>> > Thx for prompt reply.
>>>>> >
>>>>> > The main task is to be able to use Mysql db in conjunction with VM
>>>>> > (but not only voicemail_msgs, voicemail_prefs).
>>>>> >
>>>>> > Lets imagine sb is calling 1000 and wants to record the message.
>>>>> > According to mod_voicemail settings message should be sent to some
>>>>> > email address.
>>>>> > But the information about user 1000 and his settings like email
>>>>> > address, passwd, quota should be fetched from Mysql db, not from
>>>>> > directory/default/1000.xml.
>>>>> > That's why I am using <action application="python" data="test"/> in my
>>>>> > dialplan to work with python script which in turn should do the magic.
>>>>> > The script should be able to gather all necessery data about user 1000
>>>>> > (like email address in shown example) and use them in VM.
>>>>> >
>>>>> > So the problem is how to modify the script to force voicemail app to
>>>>> > use data from DB.
>>>>> > Currently ?session.execute("voicemail", "default ${domain} " +
>>>>> > the_dest) is still using .xml files.
>>>>> >
>>>>> > Thx,
>>>>> > Maciej.
>>>>> >
>>>>> >
>>>>> > 2010/2/13 Michael Jerris <mike at jerris.com>:
>>>>> >> Can you describe what your trying to accomplish, I don't understand
>>>>> >> what the goal is. ?What feature are you looking for that does not already
>>>>> >> exist in mod_voiceamil.
>>>>> >>
>>>>> >> Mike
>>>>> >>
>>>>> >> On Feb 13, 2010, at 3:03 PM, mbsip wrote:
>>>>> >>
>>>>> >>> Hello,
>>>>> >>>
>>>>> >>> I am trying to use mod_python to fetch data from Mysql db (through
>>>>> >>> ODBC) and execute voicemail application.
>>>>> >>> Below a part of my script:
>>>>> >>>
>>>>> >>> db=MySQLdb.connect("localhost","root","","test")
>>>>> >>> ? ? ? Cursor=db.cursor()
>>>>> >>> ? ? ? sql = "SELECT email FROM VM WHERE called_num=%s" % the_dest
>>>>> >>> ? ? ? Cursor.execute(sql)
>>>>> >>> ? ? ? while (1):
>>>>> >>> ? ? ? ? ? ? ? Results = Cursor.fetchone()
>>>>> >>> ? ? ? ? ? ? ? if Results == None:
>>>>> >>> ? ? ? ? ? ? ? ? ? ? ? break
>>>>> >>> ? ? ? ? ? ? ? consoleLog("debug", "Found email " + Results[0] +"\n")
>>>>> >>> ? ? ? ? ? ? ? the_recipient = Results[0]
>>>>> >>> ? ? ? db.close()
>>>>> >>>
>>>>> >>> Now i have email address corresponding with called number. The
>>>>> >>> question is how to use it for voicemail application?
>>>>> >>> So it also means how to omit all /directory/default/....xml, where
>>>>> >>> there are all VM parameters set and use fetched data.
>>>>> >>>
>>>>> >>> ? ? ? session.answer()
>>>>> >>> ? ? ? session.execute("voicemail", "default ${domain} " + the_dest)
>>>>> >>>
>>>>> >>> Is this possible or should I start all VM app in python from the
>>>>> >>> scratch?
>>>>> >>
>>>>> >>
>>>>> >> _______________________________________________
>>>>> >> FreeSWITCH-users mailing list
>>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> >>
>>>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> >> http://www.freeswitch.org
>>>>> >>
>>>>> >
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>
>



From paul at apcl.us  Mon Feb 15 04:46:10 2010
From: paul at apcl.us (Paul Levin)
Date: Mon, 15 Feb 2010 07:46:10 -0500
Subject: [Freeswitch-users] can vm settings go in
 conf\directory\default\default.xml?
Message-ID: <4B794212.9010406@apcl.us>

An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100215/a0454a49/attachment-0002.html 

From tculjaga at gmail.com  Mon Feb 15 05:49:43 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Mon, 15 Feb 2010 14:49:43 +0100
Subject: [Freeswitch-users] h323 - sip call is not working
In-Reply-To: <20100215094600.AED9312036@mail.nstel.ru>
References: <65d96fc81002121246r48e867abp3c11f7f72a0ee906@mail.gmail.com>
	<20100215094600.AED9312036@mail.nstel.ru>
Message-ID: <65d96fc81002150549j4cd129a8m8566e442e17de1f8@mail.gmail.com>

On Mon, Feb 15, 2010 at 10:46 AM, Nikolay Kondratyev <kond at nstel.ru> wrote:

>  Tihomir,
>
> I?ve just sent the trace to your gmail address?
>

didn't get anything...


> Nikolay.
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100215/fe2c30b5/attachment-0002.html 

From brian at freeswitch.org  Mon Feb 15 06:36:20 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 15 Feb 2010 08:36:20 -0600
Subject: [Freeswitch-users] Forbidden using UDP, works with TCP/TLS
In-Reply-To: <33c87fa31002142109l5fcabed3ydba21b865ed6cbbb@mail.gmail.com>
References: <33c87fa31002101824h21e3bf7lb13b68e8d4fd36a2@mail.gmail.com>
	<0A905834-850F-4D1F-AEC5-D3A87CE4141B@jerris.com>
	<33c87fa31002121629k74f26807i131841aa4252461b@mail.gmail.com>
	<33c87fa31002130343k7a01e9abt9c33360399f246f8@mail.gmail.com>
	<C37AB6F3-F712-41B6-9D6D-7A8AE3C6CABB@freeswitch.org>
	<33c87fa31002131336j7b05db7aof6ba6e3439aebc41@mail.gmail.com>
	<9D702362-BB5F-4C0F-9EE7-DABA3B926C36@freeswitch.org>
	<33c87fa31002141708s72cb7246h5372ecea40be615d@mail.gmail.com>
	<1888B1CA-D0A8-407C-9C44-B86E0F1925D6@freeswitch.org>
	<33c87fa31002142017v49d8e7c5nf572d086ec89282b@mail.gmail.com>
	<33c87fa31002142109l5fcabed3ydba21b865ed6cbbb@mail.gmail.com>
Message-ID: <D8F079A5-3EFA-41C1-8F37-780DBF5DDD44@freeswitch.org>

Interesting... I wonder if we have to echo back the qop token?

/b

On Feb 14, 2010, at 11:09 PM, Mark Campbell-Smith wrote:

> A little more testing.    I noticed that the Authorization field
> differs when TCP or UDP:
> 
> UDP (fails)
> Digest username=\"2010\", realm=\"mydns.dyndns.org\",
> nonce=\"e5f119c6-19e9-11df-bd09-773b7a755f78\",
> uri=\"sip:mydns.dyndns.org:5060\",
> response=\"e37be3e49c159d4f98e8bd04b36f2bd7\", algorithm=MD5,
> qop=\"1fffcc9f\"
> 
> TCP (works)
> Digest username=\"2010\",realm=\"mydns.dyndns.org\",nonce=\"5d9e75c2-19ea-11df-bd0b-773b7a755f78\",uri=\"sip:mydns.dyndns.org:5060\",algorithm=MD5,response=\"45ba55d3fbafcbf2bc2aa6418656ecc2\",qop=auth,nc=00000001,cnonce=\"3a650454\"
> 
> Is qop = 1fffcc9f valid in SIP?  Does a cnonce need to be included also?
> 
> FS sends qop=auth in the Unauthorized response.
> 
> Thanks

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100215/adb9e06f/attachment-0002.html 

From brian at freeswitch.org  Mon Feb 15 07:03:22 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 15 Feb 2010 09:03:22 -0600
Subject: [Freeswitch-users] Forbidden using UDP, works with TCP/TLS
In-Reply-To: <33c87fa31002142109l5fcabed3ydba21b865ed6cbbb@mail.gmail.com>
References: <33c87fa31002101824h21e3bf7lb13b68e8d4fd36a2@mail.gmail.com>
	<0A905834-850F-4D1F-AEC5-D3A87CE4141B@jerris.com>
	<33c87fa31002121629k74f26807i131841aa4252461b@mail.gmail.com>
	<33c87fa31002130343k7a01e9abt9c33360399f246f8@mail.gmail.com>
	<C37AB6F3-F712-41B6-9D6D-7A8AE3C6CABB@freeswitch.org>
	<33c87fa31002131336j7b05db7aof6ba6e3439aebc41@mail.gmail.com>
	<9D702362-BB5F-4C0F-9EE7-DABA3B926C36@freeswitch.org>
	<33c87fa31002141708s72cb7246h5372ecea40be615d@mail.gmail.com>
	<1888B1CA-D0A8-407C-9C44-B86E0F1925D6@freeswitch.org>
	<33c87fa31002142017v49d8e7c5nf572d086ec89282b@mail.gmail.com>
	<33c87fa31002142109l5fcabed3ydba21b865ed6cbbb@mail.gmail.com>
Message-ID: <147993F8-9C9E-4D96-9AFE-5EEE4E8F93F0@freeswitch.org>

Ok looks like the token is not used at all in digest auth.  This is the first time I have seen a device send back something other than auth or auth-int.

/b

On Feb 14, 2010, at 11:09 PM, Mark Campbell-Smith wrote:

> A little more testing.    I noticed that the Authorization field
> differs when TCP or UDP:
> 
> UDP (fails)
> Digest username=\"2010\", realm=\"mydns.dyndns.org\",
> nonce=\"e5f119c6-19e9-11df-bd09-773b7a755f78\",
> uri=\"sip:mydns.dyndns.org:5060\",
> response=\"e37be3e49c159d4f98e8bd04b36f2bd7\", algorithm=MD5,
> qop=\"1fffcc9f\"
> 
> TCP (works)
> Digest username=\"2010\",realm=\"mydns.dyndns.org\",nonce=\"5d9e75c2-19ea-11df-bd0b-773b7a755f78\",uri=\"sip:mydns.dyndns.org:5060\",algorithm=MD5,response=\"45ba55d3fbafcbf2bc2aa6418656ecc2\",qop=auth,nc=00000001,cnonce=\"3a650454\"
> 
> Is qop = 1fffcc9f valid in SIP?  Does a cnonce need to be included also?
> 
> FS sends qop=auth in the Unauthorized response.
> 
> Thanks

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100215/27cdb563/attachment-0002.html 

From kond at nstel.ru  Mon Feb 15 07:26:01 2010
From: kond at nstel.ru (Nikolay Kondratyev)
Date: Mon, 15 Feb 2010 18:26:01 +0300
Subject: [Freeswitch-users] h323 - sip call is not working
In-Reply-To: <65d96fc81002150549j4cd129a8m8566e442e17de1f8@mail.gmail.com>
Message-ID: <20100215152601.D569811FA6@mail.nstel.ru>

Sent again..

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Tihomir
Culjaga
Sent: Monday, February 15, 2010 4:50 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] h323 - sip call is not working

 

 

On Mon, Feb 15, 2010 at 10:46 AM, Nikolay Kondratyev <kond at nstel.ru> wrote:

Tihomir, 

I've just sent the trace to your gmail address.


didn't get anything...
 

Nikolay.

 

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100215/60631e74/attachment-0002.html 

From tculjaga at gmail.com  Mon Feb 15 07:39:38 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Mon, 15 Feb 2010 16:39:38 +0100
Subject: [Freeswitch-users] h323 - sip call is not working
In-Reply-To: <20100215152601.D569811FA6@mail.nstel.ru>
References: <65d96fc81002150549j4cd129a8m8566e442e17de1f8@mail.gmail.com>
	<20100215152601.D569811FA6@mail.nstel.ru>
Message-ID: <65d96fc81002150739l242407d2u2d5cb5081f96e7eb@mail.gmail.com>

It looks like audio level issue .. can you lower the gain on IPO ?

T.

On Mon, Feb 15, 2010 at 4:26 PM, Nikolay Kondratyev <kond at nstel.ru> wrote:

>  Sent again..
>
>
>   ------------------------------
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Tihomir
> Culjaga
> *Sent:* Monday, February 15, 2010 4:50 PM
>
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] h323 - sip call is not working
>
>
>
>
>
> On Mon, Feb 15, 2010 at 10:46 AM, Nikolay Kondratyev <kond at nstel.ru>
> wrote:
>
> Tihomir,
>
> I?ve just sent the trace to your gmail address?
>
>
> didn't get anything...
>
>
>  Nikolay.
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100215/af17b8d2/attachment-0002.html 

From vkozak at abisoft.spb.ru  Mon Feb 15 03:54:17 2010
From: vkozak at abisoft.spb.ru (Kozak Vladimir)
Date: Mon, 15 Feb 2010 14:54:17 +0300
Subject: [Freeswitch-users] CS_REPORTING state and CHANNEL_HANGUP event
Message-ID: <D0F4BBA35AD84876B35F99FCD443DAF9@abisoft.biz>

Hello all.
I have the following questions:
    1) Sometime channel whis CS_REPORTING state remain. What is channel whis CS_REPORTING state mean? for example: 2c90ac6b-7147-4aac-82fb-a23d6d5c4185,outbound,2010-02-10         11:53:02,1265820782,sofia/internal/sip:7100 at 76.74.160.163:57312,CS_REPORTING,FreeSWITCH,sipp,,7100,,,,default,,,,,,pst01.localdomain.com,, 
How can I kill this channel?  uuid_kill 2c90ac6b-7147-4aac-82fb-a23d6d5c4185 is not work.
    2) Sometime I get CHANNEL_HANGUP event but real call in eyeBeam is live for the present. When come CHANNEL_HANGUP event and response on api uuid_kill command? It's come when FS send hangup to phone or when phone confirm hangup?
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100215/924552f0/attachment-0002.html 

From kond at nstel.ru  Mon Feb 15 08:14:08 2010
From: kond at nstel.ru (Nikolay Kondratyev)
Date: Mon, 15 Feb 2010 19:14:08 +0300
Subject: [Freeswitch-users] h323 - sip call is not working
In-Reply-To: <65d96fc81002150739l242407d2u2d5cb5081f96e7eb@mail.gmail.com>
Message-ID: <20100215161410.BF0D911F5E@mail.nstel.ru>

Mmm. I doubt that it is an audio level problem.

Look at the picture that wireshark "voip calls -> player" shows for that
wheezy rbt.

Here it is:



 

This graph shows that FS really plays "stutter" tone instead of normal rbt.
Or am I mistaken?

 

And wireshark analysis of that rtp stream is strange.it shows 22 _seconds_
for that rtp stream..

Here is the text summary of wireshark rtp stream analysis of the stream from
FS to IPO:

Max delta = 50,65 ms at packet no. 1147 

Max jitter = 22556,56 ms. Mean jitter = 238,60 ms.

Max skew = 355392,29 ms.

Total RTP packets = 1538   (expected 1538)   Lost RTP packets = 0 (0,00%)
Sequence errors = 0 

Duration 36,14 s (513134 ms clock drift, corresponding to 121603 Hz
(+1420,03%)

 

I think the problem is on FS site.

 

Thanks and regards,

Nikolay.

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Tihomir
Culjaga
Sent: Monday, February 15, 2010 6:40 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] h323 - sip call is not working

 

It looks like audio level issue .. can you lower the gain on IPO ?

T.

On Mon, Feb 15, 2010 at 4:26 PM, Nikolay Kondratyev <kond at nstel.ru> wrote:

Sent again..

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Tihomir
Culjaga
Sent: Monday, February 15, 2010 4:50 PM


To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] h323 - sip call is not working

 

 

On Mon, Feb 15, 2010 at 10:46 AM, Nikolay Kondratyev <kond at nstel.ru> wrote:

Tihomir, 

I've just sent the trace to your gmail address.


didn't get anything...
 

Nikolay.

 

 


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100215/3ef129a9/attachment-0002.html 
-------------- next part --------------
A non-text attachment was scrubbed...
Name: not available
Type: image/jpeg
Size: 75817 bytes
Desc: not available
Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100215/3ef129a9/attachment-0002.jpe 

From michal.kalinowski at interia.pl  Mon Feb 15 08:24:07 2010
From: michal.kalinowski at interia.pl (michal kalinowski)
Date: Mon, 15 Feb 2010 17:24:07 +0100
Subject: [Freeswitch-users]  ivr from mysql
Message-ID: <7c74f5761002150824i41a2ed5did008f36cbb4499a4@mail.gmail.com>

Hi,

I need build ivr script/aplication which will take dynamically
configuration from mysql db (ivr menu, prompts, etc.).
My first idea is generate xml from python script. But it's not working properly.

Anybody has some idea or have this aplication already done ?

BR,
Micha?



From anthony.minessale at gmail.com  Mon Feb 15 08:32:11 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 15 Feb 2010 10:32:11 -0600
Subject: [Freeswitch-users] Broadsoft SCA w/ Cisco SPA-5xx series
In-Reply-To: <8b1c9cda1002142159h5efcc765q9a41baa1bf694c8c@mail.gmail.com>
References: <8b1c9cda1002141103te523643ob16db25e3bfa877f@mail.gmail.com>
	<0D67132C-69D8-43AD-B7E6-C7545B544322@freeswitch.org>
	<8b1c9cda1002142159h5efcc765q9a41baa1bf694c8c@mail.gmail.com>
Message-ID: <191c3a031002150832x7faa9a78xbc160cde63f09d09@mail.gmail.com>

it should be active not seized.
seized is when you take it off hook.

We need some more debugging to be sure.
Can we work in real time on it or can you get a more detailed log?

edit sofia.conf.xml and add the param to the "settings" section.

<param name="debug-sla" value="100"/>


then restart and enable sip trace and debug level

//do this for every profile involved in the call.
sofia profile <profilename> siptrace on

//also do this
console loglevel debug


if you can let us ssh, we can do all the for you if you can make the test
calls.




On Sun, Feb 14, 2010 at 11:59 PM, Gabriel Kuri <gkuri at ieee.org> wrote:

> BTW, here's a copy of the NOTIFY (event call-info) sent to the other
> phone after the first phone is answered, should this have a Call-Info
> line with an "appearance-state=seized" to turn on the light on the
> other phone?
>
>
> NOTIFY sip:2551@<removed>:54446 SIP/2.0.
> Via: SIP/2.0/UDP <removed>:9430;rport;branch=z9hG4bK71pN2cXgH851K.
> Max-Forwards: 70.
> From: <sip:2551 at mydomain.com <sip%3A2551 at mydomain.com>>;tag=XeB6ZrKDevpHp.
> To: <sip:2551 at mydomain.com <sip%3A2551 at mydomain.com>>;tag=c2d34993aac6ea.
> Call-ID: 34c34987-8b6fa786@<removed>.
> CSeq: 126950830 NOTIFY.
> Contact: <sip:2551@<removed>:9430>.
> Expires: 3959.
> Call-Info: <sip:mydomain.com>;appearance-index=*;appearance-state=idle.
> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16633M.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, PRACK, NOTIFY, PUBLISH, SUBSCRIBE.
> Supported: 100rel, timer, precondition, path, replaces.
> Event: call-info.
> Allow-Events: talk, presence, dialog, line-seize, call-info, sla,
> include-session-description, presence.winfo, message-summary, refer.
> Subscription-State: active;expires=3959.
> Content-Length: 0.
>
>
>
> On Sun, Feb 14, 2010 at 7:58 PM, Brian West <brian at freeswitch.org> wrote:
> > Works fine here... is your box slow or something?
> >
> > /b
> >
> > On Feb 14, 2010, at 1:03 PM, Gabriel Kuri wrote:
> >
> >> I followed Brian's directions from one of the previous threads on
> >> configuring the SPA-5xx series phones for Broadsoft SCA and set
> >> manage-shared-appearance=true in the internal profile. SCA appears to
> >> be working on outgoing calls between two phones, the line key starts
> >> flashing red on the second phone when the first phone picks up the
> >> receiver to make a call. However on incoming calls, both phones ring
> >> (same extension), however when one of the phones picks up the line,
> >> the second phone's line key doesn't flash red or show the first phone
> >> on that incoming call. Any ideas? Does shared appearance only work on
> >> outgoing phone calls?
> >>
> >> Thanks,
> >> Gabe
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100215/23d10814/attachment-0002.html 

From bottleman at icf.org.ru  Mon Feb 15 08:36:05 2010
From: bottleman at icf.org.ru (Georgiewskiy Yuriy)
Date: Mon, 15 Feb 2010 19:36:05 +0300 (MSK)
Subject: [Freeswitch-users] FreeSWITCH Debian packages - build issues
In-Reply-To: <20100214042033.GA19822@jdc.jasonjgw.net>
References: <20100213030812.GA19108@jdc.jasonjgw.net>
	<201002130957.11633.errotan@gmail.com>
	<20100213090907.GA29452@jdc.jasonjgw.net>
	<201002131109.35877.errotan@gmail.com>
	<20100213225320.GA4990@jdc.jasonjgw.net>
	<20100214042033.GA19822@jdc.jasonjgw.net>
Message-ID: <alpine.DEB.2.00.1002151931200.6101@srv.icf.org.ru>

On 2010-02-14 15:20 +1100, Jason White wrote freeswitch-users at lists.freeswi...:

JW>Just to close this thread for now, FreeSWITCH builds correctly if I remove the
JW>memcache module from the Debian package files.

after remove memcache it's not build mod_say_xx modules, and sound packages too.

JW>Maybe when memcache in FreeSWITCH is updated to libmemcache 0.37 (which is in
JW>Debian unstable currently) the autoconf problem, which I'm not inclined to
JW>track down myself at the moement as I don't use memcache, will go away.
JW>
JW>
JW>_______________________________________________
JW>FreeSWITCH-users mailing list
JW>FreeSWITCH-users at lists.freeswitch.org
JW>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
JW>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
JW>http://www.freeswitch.org
JW>

C ?????????                       With Best Regards
???????????? ????.                Georgiewskiy Yuriy
+7 4872 711666                    +7 4872 711666
???? +7 4872 711143               fax +7 4872 711143
???????? ??? "?? ?? ??????"       IT Service Ltd
http://nkoort.ru                  http://nkoort.ru
JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
YG129-RIPE                        YG129-RIPE

From msc at freeswitch.org  Mon Feb 15 09:59:41 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 15 Feb 2010 09:59:41 -0800
Subject: [Freeswitch-users] ASTPP For FreeSWITCH
Message-ID: <87f2f3b91002150959q6accece5od67b262fb38ec41e@mail.gmail.com>

Hey all,

Here's a quick story <http://www.freeswitch.org/node/239> about ASTPP and
FreeSWITCH. If you are using ASTPP with FreeSWITCH please let me know how it
works. I didn't see any information on our wiki about ASTPP. If ASTPP is
viable then we should document it as best we can.

Thanks!
-Michael
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100215/dc6c763a/attachment-0002.html 

From msc at freeswitch.org  Mon Feb 15 10:02:12 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 15 Feb 2010 10:02:12 -0800
Subject: [Freeswitch-users] ivr from mysql
In-Reply-To: <7c74f5761002150824i41a2ed5did008f36cbb4499a4@mail.gmail.com>
References: <7c74f5761002150824i41a2ed5did008f36cbb4499a4@mail.gmail.com>
Message-ID: <87f2f3b91002151002v6bbc3e7fgb67dd6d3e72a915@mail.gmail.com>

When you say "xml from python" what exactly do you mean? Are you trying to
use mod_xml_curl? If not you might want to check it out. The other choice is
to use Lua from the dialplan, although I have a gut feeling that
mod_xml_curl might be better for you.
-MC

2010/2/15 michal kalinowski <michal.kalinowski at interia.pl>

> Hi,
>
> I need build ivr script/aplication which will take dynamically
> configuration from mysql db (ivr menu, prompts, etc.).
> My first idea is generate xml from python script. But it's not working
> properly.
>
> Anybody has some idea or have this aplication already done ?
>
> BR,
> Micha?
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100215/651f0453/attachment-0002.html 

From gkuri at ieee.org  Mon Feb 15 10:48:13 2010
From: gkuri at ieee.org (Gabriel Kuri)
Date: Mon, 15 Feb 2010 10:48:13 -0800
Subject: [Freeswitch-users] Broadsoft SCA w/ Cisco SPA-5xx series
In-Reply-To: <191c3a031002150832x7faa9a78xbc160cde63f09d09@mail.gmail.com>
References: <8b1c9cda1002141103te523643ob16db25e3bfa877f@mail.gmail.com>
	<0D67132C-69D8-43AD-B7E6-C7545B544322@freeswitch.org>
	<8b1c9cda1002142159h5efcc765q9a41baa1bf694c8c@mail.gmail.com>
	<191c3a031002150832x7faa9a78xbc160cde63f09d09@mail.gmail.com>
Message-ID: <8b1c9cda1002151048o4b45c034qb22732a9366024ae@mail.gmail.com>

Here's a call from ext 2551 to 2552 (two phones). I noticed a bunch of
errors related to SQL UPDATE for presence ...

     http://pastebin.freeswitch.org/12152

Thanks,
Gabe


On Mon, Feb 15, 2010 at 8:32 AM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> it should be active not seized.
> seized is when you take it off hook.
>
> We need some more debugging to be sure.
> Can we work in real time on it or can you get a more detailed log?
>
> edit sofia.conf.xml and add the param to the "settings" section.
>
> <param name="debug-sla" value="100"/>
>
>
> then restart and enable sip trace and debug level
>
> //do this for every profile involved in the call.
> sofia profile <profilename> siptrace on
>
> //also do this
> console loglevel debug
>
>
> if you can let us ssh, we can do all the for you if you can make the test
> calls.
>
>
>
>
> On Sun, Feb 14, 2010 at 11:59 PM, Gabriel Kuri <gkuri at ieee.org> wrote:
>>
>> BTW, here's a copy of the NOTIFY (event call-info) sent to the other
>> phone after the first phone is answered, should this have a Call-Info
>> line with an "appearance-state=seized" to turn on the light on the
>> other phone?
>>
>>
>> NOTIFY sip:2551@<removed>:54446 SIP/2.0.
>> Via: SIP/2.0/UDP <removed>:9430;rport;branch=z9hG4bK71pN2cXgH851K.
>> Max-Forwards: 70.
>> From: <sip:2551 at mydomain.com>;tag=XeB6ZrKDevpHp.
>> To: <sip:2551 at mydomain.com>;tag=c2d34993aac6ea.
>> Call-ID: 34c34987-8b6fa786@<removed>.
>> CSeq: 126950830 NOTIFY.
>> Contact: <sip:2551@<removed>:9430>.
>> Expires: 3959.
>> Call-Info: <sip:mydomain.com>;appearance-index=*;appearance-state=idle.
>> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16633M.
>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>> REGISTER, REFER, PRACK, NOTIFY, PUBLISH, SUBSCRIBE.
>> Supported: 100rel, timer, precondition, path, replaces.
>> Event: call-info.
>> Allow-Events: talk, presence, dialog, line-seize, call-info, sla,
>> include-session-description, presence.winfo, message-summary, refer.
>> Subscription-State: active;expires=3959.
>> Content-Length: 0.
>>
>>
>>
>> On Sun, Feb 14, 2010 at 7:58 PM, Brian West <brian at freeswitch.org> wrote:
>> > Works fine here... is your box slow or something?
>> >
>> > /b
>> >
>> > On Feb 14, 2010, at 1:03 PM, Gabriel Kuri wrote:
>> >
>> >> I followed Brian's directions from one of the previous threads on
>> >> configuring the SPA-5xx series phones for Broadsoft SCA and set
>> >> manage-shared-appearance=true in the internal profile. SCA appears to
>> >> be working on outgoing calls between two phones, the line key starts
>> >> flashing red on the second phone when the first phone picks up the
>> >> receiver to make a call. However on incoming calls, both phones ring
>> >> (same extension), however when one of the phones picks up the line,
>> >> the second phone's line key doesn't flash red or show the first phone
>> >> on that incoming call. Any ideas? Does shared appearance only work on
>> >> outgoing phone calls?
>> >>
>> >> Thanks,
>> >> Gabe
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From darren at aleph-com.net  Mon Feb 15 10:53:45 2010
From: darren at aleph-com.net (Darren Wiebe)
Date: Mon, 15 Feb 2010 11:53:45 -0700
Subject: [Freeswitch-users] ASTPP For FreeSWITCH
In-Reply-To: <87f2f3b91002150959q6accece5od67b262fb38ec41e@mail.gmail.com>
References: <87f2f3b91002150959q6accece5od67b262fb38ec41e@mail.gmail.com>
Message-ID: <4B799839.2090008@aleph-com.net>

I will comment.  We've been using ASTPP for rating freeswitch cdrs for 
some time already.  It provides lcr from a database as well as sip user 
management.  It uses the mod_xml_curl and mod_xml_cdr modules for 
routing as well as realtime rating.  It also has an application that can 
listen to freeswitch and rate calls in realtime that way.  I patched a 
couple of bugs earlier this morning and I would not say that it's bug 
free but it's certainly in testing.

Darren Wiebe
darren at aleph-com.net


On 02/15/2010 10:59 AM, Michael Collins wrote:
> Hey all,
>
> Here's a quick story <http://www.freeswitch.org/node/239> about ASTPP 
> and FreeSWITCH. If you are using ASTPP with FreeSWITCH please let me 
> know how it works. I didn't see any information on our wiki about 
> ASTPP. If ASTPP is viable then we should document it as best we can.
>
> Thanks!
> -Michael
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>    

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100215/5c92b492/attachment-0002.html 

From anthony.minessale at gmail.com  Mon Feb 15 10:55:34 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 15 Feb 2010 12:55:34 -0600
Subject: [Freeswitch-users] Broadsoft SCA w/ Cisco SPA-5xx series
In-Reply-To: <8b1c9cda1002151048o4b45c034qb22732a9366024ae@mail.gmail.com>
References: <8b1c9cda1002141103te523643ob16db25e3bfa877f@mail.gmail.com>
	<0D67132C-69D8-43AD-B7E6-C7545B544322@freeswitch.org>
	<8b1c9cda1002142159h5efcc765q9a41baa1bf694c8c@mail.gmail.com>
	<191c3a031002150832x7faa9a78xbc160cde63f09d09@mail.gmail.com>
	<8b1c9cda1002151048o4b45c034qb22732a9366024ae@mail.gmail.com>
Message-ID: <191c3a031002151055v1938ab91p89232341f4087c8e@mail.gmail.com>

we log the sql stmts on err so they are red and easier to read.



On Mon, Feb 15, 2010 at 12:48 PM, Gabriel Kuri <gkuri at ieee.org> wrote:

> Here's a call from ext 2551 to 2552 (two phones). I noticed a bunch of
> errors related to SQL UPDATE for presence ...
>
>     http://pastebin.freeswitch.org/12152
>
> Thanks,
> Gabe
>
>
> On Mon, Feb 15, 2010 at 8:32 AM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
> > it should be active not seized.
> > seized is when you take it off hook.
> >
> > We need some more debugging to be sure.
> > Can we work in real time on it or can you get a more detailed log?
> >
> > edit sofia.conf.xml and add the param to the "settings" section.
> >
> > <param name="debug-sla" value="100"/>
> >
> >
> > then restart and enable sip trace and debug level
> >
> > //do this for every profile involved in the call.
> > sofia profile <profilename> siptrace on
> >
> > //also do this
> > console loglevel debug
> >
> >
> > if you can let us ssh, we can do all the for you if you can make the test
> > calls.
> >
> >
> >
> >
> > On Sun, Feb 14, 2010 at 11:59 PM, Gabriel Kuri <gkuri at ieee.org> wrote:
> >>
> >> BTW, here's a copy of the NOTIFY (event call-info) sent to the other
> >> phone after the first phone is answered, should this have a Call-Info
> >> line with an "appearance-state=seized" to turn on the light on the
> >> other phone?
> >>
> >>
> >> NOTIFY sip:2551@<removed>:54446 SIP/2.0.
> >> Via: SIP/2.0/UDP <removed>:9430;rport;branch=z9hG4bK71pN2cXgH851K.
> >> Max-Forwards: 70.
> >> From: <sip:2551 at mydomain.com <sip%3A2551 at mydomain.com>
> >;tag=XeB6ZrKDevpHp.
> >> To: <sip:2551 at mydomain.com <sip%3A2551 at mydomain.com>
> >;tag=c2d34993aac6ea.
> >> Call-ID: 34c34987-8b6fa786@<removed>.
> >> CSeq: 126950830 NOTIFY.
> >> Contact: <sip:2551@<removed>:9430>.
> >> Expires: 3959.
> >> Call-Info: <sip:mydomain.com>;appearance-index=*;appearance-state=idle.
> >> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16633M.
> >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> >> REGISTER, REFER, PRACK, NOTIFY, PUBLISH, SUBSCRIBE.
> >> Supported: 100rel, timer, precondition, path, replaces.
> >> Event: call-info.
> >> Allow-Events: talk, presence, dialog, line-seize, call-info, sla,
> >> include-session-description, presence.winfo, message-summary, refer.
> >> Subscription-State: active;expires=3959.
> >> Content-Length: 0.
> >>
> >>
> >>
> >> On Sun, Feb 14, 2010 at 7:58 PM, Brian West <brian at freeswitch.org>
> wrote:
> >> > Works fine here... is your box slow or something?
> >> >
> >> > /b
> >> >
> >> > On Feb 14, 2010, at 1:03 PM, Gabriel Kuri wrote:
> >> >
> >> >> I followed Brian's directions from one of the previous threads on
> >> >> configuring the SPA-5xx series phones for Broadsoft SCA and set
> >> >> manage-shared-appearance=true in the internal profile. SCA appears to
> >> >> be working on outgoing calls between two phones, the line key starts
> >> >> flashing red on the second phone when the first phone picks up the
> >> >> receiver to make a call. However on incoming calls, both phones ring
> >> >> (same extension), however when one of the phones picks up the line,
> >> >> the second phone's line key doesn't flash red or show the first phone
> >> >> on that incoming call. Any ideas? Does shared appearance only work on
> >> >> outgoing phone calls?
> >> >>
> >> >> Thanks,
> >> >> Gabe
> >> >
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> > Twitter: http://twitter.com/FreeSWITCH_wire
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> > IRC: irc.freenode.net #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> > iax:guest at conference.freeswitch.org/888
> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > pstn:+19193869900
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100215/04330486/attachment-0002.html 

From anthony.minessale at gmail.com  Mon Feb 15 11:04:24 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 15 Feb 2010 13:04:24 -0600
Subject: [Freeswitch-users] Broadsoft SCA w/ Cisco SPA-5xx series
In-Reply-To: <191c3a031002151055v1938ab91p89232341f4087c8e@mail.gmail.com>
References: <8b1c9cda1002141103te523643ob16db25e3bfa877f@mail.gmail.com>
	<0D67132C-69D8-43AD-B7E6-C7545B544322@freeswitch.org>
	<8b1c9cda1002142159h5efcc765q9a41baa1bf694c8c@mail.gmail.com>
	<191c3a031002150832x7faa9a78xbc160cde63f09d09@mail.gmail.com>
	<8b1c9cda1002151048o4b45c034qb22732a9366024ae@mail.gmail.com>
	<191c3a031002151055v1938ab91p89232341f4087c8e@mail.gmail.com>
Message-ID: <191c3a031002151104u7413b17dh84e68ab1cf703cb9@mail.gmail.com>

I don't see any notifies at all in this trace do the profiles in question
have:
manage-shared-appearance set to true?
and are you on latest trunk?


On Mon, Feb 15, 2010 at 12:55 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> we log the sql stmts on err so they are red and easier to read.
>
>
>
>
> On Mon, Feb 15, 2010 at 12:48 PM, Gabriel Kuri <gkuri at ieee.org> wrote:
>
>> Here's a call from ext 2551 to 2552 (two phones). I noticed a bunch of
>> errors related to SQL UPDATE for presence ...
>>
>>     http://pastebin.freeswitch.org/12152
>>
>> Thanks,
>> Gabe
>>
>>
>> On Mon, Feb 15, 2010 at 8:32 AM, Anthony Minessale
>> <anthony.minessale at gmail.com> wrote:
>> > it should be active not seized.
>> > seized is when you take it off hook.
>> >
>> > We need some more debugging to be sure.
>> > Can we work in real time on it or can you get a more detailed log?
>> >
>> > edit sofia.conf.xml and add the param to the "settings" section.
>> >
>> > <param name="debug-sla" value="100"/>
>> >
>> >
>> > then restart and enable sip trace and debug level
>> >
>> > //do this for every profile involved in the call.
>> > sofia profile <profilename> siptrace on
>> >
>> > //also do this
>> > console loglevel debug
>> >
>> >
>> > if you can let us ssh, we can do all the for you if you can make the
>> test
>> > calls.
>> >
>> >
>> >
>> >
>> > On Sun, Feb 14, 2010 at 11:59 PM, Gabriel Kuri <gkuri at ieee.org> wrote:
>> >>
>> >> BTW, here's a copy of the NOTIFY (event call-info) sent to the other
>> >> phone after the first phone is answered, should this have a Call-Info
>> >> line with an "appearance-state=seized" to turn on the light on the
>> >> other phone?
>> >>
>> >>
>> >> NOTIFY sip:2551@<removed>:54446 SIP/2.0.
>> >> Via: SIP/2.0/UDP <removed>:9430;rport;branch=z9hG4bK71pN2cXgH851K.
>> >> Max-Forwards: 70.
>> >> From: <sip:2551 at mydomain.com <sip%3A2551 at mydomain.com>
>> >;tag=XeB6ZrKDevpHp.
>> >> To: <sip:2551 at mydomain.com <sip%3A2551 at mydomain.com>
>> >;tag=c2d34993aac6ea.
>> >> Call-ID: 34c34987-8b6fa786@<removed>.
>> >> CSeq: 126950830 NOTIFY.
>> >> Contact: <sip:2551@<removed>:9430>.
>> >> Expires: 3959.
>> >> Call-Info: <sip:mydomain.com
>> >;appearance-index=*;appearance-state=idle.
>> >> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16633M.
>> >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>> >> REGISTER, REFER, PRACK, NOTIFY, PUBLISH, SUBSCRIBE.
>> >> Supported: 100rel, timer, precondition, path, replaces.
>> >> Event: call-info.
>> >> Allow-Events: talk, presence, dialog, line-seize, call-info, sla,
>> >> include-session-description, presence.winfo, message-summary, refer.
>> >> Subscription-State: active;expires=3959.
>> >> Content-Length: 0.
>> >>
>> >>
>> >>
>> >> On Sun, Feb 14, 2010 at 7:58 PM, Brian West <brian at freeswitch.org>
>> wrote:
>> >> > Works fine here... is your box slow or something?
>> >> >
>> >> > /b
>> >> >
>> >> > On Feb 14, 2010, at 1:03 PM, Gabriel Kuri wrote:
>> >> >
>> >> >> I followed Brian's directions from one of the previous threads on
>> >> >> configuring the SPA-5xx series phones for Broadsoft SCA and set
>> >> >> manage-shared-appearance=true in the internal profile. SCA appears
>> to
>> >> >> be working on outgoing calls between two phones, the line key starts
>> >> >> flashing red on the second phone when the first phone picks up the
>> >> >> receiver to make a call. However on incoming calls, both phones ring
>> >> >> (same extension), however when one of the phones picks up the line,
>> >> >> the second phone's line key doesn't flash red or show the first
>> phone
>> >> >> on that incoming call. Any ideas? Does shared appearance only work
>> on
>> >> >> outgoing phone calls?
>> >> >>
>> >> >> Thanks,
>> >> >> Gabe
>> >> >
>> >> >
>> >> > _______________________________________________
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> > http://www.freeswitch.org
>> >> >
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> >
>> > --
>> > Anthony Minessale II
>> >
>> > FreeSWITCH http://www.freeswitch.org/
>> > ClueCon http://www.cluecon.com/
>> > Twitter: http://twitter.com/FreeSWITCH_wire
>> >
>> > AIM: anthm
>> > MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> > IRC: irc.freenode.net #freeswitch
>> >
>> > FreeSWITCH Developer Conference
>> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> > iax:guest at conference.freeswitch.org/888
>> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> > pstn:+19193869900
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100215/1563828c/attachment-0002.html 

From gkuri at ieee.org  Mon Feb 15 11:47:49 2010
From: gkuri at ieee.org (Gabriel Kuri)
Date: Mon, 15 Feb 2010 11:47:49 -0800
Subject: [Freeswitch-users] Broadsoft SCA w/ Cisco SPA-5xx series
In-Reply-To: <191c3a031002151104u7413b17dh84e68ab1cf703cb9@mail.gmail.com>
References: <8b1c9cda1002141103te523643ob16db25e3bfa877f@mail.gmail.com>
	<0D67132C-69D8-43AD-B7E6-C7545B544322@freeswitch.org>
	<8b1c9cda1002142159h5efcc765q9a41baa1bf694c8c@mail.gmail.com>
	<191c3a031002150832x7faa9a78xbc160cde63f09d09@mail.gmail.com>
	<8b1c9cda1002151048o4b45c034qb22732a9366024ae@mail.gmail.com>
	<191c3a031002151055v1938ab91p89232341f4087c8e@mail.gmail.com>
	<191c3a031002151104u7413b17dh84e68ab1cf703cb9@mail.gmail.com>
Message-ID: <8b1c9cda1002151147l1e072d00y69c0022e3924c3b4@mail.gmail.com>

OK, I don't know what happened there, here's another one with the
NOTIFYs. I'm on trunk rev 16633 and I have
"managed-shared-appeareance=true" on the internal profile. I'm just
making calls between internal phones.

     http://pastebin.freeswitch.org/12153

Thanks,
Gabe

On Mon, Feb 15, 2010 at 11:04 AM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> I don't see any notifies at all in this trace do the profiles in question
> have:
> manage-shared-appearance set to true?
> and are you on latest trunk?
>
>
> On Mon, Feb 15, 2010 at 12:55 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>>
>> we log the sql stmts on err so they are red and easier to read.
>>
>>
>>
>> On Mon, Feb 15, 2010 at 12:48 PM, Gabriel Kuri <gkuri at ieee.org> wrote:
>>>
>>> Here's a call from ext 2551 to 2552 (two phones). I noticed a bunch of
>>> errors related to SQL UPDATE for presence ...
>>>
>>> ? ? http://pastebin.freeswitch.org/12152
>>>
>>> Thanks,
>>> Gabe
>>>
>>>
>>> On Mon, Feb 15, 2010 at 8:32 AM, Anthony Minessale
>>> <anthony.minessale at gmail.com> wrote:
>>> > it should be active not seized.
>>> > seized is when you take it off hook.
>>> >
>>> > We need some more debugging to be sure.
>>> > Can we work in real time on it or can you get a more detailed log?
>>> >
>>> > edit sofia.conf.xml and add the param to the "settings" section.
>>> >
>>> > <param name="debug-sla" value="100"/>
>>> >
>>> >
>>> > then restart and enable sip trace and debug level
>>> >
>>> > //do this for every profile involved in the call.
>>> > sofia profile <profilename> siptrace on
>>> >
>>> > //also do this
>>> > console loglevel debug
>>> >
>>> >
>>> > if you can let us ssh, we can do all the for you if you can make the
>>> > test
>>> > calls.
>>> >
>>> >
>>> >
>>> >
>>> > On Sun, Feb 14, 2010 at 11:59 PM, Gabriel Kuri <gkuri at ieee.org> wrote:
>>> >>
>>> >> BTW, here's a copy of the NOTIFY (event call-info) sent to the other
>>> >> phone after the first phone is answered, should this have a Call-Info
>>> >> line with an "appearance-state=seized" to turn on the light on the
>>> >> other phone?
>>> >>
>>> >>
>>> >> NOTIFY sip:2551@<removed>:54446 SIP/2.0.
>>> >> Via: SIP/2.0/UDP <removed>:9430;rport;branch=z9hG4bK71pN2cXgH851K.
>>> >> Max-Forwards: 70.
>>> >> From: <sip:2551 at mydomain.com>;tag=XeB6ZrKDevpHp.
>>> >> To: <sip:2551 at mydomain.com>;tag=c2d34993aac6ea.
>>> >> Call-ID: 34c34987-8b6fa786@<removed>.
>>> >> CSeq: 126950830 NOTIFY.
>>> >> Contact: <sip:2551@<removed>:9430>.
>>> >> Expires: 3959.
>>> >> Call-Info:
>>> >> <sip:mydomain.com>;appearance-index=*;appearance-state=idle.
>>> >> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16633M.
>>> >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>>> >> REGISTER, REFER, PRACK, NOTIFY, PUBLISH, SUBSCRIBE.
>>> >> Supported: 100rel, timer, precondition, path, replaces.
>>> >> Event: call-info.
>>> >> Allow-Events: talk, presence, dialog, line-seize, call-info, sla,
>>> >> include-session-description, presence.winfo, message-summary, refer.
>>> >> Subscription-State: active;expires=3959.
>>> >> Content-Length: 0.
>>> >>
>>> >>
>>> >>
>>> >> On Sun, Feb 14, 2010 at 7:58 PM, Brian West <brian at freeswitch.org>
>>> >> wrote:
>>> >> > Works fine here... is your box slow or something?
>>> >> >
>>> >> > /b
>>> >> >
>>> >> > On Feb 14, 2010, at 1:03 PM, Gabriel Kuri wrote:
>>> >> >
>>> >> >> I followed Brian's directions from one of the previous threads on
>>> >> >> configuring the SPA-5xx series phones for Broadsoft SCA and set
>>> >> >> manage-shared-appearance=true in the internal profile. SCA appears
>>> >> >> to
>>> >> >> be working on outgoing calls between two phones, the line key
>>> >> >> starts
>>> >> >> flashing red on the second phone when the first phone picks up the
>>> >> >> receiver to make a call. However on incoming calls, both phones
>>> >> >> ring
>>> >> >> (same extension), however when one of the phones picks up the line,
>>> >> >> the second phone's line key doesn't flash red or show the first
>>> >> >> phone
>>> >> >> on that incoming call. Any ideas? Does shared appearance only work
>>> >> >> on
>>> >> >> outgoing phone calls?
>>> >> >>
>>> >> >> Thanks,
>>> >> >> Gabe
>>> >> >
>>> >> >
>>> >> > _______________________________________________
>>> >> > FreeSWITCH-users mailing list
>>> >> > FreeSWITCH-users at lists.freeswitch.org
>>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> >
>>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> > http://www.freeswitch.org
>>> >> >
>>> >>
>>> >> _______________________________________________
>>> >> FreeSWITCH-users mailing list
>>> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>
>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> http://www.freeswitch.org
>>> >
>>> >
>>> >
>>> > --
>>> > Anthony Minessale II
>>> >
>>> > FreeSWITCH http://www.freeswitch.org/
>>> > ClueCon http://www.cluecon.com/
>>> > Twitter: http://twitter.com/FreeSWITCH_wire
>>> >
>>> > AIM: anthm
>>> > MSN:anthony_minessale at hotmail.com
>>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>> > IRC: irc.freenode.net #freeswitch
>>> >
>>> > FreeSWITCH Developer Conference
>>> > sip:888 at conference.freeswitch.org
>>> > iax:guest at conference.freeswitch.org/888
>>> > googletalk:conf+888 at conference.freeswitch.org
>>> > pstn:+19193869900
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >
>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>> >
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From tculjaga at gmail.com  Mon Feb 15 12:02:15 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Mon, 15 Feb 2010 21:02:15 +0100
Subject: [Freeswitch-users] h323 - sip call is not working
In-Reply-To: <20100215161410.BF0D911F5E@mail.nstel.ru>
References: <65d96fc81002150739l242407d2u2d5cb5081f96e7eb@mail.gmail.com>
	<20100215161410.BF0D911F5E@mail.nstel.ru>
Message-ID: <65d96fc81002151202r4fb18b6dma3cfb34fc98adc23@mail.gmail.com>

ok, than try this:


edit h323plus/src/h323caps.cxx, grep it for "H323AudioCapability(240, 30) //
240ms max, 30ms desired" ... it should be at line 2598....


replace 30 with 20, recompile (make && make install) make sure you use the
new compiled library and start FS.

let me know if you still have audio issues.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100215/1dc5a80e/attachment-0002.html 

From brian at freeswitch.org  Mon Feb 15 12:28:55 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 15 Feb 2010 14:28:55 -0600
Subject: [Freeswitch-users] Broadsoft SCA w/ Cisco SPA-5xx series
In-Reply-To: <8b1c9cda1002151147l1e072d00y69c0022e3924c3b4@mail.gmail.com>
References: <8b1c9cda1002141103te523643ob16db25e3bfa877f@mail.gmail.com>
	<0D67132C-69D8-43AD-B7E6-C7545B544322@freeswitch.org>
	<8b1c9cda1002142159h5efcc765q9a41baa1bf694c8c@mail.gmail.com>
	<191c3a031002150832x7faa9a78xbc160cde63f09d09@mail.gmail.com>
	<8b1c9cda1002151048o4b45c034qb22732a9366024ae@mail.gmail.com>
	<191c3a031002151055v1938ab91p89232341f4087c8e@mail.gmail.com>
	<191c3a031002151104u7413b17dh84e68ab1cf703cb9@mail.gmail.com>
	<8b1c9cda1002151147l1e072d00y69c0022e3924c3b4@mail.gmail.com>
Message-ID: <7C6C0110-C6AE-45CA-A1B4-413E39223286@freeswitch.org>

Can you outline the topology a bit better...  I sense you have FS behind nat... devices behind nat and ext-rtp-ip and ext-sip-ip set.

Can you confirm this?

/b

On Feb 15, 2010, at 1:47 PM, Gabriel Kuri wrote:

> OK, I don't know what happened there, here's another one with the
> NOTIFYs. I'm on trunk rev 16633 and I have
> "managed-shared-appeareance=true" on the internal profile. I'm just
> making calls between internal phones.
> 
>     http://pastebin.freeswitch.org/12153
> 
> Thanks,
> Gabe

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100215/60cf05e1/attachment-0002.html 

From peder at networkoblivion.com  Mon Feb 15 12:34:12 2010
From: peder at networkoblivion.com (Peder)
Date: Mon, 15 Feb 2010 14:34:12 -0600
Subject: [Freeswitch-users] Broadsoft SCA w/ Cisco SPA-5xx series
In-Reply-To: <8b1c9cda1002151147l1e072d00y69c0022e3924c3b4@mail.gmail.com>
References: <8b1c9cda1002141103te523643ob16db25e3bfa877f@mail.gmail.com>	<0D67132C-69D8-43AD-B7E6-C7545B544322@freeswitch.org>	<8b1c9cda1002142159h5efcc765q9a41baa1bf694c8c@mail.gmail.com>	<191c3a031002150832x7faa9a78xbc160cde63f09d09@mail.gmail.com>	<8b1c9cda1002151048o4b45c034qb22732a9366024ae@mail.gmail.com>	<191c3a031002151055v1938ab91p89232341f4087c8e@mail.gmail.com>	<191c3a031002151104u7413b17dh84e68ab1cf703cb9@mail.gmail.com>
	<8b1c9cda1002151147l1e072d00y69c0022e3924c3b4@mail.gmail.com>
Message-ID: <0dd701caae7e$3c108b70$b431a250$@com>

Is this a typo "managed-shared-appeareance=true" or is there an extra e in
appearance in your config?

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Gabriel
Kuri
Sent: Monday, February 15, 2010 1:48 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Broadsoft SCA w/ Cisco SPA-5xx series

OK, I don't know what happened there, here's another one with the
NOTIFYs. I'm on trunk rev 16633 and I have
"managed-shared-appeareance=true" on the internal profile. I'm just
making calls between internal phones.

     http://pastebin.freeswitch.org/12153

Thanks,
Gabe

On Mon, Feb 15, 2010 at 11:04 AM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> I don't see any notifies at all in this trace do the profiles in question
> have:
> manage-shared-appearance set to true?
> and are you on latest trunk?
>
>
> On Mon, Feb 15, 2010 at 12:55 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>>
>> we log the sql stmts on err so they are red and easier to read.
>>
>>
>>
>> On Mon, Feb 15, 2010 at 12:48 PM, Gabriel Kuri <gkuri at ieee.org> wrote:
>>>
>>> Here's a call from ext 2551 to 2552 (two phones). I noticed a bunch of
>>> errors related to SQL UPDATE for presence ...
>>>
>>> ? ? http://pastebin.freeswitch.org/12152
>>>
>>> Thanks,
>>> Gabe
>>>
>>>
>>> On Mon, Feb 15, 2010 at 8:32 AM, Anthony Minessale
>>> <anthony.minessale at gmail.com> wrote:
>>> > it should be active not seized.
>>> > seized is when you take it off hook.
>>> >
>>> > We need some more debugging to be sure.
>>> > Can we work in real time on it or can you get a more detailed log?
>>> >
>>> > edit sofia.conf.xml and add the param to the "settings" section.
>>> >
>>> > <param name="debug-sla" value="100"/>
>>> >
>>> >
>>> > then restart and enable sip trace and debug level
>>> >
>>> > //do this for every profile involved in the call.
>>> > sofia profile <profilename> siptrace on
>>> >
>>> > //also do this
>>> > console loglevel debug
>>> >
>>> >
>>> > if you can let us ssh, we can do all the for you if you can make the
>>> > test
>>> > calls.
>>> >
>>> >
>>> >
>>> >
>>> > On Sun, Feb 14, 2010 at 11:59 PM, Gabriel Kuri <gkuri at ieee.org> wrote:
>>> >>
>>> >> BTW, here's a copy of the NOTIFY (event call-info) sent to the other
>>> >> phone after the first phone is answered, should this have a Call-Info
>>> >> line with an "appearance-state=seized" to turn on the light on the
>>> >> other phone?
>>> >>
>>> >>
>>> >> NOTIFY sip:2551@<removed>:54446 SIP/2.0.
>>> >> Via: SIP/2.0/UDP <removed>:9430;rport;branch=z9hG4bK71pN2cXgH851K.
>>> >> Max-Forwards: 70.
>>> >> From: <sip:2551 at mydomain.com>;tag=XeB6ZrKDevpHp.
>>> >> To: <sip:2551 at mydomain.com>;tag=c2d34993aac6ea.
>>> >> Call-ID: 34c34987-8b6fa786@<removed>.
>>> >> CSeq: 126950830 NOTIFY.
>>> >> Contact: <sip:2551@<removed>:9430>.
>>> >> Expires: 3959.
>>> >> Call-Info:
>>> >> <sip:mydomain.com>;appearance-index=*;appearance-state=idle.
>>> >> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16633M.
>>> >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>>> >> REGISTER, REFER, PRACK, NOTIFY, PUBLISH, SUBSCRIBE.
>>> >> Supported: 100rel, timer, precondition, path, replaces.
>>> >> Event: call-info.
>>> >> Allow-Events: talk, presence, dialog, line-seize, call-info, sla,
>>> >> include-session-description, presence.winfo, message-summary, refer.
>>> >> Subscription-State: active;expires=3959.
>>> >> Content-Length: 0.
>>> >>
>>> >>
>>> >>
>>> >> On Sun, Feb 14, 2010 at 7:58 PM, Brian West <brian at freeswitch.org>
>>> >> wrote:
>>> >> > Works fine here... is your box slow or something?
>>> >> >
>>> >> > /b
>>> >> >
>>> >> > On Feb 14, 2010, at 1:03 PM, Gabriel Kuri wrote:
>>> >> >
>>> >> >> I followed Brian's directions from one of the previous threads on
>>> >> >> configuring the SPA-5xx series phones for Broadsoft SCA and set
>>> >> >> manage-shared-appearance=true in the internal profile. SCA appears
>>> >> >> to
>>> >> >> be working on outgoing calls between two phones, the line key
>>> >> >> starts
>>> >> >> flashing red on the second phone when the first phone picks up the
>>> >> >> receiver to make a call. However on incoming calls, both phones
>>> >> >> ring
>>> >> >> (same extension), however when one of the phones picks up the
line,
>>> >> >> the second phone's line key doesn't flash red or show the first
>>> >> >> phone
>>> >> >> on that incoming call. Any ideas? Does shared appearance only work
>>> >> >> on
>>> >> >> outgoing phone calls?
>>> >> >>
>>> >> >> Thanks,
>>> >> >> Gabe
>>> >> >
>>> >> >
>>> >> > _______________________________________________
>>> >> > FreeSWITCH-users mailing list
>>> >> > FreeSWITCH-users at lists.freeswitch.org
>>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> >
>>> >> >
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> > http://www.freeswitch.org
>>> >> >
>>> >>
>>> >> _______________________________________________
>>> >> FreeSWITCH-users mailing list
>>> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>
>>> >>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> http://www.freeswitch.org
>>> >
>>> >
>>> >
>>> > --
>>> > Anthony Minessale II
>>> >
>>> > FreeSWITCH http://www.freeswitch.org/
>>> > ClueCon http://www.cluecon.com/
>>> > Twitter: http://twitter.com/FreeSWITCH_wire
>>> >
>>> > AIM: anthm
>>> > MSN:anthony_minessale at hotmail.com
>>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>> > IRC: irc.freenode.net #freeswitch
>>> >
>>> > FreeSWITCH Developer Conference
>>> > sip:888 at conference.freeswitch.org
>>> > iax:guest at conference.freeswitch.org/888
>>> > googletalk:conf+888 at conference.freeswitch.org
>>> > pstn:+19193869900
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >
>>> >
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>> >
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




From msc at freeswitch.org  Mon Feb 15 12:45:45 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 15 Feb 2010 12:45:45 -0800
Subject: [Freeswitch-users] Broadsoft SCA w/ Cisco SPA-5xx series
In-Reply-To: <0dd701caae7e$3c108b70$b431a250$@com>
References: <8b1c9cda1002141103te523643ob16db25e3bfa877f@mail.gmail.com>
	<0D67132C-69D8-43AD-B7E6-C7545B544322@freeswitch.org>
	<8b1c9cda1002142159h5efcc765q9a41baa1bf694c8c@mail.gmail.com>
	<191c3a031002150832x7faa9a78xbc160cde63f09d09@mail.gmail.com>
	<8b1c9cda1002151048o4b45c034qb22732a9366024ae@mail.gmail.com>
	<191c3a031002151055v1938ab91p89232341f4087c8e@mail.gmail.com>
	<191c3a031002151104u7413b17dh84e68ab1cf703cb9@mail.gmail.com>
	<8b1c9cda1002151147l1e072d00y69c0022e3924c3b4@mail.gmail.com>
	<0dd701caae7e$3c108b70$b431a250$@com>
Message-ID: <87f2f3b91002151245s3987beecy41853d95d522ceb0@mail.gmail.com>

Yes, it is a typo from the original article I wrote. Also "appearance" is
misspelled. The correct line is:
<param name="manage-shared-appearance" value="true"/>

BTW, in internal.xml this line is commented out so if you need to copy &
paste just look in there...
-MC

On Mon, Feb 15, 2010 at 12:34 PM, Peder <peder at networkoblivion.com> wrote:

> Is this a typo "managed-shared-appeareance=true" or is there an extra e in
> appearance in your config?
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Gabriel
> Kuri
> Sent: Monday, February 15, 2010 1:48 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Broadsoft SCA w/ Cisco SPA-5xx series
>
> OK, I don't know what happened there, here's another one with the
> NOTIFYs. I'm on trunk rev 16633 and I have
> "managed-shared-appeareance=true" on the internal profile. I'm just
> making calls between internal phones.
>
>     http://pastebin.freeswitch.org/12153
>
> Thanks,
> Gabe
>
> On Mon, Feb 15, 2010 at 11:04 AM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
> > I don't see any notifies at all in this trace do the profiles in question
> > have:
> > manage-shared-appearance set to true?
> > and are you on latest trunk?
> >
> >
> > On Mon, Feb 15, 2010 at 12:55 PM, Anthony Minessale
> > <anthony.minessale at gmail.com> wrote:
> >>
> >> we log the sql stmts on err so they are red and easier to read.
> >>
> >>
> >>
> >> On Mon, Feb 15, 2010 at 12:48 PM, Gabriel Kuri <gkuri at ieee.org> wrote:
> >>>
> >>> Here's a call from ext 2551 to 2552 (two phones). I noticed a bunch of
> >>> errors related to SQL UPDATE for presence ...
> >>>
> >>>     http://pastebin.freeswitch.org/12152
> >>>
> >>> Thanks,
> >>> Gabe
> >>>
> >>>
> >>> On Mon, Feb 15, 2010 at 8:32 AM, Anthony Minessale
> >>> <anthony.minessale at gmail.com> wrote:
> >>> > it should be active not seized.
> >>> > seized is when you take it off hook.
> >>> >
> >>> > We need some more debugging to be sure.
> >>> > Can we work in real time on it or can you get a more detailed log?
> >>> >
> >>> > edit sofia.conf.xml and add the param to the "settings" section.
> >>> >
> >>> > <param name="debug-sla" value="100"/>
> >>> >
> >>> >
> >>> > then restart and enable sip trace and debug level
> >>> >
> >>> > //do this for every profile involved in the call.
> >>> > sofia profile <profilename> siptrace on
> >>> >
> >>> > //also do this
> >>> > console loglevel debug
> >>> >
> >>> >
> >>> > if you can let us ssh, we can do all the for you if you can make the
> >>> > test
> >>> > calls.
> >>> >
> >>> >
> >>> >
> >>> >
> >>> > On Sun, Feb 14, 2010 at 11:59 PM, Gabriel Kuri <gkuri at ieee.org>
> wrote:
> >>> >>
> >>> >> BTW, here's a copy of the NOTIFY (event call-info) sent to the other
> >>> >> phone after the first phone is answered, should this have a
> Call-Info
> >>> >> line with an "appearance-state=seized" to turn on the light on the
> >>> >> other phone?
> >>> >>
> >>> >>
> >>> >> NOTIFY sip:2551@<removed>:54446 SIP/2.0.
> >>> >> Via: SIP/2.0/UDP <removed>:9430;rport;branch=z9hG4bK71pN2cXgH851K.
> >>> >> Max-Forwards: 70.
> >>> >> From: <sip:2551 at mydomain.com <sip%3A2551 at mydomain.com>
> >;tag=XeB6ZrKDevpHp.
> >>> >> To: <sip:2551 at mydomain.com <sip%3A2551 at mydomain.com>
> >;tag=c2d34993aac6ea.
> >>> >> Call-ID: 34c34987-8b6fa786@<removed>.
> >>> >> CSeq: 126950830 NOTIFY.
> >>> >> Contact: <sip:2551@<removed>:9430>.
> >>> >> Expires: 3959.
> >>> >> Call-Info:
> >>> >> <sip:mydomain.com>;appearance-index=*;appearance-state=idle.
> >>> >> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16633M.
> >>> >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> >>> >> REGISTER, REFER, PRACK, NOTIFY, PUBLISH, SUBSCRIBE.
> >>> >> Supported: 100rel, timer, precondition, path, replaces.
> >>> >> Event: call-info.
> >>> >> Allow-Events: talk, presence, dialog, line-seize, call-info, sla,
> >>> >> include-session-description, presence.winfo, message-summary, refer.
> >>> >> Subscription-State: active;expires=3959.
> >>> >> Content-Length: 0.
> >>> >>
> >>> >>
> >>> >>
> >>> >> On Sun, Feb 14, 2010 at 7:58 PM, Brian West <brian at freeswitch.org>
> >>> >> wrote:
> >>> >> > Works fine here... is your box slow or something?
> >>> >> >
> >>> >> > /b
> >>> >> >
> >>> >> > On Feb 14, 2010, at 1:03 PM, Gabriel Kuri wrote:
> >>> >> >
> >>> >> >> I followed Brian's directions from one of the previous threads on
> >>> >> >> configuring the SPA-5xx series phones for Broadsoft SCA and set
> >>> >> >> manage-shared-appearance=true in the internal profile. SCA
> appears
> >>> >> >> to
> >>> >> >> be working on outgoing calls between two phones, the line key
> >>> >> >> starts
> >>> >> >> flashing red on the second phone when the first phone picks up
> the
> >>> >> >> receiver to make a call. However on incoming calls, both phones
> >>> >> >> ring
> >>> >> >> (same extension), however when one of the phones picks up the
> line,
> >>> >> >> the second phone's line key doesn't flash red or show the first
> >>> >> >> phone
> >>> >> >> on that incoming call. Any ideas? Does shared appearance only
> work
> >>> >> >> on
> >>> >> >> outgoing phone calls?
> >>> >> >>
> >>> >> >> Thanks,
> >>> >> >> Gabe
> >>> >> >
> >>> >> >
> >>> >> > _______________________________________________
> >>> >> > FreeSWITCH-users mailing list
> >>> >> > FreeSWITCH-users at lists.freeswitch.org
> >>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> >> >
> >>> >> >
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> >> > http://www.freeswitch.org
> >>> >> >
> >>> >>
> >>> >> _______________________________________________
> >>> >> FreeSWITCH-users mailing list
> >>> >> FreeSWITCH-users at lists.freeswitch.org
> >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> >>
> >>> >>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> >> http://www.freeswitch.org
> >>> >
> >>> >
> >>> >
> >>> > --
> >>> > Anthony Minessale II
> >>> >
> >>> > FreeSWITCH http://www.freeswitch.org/
> >>> > ClueCon http://www.cluecon.com/
> >>> > Twitter: http://twitter.com/FreeSWITCH_wire
> >>> >
> >>> > AIM: anthm
> >>> > MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
> >>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >>> > IRC: irc.freenode.net #freeswitch
> >>> >
> >>> > FreeSWITCH Developer Conference
> >>> > sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> >>> > iax:guest at conference.freeswitch.org/888
> >>> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >>> > pstn:+19193869900
> >>> >
> >>> > _______________________________________________
> >>> > FreeSWITCH-users mailing list
> >>> > FreeSWITCH-users at lists.freeswitch.org
> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> >
> >>> >
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> > http://www.freeswitch.org
> >>> >
> >>> >
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>
> >>
> >>
> >> --
> >> Anthony Minessale II
> >>
> >> FreeSWITCH http://www.freeswitch.org/
> >> ClueCon http://www.cluecon.com/
> >> Twitter: http://twitter.com/FreeSWITCH_wire
> >>
> >> AIM: anthm
> >> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >> IRC: irc.freenode.net #freeswitch
> >>
> >> FreeSWITCH Developer Conference
> >> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> >> iax:guest at conference.freeswitch.org/888
> >> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >> pstn:+19193869900
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> > Twitter: http://twitter.com/FreeSWITCH_wire
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> > IRC: irc.freenode.net #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> > iax:guest at conference.freeswitch.org/888
> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > pstn:+19193869900
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100215/391884c7/attachment-0002.html 

From jerry.richards at teotech.com  Mon Feb 15 12:49:38 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Mon, 15 Feb 2010 12:49:38 -0800
Subject: [Freeswitch-users] external_sip_address and external_rtp_address
	Question
Message-ID: <CDD4A2D8AA494ED8BF64401C5C953F72@greyhawk.tonecommander.com>

I only see one example for setting of external_sip_address and
external_rtp_address tags.  Is it true they are used to specify a SIP
provider outside of a LAN (i.e. through a router)?  If so, then can these
tags be set for each sip_profile?  So, if I have multiple external SIP
providers that are accessed through NAT, they would each have their own
external_sip_address and external_rtp_address?

Best Regards,
Jerry





From gkuri at ieee.org  Mon Feb 15 12:52:58 2010
From: gkuri at ieee.org (Gabriel Kuri)
Date: Mon, 15 Feb 2010 12:52:58 -0800
Subject: [Freeswitch-users] Broadsoft SCA w/ Cisco SPA-5xx series
In-Reply-To: <7C6C0110-C6AE-45CA-A1B4-413E39223286@freeswitch.org>
References: <8b1c9cda1002141103te523643ob16db25e3bfa877f@mail.gmail.com>
	<0D67132C-69D8-43AD-B7E6-C7545B544322@freeswitch.org>
	<8b1c9cda1002142159h5efcc765q9a41baa1bf694c8c@mail.gmail.com>
	<191c3a031002150832x7faa9a78xbc160cde63f09d09@mail.gmail.com>
	<8b1c9cda1002151048o4b45c034qb22732a9366024ae@mail.gmail.com>
	<191c3a031002151055v1938ab91p89232341f4087c8e@mail.gmail.com>
	<191c3a031002151104u7413b17dh84e68ab1cf703cb9@mail.gmail.com>
	<8b1c9cda1002151147l1e072d00y69c0022e3924c3b4@mail.gmail.com>
	<7C6C0110-C6AE-45CA-A1B4-413E39223286@freeswitch.org>
Message-ID: <8b1c9cda1002151252jbc500bax3f137a882171485f@mail.gmail.com>

Yes, that is correct. FS is behind a NAT and the phones behind another NAT.

I have ext-rtp-ip and ext-sip-ip set to the public IP address. Phones
calls and everything else seem to be working.

Thanks,
Gabe

On Mon, Feb 15, 2010 at 12:28 PM, Brian West <brian at freeswitch.org> wrote:
> Can you outline the topology a bit better... ?I sense you have FS behind
> nat... devices behind nat and ext-rtp-ip and ext-sip-ip set.
> Can you confirm this?
> /b
> On Feb 15, 2010, at 1:47 PM, Gabriel Kuri wrote:
>
> OK, I don't know what happened there, here's another one with the
> NOTIFYs. I'm on trunk rev 16633 and I have
> "managed-shared-appeareance=true" on the internal profile. I'm just
> making calls between internal phones.
>
> ????http://pastebin.freeswitch.org/12153
>
> Thanks,
> Gabe
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From gkuri at ieee.org  Mon Feb 15 12:53:40 2010
From: gkuri at ieee.org (Gabriel Kuri)
Date: Mon, 15 Feb 2010 12:53:40 -0800
Subject: [Freeswitch-users] Broadsoft SCA w/ Cisco SPA-5xx series
In-Reply-To: <0dd701caae7e$3c108b70$b431a250$@com>
References: <8b1c9cda1002141103te523643ob16db25e3bfa877f@mail.gmail.com>
	<0D67132C-69D8-43AD-B7E6-C7545B544322@freeswitch.org>
	<8b1c9cda1002142159h5efcc765q9a41baa1bf694c8c@mail.gmail.com>
	<191c3a031002150832x7faa9a78xbc160cde63f09d09@mail.gmail.com>
	<8b1c9cda1002151048o4b45c034qb22732a9366024ae@mail.gmail.com>
	<191c3a031002151055v1938ab91p89232341f4087c8e@mail.gmail.com>
	<191c3a031002151104u7413b17dh84e68ab1cf703cb9@mail.gmail.com>
	<8b1c9cda1002151147l1e072d00y69c0022e3924c3b4@mail.gmail.com>
	<0dd701caae7e$3c108b70$b431a250$@com>
Message-ID: <8b1c9cda1002151253t32c22663wf66245e0220a4e6@mail.gmail.com>

No, that was a typo. I have it correct in the config file.

Gabe

On Mon, Feb 15, 2010 at 12:34 PM, Peder <peder at networkoblivion.com> wrote:
> Is this a typo "managed-shared-appeareance=true" or is there an extra e in
> appearance in your config?
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Gabriel
> Kuri
> Sent: Monday, February 15, 2010 1:48 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Broadsoft SCA w/ Cisco SPA-5xx series
>
> OK, I don't know what happened there, here's another one with the
> NOTIFYs. I'm on trunk rev 16633 and I have
> "managed-shared-appeareance=true" on the internal profile. I'm just
> making calls between internal phones.
>
> ? ? http://pastebin.freeswitch.org/12153
>
> Thanks,
> Gabe
>
> On Mon, Feb 15, 2010 at 11:04 AM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>> I don't see any notifies at all in this trace do the profiles in question
>> have:
>> manage-shared-appearance set to true?
>> and are you on latest trunk?
>>
>>
>> On Mon, Feb 15, 2010 at 12:55 PM, Anthony Minessale
>> <anthony.minessale at gmail.com> wrote:
>>>
>>> we log the sql stmts on err so they are red and easier to read.
>>>
>>>
>>>
>>> On Mon, Feb 15, 2010 at 12:48 PM, Gabriel Kuri <gkuri at ieee.org> wrote:
>>>>
>>>> Here's a call from ext 2551 to 2552 (two phones). I noticed a bunch of
>>>> errors related to SQL UPDATE for presence ...
>>>>
>>>> ? ? http://pastebin.freeswitch.org/12152
>>>>
>>>> Thanks,
>>>> Gabe
>>>>
>>>>
>>>> On Mon, Feb 15, 2010 at 8:32 AM, Anthony Minessale
>>>> <anthony.minessale at gmail.com> wrote:
>>>> > it should be active not seized.
>>>> > seized is when you take it off hook.
>>>> >
>>>> > We need some more debugging to be sure.
>>>> > Can we work in real time on it or can you get a more detailed log?
>>>> >
>>>> > edit sofia.conf.xml and add the param to the "settings" section.
>>>> >
>>>> > <param name="debug-sla" value="100"/>
>>>> >
>>>> >
>>>> > then restart and enable sip trace and debug level
>>>> >
>>>> > //do this for every profile involved in the call.
>>>> > sofia profile <profilename> siptrace on
>>>> >
>>>> > //also do this
>>>> > console loglevel debug
>>>> >
>>>> >
>>>> > if you can let us ssh, we can do all the for you if you can make the
>>>> > test
>>>> > calls.
>>>> >
>>>> >
>>>> >
>>>> >
>>>> > On Sun, Feb 14, 2010 at 11:59 PM, Gabriel Kuri <gkuri at ieee.org> wrote:
>>>> >>
>>>> >> BTW, here's a copy of the NOTIFY (event call-info) sent to the other
>>>> >> phone after the first phone is answered, should this have a Call-Info
>>>> >> line with an "appearance-state=seized" to turn on the light on the
>>>> >> other phone?
>>>> >>
>>>> >>
>>>> >> NOTIFY sip:2551@<removed>:54446 SIP/2.0.
>>>> >> Via: SIP/2.0/UDP <removed>:9430;rport;branch=z9hG4bK71pN2cXgH851K.
>>>> >> Max-Forwards: 70.
>>>> >> From: <sip:2551 at mydomain.com>;tag=XeB6ZrKDevpHp.
>>>> >> To: <sip:2551 at mydomain.com>;tag=c2d34993aac6ea.
>>>> >> Call-ID: 34c34987-8b6fa786@<removed>.
>>>> >> CSeq: 126950830 NOTIFY.
>>>> >> Contact: <sip:2551@<removed>:9430>.
>>>> >> Expires: 3959.
>>>> >> Call-Info:
>>>> >> <sip:mydomain.com>;appearance-index=*;appearance-state=idle.
>>>> >> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16633M.
>>>> >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>>>> >> REGISTER, REFER, PRACK, NOTIFY, PUBLISH, SUBSCRIBE.
>>>> >> Supported: 100rel, timer, precondition, path, replaces.
>>>> >> Event: call-info.
>>>> >> Allow-Events: talk, presence, dialog, line-seize, call-info, sla,
>>>> >> include-session-description, presence.winfo, message-summary, refer.
>>>> >> Subscription-State: active;expires=3959.
>>>> >> Content-Length: 0.
>>>> >>
>>>> >>
>>>> >>
>>>> >> On Sun, Feb 14, 2010 at 7:58 PM, Brian West <brian at freeswitch.org>
>>>> >> wrote:
>>>> >> > Works fine here... is your box slow or something?
>>>> >> >
>>>> >> > /b
>>>> >> >
>>>> >> > On Feb 14, 2010, at 1:03 PM, Gabriel Kuri wrote:
>>>> >> >
>>>> >> >> I followed Brian's directions from one of the previous threads on
>>>> >> >> configuring the SPA-5xx series phones for Broadsoft SCA and set
>>>> >> >> manage-shared-appearance=true in the internal profile. SCA appears
>>>> >> >> to
>>>> >> >> be working on outgoing calls between two phones, the line key
>>>> >> >> starts
>>>> >> >> flashing red on the second phone when the first phone picks up the
>>>> >> >> receiver to make a call. However on incoming calls, both phones
>>>> >> >> ring
>>>> >> >> (same extension), however when one of the phones picks up the
> line,
>>>> >> >> the second phone's line key doesn't flash red or show the first
>>>> >> >> phone
>>>> >> >> on that incoming call. Any ideas? Does shared appearance only work
>>>> >> >> on
>>>> >> >> outgoing phone calls?
>>>> >> >>
>>>> >> >> Thanks,
>>>> >> >> Gabe
>>>> >> >
>>>> >> >
>>>> >> > _______________________________________________
>>>> >> > FreeSWITCH-users mailing list
>>>> >> > FreeSWITCH-users at lists.freeswitch.org
>>>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >> >
>>>> >> >
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> >> > http://www.freeswitch.org
>>>> >> >
>>>> >>
>>>> >> _______________________________________________
>>>> >> FreeSWITCH-users mailing list
>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >>
>>>> >>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> >> http://www.freeswitch.org
>>>> >
>>>> >
>>>> >
>>>> > --
>>>> > Anthony Minessale II
>>>> >
>>>> > FreeSWITCH http://www.freeswitch.org/
>>>> > ClueCon http://www.cluecon.com/
>>>> > Twitter: http://twitter.com/FreeSWITCH_wire
>>>> >
>>>> > AIM: anthm
>>>> > MSN:anthony_minessale at hotmail.com
>>>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>>> > IRC: irc.freenode.net #freeswitch
>>>> >
>>>> > FreeSWITCH Developer Conference
>>>> > sip:888 at conference.freeswitch.org
>>>> > iax:guest at conference.freeswitch.org/888
>>>> > googletalk:conf+888 at conference.freeswitch.org
>>>> > pstn:+19193869900
>>>> >
>>>> > _______________________________________________
>>>> > FreeSWITCH-users mailing list
>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >
>>>> >
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> > http://www.freeswitch.org
>>>> >
>>>> >
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org
>>> iax:guest at conference.freeswitch.org/888
>>> googletalk:conf+888 at conference.freeswitch.org
>>> pstn:+19193869900
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From anthony.minessale at gmail.com  Mon Feb 15 13:04:01 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 15 Feb 2010 15:04:01 -0600
Subject: [Freeswitch-users] Broadsoft SCA w/ Cisco SPA-5xx series
In-Reply-To: <8b1c9cda1002151253t32c22663wf66245e0220a4e6@mail.gmail.com>
References: <8b1c9cda1002141103te523643ob16db25e3bfa877f@mail.gmail.com>
	<0D67132C-69D8-43AD-B7E6-C7545B544322@freeswitch.org>
	<8b1c9cda1002142159h5efcc765q9a41baa1bf694c8c@mail.gmail.com>
	<191c3a031002150832x7faa9a78xbc160cde63f09d09@mail.gmail.com>
	<8b1c9cda1002151048o4b45c034qb22732a9366024ae@mail.gmail.com>
	<191c3a031002151055v1938ab91p89232341f4087c8e@mail.gmail.com>
	<191c3a031002151104u7413b17dh84e68ab1cf703cb9@mail.gmail.com>
	<8b1c9cda1002151147l1e072d00y69c0022e3924c3b4@mail.gmail.com>
	<0dd701caae7e$3c108b70$b431a250$@com>
	<8b1c9cda1002151253t32c22663wf66245e0220a4e6@mail.gmail.com>
Message-ID: <191c3a031002151304h54c262c5o8ffd5c195e0512d2@mail.gmail.com>

you are missing something because you have no seize events when you go on
and off hook.
is every phone in the correct mode?


On Mon, Feb 15, 2010 at 2:53 PM, Gabriel Kuri <gkuri at ieee.org> wrote:

> No, that was a typo. I have it correct in the config file.
>
> Gabe
>
> On Mon, Feb 15, 2010 at 12:34 PM, Peder <peder at networkoblivion.com> wrote:
> > Is this a typo "managed-shared-appeareance=true" or is there an extra e
> in
> > appearance in your config?
> >
> > -----Original Message-----
> > From: freeswitch-users-bounces at lists.freeswitch.org
> > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Gabriel
> > Kuri
> > Sent: Monday, February 15, 2010 1:48 PM
> > To: freeswitch-users at lists.freeswitch.org
> > Subject: Re: [Freeswitch-users] Broadsoft SCA w/ Cisco SPA-5xx series
> >
> > OK, I don't know what happened there, here's another one with the
> > NOTIFYs. I'm on trunk rev 16633 and I have
> > "managed-shared-appeareance=true" on the internal profile. I'm just
> > making calls between internal phones.
> >
> >     http://pastebin.freeswitch.org/12153
> >
> > Thanks,
> > Gabe
> >
> > On Mon, Feb 15, 2010 at 11:04 AM, Anthony Minessale
> > <anthony.minessale at gmail.com> wrote:
> >> I don't see any notifies at all in this trace do the profiles in
> question
> >> have:
> >> manage-shared-appearance set to true?
> >> and are you on latest trunk?
> >>
> >>
> >> On Mon, Feb 15, 2010 at 12:55 PM, Anthony Minessale
> >> <anthony.minessale at gmail.com> wrote:
> >>>
> >>> we log the sql stmts on err so they are red and easier to read.
> >>>
> >>>
> >>>
> >>> On Mon, Feb 15, 2010 at 12:48 PM, Gabriel Kuri <gkuri at ieee.org> wrote:
> >>>>
> >>>> Here's a call from ext 2551 to 2552 (two phones). I noticed a bunch of
> >>>> errors related to SQL UPDATE for presence ...
> >>>>
> >>>>     http://pastebin.freeswitch.org/12152
> >>>>
> >>>> Thanks,
> >>>> Gabe
> >>>>
> >>>>
> >>>> On Mon, Feb 15, 2010 at 8:32 AM, Anthony Minessale
> >>>> <anthony.minessale at gmail.com> wrote:
> >>>> > it should be active not seized.
> >>>> > seized is when you take it off hook.
> >>>> >
> >>>> > We need some more debugging to be sure.
> >>>> > Can we work in real time on it or can you get a more detailed log?
> >>>> >
> >>>> > edit sofia.conf.xml and add the param to the "settings" section.
> >>>> >
> >>>> > <param name="debug-sla" value="100"/>
> >>>> >
> >>>> >
> >>>> > then restart and enable sip trace and debug level
> >>>> >
> >>>> > //do this for every profile involved in the call.
> >>>> > sofia profile <profilename> siptrace on
> >>>> >
> >>>> > //also do this
> >>>> > console loglevel debug
> >>>> >
> >>>> >
> >>>> > if you can let us ssh, we can do all the for you if you can make the
> >>>> > test
> >>>> > calls.
> >>>> >
> >>>> >
> >>>> >
> >>>> >
> >>>> > On Sun, Feb 14, 2010 at 11:59 PM, Gabriel Kuri <gkuri at ieee.org>
> wrote:
> >>>> >>
> >>>> >> BTW, here's a copy of the NOTIFY (event call-info) sent to the
> other
> >>>> >> phone after the first phone is answered, should this have a
> Call-Info
> >>>> >> line with an "appearance-state=seized" to turn on the light on the
> >>>> >> other phone?
> >>>> >>
> >>>> >>
> >>>> >> NOTIFY sip:2551@<removed>:54446 SIP/2.0.
> >>>> >> Via: SIP/2.0/UDP <removed>:9430;rport;branch=z9hG4bK71pN2cXgH851K.
> >>>> >> Max-Forwards: 70.
> >>>> >> From: <sip:2551 at mydomain.com <sip%3A2551 at mydomain.com>
> >;tag=XeB6ZrKDevpHp.
> >>>> >> To: <sip:2551 at mydomain.com <sip%3A2551 at mydomain.com>
> >;tag=c2d34993aac6ea.
> >>>> >> Call-ID: 34c34987-8b6fa786@<removed>.
> >>>> >> CSeq: 126950830 NOTIFY.
> >>>> >> Contact: <sip:2551@<removed>:9430>.
> >>>> >> Expires: 3959.
> >>>> >> Call-Info:
> >>>> >> <sip:mydomain.com>;appearance-index=*;appearance-state=idle.
> >>>> >> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16633M.
> >>>> >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> >>>> >> REGISTER, REFER, PRACK, NOTIFY, PUBLISH, SUBSCRIBE.
> >>>> >> Supported: 100rel, timer, precondition, path, replaces.
> >>>> >> Event: call-info.
> >>>> >> Allow-Events: talk, presence, dialog, line-seize, call-info, sla,
> >>>> >> include-session-description, presence.winfo, message-summary,
> refer.
> >>>> >> Subscription-State: active;expires=3959.
> >>>> >> Content-Length: 0.
> >>>> >>
> >>>> >>
> >>>> >>
> >>>> >> On Sun, Feb 14, 2010 at 7:58 PM, Brian West <brian at freeswitch.org>
> >>>> >> wrote:
> >>>> >> > Works fine here... is your box slow or something?
> >>>> >> >
> >>>> >> > /b
> >>>> >> >
> >>>> >> > On Feb 14, 2010, at 1:03 PM, Gabriel Kuri wrote:
> >>>> >> >
> >>>> >> >> I followed Brian's directions from one of the previous threads
> on
> >>>> >> >> configuring the SPA-5xx series phones for Broadsoft SCA and set
> >>>> >> >> manage-shared-appearance=true in the internal profile. SCA
> appears
> >>>> >> >> to
> >>>> >> >> be working on outgoing calls between two phones, the line key
> >>>> >> >> starts
> >>>> >> >> flashing red on the second phone when the first phone picks up
> the
> >>>> >> >> receiver to make a call. However on incoming calls, both phones
> >>>> >> >> ring
> >>>> >> >> (same extension), however when one of the phones picks up the
> > line,
> >>>> >> >> the second phone's line key doesn't flash red or show the first
> >>>> >> >> phone
> >>>> >> >> on that incoming call. Any ideas? Does shared appearance only
> work
> >>>> >> >> on
> >>>> >> >> outgoing phone calls?
> >>>> >> >>
> >>>> >> >> Thanks,
> >>>> >> >> Gabe
> >>>> >> >
> >>>> >> >
> >>>> >> > _______________________________________________
> >>>> >> > FreeSWITCH-users mailing list
> >>>> >> > FreeSWITCH-users at lists.freeswitch.org
> >>>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>> >> >
> >>>> >> >
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>> >> > http://www.freeswitch.org
> >>>> >> >
> >>>> >>
> >>>> >> _______________________________________________
> >>>> >> FreeSWITCH-users mailing list
> >>>> >> FreeSWITCH-users at lists.freeswitch.org
> >>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>> >>
> >>>> >>
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>> >> http://www.freeswitch.org
> >>>> >
> >>>> >
> >>>> >
> >>>> > --
> >>>> > Anthony Minessale II
> >>>> >
> >>>> > FreeSWITCH http://www.freeswitch.org/
> >>>> > ClueCon http://www.cluecon.com/
> >>>> > Twitter: http://twitter.com/FreeSWITCH_wire
> >>>> >
> >>>> > AIM: anthm
> >>>> > MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
> >>>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >>>> > IRC: irc.freenode.net #freeswitch
> >>>> >
> >>>> > FreeSWITCH Developer Conference
> >>>> > sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> >>>> > iax:guest at conference.freeswitch.org/888
> >>>> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >>>> > pstn:+19193869900
> >>>> >
> >>>> > _______________________________________________
> >>>> > FreeSWITCH-users mailing list
> >>>> > FreeSWITCH-users at lists.freeswitch.org
> >>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>> >
> >>>> >
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>> > http://www.freeswitch.org
> >>>> >
> >>>> >
> >>>>
> >>>> _______________________________________________
> >>>> FreeSWITCH-users mailing list
> >>>> FreeSWITCH-users at lists.freeswitch.org
> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>> http://www.freeswitch.org
> >>>
> >>>
> >>>
> >>> --
> >>> Anthony Minessale II
> >>>
> >>> FreeSWITCH http://www.freeswitch.org/
> >>> ClueCon http://www.cluecon.com/
> >>> Twitter: http://twitter.com/FreeSWITCH_wire
> >>>
> >>> AIM: anthm
> >>> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
> >>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >>> IRC: irc.freenode.net #freeswitch
> >>>
> >>> FreeSWITCH Developer Conference
> >>> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> >>> iax:guest at conference.freeswitch.org/888
> >>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >>> pstn:+19193869900
> >>
> >>
> >>
> >> --
> >> Anthony Minessale II
> >>
> >> FreeSWITCH http://www.freeswitch.org/
> >> ClueCon http://www.cluecon.com/
> >> Twitter: http://twitter.com/FreeSWITCH_wire
> >>
> >> AIM: anthm
> >> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >> IRC: irc.freenode.net #freeswitch
> >>
> >> FreeSWITCH Developer Conference
> >> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> >> iax:guest at conference.freeswitch.org/888
> >> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >> pstn:+19193869900
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100215/f04cb603/attachment-0002.html 

From peder at networkoblivion.com  Mon Feb 15 13:17:57 2010
From: peder at networkoblivion.com (Peder)
Date: Mon, 15 Feb 2010 15:17:57 -0600
Subject: [Freeswitch-users] Broadsoft SCA w/ Cisco SPA-5xx series
In-Reply-To: <191c3a031002151304h54c262c5o8ffd5c195e0512d2@mail.gmail.com>
References: <8b1c9cda1002141103te523643ob16db25e3bfa877f@mail.gmail.com>	<0D67132C-69D8-43AD-B7E6-C7545B544322@freeswitch.org>	<8b1c9cda1002142159h5efcc765q9a41baa1bf694c8c@mail.gmail.com>	<191c3a031002150832x7faa9a78xbc160cde63f09d09@mail.gmail.com>	<8b1c9cda1002151048o4b45c034qb22732a9366024ae@mail.gmail.com>	<191c3a031002151055v1938ab91p89232341f4087c8e@mail.gmail.com>	<191c3a031002151104u7413b17dh84e68ab1cf703cb9@mail.gmail.com>	<8b1c9cda1002151147l1e072d00y69c0022e3924c3b4@mail.gmail.com>	<0dd701caae7e$3c108b70$b431a250$@com>	<8b1c9cda1002151253t32c22663wf66245e0220a4e6@mail.gmail.com>
	<191c3a031002151304h54c262c5o8ffd5c195e0512d2@mail.gmail.com>
Message-ID: <0e7301caae84$58b47ce0$0a1d76a0$@com>

On the phone itself, do you have the line set to shared and "Broadsoft SCA"
enabled?

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Monday, February 15, 2010 3:04 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Broadsoft SCA w/ Cisco SPA-5xx series

 

you are missing something because you have no seize events when you go on
and off hook.
is every phone in the correct mode?



On Mon, Feb 15, 2010 at 2:53 PM, Gabriel Kuri <gkuri at ieee.org> wrote:

No, that was a typo. I have it correct in the config file.

Gabe


On Mon, Feb 15, 2010 at 12:34 PM, Peder <peder at networkoblivion.com> wrote:

> Is this a typo "managed-shared-appeareance=true" or is there an extra e in
> appearance in your config?
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Gabriel
> Kuri
> Sent: Monday, February 15, 2010 1:48 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Broadsoft SCA w/ Cisco SPA-5xx series
>
> OK, I don't know what happened there, here's another one with the
> NOTIFYs. I'm on trunk rev 16633 and I have
> "managed-shared-appeareance=true" on the internal profile. I'm just
> making calls between internal phones.
>
>     http://pastebin.freeswitch.org/12153
>
> Thanks,
> Gabe
>
> On Mon, Feb 15, 2010 at 11:04 AM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>> I don't see any notifies at all in this trace do the profiles in question
>> have:
>> manage-shared-appearance set to true?
>> and are you on latest trunk?
>>
>>
>> On Mon, Feb 15, 2010 at 12:55 PM, Anthony Minessale
>> <anthony.minessale at gmail.com> wrote:
>>>
>>> we log the sql stmts on err so they are red and easier to read.
>>>
>>>
>>>
>>> On Mon, Feb 15, 2010 at 12:48 PM, Gabriel Kuri <gkuri at ieee.org> wrote:
>>>>
>>>> Here's a call from ext 2551 to 2552 (two phones). I noticed a bunch of
>>>> errors related to SQL UPDATE for presence ...
>>>>
>>>>     http://pastebin.freeswitch.org/12152
>>>>
>>>> Thanks,
>>>> Gabe
>>>>
>>>>
>>>> On Mon, Feb 15, 2010 at 8:32 AM, Anthony Minessale
>>>> <anthony.minessale at gmail.com> wrote:
>>>> > it should be active not seized.
>>>> > seized is when you take it off hook.
>>>> >
>>>> > We need some more debugging to be sure.
>>>> > Can we work in real time on it or can you get a more detailed log?
>>>> >
>>>> > edit sofia.conf.xml and add the param to the "settings" section.
>>>> >
>>>> > <param name="debug-sla" value="100"/>
>>>> >
>>>> >
>>>> > then restart and enable sip trace and debug level
>>>> >
>>>> > //do this for every profile involved in the call.
>>>> > sofia profile <profilename> siptrace on
>>>> >
>>>> > //also do this
>>>> > console loglevel debug
>>>> >
>>>> >
>>>> > if you can let us ssh, we can do all the for you if you can make the
>>>> > test
>>>> > calls.
>>>> >
>>>> >
>>>> >
>>>> >
>>>> > On Sun, Feb 14, 2010 at 11:59 PM, Gabriel Kuri <gkuri at ieee.org>
wrote:
>>>> >>
>>>> >> BTW, here's a copy of the NOTIFY (event call-info) sent to the other
>>>> >> phone after the first phone is answered, should this have a
Call-Info
>>>> >> line with an "appearance-state=seized" to turn on the light on the
>>>> >> other phone?
>>>> >>
>>>> >>
>>>> >> NOTIFY sip:2551@<removed>:54446 SIP/2.0.
>>>> >> Via: SIP/2.0/UDP <removed>:9430;rport;branch=z9hG4bK71pN2cXgH851K.
>>>> >> Max-Forwards: 70.
>>>> >> From: <sip:2551 at mydomain.com <mailto:sip%3A2551 at mydomain.com>
>;tag=XeB6ZrKDevpHp.
>>>> >> To: <sip:2551 at mydomain.com <mailto:sip%3A2551 at mydomain.com>
>;tag=c2d34993aac6ea.
>>>> >> Call-ID: 34c34987-8b6fa786@<removed>.
>>>> >> CSeq: 126950830 NOTIFY.
>>>> >> Contact: <sip:2551@<removed>:9430>.
>>>> >> Expires: 3959.
>>>> >> Call-Info:
>>>> >> <sip:mydomain.com>;appearance-index=*;appearance-state=idle.
>>>> >> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16633M.
>>>> >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>>>> >> REGISTER, REFER, PRACK, NOTIFY, PUBLISH, SUBSCRIBE.
>>>> >> Supported: 100rel, timer, precondition, path, replaces.
>>>> >> Event: call-info.
>>>> >> Allow-Events: talk, presence, dialog, line-seize, call-info, sla,
>>>> >> include-session-description, presence.winfo, message-summary, refer.
>>>> >> Subscription-State: active;expires=3959.
>>>> >> Content-Length: 0.
>>>> >>
>>>> >>
>>>> >>
>>>> >> On Sun, Feb 14, 2010 at 7:58 PM, Brian West <brian at freeswitch.org>
>>>> >> wrote:
>>>> >> > Works fine here... is your box slow or something?
>>>> >> >
>>>> >> > /b
>>>> >> >
>>>> >> > On Feb 14, 2010, at 1:03 PM, Gabriel Kuri wrote:
>>>> >> >
>>>> >> >> I followed Brian's directions from one of the previous threads on
>>>> >> >> configuring the SPA-5xx series phones for Broadsoft SCA and set
>>>> >> >> manage-shared-appearance=true in the internal profile. SCA
appears
>>>> >> >> to
>>>> >> >> be working on outgoing calls between two phones, the line key
>>>> >> >> starts
>>>> >> >> flashing red on the second phone when the first phone picks up
the
>>>> >> >> receiver to make a call. However on incoming calls, both phones
>>>> >> >> ring
>>>> >> >> (same extension), however when one of the phones picks up the
> line,
>>>> >> >> the second phone's line key doesn't flash red or show the first
>>>> >> >> phone
>>>> >> >> on that incoming call. Any ideas? Does shared appearance only
work
>>>> >> >> on
>>>> >> >> outgoing phone calls?
>>>> >> >>
>>>> >> >> Thanks,
>>>> >> >> Gabe
>>>> >> >
>>>> >> >
>>>> >> > _______________________________________________
>>>> >> > FreeSWITCH-users mailing list
>>>> >> > FreeSWITCH-users at lists.freeswitch.org
>>>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >> >
>>>> >> >
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> >> > http://www.freeswitch.org
>>>> >> >
>>>> >>
>>>> >> _______________________________________________
>>>> >> FreeSWITCH-users mailing list
>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >>
>>>> >>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> >> http://www.freeswitch.org
>>>> >
>>>> >
>>>> >
>>>> > --
>>>> > Anthony Minessale II
>>>> >
>>>> > FreeSWITCH http://www.freeswitch.org/
>>>> > ClueCon http://www.cluecon.com/
>>>> > Twitter: http://twitter.com/FreeSWITCH_wire
>>>> >
>>>> > AIM: anthm
>>>> > MSN:anthony_minessale at hotmail.com
<mailto:MSN%3Aanthony_minessale at hotmail.com> 
>>>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
<mailto:PAYPAL%3Aanthony.minessale at gmail.com> 
>>>> > IRC: irc.freenode.net #freeswitch
>>>> >
>>>> > FreeSWITCH Developer Conference
>>>> > sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
>>>> > iax:guest at conference.freeswitch.org/888
>>>> > googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
>>>> > pstn:+19193869900
>>>> >
>>>> > _______________________________________________
>>>> > FreeSWITCH-users mailing list
>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >
>>>> >
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> > http://www.freeswitch.org
>>>> >
>>>> >
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com
<mailto:MSN%3Aanthony_minessale at hotmail.com> 
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
<mailto:PAYPAL%3Aanthony.minessale at gmail.com> 
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
>>> iax:guest at conference.freeswitch.org/888
>>> googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
>>> pstn:+19193869900
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
<mailto:MSN%3Aanthony_minessale at hotmail.com> 
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
<mailto:PAYPAL%3Aanthony.minessale at gmail.com> 
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
<mailto:MSN%3Aanthony_minessale at hotmail.com> 
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
<mailto:PAYPAL%3Aanthony.minessale at gmail.com> 
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:+19193869900

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100215/2d11e79a/attachment-0002.html 

From rupa at rupa.com  Mon Feb 15 13:39:12 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Mon, 15 Feb 2010 15:39:12 -0600
Subject: [Freeswitch-users] CS_REPORTING state and CHANNEL_HANGUP event
In-Reply-To: <D0F4BBA35AD84876B35F99FCD443DAF9@abisoft.biz>
References: <D0F4BBA35AD84876B35F99FCD443DAF9@abisoft.biz>
Message-ID: <b39273531002151339k28aadcffyfd1e04df48d01da7@mail.gmail.com>

CS_REPORTING is where CDRs are generated.   How are you doing CDRs or are
you trying to bill at call hangup time?

2010/2/15 Kozak Vladimir <vkozak at abisoft.spb.ru>

>  Hello all.
> I have the following questions:
>     1) Sometime channel whis CS_REPORTING state remain. What is channel
> whis CS_REPORTING state mean? for example: 2c90ac6b-7147-4aac-82fb-a23d6d5c4185,outbound,2010-02-10
> *        11:53:02,1265820782,sofia/internal/sip:7100 at 76.74.160.163:57312
> ,CS_REPORTING,FreeSWITCH,sipp,,7100,,,,default,,,,,,pst01.localdomain.com,
> *,
> How can I kill this channel?  uuid_kill
> 2c90ac6b-7147-4aac-82fb-a23d6d5c4185 is not work.
>     2) Sometime I get CHANNEL_HANGUP event but real call in eyeBeam is
> live for the present. When come CHANNEL_HANGUP event and response on api
> uuid_kill command? It's come when FS send hangup to phone or when phone
> confirm hangup?
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100215/25cff3e1/attachment-0002.html 

From errotan at gmail.com  Mon Feb 15 13:57:28 2010
From: errotan at gmail.com (=?iso-8859-1?q?Pusk=E1s_Zsolt?=)
Date: Mon, 15 Feb 2010 22:57:28 +0100
Subject: [Freeswitch-users] FreeSWITCH Debian packages - build issues
In-Reply-To: <20100215064616.GA32700@jdc.jasonjgw.net>
References: <20100213030812.GA19108@jdc.jasonjgw.net>
	<9CF8766D-2367-4275-ACCF-0F9C6ABC7A09@jerris.com>
	<20100215064616.GA32700@jdc.jasonjgw.net>
Message-ID: <201002152257.28593.errotan@gmail.com>

2010. febru?r 15. 07.46.16 Jason White d?tummal ezt ?rta:
> Michael Jerris <mike at jerris.com> wrote:
> > Did anyone bother opening a bug on jira for this or are we going to just
> > tag 1.0.5 without deb packages?
> 
> Has anyone tried building these on Ubuntu 9.10 or Debian 5.0? I'm not in a
> position to do so at the moment.
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 

Just done with dpkg-buildpackage on Debian 5.0 "lenny" on x86.
It built without errors and everything looks ok and all modules exists, but 
haven't tried to install or run it.



From michal.kalinowski at interia.pl  Mon Feb 15 14:25:58 2010
From: michal.kalinowski at interia.pl (michal kalinowski)
Date: Mon, 15 Feb 2010 23:25:58 +0100
Subject: [Freeswitch-users] ivr from mysql
In-Reply-To: <87f2f3b91002151002v6bbc3e7fgb67dd6d3e72a915@mail.gmail.com>
References: <7c74f5761002150824i41a2ed5did008f36cbb4499a4@mail.gmail.com>
	<87f2f3b91002151002v6bbc3e7fgb67dd6d3e72a915@mail.gmail.com>
Message-ID: <7c74f5761002151425g512fa685n13c58d8f4b894740@mail.gmail.com>

I'm trying use in python something like this:

from freeswitch import *

def xml_fetch( param1, param2 ):
        xml = '''
<?xml version="1.0" encoding="UTF-8" standalone="no"?>
<document type="freeswitch/xml">
  <section name="dialplan" description="RE Dial Plan For FreeSwitch">
    <context name="default">
      <extension name="generated">
        <condition field="destination_number" expression="^9992$">>
         <action application="answer"/>
         <action application="playback" data="${hold_music}"/>
        </condition>
      </extension>
    </context>
  </section>
</document>
'''
        return xml

Of course XML context is with ivr parameters.

So I will try mod_xml_curl in my configuration.

BR,
Micha?


W dniu 15 lutego 2010 19:02 u?ytkownik Michael Collins
<msc at freeswitch.org> napisa?:
> When you say "xml from python" what exactly do you mean? Are you trying to
> use mod_xml_curl? If not you might want to check it out. The other choice is
> to use Lua from the dialplan, although I have a gut feeling that
> mod_xml_curl might be better for you.
> -MC
>
> 2010/2/15 michal kalinowski <michal.kalinowski at interia.pl>
>>
>> Hi,
>>
>> I need build ivr script/aplication which will take dynamically
>> configuration from mysql db (ivr menu, prompts, etc.).
>> My first idea is generate xml from python script. But it's not working
>> properly.
>>
>> Anybody has some idea or have this aplication already done ?
>>
>> BR,
>> Micha?
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From mrene_lists at avgs.ca  Mon Feb 15 14:32:48 2010
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Mon, 15 Feb 2010 17:32:48 -0500
Subject: [Freeswitch-users] ivr from mysql
In-Reply-To: <7c74f5761002151425g512fa685n13c58d8f4b894740@mail.gmail.com>
References: <7c74f5761002150824i41a2ed5did008f36cbb4499a4@mail.gmail.com>
	<87f2f3b91002151002v6bbc3e7fgb67dd6d3e72a915@mail.gmail.com>
	<7c74f5761002151425g512fa685n13c58d8f4b894740@mail.gmail.com>
Message-ID: <0FD30AAA-C2E2-4228-A09B-DB09F4E26811@avgs.ca>

You need to examine the parameters passed into the xml_fetch callback.  
The request will be in the configuration section for "ivr.conf" and  
its parameters will tell you which ivr should be loaded.

Also note that you have to return all the submenus at the same time  
(this is also valid for xml_curl) as freeswitch loads all of them at  
the same time to optimize the amount of times it does queries.

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 15-Feb-10, at 5:25 PM, michal kalinowski wrote:

> I'm trying use in python something like this:
>
> from freeswitch import *
>
> def xml_fetch( param1, param2 ):
>        xml = '''
> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
> <document type="freeswitch/xml">
>  <section name="dialplan" description="RE Dial Plan For FreeSwitch">
>    <context name="default">
>      <extension name="generated">
>        <condition field="destination_number" expression="^9992$">>
>         <action application="answer"/>
>         <action application="playback" data="${hold_music}"/>
>        </condition>
>      </extension>
>    </context>
>  </section>
> </document>
> '''
>        return xml
>
> Of course XML context is with ivr parameters.
>
> So I will try mod_xml_curl in my configuration.
>
> BR,
> Micha?
>
>
> W dniu 15 lutego 2010 19:02 u?ytkownik Michael Collins
> <msc at freeswitch.org> napisa?:
>> When you say "xml from python" what exactly do you mean? Are you  
>> trying to
>> use mod_xml_curl? If not you might want to check it out. The other  
>> choice is
>> to use Lua from the dialplan, although I have a gut feeling that
>> mod_xml_curl might be better for you.
>> -MC
>>
>> 2010/2/15 michal kalinowski <michal.kalinowski at interia.pl>
>>>
>>> Hi,
>>>
>>> I need build ivr script/aplication which will take dynamically
>>> configuration from mysql db (ivr menu, prompts, etc.).
>>> My first idea is generate xml from python script. But it's not  
>>> working
>>> properly.
>>>
>>> Anybody has some idea or have this aplication already done ?
>>>
>>> BR,
>>> Micha?
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From anthony.minessale at gmail.com  Mon Feb 15 14:49:25 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 15 Feb 2010 16:49:25 -0600
Subject: [Freeswitch-users] CS_REPORTING state and CHANNEL_HANGUP event
In-Reply-To: <b39273531002151339k28aadcffyfd1e04df48d01da7@mail.gmail.com>
References: <D0F4BBA35AD84876B35F99FCD443DAF9@abisoft.biz>
	<b39273531002151339k28aadcffyfd1e04df48d01da7@mail.gmail.com>
Message-ID: <191c3a031002151449i3f575a89me9ce23d4fc50dc6b@mail.gmail.com>

make sure you have this problem on latest trunk

On Mon, Feb 15, 2010 at 3:39 PM, Rupa Schomaker <rupa at rupa.com> wrote:

> CS_REPORTING is where CDRs are generated.   How are you doing CDRs or are
> you trying to bill at call hangup time?
>
> 2010/2/15 Kozak Vladimir <vkozak at abisoft.spb.ru>
>
>>  Hello all.
>> I have the following questions:
>>     1) Sometime channel whis CS_REPORTING state remain. What is channel
>> whis CS_REPORTING state mean? for example: 2c90ac6b-7147-4aac-82fb-a23d6d5c4185,outbound,2010-02-10
>> *        11:53:02,1265820782,sofia/internal/sip:7100 at 76.74.160.163:57312
>> ,CS_REPORTING,FreeSWITCH,sipp,,7100,,,,default,,,,,,pst01.localdomain.com
>> ,*,
>> How can I kill this channel?  uuid_kill
>> 2c90ac6b-7147-4aac-82fb-a23d6d5c4185 is not work.
>>     2) Sometime I get CHANNEL_HANGUP event but real call in eyeBeam is
>> live for the present. When come CHANNEL_HANGUP event and response on api
>> uuid_kill command? It's come when FS send hangup to phone or when phone
>> confirm hangup?
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100215/3adbe950/attachment-0002.html 

From errotan at gmail.com  Mon Feb 15 14:53:43 2010
From: errotan at gmail.com (=?iso-8859-1?q?Pusk=E1s_Zsolt?=)
Date: Mon, 15 Feb 2010 23:53:43 +0100
Subject: [Freeswitch-users] FreeSWITCH.Managed.dll deletes on make distclean
Message-ID: <201002152353.43666.errotan@gmail.com>

I usually do svn-clean than svn up when i compile a new version of fs.
I noticed that if i do make distclean the file @ 
src/mod/languages/mod_managed/managed/FreeSWITCH.Managed.dll got deleted.
When i do svn up it gets 'restored':

Restored 'src/mod/languages/mod_managed/managed/FreeSWITCH.Managed.dll'

Is this normal ?

I don't use mod_managed btw.



From gkuri at ieee.org  Mon Feb 15 14:54:12 2010
From: gkuri at ieee.org (Gabriel Kuri)
Date: Mon, 15 Feb 2010 14:54:12 -0800
Subject: [Freeswitch-users] Broadsoft SCA w/ Cisco SPA-5xx series
In-Reply-To: <191c3a031002151304h54c262c5o8ffd5c195e0512d2@mail.gmail.com>
References: <8b1c9cda1002141103te523643ob16db25e3bfa877f@mail.gmail.com>
	<8b1c9cda1002142159h5efcc765q9a41baa1bf694c8c@mail.gmail.com>
	<191c3a031002150832x7faa9a78xbc160cde63f09d09@mail.gmail.com>
	<8b1c9cda1002151048o4b45c034qb22732a9366024ae@mail.gmail.com>
	<191c3a031002151055v1938ab91p89232341f4087c8e@mail.gmail.com>
	<191c3a031002151104u7413b17dh84e68ab1cf703cb9@mail.gmail.com>
	<8b1c9cda1002151147l1e072d00y69c0022e3924c3b4@mail.gmail.com>
	<0dd701caae7e$3c108b70$b431a250$@com>
	<8b1c9cda1002151253t32c22663wf66245e0220a4e6@mail.gmail.com>
	<191c3a031002151304h54c262c5o8ffd5c195e0512d2@mail.gmail.com>
Message-ID: <8b1c9cda1002151454m45afd322o9fd3bc9f0a6fa58f@mail.gmail.com>

The phone I am dialing from (ext 2552) is an SPA-942 and I do not have
it configured for SCA. I only have the two phones than I am calling
(ext 2551) configured for SCA, so that's probably why you're not
seeing the seize events from the SPA-942.

Thanks,
Gabe



On Mon, Feb 15, 2010 at 1:04 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> you are missing something because you have no seize events when you go on
> and off hook.
> is every phone in the correct mode?
>
>
> On Mon, Feb 15, 2010 at 2:53 PM, Gabriel Kuri <gkuri at ieee.org> wrote:
>>
>> No, that was a typo. I have it correct in the config file.
>>
>> Gabe
>>
>> On Mon, Feb 15, 2010 at 12:34 PM, Peder <peder at networkoblivion.com> wrote:
>> > Is this a typo "managed-shared-appeareance=true" or is there an extra e
>> > in
>> > appearance in your config?
>> >
>> > -----Original Message-----
>> > From: freeswitch-users-bounces at lists.freeswitch.org
>> > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
>> > Gabriel
>> > Kuri
>> > Sent: Monday, February 15, 2010 1:48 PM
>> > To: freeswitch-users at lists.freeswitch.org
>> > Subject: Re: [Freeswitch-users] Broadsoft SCA w/ Cisco SPA-5xx series
>> >
>> > OK, I don't know what happened there, here's another one with the
>> > NOTIFYs. I'm on trunk rev 16633 and I have
>> > "managed-shared-appeareance=true" on the internal profile. I'm just
>> > making calls between internal phones.
>> >
>> > ? ? http://pastebin.freeswitch.org/12153
>> >
>> > Thanks,
>> > Gabe
>> >
>> > On Mon, Feb 15, 2010 at 11:04 AM, Anthony Minessale
>> > <anthony.minessale at gmail.com> wrote:
>> >> I don't see any notifies at all in this trace do the profiles in
>> >> question
>> >> have:
>> >> manage-shared-appearance set to true?
>> >> and are you on latest trunk?
>> >>
>> >>
>> >> On Mon, Feb 15, 2010 at 12:55 PM, Anthony Minessale
>> >> <anthony.minessale at gmail.com> wrote:
>> >>>
>> >>> we log the sql stmts on err so they are red and easier to read.
>> >>>
>> >>>
>> >>>
>> >>> On Mon, Feb 15, 2010 at 12:48 PM, Gabriel Kuri <gkuri at ieee.org> wrote:
>> >>>>
>> >>>> Here's a call from ext 2551 to 2552 (two phones). I noticed a bunch
>> >>>> of
>> >>>> errors related to SQL UPDATE for presence ...
>> >>>>
>> >>>> ? ? http://pastebin.freeswitch.org/12152
>> >>>>
>> >>>> Thanks,
>> >>>> Gabe
>> >>>>
>> >>>>
>> >>>> On Mon, Feb 15, 2010 at 8:32 AM, Anthony Minessale
>> >>>> <anthony.minessale at gmail.com> wrote:
>> >>>> > it should be active not seized.
>> >>>> > seized is when you take it off hook.
>> >>>> >
>> >>>> > We need some more debugging to be sure.
>> >>>> > Can we work in real time on it or can you get a more detailed log?
>> >>>> >
>> >>>> > edit sofia.conf.xml and add the param to the "settings" section.
>> >>>> >
>> >>>> > <param name="debug-sla" value="100"/>
>> >>>> >
>> >>>> >
>> >>>> > then restart and enable sip trace and debug level
>> >>>> >
>> >>>> > //do this for every profile involved in the call.
>> >>>> > sofia profile <profilename> siptrace on
>> >>>> >
>> >>>> > //also do this
>> >>>> > console loglevel debug
>> >>>> >
>> >>>> >
>> >>>> > if you can let us ssh, we can do all the for you if you can make
>> >>>> > the
>> >>>> > test
>> >>>> > calls.
>> >>>> >
>> >>>> >
>> >>>> >
>> >>>> >
>> >>>> > On Sun, Feb 14, 2010 at 11:59 PM, Gabriel Kuri <gkuri at ieee.org>
>> >>>> > wrote:
>> >>>> >>
>> >>>> >> BTW, here's a copy of the NOTIFY (event call-info) sent to the
>> >>>> >> other
>> >>>> >> phone after the first phone is answered, should this have a
>> >>>> >> Call-Info
>> >>>> >> line with an "appearance-state=seized" to turn on the light on the
>> >>>> >> other phone?
>> >>>> >>
>> >>>> >>
>> >>>> >> NOTIFY sip:2551@<removed>:54446 SIP/2.0.
>> >>>> >> Via: SIP/2.0/UDP <removed>:9430;rport;branch=z9hG4bK71pN2cXgH851K.
>> >>>> >> Max-Forwards: 70.
>> >>>> >> From: <sip:2551 at mydomain.com>;tag=XeB6ZrKDevpHp.
>> >>>> >> To: <sip:2551 at mydomain.com>;tag=c2d34993aac6ea.
>> >>>> >> Call-ID: 34c34987-8b6fa786@<removed>.
>> >>>> >> CSeq: 126950830 NOTIFY.
>> >>>> >> Contact: <sip:2551@<removed>:9430>.
>> >>>> >> Expires: 3959.
>> >>>> >> Call-Info:
>> >>>> >> <sip:mydomain.com>;appearance-index=*;appearance-state=idle.
>> >>>> >> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16633M.
>> >>>> >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>> >>>> >> REGISTER, REFER, PRACK, NOTIFY, PUBLISH, SUBSCRIBE.
>> >>>> >> Supported: 100rel, timer, precondition, path, replaces.
>> >>>> >> Event: call-info.
>> >>>> >> Allow-Events: talk, presence, dialog, line-seize, call-info, sla,
>> >>>> >> include-session-description, presence.winfo, message-summary,
>> >>>> >> refer.
>> >>>> >> Subscription-State: active;expires=3959.
>> >>>> >> Content-Length: 0.
>> >>>> >>
>> >>>> >>
>> >>>> >>
>> >>>> >> On Sun, Feb 14, 2010 at 7:58 PM, Brian West <brian at freeswitch.org>
>> >>>> >> wrote:
>> >>>> >> > Works fine here... is your box slow or something?
>> >>>> >> >
>> >>>> >> > /b
>> >>>> >> >
>> >>>> >> > On Feb 14, 2010, at 1:03 PM, Gabriel Kuri wrote:
>> >>>> >> >
>> >>>> >> >> I followed Brian's directions from one of the previous threads
>> >>>> >> >> on
>> >>>> >> >> configuring the SPA-5xx series phones for Broadsoft SCA and set
>> >>>> >> >> manage-shared-appearance=true in the internal profile. SCA
>> >>>> >> >> appears
>> >>>> >> >> to
>> >>>> >> >> be working on outgoing calls between two phones, the line key
>> >>>> >> >> starts
>> >>>> >> >> flashing red on the second phone when the first phone picks up
>> >>>> >> >> the
>> >>>> >> >> receiver to make a call. However on incoming calls, both phones
>> >>>> >> >> ring
>> >>>> >> >> (same extension), however when one of the phones picks up the
>> > line,
>> >>>> >> >> the second phone's line key doesn't flash red or show the first
>> >>>> >> >> phone
>> >>>> >> >> on that incoming call. Any ideas? Does shared appearance only
>> >>>> >> >> work
>> >>>> >> >> on
>> >>>> >> >> outgoing phone calls?
>> >>>> >> >>
>> >>>> >> >> Thanks,
>> >>>> >> >> Gabe
>> >>>> >> >
>> >>>> >> >
>> >>>> >> > _______________________________________________
>> >>>> >> > FreeSWITCH-users mailing list
>> >>>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >>>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>> >> >
>> >>>> >> >
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>>> >> > http://www.freeswitch.org
>> >>>> >> >
>> >>>> >>
>> >>>> >> _______________________________________________
>> >>>> >> FreeSWITCH-users mailing list
>> >>>> >> FreeSWITCH-users at lists.freeswitch.org
>> >>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>> >>
>> >>>> >>
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>>> >> http://www.freeswitch.org
>> >>>> >
>> >>>> >
>> >>>> >
>> >>>> > --
>> >>>> > Anthony Minessale II
>> >>>> >
>> >>>> > FreeSWITCH http://www.freeswitch.org/
>> >>>> > ClueCon http://www.cluecon.com/
>> >>>> > Twitter: http://twitter.com/FreeSWITCH_wire
>> >>>> >
>> >>>> > AIM: anthm
>> >>>> > MSN:anthony_minessale at hotmail.com
>> >>>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> >>>> > IRC: irc.freenode.net #freeswitch
>> >>>> >
>> >>>> > FreeSWITCH Developer Conference
>> >>>> > sip:888 at conference.freeswitch.org
>> >>>> > iax:guest at conference.freeswitch.org/888
>> >>>> > googletalk:conf+888 at conference.freeswitch.org
>> >>>> > pstn:+19193869900
>> >>>> >
>> >>>> > _______________________________________________
>> >>>> > FreeSWITCH-users mailing list
>> >>>> > FreeSWITCH-users at lists.freeswitch.org
>> >>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>> >
>> >>>> >
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>>> > http://www.freeswitch.org
>> >>>> >
>> >>>> >
>> >>>>
>> >>>> _______________________________________________
>> >>>> FreeSWITCH-users mailing list
>> >>>> FreeSWITCH-users at lists.freeswitch.org
>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>>
>> >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>>> http://www.freeswitch.org
>> >>>
>> >>>
>> >>>
>> >>> --
>> >>> Anthony Minessale II
>> >>>
>> >>> FreeSWITCH http://www.freeswitch.org/
>> >>> ClueCon http://www.cluecon.com/
>> >>> Twitter: http://twitter.com/FreeSWITCH_wire
>> >>>
>> >>> AIM: anthm
>> >>> MSN:anthony_minessale at hotmail.com
>> >>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> >>> IRC: irc.freenode.net #freeswitch
>> >>>
>> >>> FreeSWITCH Developer Conference
>> >>> sip:888 at conference.freeswitch.org
>> >>> iax:guest at conference.freeswitch.org/888
>> >>> googletalk:conf+888 at conference.freeswitch.org
>> >>> pstn:+19193869900
>> >>
>> >>
>> >>
>> >> --
>> >> Anthony Minessale II
>> >>
>> >> FreeSWITCH http://www.freeswitch.org/
>> >> ClueCon http://www.cluecon.com/
>> >> Twitter: http://twitter.com/FreeSWITCH_wire
>> >>
>> >> AIM: anthm
>> >> MSN:anthony_minessale at hotmail.com
>> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> >> IRC: irc.freenode.net #freeswitch
>> >>
>> >> FreeSWITCH Developer Conference
>> >> sip:888 at conference.freeswitch.org
>> >> iax:guest at conference.freeswitch.org/888
>> >> googletalk:conf+888 at conference.freeswitch.org
>> >> pstn:+19193869900
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >>
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From gkuri at ieee.org  Mon Feb 15 14:56:07 2010
From: gkuri at ieee.org (Gabriel Kuri)
Date: Mon, 15 Feb 2010 14:56:07 -0800
Subject: [Freeswitch-users] Broadsoft SCA w/ Cisco SPA-5xx series
In-Reply-To: <0e7301caae84$58b47ce0$0a1d76a0$@com>
References: <8b1c9cda1002141103te523643ob16db25e3bfa877f@mail.gmail.com>
	<191c3a031002150832x7faa9a78xbc160cde63f09d09@mail.gmail.com>
	<8b1c9cda1002151048o4b45c034qb22732a9366024ae@mail.gmail.com>
	<191c3a031002151055v1938ab91p89232341f4087c8e@mail.gmail.com>
	<191c3a031002151104u7413b17dh84e68ab1cf703cb9@mail.gmail.com>
	<8b1c9cda1002151147l1e072d00y69c0022e3924c3b4@mail.gmail.com>
	<0dd701caae7e$3c108b70$b431a250$@com>
	<8b1c9cda1002151253t32c22663wf66245e0220a4e6@mail.gmail.com>
	<191c3a031002151304h54c262c5o8ffd5c195e0512d2@mail.gmail.com>
	<0e7301caae84$58b47ce0$0a1d76a0$@com>
Message-ID: <8b1c9cda1002151456s5dec6dbbtaa2063e0f70fe5dd@mail.gmail.com>

Yes, the two phones being called (SPA-509Gs) have SCA enabled and it
works when making outgoing calls from those phones. But incoming calls
to those two phones don't seem to have the line key light up on the
other phone when one of the phones is answered (same extension).

Thanks,
Gabe

On Mon, Feb 15, 2010 at 1:17 PM, Peder <peder at networkoblivion.com> wrote:
> On the phone itself, do you have the line set to shared and ?Broadsoft SCA?
> enabled?
>
>
>
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
> Minessale
> Sent: Monday, February 15, 2010 3:04 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Broadsoft SCA w/ Cisco SPA-5xx series
>
>
>
> you are missing something because you have no seize events when you go on
> and off hook.
> is every phone in the correct mode?
>
> On Mon, Feb 15, 2010 at 2:53 PM, Gabriel Kuri <gkuri at ieee.org> wrote:
>
> No, that was a typo. I have it correct in the config file.
>
> Gabe
>
> On Mon, Feb 15, 2010 at 12:34 PM, Peder <peder at networkoblivion.com> wrote:
>
>> Is this a typo "managed-shared-appeareance=true" or is there an extra e in
>> appearance in your config?
>>
>> -----Original Message-----
>> From: freeswitch-users-bounces at lists.freeswitch.org
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
>> Gabriel
>> Kuri
>> Sent: Monday, February 15, 2010 1:48 PM
>> To: freeswitch-users at lists.freeswitch.org
>> Subject: Re: [Freeswitch-users] Broadsoft SCA w/ Cisco SPA-5xx series
>>
>> OK, I don't know what happened there, here's another one with the
>> NOTIFYs. I'm on trunk rev 16633 and I have
>> "managed-shared-appeareance=true" on the internal profile. I'm just
>> making calls between internal phones.
>>
>> ? ? http://pastebin.freeswitch.org/12153
>>
>> Thanks,
>> Gabe
>>
>> On Mon, Feb 15, 2010 at 11:04 AM, Anthony Minessale
>> <anthony.minessale at gmail.com> wrote:
>>> I don't see any notifies at all in this trace do the profiles in question
>>> have:
>>> manage-shared-appearance set to true?
>>> and are you on latest trunk?
>>>
>>>
>>> On Mon, Feb 15, 2010 at 12:55 PM, Anthony Minessale
>>> <anthony.minessale at gmail.com> wrote:
>>>>
>>>> we log the sql stmts on err so they are red and easier to read.
>>>>
>>>>
>>>>
>>>> On Mon, Feb 15, 2010 at 12:48 PM, Gabriel Kuri <gkuri at ieee.org> wrote:
>>>>>
>>>>> Here's a call from ext 2551 to 2552 (two phones). I noticed a bunch of
>>>>> errors related to SQL UPDATE for presence ...
>>>>>
>>>>> ? ? http://pastebin.freeswitch.org/12152
>>>>>
>>>>> Thanks,
>>>>> Gabe
>>>>>
>>>>>
>>>>> On Mon, Feb 15, 2010 at 8:32 AM, Anthony Minessale
>>>>> <anthony.minessale at gmail.com> wrote:
>>>>> > it should be active not seized.
>>>>> > seized is when you take it off hook.
>>>>> >
>>>>> > We need some more debugging to be sure.
>>>>> > Can we work in real time on it or can you get a more detailed log?
>>>>> >
>>>>> > edit sofia.conf.xml and add the param to the "settings" section.
>>>>> >
>>>>> > <param name="debug-sla" value="100"/>
>>>>> >
>>>>> >
>>>>> > then restart and enable sip trace and debug level
>>>>> >
>>>>> > //do this for every profile involved in the call.
>>>>> > sofia profile <profilename> siptrace on
>>>>> >
>>>>> > //also do this
>>>>> > console loglevel debug
>>>>> >
>>>>> >
>>>>> > if you can let us ssh, we can do all the for you if you can make the
>>>>> > test
>>>>> > calls.
>>>>> >
>>>>> >
>>>>> >
>>>>> >
>>>>> > On Sun, Feb 14, 2010 at 11:59 PM, Gabriel Kuri <gkuri at ieee.org>
>>>>> > wrote:
>>>>> >>
>>>>> >> BTW, here's a copy of the NOTIFY (event call-info) sent to the other
>>>>> >> phone after the first phone is answered, should this have a
>>>>> >> Call-Info
>>>>> >> line with an "appearance-state=seized" to turn on the light on the
>>>>> >> other phone?
>>>>> >>
>>>>> >>
>>>>> >> NOTIFY sip:2551@<removed>:54446 SIP/2.0.
>>>>> >> Via: SIP/2.0/UDP <removed>:9430;rport;branch=z9hG4bK71pN2cXgH851K.
>>>>> >> Max-Forwards: 70.
>>>>> >> From: <sip:2551 at mydomain.com>;tag=XeB6ZrKDevpHp.
>>>>> >> To: <sip:2551 at mydomain.com>;tag=c2d34993aac6ea.
>>>>> >> Call-ID: 34c34987-8b6fa786@<removed>.
>>>>> >> CSeq: 126950830 NOTIFY.
>>>>> >> Contact: <sip:2551@<removed>:9430>.
>>>>> >> Expires: 3959.
>>>>> >> Call-Info:
>>>>> >> <sip:mydomain.com>;appearance-index=*;appearance-state=idle.
>>>>> >> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16633M.
>>>>> >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>>>>> >> REGISTER, REFER, PRACK, NOTIFY, PUBLISH, SUBSCRIBE.
>>>>> >> Supported: 100rel, timer, precondition, path, replaces.
>>>>> >> Event: call-info.
>>>>> >> Allow-Events: talk, presence, dialog, line-seize, call-info, sla,
>>>>> >> include-session-description, presence.winfo, message-summary, refer.
>>>>> >> Subscription-State: active;expires=3959.
>>>>> >> Content-Length: 0.
>>>>> >>
>>>>> >>
>>>>> >>
>>>>> >> On Sun, Feb 14, 2010 at 7:58 PM, Brian West <brian at freeswitch.org>
>>>>> >> wrote:
>>>>> >> > Works fine here... is your box slow or something?
>>>>> >> >
>>>>> >> > /b
>>>>> >> >
>>>>> >> > On Feb 14, 2010, at 1:03 PM, Gabriel Kuri wrote:
>>>>> >> >
>>>>> >> >> I followed Brian's directions from one of the previous threads on
>>>>> >> >> configuring the SPA-5xx series phones for Broadsoft SCA and set
>>>>> >> >> manage-shared-appearance=true in the internal profile. SCA
>>>>> >> >> appears
>>>>> >> >> to
>>>>> >> >> be working on outgoing calls between two phones, the line key
>>>>> >> >> starts
>>>>> >> >> flashing red on the second phone when the first phone picks up
>>>>> >> >> the
>>>>> >> >> receiver to make a call. However on incoming calls, both phones
>>>>> >> >> ring
>>>>> >> >> (same extension), however when one of the phones picks up the
>> line,
>>>>> >> >> the second phone's line key doesn't flash red or show the first
>>>>> >> >> phone
>>>>> >> >> on that incoming call. Any ideas? Does shared appearance only
>>>>> >> >> work
>>>>> >> >> on
>>>>> >> >> outgoing phone calls?
>>>>> >> >>
>>>>> >> >> Thanks,
>>>>> >> >> Gabe
>>>>> >> >
>>>>> >> >
>>>>> >> > _______________________________________________
>>>>> >> > FreeSWITCH-users mailing list
>>>>> >> > FreeSWITCH-users at lists.freeswitch.org
>>>>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> >> >
>>>>> >> >
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> >> > http://www.freeswitch.org
>>>>> >> >
>>>>> >>
>>>>> >> _______________________________________________
>>>>> >> FreeSWITCH-users mailing list
>>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> >>
>>>>> >>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> >> http://www.freeswitch.org
>>>>> >
>>>>> >
>>>>> >
>>>>> > --
>>>>> > Anthony Minessale II
>>>>> >
>>>>> > FreeSWITCH http://www.freeswitch.org/
>>>>> > ClueCon http://www.cluecon.com/
>>>>> > Twitter: http://twitter.com/FreeSWITCH_wire
>>>>> >
>>>>> > AIM: anthm
>>>>> > MSN:anthony_minessale at hotmail.com
>>>>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>>>> > IRC: irc.freenode.net #freeswitch
>>>>> >
>>>>> > FreeSWITCH Developer Conference
>>>>> > sip:888 at conference.freeswitch.org
>>>>> > iax:guest at conference.freeswitch.org/888
>>>>> > googletalk:conf+888 at conference.freeswitch.org
>>>>> > pstn:+19193869900
>>>>> >
>>>>> > _______________________________________________
>>>>> > FreeSWITCH-users mailing list
>>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> >
>>>>> >
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> > http://www.freeswitch.org
>>>>> >
>>>>> >
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>> --
>>>> Anthony Minessale II
>>>>
>>>> FreeSWITCH http://www.freeswitch.org/
>>>> ClueCon http://www.cluecon.com/
>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>
>>>> AIM: anthm
>>>> MSN:anthony_minessale at hotmail.com
>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>>> IRC: irc.freenode.net #freeswitch
>>>>
>>>> FreeSWITCH Developer Conference
>>>> sip:888 at conference.freeswitch.org
>>>> iax:guest at conference.freeswitch.org/888
>>>> googletalk:conf+888 at conference.freeswitch.org
>>>> pstn:+19193869900
>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org
>>> iax:guest at conference.freeswitch.org/888
>>> googletalk:conf+888 at conference.freeswitch.org
>>> pstn:+19193869900
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From brian at freeswitch.org  Mon Feb 15 14:56:42 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 15 Feb 2010 16:56:42 -0600
Subject: [Freeswitch-users] FreeSWITCH.Managed.dll deletes on make
	distclean
In-Reply-To: <201002152353.43666.errotan@gmail.com>
References: <201002152353.43666.errotan@gmail.com>
Message-ID: <D5940D2A-0A92-46E4-B83C-14FB8AF69D9D@freeswitch.org>

YES.

/b

On Feb 15, 2010, at 4:53 PM, Pusk?s Zsolt wrote:

> Is this normal ?




From brian at freeswitch.org  Mon Feb 15 14:57:15 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 15 Feb 2010 16:57:15 -0600
Subject: [Freeswitch-users] Broadsoft SCA w/ Cisco SPA-5xx series
In-Reply-To: <8b1c9cda1002151454m45afd322o9fd3bc9f0a6fa58f@mail.gmail.com>
References: <8b1c9cda1002141103te523643ob16db25e3bfa877f@mail.gmail.com>
	<8b1c9cda1002142159h5efcc765q9a41baa1bf694c8c@mail.gmail.com>
	<191c3a031002150832x7faa9a78xbc160cde63f09d09@mail.gmail.com>
	<8b1c9cda1002151048o4b45c034qb22732a9366024ae@mail.gmail.com>
	<191c3a031002151055v1938ab91p89232341f4087c8e@mail.gmail.com>
	<191c3a031002151104u7413b17dh84e68ab1cf703cb9@mail.gmail.com>
	<8b1c9cda1002151147l1e072d00y69c0022e3924c3b4@mail.gmail.com>
	<0dd701caae7e$3c108b70$b431a250$@com>
	<8b1c9cda1002151253t32c22663wf66245e0220a4e6@mail.gmail.com>
	<191c3a031002151304h54c262c5o8ffd5c195e0512d2@mail.gmail.com>
	<8b1c9cda1002151454m45afd322o9fd3bc9f0a6fa58f@mail.gmail.com>
Message-ID: <AC324FC3-2298-4765-887C-451F6F17E951@freeswitch.org>

You'll see it when you life the handset... thats the going to happen no matter what.. are you lifting the handset at all?

/b

On Feb 15, 2010, at 4:54 PM, Gabriel Kuri wrote:

> The phone I am dialing from (ext 2552) is an SPA-942 and I do not have
> it configured for SCA. I only have the two phones than I am calling
> (ext 2551) configured for SCA, so that's probably why you're not
> seeing the seize events from the SPA-942.
> 
> Thanks,
> Gabe




From gorand at donevtechconsulting.com  Mon Feb 15 15:12:41 2010
From: gorand at donevtechconsulting.com (Goran Donev)
Date: Mon, 15 Feb 2010 17:12:41 -0600
Subject: [Freeswitch-users] Version 1.05 release
In-Reply-To: <mailman.6410.1266268685.2495.freeswitch-users@lists.freeswitch.org>
References: <mailman.6410.1266268685.2495.freeswitch-users@lists.freeswitch.org>
Message-ID: <054c01caae94$6131b1c0$23951540$@com>

I really didn't get a definitive answer on when 1.05 is slated to be
released. We are running into some issues that I hope that 1.05 fixes. Do we
have an eta?

Thx




From brian at freeswitch.org  Mon Feb 15 15:17:26 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 15 Feb 2010 17:17:26 -0600
Subject: [Freeswitch-users] Version 1.05 release
In-Reply-To: <054c01caae94$6131b1c0$23951540$@com>
References: <mailman.6410.1266268685.2495.freeswitch-users@lists.freeswitch.org>
	<054c01caae94$6131b1c0$23951540$@com>
Message-ID: <DA13E7EA-D100-4A19-8477-25D5B57A1ECB@freeswitch.org>

Well if you're running into issues you should have already updated to SVN trunk.  1.0.5 is marching to release.  You're better off being on SVN trunk right now anyway. 

/b

On Feb 15, 2010, at 5:12 PM, Goran Donev wrote:

> I really didn't get a definitive answer on when 1.05 is slated to be
> released. We are running into some issues that I hope that 1.05 fixes. Do we
> have an eta?
> 
> Thx
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From brian at freeswitch.org  Mon Feb 15 15:22:04 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 15 Feb 2010 17:22:04 -0600
Subject: [Freeswitch-users] Broadsoft SCA w/ Cisco SPA-5xx series
In-Reply-To: <8b1c9cda1002151252jbc500bax3f137a882171485f@mail.gmail.com>
References: <8b1c9cda1002141103te523643ob16db25e3bfa877f@mail.gmail.com>
	<0D67132C-69D8-43AD-B7E6-C7545B544322@freeswitch.org>
	<8b1c9cda1002142159h5efcc765q9a41baa1bf694c8c@mail.gmail.com>
	<191c3a031002150832x7faa9a78xbc160cde63f09d09@mail.gmail.com>
	<8b1c9cda1002151048o4b45c034qb22732a9366024ae@mail.gmail.com>
	<191c3a031002151055v1938ab91p89232341f4087c8e@mail.gmail.com>
	<191c3a031002151104u7413b17dh84e68ab1cf703cb9@mail.gmail.com>
	<8b1c9cda1002151147l1e072d00y69c0022e3924c3b4@mail.gmail.com>
	<7C6C0110-C6AE-45CA-A1B4-413E39223286@freeswitch.org>
	<8b1c9cda1002151252jbc500bax3f137a882171485f@mail.gmail.com>
Message-ID: <D62B55B2-5DE9-4585-A533-0F77593FA125@freeswitch.org>

Can you try something for me?

Connect to your sofia db and dump its contents and put it on pastebin please.

/b
On Feb 15, 2010, at 2:52 PM, Gabriel Kuri wrote:

> Yes, that is correct. FS is behind a NAT and the phones behind another NAT.
> 
> I have ext-rtp-ip and ext-sip-ip set to the public IP address. Phones
> calls and everything else seem to be working.
> 
> Thanks,
> Gabe




From michal.kalinowski at interia.pl  Mon Feb 15 15:25:32 2010
From: michal.kalinowski at interia.pl (michal kalinowski)
Date: Tue, 16 Feb 2010 00:25:32 +0100
Subject: [Freeswitch-users] ivr from mysql
In-Reply-To: <0FD30AAA-C2E2-4228-A09B-DB09F4E26811@avgs.ca>
References: <7c74f5761002150824i41a2ed5did008f36cbb4499a4@mail.gmail.com>
	<87f2f3b91002151002v6bbc3e7fgb67dd6d3e72a915@mail.gmail.com>
	<7c74f5761002151425g512fa685n13c58d8f4b894740@mail.gmail.com>
	<0FD30AAA-C2E2-4228-A09B-DB09F4E26811@avgs.ca>
Message-ID: <7c74f5761002151525p7cc690ces632d46c6ab5e236c@mail.gmail.com>

Mathieu,

Could you insert several examples here?

BR,
Micha?

W dniu 15 lutego 2010 23:32 u?ytkownik Mathieu Rene
<mrene_lists at avgs.ca> napisa?:
> You need to examine the parameters passed into the xml_fetch callback.
> The request will be in the configuration section for "ivr.conf" and
> its parameters will tell you which ivr should be loaded.
>
> Also note that you have to return all the submenus at the same time
> (this is also valid for xml_curl) as freeswitch loads all of them at
> the same time to optimize the amount of times it does queries.
>
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>
>
>
>
> On 15-Feb-10, at 5:25 PM, michal kalinowski wrote:
>
>> I'm trying use in python something like this:
>>
>> from freeswitch import *
>>
>> def xml_fetch( param1, param2 ):
>> ? ? ? ?xml = '''
>> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
>> <document type="freeswitch/xml">
>> ?<section name="dialplan" description="RE Dial Plan For FreeSwitch">
>> ? ?<context name="default">
>> ? ? ?<extension name="generated">
>> ? ? ? ?<condition field="destination_number" expression="^9992$">>
>> ? ? ? ? <action application="answer"/>
>> ? ? ? ? <action application="playback" data="${hold_music}"/>
>> ? ? ? ?</condition>
>> ? ? ?</extension>
>> ? ?</context>
>> ?</section>
>> </document>
>> '''
>> ? ? ? ?return xml
>>
>> Of course XML context is with ivr parameters.
>>
>> So I will try mod_xml_curl in my configuration.
>>
>> BR,
>> Micha?
>>
>>
>> W dniu 15 lutego 2010 19:02 u?ytkownik Michael Collins
>> <msc at freeswitch.org> napisa?:
>>> When you say "xml from python" what exactly do you mean? Are you
>>> trying to
>>> use mod_xml_curl? If not you might want to check it out. The other
>>> choice is
>>> to use Lua from the dialplan, although I have a gut feeling that
>>> mod_xml_curl might be better for you.
>>> -MC
>>>
>>> 2010/2/15 michal kalinowski <michal.kalinowski at interia.pl>
>>>>
>>>> Hi,
>>>>
>>>> I need build ivr script/aplication which will take dynamically
>>>> configuration from mysql db (ivr menu, prompts, etc.).
>>>> My first idea is generate xml from python script. But it's not
>>>> working
>>>> properly.
>>>>
>>>> Anybody has some idea or have this aplication already done ?
>>>>
>>>> BR,
>>>> Micha?
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From brian at freeswitch.org  Mon Feb 15 15:29:27 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 15 Feb 2010 17:29:27 -0600
Subject: [Freeswitch-users] ivr from mysql
In-Reply-To: <7c74f5761002151525p7cc690ces632d46c6ab5e236c@mail.gmail.com>
References: <7c74f5761002150824i41a2ed5did008f36cbb4499a4@mail.gmail.com>
	<87f2f3b91002151002v6bbc3e7fgb67dd6d3e72a915@mail.gmail.com>
	<7c74f5761002151425g512fa685n13c58d8f4b894740@mail.gmail.com>
	<0FD30AAA-C2E2-4228-A09B-DB09F4E26811@avgs.ca>
	<7c74f5761002151525p7cc690ces632d46c6ab5e236c@mail.gmail.com>
Message-ID: <F2D49DD4-79DC-4F52-A1C1-4C48C79B4637@freeswitch.org>

Examples?  like we need more of those :)  where is the fun in that?

/b
PS: yes we need more docs... and code samples... anyone willing to help out?
On Feb 15, 2010, at 5:25 PM, michal kalinowski wrote:

> Mathieu,
> 
> Could you insert several examples here?
> 
> BR,
> Micha?




From anthony.minessale at gmail.com  Mon Feb 15 15:31:04 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 15 Feb 2010 17:31:04 -0600
Subject: [Freeswitch-users] Broadsoft SCA w/ Cisco SPA-5xx series
In-Reply-To: <8b1c9cda1002151456s5dec6dbbtaa2063e0f70fe5dd@mail.gmail.com>
References: <8b1c9cda1002141103te523643ob16db25e3bfa877f@mail.gmail.com>
	<8b1c9cda1002151048o4b45c034qb22732a9366024ae@mail.gmail.com>
	<191c3a031002151055v1938ab91p89232341f4087c8e@mail.gmail.com>
	<191c3a031002151104u7413b17dh84e68ab1cf703cb9@mail.gmail.com>
	<8b1c9cda1002151147l1e072d00y69c0022e3924c3b4@mail.gmail.com>
	<0dd701caae7e$3c108b70$b431a250$@com>
	<8b1c9cda1002151253t32c22663wf66245e0220a4e6@mail.gmail.com>
	<191c3a031002151304h54c262c5o8ffd5c195e0512d2@mail.gmail.com>
	<0e7301caae84$58b47ce0$0a1d76a0$@com>
	<8b1c9cda1002151456s5dec6dbbtaa2063e0f70fe5dd@mail.gmail.com>
Message-ID: <191c3a031002151531k148d8504w15e6f4c6c1e3ccd@mail.gmail.com>

Do the domain names match on what the remote phones are using?

When the call is active, can you attach to sqlite with the sqlite3 app and
select * from sip_dialogs

sqlite3 /usr/local/freeswitch/db/sofia_reg_internal.db
> select * from sip_dialogs;

remember to do it while the call is up.


I am going to bet the domain name in that table is not the same as your
actual domain.


I tried to make this easier by asking to ssh to your box and work with you
to fix it but now 9 hours later its starting to resemble diffusing a bomb
over a telegraph wire.




On Mon, Feb 15, 2010 at 4:56 PM, Gabriel Kuri <gkuri at ieee.org> wrote:

> Yes, the two phones being called (SPA-509Gs) have SCA enabled and it
> works when making outgoing calls from those phones. But incoming calls
> to those two phones don't seem to have the line key light up on the
> other phone when one of the phones is answered (same extension).
>
> Thanks,
> Gabe
>
> On Mon, Feb 15, 2010 at 1:17 PM, Peder <peder at networkoblivion.com> wrote:
> > On the phone itself, do you have the line set to shared and ?Broadsoft
> SCA?
> > enabled?
> >
> >
> >
> > From: freeswitch-users-bounces at lists.freeswitch.org
> > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Anthony
> > Minessale
> > Sent: Monday, February 15, 2010 3:04 PM
> > To: freeswitch-users at lists.freeswitch.org
> > Subject: Re: [Freeswitch-users] Broadsoft SCA w/ Cisco SPA-5xx series
> >
> >
> >
> > you are missing something because you have no seize events when you go on
> > and off hook.
> > is every phone in the correct mode?
> >
> > On Mon, Feb 15, 2010 at 2:53 PM, Gabriel Kuri <gkuri at ieee.org> wrote:
> >
> > No, that was a typo. I have it correct in the config file.
> >
> > Gabe
> >
> > On Mon, Feb 15, 2010 at 12:34 PM, Peder <peder at networkoblivion.com>
> wrote:
> >
> >> Is this a typo "managed-shared-appeareance=true" or is there an extra e
> in
> >> appearance in your config?
> >>
> >> -----Original Message-----
> >> From: freeswitch-users-bounces at lists.freeswitch.org
> >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> >> Gabriel
> >> Kuri
> >> Sent: Monday, February 15, 2010 1:48 PM
> >> To: freeswitch-users at lists.freeswitch.org
> >> Subject: Re: [Freeswitch-users] Broadsoft SCA w/ Cisco SPA-5xx series
> >>
> >> OK, I don't know what happened there, here's another one with the
> >> NOTIFYs. I'm on trunk rev 16633 and I have
> >> "managed-shared-appeareance=true" on the internal profile. I'm just
> >> making calls between internal phones.
> >>
> >>     http://pastebin.freeswitch.org/12153
> >>
> >> Thanks,
> >> Gabe
> >>
> >> On Mon, Feb 15, 2010 at 11:04 AM, Anthony Minessale
> >> <anthony.minessale at gmail.com> wrote:
> >>> I don't see any notifies at all in this trace do the profiles in
> question
> >>> have:
> >>> manage-shared-appearance set to true?
> >>> and are you on latest trunk?
> >>>
> >>>
> >>> On Mon, Feb 15, 2010 at 12:55 PM, Anthony Minessale
> >>> <anthony.minessale at gmail.com> wrote:
> >>>>
> >>>> we log the sql stmts on err so they are red and easier to read.
> >>>>
> >>>>
> >>>>
> >>>> On Mon, Feb 15, 2010 at 12:48 PM, Gabriel Kuri <gkuri at ieee.org>
> wrote:
> >>>>>
> >>>>> Here's a call from ext 2551 to 2552 (two phones). I noticed a bunch
> of
> >>>>> errors related to SQL UPDATE for presence ...
> >>>>>
> >>>>>     http://pastebin.freeswitch.org/12152
> >>>>>
> >>>>> Thanks,
> >>>>> Gabe
> >>>>>
> >>>>>
> >>>>> On Mon, Feb 15, 2010 at 8:32 AM, Anthony Minessale
> >>>>> <anthony.minessale at gmail.com> wrote:
> >>>>> > it should be active not seized.
> >>>>> > seized is when you take it off hook.
> >>>>> >
> >>>>> > We need some more debugging to be sure.
> >>>>> > Can we work in real time on it or can you get a more detailed log?
> >>>>> >
> >>>>> > edit sofia.conf.xml and add the param to the "settings" section.
> >>>>> >
> >>>>> > <param name="debug-sla" value="100"/>
> >>>>> >
> >>>>> >
> >>>>> > then restart and enable sip trace and debug level
> >>>>> >
> >>>>> > //do this for every profile involved in the call.
> >>>>> > sofia profile <profilename> siptrace on
> >>>>> >
> >>>>> > //also do this
> >>>>> > console loglevel debug
> >>>>> >
> >>>>> >
> >>>>> > if you can let us ssh, we can do all the for you if you can make
> the
> >>>>> > test
> >>>>> > calls.
> >>>>> >
> >>>>> >
> >>>>> >
> >>>>> >
> >>>>> > On Sun, Feb 14, 2010 at 11:59 PM, Gabriel Kuri <gkuri at ieee.org>
> >>>>> > wrote:
> >>>>> >>
> >>>>> >> BTW, here's a copy of the NOTIFY (event call-info) sent to the
> other
> >>>>> >> phone after the first phone is answered, should this have a
> >>>>> >> Call-Info
> >>>>> >> line with an "appearance-state=seized" to turn on the light on the
> >>>>> >> other phone?
> >>>>> >>
> >>>>> >>
> >>>>> >> NOTIFY sip:2551@<removed>:54446 SIP/2.0.
> >>>>> >> Via: SIP/2.0/UDP <removed>:9430;rport;branch=z9hG4bK71pN2cXgH851K.
> >>>>> >> Max-Forwards: 70.
> >>>>> >> From: <sip:2551 at mydomain.com <sip%3A2551 at mydomain.com>
> >;tag=XeB6ZrKDevpHp.
> >>>>> >> To: <sip:2551 at mydomain.com <sip%3A2551 at mydomain.com>
> >;tag=c2d34993aac6ea.
> >>>>> >> Call-ID: 34c34987-8b6fa786@<removed>.
> >>>>> >> CSeq: 126950830 NOTIFY.
> >>>>> >> Contact: <sip:2551@<removed>:9430>.
> >>>>> >> Expires: 3959.
> >>>>> >> Call-Info:
> >>>>> >> <sip:mydomain.com>;appearance-index=*;appearance-state=idle.
> >>>>> >> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16633M.
> >>>>> >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> >>>>> >> REGISTER, REFER, PRACK, NOTIFY, PUBLISH, SUBSCRIBE.
> >>>>> >> Supported: 100rel, timer, precondition, path, replaces.
> >>>>> >> Event: call-info.
> >>>>> >> Allow-Events: talk, presence, dialog, line-seize, call-info, sla,
> >>>>> >> include-session-description, presence.winfo, message-summary,
> refer.
> >>>>> >> Subscription-State: active;expires=3959.
> >>>>> >> Content-Length: 0.
> >>>>> >>
> >>>>> >>
> >>>>> >>
> >>>>> >> On Sun, Feb 14, 2010 at 7:58 PM, Brian West <brian at freeswitch.org
> >
> >>>>> >> wrote:
> >>>>> >> > Works fine here... is your box slow or something?
> >>>>> >> >
> >>>>> >> > /b
> >>>>> >> >
> >>>>> >> > On Feb 14, 2010, at 1:03 PM, Gabriel Kuri wrote:
> >>>>> >> >
> >>>>> >> >> I followed Brian's directions from one of the previous threads
> on
> >>>>> >> >> configuring the SPA-5xx series phones for Broadsoft SCA and set
> >>>>> >> >> manage-shared-appearance=true in the internal profile. SCA
> >>>>> >> >> appears
> >>>>> >> >> to
> >>>>> >> >> be working on outgoing calls between two phones, the line key
> >>>>> >> >> starts
> >>>>> >> >> flashing red on the second phone when the first phone picks up
> >>>>> >> >> the
> >>>>> >> >> receiver to make a call. However on incoming calls, both phones
> >>>>> >> >> ring
> >>>>> >> >> (same extension), however when one of the phones picks up the
> >> line,
> >>>>> >> >> the second phone's line key doesn't flash red or show the first
> >>>>> >> >> phone
> >>>>> >> >> on that incoming call. Any ideas? Does shared appearance only
> >>>>> >> >> work
> >>>>> >> >> on
> >>>>> >> >> outgoing phone calls?
> >>>>> >> >>
> >>>>> >> >> Thanks,
> >>>>> >> >> Gabe
> >>>>> >> >
> >>>>> >> >
> >>>>> >> > _______________________________________________
> >>>>> >> > FreeSWITCH-users mailing list
> >>>>> >> > FreeSWITCH-users at lists.freeswitch.org
> >>>>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>> >> >
> >>>>> >> >
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>> >> > http://www.freeswitch.org
> >>>>> >> >
> >>>>> >>
> >>>>> >> _______________________________________________
> >>>>> >> FreeSWITCH-users mailing list
> >>>>> >> FreeSWITCH-users at lists.freeswitch.org
> >>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>> >>
> >>>>> >>
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>> >> http://www.freeswitch.org
> >>>>> >
> >>>>> >
> >>>>> >
> >>>>> > --
> >>>>> > Anthony Minessale II
> >>>>> >
> >>>>> > FreeSWITCH http://www.freeswitch.org/
> >>>>> > ClueCon http://www.cluecon.com/
> >>>>> > Twitter: http://twitter.com/FreeSWITCH_wire
> >>>>> >
> >>>>> > AIM: anthm
> >>>>> > MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
> >>>>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >>>>> > IRC: irc.freenode.net #freeswitch
> >>>>> >
> >>>>> > FreeSWITCH Developer Conference
> >>>>> > sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> >>>>> > iax:guest at conference.freeswitch.org/888
> >>>>> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >>>>> > pstn:+19193869900
> >>>>> >
> >>>>> > _______________________________________________
> >>>>> > FreeSWITCH-users mailing list
> >>>>> > FreeSWITCH-users at lists.freeswitch.org
> >>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>> >
> >>>>> >
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>> > http://www.freeswitch.org
> >>>>> >
> >>>>> >
> >>>>>
> >>>>> _______________________________________________
> >>>>> FreeSWITCH-users mailing list
> >>>>> FreeSWITCH-users at lists.freeswitch.org
> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>>
> >>>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>> http://www.freeswitch.org
> >>>>
> >>>>
> >>>>
> >>>> --
> >>>> Anthony Minessale II
> >>>>
> >>>> FreeSWITCH http://www.freeswitch.org/
> >>>> ClueCon http://www.cluecon.com/
> >>>> Twitter: http://twitter.com/FreeSWITCH_wire
> >>>>
> >>>> AIM: anthm
> >>>> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
> >>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >>>> IRC: irc.freenode.net #freeswitch
> >>>>
> >>>> FreeSWITCH Developer Conference
> >>>> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> >>>> iax:guest at conference.freeswitch.org/888
> >>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >>>> pstn:+19193869900
> >>>
> >>>
> >>>
> >>> --
> >>> Anthony Minessale II
> >>>
> >>> FreeSWITCH http://www.freeswitch.org/
> >>> ClueCon http://www.cluecon.com/
> >>> Twitter: http://twitter.com/FreeSWITCH_wire
> >>>
> >>> AIM: anthm
> >>> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
> >>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >>> IRC: irc.freenode.net #freeswitch
> >>>
> >>> FreeSWITCH Developer Conference
> >>> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> >>> iax:guest at conference.freeswitch.org/888
> >>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >>> pstn:+19193869900
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>>
> >>>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> > Twitter: http://twitter.com/FreeSWITCH_wire
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> > IRC: irc.freenode.net #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> > iax:guest at conference.freeswitch.org/888
> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > pstn:+19193869900
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100215/122fa7df/attachment-0002.html 

From anthony.minessale at gmail.com  Mon Feb 15 15:43:03 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 15 Feb 2010 17:43:03 -0600
Subject: [Freeswitch-users] Version 1.05 release
In-Reply-To: <054c01caae94$6131b1c0$23951540$@com>
References: <mailman.6410.1266268685.2495.freeswitch-users@lists.freeswitch.org>
	<054c01caae94$6131b1c0$23951540$@com>
Message-ID: <191c3a031002151543q6a00a936vf7a5e9d4b69d0ebc@mail.gmail.com>

Since you are advertising services on your website most likely all provided
by our free software, I hope you can learn some patience while you wait for
us to provide you with your next release to sell to your customers.

The release will be as soon as we make sure all the bugs are fixed, unless
you prefer it to be buggy.

BTW,

If your issues are still present in 1.0.5 and it's because you never
reported them, then we will, of course, be very unhappy.

If this is unacceptable we do offer a triple-your-money-back guarantee that
we will upgrade to quadruple if you act now.


On Mon, Feb 15, 2010 at 5:12 PM, Goran Donev <gorand at donevtechconsulting.com
> wrote:

> I really didn't get a definitive answer on when 1.05 is slated to be
> released. We are running into some issues that I hope that 1.05 fixes. Do
> we
> have an eta?
>
> Thx
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100215/3da9decd/attachment-0002.html 

From rupa at rupa.com  Mon Feb 15 16:00:51 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Mon, 15 Feb 2010 18:00:51 -0600
Subject: [Freeswitch-users] FreeSWITCH Debian packages - build issues
In-Reply-To: <201002152257.28593.errotan@gmail.com>
References: <20100213030812.GA19108@jdc.jasonjgw.net>
	<9CF8766D-2367-4275-ACCF-0F9C6ABC7A09@jerris.com>
	<20100215064616.GA32700@jdc.jasonjgw.net>
	<201002152257.28593.errotan@gmail.com>
Message-ID: <b39273531002151600l5c5620eapd2eb745d7bb3bbc0@mail.gmail.com>

I made a minor change to mod_memcache's Makefile -- can you try re-enabling
mod_memcache in your debian build and see if it builds for you?

On Mon, Feb 15, 2010 at 3:57 PM, Pusk?s Zsolt <errotan at gmail.com> wrote:

> 2010. febru?r 15. 07.46.16 Jason White d?tummal ezt ?rta:
> > Michael Jerris <mike at jerris.com> wrote:
> > > Did anyone bother opening a bug on jira for this or are we going to
> just
> > > tag 1.0.5 without deb packages?
> >
> > Has anyone tried building these on Ubuntu 9.10 or Debian 5.0? I'm not in
> a
> > position to do so at the moment.
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> Just done with dpkg-buildpackage on Debian 5.0 "lenny" on x86.
> It built without errors and everything looks ok and all modules exists, but
> haven't tried to install or run it.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
-Rupa
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100215/63b31383/attachment-0002.html 

From jason at jasonjgw.net  Mon Feb 15 18:09:34 2010
From: jason at jasonjgw.net (Jason White)
Date: Tue, 16 Feb 2010 13:09:34 +1100
Subject: [Freeswitch-users] FreeSWITCH Debian packages - build issues
In-Reply-To: <b39273531002151600l5c5620eapd2eb745d7bb3bbc0@mail.gmail.com>
References: <20100213030812.GA19108@jdc.jasonjgw.net>
	<9CF8766D-2367-4275-ACCF-0F9C6ABC7A09@jerris.com>
	<20100215064616.GA32700@jdc.jasonjgw.net>
	<201002152257.28593.errotan@gmail.com>
	<b39273531002151600l5c5620eapd2eb745d7bb3bbc0@mail.gmail.com>
Message-ID: <20100216020934.GA10717@jdc.jasonjgw.net>

Rupa Schomaker <rupa at rupa.com> wrote:
> I made a minor change to mod_memcache's Makefile -- can you try re-enabling
> mod_memcache in your debian build and see if it builds for you?

It built this time:

-rw-r----- 1 freeswitch daemon 73552 Feb 16 02:02 mod_memcache.so




From gkuri at ieee.org  Mon Feb 15 22:20:59 2010
From: gkuri at ieee.org (Gabriel Kuri)
Date: Mon, 15 Feb 2010 22:20:59 -0800
Subject: [Freeswitch-users] Broadsoft SCA w/ Cisco SPA-5xx series
In-Reply-To: <D62B55B2-5DE9-4585-A533-0F77593FA125@freeswitch.org>
References: <8b1c9cda1002141103te523643ob16db25e3bfa877f@mail.gmail.com>
	<8b1c9cda1002142159h5efcc765q9a41baa1bf694c8c@mail.gmail.com>
	<191c3a031002150832x7faa9a78xbc160cde63f09d09@mail.gmail.com>
	<8b1c9cda1002151048o4b45c034qb22732a9366024ae@mail.gmail.com>
	<191c3a031002151055v1938ab91p89232341f4087c8e@mail.gmail.com>
	<191c3a031002151104u7413b17dh84e68ab1cf703cb9@mail.gmail.com>
	<8b1c9cda1002151147l1e072d00y69c0022e3924c3b4@mail.gmail.com>
	<7C6C0110-C6AE-45CA-A1B4-413E39223286@freeswitch.org>
	<8b1c9cda1002151252jbc500bax3f137a882171485f@mail.gmail.com>
	<D62B55B2-5DE9-4585-A533-0F77593FA125@freeswitch.org>
Message-ID: <8b1c9cda1002152220i710d7174n9ad7cdaa22c6452@mail.gmail.com>

Here's the database dump ...

     http://pastebin.freeswitch.org/12158

Thanks,
Gabe


On Mon, Feb 15, 2010 at 3:22 PM, Brian West <brian at freeswitch.org> wrote:
> Can you try something for me?
>
> Connect to your sofia db and dump its contents and put it on pastebin please.
>
> /b
> On Feb 15, 2010, at 2:52 PM, Gabriel Kuri wrote:
>
>> Yes, that is correct. FS is behind a NAT and the phones behind another NAT.
>>
>> I have ext-rtp-ip and ext-sip-ip set to the public IP address. Phones
>> calls and everything else seem to be working.
>>
>> Thanks,
>> Gabe
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From gkuri at ieee.org  Mon Feb 15 22:25:56 2010
From: gkuri at ieee.org (Gabriel Kuri)
Date: Mon, 15 Feb 2010 22:25:56 -0800
Subject: [Freeswitch-users] Broadsoft SCA w/ Cisco SPA-5xx series
In-Reply-To: <191c3a031002151531k148d8504w15e6f4c6c1e3ccd@mail.gmail.com>
References: <8b1c9cda1002141103te523643ob16db25e3bfa877f@mail.gmail.com>
	<191c3a031002151055v1938ab91p89232341f4087c8e@mail.gmail.com>
	<191c3a031002151104u7413b17dh84e68ab1cf703cb9@mail.gmail.com>
	<8b1c9cda1002151147l1e072d00y69c0022e3924c3b4@mail.gmail.com>
	<0dd701caae7e$3c108b70$b431a250$@com>
	<8b1c9cda1002151253t32c22663wf66245e0220a4e6@mail.gmail.com>
	<191c3a031002151304h54c262c5o8ffd5c195e0512d2@mail.gmail.com>
	<0e7301caae84$58b47ce0$0a1d76a0$@com>
	<8b1c9cda1002151456s5dec6dbbtaa2063e0f70fe5dd@mail.gmail.com>
	<191c3a031002151531k148d8504w15e6f4c6c1e3ccd@mail.gmail.com>
Message-ID: <8b1c9cda1002152225i7700bdfcgd2c112bf8180e36@mail.gmail.com>

Yeah, the domain name matches on the internal profile.

Thanks for all your help, I can arrange ssh access tomorrow, today
just wasn't one of those good days to do so, I've been running in and
out too much to coordinate it.

Here's the pastebin for the sip_dialogs table while the call is up ...

     http://pastebin.freeswitch.org/12159

Thanks,
Gabe


On Mon, Feb 15, 2010 at 3:31 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> Do the domain names match on what the remote phones are using?
>
> When the call is active, can you attach to sqlite with the sqlite3 app and
> select * from sip_dialogs
>
> sqlite3 /usr/local/freeswitch/db/sofia_reg_internal.db
>> select * from sip_dialogs;
>
> remember to do it while the call is up.
>
>
> I am going to bet the domain name in that table is not the same as your
> actual domain.
>
>
> I tried to make this easier by asking to ssh to your box and work with you
> to fix it but now 9 hours later its starting to resemble diffusing a bomb
> over a telegraph wire.
>
>
>
>
> On Mon, Feb 15, 2010 at 4:56 PM, Gabriel Kuri <gkuri at ieee.org> wrote:
>>
>> Yes, the two phones being called (SPA-509Gs) have SCA enabled and it
>> works when making outgoing calls from those phones. But incoming calls
>> to those two phones don't seem to have the line key light up on the
>> other phone when one of the phones is answered (same extension).
>>
>> Thanks,
>> Gabe
>>
>> On Mon, Feb 15, 2010 at 1:17 PM, Peder <peder at networkoblivion.com> wrote:
>> > On the phone itself, do you have the line set to shared and ?Broadsoft
>> > SCA?
>> > enabled?
>> >
>> >
>> >
>> > From: freeswitch-users-bounces at lists.freeswitch.org
>> > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
>> > Anthony
>> > Minessale
>> > Sent: Monday, February 15, 2010 3:04 PM
>> > To: freeswitch-users at lists.freeswitch.org
>> > Subject: Re: [Freeswitch-users] Broadsoft SCA w/ Cisco SPA-5xx series
>> >
>> >
>> >
>> > you are missing something because you have no seize events when you go
>> > on
>> > and off hook.
>> > is every phone in the correct mode?
>> >
>> > On Mon, Feb 15, 2010 at 2:53 PM, Gabriel Kuri <gkuri at ieee.org> wrote:
>> >
>> > No, that was a typo. I have it correct in the config file.
>> >
>> > Gabe
>> >
>> > On Mon, Feb 15, 2010 at 12:34 PM, Peder <peder at networkoblivion.com>
>> > wrote:
>> >
>> >> Is this a typo "managed-shared-appeareance=true" or is there an extra e
>> >> in
>> >> appearance in your config?
>> >>
>> >> -----Original Message-----
>> >> From: freeswitch-users-bounces at lists.freeswitch.org
>> >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
>> >> Gabriel
>> >> Kuri
>> >> Sent: Monday, February 15, 2010 1:48 PM
>> >> To: freeswitch-users at lists.freeswitch.org
>> >> Subject: Re: [Freeswitch-users] Broadsoft SCA w/ Cisco SPA-5xx series
>> >>
>> >> OK, I don't know what happened there, here's another one with the
>> >> NOTIFYs. I'm on trunk rev 16633 and I have
>> >> "managed-shared-appeareance=true" on the internal profile. I'm just
>> >> making calls between internal phones.
>> >>
>> >> ? ? http://pastebin.freeswitch.org/12153
>> >>
>> >> Thanks,
>> >> Gabe
>> >>
>> >> On Mon, Feb 15, 2010 at 11:04 AM, Anthony Minessale
>> >> <anthony.minessale at gmail.com> wrote:
>> >>> I don't see any notifies at all in this trace do the profiles in
>> >>> question
>> >>> have:
>> >>> manage-shared-appearance set to true?
>> >>> and are you on latest trunk?
>> >>>
>> >>>
>> >>> On Mon, Feb 15, 2010 at 12:55 PM, Anthony Minessale
>> >>> <anthony.minessale at gmail.com> wrote:
>> >>>>
>> >>>> we log the sql stmts on err so they are red and easier to read.
>> >>>>
>> >>>>
>> >>>>
>> >>>> On Mon, Feb 15, 2010 at 12:48 PM, Gabriel Kuri <gkuri at ieee.org>
>> >>>> wrote:
>> >>>>>
>> >>>>> Here's a call from ext 2551 to 2552 (two phones). I noticed a bunch
>> >>>>> of
>> >>>>> errors related to SQL UPDATE for presence ...
>> >>>>>
>> >>>>> ? ? http://pastebin.freeswitch.org/12152
>> >>>>>
>> >>>>> Thanks,
>> >>>>> Gabe
>> >>>>>
>> >>>>>
>> >>>>> On Mon, Feb 15, 2010 at 8:32 AM, Anthony Minessale
>> >>>>> <anthony.minessale at gmail.com> wrote:
>> >>>>> > it should be active not seized.
>> >>>>> > seized is when you take it off hook.
>> >>>>> >
>> >>>>> > We need some more debugging to be sure.
>> >>>>> > Can we work in real time on it or can you get a more detailed log?
>> >>>>> >
>> >>>>> > edit sofia.conf.xml and add the param to the "settings" section.
>> >>>>> >
>> >>>>> > <param name="debug-sla" value="100"/>
>> >>>>> >
>> >>>>> >
>> >>>>> > then restart and enable sip trace and debug level
>> >>>>> >
>> >>>>> > //do this for every profile involved in the call.
>> >>>>> > sofia profile <profilename> siptrace on
>> >>>>> >
>> >>>>> > //also do this
>> >>>>> > console loglevel debug
>> >>>>> >
>> >>>>> >
>> >>>>> > if you can let us ssh, we can do all the for you if you can make
>> >>>>> > the
>> >>>>> > test
>> >>>>> > calls.
>> >>>>> >
>> >>>>> >
>> >>>>> >
>> >>>>> >
>> >>>>> > On Sun, Feb 14, 2010 at 11:59 PM, Gabriel Kuri <gkuri at ieee.org>
>> >>>>> > wrote:
>> >>>>> >>
>> >>>>> >> BTW, here's a copy of the NOTIFY (event call-info) sent to the
>> >>>>> >> other
>> >>>>> >> phone after the first phone is answered, should this have a
>> >>>>> >> Call-Info
>> >>>>> >> line with an "appearance-state=seized" to turn on the light on
>> >>>>> >> the
>> >>>>> >> other phone?
>> >>>>> >>
>> >>>>> >>
>> >>>>> >> NOTIFY sip:2551@<removed>:54446 SIP/2.0.
>> >>>>> >> Via: SIP/2.0/UDP
>> >>>>> >> <removed>:9430;rport;branch=z9hG4bK71pN2cXgH851K.
>> >>>>> >> Max-Forwards: 70.
>> >>>>> >> From: <sip:2551 at mydomain.com>;tag=XeB6ZrKDevpHp.
>> >>>>> >> To: <sip:2551 at mydomain.com>;tag=c2d34993aac6ea.
>> >>>>> >> Call-ID: 34c34987-8b6fa786@<removed>.
>> >>>>> >> CSeq: 126950830 NOTIFY.
>> >>>>> >> Contact: <sip:2551@<removed>:9430>.
>> >>>>> >> Expires: 3959.
>> >>>>> >> Call-Info:
>> >>>>> >> <sip:mydomain.com>;appearance-index=*;appearance-state=idle.
>> >>>>> >> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16633M.
>> >>>>> >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>> >>>>> >> REGISTER, REFER, PRACK, NOTIFY, PUBLISH, SUBSCRIBE.
>> >>>>> >> Supported: 100rel, timer, precondition, path, replaces.
>> >>>>> >> Event: call-info.
>> >>>>> >> Allow-Events: talk, presence, dialog, line-seize, call-info, sla,
>> >>>>> >> include-session-description, presence.winfo, message-summary,
>> >>>>> >> refer.
>> >>>>> >> Subscription-State: active;expires=3959.
>> >>>>> >> Content-Length: 0.
>> >>>>> >>
>> >>>>> >>
>> >>>>> >>
>> >>>>> >> On Sun, Feb 14, 2010 at 7:58 PM, Brian West
>> >>>>> >> <brian at freeswitch.org>
>> >>>>> >> wrote:
>> >>>>> >> > Works fine here... is your box slow or something?
>> >>>>> >> >
>> >>>>> >> > /b
>> >>>>> >> >
>> >>>>> >> > On Feb 14, 2010, at 1:03 PM, Gabriel Kuri wrote:
>> >>>>> >> >
>> >>>>> >> >> I followed Brian's directions from one of the previous threads
>> >>>>> >> >> on
>> >>>>> >> >> configuring the SPA-5xx series phones for Broadsoft SCA and
>> >>>>> >> >> set
>> >>>>> >> >> manage-shared-appearance=true in the internal profile. SCA
>> >>>>> >> >> appears
>> >>>>> >> >> to
>> >>>>> >> >> be working on outgoing calls between two phones, the line key
>> >>>>> >> >> starts
>> >>>>> >> >> flashing red on the second phone when the first phone picks up
>> >>>>> >> >> the
>> >>>>> >> >> receiver to make a call. However on incoming calls, both
>> >>>>> >> >> phones
>> >>>>> >> >> ring
>> >>>>> >> >> (same extension), however when one of the phones picks up the
>> >> line,
>> >>>>> >> >> the second phone's line key doesn't flash red or show the
>> >>>>> >> >> first
>> >>>>> >> >> phone
>> >>>>> >> >> on that incoming call. Any ideas? Does shared appearance only
>> >>>>> >> >> work
>> >>>>> >> >> on
>> >>>>> >> >> outgoing phone calls?
>> >>>>> >> >>
>> >>>>> >> >> Thanks,
>> >>>>> >> >> Gabe
>> >>>>> >> >
>> >>>>> >> >
>> >>>>> >> > _______________________________________________
>> >>>>> >> > FreeSWITCH-users mailing list
>> >>>>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >>>>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>>> >> >
>> >>>>> >> >
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>>>> >> > http://www.freeswitch.org
>> >>>>> >> >
>> >>>>> >>
>> >>>>> >> _______________________________________________
>> >>>>> >> FreeSWITCH-users mailing list
>> >>>>> >> FreeSWITCH-users at lists.freeswitch.org
>> >>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>>> >>
>> >>>>> >>
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>>>> >> http://www.freeswitch.org
>> >>>>> >
>> >>>>> >
>> >>>>> >
>> >>>>> > --
>> >>>>> > Anthony Minessale II
>> >>>>> >
>> >>>>> > FreeSWITCH http://www.freeswitch.org/
>> >>>>> > ClueCon http://www.cluecon.com/
>> >>>>> > Twitter: http://twitter.com/FreeSWITCH_wire
>> >>>>> >
>> >>>>> > AIM: anthm
>> >>>>> > MSN:anthony_minessale at hotmail.com
>> >>>>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> >>>>> > IRC: irc.freenode.net #freeswitch
>> >>>>> >
>> >>>>> > FreeSWITCH Developer Conference
>> >>>>> > sip:888 at conference.freeswitch.org
>> >>>>> > iax:guest at conference.freeswitch.org/888
>> >>>>> > googletalk:conf+888 at conference.freeswitch.org
>> >>>>> > pstn:+19193869900
>> >>>>> >
>> >>>>> > _______________________________________________
>> >>>>> > FreeSWITCH-users mailing list
>> >>>>> > FreeSWITCH-users at lists.freeswitch.org
>> >>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>>> >
>> >>>>> >
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>>>> > http://www.freeswitch.org
>> >>>>> >
>> >>>>> >
>> >>>>>
>> >>>>> _______________________________________________
>> >>>>> FreeSWITCH-users mailing list
>> >>>>> FreeSWITCH-users at lists.freeswitch.org
>> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>>>
>> >>>>>
>> >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>>>> http://www.freeswitch.org
>> >>>>
>> >>>>
>> >>>>
>> >>>> --
>> >>>> Anthony Minessale II
>> >>>>
>> >>>> FreeSWITCH http://www.freeswitch.org/
>> >>>> ClueCon http://www.cluecon.com/
>> >>>> Twitter: http://twitter.com/FreeSWITCH_wire
>> >>>>
>> >>>> AIM: anthm
>> >>>> MSN:anthony_minessale at hotmail.com
>> >>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> >>>> IRC: irc.freenode.net #freeswitch
>> >>>>
>> >>>> FreeSWITCH Developer Conference
>> >>>> sip:888 at conference.freeswitch.org
>> >>>> iax:guest at conference.freeswitch.org/888
>> >>>> googletalk:conf+888 at conference.freeswitch.org
>> >>>> pstn:+19193869900
>> >>>
>> >>>
>> >>>
>> >>> --
>> >>> Anthony Minessale II
>> >>>
>> >>> FreeSWITCH http://www.freeswitch.org/
>> >>> ClueCon http://www.cluecon.com/
>> >>> Twitter: http://twitter.com/FreeSWITCH_wire
>> >>>
>> >>> AIM: anthm
>> >>> MSN:anthony_minessale at hotmail.com
>> >>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> >>> IRC: irc.freenode.net #freeswitch
>> >>>
>> >>> FreeSWITCH Developer Conference
>> >>> sip:888 at conference.freeswitch.org
>> >>> iax:guest at conference.freeswitch.org/888
>> >>> googletalk:conf+888 at conference.freeswitch.org
>> >>> pstn:+19193869900
>> >>>
>> >>> _______________________________________________
>> >>> FreeSWITCH-users mailing list
>> >>> FreeSWITCH-users at lists.freeswitch.org
>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>
>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> http://www.freeswitch.org
>> >>>
>> >>>
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>> > --
>> > Anthony Minessale II
>> >
>> > FreeSWITCH http://www.freeswitch.org/
>> > ClueCon http://www.cluecon.com/
>> > Twitter: http://twitter.com/FreeSWITCH_wire
>> >
>> > AIM: anthm
>> > MSN:anthony_minessale at hotmail.com
>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> > IRC: irc.freenode.net #freeswitch
>> >
>> > FreeSWITCH Developer Conference
>> > sip:888 at conference.freeswitch.org
>> > iax:guest at conference.freeswitch.org/888
>> > googletalk:conf+888 at conference.freeswitch.org
>> > pstn:+19193869900
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From tculjaga at gmail.com  Tue Feb 16 00:10:30 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 16 Feb 2010 09:10:30 +0100
Subject: [Freeswitch-users] DTMF problem
In-Reply-To: <1265922753047-4557446.post@n2.nabble.com>
References: <89c9bbf81002111209v6e2f36b3x86a628c5e1390599@mail.gmail.com>
	<F960CC45-C9D2-4150-A108-C0E8B230284D@freeswitch.org>
	<1265922753047-4557446.post@n2.nabble.com>
Message-ID: <65d96fc81002160010j798353b4y5d38894e774b85e0@mail.gmail.com>

what DTMF method are you using ?

InBand as pure voice or by rfc2833?


if it is as pure voice than you will need this "<action
application="start_dtmf" />" in FS dialplan receiving the call.

If you are using 2833, than you should check your FS config for DTMF method.

T.


On Thu, Feb 11, 2010 at 10:12 PM, maxim.tsvetov <maxim.tsvetov at gmail.com>wrote:

>
> I already added "dtmf-relay rtp-nte" and this doesn't work.
>
> Also I don't have "dtmf-interworking rtp-nte" command in Cisco.
> --
> View this message in context:
> http://n2.nabble.com/DTMF-problem-tp4557122p4557446.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100216/7595082c/attachment-0002.html 

From kond at nstel.ru  Tue Feb 16 01:08:39 2010
From: kond at nstel.ru (Nikolay Kondratyev)
Date: Tue, 16 Feb 2010 12:08:39 +0300
Subject: [Freeswitch-users] h323 - sip call is not working
In-Reply-To: <65d96fc81002151202r4fb18b6dma3cfb34fc98adc23@mail.gmail.com>
Message-ID: <20100216090836.C841011FDD@mail.nstel.ru>

Tihomir,

Thanks a lot, I recompiled h323plus libs as you told me and rbt and voice is
ok now.

Thanks again,

Nikolay.

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Tihomir
Culjaga
Sent: Monday, February 15, 2010 11:02 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] h323 - sip call is not working

 

ok, than try this:


edit h323plus/src/h323caps.cxx, grep it for "H323AudioCapability(240, 30) //
240ms max, 30ms desired" ... it should be at line 2598.... 


replace 30 with 20, recompile (make && make install) make sure you use the
new compiled library and start FS.

let me know if you still have audio issues.




-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100216/648dbf2d/attachment-0002.html 

From yehavi.bourvine at gmail.com  Tue Feb 16 01:09:03 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Tue, 16 Feb 2010 11:09:03 +0200
Subject: [Freeswitch-users] DTMF problem
In-Reply-To: <65d96fc81002160010j798353b4y5d38894e774b85e0@mail.gmail.com>
References: <89c9bbf81002111209v6e2f36b3x86a628c5e1390599@mail.gmail.com>
	<F960CC45-C9D2-4150-A108-C0E8B230284D@freeswitch.org>
	<1265922753047-4557446.post@n2.nabble.com>
	<65d96fc81002160010j798353b4y5d38894e774b85e0@mail.gmail.com>
Message-ID: <d361ca7b1002160109r539f98dfra29669a8d7c28e60@mail.gmail.com>

I have a similar problem. In debug mode I see that the Cisco decodes the
DTMFs but does *not* send them to the PSTN.

The only way I managed to make it working is setting the DTMF mode to be
INFO on Freeswitch. This is problematic when you have both phones and the
gateway on the same profile, and the phones use RFC-2833. The solution is to
have a separate profile for the gateway.

                      Regards, __Yehavi:

 2010/2/16 Tihomir Culjaga <tculjaga at gmail.com>

> what DTMF method are you using ?
>
> InBand as pure voice or by rfc2833?
>
>
> if it is as pure voice than you will need this "<action
> application="start_dtmf" />" in FS dialplan receiving the call.
>
> If you are using 2833, than you should check your FS config for DTMF
> method.
>
> T.
>
>
>
> On Thu, Feb 11, 2010 at 10:12 PM, maxim.tsvetov <maxim.tsvetov at gmail.com>wrote:
>
>>
>> I already added "dtmf-relay rtp-nte" and this doesn't work.
>>
>> Also I don't have "dtmf-interworking rtp-nte" command in Cisco.
>> --
>> View this message in context:
>> http://n2.nabble.com/DTMF-problem-tp4557122p4557446.html
>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>
>> _______________________________________________
>>  FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100216/21ffa450/attachment-0002.html 

From devel at thom.fr.eu.org  Tue Feb 16 01:33:44 2010
From: devel at thom.fr.eu.org (=?UTF-8?Q?Fran=C3=A7ois_Legal?=)
Date: Tue, 16 Feb 2010 10:33:44 +0100
Subject: [Freeswitch-users] Sending message notifications with openzap
Message-ID: <e749d5a76cc31b4a3bb2459aaf8b78d9@thom.fr.eu.org>



Hello, 

I was wondering whether or not a facility was available to send
message waiting indicator (from mpd_voicemail) using openzap. 

I know this
feature is not available in openzap (I'm in the process of coding it for
analog channels) but as message indication can be sent via mod_sofia, I
wonder if a frame is available in FS core. 

Anybody can shed me some light
? 

Fran?ois
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100216/754ff240/attachment-0002.html 

From tculjaga at gmail.com  Tue Feb 16 01:39:08 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 16 Feb 2010 10:39:08 +0100
Subject: [Freeswitch-users] h323 - sip call is not working
In-Reply-To: <20100216090836.C841011FDD@mail.nstel.ru>
References: <65d96fc81002151202r4fb18b6dma3cfb34fc98adc23@mail.gmail.com>
	<20100216090836.C841011FDD@mail.nstel.ru>
Message-ID: <65d96fc81002160139x7d002c5al6292e78539b5094c@mail.gmail.com>

On Tue, Feb 16, 2010 at 10:08 AM, Nikolay Kondratyev <kond at nstel.ru> wrote:

>  Tihomir,
>
> Thanks a lot, I recompiled h323plus libs as you told me and rbt and voice
> is ok now.
>

I was afraid so, we will need to work on mod_h323 to allow async codec
framing.


> Thanks again,
>
> Nikolay.
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100216/54b4e03e/attachment-0002.html 

From maxim.tsvetov at gmail.com  Tue Feb 16 02:27:03 2010
From: maxim.tsvetov at gmail.com (maxim.tsvetov)
Date: Tue, 16 Feb 2010 02:27:03 -0800 (PST)
Subject: [Freeswitch-users] DTMF problem
In-Reply-To: <65d96fc81002160010j798353b4y5d38894e774b85e0@mail.gmail.com>
References: <89c9bbf81002111209v6e2f36b3x86a628c5e1390599@mail.gmail.com>
	<F960CC45-C9D2-4150-A108-C0E8B230284D@freeswitch.org>
	<1265922753047-4557446.post@n2.nabble.com>
	<65d96fc81002160010j798353b4y5d38894e774b85e0@mail.gmail.com>
Message-ID: <89c9bbf81002160226u61df2a95j695fcd960f9b2449@mail.gmail.com>


I tried both inband with "start_dtmf" and rfc2833. They are not working.

Maybe there is a method how can I check whether DTMF tones come to FS server
or not ?
(like logs or sniffer)

On Tue, Feb 16, 2010 at 11:18 AM, Tihomir Culjaga [via freeswitch-users] <
ml-node+4579097-1272436358 at n2.nabble.com<ml-node%2B4579097-1272436358 at n2.nabble.com>
> wrote:

> what DTMF method are you using ?
>
> InBand as pure voice or by rfc2833?
>
>
> if it is as pure voice than you will need this "<action
> application="start_dtmf" />" in FS dialplan receiving the call.
>
> If you are using 2833, than you should check your FS config for DTMF
> method.
>
> T.
>
>
> On Thu, Feb 11, 2010 at 10:12 PM, maxim.tsvetov <[hidden email]<http://n2.nabble.com/user/SendEmail.jtp?type=node&node=4579097&i=0>
> > wrote:
>
>>
>> I already added "dtmf-relay rtp-nte" and this doesn't work.
>>
>> Also I don't have "dtmf-interworking rtp-nte" command in Cisco.
>> --
>> View this message in context:
>> http://n2.nabble.com/DTMF-problem-tp4557122p4557446.html
>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> [hidden email]<http://n2.nabble.com/user/SendEmail.jtp?type=node&node=4579097&i=1>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> [hidden email]<http://n2.nabble.com/user/SendEmail.jtp?type=node&node=4579097&i=2>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> ------------------------------
>  View message @ http://n2.nabble.com/DTMF-problem-tp4557122p4579097.html
> To unsubscribe from Re: DTMF problem, click here< (link removed) >.
>
>
>

-- 
View this message in context: http://n2.nabble.com/DTMF-problem-tp4557122p4579539.html
Sent from the freeswitch-users mailing list archive at Nabble.com.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100216/01a50173/attachment-0002.html 

From maxim.tsvetov at gmail.com  Tue Feb 16 02:32:26 2010
From: maxim.tsvetov at gmail.com (maxim.tsvetov)
Date: Tue, 16 Feb 2010 02:32:26 -0800 (PST)
Subject: [Freeswitch-users] DTMF problem
In-Reply-To: <d361ca7b1002160109r539f98dfra29669a8d7c28e60@mail.gmail.com>
References: <89c9bbf81002111209v6e2f36b3x86a628c5e1390599@mail.gmail.com>
	<F960CC45-C9D2-4150-A108-C0E8B230284D@freeswitch.org>
	<1265922753047-4557446.post@n2.nabble.com>
	<65d96fc81002160010j798353b4y5d38894e774b85e0@mail.gmail.com>
	<d361ca7b1002160109r539f98dfra29669a8d7c28e60@mail.gmail.com>
Message-ID: <1266316346819-4579554.post@n2.nabble.com>


Could you please send me example of your FS and Cisco config?

Regards,
Maxim Tsvetov
-- 
View this message in context: http://n2.nabble.com/DTMF-problem-tp4557122p4579554.html
Sent from the freeswitch-users mailing list archive at Nabble.com.



From tculjaga at gmail.com  Tue Feb 16 05:22:43 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 16 Feb 2010 14:22:43 +0100
Subject: [Freeswitch-users] DTMF problem
In-Reply-To: <1266316346819-4579554.post@n2.nabble.com>
References: <89c9bbf81002111209v6e2f36b3x86a628c5e1390599@mail.gmail.com>
	<F960CC45-C9D2-4150-A108-C0E8B230284D@freeswitch.org>
	<1265922753047-4557446.post@n2.nabble.com>
	<65d96fc81002160010j798353b4y5d38894e774b85e0@mail.gmail.com>
	<d361ca7b1002160109r539f98dfra29669a8d7c28e60@mail.gmail.com>
	<1266316346819-4579554.post@n2.nabble.com>
Message-ID: <65d96fc81002160522y14ae88eex810928cadf496fe7@mail.gmail.com>

Please send me a wireshark sniff taken on FS for the call establishment and
DTMF transmition.

not filtered please!

T.

On Tue, Feb 16, 2010 at 11:32 AM, maxim.tsvetov <maxim.tsvetov at gmail.com>wrote:

>
> Could you please send me example of your FS and Cisco config?
>
> Regards,
> Maxim Tsvetov
> --
> View this message in context:
> http://n2.nabble.com/DTMF-problem-tp4557122p4579554.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100216/2862a3f5/attachment-0002.html 

From maxim.tsvetov at gmail.com  Tue Feb 16 05:54:22 2010
From: maxim.tsvetov at gmail.com (maxim.tsvetov)
Date: Tue, 16 Feb 2010 05:54:22 -0800 (PST)
Subject: [Freeswitch-users] DTMF problem
In-Reply-To: <1266316346819-4579554.post@n2.nabble.com>
References: <89c9bbf81002111209v6e2f36b3x86a628c5e1390599@mail.gmail.com>
	<F960CC45-C9D2-4150-A108-C0E8B230284D@freeswitch.org>
	<1265922753047-4557446.post@n2.nabble.com>
	<65d96fc81002160010j798353b4y5d38894e774b85e0@mail.gmail.com>
	<d361ca7b1002160109r539f98dfra29669a8d7c28e60@mail.gmail.com>
	<1266316346819-4579554.post@n2.nabble.com>
Message-ID: <1266328462314-4580309.post@n2.nabble.com>


http://n2.nabble.com/file/n4580309/fs.pcap fs.pcap 
-- 
View this message in context: http://n2.nabble.com/DTMF-problem-tp4557122p4580309.html
Sent from the freeswitch-users mailing list archive at Nabble.com.



From maxim.tsvetov at gmail.com  Tue Feb 16 05:55:06 2010
From: maxim.tsvetov at gmail.com (maxim.tsvetov)
Date: Tue, 16 Feb 2010 05:55:06 -0800 (PST)
Subject: [Freeswitch-users] DTMF problem
In-Reply-To: <65d96fc81002160522y14ae88eex810928cadf496fe7@mail.gmail.com>
References: <89c9bbf81002111209v6e2f36b3x86a628c5e1390599@mail.gmail.com>
	<F960CC45-C9D2-4150-A108-C0E8B230284D@freeswitch.org>
	<1265922753047-4557446.post@n2.nabble.com>
	<65d96fc81002160010j798353b4y5d38894e774b85e0@mail.gmail.com>
	<d361ca7b1002160109r539f98dfra29669a8d7c28e60@mail.gmail.com>
	<1266316346819-4579554.post@n2.nabble.com>
	<65d96fc81002160522y14ae88eex810928cadf496fe7@mail.gmail.com>
Message-ID: <1266328506892-4580313.post@n2.nabble.com>


http://n2.nabble.com/file/n4580313/fs.pcap fs.pcap 
-- 
View this message in context: http://n2.nabble.com/DTMF-problem-tp4557122p4580313.html
Sent from the freeswitch-users mailing list archive at Nabble.com.



From kond at nstel.ru  Tue Feb 16 06:16:34 2010
From: kond at nstel.ru (Nikolay Kondratyev)
Date: Tue, 16 Feb 2010 17:16:34 +0300
Subject: [Freeswitch-users] How to tie context to a gateway?
Message-ID: <20100216141634.75B3511FC6@mail.nstel.ru>

Hi all,

I have several gateways in the external profile.

Let's say GW1 and GW2. I'd like to process calls from the GW1 in the context
C1 and calls from GW2 in the context C2.

Parameter "context", as far as I understand works for the whole profile, not
for individual gateways in the profile.

How do send calls from GW1 into context C1?

What will be a good practice to do that?

Thanks in advance,

Nikolay.

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100216/41af1172/attachment-0002.html 

From tculjaga at gmail.com  Tue Feb 16 06:27:17 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 16 Feb 2010 15:27:17 +0100
Subject: [Freeswitch-users] DTMF problem
In-Reply-To: <1266328506892-4580313.post@n2.nabble.com>
References: <89c9bbf81002111209v6e2f36b3x86a628c5e1390599@mail.gmail.com>
	<F960CC45-C9D2-4150-A108-C0E8B230284D@freeswitch.org>
	<1265922753047-4557446.post@n2.nabble.com>
	<65d96fc81002160010j798353b4y5d38894e774b85e0@mail.gmail.com>
	<d361ca7b1002160109r539f98dfra29669a8d7c28e60@mail.gmail.com>
	<1266316346819-4579554.post@n2.nabble.com>
	<65d96fc81002160522y14ae88eex810928cadf496fe7@mail.gmail.com>
	<1266328506892-4580313.post@n2.nabble.com>
Message-ID: <65d96fc81002160627o79869d84mee3773b34d993d4d@mail.gmail.com>

On Tue, Feb 16, 2010 at 2:55 PM, maxim.tsvetov <maxim.tsvetov at gmail.com>wrote:

>
> http://n2.nabble.com/file/n4580313/fs.pcap fs.pcap
>

the call is not even established so no DTMF can be exchanged :)
please establish a call and send me the sniff.

T.



> --
> View this message in context:
> http://n2.nabble.com/DTMF-problem-tp4557122p4580313.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100216/31299b1e/attachment-0002.html 

From anthony.minessale at gmail.com  Tue Feb 16 06:41:14 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 16 Feb 2010 08:41:14 -0600
Subject: [Freeswitch-users] Broadsoft SCA w/ Cisco SPA-5xx series
In-Reply-To: <8b1c9cda1002152225i7700bdfcgd2c112bf8180e36@mail.gmail.com>
References: <8b1c9cda1002141103te523643ob16db25e3bfa877f@mail.gmail.com>
	<191c3a031002151104u7413b17dh84e68ab1cf703cb9@mail.gmail.com>
	<8b1c9cda1002151147l1e072d00y69c0022e3924c3b4@mail.gmail.com>
	<0dd701caae7e$3c108b70$b431a250$@com>
	<8b1c9cda1002151253t32c22663wf66245e0220a4e6@mail.gmail.com>
	<191c3a031002151304h54c262c5o8ffd5c195e0512d2@mail.gmail.com>
	<0e7301caae84$58b47ce0$0a1d76a0$@com>
	<8b1c9cda1002151456s5dec6dbbtaa2063e0f70fe5dd@mail.gmail.com>
	<191c3a031002151531k148d8504w15e6f4c6c1e3ccd@mail.gmail.com>
	<8b1c9cda1002152225i7700bdfcgd2c112bf8180e36@mail.gmail.com>
Message-ID: <191c3a031002160641q61cb1c53r38add08fc32f3a5a@mail.gmail.com>

as I expected, you have IP addrs in the table which do not match your domain
name.
the phones behind nat should have your domain name in them same as the local
phones.
And the proxy addr should be set to the ip.

If the IP and the DOMAIN do not match you will get mismatches.
Most people make the false assumption that this is like dns where the ip and
hostname are interchangeable.

We can look at making a patch to force the hostname to always be the right
value in the db like we do for reg possibly.



On Tue, Feb 16, 2010 at 12:25 AM, Gabriel Kuri <gkuri at ieee.org> wrote:

> Yeah, the domain name matches on the internal profile.
>
> Thanks for all your help, I can arrange ssh access tomorrow, today
> just wasn't one of those good days to do so, I've been running in and
> out too much to coordinate it.
>
> Here's the pastebin for the sip_dialogs table while the call is up ...
>
>     http://pastebin.freeswitch.org/12159
>
> Thanks,
> Gabe
>
>
> On Mon, Feb 15, 2010 at 3:31 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
> > Do the domain names match on what the remote phones are using?
> >
> > When the call is active, can you attach to sqlite with the sqlite3 app
> and
> > select * from sip_dialogs
> >
> > sqlite3 /usr/local/freeswitch/db/sofia_reg_internal.db
> >> select * from sip_dialogs;
> >
> > remember to do it while the call is up.
> >
> >
> > I am going to bet the domain name in that table is not the same as your
> > actual domain.
> >
> >
> > I tried to make this easier by asking to ssh to your box and work with
> you
> > to fix it but now 9 hours later its starting to resemble diffusing a bomb
> > over a telegraph wire.
> >
> >
> >
> >
> > On Mon, Feb 15, 2010 at 4:56 PM, Gabriel Kuri <gkuri at ieee.org> wrote:
> >>
> >> Yes, the two phones being called (SPA-509Gs) have SCA enabled and it
> >> works when making outgoing calls from those phones. But incoming calls
> >> to those two phones don't seem to have the line key light up on the
> >> other phone when one of the phones is answered (same extension).
> >>
> >> Thanks,
> >> Gabe
> >>
> >> On Mon, Feb 15, 2010 at 1:17 PM, Peder <peder at networkoblivion.com>
> wrote:
> >> > On the phone itself, do you have the line set to shared and ?Broadsoft
> >> > SCA?
> >> > enabled?
> >> >
> >> >
> >> >
> >> > From: freeswitch-users-bounces at lists.freeswitch.org
> >> > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> >> > Anthony
> >> > Minessale
> >> > Sent: Monday, February 15, 2010 3:04 PM
> >> > To: freeswitch-users at lists.freeswitch.org
> >> > Subject: Re: [Freeswitch-users] Broadsoft SCA w/ Cisco SPA-5xx series
> >> >
> >> >
> >> >
> >> > you are missing something because you have no seize events when you go
> >> > on
> >> > and off hook.
> >> > is every phone in the correct mode?
> >> >
> >> > On Mon, Feb 15, 2010 at 2:53 PM, Gabriel Kuri <gkuri at ieee.org> wrote:
> >> >
> >> > No, that was a typo. I have it correct in the config file.
> >> >
> >> > Gabe
> >> >
> >> > On Mon, Feb 15, 2010 at 12:34 PM, Peder <peder at networkoblivion.com>
> >> > wrote:
> >> >
> >> >> Is this a typo "managed-shared-appeareance=true" or is there an extra
> e
> >> >> in
> >> >> appearance in your config?
> >> >>
> >> >> -----Original Message-----
> >> >> From: freeswitch-users-bounces at lists.freeswitch.org
> >> >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> >> >> Gabriel
> >> >> Kuri
> >> >> Sent: Monday, February 15, 2010 1:48 PM
> >> >> To: freeswitch-users at lists.freeswitch.org
> >> >> Subject: Re: [Freeswitch-users] Broadsoft SCA w/ Cisco SPA-5xx series
> >> >>
> >> >> OK, I don't know what happened there, here's another one with the
> >> >> NOTIFYs. I'm on trunk rev 16633 and I have
> >> >> "managed-shared-appeareance=true" on the internal profile. I'm just
> >> >> making calls between internal phones.
> >> >>
> >> >>     http://pastebin.freeswitch.org/12153
> >> >>
> >> >> Thanks,
> >> >> Gabe
> >> >>
> >> >> On Mon, Feb 15, 2010 at 11:04 AM, Anthony Minessale
> >> >> <anthony.minessale at gmail.com> wrote:
> >> >>> I don't see any notifies at all in this trace do the profiles in
> >> >>> question
> >> >>> have:
> >> >>> manage-shared-appearance set to true?
> >> >>> and are you on latest trunk?
> >> >>>
> >> >>>
> >> >>> On Mon, Feb 15, 2010 at 12:55 PM, Anthony Minessale
> >> >>> <anthony.minessale at gmail.com> wrote:
> >> >>>>
> >> >>>> we log the sql stmts on err so they are red and easier to read.
> >> >>>>
> >> >>>>
> >> >>>>
> >> >>>> On Mon, Feb 15, 2010 at 12:48 PM, Gabriel Kuri <gkuri at ieee.org>
> >> >>>> wrote:
> >> >>>>>
> >> >>>>> Here's a call from ext 2551 to 2552 (two phones). I noticed a
> bunch
> >> >>>>> of
> >> >>>>> errors related to SQL UPDATE for presence ...
> >> >>>>>
> >> >>>>>     http://pastebin.freeswitch.org/12152
> >> >>>>>
> >> >>>>> Thanks,
> >> >>>>> Gabe
> >> >>>>>
> >> >>>>>
> >> >>>>> On Mon, Feb 15, 2010 at 8:32 AM, Anthony Minessale
> >> >>>>> <anthony.minessale at gmail.com> wrote:
> >> >>>>> > it should be active not seized.
> >> >>>>> > seized is when you take it off hook.
> >> >>>>> >
> >> >>>>> > We need some more debugging to be sure.
> >> >>>>> > Can we work in real time on it or can you get a more detailed
> log?
> >> >>>>> >
> >> >>>>> > edit sofia.conf.xml and add the param to the "settings" section.
> >> >>>>> >
> >> >>>>> > <param name="debug-sla" value="100"/>
> >> >>>>> >
> >> >>>>> >
> >> >>>>> > then restart and enable sip trace and debug level
> >> >>>>> >
> >> >>>>> > //do this for every profile involved in the call.
> >> >>>>> > sofia profile <profilename> siptrace on
> >> >>>>> >
> >> >>>>> > //also do this
> >> >>>>> > console loglevel debug
> >> >>>>> >
> >> >>>>> >
> >> >>>>> > if you can let us ssh, we can do all the for you if you can make
> >> >>>>> > the
> >> >>>>> > test
> >> >>>>> > calls.
> >> >>>>> >
> >> >>>>> >
> >> >>>>> >
> >> >>>>> >
> >> >>>>> > On Sun, Feb 14, 2010 at 11:59 PM, Gabriel Kuri <gkuri at ieee.org>
> >> >>>>> > wrote:
> >> >>>>> >>
> >> >>>>> >> BTW, here's a copy of the NOTIFY (event call-info) sent to the
> >> >>>>> >> other
> >> >>>>> >> phone after the first phone is answered, should this have a
> >> >>>>> >> Call-Info
> >> >>>>> >> line with an "appearance-state=seized" to turn on the light on
> >> >>>>> >> the
> >> >>>>> >> other phone?
> >> >>>>> >>
> >> >>>>> >>
> >> >>>>> >> NOTIFY sip:2551@<removed>:54446 SIP/2.0.
> >> >>>>> >> Via: SIP/2.0/UDP
> >> >>>>> >> <removed>:9430;rport;branch=z9hG4bK71pN2cXgH851K.
> >> >>>>> >> Max-Forwards: 70.
> >> >>>>> >> From: <sip:2551 at mydomain.com <sip%3A2551 at mydomain.com>
> >;tag=XeB6ZrKDevpHp.
> >> >>>>> >> To: <sip:2551 at mydomain.com <sip%3A2551 at mydomain.com>
> >;tag=c2d34993aac6ea.
> >> >>>>> >> Call-ID: 34c34987-8b6fa786@<removed>.
> >> >>>>> >> CSeq: 126950830 NOTIFY.
> >> >>>>> >> Contact: <sip:2551@<removed>:9430>.
> >> >>>>> >> Expires: 3959.
> >> >>>>> >> Call-Info:
> >> >>>>> >> <sip:mydomain.com>;appearance-index=*;appearance-state=idle.
> >> >>>>> >> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16633M.
> >> >>>>> >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE,
> INFO,
> >> >>>>> >> REGISTER, REFER, PRACK, NOTIFY, PUBLISH, SUBSCRIBE.
> >> >>>>> >> Supported: 100rel, timer, precondition, path, replaces.
> >> >>>>> >> Event: call-info.
> >> >>>>> >> Allow-Events: talk, presence, dialog, line-seize, call-info,
> sla,
> >> >>>>> >> include-session-description, presence.winfo, message-summary,
> >> >>>>> >> refer.
> >> >>>>> >> Subscription-State: active;expires=3959.
> >> >>>>> >> Content-Length: 0.
> >> >>>>> >>
> >> >>>>> >>
> >> >>>>> >>
> >> >>>>> >> On Sun, Feb 14, 2010 at 7:58 PM, Brian West
> >> >>>>> >> <brian at freeswitch.org>
> >> >>>>> >> wrote:
> >> >>>>> >> > Works fine here... is your box slow or something?
> >> >>>>> >> >
> >> >>>>> >> > /b
> >> >>>>> >> >
> >> >>>>> >> > On Feb 14, 2010, at 1:03 PM, Gabriel Kuri wrote:
> >> >>>>> >> >
> >> >>>>> >> >> I followed Brian's directions from one of the previous
> threads
> >> >>>>> >> >> on
> >> >>>>> >> >> configuring the SPA-5xx series phones for Broadsoft SCA and
> >> >>>>> >> >> set
> >> >>>>> >> >> manage-shared-appearance=true in the internal profile. SCA
> >> >>>>> >> >> appears
> >> >>>>> >> >> to
> >> >>>>> >> >> be working on outgoing calls between two phones, the line
> key
> >> >>>>> >> >> starts
> >> >>>>> >> >> flashing red on the second phone when the first phone picks
> up
> >> >>>>> >> >> the
> >> >>>>> >> >> receiver to make a call. However on incoming calls, both
> >> >>>>> >> >> phones
> >> >>>>> >> >> ring
> >> >>>>> >> >> (same extension), however when one of the phones picks up
> the
> >> >> line,
> >> >>>>> >> >> the second phone's line key doesn't flash red or show the
> >> >>>>> >> >> first
> >> >>>>> >> >> phone
> >> >>>>> >> >> on that incoming call. Any ideas? Does shared appearance
> only
> >> >>>>> >> >> work
> >> >>>>> >> >> on
> >> >>>>> >> >> outgoing phone calls?
> >> >>>>> >> >>
> >> >>>>> >> >> Thanks,
> >> >>>>> >> >> Gabe
> >> >>>>> >> >
> >> >>>>> >> >
> >> >>>>> >> > _______________________________________________
> >> >>>>> >> > FreeSWITCH-users mailing list
> >> >>>>> >> > FreeSWITCH-users at lists.freeswitch.org
> >> >>>>> >> >
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>>>> >> >
> >> >>>>> >> >
> >> >>
> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >>>>> >> > http://www.freeswitch.org
> >> >>>>> >> >
> >> >>>>> >>
> >> >>>>> >> _______________________________________________
> >> >>>>> >> FreeSWITCH-users mailing list
> >> >>>>> >> FreeSWITCH-users at lists.freeswitch.org
> >> >>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>>>> >>
> >> >>>>> >>
> >> >>
> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >>>>> >> http://www.freeswitch.org
> >> >>>>> >
> >> >>>>> >
> >> >>>>> >
> >> >>>>> > --
> >> >>>>> > Anthony Minessale II
> >> >>>>> >
> >> >>>>> > FreeSWITCH http://www.freeswitch.org/
> >> >>>>> > ClueCon http://www.cluecon.com/
> >> >>>>> > Twitter: http://twitter.com/FreeSWITCH_wire
> >> >>>>> >
> >> >>>>> > AIM: anthm
> >> >>>>> > MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
> >> >>>>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >> >>>>> > IRC: irc.freenode.net #freeswitch
> >> >>>>> >
> >> >>>>> > FreeSWITCH Developer Conference
> >> >>>>> > sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> >> >>>>> > iax:guest at conference.freeswitch.org/888
> >> >>>>> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >> >>>>> > pstn:+19193869900
> >> >>>>> >
> >> >>>>> > _______________________________________________
> >> >>>>> > FreeSWITCH-users mailing list
> >> >>>>> > FreeSWITCH-users at lists.freeswitch.org
> >> >>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>>>> >
> >> >>>>> >
> >> >>
> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >>>>> > http://www.freeswitch.org
> >> >>>>> >
> >> >>>>> >
> >> >>>>>
> >> >>>>> _______________________________________________
> >> >>>>> FreeSWITCH-users mailing list
> >> >>>>> FreeSWITCH-users at lists.freeswitch.org
> >> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>>>>
> >> >>>>>
> >> >>>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >>>>> http://www.freeswitch.org
> >> >>>>
> >> >>>>
> >> >>>>
> >> >>>> --
> >> >>>> Anthony Minessale II
> >> >>>>
> >> >>>> FreeSWITCH http://www.freeswitch.org/
> >> >>>> ClueCon http://www.cluecon.com/
> >> >>>> Twitter: http://twitter.com/FreeSWITCH_wire
> >> >>>>
> >> >>>> AIM: anthm
> >> >>>> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
> >> >>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >> >>>> IRC: irc.freenode.net #freeswitch
> >> >>>>
> >> >>>> FreeSWITCH Developer Conference
> >> >>>> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> >> >>>> iax:guest at conference.freeswitch.org/888
> >> >>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >> >>>> pstn:+19193869900
> >> >>>
> >> >>>
> >> >>>
> >> >>> --
> >> >>> Anthony Minessale II
> >> >>>
> >> >>> FreeSWITCH http://www.freeswitch.org/
> >> >>> ClueCon http://www.cluecon.com/
> >> >>> Twitter: http://twitter.com/FreeSWITCH_wire
> >> >>>
> >> >>> AIM: anthm
> >> >>> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
> >> >>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >> >>> IRC: irc.freenode.net #freeswitch
> >> >>>
> >> >>> FreeSWITCH Developer Conference
> >> >>> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> >> >>> iax:guest at conference.freeswitch.org/888
> >> >>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >> >>> pstn:+19193869900
> >> >>>
> >> >>> _______________________________________________
> >> >>> FreeSWITCH-users mailing list
> >> >>> FreeSWITCH-users at lists.freeswitch.org
> >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>>
> >> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >>> http://www.freeswitch.org
> >> >>>
> >> >>>
> >> >>
> >> >> _______________________________________________
> >> >> FreeSWITCH-users mailing list
> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>
> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> http://www.freeswitch.org
> >> >>
> >> >>
> >> >> _______________________________________________
> >> >> FreeSWITCH-users mailing list
> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>
> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> http://www.freeswitch.org
> >> >>
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >> > --
> >> > Anthony Minessale II
> >> >
> >> > FreeSWITCH http://www.freeswitch.org/
> >> > ClueCon http://www.cluecon.com/
> >> > Twitter: http://twitter.com/FreeSWITCH_wire
> >> >
> >> > AIM: anthm
> >> > MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
> >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >> > IRC: irc.freenode.net #freeswitch
> >> >
> >> > FreeSWITCH Developer Conference
> >> > sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> >> > iax:guest at conference.freeswitch.org/888
> >> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >> > pstn:+19193869900
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> > Twitter: http://twitter.com/FreeSWITCH_wire
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> > IRC: irc.freenode.net #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> > iax:guest at conference.freeswitch.org/888
> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > pstn:+19193869900
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100216/0d5c4dcb/attachment-0002.html 

From yehavi.bourvine at gmail.com  Tue Feb 16 06:56:22 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Tue, 16 Feb 2010 16:56:22 +0200
Subject: [Freeswitch-users] SQL ERR: library routine called out of sequence
Message-ID: <d361ca7b1002160656w7d616f6dn5e80087ebde94dbe@mail.gmail.com>

Hello,

  After upgrading from 1.0.5pre10 to 1.0.5-20100216-0400 we started getting
the above errors (I append bellow two samples). It seems Freeswitch fails to
read a database using Sqlite.
Anyone have seen this?

Other details: Fedora 10, SQlite 3.5.9.
We also do SQLite quesries during call setup via LUA from CoreDB. Is it an
SQLite problem?

                         Thanks! __Yehavi:

The samples:
2010-02-16 16:22:13.762679 [ERR] switch_core_sqldb.c:404 SQL ERR [library
routin
e called out of sequence]
delete from sip_dialogs where call_id='
8656841832142-120172129116107 at 10.64.1.2'

2010-02-16 16:42:00.802442 [ERR] switch_core_sqldb.c:722 SQL ERR: [select
call_i
d,sip_user,sip_host,contact,status,rpid,expires,user_agent,server_user,server_ho
st,profile_name,hostname,network_ip,network_port,sip_username,sip_realm,mwi_user
,mwi_host from sip_registrations where profile_name='phones' and contact
like '%
80635%'] library routine called out of sequence
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100216/11d23da8/attachment-0002.html 

From anthony.minessale at gmail.com  Tue Feb 16 07:05:12 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 16 Feb 2010 09:05:12 -0600
Subject: [Freeswitch-users] SQL ERR: library routine called out of
	sequence
In-Reply-To: <d361ca7b1002160656w7d616f6dn5e80087ebde94dbe@mail.gmail.com>
References: <d361ca7b1002160656w7d616f6dn5e80087ebde94dbe@mail.gmail.com>
Message-ID: <191c3a031002160705w4050c3cdk2bf2922e9aff5ba3@mail.gmail.com>

try removing all the .db files from /usr/local/freeswitch/db


On Tue, Feb 16, 2010 at 8:56 AM, Yehavi Bourvine
<yehavi.bourvine at gmail.com>wrote:

> Hello,
>
>   After upgrading from 1.0.5pre10 to 1.0.5-20100216-0400 we started getting
> the above errors (I append bellow two samples). It seems Freeswitch fails to
> read a database using Sqlite.
> Anyone have seen this?
>
> Other details: Fedora 10, SQlite 3.5.9.
> We also do SQLite quesries during call setup via LUA from CoreDB. Is it an
> SQLite problem?
>
>                          Thanks! __Yehavi:
>
> The samples:
> 2010-02-16 16:22:13.762679 [ERR] switch_core_sqldb.c:404 SQL ERR [library
> routin
> e called out of sequence]
> delete from sip_dialogs where call_id='
> 8656841832142-120172129116107 at 10.64.1.2'
>
> 2010-02-16 16:42:00.802442 [ERR] switch_core_sqldb.c:722 SQL ERR: [select
> call_i
>
> d,sip_user,sip_host,contact,status,rpid,expires,user_agent,server_user,server_ho
>
> st,profile_name,hostname,network_ip,network_port,sip_username,sip_realm,mwi_user
> ,mwi_host from sip_registrations where profile_name='phones' and contact
> like '%
> 80635%'] library routine called out of sequence
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100216/5112a388/attachment-0002.html 

From yehavi.bourvine at gmail.com  Tue Feb 16 07:27:50 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Tue, 16 Feb 2010 17:27:50 +0200
Subject: [Freeswitch-users] SQL ERR: library routine called out of
	sequence
In-Reply-To: <191c3a031002160705w4050c3cdk2bf2922e9aff5ba3@mail.gmail.com>
References: <d361ca7b1002160656w7d616f6dn5e80087ebde94dbe@mail.gmail.com>
	<191c3a031002160705w4050c3cdk2bf2922e9aff5ba3@mail.gmail.com>
Message-ID: <d361ca7b1002160727ucb6e0c8id62e803757333213@mail.gmail.com>

Tried this, but it didn't help. I delete these DB files before any upgrade
just to be sure.

                  Thanks! __Yehavi:

 2010/2/16 Anthony Minessale <anthony.minessale at gmail.com>

> try removing all the .db files from /usr/local/freeswitch/db
>
>
>   On Tue, Feb 16, 2010 at 8:56 AM, Yehavi Bourvine <
> yehavi.bourvine at gmail.com> wrote:
>
>>   Hello,
>>
>>   After upgrading from 1.0.5pre10 to 1.0.5-20100216-0400 we started
>> getting the above errors (I append bellow two samples). It seems Freeswitch
>> fails to read a database using Sqlite.
>> Anyone have seen this?
>>
>> Other details: Fedora 10, SQlite 3.5.9.
>> We also do SQLite quesries during call setup via LUA from CoreDB. Is it an
>> SQLite problem?
>>
>>                          Thanks! __Yehavi:
>>
>> The samples:
>> 2010-02-16 16:22:13.762679 [ERR] switch_core_sqldb.c:404 SQL ERR [library
>> routin
>> e called out of sequence]
>> delete from sip_dialogs where call_id='
>> 8656841832142-120172129116107 at 10.64.1.2'
>>
>> 2010-02-16 16:42:00.802442 [ERR] switch_core_sqldb.c:722 SQL ERR: [select
>> call_i
>>
>> d,sip_user,sip_host,contact,status,rpid,expires,user_agent,server_user,server_ho
>>
>> st,profile_name,hostname,network_ip,network_port,sip_username,sip_realm,mwi_user
>> ,mwi_host from sip_registrations where profile_name='phones' and contact
>> like '%
>> 80635%'] library routine called out of sequence
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100216/3ef8c588/attachment-0002.html 

From gkuri at ieee.org  Tue Feb 16 07:38:27 2010
From: gkuri at ieee.org (Gabriel Kuri)
Date: Tue, 16 Feb 2010 07:38:27 -0800
Subject: [Freeswitch-users] Broadsoft SCA w/ Cisco SPA-5xx series
In-Reply-To: <191c3a031002160641q61cb1c53r38add08fc32f3a5a@mail.gmail.com>
References: <8b1c9cda1002141103te523643ob16db25e3bfa877f@mail.gmail.com>
	<8b1c9cda1002151147l1e072d00y69c0022e3924c3b4@mail.gmail.com>
	<0dd701caae7e$3c108b70$b431a250$@com>
	<8b1c9cda1002151253t32c22663wf66245e0220a4e6@mail.gmail.com>
	<191c3a031002151304h54c262c5o8ffd5c195e0512d2@mail.gmail.com>
	<0e7301caae84$58b47ce0$0a1d76a0$@com>
	<8b1c9cda1002151456s5dec6dbbtaa2063e0f70fe5dd@mail.gmail.com>
	<191c3a031002151531k148d8504w15e6f4c6c1e3ccd@mail.gmail.com>
	<8b1c9cda1002152225i7700bdfcgd2c112bf8180e36@mail.gmail.com>
	<191c3a031002160641q61cb1c53r38add08fc32f3a5a@mail.gmail.com>
Message-ID: <8b1c9cda1002160738r619b2a3cs3bf5dd7d1322121e@mail.gmail.com>

The phones are currently setup with the domain in their "Proxy" field
and set to use SRV to lookup the IP. The "Outbound Proxy" field is
left empty. How should the phones be setup? The Proxy field with the
domain and Outbound Proxy set to the IP?

Thanks,
Gabe

On Tue, Feb 16, 2010 at 6:41 AM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> as I expected, you have IP addrs in the table which do not match your domain
> name.
> the phones behind nat should have your domain name in them same as the local
> phones.
> And the proxy addr should be set to the ip.
>
> If the IP and the DOMAIN do not match you will get mismatches.
> Most people make the false assumption that this is like dns where the ip and
> hostname are interchangeable.
>
> We can look at making a patch to force the hostname to always be the right
> value in the db like we do for reg possibly.
>
>
>
> On Tue, Feb 16, 2010 at 12:25 AM, Gabriel Kuri <gkuri at ieee.org> wrote:
>>
>> Yeah, the domain name matches on the internal profile.
>>
>> Thanks for all your help, I can arrange ssh access tomorrow, today
>> just wasn't one of those good days to do so, I've been running in and
>> out too much to coordinate it.
>>
>> Here's the pastebin for the sip_dialogs table while the call is up ...
>>
>> ? ? http://pastebin.freeswitch.org/12159
>>
>> Thanks,
>> Gabe
>>
>>
>> On Mon, Feb 15, 2010 at 3:31 PM, Anthony Minessale
>> <anthony.minessale at gmail.com> wrote:
>> > Do the domain names match on what the remote phones are using?
>> >
>> > When the call is active, can you attach to sqlite with the sqlite3 app
>> > and
>> > select * from sip_dialogs
>> >
>> > sqlite3 /usr/local/freeswitch/db/sofia_reg_internal.db
>> >> select * from sip_dialogs;
>> >
>> > remember to do it while the call is up.
>> >
>> >
>> > I am going to bet the domain name in that table is not the same as your
>> > actual domain.
>> >
>> >
>> > I tried to make this easier by asking to ssh to your box and work with
>> > you
>> > to fix it but now 9 hours later its starting to resemble diffusing a
>> > bomb
>> > over a telegraph wire.
>> >
>> >
>> >
>> >
>> > On Mon, Feb 15, 2010 at 4:56 PM, Gabriel Kuri <gkuri at ieee.org> wrote:
>> >>
>> >> Yes, the two phones being called (SPA-509Gs) have SCA enabled and it
>> >> works when making outgoing calls from those phones. But incoming calls
>> >> to those two phones don't seem to have the line key light up on the
>> >> other phone when one of the phones is answered (same extension).
>> >>
>> >> Thanks,
>> >> Gabe
>> >>
>> >> On Mon, Feb 15, 2010 at 1:17 PM, Peder <peder at networkoblivion.com>
>> >> wrote:
>> >> > On the phone itself, do you have the line set to shared and
>> >> > ?Broadsoft
>> >> > SCA?
>> >> > enabled?
>> >> >
>> >> >
>> >> >
>> >> > From: freeswitch-users-bounces at lists.freeswitch.org
>> >> > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
>> >> > Anthony
>> >> > Minessale
>> >> > Sent: Monday, February 15, 2010 3:04 PM
>> >> > To: freeswitch-users at lists.freeswitch.org
>> >> > Subject: Re: [Freeswitch-users] Broadsoft SCA w/ Cisco SPA-5xx series
>> >> >
>> >> >
>> >> >
>> >> > you are missing something because you have no seize events when you
>> >> > go
>> >> > on
>> >> > and off hook.
>> >> > is every phone in the correct mode?
>> >> >
>> >> > On Mon, Feb 15, 2010 at 2:53 PM, Gabriel Kuri <gkuri at ieee.org> wrote:
>> >> >
>> >> > No, that was a typo. I have it correct in the config file.
>> >> >
>> >> > Gabe
>> >> >
>> >> > On Mon, Feb 15, 2010 at 12:34 PM, Peder <peder at networkoblivion.com>
>> >> > wrote:
>> >> >
>> >> >> Is this a typo "managed-shared-appeareance=true" or is there an
>> >> >> extra e
>> >> >> in
>> >> >> appearance in your config?
>> >> >>
>> >> >> -----Original Message-----
>> >> >> From: freeswitch-users-bounces at lists.freeswitch.org
>> >> >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
>> >> >> Gabriel
>> >> >> Kuri
>> >> >> Sent: Monday, February 15, 2010 1:48 PM
>> >> >> To: freeswitch-users at lists.freeswitch.org
>> >> >> Subject: Re: [Freeswitch-users] Broadsoft SCA w/ Cisco SPA-5xx
>> >> >> series
>> >> >>
>> >> >> OK, I don't know what happened there, here's another one with the
>> >> >> NOTIFYs. I'm on trunk rev 16633 and I have
>> >> >> "managed-shared-appeareance=true" on the internal profile. I'm just
>> >> >> making calls between internal phones.
>> >> >>
>> >> >> ? ? http://pastebin.freeswitch.org/12153
>> >> >>
>> >> >> Thanks,
>> >> >> Gabe
>> >> >>
>> >> >> On Mon, Feb 15, 2010 at 11:04 AM, Anthony Minessale
>> >> >> <anthony.minessale at gmail.com> wrote:
>> >> >>> I don't see any notifies at all in this trace do the profiles in
>> >> >>> question
>> >> >>> have:
>> >> >>> manage-shared-appearance set to true?
>> >> >>> and are you on latest trunk?
>> >> >>>
>> >> >>>
>> >> >>> On Mon, Feb 15, 2010 at 12:55 PM, Anthony Minessale
>> >> >>> <anthony.minessale at gmail.com> wrote:
>> >> >>>>
>> >> >>>> we log the sql stmts on err so they are red and easier to read.
>> >> >>>>
>> >> >>>>
>> >> >>>>
>> >> >>>> On Mon, Feb 15, 2010 at 12:48 PM, Gabriel Kuri <gkuri at ieee.org>
>> >> >>>> wrote:
>> >> >>>>>
>> >> >>>>> Here's a call from ext 2551 to 2552 (two phones). I noticed a
>> >> >>>>> bunch
>> >> >>>>> of
>> >> >>>>> errors related to SQL UPDATE for presence ...
>> >> >>>>>
>> >> >>>>> ? ? http://pastebin.freeswitch.org/12152
>> >> >>>>>
>> >> >>>>> Thanks,
>> >> >>>>> Gabe
>> >> >>>>>
>> >> >>>>>
>> >> >>>>> On Mon, Feb 15, 2010 at 8:32 AM, Anthony Minessale
>> >> >>>>> <anthony.minessale at gmail.com> wrote:
>> >> >>>>> > it should be active not seized.
>> >> >>>>> > seized is when you take it off hook.
>> >> >>>>> >
>> >> >>>>> > We need some more debugging to be sure.
>> >> >>>>> > Can we work in real time on it or can you get a more detailed
>> >> >>>>> > log?
>> >> >>>>> >
>> >> >>>>> > edit sofia.conf.xml and add the param to the "settings"
>> >> >>>>> > section.
>> >> >>>>> >
>> >> >>>>> > <param name="debug-sla" value="100"/>
>> >> >>>>> >
>> >> >>>>> >
>> >> >>>>> > then restart and enable sip trace and debug level
>> >> >>>>> >
>> >> >>>>> > //do this for every profile involved in the call.
>> >> >>>>> > sofia profile <profilename> siptrace on
>> >> >>>>> >
>> >> >>>>> > //also do this
>> >> >>>>> > console loglevel debug
>> >> >>>>> >
>> >> >>>>> >
>> >> >>>>> > if you can let us ssh, we can do all the for you if you can
>> >> >>>>> > make
>> >> >>>>> > the
>> >> >>>>> > test
>> >> >>>>> > calls.
>> >> >>>>> >
>> >> >>>>> >
>> >> >>>>> >
>> >> >>>>> >
>> >> >>>>> > On Sun, Feb 14, 2010 at 11:59 PM, Gabriel Kuri <gkuri at ieee.org>
>> >> >>>>> > wrote:
>> >> >>>>> >>
>> >> >>>>> >> BTW, here's a copy of the NOTIFY (event call-info) sent to the
>> >> >>>>> >> other
>> >> >>>>> >> phone after the first phone is answered, should this have a
>> >> >>>>> >> Call-Info
>> >> >>>>> >> line with an "appearance-state=seized" to turn on the light on
>> >> >>>>> >> the
>> >> >>>>> >> other phone?
>> >> >>>>> >>
>> >> >>>>> >>
>> >> >>>>> >> NOTIFY sip:2551@<removed>:54446 SIP/2.0.
>> >> >>>>> >> Via: SIP/2.0/UDP
>> >> >>>>> >> <removed>:9430;rport;branch=z9hG4bK71pN2cXgH851K.
>> >> >>>>> >> Max-Forwards: 70.
>> >> >>>>> >> From: <sip:2551 at mydomain.com>;tag=XeB6ZrKDevpHp.
>> >> >>>>> >> To: <sip:2551 at mydomain.com>;tag=c2d34993aac6ea.
>> >> >>>>> >> Call-ID: 34c34987-8b6fa786@<removed>.
>> >> >>>>> >> CSeq: 126950830 NOTIFY.
>> >> >>>>> >> Contact: <sip:2551@<removed>:9430>.
>> >> >>>>> >> Expires: 3959.
>> >> >>>>> >> Call-Info:
>> >> >>>>> >> <sip:mydomain.com>;appearance-index=*;appearance-state=idle.
>> >> >>>>> >> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16633M.
>> >> >>>>> >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE,
>> >> >>>>> >> INFO,
>> >> >>>>> >> REGISTER, REFER, PRACK, NOTIFY, PUBLISH, SUBSCRIBE.
>> >> >>>>> >> Supported: 100rel, timer, precondition, path, replaces.
>> >> >>>>> >> Event: call-info.
>> >> >>>>> >> Allow-Events: talk, presence, dialog, line-seize, call-info,
>> >> >>>>> >> sla,
>> >> >>>>> >> include-session-description, presence.winfo, message-summary,
>> >> >>>>> >> refer.
>> >> >>>>> >> Subscription-State: active;expires=3959.
>> >> >>>>> >> Content-Length: 0.
>> >> >>>>> >>
>> >> >>>>> >>
>> >> >>>>> >>
>> >> >>>>> >> On Sun, Feb 14, 2010 at 7:58 PM, Brian West
>> >> >>>>> >> <brian at freeswitch.org>
>> >> >>>>> >> wrote:
>> >> >>>>> >> > Works fine here... is your box slow or something?
>> >> >>>>> >> >
>> >> >>>>> >> > /b
>> >> >>>>> >> >
>> >> >>>>> >> > On Feb 14, 2010, at 1:03 PM, Gabriel Kuri wrote:
>> >> >>>>> >> >
>> >> >>>>> >> >> I followed Brian's directions from one of the previous
>> >> >>>>> >> >> threads
>> >> >>>>> >> >> on
>> >> >>>>> >> >> configuring the SPA-5xx series phones for Broadsoft SCA and
>> >> >>>>> >> >> set
>> >> >>>>> >> >> manage-shared-appearance=true in the internal profile. SCA
>> >> >>>>> >> >> appears
>> >> >>>>> >> >> to
>> >> >>>>> >> >> be working on outgoing calls between two phones, the line
>> >> >>>>> >> >> key
>> >> >>>>> >> >> starts
>> >> >>>>> >> >> flashing red on the second phone when the first phone picks
>> >> >>>>> >> >> up
>> >> >>>>> >> >> the
>> >> >>>>> >> >> receiver to make a call. However on incoming calls, both
>> >> >>>>> >> >> phones
>> >> >>>>> >> >> ring
>> >> >>>>> >> >> (same extension), however when one of the phones picks up
>> >> >>>>> >> >> the
>> >> >> line,
>> >> >>>>> >> >> the second phone's line key doesn't flash red or show the
>> >> >>>>> >> >> first
>> >> >>>>> >> >> phone
>> >> >>>>> >> >> on that incoming call. Any ideas? Does shared appearance
>> >> >>>>> >> >> only
>> >> >>>>> >> >> work
>> >> >>>>> >> >> on
>> >> >>>>> >> >> outgoing phone calls?
>> >> >>>>> >> >>
>> >> >>>>> >> >> Thanks,
>> >> >>>>> >> >> Gabe
>> >> >>>>> >> >
>> >> >>>>> >> >
>> >> >>>>> >> > _______________________________________________
>> >> >>>>> >> > FreeSWITCH-users mailing list
>> >> >>>>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> >>>>> >> >
>> >> >>>>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>>>> >> >
>> >> >>>>> >> >
>> >> >>
>> >> >>
>> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >>>>> >> > http://www.freeswitch.org
>> >> >>>>> >> >
>> >> >>>>> >>
>> >> >>>>> >> _______________________________________________
>> >> >>>>> >> FreeSWITCH-users mailing list
>> >> >>>>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> >>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>>>> >>
>> >> >>>>> >>
>> >> >>
>> >> >>
>> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >>>>> >> http://www.freeswitch.org
>> >> >>>>> >
>> >> >>>>> >
>> >> >>>>> >
>> >> >>>>> > --
>> >> >>>>> > Anthony Minessale II
>> >> >>>>> >
>> >> >>>>> > FreeSWITCH http://www.freeswitch.org/
>> >> >>>>> > ClueCon http://www.cluecon.com/
>> >> >>>>> > Twitter: http://twitter.com/FreeSWITCH_wire
>> >> >>>>> >
>> >> >>>>> > AIM: anthm
>> >> >>>>> > MSN:anthony_minessale at hotmail.com
>> >> >>>>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> >> >>>>> > IRC: irc.freenode.net #freeswitch
>> >> >>>>> >
>> >> >>>>> > FreeSWITCH Developer Conference
>> >> >>>>> > sip:888 at conference.freeswitch.org
>> >> >>>>> > iax:guest at conference.freeswitch.org/888
>> >> >>>>> > googletalk:conf+888 at conference.freeswitch.org
>> >> >>>>> > pstn:+19193869900
>> >> >>>>> >
>> >> >>>>> > _______________________________________________
>> >> >>>>> > FreeSWITCH-users mailing list
>> >> >>>>> > FreeSWITCH-users at lists.freeswitch.org
>> >> >>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>>>> >
>> >> >>>>> >
>> >> >>
>> >> >>
>> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >>>>> > http://www.freeswitch.org
>> >> >>>>> >
>> >> >>>>> >
>> >> >>>>>
>> >> >>>>> _______________________________________________
>> >> >>>>> FreeSWITCH-users mailing list
>> >> >>>>> FreeSWITCH-users at lists.freeswitch.org
>> >> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>>>>
>> >> >>>>>
>> >> >>>>>
>> >> >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >>>>> http://www.freeswitch.org
>> >> >>>>
>> >> >>>>
>> >> >>>>
>> >> >>>> --
>> >> >>>> Anthony Minessale II
>> >> >>>>
>> >> >>>> FreeSWITCH http://www.freeswitch.org/
>> >> >>>> ClueCon http://www.cluecon.com/
>> >> >>>> Twitter: http://twitter.com/FreeSWITCH_wire
>> >> >>>>
>> >> >>>> AIM: anthm
>> >> >>>> MSN:anthony_minessale at hotmail.com
>> >> >>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> >> >>>> IRC: irc.freenode.net #freeswitch
>> >> >>>>
>> >> >>>> FreeSWITCH Developer Conference
>> >> >>>> sip:888 at conference.freeswitch.org
>> >> >>>> iax:guest at conference.freeswitch.org/888
>> >> >>>> googletalk:conf+888 at conference.freeswitch.org
>> >> >>>> pstn:+19193869900
>> >> >>>
>> >> >>>
>> >> >>>
>> >> >>> --
>> >> >>> Anthony Minessale II
>> >> >>>
>> >> >>> FreeSWITCH http://www.freeswitch.org/
>> >> >>> ClueCon http://www.cluecon.com/
>> >> >>> Twitter: http://twitter.com/FreeSWITCH_wire
>> >> >>>
>> >> >>> AIM: anthm
>> >> >>> MSN:anthony_minessale at hotmail.com
>> >> >>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> >> >>> IRC: irc.freenode.net #freeswitch
>> >> >>>
>> >> >>> FreeSWITCH Developer Conference
>> >> >>> sip:888 at conference.freeswitch.org
>> >> >>> iax:guest at conference.freeswitch.org/888
>> >> >>> googletalk:conf+888 at conference.freeswitch.org
>> >> >>> pstn:+19193869900
>> >> >>>
>> >> >>> _______________________________________________
>> >> >>> FreeSWITCH-users mailing list
>> >> >>> FreeSWITCH-users at lists.freeswitch.org
>> >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>>
>> >> >>>
>> >> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >>> http://www.freeswitch.org
>> >> >>>
>> >> >>>
>> >> >>
>> >> >> _______________________________________________
>> >> >> FreeSWITCH-users mailing list
>> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>
>> >> >>
>> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> http://www.freeswitch.org
>> >> >>
>> >> >>
>> >> >> _______________________________________________
>> >> >> FreeSWITCH-users mailing list
>> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>
>> >> >>
>> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> http://www.freeswitch.org
>> >> >>
>> >> >
>> >> > _______________________________________________
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >
>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> > http://www.freeswitch.org
>> >> >
>> >> >
>> >> > --
>> >> > Anthony Minessale II
>> >> >
>> >> > FreeSWITCH http://www.freeswitch.org/
>> >> > ClueCon http://www.cluecon.com/
>> >> > Twitter: http://twitter.com/FreeSWITCH_wire
>> >> >
>> >> > AIM: anthm
>> >> > MSN:anthony_minessale at hotmail.com
>> >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> >> > IRC: irc.freenode.net #freeswitch
>> >> >
>> >> > FreeSWITCH Developer Conference
>> >> > sip:888 at conference.freeswitch.org
>> >> > iax:guest at conference.freeswitch.org/888
>> >> > googletalk:conf+888 at conference.freeswitch.org
>> >> > pstn:+19193869900
>> >> >
>> >> > _______________________________________________
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >
>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> > http://www.freeswitch.org
>> >> >
>> >> >
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> >
>> > --
>> > Anthony Minessale II
>> >
>> > FreeSWITCH http://www.freeswitch.org/
>> > ClueCon http://www.cluecon.com/
>> > Twitter: http://twitter.com/FreeSWITCH_wire
>> >
>> > AIM: anthm
>> > MSN:anthony_minessale at hotmail.com
>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> > IRC: irc.freenode.net #freeswitch
>> >
>> > FreeSWITCH Developer Conference
>> > sip:888 at conference.freeswitch.org
>> > iax:guest at conference.freeswitch.org/888
>> > googletalk:conf+888 at conference.freeswitch.org
>> > pstn:+19193869900
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From anthony.minessale at gmail.com  Tue Feb 16 07:55:02 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 16 Feb 2010 09:55:02 -0600
Subject: [Freeswitch-users] SQL ERR: library routine called out of
	sequence
In-Reply-To: <d361ca7b1002160727ucb6e0c8id62e803757333213@mail.gmail.com>
References: <d361ca7b1002160656w7d616f6dn5e80087ebde94dbe@mail.gmail.com>
	<191c3a031002160705w4050c3cdk2bf2922e9aff5ba3@mail.gmail.com>
	<d361ca7b1002160727ucb6e0c8id62e803757333213@mail.gmail.com>
Message-ID: <191c3a031002160755q569fe2ednfefb01b6d9cf7f0e@mail.gmail.com>

you may want to do a clean wipe of all files related to FS then.
you clearly have some problem with legacy something or other because we
don't see that on dozens of dev boxes.

What os is it?


On Tue, Feb 16, 2010 at 9:27 AM, Yehavi Bourvine
<yehavi.bourvine at gmail.com>wrote:

> Tried this, but it didn't help. I delete these DB files before any upgrade
> just to be sure.
>
>                   Thanks! __Yehavi:
>
>  2010/2/16 Anthony Minessale <anthony.minessale at gmail.com>
>
>> try removing all the .db files from /usr/local/freeswitch/db
>>
>>
>>   On Tue, Feb 16, 2010 at 8:56 AM, Yehavi Bourvine <
>> yehavi.bourvine at gmail.com> wrote:
>>
>>>   Hello,
>>>
>>>   After upgrading from 1.0.5pre10 to 1.0.5-20100216-0400 we started
>>> getting the above errors (I append bellow two samples). It seems Freeswitch
>>> fails to read a database using Sqlite.
>>> Anyone have seen this?
>>>
>>> Other details: Fedora 10, SQlite 3.5.9.
>>> We also do SQLite quesries during call setup via LUA from CoreDB. Is it
>>> an SQLite problem?
>>>
>>>                          Thanks! __Yehavi:
>>>
>>> The samples:
>>> 2010-02-16 16:22:13.762679 [ERR] switch_core_sqldb.c:404 SQL ERR [library
>>> routin
>>> e called out of sequence]
>>> delete from sip_dialogs where call_id='
>>> 8656841832142-120172129116107 at 10.64.1.2'
>>>
>>> 2010-02-16 16:42:00.802442 [ERR] switch_core_sqldb.c:722 SQL ERR: [select
>>> call_i
>>>
>>> d,sip_user,sip_host,contact,status,rpid,expires,user_agent,server_user,server_ho
>>>
>>> st,profile_name,hostname,network_ip,network_port,sip_username,sip_realm,mwi_user
>>> ,mwi_host from sip_registrations where profile_name='phones' and contact
>>> like '%
>>> 80635%'] library routine called out of sequence
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100216/b1e80285/attachment-0002.html 

From freeswitchlistreceiver at gmail.com  Mon Feb 15 23:41:09 2010
From: freeswitchlistreceiver at gmail.com (Thomas Switch)
Date: Tue, 16 Feb 2010 08:41:09 +0100
Subject: [Freeswitch-users] Call Monitoring and Recording with FS?
Message-ID: <106852de1002152341v5ec56da9s2fc30b580a989a18@mail.gmail.com>

Hello FreeSWITCH folks,

I was asked to join a project in the VoIP field. Being a newbie to VoIP, I
read a couple of books, many web pages and came across FreeSWITCH.

Hope you don't mind answering two questions of mine:


a) Similar to a call centre application, I'd need to record *all*conversation.
Like "For quality assurance, all conversations will be recorded..."... Could
I do that with FS or do I need an additional piece of software or hardware?
If I cannot record all conversation easily, I might be able to water the
requirement down to an "Operator monitoring a call, can record the call on
demand" (see also the second question)? What about that?

b) As an operator, I need to be able to monitor any call.
I understand, that one can get the active connection from FS. Is there a
possibility to get into these calls?
Or do I need to hack a "standard" call silently into a conference call with
the operator? If yes, is it possible to do that without the participants in
the call noticing it?

Thanks a lot for your time and patience.

Daniel
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100216/f4ed72f7/attachment-0002.html 

From Russell.Mosemann at cune.org  Tue Feb 16 08:13:09 2010
From: Russell.Mosemann at cune.org (Russell.Mosemann at cune.org)
Date: Tue, 16 Feb 2010 16:13:09 -0000
Subject: [Freeswitch-users] Call Monitoring and Recording with FS?
In-Reply-To: <106852de1002152341v5ec56da9s2fc30b580a989a18@mail.gmail.com>
Message-ID: <20100216161309.5D0D6156285@cuneorg-email.cune.pri>

Thomas Switch <freeswitchlistreceiver at gmail.com> said:

> I was asked to join a project in the VoIP field. Being a newbie to VoIP, I
> read a couple of books, many web pages and came across FreeSWITCH.

Well, then, you are primed and ready for this.

http://wiki.freeswitch.org/

-- 
Russell Mosemann



________________________________________________________
Concordia University, Nebraska
See http://www.cune.edu/ for the latest news and events!




From rupa at rupa.com  Tue Feb 16 08:22:06 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Tue, 16 Feb 2010 10:22:06 -0600
Subject: [Freeswitch-users] Call Monitoring and Recording with FS?
In-Reply-To: <106852de1002152341v5ec56da9s2fc30b580a989a18@mail.gmail.com>
References: <106852de1002152341v5ec56da9s2fc30b580a989a18@mail.gmail.com>
Message-ID: <b39273531002160822s52e2b2e1u86c734c47f5e904e@mail.gmail.com>

a) http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session
<http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session>(I'd
suggest recording to ${uuid}.wav, you can then use the uuid from the CDR to
find the wav file)

b) http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_eavesdrop

<http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_eavesdrop>The second
(at least) is in the default configs

On Tue, Feb 16, 2010 at 1:41 AM, Thomas Switch <
freeswitchlistreceiver at gmail.com> wrote:

> Hello FreeSWITCH folks,
>
> I was asked to join a project in the VoIP field. Being a newbie to VoIP, I
> read a couple of books, many web pages and came across FreeSWITCH.
>
> Hope you don't mind answering two questions of mine:
>
>
> a) Similar to a call centre application, I'd need to record *all*conversation.
> Like "For quality assurance, all conversations will be recorded..."...
> Could I do that with FS or do I need an additional piece of software or
> hardware?
> If I cannot record all conversation easily, I might be able to water the
> requirement down to an "Operator monitoring a call, can record the call on
> demand" (see also the second question)? What about that?
>
> b) As an operator, I need to be able to monitor any call.
> I understand, that one can get the active connection from FS. Is there a
> possibility to get into these calls?
> Or do I need to hack a "standard" call silently into a conference call with
> the operator? If yes, is it possible to do that without the participants in
> the call noticing it?
>
> Thanks a lot for your time and patience.
>
> Daniel
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100216/041a8ac2/attachment-0002.html 

From rob4manhere at gmail.com  Tue Feb 16 08:22:52 2010
From: rob4manhere at gmail.com (Rob Forman)
Date: Tue, 16 Feb 2010 10:22:52 -0600
Subject: [Freeswitch-users] Call Monitoring and Recording with FS?
In-Reply-To: <106852de1002152341v5ec56da9s2fc30b580a989a18@mail.gmail.com>
References: <106852de1002152341v5ec56da9s2fc30b580a989a18@mail.gmail.com>
Message-ID: <AB27AF06-74B2-49FE-B943-CD5F007A3F13@gmail.com>

Hi Daniel,

FreeSWITCH can definitely handle those requirements, and more, without  
hacking.  I would encourage you to search around the wiki.

For recording:
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session

You can enable it on any and all calls via your dialplan.

For call monitoring:
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_eavesdrop


Good luck!
Rob

On Feb 16, 2010, at 1:41 AM, Thomas Switch wrote:

> Hello FreeSWITCH folks,
>
> I was asked to join a project in the VoIP field. Being a newbie to  
> VoIP, I read a couple of books, many web pages and came across  
> FreeSWITCH.
>
> Hope you don't mind answering two questions of mine:
>
>
> a) Similar to a call centre application, I'd need to record all  
> conversation.
> Like "For quality assurance, all conversations will be  
> recorded..."... Could I do that with FS or do I need an additional  
> piece of software or hardware?
> If I cannot record all conversation easily, I might be able to water  
> the requirement down to an "Operator monitoring a call, can record  
> the call on demand" (see also the second question)? What about that?
>
> b) As an operator, I need to be able to monitor any call.
> I understand, that one can get the active connection from FS. Is  
> there a possibility to get into these calls?
> Or do I need to hack a "standard" call silently into a conference  
> call with the operator? If yes, is it possible to do that without  
> the participants in the call noticing it?
>
> Thanks a lot for your time and patience.
>
> Daniel
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100216/74955172/attachment-0002.html 

From brian at freeswitch.org  Tue Feb 16 08:23:40 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 16 Feb 2010 10:23:40 -0600
Subject: [Freeswitch-users] Call Monitoring and Recording with FS?
In-Reply-To: <106852de1002152341v5ec56da9s2fc30b580a989a18@mail.gmail.com>
References: <106852de1002152341v5ec56da9s2fc30b580a989a18@mail.gmail.com>
Message-ID: <23F6840C-4278-4A86-8615-E8D69A3147C8@freeswitch.org>

Word of advice... our community is a unique one.  Just don't take anything personally and roll with the punches.  You'll fit right in if you take that advice.

/b
PS:  We have you now... you'll be hooked before long. muahahahah

On Feb 16, 2010, at 1:41 AM, Thomas Switch wrote:

> Hello FreeSWITCH folks,
> 
> I was asked to join a project in the VoIP field. Being a newbie to VoIP, I read a couple of books, many web pages and came across FreeSWITCH. 
> 
> Hope you don't mind answering two questions of mine:
> 
> 
> a) Similar to a call centre application, I'd need to record all conversation. 
> Like "For quality assurance, all conversations will be recorded..."... Could I do that with FS or do I need an additional piece of software or hardware? 
> If I cannot record all conversation easily, I might be able to water the requirement down to an "Operator monitoring a call, can record the call on demand" (see also the second question)? What about that?
> 
> b) As an operator, I need to be able to monitor any call. 
> I understand, that one can get the active connection from FS. Is there a possibility to get into these calls?
> Or do I need to hack a "standard" call silently into a conference call with the operator? If yes, is it possible to do that without the participants in the call noticing it?
> 
> Thanks a lot for your time and patience.
> 
> Daniel

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100216/f3e15fa0/attachment-0002.html 

From moizchinoy at gmail.com  Tue Feb 16 08:35:17 2010
From: moizchinoy at gmail.com (Moiz Chinoy)
Date: Tue, 16 Feb 2010 20:35:17 +0400
Subject: [Freeswitch-users] Dingaling - external rtp supported?
Message-ID: <29b888f81002160835y6e290319ub0d04c088caa1b79@mail.gmail.com>

Hi All,

In mod_dingaling <param name="ext-rtp-ip"
value="$${external_rtp_ip}"/> supported? Whenever I uncomment this
line in client.xml (jingle profile) FS crashes as soon a call lands
(sip call) and dialplan bridges the call to a gtalk user.

I am running FS on windows and build is 16642.

-- 
Regards,
Moiz Chinoy.



From brian at freeswitch.org  Tue Feb 16 08:41:04 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 16 Feb 2010 10:41:04 -0600
Subject: [Freeswitch-users] Dingaling - external rtp supported?
In-Reply-To: <29b888f81002160835y6e290319ub0d04c088caa1b79@mail.gmail.com>
References: <29b888f81002160835y6e290319ub0d04c088caa1b79@mail.gmail.com>
Message-ID: <DEA39AD1-0EE9-473A-B78C-CDDF3F100578@freeswitch.org>

can you please update, try again and post a jira?

/b

On Feb 16, 2010, at 10:35 AM, Moiz Chinoy wrote:

> Hi All,
> 
> In mod_dingaling <param name="ext-rtp-ip"
> value="$${external_rtp_ip}"/> supported? Whenever I uncomment this
> line in client.xml (jingle profile) FS crashes as soon a call lands
> (sip call) and dialplan bridges the call to a gtalk user.
> 
> I am running FS on windows and build is 16642.
> 
> -- 
> Regards,
> Moiz Chinoy.




From jerry.richards at teotech.com  Tue Feb 16 09:11:31 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Tue, 16 Feb 2010 09:11:31 -0800
Subject: [Freeswitch-users] Presence PUBLISH Not Updating AfterSoftphone
	OffLine Then Available
In-Reply-To: <191c3a031002091557jfc9b486lb832ea538883bce9@mail.gmail.com>
References: <31FEE9072F104AE18B26812978FCA341@greyhawk.tonecommander.com><45963EB5-DFE5-43FA-AFCC-857E1E719E2D@jerris.com>
	<191c3a031002091557jfc9b486lb832ea538883bce9@mail.gmail.com>
Message-ID: <68A4C011A72B4ABA91A185F32480EB99@greyhawk.tonecommander.com>

I got version freeswitch-1.0.5-20100215-0400, built it, and ran it, and I am
seeing the same issue.  That is, once I set the Bria softphone status to
'Appear Offline', FS does not forward presence states until resubscription
time (i.e. tens of minutes later).  I posted a trace at
http://pastebin.freeswitch.org/12164.  At line 359 of the trace, FS is
logging an ERR at sofia_presence.c:662.  Here is the scenario:
 
1) Set Bria softphone presence state to 'Appear Offline'
2) Subscibing softphones reflect offline status
3) Set Bria softphone presence state to 'Available'
4) *** Subscibing softphones do not get status update ***
 
Thanks And Best Regards,
Jerry
 


  _____  

From: Anthony Minessale [mailto:anthony.minessale at gmail.com] 
Sent: Tuesday, February 09, 2010 3:58 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Presence PUBLISH Not Updating AfterSoftphone
OffLine Then Available


he means update to trunk first then try it again obviously.



On Tue, Feb 9, 2010 at 3:10 PM, Michael Jerris <mike at jerris.com> wrote:


Try this again, I think I saw changes go in for this issue.

Mike


On Feb 5, 2010, at 2:38 PM, Jerry Richards wrote:

> I found a scenario where presence status is not distributed to
subscribers.
> This is using the latest changes (as of Feb 03, 2010).  The scenario
> follows:
>
> 1) Register two Bria softphones (A and B), which each have the other as a
> contact.
> 2) Set softphone B's presence status to 'Appear Offline'.
> 3) Softphone A correctly reflects contact B is offline.
> 4) Set softphone B's presence status to 'Available'.
> 5) ******* There is no change to contact B's status at softphone A *******
>
> I posted a log at http://pastebin.freeswitch.org/12054.  At line 773,
there
> is an error when FS is processing the PUBLISH from softphone B:
>
> 773.2010-02-05 10:29:21.254221 [ERR] sofia_presence.c:674 DUMP PRESENCE
SQL:
>
> I did notice that after about 30 minutes, softphone B's status gets
> reflected at softphone A.



_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org





-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
<mailto:MSN%3Aanthony_minessale at hotmail.com> 
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
<mailto:PAYPAL%3Aanthony.minessale at gmail.com> 
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:+19193869900


-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100216/b3237ee4/attachment-0002.html 

From msc at freeswitch.org  Tue Feb 16 09:31:14 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 16 Feb 2010 09:31:14 -0800
Subject: [Freeswitch-users] Call Monitoring and Recording with FS?
In-Reply-To: <106852de1002152341v5ec56da9s2fc30b580a989a18@mail.gmail.com>
References: <106852de1002152341v5ec56da9s2fc30b580a989a18@mail.gmail.com>
Message-ID: <87f2f3b91002160931g7fac3f52m4ecd95b617eece87@mail.gmail.com>

Daniel,

Welcome to the wonderful and crazy world of VoIP! FreeSWITCH is totally
awesome and can do all sorts of things. It is like a cross between a Hemi
and a box of Lego bricks: it is extremely powerful and you can build all
sorts of things with it. Here's the caveat: there probably isn't a
pre-rolled solution for your setup. That being said, if you have any
programming skills, or if you have access to some IT resources at your firm,
then you can probably build something for your enterprise. Alternatively,
you can email consulting at freeswitch.org and seek professional (i.e. paid)
help.

My recommendation is to learn more about FreeSWITCH from these resources:
wiki.freeswitch.org
lists.freeswitch.org (freeswitch-users is the main list)
#freeswitch channel on irc.freenode.net
Community conf call on Fridays:
http://wiki.freeswitch.org/wiki/Weekly_Conference_Call

A very gentle intro to FreeSWITCH can be found here: http://bit.ly/EpVrv
After that then it's off to the wiki. There is a FreeSWITCH book in writing,
probably due out by summer time. In the meantime the community will be happy
to answer questions, especially if you roll up your sleeves and try things
before you ask.

Last pieces of advice: Use Linux distro CentOS 5.3, and use x86_64 if you
have 64bit hardware. We've seen crazy things happen with using 32bit Linux
on 64bit hardware. Also, use latest SVN trunk or download from
latest.freeswitch.org. FreeSWITCH is a unique project where the latest SVN
trunk is almost always more stable than the "latest stable" release. Hint:
if you want to update your FreeSWITCH installation to "the latest" then just
go to your source directory and type "make current" and it will do the rest.


Have fun!
-Michael

On Mon, Feb 15, 2010 at 11:41 PM, Thomas Switch <
freeswitchlistreceiver at gmail.com> wrote:

> Hello FreeSWITCH folks,
>
> I was asked to join a project in the VoIP field. Being a newbie to VoIP, I
> read a couple of books, many web pages and came across FreeSWITCH.
>
> Hope you don't mind answering two questions of mine:
>
>
> a) Similar to a call centre application, I'd need to record *all*conversation.
> Like "For quality assurance, all conversations will be recorded..."...
> Could I do that with FS or do I need an additional piece of software or
> hardware?
> If I cannot record all conversation easily, I might be able to water the
> requirement down to an "Operator monitoring a call, can record the call on
> demand" (see also the second question)? What about that?
>
> b) As an operator, I need to be able to monitor any call.
> I understand, that one can get the active connection from FS. Is there a
> possibility to get into these calls?
> Or do I need to hack a "standard" call silently into a conference call with
> the operator? If yes, is it possible to do that without the participants in
> the call noticing it?
>
> Thanks a lot for your time and patience.
>
> Daniel
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100216/67c9317f/attachment-0002.html 

From ivan at myrvold.org  Tue Feb 16 09:41:15 2010
From: ivan at myrvold.org (Ivan C Myrvold)
Date: Tue, 16 Feb 2010 18:41:15 +0100
Subject: [Freeswitch-users] mod_zeroconf
Message-ID: <8FFE625C-FFBE-4414-A95B-C54C1D21BFBF@myrvold.org>

How can I build FreeSWITCH with mod_zeroconf? I can't find any mod_zeroconf to uncomment in modules.conf

Ivan


From mrene_lists at avgs.ca  Tue Feb 16 09:44:55 2010
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Tue, 16 Feb 2010 12:44:55 -0500
Subject: [Freeswitch-users] mod_zeroconf
In-Reply-To: <8FFE625C-FFBE-4414-A95B-C54C1D21BFBF@myrvold.org>
References: <8FFE625C-FFBE-4414-A95B-C54C1D21BFBF@myrvold.org>
Message-ID: <45AE1465-DD7C-4101-A71A-B148A7DBCD74@avgs.ca>

mod_zeroconf has been moved to unsupported. you can still get it, cd  
to src/mod/applications/, type in svn co http://svn.freeswitch.org/svn/unsupported/mod_zeroconf/ 
  and then add a line in modules.conf it should then build  
auttomatically when you do make.

You can also speed that up and type make mod_zeroconf / make  
mod_zeroconf-install once the line in modules.conf has been added

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 16-Feb-10, at 12:41 PM, Ivan C Myrvold wrote:

> How can I build FreeSWITCH with mod_zeroconf? I can't find any  
> mod_zeroconf to uncomment in modules.conf
>
> Ivan
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From lists at redbonez.net  Tue Feb 16 10:03:52 2010
From: lists at redbonez.net (Adam Ford)
Date: Tue, 16 Feb 2010 11:03:52 -0700
Subject: [Freeswitch-users] Version 1.05 release
In-Reply-To: <191c3a031002151543q6a00a936vf7a5e9d4b69d0ebc@mail.gmail.com>
References: <mailman.6410.1266268685.2495.freeswitch-users@lists.freeswitch.org>	<054c01caae94$6131b1c0$23951540$@com>
	<191c3a031002151543q6a00a936vf7a5e9d4b69d0ebc@mail.gmail.com>
Message-ID: <005101caaf32$679da250$36d8e6f0$@net>

I'm sure you guys get this all the time, but I just wanted to throw in and
say I appreciate what all you FreeSWITCH devs have done in creating this
software.   Getting to learn and implement a FreeSWITCH system for our
office has been the most fun project I have had in years as a sys/network
admin.

 

-Adam

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Monday, February 15, 2010 4:43 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Version 1.05 release

 


Since you are advertising services on your website most likely all provided
by our free software, I hope you can learn some patience while you wait for
us to provide you with your next release to sell to your customers.

The release will be as soon as we make sure all the bugs are fixed, unless
you prefer it to be buggy.

BTW,

If your issues are still present in 1.0.5 and it's because you never
reported them, then we will, of course, be very unhappy.

If this is unacceptable we do offer a triple-your-money-back guarantee that
we will upgrade to quadruple if you act now.



On Mon, Feb 15, 2010 at 5:12 PM, Goran Donev
<gorand at donevtechconsulting.com> wrote:

I really didn't get a definitive answer on when 1.05 is slated to be
released. We are running into some issues that I hope that 1.05 fixes. Do we
have an eta?

Thx


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
<mailto:MSN%3Aanthony_minessale at hotmail.com> 
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
<mailto:PAYPAL%3Aanthony.minessale at gmail.com> 
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:+19193869900

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100216/594f00d7/attachment-0002.html 

From brian at freeswitch.org  Tue Feb 16 10:12:20 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 16 Feb 2010 12:12:20 -0600
Subject: [Freeswitch-users] Version 1.05 release
In-Reply-To: <005101caaf32$679da250$36d8e6f0$@net>
References: <mailman.6410.1266268685.2495.freeswitch-users@lists.freeswitch.org>	<054c01caae94$6131b1c0$23951540$@com>
	<191c3a031002151543q6a00a936vf7a5e9d4b69d0ebc@mail.gmail.com>
	<005101caaf32$679da250$36d8e6f0$@net>
Message-ID: <1C2E630C-E859-42ED-B5CE-BFF89212B71B@freeswitch.org>

Thank you... hope to see you at ClueCon this year too... Registration is open but I'm working out the last few minor details with the Trump hotel in Chicago. 

http://www.cluecon.com  (website is being worked on and updated as we move forward) 

Thanks,

/b

On Feb 16, 2010, at 12:03 PM, Adam Ford wrote:

> I?m sure you guys get this all the time, but I just wanted to throw in and say I appreciate what all you FreeSWITCH devs have done in creating this software.   Getting to learn and implement a FreeSWITCH system for our office has been the most fun project I have had in years as a sys/network admin.
>  
> -Adam
>  

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100216/a791c50f/attachment-0002.html 

From msc at freeswitch.org  Tue Feb 16 10:18:01 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 16 Feb 2010 10:18:01 -0800
Subject: [Freeswitch-users] Version 1.05 release
In-Reply-To: <005101caaf32$679da250$36d8e6f0$@net>
References: <mailman.6410.1266268685.2495.freeswitch-users@lists.freeswitch.org>
	<054c01caae94$6131b1c0$23951540$@com>
	<191c3a031002151543q6a00a936vf7a5e9d4b69d0ebc@mail.gmail.com>
	<005101caaf32$679da250$36d8e6f0$@net>
Message-ID: <87f2f3b91002161018m5b0e0517xda3a6e95467cd496@mail.gmail.com>

On Tue, Feb 16, 2010 at 10:03 AM, Adam Ford <lists at redbonez.net> wrote:

>  I?m sure you guys get this all the time, but I just wanted to throw in
> and say I appreciate what all you FreeSWITCH devs have done in creating this
> software.   Getting to learn and implement a FreeSWITCH system for our
> office has been the most fun project I have had in years as a sys/network
> admin.
>
> The devs don't get this kind of email often enough, so many thanks for your
recognition. The guys work extremely hard on FreeSWITCH and related
projects, not the least of which is ClueCon. Thanks for chiming in.

BTW, the dev dinner was great! Thanks to all who give active support to the
community.
-MC

>
>
> -Adam
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony
> Minessale
> *Sent:* Monday, February 15, 2010 4:43 PM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Version 1.05 release
>
>
>
>
> Since you are advertising services on your website most likely all provided
> by our free software, I hope you can learn some patience while you wait for
> us to provide you with your next release to sell to your customers.
>
> The release will be as soon as we make sure all the bugs are fixed, unless
> you prefer it to be buggy.
>
> BTW,
>
> If your issues are still present in 1.0.5 and it's because you never
> reported them, then we will, of course, be very unhappy.
>
> If this is unacceptable we do offer a triple-your-money-back guarantee that
> we will upgrade to quadruple if you act now.
>
>  On Mon, Feb 15, 2010 at 5:12 PM, Goran Donev <
> gorand at donevtechconsulting.com> wrote:
>
> I really didn't get a definitive answer on when 1.05 is slated to be
> released. We are running into some issues that I hope that 1.05 fixes. Do
> we
> have an eta?
>
> Thx
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100216/f76a00fd/attachment-0002.html 

From freeswitch at peely.com  Tue Feb 16 10:55:50 2010
From: freeswitch at peely.com (peely)
Date: Tue, 16 Feb 2010 10:55:50 -0800 (PST)
Subject: [Freeswitch-users] event-socket outbound: Dialplan failover on
 socket error?
Message-ID: <27613245.post@talk.nabble.com>


Hi,

I'd like to implement a dialplan entry to use a secondary event-socket
application if the first esl server is down or could not be connected to.

Could somebody please tell me what continue_on_fail options I could set so
that I will continue ONLY if the event-socket outbound connection was
unsuccessful i.e. an "[ERR] mod_event_socket.c:414 Socket Error!"? By
default it seems to continue if the outbound bridge is unsuccessful however
I don;t want to do that.

Alternatively, is there something I can set from the event-socket session to
stop the continue_on_fail from failing over?


Thanks,



Neil.
-- 
View this message in context: http://old.nabble.com/event-socket-outbound%3A-Dialplan-failover-on-socket-error--tp27613245p27613245.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From yehavi.bourvine at gmail.com  Tue Feb 16 10:59:26 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Tue, 16 Feb 2010 20:59:26 +0200
Subject: [Freeswitch-users] SQL ERR: library routine called out of
	sequence
In-Reply-To: <191c3a031002160755q569fe2ednfefb01b6d9cf7f0e@mail.gmail.com>
References: <d361ca7b1002160656w7d616f6dn5e80087ebde94dbe@mail.gmail.com>
	<191c3a031002160705w4050c3cdk2bf2922e9aff5ba3@mail.gmail.com>
	<d361ca7b1002160727ucb6e0c8id62e803757333213@mail.gmail.com>
	<191c3a031002160755q569fe2ednfefb01b6d9cf7f0e@mail.gmail.com>
Message-ID: <d361ca7b1002161059w1ef9484dr87e69a33aa2b62fc@mail.gmail.com>

The OS is Fedora-10 (soon to be upgraded to 12).

What I do when I want to test a new version:

   - Download the latest one into a fresh directory
   - bootstrap.sh, configure and make
   - stop Freeswitch, delete everything in lib, mod, bin ,db
   - make install and run it.


Is there additional place to clean?

                        Thanks! __Yehavi:

 2010/2/16 Anthony Minessale <anthony.minessale at gmail.com>

> you may want to do a clean wipe of all files related to FS then.
> you clearly have some problem with legacy something or other because we
> don't see that on dozens of dev boxes.
>
> What os is it?
>
>
>
> On Tue, Feb 16, 2010 at 9:27 AM, Yehavi Bourvine <
> yehavi.bourvine at gmail.com> wrote:
>
>>  Tried this, but it didn't help. I delete these DB files before any
>> upgrade just to be sure.
>>
>>                   Thanks! __Yehavi:
>>
>>  2010/2/16 Anthony Minessale <anthony.minessale at gmail.com>
>>
>>> try removing all the .db files from /usr/local/freeswitch/db
>>>
>>>
>>>   On Tue, Feb 16, 2010 at 8:56 AM, Yehavi Bourvine <
>>> yehavi.bourvine at gmail.com> wrote:
>>>
>>>>   Hello,
>>>>
>>>>   After upgrading from 1.0.5pre10 to 1.0.5-20100216-0400 we started
>>>> getting the above errors (I append bellow two samples). It seems Freeswitch
>>>> fails to read a database using Sqlite.
>>>> Anyone have seen this?
>>>>
>>>> Other details: Fedora 10, SQlite 3.5.9.
>>>> We also do SQLite quesries during call setup via LUA from CoreDB. Is it
>>>> an SQLite problem?
>>>>
>>>>                          Thanks! __Yehavi:
>>>>
>>>> The samples:
>>>> 2010-02-16 16:22:13.762679 [ERR] switch_core_sqldb.c:404 SQL ERR
>>>> [library routin
>>>> e called out of sequence]
>>>> delete from sip_dialogs where call_id='
>>>> 8656841832142-120172129116107 at 10.64.1.2'
>>>>
>>>> 2010-02-16 16:42:00.802442 [ERR] switch_core_sqldb.c:722 SQL ERR:
>>>> [select call_i
>>>>
>>>> d,sip_user,sip_host,contact,status,rpid,expires,user_agent,server_user,server_ho
>>>>
>>>> st,profile_name,hostname,network_ip,network_port,sip_username,sip_realm,mwi_user
>>>> ,mwi_host from sip_registrations where profile_name='phones' and contact
>>>> like '%
>>>> 80635%'] library routine called out of sequence
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>> iax:guest at conference.freeswitch.org/888
>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> pstn:+19193869900
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100216/c8a4c162/attachment-0002.html 

From anthony.minessale at gmail.com  Tue Feb 16 11:08:54 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 16 Feb 2010 13:08:54 -0600
Subject: [Freeswitch-users] event-socket outbound: Dialplan failover on
	socket error?
In-Reply-To: <27613245.post@talk.nabble.com>
References: <27613245.post@talk.nabble.com>
Message-ID: <191c3a031002161108j1f77925fpd58b4497b9b4d54a@mail.gmail.com>

continue_on_fail only applies to origination from the bridge app
the socket app should always continue to the next entry in the dp when it
fails.


On Tue, Feb 16, 2010 at 12:55 PM, peely <freeswitch at peely.com> wrote:

>
> Hi,
>
> I'd like to implement a dialplan entry to use a secondary event-socket
> application if the first esl server is down or could not be connected to.
>
> Could somebody please tell me what continue_on_fail options I could set so
> that I will continue ONLY if the event-socket outbound connection was
> unsuccessful i.e. an "[ERR] mod_event_socket.c:414 Socket Error!"? By
> default it seems to continue if the outbound bridge is unsuccessful however
> I don;t want to do that.
>
> Alternatively, is there something I can set from the event-socket session
> to
> stop the continue_on_fail from failing over?
>
>
> Thanks,
>
>
>
> Neil.
> --
> View this message in context:
> http://old.nabble.com/event-socket-outbound%3A-Dialplan-failover-on-socket-error--tp27613245p27613245.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100216/64686c89/attachment-0002.html 

From anthony.minessale at gmail.com  Tue Feb 16 11:10:12 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 16 Feb 2010 13:10:12 -0600
Subject: [Freeswitch-users] SQL ERR: library routine called out of
	sequence
In-Reply-To: <d361ca7b1002161059w1ef9484dr87e69a33aa2b62fc@mail.gmail.com>
References: <d361ca7b1002160656w7d616f6dn5e80087ebde94dbe@mail.gmail.com>
	<191c3a031002160705w4050c3cdk2bf2922e9aff5ba3@mail.gmail.com>
	<d361ca7b1002160727ucb6e0c8id62e803757333213@mail.gmail.com>
	<191c3a031002160755q569fe2ednfefb01b6d9cf7f0e@mail.gmail.com>
	<d361ca7b1002161059w1ef9484dr87e69a33aa2b62fc@mail.gmail.com>
Message-ID: <191c3a031002161110s637a48a2r717ae94557fe9e9a@mail.gmail.com>

That sounds about right.

That error usually has something to do with using db calls on a closed file
or something along those lines.
Maybe you have a permission problem on the directory where the db files are?


On Tue, Feb 16, 2010 at 12:59 PM, Yehavi Bourvine <yehavi.bourvine at gmail.com
> wrote:

> The OS is Fedora-10 (soon to be upgraded to 12).
>
> What I do when I want to test a new version:
>
>    - Download the latest one into a fresh directory
>    - bootstrap.sh, configure and make
>    - stop Freeswitch, delete everything in lib, mod, bin ,db
>    - make install and run it.
>
>
> Is there additional place to clean?
>
>                         Thanks! __Yehavi:
>
>  2010/2/16 Anthony Minessale <anthony.minessale at gmail.com>
>
>> you may want to do a clean wipe of all files related to FS then.
>> you clearly have some problem with legacy something or other because we
>> don't see that on dozens of dev boxes.
>>
>> What os is it?
>>
>>
>>
>> On Tue, Feb 16, 2010 at 9:27 AM, Yehavi Bourvine <
>> yehavi.bourvine at gmail.com> wrote:
>>
>>>  Tried this, but it didn't help. I delete these DB files before any
>>> upgrade just to be sure.
>>>
>>>                   Thanks! __Yehavi:
>>>
>>>  2010/2/16 Anthony Minessale <anthony.minessale at gmail.com>
>>>
>>>> try removing all the .db files from /usr/local/freeswitch/db
>>>>
>>>>
>>>>   On Tue, Feb 16, 2010 at 8:56 AM, Yehavi Bourvine <
>>>> yehavi.bourvine at gmail.com> wrote:
>>>>
>>>>>   Hello,
>>>>>
>>>>>   After upgrading from 1.0.5pre10 to 1.0.5-20100216-0400 we started
>>>>> getting the above errors (I append bellow two samples). It seems Freeswitch
>>>>> fails to read a database using Sqlite.
>>>>> Anyone have seen this?
>>>>>
>>>>> Other details: Fedora 10, SQlite 3.5.9.
>>>>> We also do SQLite quesries during call setup via LUA from CoreDB. Is it
>>>>> an SQLite problem?
>>>>>
>>>>>                          Thanks! __Yehavi:
>>>>>
>>>>> The samples:
>>>>> 2010-02-16 16:22:13.762679 [ERR] switch_core_sqldb.c:404 SQL ERR
>>>>> [library routin
>>>>> e called out of sequence]
>>>>> delete from sip_dialogs where call_id='
>>>>> 8656841832142-120172129116107 at 10.64.1.2'
>>>>>
>>>>> 2010-02-16 16:42:00.802442 [ERR] switch_core_sqldb.c:722 SQL ERR:
>>>>> [select call_i
>>>>>
>>>>> d,sip_user,sip_host,contact,status,rpid,expires,user_agent,server_user,server_ho
>>>>>
>>>>> st,profile_name,hostname,network_ip,network_port,sip_username,sip_realm,mwi_user
>>>>> ,mwi_host from sip_registrations where profile_name='phones' and
>>>>> contact like '%
>>>>> 80635%'] library routine called out of sequence
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>>
>>>> --
>>>> Anthony Minessale II
>>>>
>>>> FreeSWITCH http://www.freeswitch.org/
>>>> ClueCon http://www.cluecon.com/
>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>
>>>> AIM: anthm
>>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>> IRC: irc.freenode.net #freeswitch
>>>>
>>>> FreeSWITCH Developer Conference
>>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>>> iax:guest at conference.freeswitch.org/888
>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>> pstn:+19193869900
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100216/ae8618dc/attachment-0002.html 

From yehavi.bourvine at gmail.com  Tue Feb 16 11:30:33 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Tue, 16 Feb 2010 21:30:33 +0200
Subject: [Freeswitch-users] SQL ERR: library routine called out of
	sequence
In-Reply-To: <191c3a031002161110s637a48a2r717ae94557fe9e9a@mail.gmail.com>
References: <d361ca7b1002160656w7d616f6dn5e80087ebde94dbe@mail.gmail.com>
	<191c3a031002160705w4050c3cdk2bf2922e9aff5ba3@mail.gmail.com>
	<d361ca7b1002160727ucb6e0c8id62e803757333213@mail.gmail.com>
	<191c3a031002160755q569fe2ednfefb01b6d9cf7f0e@mail.gmail.com>
	<d361ca7b1002161059w1ef9484dr87e69a33aa2b62fc@mail.gmail.com>
	<191c3a031002161110s637a48a2r717ae94557fe9e9a@mail.gmail.com>
Message-ID: <d361ca7b1002161130k3eb2d88by3100b30a9d29bd51@mail.gmail.com>

Most of the queries are ok, only some fail, thus it doesn't look like
permission problem. Furthermore, under 1.0.5pre10 it works for months.

Might it be thread unsafe function calls? I've found the following while
searching the WEB:

*According to the MSDN docs, System.Timers.Timer operates in a thread pool.
If that's the case, your code is breaking the "connections cannot be shared
across threads" rule for SQLit*

Although it quotes MSDN, it might be related to Linux as well.

                    Thanks, __Yehavi:

 2010/2/16 Anthony Minessale <anthony.minessale at gmail.com>

> That sounds about right.
>
> That error usually has something to do with using db calls on a closed file
> or something along those lines.
> Maybe you have a permission problem on the directory where the db files
> are?
>
>
>
> On Tue, Feb 16, 2010 at 12:59 PM, Yehavi Bourvine <
> yehavi.bourvine at gmail.com> wrote:
>
>>  The OS is Fedora-10 (soon to be upgraded to 12).
>>
>> What I do when I want to test a new version:
>>
>>    - Download the latest one into a fresh directory
>>    - bootstrap.sh, configure and make
>>    - stop Freeswitch, delete everything in lib, mod, bin ,db
>>    - make install and run it.
>>
>>
>> Is there additional place to clean?
>>
>>                         Thanks! __Yehavi:
>>
>>  2010/2/16 Anthony Minessale <anthony.minessale at gmail.com>
>>
>>> you may want to do a clean wipe of all files related to FS then.
>>> you clearly have some problem with legacy something or other because we
>>> don't see that on dozens of dev boxes.
>>>
>>> What os is it?
>>>
>>>
>>>
>>> On Tue, Feb 16, 2010 at 9:27 AM, Yehavi Bourvine <
>>> yehavi.bourvine at gmail.com> wrote:
>>>
>>>>  Tried this, but it didn't help. I delete these DB files before any
>>>> upgrade just to be sure.
>>>>
>>>>                   Thanks! __Yehavi:
>>>>
>>>>  2010/2/16 Anthony Minessale <anthony.minessale at gmail.com>
>>>>
>>>>> try removing all the .db files from /usr/local/freeswitch/db
>>>>>
>>>>>
>>>>>   On Tue, Feb 16, 2010 at 8:56 AM, Yehavi Bourvine <
>>>>> yehavi.bourvine at gmail.com> wrote:
>>>>>
>>>>>>   Hello,
>>>>>>
>>>>>>   After upgrading from 1.0.5pre10 to 1.0.5-20100216-0400 we started
>>>>>> getting the above errors (I append bellow two samples). It seems Freeswitch
>>>>>> fails to read a database using Sqlite.
>>>>>> Anyone have seen this?
>>>>>>
>>>>>> Other details: Fedora 10, SQlite 3.5.9.
>>>>>> We also do SQLite quesries during call setup via LUA from CoreDB. Is
>>>>>> it an SQLite problem?
>>>>>>
>>>>>>                          Thanks! __Yehavi:
>>>>>>
>>>>>> The samples:
>>>>>> 2010-02-16 16:22:13.762679 [ERR] switch_core_sqldb.c:404 SQL ERR
>>>>>> [library routin
>>>>>> e called out of sequence]
>>>>>> delete from sip_dialogs where call_id='
>>>>>> 8656841832142-120172129116107 at 10.64.1.2'
>>>>>>
>>>>>> 2010-02-16 16:42:00.802442 [ERR] switch_core_sqldb.c:722 SQL ERR:
>>>>>> [select call_i
>>>>>>
>>>>>> d,sip_user,sip_host,contact,status,rpid,expires,user_agent,server_user,server_ho
>>>>>>
>>>>>> st,profile_name,hostname,network_ip,network_port,sip_username,sip_realm,mwi_user
>>>>>> ,mwi_host from sip_registrations where profile_name='phones' and
>>>>>> contact like '%
>>>>>> 80635%'] library routine called out of sequence
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> Anthony Minessale II
>>>>>
>>>>> FreeSWITCH http://www.freeswitch.org/
>>>>> ClueCon http://www.cluecon.com/
>>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>>
>>>>> AIM: anthm
>>>>> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>>> IRC: irc.freenode.net #freeswitch
>>>>>
>>>>> FreeSWITCH Developer Conference
>>>>> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>>>>> iax:guest at conference.freeswitch.org/888
>>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>>> pstn:+19193869900
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>> iax:guest at conference.freeswitch.org/888
>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> pstn:+19193869900
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100216/d3a65942/attachment-0002.html 

From anthony.minessale at gmail.com  Tue Feb 16 12:30:41 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 16 Feb 2010 14:30:41 -0600
Subject: [Freeswitch-users] SQL ERR: library routine called out of
	sequence
In-Reply-To: <d361ca7b1002161130k3eb2d88by3100b30a9d29bd51@mail.gmail.com>
References: <d361ca7b1002160656w7d616f6dn5e80087ebde94dbe@mail.gmail.com>
	<191c3a031002160705w4050c3cdk2bf2922e9aff5ba3@mail.gmail.com>
	<d361ca7b1002160727ucb6e0c8id62e803757333213@mail.gmail.com>
	<191c3a031002160755q569fe2ednfefb01b6d9cf7f0e@mail.gmail.com>
	<d361ca7b1002161059w1ef9484dr87e69a33aa2b62fc@mail.gmail.com>
	<191c3a031002161110s637a48a2r717ae94557fe9e9a@mail.gmail.com>
	<d361ca7b1002161130k3eb2d88by3100b30a9d29bd51@mail.gmail.com>
Message-ID: <191c3a031002161230h2b6e6f9wf3f65ddb71f08297@mail.gmail.com>

Strange, even on abusive testing we have not seen this problem.

please update to latest trunk.
There was only one change I can think of that may cause your issue and I
added a patch for it.
If it persists try setting the sql-in-transactions profile param to false.




On Tue, Feb 16, 2010 at 1:30 PM, Yehavi Bourvine
<yehavi.bourvine at gmail.com>wrote:

> Most of the queries are ok, only some fail, thus it doesn't look like
> permission problem. Furthermore, under 1.0.5pre10 it works for months.
>
> Might it be thread unsafe function calls? I've found the following while
> searching the WEB:
>
> *According to the MSDN docs, System.Timers.Timer operates in a thread
> pool.  If that's the case, your code is breaking the "connections cannot be
> shared across threads" rule for SQLit*
>
> Although it quotes MSDN, it might be related to Linux as well.
>
>                     Thanks, __Yehavi:
>
>  2010/2/16 Anthony Minessale <anthony.minessale at gmail.com>
>
>> That sounds about right.
>>
>> That error usually has something to do with using db calls on a closed
>> file or something along those lines.
>> Maybe you have a permission problem on the directory where the db files
>> are?
>>
>>
>>
>> On Tue, Feb 16, 2010 at 12:59 PM, Yehavi Bourvine <
>> yehavi.bourvine at gmail.com> wrote:
>>
>>>  The OS is Fedora-10 (soon to be upgraded to 12).
>>>
>>> What I do when I want to test a new version:
>>>
>>>    - Download the latest one into a fresh directory
>>>    - bootstrap.sh, configure and make
>>>    - stop Freeswitch, delete everything in lib, mod, bin ,db
>>>    - make install and run it.
>>>
>>>
>>> Is there additional place to clean?
>>>
>>>                         Thanks! __Yehavi:
>>>
>>>  2010/2/16 Anthony Minessale <anthony.minessale at gmail.com>
>>>
>>>> you may want to do a clean wipe of all files related to FS then.
>>>> you clearly have some problem with legacy something or other because we
>>>> don't see that on dozens of dev boxes.
>>>>
>>>> What os is it?
>>>>
>>>>
>>>>
>>>> On Tue, Feb 16, 2010 at 9:27 AM, Yehavi Bourvine <
>>>> yehavi.bourvine at gmail.com> wrote:
>>>>
>>>>>  Tried this, but it didn't help. I delete these DB files before any
>>>>> upgrade just to be sure.
>>>>>
>>>>>                   Thanks! __Yehavi:
>>>>>
>>>>>  2010/2/16 Anthony Minessale <anthony.minessale at gmail.com>
>>>>>
>>>>>> try removing all the .db files from /usr/local/freeswitch/db
>>>>>>
>>>>>>
>>>>>>   On Tue, Feb 16, 2010 at 8:56 AM, Yehavi Bourvine <
>>>>>> yehavi.bourvine at gmail.com> wrote:
>>>>>>
>>>>>>>   Hello,
>>>>>>>
>>>>>>>   After upgrading from 1.0.5pre10 to 1.0.5-20100216-0400 we started
>>>>>>> getting the above errors (I append bellow two samples). It seems Freeswitch
>>>>>>> fails to read a database using Sqlite.
>>>>>>> Anyone have seen this?
>>>>>>>
>>>>>>> Other details: Fedora 10, SQlite 3.5.9.
>>>>>>> We also do SQLite quesries during call setup via LUA from CoreDB. Is
>>>>>>> it an SQLite problem?
>>>>>>>
>>>>>>>                          Thanks! __Yehavi:
>>>>>>>
>>>>>>> The samples:
>>>>>>> 2010-02-16 16:22:13.762679 [ERR] switch_core_sqldb.c:404 SQL ERR
>>>>>>> [library routin
>>>>>>> e called out of sequence]
>>>>>>> delete from sip_dialogs where call_id='
>>>>>>> 8656841832142-120172129116107 at 10.64.1.2'
>>>>>>>
>>>>>>> 2010-02-16 16:42:00.802442 [ERR] switch_core_sqldb.c:722 SQL ERR:
>>>>>>> [select call_i
>>>>>>>
>>>>>>> d,sip_user,sip_host,contact,status,rpid,expires,user_agent,server_user,server_ho
>>>>>>>
>>>>>>> st,profile_name,hostname,network_ip,network_port,sip_username,sip_realm,mwi_user
>>>>>>> ,mwi_host from sip_registrations where profile_name='phones' and
>>>>>>> contact like '%
>>>>>>> 80635%'] library routine called out of sequence
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>> Anthony Minessale II
>>>>>>
>>>>>> FreeSWITCH http://www.freeswitch.org/
>>>>>> ClueCon http://www.cluecon.com/
>>>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>>>
>>>>>> AIM: anthm
>>>>>> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
>>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>>>> IRC: irc.freenode.net #freeswitch
>>>>>>
>>>>>> FreeSWITCH Developer Conference
>>>>>> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>>>>>> iax:guest at conference.freeswitch.org/888
>>>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>>>> pstn:+19193869900
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>>
>>>> --
>>>> Anthony Minessale II
>>>>
>>>> FreeSWITCH http://www.freeswitch.org/
>>>> ClueCon http://www.cluecon.com/
>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>
>>>> AIM: anthm
>>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>> IRC: irc.freenode.net #freeswitch
>>>>
>>>> FreeSWITCH Developer Conference
>>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>>> iax:guest at conference.freeswitch.org/888
>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>> pstn:+19193869900
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100216/2fcd1a2f/attachment-0002.html 

From brian at freeswitch.org  Tue Feb 16 12:36:45 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 16 Feb 2010 14:36:45 -0600
Subject: [Freeswitch-users] SQL ERR: library routine called out of
	sequence
In-Reply-To: <191c3a031002161230h2b6e6f9wf3f65ddb71f08297@mail.gmail.com>
References: <d361ca7b1002160656w7d616f6dn5e80087ebde94dbe@mail.gmail.com>
	<191c3a031002160705w4050c3cdk2bf2922e9aff5ba3@mail.gmail.com>
	<d361ca7b1002160727ucb6e0c8id62e803757333213@mail.gmail.com>
	<191c3a031002160755q569fe2ednfefb01b6d9cf7f0e@mail.gmail.com>
	<d361ca7b1002161059w1ef9484dr87e69a33aa2b62fc@mail.gmail.com>
	<191c3a031002161110s637a48a2r717ae94557fe9e9a@mail.gmail.com>
	<d361ca7b1002161130k3eb2d88by3100b30a9d29bd51@mail.gmail.com>
	<191c3a031002161230h2b6e6f9wf3f65ddb71f08297@mail.gmail.com>
Message-ID: <CBE5A464-0321-4A55-9195-7CAB0E3405AE@freeswitch.org>

What distro are you on and kernel version?

cat /proc/cpuinfo
uname -a

and such

/b

On Feb 16, 2010, at 2:30 PM, Anthony Minessale wrote:

> Strange, even on abusive testing we have not seen this problem.
> 
> please update to latest trunk.
> There was only one change I can think of that may cause your issue and I added a patch for it.
> If it persists try setting the sql-in-transactions profile param to false.
> 
> 
> 
> 
> On Tue, Feb 16, 2010 at 1:30 PM, Yehavi Bourvine <yehavi.bourvine at gmail.com> wrote:
> Most of the queries are ok, only some fail, thus it doesn't look like permission problem. Furthermore, under 1.0.5pre10 it works for months.
>  
> Might it be thread unsafe function calls? I've found the following while searching the WEB:
>  
> According to the MSDN docs, System.Timers.Timer operates in a thread pool.  If that's the case, your code is breaking the "connections cannot be shared across threads" rule for SQLit
>  

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100216/6bbf2331/attachment-0002.html 

From lists at redbonez.net  Tue Feb 16 13:39:35 2010
From: lists at redbonez.net (Adam Ford)
Date: Tue, 16 Feb 2010 14:39:35 -0700
Subject: [Freeswitch-users] Multiple versions of FreeSWITCH on one box
Message-ID: <00cb01caaf50$8a1dbe50$9e593af0$@net>

According to the wiki it is possible to run multiple FreeSWITCH instances on
one box.  Are there any known issues with having those instances be
different versions? 

 

Unfortunately I don't have a separate box to test the migration from 1.0.4
to 1.0.5, so I am looking to build a separate instance of 1.0.5 on the same
box.   

 

-Adam

 

 

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100216/db6524c3/attachment-0002.html 

From brian at freeswitch.org  Tue Feb 16 13:45:10 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 16 Feb 2010 15:45:10 -0600
Subject: [Freeswitch-users] Multiple versions of FreeSWITCH on one box
In-Reply-To: <00cb01caaf50$8a1dbe50$9e593af0$@net>
References: <00cb01caaf50$8a1dbe50$9e593af0$@net>
Message-ID: <E7A31EA6-D007-41D1-BACE-C113064E3860@freeswitch.org>

./configure --prefix=/usr/local/freeswitch-1.0.5

then edit for different ip's and ports 

/b

On Feb 16, 2010, at 3:39 PM, Adam Ford wrote:

> According to the wiki it is possible to run multiple FreeSWITCH instances on one box.  Are there any known issues with having those instances be different versions?
>  
> Unfortunately I don?t have a separate box to test the migration from 1.0.4 to 1.0.5, so I am looking to build a separate instance of 1.0.5 on the same box.   
>  
> -Adam

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100216/3bb746f0/attachment-0002.html 

From leo.zibi at gmail.com  Tue Feb 16 14:50:25 2010
From: leo.zibi at gmail.com (leo.zibi at gmail.com)
Date: Tue, 16 Feb 2010 23:50:25 +0100
Subject: [Freeswitch-users] Multiple versions of FreeSWITCH on one box
In-Reply-To: <00cb01caaf50$8a1dbe50$9e593af0$@net>
References: <00cb01caaf50$8a1dbe50$9e593af0$@net>
Message-ID: <4B7B2131.4010003@gmail.com>

An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100216/9f1997be/attachment-0002.html 

From lists at redbonez.net  Tue Feb 16 14:57:39 2010
From: lists at redbonez.net (Adam Ford)
Date: Tue, 16 Feb 2010 15:57:39 -0700
Subject: [Freeswitch-users] Multiple versions of FreeSWITCH on one box
In-Reply-To: <4B7B2131.4010003@gmail.com>
References: <00cb01caaf50$8a1dbe50$9e593af0$@net> <4B7B2131.4010003@gmail.com>
Message-ID: <00e401caaf5b$71cdfc10$5569f430$@net>

Yeah, that is what I was referencing when I said 'According to the wiki it
is possible..' My question was if it matter if they were different versions
of FreeSWITCH (1.0.4 and 1.0.5).  I will take this as a no.

 

Thanks you guys.

-Adam

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
leo.zibi at gmail.com
Sent: Tuesday, February 16, 2010 3:50 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Multiple versions of FreeSWITCH on one box

 

http://wiki.freeswitch.org/wiki/Deployment_Setup

Adam Ford wrote: 

According to the wiki it is possible to run multiple FreeSWITCH instances on
one box.  Are there any known issues with having those instances be
different versions? 

 

Unfortunately I don't have a separate box to test the migration from 1.0.4
to 1.0.5, so I am looking to build a separate instance of 1.0.5 on the same
box.   

 

-Adam

 

 

 

 



  _____  



 
_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
  

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100216/422c1747/attachment-0002.html 

From mike at jerris.com  Tue Feb 16 15:56:37 2010
From: mike at jerris.com (Michael Jerris)
Date: Tue, 16 Feb 2010 18:56:37 -0500
Subject: [Freeswitch-users] external_sip_address and
	external_rtp_address Question
In-Reply-To: <CDD4A2D8AA494ED8BF64401C5C953F72@greyhawk.tonecommander.com>
References: <CDD4A2D8AA494ED8BF64401C5C953F72@greyhawk.tonecommander.com>
Message-ID: <6D6A08D3-68FD-4B19-90A5-A19A9BC9100E@jerris.com>

not sure what those vars are, in the sip profiles we have ext-sip-ip and ext-rtp-ip.  They should be documented in the default configs and on the wiki.  These are per-profile settings, not per provider or gateway (unless of course you have a profile for each).

Mike

On Feb 15, 2010, at 3:49 PM, Jerry Richards wrote:

> I only see one example for setting of external_sip_address and
> external_rtp_address tags.  Is it true they are used to specify a SIP
> provider outside of a LAN (i.e. through a router)?  If so, then can these
> tags be set for each sip_profile?  So, if I have multiple external SIP
> providers that are accessed through NAT, they would each have their own
> external_sip_address and external_rtp_address?




From robert.hadley at teotech.com  Tue Feb 16 17:01:45 2010
From: robert.hadley at teotech.com (Robert Hadley)
Date: Tue, 16 Feb 2010 17:01:45 -0800
Subject: [Freeswitch-users] mod_fax receives fax to file but logs error msg
Message-ID: <02F74983D688435EA3F6954B48A14D6D@greyhawk.tonecommander.com>

I have been playing around with mod_fax and can successfully receive a fax
to file.  However, while doing so mod_fax is logging an error message.  Does
anybody know what this error means?

 

Dialplan: OpenZAP/2:1/1011 parsing [default->REH_test_fax_receive]
continue=false

Dialplan: OpenZAP/2:1/1011 Regex (PASS) [REH_test_fax_receive]
destination_number(1011) =~ /^1011$/ break=on-false

Dialplan: OpenZAP/2:1/1011 Action disable_ec()

Dialplan: OpenZAP/2:1/1011 Action answer()

Dialplan: OpenZAP/2:1/1011 Action playback(silence_stream://2000)

Dialplan: OpenZAP/2:1/1011 Action rxfax(/tmp/rxfax.tif)

Dialplan: OpenZAP/2:1/1011 Action hangup()

2010-02-16 16:27:08.542746 [DEBUG] switch_core_state_machine.c:122
(OpenZAP/2:1/1011) State Change CS_ROUTING -> CS_EXECUTE

2010-02-16 16:27:08.542746 [DEBUG] switch_core_session.c:1019 Send signal
OpenZAP/2:1/1011 [BREAK]

2010-02-16 16:27:08.542746 [DEBUG] switch_core_state_machine.c:341
(OpenZAP/2:1/1011) State ROUTING going to sleep

2010-02-16 16:27:08.542746 [DEBUG] switch_core_state_machine.c:314
(OpenZAP/2:1/1011) Running State Change CS_EXECUTE

2010-02-16 16:27:08.542746 [DEBUG] switch_core_state_machine.c:348
(OpenZAP/2:1/1011) State EXECUTE

2010-02-16 16:27:08.542746 [DEBUG] mod_openzap.c:434 OpenZAP/2:1/1011
CHANNEL EXECUTE

2010-02-16 16:27:08.542746 [DEBUG] switch_core_state_machine.c:159
OpenZAP/2:1/1011 Standard EXECUTE

2010-02-16 16:27:08.542746 [DEBUG] switch_core_session.c:1521 Application
disable_ec Requires media! pre_answering channel OpenZAP/2:1/1011

2010-02-16 16:27:08.542746 [DEBUG] switch_core_session.c:1523
OpenZAP/2:1/1011 receive message [PROGRESS]

2010-02-16 16:27:08.542746 [DEBUG] mod_openzap.c:960 Changing state on 2:1
from IDLE to UP

2010-02-16 16:27:08.542746 [NOTICE] mod_openzap.c:961 Channel
[OpenZAP/2:1/1011] has been answered

2010-02-16 16:27:08.542746 [DEBUG] switch_channel.c:182 OpenZAP/2:1/1011
receive message [AUDIO_SYNC]

2010-02-16 16:27:08.542746 [DEBUG] switch_core_session.c:634 Send signal
OpenZAP/2:1/1011 [BREAK]

EXECUTE OpenZAP/2:1/1011 disable_ec()

2010-02-16 16:27:08.542746 [INFO] mod_openzap.c:2951 Echo Canceller Disabled

EXECUTE OpenZAP/2:1/1011 answer()

EXECUTE OpenZAP/2:1/1011 playback(silence_stream://2000)

2010-02-16 16:27:08.542746 [DEBUG] switch_ivr_play_say.c:1162 Codec
Activated L16 at 8000hz 1 channels 20ms

2010-02-16 16:27:08.542746 [DEBUG] switch_core_io.c:652 OpenZAP/2:1/1011
receive message [TRANSCODING_NECESSARY]

2010-02-16 16:27:08.562747 [DEBUG] ozmod_analog.c:450 Executing state
handler on 2:1 for UP

2010-02-16 16:27:08.562747 [DEBUG] mod_openzap.c:1463 got FXS sig [UP]

2010-02-16 16:27:10.522277 [DEBUG] switch_ivr_play_say.c:1454 done playing
file

EXECUTE OpenZAP/2:1/1011 rxfax(/tmp/rxfax.tif)

2010-02-16 16:27:10.522277 [DEBUG] mod_fax.c:591 Raw read codec activation
Success L16 20000

2010-02-16 16:27:10.522277 [DEBUG] switch_core_codec.c:112 OpenZAP/2:1/1011
Push codec L16:10

2010-02-16 16:27:10.522277 [DEBUG] mod_fax.c:607 Raw write codec activation
Success L16

2010-02-16 16:27:10.522277 [DEBUG] switch_channel.c:182 OpenZAP/2:1/1011
receive message [AUDIO_SYNC]

2010-02-16 16:27:10.542275 [DEBUG] switch_core_io.c:234 OpenZAP/2:1/1011
receive message [TRANSCODING_NECESSARY]

2010-02-16 16:28:14.982013 [DEBUG] ozmod_analog.c:788 EVENT [ONHOOK][2:1]
STATE [UP]

2010-02-16 16:28:14.982013 [DEBUG] ozmod_analog.c:824 Changing state on 2:1
from UP to DOWN

2010-02-16 16:28:14.992014 [DEBUG] ozmod_analog.c:450 Executing state
handler on 2:1 for DOWN

2010-02-16 16:28:14.992014 [DEBUG] mod_openzap.c:1463 got FXS sig [STOP]

2010-02-16 16:28:14.992014 [NOTICE] mod_openzap.c:1554 Hangup
OpenZAP/2:1/1011 [CS_EXECUTE] [NORMAL_CLEARING]

2010-02-16 16:28:14.992014 [DEBUG] switch_channel.c:1976 Send signal
OpenZAP/2:1/1011 [KILL]

2010-02-16 16:28:15.002011 [ERR] mod_fax.c:666 Cannot write frame [datalen:
320, samples: 160]

2010-02-16 16:28:15.002011 [DEBUG] mod_fax.c:167
============================================================================
==

2010-02-16 16:28:15.002011 [DEBUG] mod_fax.c:174 Fax successfully received.

2010-02-16 16:28:15.002011 [DEBUG] mod_fax.c:185 Remote station id: 206 742
3831

2010-02-16 16:28:15.002011 [DEBUG] mod_fax.c:186 Local station id:  SpanDSP
Fax Ident

2010-02-16 16:28:15.002011 [DEBUG] mod_fax.c:187 Pages transferred: 1

2010-02-16 16:28:15.002011 [DEBUG] mod_fax.c:189 Total fax pages:   1

2010-02-16 16:28:15.002011 [DEBUG] mod_fax.c:190 Image resolution:
8031x3850

2010-02-16 16:28:15.002011 [DEBUG] mod_fax.c:191 Transfer Rate:     9600

2010-02-16 16:28:15.002011 [DEBUG] mod_fax.c:193 ECM status         off

2010-02-16 16:28:15.002011 [DEBUG] mod_fax.c:194 remote country:

2010-02-16 16:28:15.002011 [DEBUG] mod_fax.c:195 remote vendor:

2010-02-16 16:28:15.002011 [DEBUG] mod_fax.c:196 remote model:

2010-02-16 16:28:15.002011 [DEBUG] mod_fax.c:198
============================================================================
==

 

I traced the error message as coming from switch_core_session_write_frame in
src/switch_core_io.c which returns SWITCH_STATUS_FALSE.

 

Thanks for any information,

Robert

 

 

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100216/5d51c234/attachment-0002.html 

From brian at freeswitch.org  Tue Feb 16 17:07:29 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 16 Feb 2010 19:07:29 -0600
Subject: [Freeswitch-users] mod_fax receives fax to file but logs error
	msg
In-Reply-To: <02F74983D688435EA3F6954B48A14D6D@greyhawk.tonecommander.com>
References: <02F74983D688435EA3F6954B48A14D6D@greyhawk.tonecommander.com>
Message-ID: <80AC3154-4DC5-4432-BEF7-442C4DB47553@freeswitch.org>

Usually means what it says... I think thats harmless if the fax worked.

/b

On Feb 16, 2010, at 7:01 PM, Robert Hadley wrote:

> I have been playing around with mod_fax and can successfully receive a fax to file.  However, while doing so mod_fax is logging an error message.  Does anybody know what this error means?

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100216/a4d051e5/attachment-0002.html 

From anthony.minessale at gmail.com  Tue Feb 16 17:51:15 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 16 Feb 2010 19:51:15 -0600
Subject: [Freeswitch-users] Multiple versions of FreeSWITCH on one box
In-Reply-To: <00e401caaf5b$71cdfc10$5569f430$@net>
References: <00cb01caaf50$8a1dbe50$9e593af0$@net> <4B7B2131.4010003@gmail.com>
	<00e401caaf5b$71cdfc10$5569f430$@net>
Message-ID: <191c3a031002161751p719a177ah8acac55864817c37@mail.gmail.com>

it makes no difference at all what version is was as long as it has
alternate config paths and uses a different ip/port


On Tue, Feb 16, 2010 at 4:57 PM, Adam Ford <lists at redbonez.net> wrote:

>  Yeah, that is what I was referencing when I said ?According to the wiki
> it is possible?.? My question was if it matter if they were different
> versions of FreeSWITCH (1.0.4 and 1.0.5).  I will take this as a no.
>
>
>
> Thanks you guys.
>
> -Adam
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *
> leo.zibi at gmail.com
> *Sent:* Tuesday, February 16, 2010 3:50 PM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Multiple versions of FreeSWITCH on one
> box
>
>
>
> http://wiki.freeswitch.org/wiki/Deployment_Setup
>
> Adam Ford wrote:
>
> According to the wiki it is possible to run multiple FreeSWITCH instances
> on one box.  Are there any known issues with having those instances be
> different versions?
>
>
>
> Unfortunately I don?t have a separate box to test the migration from 1.0.4
> to 1.0.5, so I am looking to build a separate instance of 1.0.5 on the same
> box.
>
>
>
> -Adam
>
>
>
>
>
>
>
>
>
> ------------------------------
>
>
>
> _______________________________________________
>
> FreeSWITCH-users mailing list
>
> FreeSWITCH-users at lists.freeswitch.org
>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
> http://www.freeswitch.org
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100216/1160aa68/attachment-0002.html 

From anthony.minessale at gmail.com  Tue Feb 16 17:53:30 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 16 Feb 2010 19:53:30 -0600
Subject: [Freeswitch-users] Broadsoft SCA w/ Cisco SPA-5xx series
In-Reply-To: <8b1c9cda1002160738r619b2a3cs3bf5dd7d1322121e@mail.gmail.com>
References: <8b1c9cda1002141103te523643ob16db25e3bfa877f@mail.gmail.com>
	<0dd701caae7e$3c108b70$b431a250$@com>
	<8b1c9cda1002151253t32c22663wf66245e0220a4e6@mail.gmail.com>
	<191c3a031002151304h54c262c5o8ffd5c195e0512d2@mail.gmail.com>
	<0e7301caae84$58b47ce0$0a1d76a0$@com>
	<8b1c9cda1002151456s5dec6dbbtaa2063e0f70fe5dd@mail.gmail.com>
	<191c3a031002151531k148d8504w15e6f4c6c1e3ccd@mail.gmail.com>
	<8b1c9cda1002152225i7700bdfcgd2c112bf8180e36@mail.gmail.com>
	<191c3a031002160641q61cb1c53r38add08fc32f3a5a@mail.gmail.com>
	<8b1c9cda1002160738r619b2a3cs3bf5dd7d1322121e@mail.gmail.com>
Message-ID: <191c3a031002161753g27878c85nb20ebb86d9646dc1@mail.gmail.com>

ideally they should have the host name in the packets to match what your
configured domain is so when it does invites the domain is in the to from
etc.  your external phones are putting the ip in the packets which do not
match the domain name.

Another solution is to set the domain to be that IP addr in your config.



On Tue, Feb 16, 2010 at 9:38 AM, Gabriel Kuri <gkuri at ieee.org> wrote:

> The phones are currently setup with the domain in their "Proxy" field
> and set to use SRV to lookup the IP. The "Outbound Proxy" field is
> left empty. How should the phones be setup? The Proxy field with the
> domain and Outbound Proxy set to the IP?
>
> Thanks,
> Gabe
>
> On Tue, Feb 16, 2010 at 6:41 AM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
> > as I expected, you have IP addrs in the table which do not match your
> domain
> > name.
> > the phones behind nat should have your domain name in them same as the
> local
> > phones.
> > And the proxy addr should be set to the ip.
> >
> > If the IP and the DOMAIN do not match you will get mismatches.
> > Most people make the false assumption that this is like dns where the ip
> and
> > hostname are interchangeable.
> >
> > We can look at making a patch to force the hostname to always be the
> right
> > value in the db like we do for reg possibly.
> >
> >
> >
> > On Tue, Feb 16, 2010 at 12:25 AM, Gabriel Kuri <gkuri at ieee.org> wrote:
> >>
> >> Yeah, the domain name matches on the internal profile.
> >>
> >> Thanks for all your help, I can arrange ssh access tomorrow, today
> >> just wasn't one of those good days to do so, I've been running in and
> >> out too much to coordinate it.
> >>
> >> Here's the pastebin for the sip_dialogs table while the call is up ...
> >>
> >>     http://pastebin.freeswitch.org/12159
> >>
> >> Thanks,
> >> Gabe
> >>
> >>
> >> On Mon, Feb 15, 2010 at 3:31 PM, Anthony Minessale
> >> <anthony.minessale at gmail.com> wrote:
> >> > Do the domain names match on what the remote phones are using?
> >> >
> >> > When the call is active, can you attach to sqlite with the sqlite3 app
> >> > and
> >> > select * from sip_dialogs
> >> >
> >> > sqlite3 /usr/local/freeswitch/db/sofia_reg_internal.db
> >> >> select * from sip_dialogs;
> >> >
> >> > remember to do it while the call is up.
> >> >
> >> >
> >> > I am going to bet the domain name in that table is not the same as
> your
> >> > actual domain.
> >> >
> >> >
> >> > I tried to make this easier by asking to ssh to your box and work with
> >> > you
> >> > to fix it but now 9 hours later its starting to resemble diffusing a
> >> > bomb
> >> > over a telegraph wire.
> >> >
> >> >
> >> >
> >> >
> >> > On Mon, Feb 15, 2010 at 4:56 PM, Gabriel Kuri <gkuri at ieee.org> wrote:
> >> >>
> >> >> Yes, the two phones being called (SPA-509Gs) have SCA enabled and it
> >> >> works when making outgoing calls from those phones. But incoming
> calls
> >> >> to those two phones don't seem to have the line key light up on the
> >> >> other phone when one of the phones is answered (same extension).
> >> >>
> >> >> Thanks,
> >> >> Gabe
> >> >>
> >> >> On Mon, Feb 15, 2010 at 1:17 PM, Peder <peder at networkoblivion.com>
> >> >> wrote:
> >> >> > On the phone itself, do you have the line set to shared and
> >> >> > ?Broadsoft
> >> >> > SCA?
> >> >> > enabled?
> >> >> >
> >> >> >
> >> >> >
> >> >> > From: freeswitch-users-bounces at lists.freeswitch.org
> >> >> > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf
> Of
> >> >> > Anthony
> >> >> > Minessale
> >> >> > Sent: Monday, February 15, 2010 3:04 PM
> >> >> > To: freeswitch-users at lists.freeswitch.org
> >> >> > Subject: Re: [Freeswitch-users] Broadsoft SCA w/ Cisco SPA-5xx
> series
> >> >> >
> >> >> >
> >> >> >
> >> >> > you are missing something because you have no seize events when you
> >> >> > go
> >> >> > on
> >> >> > and off hook.
> >> >> > is every phone in the correct mode?
> >> >> >
> >> >> > On Mon, Feb 15, 2010 at 2:53 PM, Gabriel Kuri <gkuri at ieee.org>
> wrote:
> >> >> >
> >> >> > No, that was a typo. I have it correct in the config file.
> >> >> >
> >> >> > Gabe
> >> >> >
> >> >> > On Mon, Feb 15, 2010 at 12:34 PM, Peder <peder at networkoblivion.com
> >
> >> >> > wrote:
> >> >> >
> >> >> >> Is this a typo "managed-shared-appeareance=true" or is there an
> >> >> >> extra e
> >> >> >> in
> >> >> >> appearance in your config?
> >> >> >>
> >> >> >> -----Original Message-----
> >> >> >> From: freeswitch-users-bounces at lists.freeswitch.org
> >> >> >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf
> Of
> >> >> >> Gabriel
> >> >> >> Kuri
> >> >> >> Sent: Monday, February 15, 2010 1:48 PM
> >> >> >> To: freeswitch-users at lists.freeswitch.org
> >> >> >> Subject: Re: [Freeswitch-users] Broadsoft SCA w/ Cisco SPA-5xx
> >> >> >> series
> >> >> >>
> >> >> >> OK, I don't know what happened there, here's another one with the
> >> >> >> NOTIFYs. I'm on trunk rev 16633 and I have
> >> >> >> "managed-shared-appeareance=true" on the internal profile. I'm
> just
> >> >> >> making calls between internal phones.
> >> >> >>
> >> >> >>     http://pastebin.freeswitch.org/12153
> >> >> >>
> >> >> >> Thanks,
> >> >> >> Gabe
> >> >> >>
> >> >> >> On Mon, Feb 15, 2010 at 11:04 AM, Anthony Minessale
> >> >> >> <anthony.minessale at gmail.com> wrote:
> >> >> >>> I don't see any notifies at all in this trace do the profiles in
> >> >> >>> question
> >> >> >>> have:
> >> >> >>> manage-shared-appearance set to true?
> >> >> >>> and are you on latest trunk?
> >> >> >>>
> >> >> >>>
> >> >> >>> On Mon, Feb 15, 2010 at 12:55 PM, Anthony Minessale
> >> >> >>> <anthony.minessale at gmail.com> wrote:
> >> >> >>>>
> >> >> >>>> we log the sql stmts on err so they are red and easier to read.
> >> >> >>>>
> >> >> >>>>
> >> >> >>>>
> >> >> >>>> On Mon, Feb 15, 2010 at 12:48 PM, Gabriel Kuri <gkuri at ieee.org>
> >> >> >>>> wrote:
> >> >> >>>>>
> >> >> >>>>> Here's a call from ext 2551 to 2552 (two phones). I noticed a
> >> >> >>>>> bunch
> >> >> >>>>> of
> >> >> >>>>> errors related to SQL UPDATE for presence ...
> >> >> >>>>>
> >> >> >>>>>     http://pastebin.freeswitch.org/12152
> >> >> >>>>>
> >> >> >>>>> Thanks,
> >> >> >>>>> Gabe
> >> >> >>>>>
> >> >> >>>>>
> >> >> >>>>> On Mon, Feb 15, 2010 at 8:32 AM, Anthony Minessale
> >> >> >>>>> <anthony.minessale at gmail.com> wrote:
> >> >> >>>>> > it should be active not seized.
> >> >> >>>>> > seized is when you take it off hook.
> >> >> >>>>> >
> >> >> >>>>> > We need some more debugging to be sure.
> >> >> >>>>> > Can we work in real time on it or can you get a more detailed
> >> >> >>>>> > log?
> >> >> >>>>> >
> >> >> >>>>> > edit sofia.conf.xml and add the param to the "settings"
> >> >> >>>>> > section.
> >> >> >>>>> >
> >> >> >>>>> > <param name="debug-sla" value="100"/>
> >> >> >>>>> >
> >> >> >>>>> >
> >> >> >>>>> > then restart and enable sip trace and debug level
> >> >> >>>>> >
> >> >> >>>>> > //do this for every profile involved in the call.
> >> >> >>>>> > sofia profile <profilename> siptrace on
> >> >> >>>>> >
> >> >> >>>>> > //also do this
> >> >> >>>>> > console loglevel debug
> >> >> >>>>> >
> >> >> >>>>> >
> >> >> >>>>> > if you can let us ssh, we can do all the for you if you can
> >> >> >>>>> > make
> >> >> >>>>> > the
> >> >> >>>>> > test
> >> >> >>>>> > calls.
> >> >> >>>>> >
> >> >> >>>>> >
> >> >> >>>>> >
> >> >> >>>>> >
> >> >> >>>>> > On Sun, Feb 14, 2010 at 11:59 PM, Gabriel Kuri <
> gkuri at ieee.org>
> >> >> >>>>> > wrote:
> >> >> >>>>> >>
> >> >> >>>>> >> BTW, here's a copy of the NOTIFY (event call-info) sent to
> the
> >> >> >>>>> >> other
> >> >> >>>>> >> phone after the first phone is answered, should this have a
> >> >> >>>>> >> Call-Info
> >> >> >>>>> >> line with an "appearance-state=seized" to turn on the light
> on
> >> >> >>>>> >> the
> >> >> >>>>> >> other phone?
> >> >> >>>>> >>
> >> >> >>>>> >>
> >> >> >>>>> >> NOTIFY sip:2551@<removed>:54446 SIP/2.0.
> >> >> >>>>> >> Via: SIP/2.0/UDP
> >> >> >>>>> >> <removed>:9430;rport;branch=z9hG4bK71pN2cXgH851K.
> >> >> >>>>> >> Max-Forwards: 70.
> >> >> >>>>> >> From: <sip:2551 at mydomain.com <sip%3A2551 at mydomain.com>
> >;tag=XeB6ZrKDevpHp.
> >> >> >>>>> >> To: <sip:2551 at mydomain.com <sip%3A2551 at mydomain.com>
> >;tag=c2d34993aac6ea.
> >> >> >>>>> >> Call-ID: 34c34987-8b6fa786@<removed>.
> >> >> >>>>> >> CSeq: 126950830 NOTIFY.
> >> >> >>>>> >> Contact: <sip:2551@<removed>:9430>.
> >> >> >>>>> >> Expires: 3959.
> >> >> >>>>> >> Call-Info:
> >> >> >>>>> >> <sip:mydomain.com
> >;appearance-index=*;appearance-state=idle.
> >> >> >>>>> >> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16633M.
> >> >> >>>>> >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE,
> >> >> >>>>> >> INFO,
> >> >> >>>>> >> REGISTER, REFER, PRACK, NOTIFY, PUBLISH, SUBSCRIBE.
> >> >> >>>>> >> Supported: 100rel, timer, precondition, path, replaces.
> >> >> >>>>> >> Event: call-info.
> >> >> >>>>> >> Allow-Events: talk, presence, dialog, line-seize, call-info,
> >> >> >>>>> >> sla,
> >> >> >>>>> >> include-session-description, presence.winfo,
> message-summary,
> >> >> >>>>> >> refer.
> >> >> >>>>> >> Subscription-State: active;expires=3959.
> >> >> >>>>> >> Content-Length: 0.
> >> >> >>>>> >>
> >> >> >>>>> >>
> >> >> >>>>> >>
> >> >> >>>>> >> On Sun, Feb 14, 2010 at 7:58 PM, Brian West
> >> >> >>>>> >> <brian at freeswitch.org>
> >> >> >>>>> >> wrote:
> >> >> >>>>> >> > Works fine here... is your box slow or something?
> >> >> >>>>> >> >
> >> >> >>>>> >> > /b
> >> >> >>>>> >> >
> >> >> >>>>> >> > On Feb 14, 2010, at 1:03 PM, Gabriel Kuri wrote:
> >> >> >>>>> >> >
> >> >> >>>>> >> >> I followed Brian's directions from one of the previous
> >> >> >>>>> >> >> threads
> >> >> >>>>> >> >> on
> >> >> >>>>> >> >> configuring the SPA-5xx series phones for Broadsoft SCA
> and
> >> >> >>>>> >> >> set
> >> >> >>>>> >> >> manage-shared-appearance=true in the internal profile.
> SCA
> >> >> >>>>> >> >> appears
> >> >> >>>>> >> >> to
> >> >> >>>>> >> >> be working on outgoing calls between two phones, the line
> >> >> >>>>> >> >> key
> >> >> >>>>> >> >> starts
> >> >> >>>>> >> >> flashing red on the second phone when the first phone
> picks
> >> >> >>>>> >> >> up
> >> >> >>>>> >> >> the
> >> >> >>>>> >> >> receiver to make a call. However on incoming calls, both
> >> >> >>>>> >> >> phones
> >> >> >>>>> >> >> ring
> >> >> >>>>> >> >> (same extension), however when one of the phones picks up
> >> >> >>>>> >> >> the
> >> >> >> line,
> >> >> >>>>> >> >> the second phone's line key doesn't flash red or show the
> >> >> >>>>> >> >> first
> >> >> >>>>> >> >> phone
> >> >> >>>>> >> >> on that incoming call. Any ideas? Does shared appearance
> >> >> >>>>> >> >> only
> >> >> >>>>> >> >> work
> >> >> >>>>> >> >> on
> >> >> >>>>> >> >> outgoing phone calls?
> >> >> >>>>> >> >>
> >> >> >>>>> >> >> Thanks,
> >> >> >>>>> >> >> Gabe
> >> >> >>>>> >> >
> >> >> >>>>> >> >
> >> >> >>>>> >> > _______________________________________________
> >> >> >>>>> >> > FreeSWITCH-users mailing list
> >> >> >>>>> >> > FreeSWITCH-users at lists.freeswitch.org
> >> >> >>>>> >> >
> >> >> >>>>> >> >
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >>>>> >> >
> >> >> >>>>> >> >
> >> >> >>
> >> >> >>
> >> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >>>>> >> > http://www.freeswitch.org
> >> >> >>>>> >> >
> >> >> >>>>> >>
> >> >> >>>>> >> _______________________________________________
> >> >> >>>>> >> FreeSWITCH-users mailing list
> >> >> >>>>> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> >>>>> >>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >>>>> >>
> >> >> >>>>> >>
> >> >> >>
> >> >> >>
> >> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >>>>> >> http://www.freeswitch.org
> >> >> >>>>> >
> >> >> >>>>> >
> >> >> >>>>> >
> >> >> >>>>> > --
> >> >> >>>>> > Anthony Minessale II
> >> >> >>>>> >
> >> >> >>>>> > FreeSWITCH http://www.freeswitch.org/
> >> >> >>>>> > ClueCon http://www.cluecon.com/
> >> >> >>>>> > Twitter: http://twitter.com/FreeSWITCH_wire
> >> >> >>>>> >
> >> >> >>>>> > AIM: anthm
> >> >> >>>>> > MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
> >> >> >>>>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >> >> >>>>> > IRC: irc.freenode.net #freeswitch
> >> >> >>>>> >
> >> >> >>>>> > FreeSWITCH Developer Conference
> >> >> >>>>> > sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> >> >> >>>>> > iax:guest at conference.freeswitch.org/888
> >> >> >>>>> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >> >> >>>>> > pstn:+19193869900
> >> >> >>>>> >
> >> >> >>>>> > _______________________________________________
> >> >> >>>>> > FreeSWITCH-users mailing list
> >> >> >>>>> > FreeSWITCH-users at lists.freeswitch.org
> >> >> >>>>> >
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >>>>> >
> >> >> >>>>> >
> >> >> >>
> >> >> >>
> >> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >>>>> > http://www.freeswitch.org
> >> >> >>>>> >
> >> >> >>>>> >
> >> >> >>>>>
> >> >> >>>>> _______________________________________________
> >> >> >>>>> FreeSWITCH-users mailing list
> >> >> >>>>> FreeSWITCH-users at lists.freeswitch.org
> >> >> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >>>>>
> >> >> >>>>>
> >> >> >>>>>
> >> >> >>>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >>>>> http://www.freeswitch.org
> >> >> >>>>
> >> >> >>>>
> >> >> >>>>
> >> >> >>>> --
> >> >> >>>> Anthony Minessale II
> >> >> >>>>
> >> >> >>>> FreeSWITCH http://www.freeswitch.org/
> >> >> >>>> ClueCon http://www.cluecon.com/
> >> >> >>>> Twitter: http://twitter.com/FreeSWITCH_wire
> >> >> >>>>
> >> >> >>>> AIM: anthm
> >> >> >>>> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
> >> >> >>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >> >> >>>> IRC: irc.freenode.net #freeswitch
> >> >> >>>>
> >> >> >>>> FreeSWITCH Developer Conference
> >> >> >>>> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> >> >> >>>> iax:guest at conference.freeswitch.org/888
> >> >> >>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >> >> >>>> pstn:+19193869900
> >> >> >>>
> >> >> >>>
> >> >> >>>
> >> >> >>> --
> >> >> >>> Anthony Minessale II
> >> >> >>>
> >> >> >>> FreeSWITCH http://www.freeswitch.org/
> >> >> >>> ClueCon http://www.cluecon.com/
> >> >> >>> Twitter: http://twitter.com/FreeSWITCH_wire
> >> >> >>>
> >> >> >>> AIM: anthm
> >> >> >>> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
> >> >> >>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >> >> >>> IRC: irc.freenode.net #freeswitch
> >> >> >>>
> >> >> >>> FreeSWITCH Developer Conference
> >> >> >>> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> >> >> >>> iax:guest at conference.freeswitch.org/888
> >> >> >>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >> >> >>> pstn:+19193869900
> >> >> >>>
> >> >> >>> _______________________________________________
> >> >> >>> FreeSWITCH-users mailing list
> >> >> >>> FreeSWITCH-users at lists.freeswitch.org
> >> >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >>>
> >> >> >>>
> >> >> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >>> http://www.freeswitch.org
> >> >> >>>
> >> >> >>>
> >> >> >>
> >> >> >> _______________________________________________
> >> >> >> FreeSWITCH-users mailing list
> >> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >>
> >> >> >>
> >> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >> http://www.freeswitch.org
> >> >> >>
> >> >> >>
> >> >> >> _______________________________________________
> >> >> >> FreeSWITCH-users mailing list
> >> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >>
> >> >> >>
> >> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >> http://www.freeswitch.org
> >> >> >>
> >> >> >
> >> >> > _______________________________________________
> >> >> > FreeSWITCH-users mailing list
> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >
> >> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> > http://www.freeswitch.org
> >> >> >
> >> >> >
> >> >> > --
> >> >> > Anthony Minessale II
> >> >> >
> >> >> > FreeSWITCH http://www.freeswitch.org/
> >> >> > ClueCon http://www.cluecon.com/
> >> >> > Twitter: http://twitter.com/FreeSWITCH_wire
> >> >> >
> >> >> > AIM: anthm
> >> >> > MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
> >> >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >> >> > IRC: irc.freenode.net #freeswitch
> >> >> >
> >> >> > FreeSWITCH Developer Conference
> >> >> > sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> >> >> > iax:guest at conference.freeswitch.org/888
> >> >> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >> >> > pstn:+19193869900
> >> >> >
> >> >> > _______________________________________________
> >> >> > FreeSWITCH-users mailing list
> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >
> >> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> > http://www.freeswitch.org
> >> >> >
> >> >> >
> >> >>
> >> >> _______________________________________________
> >> >> FreeSWITCH-users mailing list
> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>
> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> http://www.freeswitch.org
> >> >
> >> >
> >> >
> >> > --
> >> > Anthony Minessale II
> >> >
> >> > FreeSWITCH http://www.freeswitch.org/
> >> > ClueCon http://www.cluecon.com/
> >> > Twitter: http://twitter.com/FreeSWITCH_wire
> >> >
> >> > AIM: anthm
> >> > MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
> >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >> > IRC: irc.freenode.net #freeswitch
> >> >
> >> > FreeSWITCH Developer Conference
> >> > sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> >> > iax:guest at conference.freeswitch.org/888
> >> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >> > pstn:+19193869900
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> > Twitter: http://twitter.com/FreeSWITCH_wire
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> > IRC: irc.freenode.net #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> > iax:guest at conference.freeswitch.org/888
> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > pstn:+19193869900
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100216/b3ccf694/attachment-0002.html 

From anthony.minessale at gmail.com  Tue Feb 16 18:08:38 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 16 Feb 2010 20:08:38 -0600
Subject: [Freeswitch-users] mod_fax receives fax to file but logs error
	msg
In-Reply-To: <80AC3154-4DC5-4432-BEF7-442C4DB47553@freeswitch.org>
References: <02F74983D688435EA3F6954B48A14D6D@greyhawk.tonecommander.com>
	<80AC3154-4DC5-4432-BEF7-442C4DB47553@freeswitch.org>
Message-ID: <191c3a031002161808u32282f7esbee131305a607786@mail.gmail.com>

when a channel is hungup the read and write will fail to stop the media,
this is typical.


On Tue, Feb 16, 2010 at 7:07 PM, Brian West <brian at freeswitch.org> wrote:

> Usually means what it says... I think thats harmless if the fax worked.
>
> /b
>
> On Feb 16, 2010, at 7:01 PM, Robert Hadley wrote:
>
> I have been playing around with mod_fax and can successfully receive a fax
> to file.  However, while doing so mod_fax is logging an error message.  Does
> anybody know what this error means?
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100216/1c9a5e9b/attachment-0002.html 

From infos at madovsky.org  Tue Feb 16 20:02:42 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 16 Feb 2010 23:02:42 -0500
Subject: [Freeswitch-users] freeswitch as proxy
Message-ID: <82FAF11BF2BB4A0787E86706EABF755E@MOBILEE1705>

Hi,

First I'd like to felicitate the huge work of the freeswitch founders, their patience and humbleness
(Anthony Minessale (is there 1 and 2 ?), Michael Jerris, Brian West and Others ) and people need to
know and understand how hard is to maintain open source for years...

well, maybe I didn't look for very well in archives and google so I hope my request won't be an old one....
I set freeswitch with proxy_media and blind reg and auth on true since I want everybody who uses my network
use a softphone to register their own sip account (whatever domain outside my network).

user at hisdomain -> myproxy IP -> hisdomain registrar -> confirm register and wait call.
user at hisdomain calls other at otherdomain everywhere in the world.
other at otherdomain can also call user at hisdomain

is it possible ? if yes is there any link example ? I guess maybe it's a dialplan rule....

Sorry I'm novice yet

Regards

Franck
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100216/49356588/attachment-0002.html 

From yehavi.bourvine at gmail.com  Tue Feb 16 20:49:30 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Wed, 17 Feb 2010 06:49:30 +0200
Subject: [Freeswitch-users] SQL ERR: library routine called out of
	sequence
In-Reply-To: <CBE5A464-0321-4A55-9195-7CAB0E3405AE@freeswitch.org>
References: <d361ca7b1002160656w7d616f6dn5e80087ebde94dbe@mail.gmail.com>
	<191c3a031002160705w4050c3cdk2bf2922e9aff5ba3@mail.gmail.com>
	<d361ca7b1002160727ucb6e0c8id62e803757333213@mail.gmail.com>
	<191c3a031002160755q569fe2ednfefb01b6d9cf7f0e@mail.gmail.com>
	<d361ca7b1002161059w1ef9484dr87e69a33aa2b62fc@mail.gmail.com>
	<191c3a031002161110s637a48a2r717ae94557fe9e9a@mail.gmail.com>
	<d361ca7b1002161130k3eb2d88by3100b30a9d29bd51@mail.gmail.com>
	<191c3a031002161230h2b6e6f9wf3f65ddb71f08297@mail.gmail.com>
	<CBE5A464-0321-4A55-9195-7CAB0E3405AE@freeswitch.org>
Message-ID: <d361ca7b1002162049y6810073cj55765ce9fdfc2f48@mail.gmail.com>

Hello,

  I'll try the latest snapshot during the weekend as this is a production
system.

I am using FedoraCore 10 with a kernel from kernel.org (as I recall there
was some issue with Freeswitch and Fedora's kernel). Here is the output of
uname and cpuinfo:



Linux control.huji.ac.il 2.6.32.5 #1 SMP Sat Jan 23 11:17:10 IST 2010 i686
i686 i386 GNU/Linux



processor   : 0

vendor_id   : GenuineIntel

cpu family  : 15

model       : 4

model name  : Intel(R) Xeon(TM) CPU 3.00GHz

stepping    : 1

cpu MHz           : 3000.000

cache size  : 1024 KB

physical id : 0

siblings    : 2

core id           : 0

cpu cores   : 1

apicid            : 0

initial apicid    : 0

fdiv_bug    : no

hlt_bug           : no

f00f_bug    : no

coma_bug    : no

fpu         : yes

fpu_exception     : yes

cpuid level : 5

wp          : yes

flags       : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca cmov
pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe lm constant_tsc
pebs bts pni dtes64 monitor ds_cpl cid cx16 xtpr

bogomips    : 6000.32

clflush size      : 64

cache_alignment   : 128

address sizes     : 36 bits physical, 48 bits virtual

power management:



processor   : 1

vendor_id   : GenuineIntel

cpu family  : 15

model       : 4

model name  : Intel(R) Xeon(TM) CPU 3.00GHz

stepping    : 1

cpu MHz           : 3000.000

cache size  : 1024 KB

physical id : 0

siblings    : 2

core id           : 0

cpu cores   : 1

apicid            : 1

initial apicid    : 1

fdiv_bug    : no

hlt_bug           : no

f00f_bug    : no

coma_bug    : no

fpu         : yes

fpu_exception     : yes

cpuid level : 5

wp          : yes

flags       : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca cmov
pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe lm constant_tsc
pebs bts pni dtes64 monitor ds_cpl cid cx16 xtpr

bogomips    : 5999.17

clflush size      : 64

cache_alignment   : 128

address sizes     : 36 bits physical, 48 bits virtual

power management:



                Thanks! __Yehavi:


 2010/2/16 Brian West <brian at freeswitch.org>

>  What distro are you on and kernel version?
>
> cat /proc/cpuinfo
> uname -a
>
> and such
>
> /b
>
>  On Feb 16, 2010, at 2:30 PM, Anthony Minessale wrote:
>
> Strange, even on abusive testing we have not seen this problem.
>
> please update to latest trunk.
> There was only one change I can think of that may cause your issue and I
> added a patch for it.
> If it persists try setting the sql-in-transactions profile param to false.
>
>
>
>
> On Tue, Feb 16, 2010 at 1:30 PM, Yehavi Bourvine <
> yehavi.bourvine at gmail.com> wrote:
>
>>  Most of the queries are ok, only some fail, thus it doesn't look like
>> permission problem. Furthermore, under 1.0.5pre10 it works for months.
>>
>> Might it be thread unsafe function calls? I've found the following while
>> searching the WEB:
>>
>> *According to the MSDN docs, System.Timers.Timer operates in a thread
>> pool.  If that's the case, your code is breaking the "connections cannot be
>> shared across threads" rule for SQLit*
>>
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100217/d02228a0/attachment-0002.html 

From anthony.minessale at gmail.com  Tue Feb 16 21:39:53 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 16 Feb 2010 23:39:53 -0600
Subject: [Freeswitch-users] Presence PUBLISH Not Updating AfterSoftphone
	OffLine Then Available
In-Reply-To: <68A4C011A72B4ABA91A185F32480EB99@greyhawk.tonecommander.com>
References: <31FEE9072F104AE18B26812978FCA341@greyhawk.tonecommander.com>
	<45963EB5-DFE5-43FA-AFCC-857E1E719E2D@jerris.com>
	<191c3a031002091557jfc9b486lb832ea538883bce9@mail.gmail.com>
	<68A4C011A72B4ABA91A185F32480EB99@greyhawk.tonecommander.com>
Message-ID: <191c3a031002162139p57c94046y660fb8d01fd76f46@mail.gmail.com>

You see one case at the top where it sends a notify and more where it doesnt
.  You have the sql stmts right there (they are not errs just logging in red
so they are obvious) run them manually and figure out why there are no
matches.  No subscriptions maybe?

Its beginning to sound like a broken record with so many bria isssues, its a
new software afterall and not free like we are, why must we support it so
much?

Also if you are actually concerned with this issue, maybe you can come back
sooner than once every week or 2 weeks.  We quickly lose track of threads
like this that linger for a month, that's what jira is for....

Maybe you can stop by irc or keep an eye on your email client so we can
confirm what you are doing wrong or if we have an interop with bria, a pay
softphone none of us have a copy of........

On Feb 16, 2010 11:18 AM, "Jerry Richards" <jerry.richards at teotech.com>
wrote:

 I got version freeswitch-1.0.5-20100215-0400, built it, and ran it, and I
am seeing the same issue.  That is, once I set the Bria softphone status to
'Appear Offline', FS does not forward presence states until resubscription
time (i.e. tens of minutes later).  I posted a trace at
http://pastebin.freeswitch.org/12164.  At line 359 of the trace, FS is
logging an ERR at sofia_presence.c:662.  Here is the scenario:

1) Set Bria softphone presence state to 'Appear Offline'
2) Subscibing softphones reflect offline status
 3) Set Bria softphone presence state to 'Available'
4) *** Subscibing softphones do not get status update ***

Thanks And Best Regards,
Jerry


 ------------------------------
*From:* Anthony Minessale [mailto:anthony.minessale at gmail.com]
*Sent:* Tuesday, February 09, 2010 3:58 PM
*To:* freeswitch-users at lists.freeswitch.org
*Subject:* Re: [Freeswitch-users] Presence PUBLISH Not Updating
AfterSoftphone OffLine Then Available

> he means update to trunk first then try it again obviously.
>
>
> On Tue, Feb 9, 2010 at 3:10 PM, ...


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100216/4841dd82/attachment-0002.html 

From ledoktre at meanie.us  Tue Feb 16 18:53:58 2010
From: ledoktre at meanie.us (Doc)
Date: Tue, 16 Feb 2010 20:53:58 -0600
Subject: [Freeswitch-users] Greetings and a couple of questions
Message-ID: <4B7B5A46.3090304@meanie.us>

First, to all, greetings.

I am just beginning a quest in trying to setup a simple FS box to route 
my incoming skype account to a SIP ATA (SPA-1001).

I have this installed on Ubuntu 8.04 (since I read that it came with a 
stock tickless kernel with 100HZ tick).  I have (as far as I can tell) 
compiled the alsa 1.0.20 drivers and included the mod_skypiax dummy 
file.  I followed install instructions in the FS wiki for Hardy & FS & 
skypiax.

1) I am able to see a call come in, and it gets routed to the sample IVR 
to start.  The first thing off is that when I dial from PSTN -> 
Skype-In, it does not let me push any buttons.  If I launch a second 
skype client, and dial the skype user on FS directly, it works fine.  
Any ideas?

2) No matter how Skype works, when I hang up, it throws 3 or 4 errors 
(2010-02-16 20:41:45.384633 [ERR] mod_skypiax.c:826 rev 
16619M[(nil)|37     ][ERRORA  826  ][interface1][-1, 5,21] EXIT? 
sent=-1), and when I try subsequent dials, I get pages and pages of this 
error, and no audio.  Eventually it crashes FS.  Any ideas?

3) When dialing extension to extension, or even testing out the IVR, it 
all works fine - no errors.  The only thing it does make me do, and I 
haven't tracked it down yet is it waits like 30 seconds before 
responding (unless I press the # after I type the extension number).  
Any ideas?

4) When I run startskype.sh, I see these errors (and are they worth 
concerning?) :

expected keysym, got XF86KbdLightOnOff: line 70 of pc
expected keysym, got XF86KbdBrightnessDown: line 71 of pc
expected keysym, got XF86KbdBrightnessUp: line 72 of pc
Could not init font path element /usr/share/fonts/X11/cyrillic, removing 
from list!
Could not init font path element 
/var/lib/defoma/x-ttcidfont-conf.d/dirs/TrueType, removing from list!
ALSA lib pcm.c:2211:(snd_pcm_open_noupdate) Unknown PCM null
ALSA lib pcm.c:2211:(snd_pcm_open_noupdate) Unknown PCM null

Thanks for the patience in letting me email to the group.  I used 
Asterisk in the past, and even though I am finding a bit of a learning 
curve using FS, I am enjoying it.  Hopefully someone will have some 
insight at least into where or how I can settle the above items down, 
and I can start building a proper dialplan.

Thanks,

Doc



From ledoktre at meanie.us  Tue Feb 16 19:13:45 2010
From: ledoktre at meanie.us (Doc)
Date: Tue, 16 Feb 2010 21:13:45 -0600
Subject: [Freeswitch-users] One more thing..
Message-ID: <4B7B5EE9.4020705@meanie.us>

Greetings one more time,

I just remembered.  I also ran into one other issue in my testing.

When I route the incoming skype call to an extension (Sipura SPA 1001), 
it plays hold music on the callers side (good....), and when I pick up 
the extension, calling party drops, internal phone is left with a fast 
busy signal :

2010-02-16 21:09:31.687204 [NOTICE] sofia.c:4690 Channel 
[sofia/internal/sip:1001 at 10.24.72.12:5060] has been answered
2010-02-16 21:09:31.987714 [WARNING] mod_sofia.c:918 We were told to use 
ptime 3 but what they meant to say was 20
This issue has so far been identified to happen on the following broken 
platforms/devices:
Linksys/Sipura aka Cisco
ShoreTel
Sonus/L3
We will try to fix it but some of the devices on this list are so broken 
who knows what will happen..
2010-02-16 21:09:32.027243 [ERR] switch_core_io.c:121 
sofia/internal/sip:1001 at 10.24.72.12:5060 has no read codec.

First thing I notice is this unusual error about the ptime?  and the one 
that hangs me up I think is the "sofia... has no read codec".  I have 
been poking around my configuration - any suggestions?

Thanks,

Doc




From moizchinoy at gmail.com  Wed Feb 17 03:18:54 2010
From: moizchinoy at gmail.com (Moiz Chinoy)
Date: Wed, 17 Feb 2010 15:18:54 +0400
Subject: [Freeswitch-users] Dingaling - external rtp supported?
In-Reply-To: <DEA39AD1-0EE9-473A-B78C-CDDF3F100578@freeswitch.org>
References: <29b888f81002160835y6e290319ub0d04c088caa1b79@mail.gmail.com>
	<DEA39AD1-0EE9-473A-B78C-CDDF3F100578@freeswitch.org>
Message-ID: <29b888f81002170318p63622f3fhc8d26ad0e85602d2@mail.gmail.com>

Hi,

FS rev: 16673
Platform: Windows

More details:

FS is behind NAT and machine is running a VPN connection.

FS and GTalk client on the same machine:
--------------------------------------------------------------------------------------------------
jingle profile client.xml has following line:
<param name="rtp-ip" value="$${bind_server_ip}"/>

External SIP call is successfully bridged to GTalk client.


FS and GTalk client on the different machine:
--------------------------------------------------------------------------------------------------
jingle profile client.xml has following lines:
<param name="rtp-ip" value="$${bind_server_ip}"/>
<param name="ext-rtp-ip" value="$${external_rtp_ip}"/>

As soon as external SIP call land and I try to bridge the call to
GTalk client, FS crashes.


NAT Details:
---------------------------
I think my NAT does not support UpNP or PMP. The reason I say it
because when FS starts following message is displayed:

2010-02-17 09:15:42.809674 [INFO] switch_nat.c:409 Scanning for NAT
2010-02-17 09:15:42.809674 [ERR] switch_nat.c:197 Error checking for
PMP [init failed]
2010-02-17 09:15:42.809674 [DEBUG] switch_nat.c:414 Checking for UPnP
2010-02-17 09:15:56.825388 [DEBUG] switch_nat.c:114 No
InternetGatewayDevice, using first entry as default
(http://192.168.16.17:50144/).
2010-02-17 09:15:56.887889 [INFO] switch_nat.c:429 No PMP or UPnP NAT
devices detected!



On Tue, Feb 16, 2010 at 8:41 PM, Brian West <brian at freeswitch.org> wrote:
> can you please update, try again and post a jira?
>
> /b
>
> On Feb 16, 2010, at 10:35 AM, Moiz Chinoy wrote:
>
>> Hi All,
>>
>> In mod_dingaling <param name="ext-rtp-ip"
>> value="$${external_rtp_ip}"/> supported? Whenever I uncomment this
>> line in client.xml (jingle profile) FS crashes as soon a call lands
>> (sip call) and dialplan bridges the call to a gtalk user.
>>
>> I am running FS on windows and build is 16642.
>>
>> --
>> Regards,
>> Moiz Chinoy.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Regards,
Moiz Chinoy.



From scott.torr.fs at letterboxes.org  Wed Feb 17 04:08:23 2010
From: scott.torr.fs at letterboxes.org (Scott Torr)
Date: Wed, 17 Feb 2010 23:08:23 +1100
Subject: [Freeswitch-users] Greetings and a couple of questions
In-Reply-To: <4B7B5A46.3090304@meanie.us>
References: <4B7B5A46.3090304@meanie.us>
Message-ID: <1266408503.10430.1360421639@webmail.messagingengine.com>

On Tue, 16 Feb 2010 20:53 -0600, "Doc" <ledoktre at meanie.us> wrote:
> 1) I am able to see a call come in, and it gets routed to the sample IVR 
> to start.  The first thing off is that when I dial from PSTN -> 
> Skype-In, it does not let me push any buttons.  If I launch a second 
> skype client, and dial the skype user on FS directly, it works fine.  
> Any ideas?

Hi Doc,

When you dial in from the PSTN the 'push button' events are present as
"in band" audio tones.
By default the sample IVR only works on "out of band" DTMF events.

This is why when you call directly from another skype client the 'push
button' events are detected because they are passed as "out of band"
signaling.


Now,
In the dial plan you can tell FS to listen for "in band" audio DTMF
tones using <action application="start_dtmf" />

However,
This currently does not work during a skype call for some reason?
http://jira.freeswitch.org/browse/MODSKYPIAX-66


A work around,
is to sign up for the "Skype SIP Beta" product where the 'push button'
events are sent to FS "out of band".

This conversion is done at the PSTN --> Skype gateway by dedicated DTMF
tone detection hardware.


Skype has either made a business decision, or a technical over sight to
pass DTMF events 'out of band' for only addition fee products.


It has also been reported in New Zealand that even the 'in band' tones
where present one day and actually filtered out the next.
This seems extreme, but either through deliberate action or a technology
change this is what was reported on one blog.

It is unclear to me if this was a technical limitation or a blunt
business decisions, but a audio sample showed the audio tones missing? 


In any case you would not want to rely on 'In band' DTMF' tones when
passed through 'lossy' codecs anyway.

Best to stick with 'out of band' signaling for reliability.


regards,
Scott Torr









From anthony.minessale at gmail.com  Wed Feb 17 05:36:05 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 17 Feb 2010 07:36:05 -0600
Subject: [Freeswitch-users] One more thing..
In-Reply-To: <191c3a031002170535i12c82731r5829417387dd3228@mail.gmail.com>
References: <4B7B5EE9.4020705@meanie.us>
	<191c3a031002170535i12c82731r5829417387dd3228@mail.gmail.com>
Message-ID: <191c3a031002170536g297a816ch2fa94eb2dcfbd16d@mail.gmail.com>

Are you doing this with the latest revision?
You would have to supply more info like a console trace on debug level with
siptrace enabled.

On Feb 17, 2010 1:51 AM, "Doc" <ledoktre at meanie.us> wrote:

Greetings one more time,

I just remembered.  I also ran into one other issue in my testing.

When I route the incoming skype call to an extension (Sipura SPA 1001),
it plays hold music on the callers side (good....), and when I pick up
the extension, calling party drops, internal phone is left with a fast
busy signal :

2010-02-16 21:09:31.687204 [NOTICE] sofia.c:4690 Channel
[sofia/internal/sip:1001 at 10.24.72.12:5060] has been answered
2010-02-16 21:09:31.987714 [WARNING] mod_sofia.c:918 We were told to use
ptime 3 but what they meant to say was 20
This issue has so far been identified to happen on the following broken
platforms/devices:
Linksys/Sipura aka Cisco
ShoreTel
Sonus/L3
We will try to fix it but some of the devices on this list are so broken
who knows what will happen..
2010-02-16 21:09:32.027243 [ERR] switch_core_io.c:121
sofia/internal/sip:1001 at 10.24.72.12:5060 has no read codec.

First thing I notice is this unusual error about the ptime?  and the one
that hangs me up I think is the "sofia... has no read codec".  I have
been poking around my configuration - any suggestions?

Thanks,

Doc


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100217/a192135e/attachment-0002.html 

From anthony.minessale at gmail.com  Wed Feb 17 05:38:50 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 17 Feb 2010 07:38:50 -0600
Subject: [Freeswitch-users] Dingaling - external rtp supported?
In-Reply-To: <29b888f81002170318p63622f3fhc8d26ad0e85602d2@mail.gmail.com>
References: <29b888f81002160835y6e290319ub0d04c088caa1b79@mail.gmail.com>
	<DEA39AD1-0EE9-473A-B78C-CDDF3F100578@freeswitch.org>
	<29b888f81002170318p63622f3fhc8d26ad0e85602d2@mail.gmail.com>
Message-ID: <191c3a031002170538r3b029515i8a000812a70ad558@mail.gmail.com>

Obtain a stack trace from the crash.

On Feb 17, 2010 5:26 AM, "Moiz Chinoy" <moizchinoy at gmail.com> wrote:

Hi,

FS rev: 16673
Platform: Windows

More details:

FS is behind NAT and machine is running a VPN connection.

FS and GTalk client on the same machine:
--------------------------------------------------------------------------------------------------
jingle profile client.xml has following line:
<param name="rtp-ip" value="$${bind_server_ip}"/>

External SIP call is successfully bridged to GTalk client.


FS and GTalk client on the different machine:
--------------------------------------------------------------------------------------------------
jingle profile client.xml has following lines:
<param name="rtp-ip" value="$${bind_server_ip}"/>

<param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
As soon as external SIP call land and I try to bridge the call to
GTalk client, FS crashes.


NAT Details:
---------------------------
I think my NAT does not support UpNP or PMP. The reason I say it
because when FS starts following message is displayed:

2010-02-17 09:15:42.809674 [INFO] switch_nat.c:409 Scanning for NAT
2010-02-17 09:15:42.809674 [ERR] switch_nat.c:197 Error checking for
PMP [init failed]
2010-02-17 09:15:42.809674 [DEBUG] switch_nat.c:414 Checking for UPnP
2010-02-17 09:15:56.825388 [DEBUG] switch_nat.c:114 No
InternetGatewayDevice, using first entry as default
(http://192.168.16.17:50144/).
2010-02-17 09:15:56.887889 [INFO] switch_nat.c:429 No PMP or UPnP NAT
devices detected!




On Tue, Feb 16, 2010 at 8:41 PM, Brian West <brian at freeswitch.org> wrote:
> can you please update...
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100217/f54b8d97/attachment-0002.html 

From yehavi.bourvine at gmail.com  Wed Feb 17 05:49:29 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Wed, 17 Feb 2010 15:49:29 +0200
Subject: [Freeswitch-users] Announcement: FreeSWITCH Add Broadsoft SCA
	Support
In-Reply-To: <9D328485-D4C3-4B14-808A-CB63BBF06DC1@freeswitch.org>
References: <87f2f3b91001111742k691966cfs82515db77c13d4d4@mail.gmail.com>
	<d361ca7b1002082110h4e3c60d0p88412f753444ec3d@mail.gmail.com>
	<016d01caa98e$f6df25f0$e49d71d0$@com>
	<d361ca7b1002090817h4aa3c13dka4d4db8b4adc7606@mail.gmail.com>
	<1C0A1BEE-3480-4A33-916F-52067AF5B9C9@freeswitch.org>
	<d361ca7b1002090838j6c42dcd2of8affb5f281900fd@mail.gmail.com>
	<767B677E-0D19-43A9-89DE-D96EC4A715EC@freeswitch.org>
	<8D062B6E-4691-4BA6-B038-0EC6E8642DDA@jerris.com>
	<9D328485-D4C3-4B14-808A-CB63BBF06DC1@freeswitch.org>
Message-ID: <d361ca7b1002170549g608d6e16vb2eb3f087fff9752@mail.gmail.com>

I am trying to migrate from 1.0.5pre10 to the latest (FreeSWITCH Version
1.0.5-20100216-0400 (16659M)), and shared apearance stopped working...

We have Polycom phones and I use the same configuration for both versions.
With the old version I see that after the phone registers with FreeSwitch,
the server subscribes to the phone for the extension; with the latest
version this does not happen. Furthermore, the table
sip_shared_appearance_dialogs is empty.

I don't find anything I can change on the phone config (the line is already
set to shared).

Here is the relevant config from the sip profile:

    <param name="manage-presence" value="passive"/> (set to TRUE only on one
profile).
    <param name="manage-shared-appearance" value="true"/>
    <param name="dbname" value="share_presence"/>
    <param name="presence-hosts" value="pbx-gr-backup.ip-phones.huji.ac.il
"/>
Any idea where to look?
                           Thanks, __Yehavi:
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100217/f7b628b6/attachment-0002.html 

From anthony.minessale at gmail.com  Wed Feb 17 05:57:13 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 17 Feb 2010 07:57:13 -0600
Subject: [Freeswitch-users] Announcement: FreeSWITCH Add Broadsoft SCA
	Support
In-Reply-To: <191c3a031002170556p5dfd4c55le3c279f194808d92@mail.gmail.com>
References: <87f2f3b91001111742k691966cfs82515db77c13d4d4@mail.gmail.com>
	<016d01caa98e$f6df25f0$e49d71d0$@com>
	<d361ca7b1002090817h4aa3c13dka4d4db8b4adc7606@mail.gmail.com>
	<1C0A1BEE-3480-4A33-916F-52067AF5B9C9@freeswitch.org>
	<d361ca7b1002090838j6c42dcd2of8affb5f281900fd@mail.gmail.com>
	<767B677E-0D19-43A9-89DE-D96EC4A715EC@freeswitch.org>
	<8D062B6E-4691-4BA6-B038-0EC6E8642DDA@jerris.com>
	<9D328485-D4C3-4B14-808A-CB63BBF06DC1@freeswitch.org>
	<d361ca7b1002170549g608d6e16vb2eb3f087fff9752@mail.gmail.com>
	<191c3a031002170556p5dfd4c55le3c279f194808d92@mail.gmail.com>
Message-ID: <191c3a031002170557g5d8c9df9xf40d1b74b9a6f19d@mail.gmail.com>

You have to set the param to sylantro to get that mode.
Or configure your phone to use broadsoft.

On Feb 17, 2010 7:54 AM, "Yehavi Bourvine" <yehavi.bourvine at gmail.com>
wrote:

I am trying to migrate from 1.0.5pre10 to the latest (FreeSWITCH Version
1.0.5-20100216-0400 (16659M)), and shared apearance stopped working...

We have Polycom phones and I use the same configuration for both versions.
With the old version I see that after the phone registers with FreeSwitch,
the server subscribes to the phone for the extension; with the latest
version this does not happen. Furthermore, the table
sip_shared_appearance_dialogs is empty.

I don't find anything I can change on the phone config (the line is already
set to shared).

Here is the relevant config from the sip profile:

    <param name="manage-presence" value="passive"/> (set to TRUE only on one
profile).
    <param name="manage-shared-appearance" value="true"/>


    <param name="dbname" value="share_presence"/>
    <param name="presence-hosts" value="pbx-gr-backup.ip-phones.huji.ac.il
"/>
Any idea where to look?
                           Thanks, __Yehavi:


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100217/35d7982e/attachment-0002.html 

From brian at freeswitch.org  Wed Feb 17 06:26:32 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 17 Feb 2010 08:26:32 -0600
Subject: [Freeswitch-users] Dingaling - external rtp supported?
In-Reply-To: <29b888f81002170318p63622f3fhc8d26ad0e85602d2@mail.gmail.com>
References: <29b888f81002160835y6e290319ub0d04c088caa1b79@mail.gmail.com>
	<DEA39AD1-0EE9-473A-B78C-CDDF3F100578@freeswitch.org>
	<29b888f81002170318p63622f3fhc8d26ad0e85602d2@mail.gmail.com>
Message-ID: <2F1D6EF8-CA94-44D6-B947-6E4B75D14A62@freeswitch.org>

http://wiki.freeswitch.org/wiki/Reporting_Bugs

/b

On Feb 17, 2010, at 5:18 AM, Moiz Chinoy wrote:

> Hi,
> 
> FS rev: 16673
> Platform: Windows
> 
> More details:
> 
> FS is behind NAT and machine is running a VPN connection.
> 
> FS and GTalk client on the same machine:
> --------------------------------------------------------------------------------------------------
> jingle profile client.xml has following line:
> <param name="rtp-ip" value="$${bind_server_ip}"/>
> 
> External SIP call is successfully bridged to GTalk client.
> 
> 
> FS and GTalk client on the different machine:
> --------------------------------------------------------------------------------------------------
> jingle profile client.xml has following lines:
> <param name="rtp-ip" value="$${bind_server_ip}"/>
> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
> 
> As soon as external SIP call land and I try to bridge the call to
> GTalk client, FS crashes.
> 
> 
> NAT Details:
> ---------------------------
> I think my NAT does not support UpNP or PMP. The reason I say it
> because when FS starts following message is displayed:
> 
> 2010-02-17 09:15:42.809674 [INFO] switch_nat.c:409 Scanning for NAT
> 2010-02-17 09:15:42.809674 [ERR] switch_nat.c:197 Error checking for
> PMP [init failed]
> 2010-02-17 09:15:42.809674 [DEBUG] switch_nat.c:414 Checking for UPnP
> 2010-02-17 09:15:56.825388 [DEBUG] switch_nat.c:114 No
> InternetGatewayDevice, using first entry as default
> (http://192.168.16.17:50144/).
> 2010-02-17 09:15:56.887889 [INFO] switch_nat.c:429 No PMP or UPnP NAT
> devices detected!
> 
> 
> 
> On Tue, Feb 16, 2010 at 8:41 PM, Brian West <brian at freeswitch.org> wrote:
>> can you please update, try again and post a jira?
>> 
>> /b
>> 
>> On Feb 16, 2010, at 10:35 AM, Moiz Chinoy wrote:
>> 
>>> Hi All,
>>> 
>>> In mod_dingaling <param name="ext-rtp-ip"
>>> value="$${external_rtp_ip}"/> supported? Whenever I uncomment this
>>> line in client.xml (jingle profile) FS crashes as soon a call lands
>>> (sip call) and dialplan bridges the call to a gtalk user.
>>> 
>>> I am running FS on windows and build is 16642.
>>> 
>>> --
>>> Regards,
>>> Moiz Chinoy.
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
> 
> 
> 
> -- 
> Regards,
> Moiz Chinoy.
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From chris.chen2004 at gmail.com  Wed Feb 17 06:40:36 2010
From: chris.chen2004 at gmail.com (Chris Chen)
Date: Wed, 17 Feb 2010 09:40:36 -0500
Subject: [Freeswitch-users] Announcement: FreeSWITCH Add Broadsoft SCA
	Support
In-Reply-To: <191c3a031002170557g5d8c9df9xf40d1b74b9a6f19d@mail.gmail.com>
References: <87f2f3b91001111742k691966cfs82515db77c13d4d4@mail.gmail.com>
	<d361ca7b1002090817h4aa3c13dka4d4db8b4adc7606@mail.gmail.com>
	<1C0A1BEE-3480-4A33-916F-52067AF5B9C9@freeswitch.org>
	<d361ca7b1002090838j6c42dcd2of8affb5f281900fd@mail.gmail.com>
	<767B677E-0D19-43A9-89DE-D96EC4A715EC@freeswitch.org>
	<8D062B6E-4691-4BA6-B038-0EC6E8642DDA@jerris.com>
	<9D328485-D4C3-4B14-808A-CB63BBF06DC1@freeswitch.org>
	<d361ca7b1002170549g608d6e16vb2eb3f087fff9752@mail.gmail.com>
	<191c3a031002170556p5dfd4c55le3c279f194808d92@mail.gmail.com>
	<191c3a031002170557g5d8c9df9xf40d1b74b9a6f19d@mail.gmail.com>
Message-ID: <507898381002170640u5586698cg310f9cd228da5936@mail.gmail.com>

Hi Tony, do you mean in the internal.xml under
/usr/local/freeswitch/local/sip_profiles
we should set <param name="manage-shared-
appearance" value="sylantro"/>

to get the SCA mode working properly?
Please confirm
Thanks,

Chris


On Wed, Feb 17, 2010 at 8:57 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> You have to set the param to sylantro to get that mode.
> Or configure your phone to use broadsoft.
>
> On Feb 17, 2010 7:54 AM, "Yehavi Bourvine" <yehavi.bourvine at gmail.com>
> wrote:
>
> I am trying to migrate from 1.0.5pre10 to the latest (FreeSWITCH Version
> 1.0.5-20100216-0400 (16659M)), and shared apearance stopped working...
>
> We have Polycom phones and I use the same configuration for both versions.
> With the old version I see that after the phone registers with FreeSwitch,
> the server subscribes to the phone for the extension; with the latest
> version this does not happen. Furthermore, the table
> sip_shared_appearance_dialogs is empty.
>
> I don't find anything I can change on the phone config (the line is already
> set to shared).
>
> Here is the relevant config from the sip profile:
>
>     <param name="manage-presence" value="passive"/> (set to TRUE only on
> one profile).
>     <param name="manage-shared-appearance" value="true"/>
>
>
>     <param name="dbname" value="share_presence"/>
>     <param name="presence-hosts" value="pbx-gr-backup.ip-phones.huji.ac.il
> "/>
> Any idea where to look?
>                            Thanks, __Yehavi:
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100217/a6bcf3b9/attachment-0002.html 

From anthony.minessale at gmail.com  Wed Feb 17 06:51:57 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 17 Feb 2010 08:51:57 -0600
Subject: [Freeswitch-users] Announcement: FreeSWITCH Add Broadsoft SCA
	Support
In-Reply-To: <191c3a031002170649i2e7c0cb9t7b806367b781b06a@mail.gmail.com>
References: <87f2f3b91001111742k691966cfs82515db77c13d4d4@mail.gmail.com>
	<d361ca7b1002090838j6c42dcd2of8affb5f281900fd@mail.gmail.com>
	<767B677E-0D19-43A9-89DE-D96EC4A715EC@freeswitch.org>
	<8D062B6E-4691-4BA6-B038-0EC6E8642DDA@jerris.com>
	<9D328485-D4C3-4B14-808A-CB63BBF06DC1@freeswitch.org>
	<d361ca7b1002170549g608d6e16vb2eb3f087fff9752@mail.gmail.com>
	<191c3a031002170556p5dfd4c55le3c279f194808d92@mail.gmail.com>
	<191c3a031002170557g5d8c9df9xf40d1b74b9a6f19d@mail.gmail.com>
	<507898381002170640u5586698cg310f9cd228da5936@mail.gmail.com>
	<191c3a031002170649i2e7c0cb9t7b806367b781b06a@mail.gmail.com>
Message-ID: <191c3a031002170651k3f0c421cj87f3e2bd796b7711@mail.gmail.com>

If by properly you mean the previously supported way then yes.

We added support for the broadsoft method now as the default because its
supported on more phones.

You may want to try setting your phones to that mode to compare but setting
the profile param to sylantro should restore the previous default behaviour.

On Feb 17, 2010 8:46 AM, "Chris Chen" <chris.chen2004 at gmail.com> wrote:

Hi Tony, do you mean in the internal.xml under
/usr/local/freeswitch/local/sip_profiles
we should set <param name="manage-shared-
appearance" value="sylantro"/>

to get the SCA mode working properly?
Please confirm
Thanks,

Chris




On Wed, Feb 17, 2010 at 8:57 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:
>
> You h...

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100217/ff44348f/attachment-0002.html 

From yehavi.bourvine at gmail.com  Wed Feb 17 07:00:17 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Wed, 17 Feb 2010 17:00:17 +0200
Subject: [Freeswitch-users] Announcement: FreeSWITCH Add Broadsoft SCA
	Support
In-Reply-To: <191c3a031002170557g5d8c9df9xf40d1b74b9a6f19d@mail.gmail.com>
References: <87f2f3b91001111742k691966cfs82515db77c13d4d4@mail.gmail.com>
	<d361ca7b1002090817h4aa3c13dka4d4db8b4adc7606@mail.gmail.com>
	<1C0A1BEE-3480-4A33-916F-52067AF5B9C9@freeswitch.org>
	<d361ca7b1002090838j6c42dcd2of8affb5f281900fd@mail.gmail.com>
	<767B677E-0D19-43A9-89DE-D96EC4A715EC@freeswitch.org>
	<8D062B6E-4691-4BA6-B038-0EC6E8642DDA@jerris.com>
	<9D328485-D4C3-4B14-808A-CB63BBF06DC1@freeswitch.org>
	<d361ca7b1002170549g608d6e16vb2eb3f087fff9752@mail.gmail.com>
	<191c3a031002170556p5dfd4c55le3c279f194808d92@mail.gmail.com>
	<191c3a031002170557g5d8c9df9xf40d1b74b9a6f19d@mail.gmail.com>
Message-ID: <d361ca7b1002170700x15630ab2tabc6f3d91ebc3303@mail.gmail.com>

Hello Anthony,

  Since Polycom has no place to define the server type I've set
manage-shared-appearance="sylantro" and have some progress.

Now I see that both the server and the phone subscribe to each other(the
server subscribes twice), but the last NOTIFY from the phone to the server
gets rejected with "Call does not exist".

On the console I get the following error message: [ERR]
sofia_presence.c:2135 Gateway information missing.
                  Thanks, __Yehavi:

 2010/2/17 Anthony Minessale <anthony.minessale at gmail.com>

>  You have to set the param to sylantro to get that mode.
> Or configure your phone to use broadsoft.
>
>   On Feb 17, 2010 7:54 AM, "Yehavi Bourvine" <yehavi.bourvine at gmail.com>
> wrote:
>
>  I am trying to migrate from 1.0.5pre10 to the latest (FreeSWITCH Version
> 1.0.5-20100216-0400 (16659M)), and shared apearance stopped working...
>
> We have Polycom phones and I use the same configuration for both versions.
> With the old version I see that after the phone registers with FreeSwitch,
> the server subscribes to the phone for the extension; with the latest
> version this does not happen. Furthermore, the table
> sip_shared_appearance_dialogs is empty.
>
> I don't find anything I can change on the phone config (the line is already
> set to shared).
>
> Here is the relevant config from the sip profile:
>
>     <param name="manage-presence" value="passive"/> (set to TRUE only on
> one profile).
>     <param name="manage-shared-appearance" value="true"/>
>
>
>     <param name="dbname" value="share_presence"/>
>     <param name="presence-hosts" value="pbx-gr-backup.ip-phones.huji.ac.il
> "/>
> Any idea where to look?
>                            Thanks, __Yehavi:
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100217/317d66f4/attachment-0002.html 

From moizchinoy at gmail.com  Wed Feb 17 07:41:51 2010
From: moizchinoy at gmail.com (Moiz Chinoy)
Date: Wed, 17 Feb 2010 19:41:51 +0400
Subject: [Freeswitch-users] Dingaling - external rtp supported?
In-Reply-To: <191c3a031002170538r3b029515i8a000812a70ad558@mail.gmail.com>
References: <29b888f81002160835y6e290319ub0d04c088caa1b79@mail.gmail.com>
	<DEA39AD1-0EE9-473A-B78C-CDDF3F100578@freeswitch.org>
	<29b888f81002170318p63622f3fhc8d26ad0e85602d2@mail.gmail.com>
	<191c3a031002170538r3b029515i8a000812a70ad558@mail.gmail.com>
Message-ID: <29b888f81002170741y3d13cb57ycb4436a372c4eded@mail.gmail.com>

Guys I am unable to produce the crash but now both parties cannot hear
each other!

Vars.xml has following lines:
  <X-PRE-PROCESS cmd="set" data="external_rtp_ip=stun:stun.freeswitch.org"/>
  <X-PRE-PROCESS cmd="set" data="external_sip_ip=stun:stun.freeswitch.org"/>

Jingle Client.xml has following lines:
    <param name="rtp-ip" value="$${bind_server_ip}"/>
    <!-- <param name="ext-rtp-ip" value="auto-nat"/> -->
    <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
    <param name="disable-rtp-auto-adjust" value="true"/>
    <param name="local-network-acl" value="localnet.auto"/>

On Wed, Feb 17, 2010 at 5:38 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> Obtain a stack trace from the crash.
>
> On Feb 17, 2010 5:26 AM, "Moiz Chinoy" <moizchinoy at gmail.com> wrote:
>
> Hi,
>
> FS rev: 16673
> Platform: Windows
>
> More details:
>
> FS is behind NAT and machine is running a VPN connection.
>
> FS and GTalk client on the same machine:
> --------------------------------------------------------------------------------------------------
> jingle profile client.xml has following line:
> <param name="rtp-ip" value="$${bind_server_ip}"/>
>
> External SIP call is successfully bridged to GTalk client.
>
>
> FS and GTalk client on the different machine:
> --------------------------------------------------------------------------------------------------
> jingle profile client.xml has following lines:
> <param name="rtp-ip" value="$${bind_server_ip}"/>
>
> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
>
> As soon as external SIP call land and I try to bridge the call to
> GTalk client, FS crashes.
>
>
> NAT Details:
> ---------------------------
> I think my NAT does not support UpNP or PMP. The reason I say it
> because when FS starts following message is displayed:
>
> 2010-02-17 09:15:42.809674 [INFO] switch_nat.c:409 Scanning for NAT
> 2010-02-17 09:15:42.809674 [ERR] switch_nat.c:197 Error checking for
> PMP [init failed]
> 2010-02-17 09:15:42.809674 [DEBUG] switch_nat.c:414 Checking for UPnP
> 2010-02-17 09:15:56.825388 [DEBUG] switch_nat.c:114 No
> InternetGatewayDevice, using first entry as default
> (http://192.168.16.17:50144/).
> 2010-02-17 09:15:56.887889 [INFO] switch_nat.c:429 No PMP or UPnP NAT
> devices detected!
>
>
>
> On Tue, Feb 16, 2010 at 8:41 PM, Brian West <brian at freeswitch.org> wrote:
>> can you please update...
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Regards,
Moiz Chinoy.



From msc at freeswitch.org  Wed Feb 17 08:19:22 2010
From: msc at freeswitch.org (Michael S Collins)
Date: Wed, 17 Feb 2010 08:19:22 -0800
Subject: [Freeswitch-users] Dingaling - external rtp supported?
In-Reply-To: <29b888f81002170741y3d13cb57ycb4436a372c4eded@mail.gmail.com>
References: <29b888f81002160835y6e290319ub0d04c088caa1b79@mail.gmail.com>
	<DEA39AD1-0EE9-473A-B78C-CDDF3F100578@freeswitch.org>
	<29b888f81002170318p63622f3fhc8d26ad0e85602d2@mail.gmail.com>
	<191c3a031002170538r3b029515i8a000812a70ad558@mail.gmail.com>
	<29b888f81002170741y3d13cb57ycb4436a372c4eded@mail.gmail.com>
Message-ID: <055E2932-0D54-44E8-85F7-503D2B6CD592@freeswitch.org>

Get a console log and SIP trace of the call. The wiki page Brian sent  
has all the details on how to collect this information.
-MC

Sent from my iPhone

On Feb 17, 2010, at 7:41 AM, Moiz Chinoy <moizchinoy at gmail.com> wrote:

> Guys I am unable to produce the crash but now both parties cannot hear
> each other!
>
> Vars.xml has following lines:
>  <X-PRE-PROCESS cmd="set"  
> data="external_rtp_ip=stun:stun.freeswitch.org"/>
>  <X-PRE-PROCESS cmd="set"  
> data="external_sip_ip=stun:stun.freeswitch.org"/>
>
> Jingle Client.xml has following lines:
>    <param name="rtp-ip" value="$${bind_server_ip}"/>
>    <!-- <param name="ext-rtp-ip" value="auto-nat"/> -->
>    <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
>    <param name="disable-rtp-auto-adjust" value="true"/>
>    <param name="local-network-acl" value="localnet.auto"/>
>
> On Wed, Feb 17, 2010 at 5:38 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>> Obtain a stack trace from the crash.
>>
>> On Feb 17, 2010 5:26 AM, "Moiz Chinoy" <moizchinoy at gmail.com> wrote:
>>
>> Hi,
>>
>> FS rev: 16673
>> Platform: Windows
>>
>> More details:
>>
>> FS is behind NAT and machine is running a VPN connection.
>>
>> FS and GTalk client on the same machine:
>> --- 
>> --- 
>> --- 
>> --- 
>> --- 
>> --- 
>> --- 
>> --- 
>> --- 
>> --- 
>> --------------------------------------------------------------------
>> jingle profile client.xml has following line:
>> <param name="rtp-ip" value="$${bind_server_ip}"/>
>>
>> External SIP call is successfully bridged to GTalk client.
>>
>>
>> FS and GTalk client on the different machine:
>> --- 
>> --- 
>> --- 
>> --- 
>> --- 
>> --- 
>> --- 
>> --- 
>> --- 
>> --- 
>> --------------------------------------------------------------------
>> jingle profile client.xml has following lines:
>> <param name="rtp-ip" value="$${bind_server_ip}"/>
>>
>> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
>>
>> As soon as external SIP call land and I try to bridge the call to
>> GTalk client, FS crashes.
>>
>>
>> NAT Details:
>> ---------------------------
>> I think my NAT does not support UpNP or PMP. The reason I say it
>> because when FS starts following message is displayed:
>>
>> 2010-02-17 09:15:42.809674 [INFO] switch_nat.c:409 Scanning for NAT
>> 2010-02-17 09:15:42.809674 [ERR] switch_nat.c:197 Error checking for
>> PMP [init failed]
>> 2010-02-17 09:15:42.809674 [DEBUG] switch_nat.c:414 Checking for UPnP
>> 2010-02-17 09:15:56.825388 [DEBUG] switch_nat.c:114 No
>> InternetGatewayDevice, using first entry as default
>> (http://192.168.16.17:50144/).
>> 2010-02-17 09:15:56.887889 [INFO] switch_nat.c:429 No PMP or UPnP NAT
>> devices detected!
>>
>>
>>
>> On Tue, Feb 16, 2010 at 8:41 PM, Brian West <brian at freeswitch.org>  
>> wrote:
>>> can you please update...
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>> users
>> http://www.freeswitch.org
>>
>>
>
>
>
> -- 
> Regards,
> Moiz Chinoy.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org



From brian at freeswitch.org  Wed Feb 17 08:24:47 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 17 Feb 2010 10:24:47 -0600
Subject: [Freeswitch-users] Dingaling - external rtp supported?
In-Reply-To: <055E2932-0D54-44E8-85F7-503D2B6CD592@freeswitch.org>
References: <29b888f81002160835y6e290319ub0d04c088caa1b79@mail.gmail.com>
	<DEA39AD1-0EE9-473A-B78C-CDDF3F100578@freeswitch.org>
	<29b888f81002170318p63622f3fhc8d26ad0e85602d2@mail.gmail.com>
	<191c3a031002170538r3b029515i8a000812a70ad558@mail.gmail.com>
	<29b888f81002170741y3d13cb57ycb4436a372c4eded@mail.gmail.com>
	<055E2932-0D54-44E8-85F7-503D2B6CD592@freeswitch.org>
Message-ID: <D4D98039-293E-4DF4-9DB2-E695EA84F3C9@freeswitch.org>

I have tested this with empty values... and it works fine.

/b

On Feb 17, 2010, at 10:19 AM, Michael S Collins wrote:

> Get a console log and SIP trace of the call. The wiki page Brian sent  
> has all the details on how to collect this information.
> -MC
> 
> Sent from my iPhone




From e.brolman at telecats.nl  Wed Feb 17 05:56:32 2010
From: e.brolman at telecats.nl (=?iso-8859-1?Q?Eelco_Br=F6lman?=)
Date: Wed, 17 Feb 2010 14:56:32 +0100
Subject: [Freeswitch-users] Calls being parked on DTMF (2)
Message-ID: <0A1FDB5DAA23564F8758BA05D26DCD747F9A09@exchange.telecats.nl>

Hi all,

I have the same problems as described in the thread "Calls being parked on DTMF".

We have a lab-test setup, which consist of the following:

1) a box with 4 E1 trunks with a call generator application (let's call it CG) 
2) connected with a FreeSwitch box (Sangoma A104D card) (let's call that FS)

The CG dials a number on the FS box (number 921000), which is bridged out back to the CG server (number 1000). The initiating application sends some DTMF, which is replied (echo-ed) by the receiving application (both on the CG server).

In roughly about 80% of the calls, FS just disconnects the call:

2010-02-17 14:30:14.677125 [DEBUG] switch_ivr_originate.c:3105 Originate Resulted in Success: [OpenZAP/1:1/1000]
2010-02-17 14:30:14.677125 [DEBUG] switch_core_session.c:638 Send signal OpenZAP/1:1/1000 [BREAK]
2010-02-17 14:30:14.677125 [DEBUG] switch_core_session.c:638 Send signal OpenZAP/1:60/921000 [BREAK]
2010-02-17 14:30:14.677125 [DEBUG] switch_ivr_bridge.c:1175 (OpenZAP/1:1/1000) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA
2010-02-17 14:30:14.677125 [DEBUG] switch_core_session.c:1018 Send signal OpenZAP/1:1/1000 [BREAK]
2010-02-17 14:30:14.677125 [DEBUG] switch_core_state_machine.c:314 (OpenZAP/1:1/1000) Running State Change CS_EXCHANGE_MEDIA
2010-02-17 14:30:14.677125 [DEBUG] switch_core_state_machine.c:351 (OpenZAP/1:1/1000) State EXCHANGE_MEDIA
2010-02-17 14:30:14.677125 [DEBUG] mod_openzap.c:558 CHANNEL EXCHANGE_MEDIA
2010-02-17 14:30:14.687135 [DEBUG] switch_core_session.c:699 Send signal OpenZAP/1:1/1000 [BREAK]
2010-02-17 14:30:14.687135 [DEBUG] switch_core_session.c:699 Send signal OpenZAP/1:60/921000 [BREAK]
2010-02-17 14:30:15.986882 [DEBUG] mod_openzap.c:684 queue DTMF [7]
2010-02-17 14:30:16.026733 [DEBUG] zap_io.c:1870 1:1 GENERATE DTMF [7]
2010-02-17 14:30:16.106418 [DEBUG] switch_ivr_bridge.c:469 OpenZAP/1:1/1000 ending bridge by request from write function
2010-02-17 14:30:16.106418 [DEBUG] switch_core_session.c:638 Send signal OpenZAP/1:60/921000 [BREAK]
2010-02-17 14:30:16.126335 [DEBUG] switch_ivr_bridge.c:475 OpenZAP/1:1/1000 ending bridge by request from read function
2010-02-17 14:30:16.126335 [DEBUG] switch_core_session.c:638 Send signal OpenZAP/1:1/1000 [BREAK]
2010-02-17 14:30:16.126335 [DEBUG] switch_ivr_bridge.c:562 BRIDGE THREAD DONE [OpenZAP/1:60/921000]
2010-02-17 14:30:16.126335 [DEBUG] switch_ivr_bridge.c:582 Send signal OpenZAP/1:1/1000 [BREAK]
2010-02-17 14:30:16.126335 [DEBUG] switch_ivr_bridge.c:562 BRIDGE THREAD DONE [OpenZAP/1:1/1000]
2010-02-17 14:30:16.126335 [DEBUG] switch_ivr_bridge.c:582 Send signal OpenZAP/1:60/921000 [BREAK]
2010-02-17 14:30:16.126335 [NOTICE] switch_ivr_bridge.c:634 Hangup OpenZAP/1:1/1000 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
2010-02-17 14:30:16.126335 [DEBUG] switch_channel.c:2063 Send signal OpenZAP/1:1/1000 [KILL]


Where the line "ending bridge by request from write function" is curious of course.

I'm running:
- Freeswitch trunk (r16674) (updated, make clean, reconfigured and make install this morning)
- Wanpipe 3.5.10.smg-2 (see http://wiki.sangoma.com/wanpipe-SmgPriInstallation)
- Using the ozmod_sangoma_boost SCTP


If anyone needs more information, tests or config files, I'll be happy to provide them.



Kind regards, 

Eelco Br?lman




From mike at jerris.com  Wed Feb 17 08:52:20 2010
From: mike at jerris.com (Michael Jerris)
Date: Wed, 17 Feb 2010 11:52:20 -0500
Subject: [Freeswitch-users] FreeSWITCH.Managed.dll deletes on make
	distclean
In-Reply-To: <201002152353.43666.errotan@gmail.com>
References: <201002152353.43666.errotan@gmail.com>
Message-ID: <097535D1-2601-4BBE-88D8-A113685A9435@jerris.com>

we do not currently have a working distclean target.  using it will likely make your tree in an unusable state.

Mike

On Feb 15, 2010, at 5:53 PM, Pusk?s Zsolt wrote:

> I usually do svn-clean than svn up when i compile a new version of fs.
> I noticed that if i do make distclean the file @ 
> src/mod/languages/mod_managed/managed/FreeSWITCH.Managed.dll got deleted.
> When i do svn up it gets 'restored':
> 
> Restored 'src/mod/languages/mod_managed/managed/FreeSWITCH.Managed.dll'




From brian at freeswitch.org  Wed Feb 17 09:04:38 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 17 Feb 2010 11:04:38 -0600
Subject: [Freeswitch-users] One more thing..
In-Reply-To: <191c3a031002170536g297a816ch2fa94eb2dcfbd16d@mail.gmail.com>
References: <4B7B5EE9.4020705@meanie.us>
	<191c3a031002170535i12c82731r5829417387dd3228@mail.gmail.com>
	<191c3a031002170536g297a816ch2fa94eb2dcfbd16d@mail.gmail.com>
Message-ID: <2CEBE2C8-3982-49B6-A06F-9406C1D1F123@freeswitch.org>

Its the same bug the linksys SPA has... where the RTP time is set to 0.030 and an inbound call to the device doesn't 200ok with the right ptime.

/b

On Feb 17, 2010, at 7:36 AM, Anthony Minessale wrote:

> Are you doing this with the latest revision?
> You would have to supply more info like a console trace on debug level with siptrace enabled.




From ledoktre at meanie.us  Wed Feb 17 09:19:06 2010
From: ledoktre at meanie.us (Doc)
Date: Wed, 17 Feb 2010 11:19:06 -0600
Subject: [Freeswitch-users] One more thing..
In-Reply-To: <191c3a031002170536g297a816ch2fa94eb2dcfbd16d@mail.gmail.com>
References: <4B7B5EE9.4020705@meanie.us>	<191c3a031002170535i12c82731r5829417387dd3228@mail.gmail.com>
	<191c3a031002170536g297a816ch2fa94eb2dcfbd16d@mail.gmail.com>
Message-ID: <4B7C250A.3050801@meanie.us>

I need to be a little quicker on my replies :-)  FS tells me "FreeSWITCH 
Version 1.0.trunk (16619M)".  I installed it this past Saturday from SVN 
trunk as I recall.  I can supply anything you need, but I am pretty new 
to FS and definately wet behind the ears.  Thanks-

Anthony Minessale wrote:
>
> Are you doing this with the latest revision?
> You would have to supply more info like a console trace on debug level 
> with siptrace enabled.
>
>> On Feb 17, 2010 1:51 AM, "Doc" <ledoktre at meanie.us 
>> <mailto:ledoktre at meanie.us>> wrote:
>>
>> Greetings one more time,
>>
>> I just remembered.  I also ran into one other issue in my testing.
>>
>> When I route the incoming skype call to an extension (Sipura SPA 1001),
>> it plays hold music on the callers side (good....), and when I pick up
>> the extension, calling party drops, internal phone is left with a fast
>> busy signal :
>>
>> 2010-02-16 21:09:31.687204 [NOTICE] sofia.c:4690 Channel
>> [sofia/internal/sip:1001 at 10.24.72.12:5060 
>> <http://sip:1001 at 10.24.72.12:5060>] has been answered
>> 2010-02-16 21:09:31.987714 [WARNING] mod_sofia.c:918 We were told to use
>> ptime 3 but what they meant to say was 20
>> This issue has so far been identified to happen on the following broken
>> platforms/devices:
>> Linksys/Sipura aka Cisco
>> ShoreTel
>> Sonus/L3
>> We will try to fix it but some of the devices on this list are so broken
>> who knows what will happen..
>> 2010-02-16 21:09:32.027243 [ERR] switch_core_io.c:121
>> sofia/internal/sip:1001 at 10.24.72.12:5060 
>> <http://sip:1001 at 10.24.72.12:5060> has no read codec.
>>
>> First thing I notice is this unusual error about the ptime?  and the one
>> that hangs me up I think is the "sofia... has no read codec".  I have
>> been poking around my configuration - any suggestions?
>>
>> Thanks,
>>
>> Doc
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org 
>> <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> ------------------------------------------------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   



From ledoktre at meanie.us  Wed Feb 17 09:19:59 2010
From: ledoktre at meanie.us (Doc)
Date: Wed, 17 Feb 2010 11:19:59 -0600
Subject: [Freeswitch-users] One more thing..
In-Reply-To: <2CEBE2C8-3982-49B6-A06F-9406C1D1F123@freeswitch.org>
References: <4B7B5EE9.4020705@meanie.us>	<191c3a031002170535i12c82731r5829417387dd3228@mail.gmail.com>	<191c3a031002170536g297a816ch2fa94eb2dcfbd16d@mail.gmail.com>
	<2CEBE2C8-3982-49B6-A06F-9406C1D1F123@freeswitch.org>
Message-ID: <4B7C253F.9090905@meanie.us>

Is there any configuration to change in the Sipura, any changes to FS 
configuration, or does it boil down to using different SIP hardware? -Thanks

Brian West wrote:
> Its the same bug the linksys SPA has... where the RTP time is set to 0.030 and an inbound call to the device doesn't 200ok with the right ptime.
>
> /b
>
> On Feb 17, 2010, at 7:36 AM, Anthony Minessale wrote:
>
>   
>> Are you doing this with the latest revision?
>> You would have to supply more info like a console trace on debug level with siptrace enabled.
>>     
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   



From ledoktre at meanie.us  Wed Feb 17 09:23:40 2010
From: ledoktre at meanie.us (Doc)
Date: Wed, 17 Feb 2010 11:23:40 -0600
Subject: [Freeswitch-users] Greetings and a couple of questions
In-Reply-To: <1266408503.10430.1360421639@webmail.messagingengine.com>
References: <4B7B5A46.3090304@meanie.us>
	<1266408503.10430.1360421639@webmail.messagingengine.com>
Message-ID: <4B7C261C.1040802@meanie.us>

Once you mentioned the in-band versus out-of-band DTMF, it made sense.  
I've been reading on it since your post, and I'm going to try a few 
things.  Wonder if there is any app for voice controlled IVR (so many 
systems seem to support it these days, it would be a nice way to 
circumvent the DTMF issue... :-) ).

Any ideas on the rest of my points?   -Thanks,

Scott Torr wrote:
> On Tue, 16 Feb 2010 20:53 -0600, "Doc" <ledoktre at meanie.us> wrote:
>   
>> 1) I am able to see a call come in, and it gets routed to the sample IVR 
>> to start.  The first thing off is that when I dial from PSTN -> 
>> Skype-In, it does not let me push any buttons.  If I launch a second 
>> skype client, and dial the skype user on FS directly, it works fine.  
>> Any ideas?
>>     
>
> Hi Doc,
>
> When you dial in from the PSTN the 'push button' events are present as
> "in band" audio tones.
> By default the sample IVR only works on "out of band" DTMF events.
>
> This is why when you call directly from another skype client the 'push
> button' events are detected because they are passed as "out of band"
> signaling.
>
>
> Now,
> In the dial plan you can tell FS to listen for "in band" audio DTMF
> tones using <action application="start_dtmf" />
>
> However,
> This currently does not work during a skype call for some reason?
> http://jira.freeswitch.org/browse/MODSKYPIAX-66
>
>
> A work around,
> is to sign up for the "Skype SIP Beta" product where the 'push button'
> events are sent to FS "out of band".
>
> This conversion is done at the PSTN --> Skype gateway by dedicated DTMF
> tone detection hardware.
>
>
> Skype has either made a business decision, or a technical over sight to
> pass DTMF events 'out of band' for only addition fee products.
>
>
> It has also been reported in New Zealand that even the 'in band' tones
> where present one day and actually filtered out the next.
> This seems extreme, but either through deliberate action or a technology
> change this is what was reported on one blog.
>
> It is unclear to me if this was a technical limitation or a blunt
> business decisions, but a audio sample showed the audio tones missing? 
>
>
> In any case you would not want to rely on 'In band' DTMF' tones when
> passed through 'lossy' codecs anyway.
>
> Best to stick with 'out of band' signaling for reliability.
>
>
> regards,
> Scott Torr
>
>
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   



From brian at freeswitch.org  Wed Feb 17 09:26:44 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 17 Feb 2010 11:26:44 -0600
Subject: [Freeswitch-users] One more thing..
In-Reply-To: <4B7C253F.9090905@meanie.us>
References: <4B7B5EE9.4020705@meanie.us>	<191c3a031002170535i12c82731r5829417387dd3228@mail.gmail.com>	<191c3a031002170536g297a816ch2fa94eb2dcfbd16d@mail.gmail.com>
	<2CEBE2C8-3982-49B6-A06F-9406C1D1F123@freeswitch.org>
	<4B7C253F.9090905@meanie.us>
Message-ID: <5A00F97E-41C9-43AD-8CFA-1C8B87E5EECC@freeswitch.org>

change the rtp time to 0.020 from 0.030

/b

On Feb 17, 2010, at 11:19 AM, Doc wrote:

> Is there any configuration to change in the Sipura, any changes to FS 
> configuration, or does it boil down to using different SIP hardware? -Thanks




From ledoktre at meanie.us  Wed Feb 17 09:56:19 2010
From: ledoktre at meanie.us (Doc)
Date: Wed, 17 Feb 2010 11:56:19 -0600
Subject: [Freeswitch-users] One more thing..
In-Reply-To: <5A00F97E-41C9-43AD-8CFA-1C8B87E5EECC@freeswitch.org>
References: <4B7B5EE9.4020705@meanie.us>	<191c3a031002170535i12c82731r5829417387dd3228@mail.gmail.com>	<191c3a031002170536g297a816ch2fa94eb2dcfbd16d@mail.gmail.com>	<2CEBE2C8-3982-49B6-A06F-9406C1D1F123@freeswitch.org>	<4B7C253F.9090905@meanie.us>
	<5A00F97E-41C9-43AD-8CFA-1C8B87E5EECC@freeswitch.org>
Message-ID: <4B7C2DC3.20502@meanie.us>

I did try changing the rtp time in my Sipura (advanced, sip, RTP Packet 
Size) from 0.030 to 0.020, and the error went away.  I was able to then 
test (and succeed) dialing from secondary skype user to skypiax, and 
have the call bridged automatically to one of my Sipura SPA-1001's.  I 
could speak into the phone, and hear it come through skype (on my 
laptop) no problem.  When I would talk on the laptop, I would get garble 
back on the phone, but that could be a sound issue on my laptop (my 
skype sometimes does this with Pulse audio on my laptop, Ubuntu Karmic).

The interesting thing it did do, however, was eventually (within a 
minute?) It threw a couple of errors :

2010-02-17 11:47:34.007909 [ERR] mod_skypiax.c:826 rev 
16619M[(nil)|37     ][ERRORA  826  ][interface1][-1, 5,21] EXIT? sent=-1
2010-02-17 11:47:34.027776 [ERR] mod_skypiax.c:826 rev 
16619M[(nil)|37     ][ERRORA  826  ][interface1][-1, 5,21] EXIT? sent=-1
2010-02-17 11:47:34.047643 [ERR] mod_skypiax.c:826 rev 
16619M[(nil)|37     ][ERRORA  826  ][interface1][-1, 5,21] EXIT? sent=-1

And on a subsequent test, I received no audio (the above error rolling 
on the console), and within a matter of seconds, FS crashed with this :

Segmentation fault (core dumped)

I have another request open on the group actually for the above error, 
so I think it'd make sense to discuss further there (rather than 
duplicating material).

Thanks!  At any rate, the error for the missing codec, etc, seemed to be 
gone once I updated the RTP time in my ATA.  Hope this helps someone!!

Brian West wrote:
> change the rtp time to 0.020 from 0.030
>
> /b



From brian at freeswitch.org  Wed Feb 17 11:23:03 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 17 Feb 2010 13:23:03 -0600
Subject: [Freeswitch-users] Announcement: FreeSWITCH Add Broadsoft SCA
	Support
In-Reply-To: <d361ca7b1002170700x15630ab2tabc6f3d91ebc3303@mail.gmail.com>
References: <87f2f3b91001111742k691966cfs82515db77c13d4d4@mail.gmail.com>
	<d361ca7b1002090817h4aa3c13dka4d4db8b4adc7606@mail.gmail.com>
	<1C0A1BEE-3480-4A33-916F-52067AF5B9C9@freeswitch.org>
	<d361ca7b1002090838j6c42dcd2of8affb5f281900fd@mail.gmail.com>
	<767B677E-0D19-43A9-89DE-D96EC4A715EC@freeswitch.org>
	<8D062B6E-4691-4BA6-B038-0EC6E8642DDA@jerris.com>
	<9D328485-D4C3-4B14-808A-CB63BBF06DC1@freeswitch.org>
	<d361ca7b1002170549g608d6e16vb2eb3f087fff9752@mail.gmail.com>
	<191c3a031002170556p5dfd4c55le3c279f194808d92@mail.gmail.com>
	<191c3a031002170557g5d8c9df9xf40d1b74b9a6f19d@mail.gmail.com>
	<d361ca7b1002170700x15630ab2tabc6f3d91ebc3303@mail.gmail.com>
Message-ID: <ADC8B813-4161-481C-877E-93FC012858B4@freeswitch.org>

Step 1. Enable manage-shared-appearance=true

Step 2. Now in the phone's config Configure the phone as usually, set the line shared and DO NOT set the third party name.

Step 3. Reboot

It should work.

I wish someone that has this working would write some wiki docs these threads about it not working are getting rather old when I know for a fact they work fine.

The gateway info missing is a gateway you have configured getting a notify.  It has nothing to do with SCA.

/b

On Feb 17, 2010, at 9:00 AM, Yehavi Bourvine wrote:

> .




From anthony.minessale at gmail.com  Wed Feb 17 11:23:06 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 17 Feb 2010 13:23:06 -0600
Subject: [Freeswitch-users] One more thing..
In-Reply-To: <4B7C2DC3.20502@meanie.us>
References: <4B7B5EE9.4020705@meanie.us>
	<191c3a031002170535i12c82731r5829417387dd3228@mail.gmail.com>
	<191c3a031002170536g297a816ch2fa94eb2dcfbd16d@mail.gmail.com>
	<2CEBE2C8-3982-49B6-A06F-9406C1D1F123@freeswitch.org>
	<4B7C253F.9090905@meanie.us>
	<5A00F97E-41C9-43AD-8CFA-1C8B87E5EECC@freeswitch.org>
	<4B7C2DC3.20502@meanie.us>
Message-ID: <191c3a031002171123r416f6413wbecddcea498b5616@mail.gmail.com>

keep updating, the maintainer of mod_skypeiax is adding new patches every
few hours.
You can also join irc on irc.freenode.net #freeswitch and #freeswitch-dev to
interact with him live.


One hint around here when we ask if you updated we work in 1 minute
increments.



On Wed, Feb 17, 2010 at 11:56 AM, Doc <ledoktre at meanie.us> wrote:

> I did try changing the rtp time in my Sipura (advanced, sip, RTP Packet
> Size) from 0.030 to 0.020, and the error went away.  I was able to then
> test (and succeed) dialing from secondary skype user to skypiax, and
> have the call bridged automatically to one of my Sipura SPA-1001's.  I
> could speak into the phone, and hear it come through skype (on my
> laptop) no problem.  When I would talk on the laptop, I would get garble
> back on the phone, but that could be a sound issue on my laptop (my
> skype sometimes does this with Pulse audio on my laptop, Ubuntu Karmic).
>
> The interesting thing it did do, however, was eventually (within a
> minute?) It threw a couple of errors :
>
> 2010-02-17 11:47:34.007909 [ERR] mod_skypiax.c:826 rev
> 16619M[(nil)|37     ][ERRORA  826  ][interface1][-1, 5,21] EXIT? sent=-1
> 2010-02-17 11:47:34.027776 [ERR] mod_skypiax.c:826 rev
> 16619M[(nil)|37     ][ERRORA  826  ][interface1][-1, 5,21] EXIT? sent=-1
> 2010-02-17 11:47:34.047643 [ERR] mod_skypiax.c:826 rev
> 16619M[(nil)|37     ][ERRORA  826  ][interface1][-1, 5,21] EXIT? sent=-1
>
> And on a subsequent test, I received no audio (the above error rolling
> on the console), and within a matter of seconds, FS crashed with this :
>
> Segmentation fault (core dumped)
>
> I have another request open on the group actually for the above error,
> so I think it'd make sense to discuss further there (rather than
> duplicating material).
>
> Thanks!  At any rate, the error for the missing codec, etc, seemed to be
> gone once I updated the RTP time in my ATA.  Hope this helps someone!!
>
> Brian West wrote:
> > change the rtp time to 0.020 from 0.030
> >
> > /b
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100217/9657cfda/attachment-0002.html 

From msc at freeswitch.org  Wed Feb 17 11:29:40 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 17 Feb 2010 11:29:40 -0800
Subject: [Freeswitch-users] One more thing..
In-Reply-To: <4B7C2DC3.20502@meanie.us>
References: <4B7B5EE9.4020705@meanie.us>
	<191c3a031002170535i12c82731r5829417387dd3228@mail.gmail.com>
	<191c3a031002170536g297a816ch2fa94eb2dcfbd16d@mail.gmail.com>
	<2CEBE2C8-3982-49B6-A06F-9406C1D1F123@freeswitch.org>
	<4B7C253F.9090905@meanie.us>
	<5A00F97E-41C9-43AD-8CFA-1C8B87E5EECC@freeswitch.org>
	<4B7C2DC3.20502@meanie.us>
Message-ID: <87f2f3b91002171129v59518b2bg9673212cff957a6c@mail.gmail.com>

Are you able to update your FS to the latest trunk? ("make current" in the
FS src dir) Also, I notice that you are on 16619M - the "M" indicates a
modification of some kind. Did you make any changes to the FS source?

-MC

On Wed, Feb 17, 2010 at 9:56 AM, Doc <ledoktre at meanie.us> wrote:

> I did try changing the rtp time in my Sipura (advanced, sip, RTP Packet
> Size) from 0.030 to 0.020, and the error went away.  I was able to then
> test (and succeed) dialing from secondary skype user to skypiax, and
> have the call bridged automatically to one of my Sipura SPA-1001's.  I
> could speak into the phone, and hear it come through skype (on my
> laptop) no problem.  When I would talk on the laptop, I would get garble
> back on the phone, but that could be a sound issue on my laptop (my
> skype sometimes does this with Pulse audio on my laptop, Ubuntu Karmic).
>
> The interesting thing it did do, however, was eventually (within a
> minute?) It threw a couple of errors :
>
> 2010-02-17 11:47:34.007909 [ERR] mod_skypiax.c:826 rev
> 16619M[(nil)|37     ][ERRORA  826  ][interface1][-1, 5,21] EXIT? sent=-1
> 2010-02-17 11:47:34.027776 [ERR] mod_skypiax.c:826 rev
> 16619M[(nil)|37     ][ERRORA  826  ][interface1][-1, 5,21] EXIT? sent=-1
> 2010-02-17 11:47:34.047643 [ERR] mod_skypiax.c:826 rev
> 16619M[(nil)|37     ][ERRORA  826  ][interface1][-1, 5,21] EXIT? sent=-1
>
> And on a subsequent test, I received no audio (the above error rolling
> on the console), and within a matter of seconds, FS crashed with this :
>
> Segmentation fault (core dumped)
>
> I have another request open on the group actually for the above error,
> so I think it'd make sense to discuss further there (rather than
> duplicating material).
>
> Thanks!  At any rate, the error for the missing codec, etc, seemed to be
> gone once I updated the RTP time in my ATA.  Hope this helps someone!!
>
> Brian West wrote:
> > change the rtp time to 0.020 from 0.030
> >
> > /b
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100217/842ca09c/attachment-0002.html 

From ledoktre at meanie.us  Wed Feb 17 11:35:59 2010
From: ledoktre at meanie.us (Doc)
Date: Wed, 17 Feb 2010 13:35:59 -0600
Subject: [Freeswitch-users] One more thing..
In-Reply-To: <191c3a031002171123r416f6413wbecddcea498b5616@mail.gmail.com>
References: <4B7B5EE9.4020705@meanie.us>	<191c3a031002170535i12c82731r5829417387dd3228@mail.gmail.com>	<191c3a031002170536g297a816ch2fa94eb2dcfbd16d@mail.gmail.com>	<2CEBE2C8-3982-49B6-A06F-9406C1D1F123@freeswitch.org>	<4B7C253F.9090905@meanie.us>	<5A00F97E-41C9-43AD-8CFA-1C8B87E5EECC@freeswitch.org>	<4B7C2DC3.20502@meanie.us>
	<191c3a031002171123r416f6413wbecddcea498b5616@mail.gmail.com>
Message-ID: <4B7C451F.1050104@meanie.us>

I will update the software and report back.  It it good to know that you 
think in "1 minute increments" - I figured mine was not too outdated 
;-)   -Thanks,

Anthony Minessale wrote:
> keep updating, the maintainer of mod_skypeiax is adding new patches 
> every few hours.
> You can also join irc on irc.freenode.net <http://irc.freenode.net> 
> #freeswitch and #freeswitch-dev to interact with him live.
>
>
> One hint around here when we ask if you updated we work in 1 minute 
> increments.
>
>
>
> On Wed, Feb 17, 2010 at 11:56 AM, Doc <ledoktre at meanie.us 
> <mailto:ledoktre at meanie.us>> wrote:
>
>     I did try changing the rtp time in my Sipura (advanced, sip, RTP
>     Packet
>     Size) from 0.030 to 0.020, and the error went away.  I was able to
>     then
>     test (and succeed) dialing from secondary skype user to skypiax, and
>     have the call bridged automatically to one of my Sipura SPA-1001's.  I
>     could speak into the phone, and hear it come through skype (on my
>     laptop) no problem.  When I would talk on the laptop, I would get
>     garble
>     back on the phone, but that could be a sound issue on my laptop (my
>     skype sometimes does this with Pulse audio on my laptop, Ubuntu
>     Karmic).
>
>     The interesting thing it did do, however, was eventually (within a
>     minute?) It threw a couple of errors :
>
>     2010-02-17 11:47:34.007909 [ERR] mod_skypiax.c:826 rev
>     16619M[(nil)|37     ][ERRORA  826  ][interface1][-1, 5,21] EXIT?
>     sent=-1
>     2010-02-17 11:47:34.027776 [ERR] mod_skypiax.c:826 rev
>     16619M[(nil)|37     ][ERRORA  826  ][interface1][-1, 5,21] EXIT?
>     sent=-1
>     2010-02-17 11:47:34.047643 [ERR] mod_skypiax.c:826 rev
>     16619M[(nil)|37     ][ERRORA  826  ][interface1][-1, 5,21] EXIT?
>     sent=-1
>
>     And on a subsequent test, I received no audio (the above error rolling
>     on the console), and within a matter of seconds, FS crashed with
>     this :
>
>     Segmentation fault (core dumped)
>
>     I have another request open on the group actually for the above error,
>     so I think it'd make sense to discuss further there (rather than
>     duplicating material).
>
>     Thanks!  At any rate, the error for the missing codec, etc, seemed
>     to be
>     gone once I updated the RTP time in my ATA.  Hope this helps someone!!
>
>     Brian West wrote:
>     > change the rtp time to 0.020 from 0.030
>     >
>     > /b
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com 
> <mailto:MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com 
> <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org 
> <mailto:sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888 
> <http://iax:guest at conference.freeswitch.org/888>
> googletalk:conf+888 at conference.freeswitch.org 
> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
> ------------------------------------------------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   



From anthony.minessale at gmail.com  Wed Feb 17 11:36:49 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 17 Feb 2010 13:36:49 -0600
Subject: [Freeswitch-users] Dingaling - external rtp supported?
In-Reply-To: <29b888f81002170741y3d13cb57ycb4436a372c4eded@mail.gmail.com>
References: <29b888f81002160835y6e290319ub0d04c088caa1b79@mail.gmail.com>
	<DEA39AD1-0EE9-473A-B78C-CDDF3F100578@freeswitch.org>
	<29b888f81002170318p63622f3fhc8d26ad0e85602d2@mail.gmail.com>
	<191c3a031002170538r3b029515i8a000812a70ad558@mail.gmail.com>
	<29b888f81002170741y3d13cb57ycb4436a372c4eded@mail.gmail.com>
Message-ID: <191c3a031002171136q7424e9bai34379abb542daf93@mail.gmail.com>

you cant combine stun and gtalk and boxes in the same lan very easily if you
do need to do that you will need to mess with

<param name="candidate-acl" value="wan.auto"/>
<param name="local-network-acl" value="localnet.auto"/>





On Wed, Feb 17, 2010 at 9:41 AM, Moiz Chinoy <moizchinoy at gmail.com> wrote:

> Guys I am unable to produce the crash but now both parties cannot hear
> each other!
>
> Vars.xml has following lines:
>  <X-PRE-PROCESS cmd="set" data="external_rtp_ip=stun:stun.freeswitch.org
> "/>
>  <X-PRE-PROCESS cmd="set" data="external_sip_ip=stun:stun.freeswitch.org
> "/>
>
> Jingle Client.xml has following lines:
>     <param name="rtp-ip" value="$${bind_server_ip}"/>
>     <!-- <param name="ext-rtp-ip" value="auto-nat"/> -->
>     <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
>     <param name="disable-rtp-auto-adjust" value="true"/>
>    <param name="local-network-acl" value="localnet.auto"/>
>
> On Wed, Feb 17, 2010 at 5:38 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
> > Obtain a stack trace from the crash.
> >
> > On Feb 17, 2010 5:26 AM, "Moiz Chinoy" <moizchinoy at gmail.com> wrote:
> >
> > Hi,
> >
> > FS rev: 16673
> > Platform: Windows
> >
> > More details:
> >
> > FS is behind NAT and machine is running a VPN connection.
> >
> > FS and GTalk client on the same machine:
> >
> --------------------------------------------------------------------------------------------------
> > jingle profile client.xml has following line:
> > <param name="rtp-ip" value="$${bind_server_ip}"/>
> >
> > External SIP call is successfully bridged to GTalk client.
> >
> >
> > FS and GTalk client on the different machine:
> >
> --------------------------------------------------------------------------------------------------
> > jingle profile client.xml has following lines:
> > <param name="rtp-ip" value="$${bind_server_ip}"/>
> >
> > <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
> >
> > As soon as external SIP call land and I try to bridge the call to
> > GTalk client, FS crashes.
> >
> >
> > NAT Details:
> > ---------------------------
> > I think my NAT does not support UpNP or PMP. The reason I say it
> > because when FS starts following message is displayed:
> >
> > 2010-02-17 09:15:42.809674 [INFO] switch_nat.c:409 Scanning for NAT
> > 2010-02-17 09:15:42.809674 [ERR] switch_nat.c:197 Error checking for
> > PMP [init failed]
> > 2010-02-17 09:15:42.809674 [DEBUG] switch_nat.c:414 Checking for UPnP
> > 2010-02-17 09:15:56.825388 [DEBUG] switch_nat.c:114 No
> > InternetGatewayDevice, using first entry as default
> > (http://192.168.16.17:50144/).
> > 2010-02-17 09:15:56.887889 [INFO] switch_nat.c:429 No PMP or UPnP NAT
> > devices detected!
> >
> >
> >
> > On Tue, Feb 16, 2010 at 8:41 PM, Brian West <brian at freeswitch.org>
> wrote:
> >> can you please update...
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Regards,
> Moiz Chinoy.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100217/bb5f5244/attachment-0002.html 

From yehavi.bourvine at gmail.com  Wed Feb 17 11:37:52 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Wed, 17 Feb 2010 21:37:52 +0200
Subject: [Freeswitch-users] Announcement: FreeSWITCH Add Broadsoft SCA
	Support
In-Reply-To: <ADC8B813-4161-481C-877E-93FC012858B4@freeswitch.org>
References: <87f2f3b91001111742k691966cfs82515db77c13d4d4@mail.gmail.com>
	<d361ca7b1002090838j6c42dcd2of8affb5f281900fd@mail.gmail.com>
	<767B677E-0D19-43A9-89DE-D96EC4A715EC@freeswitch.org>
	<8D062B6E-4691-4BA6-B038-0EC6E8642DDA@jerris.com>
	<9D328485-D4C3-4B14-808A-CB63BBF06DC1@freeswitch.org>
	<d361ca7b1002170549g608d6e16vb2eb3f087fff9752@mail.gmail.com>
	<191c3a031002170556p5dfd4c55le3c279f194808d92@mail.gmail.com>
	<191c3a031002170557g5d8c9df9xf40d1b74b9a6f19d@mail.gmail.com>
	<d361ca7b1002170700x15630ab2tabc6f3d91ebc3303@mail.gmail.com>
	<ADC8B813-4161-481C-877E-93FC012858B4@freeswitch.org>
Message-ID: <d361ca7b1002171137u7ad51bfaia34176a3b8181f0f@mail.gmail.com>

Hello Brian,

  I'll try what you suggest on Friday, and if it works I will document it in
the wiki under the Polycom page I already wrote.

                                 Thanks, __Yehavi:

2010/2/17 Brian West <brian at freeswitch.org>

> Step 1. Enable manage-shared-appearance=true
>
> Step 2. Now in the phone's config Configure the phone as usually, set the
> line shared and DO NOT set the third party name.
>
> Step 3. Reboot
>
> It should work.
>
> I wish someone that has this working would write some wiki docs these
> threads about it not working are getting rather old when I know for a fact
> they work fine.
>
> The gateway info missing is a gateway you have configured getting a notify.
>  It has nothing to do with SCA.
>
> /b
>
> On Feb 17, 2010, at 9:00 AM, Yehavi Bourvine wrote:
>
> > .
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100217/e397f872/attachment-0002.html 

From anthony.minessale at gmail.com  Wed Feb 17 11:41:21 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 17 Feb 2010 13:41:21 -0600
Subject: [Freeswitch-users] Calls being parked on DTMF (2)
In-Reply-To: <0A1FDB5DAA23564F8758BA05D26DCD747F9A09@exchange.telecats.nl>
References: <0A1FDB5DAA23564F8758BA05D26DCD747F9A09@exchange.telecats.nl>
Message-ID: <191c3a031002171141ne49886en5552177104f10bf3@mail.gmail.com>

For the 10 gazillionth time:

Please report issues to jira, as soon as I read an email with a problem in
it, it's marked read and gets lost in a sea of 1000 other unread emails.
Jira is designed to track issues so we may help you.

Please indicate if you are using a board with an echo canceler on it or not?



On Wed, Feb 17, 2010 at 7:56 AM, Eelco Br?lman <e.brolman at telecats.nl>wrote:

> Hi all,
>
> I have the same problems as described in the thread "Calls being parked on
> DTMF".
>
> We have a lab-test setup, which consist of the following:
>
> 1) a box with 4 E1 trunks with a call generator application (let's call it
> CG)
> 2) connected with a FreeSwitch box (Sangoma A104D card) (let's call that
> FS)
>
> The CG dials a number on the FS box (number 921000), which is bridged out
> back to the CG server (number 1000). The initiating application sends some
> DTMF, which is replied (echo-ed) by the receiving application (both on the
> CG server).
>
> In roughly about 80% of the calls, FS just disconnects the call:
>
> 2010-02-17 14:30:14.677125 [DEBUG] switch_ivr_originate.c:3105 Originate
> Resulted in Success: [OpenZAP/1:1/1000]
> 2010-02-17 14:30:14.677125 [DEBUG] switch_core_session.c:638 Send signal
> OpenZAP/1:1/1000 [BREAK]
> 2010-02-17 14:30:14.677125 [DEBUG] switch_core_session.c:638 Send signal
> OpenZAP/1:60/921000 [BREAK]
> 2010-02-17 14:30:14.677125 [DEBUG] switch_ivr_bridge.c:1175
> (OpenZAP/1:1/1000) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA
> 2010-02-17 14:30:14.677125 [DEBUG] switch_core_session.c:1018 Send signal
> OpenZAP/1:1/1000 [BREAK]
> 2010-02-17 14:30:14.677125 [DEBUG] switch_core_state_machine.c:314
> (OpenZAP/1:1/1000) Running State Change CS_EXCHANGE_MEDIA
> 2010-02-17 14:30:14.677125 [DEBUG] switch_core_state_machine.c:351
> (OpenZAP/1:1/1000) State EXCHANGE_MEDIA
> 2010-02-17 14:30:14.677125 [DEBUG] mod_openzap.c:558 CHANNEL EXCHANGE_MEDIA
> 2010-02-17 14:30:14.687135 [DEBUG] switch_core_session.c:699 Send signal
> OpenZAP/1:1/1000 [BREAK]
> 2010-02-17 14:30:14.687135 [DEBUG] switch_core_session.c:699 Send signal
> OpenZAP/1:60/921000 [BREAK]
> 2010-02-17 14:30:15.986882 [DEBUG] mod_openzap.c:684 queue DTMF [7]
> 2010-02-17 14:30:16.026733 [DEBUG] zap_io.c:1870 1:1 GENERATE DTMF [7]
> 2010-02-17 14:30:16.106418 [DEBUG] switch_ivr_bridge.c:469 OpenZAP/1:1/1000
> ending bridge by request from write function
> 2010-02-17 14:30:16.106418 [DEBUG] switch_core_session.c:638 Send signal
> OpenZAP/1:60/921000 [BREAK]
> 2010-02-17 14:30:16.126335 [DEBUG] switch_ivr_bridge.c:475 OpenZAP/1:1/1000
> ending bridge by request from read function
> 2010-02-17 14:30:16.126335 [DEBUG] switch_core_session.c:638 Send signal
> OpenZAP/1:1/1000 [BREAK]
> 2010-02-17 14:30:16.126335 [DEBUG] switch_ivr_bridge.c:562 BRIDGE THREAD
> DONE [OpenZAP/1:60/921000]
> 2010-02-17 14:30:16.126335 [DEBUG] switch_ivr_bridge.c:582 Send signal
> OpenZAP/1:1/1000 [BREAK]
> 2010-02-17 14:30:16.126335 [DEBUG] switch_ivr_bridge.c:562 BRIDGE THREAD
> DONE [OpenZAP/1:1/1000]
> 2010-02-17 14:30:16.126335 [DEBUG] switch_ivr_bridge.c:582 Send signal
> OpenZAP/1:60/921000 [BREAK]
> 2010-02-17 14:30:16.126335 [NOTICE] switch_ivr_bridge.c:634 Hangup
> OpenZAP/1:1/1000 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
> 2010-02-17 14:30:16.126335 [DEBUG] switch_channel.c:2063 Send signal
> OpenZAP/1:1/1000 [KILL]
>
>
> Where the line "ending bridge by request from write function" is curious of
> course.
>
> I'm running:
> - Freeswitch trunk (r16674) (updated, make clean, reconfigured and make
> install this morning)
> - Wanpipe 3.5.10.smg-2 (see
> http://wiki.sangoma.com/wanpipe-SmgPriInstallation)
> - Using the ozmod_sangoma_boost SCTP
>
>
> If anyone needs more information, tests or config files, I'll be happy to
> provide them.
>
>
>
> Kind regards,
>
> Eelco Br?lman
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100217/62761552/attachment-0002.html 

From ledoktre at meanie.us  Wed Feb 17 11:43:06 2010
From: ledoktre at meanie.us (Doc)
Date: Wed, 17 Feb 2010 13:43:06 -0600
Subject: [Freeswitch-users] One more thing..
In-Reply-To: <87f2f3b91002171129v59518b2bg9673212cff957a6c@mail.gmail.com>
References: <4B7B5EE9.4020705@meanie.us>	<191c3a031002170535i12c82731r5829417387dd3228@mail.gmail.com>	<191c3a031002170536g297a816ch2fa94eb2dcfbd16d@mail.gmail.com>	<2CEBE2C8-3982-49B6-A06F-9406C1D1F123@freeswitch.org>	<4B7C253F.9090905@meanie.us>	<5A00F97E-41C9-43AD-8CFA-1C8B87E5EECC@freeswitch.org>	<4B7C2DC3.20502@meanie.us>
	<87f2f3b91002171129v59518b2bg9673212cff957a6c@mail.gmail.com>
Message-ID: <4B7C46CA.6060708@meanie.us>

I should be able to update to the latest trunk.  Thanks for including 
the instructions.  The only "modification" I did to the source was to 
edit the modules file and enable mod_skypiax.  I will repost with 
results once I obtain them.  -Thanks,

Michael Collins wrote:
> Are you able to update your FS to the latest trunk? ("make current" in 
> the FS src dir) Also, I notice that you are on 16619M - the "M" 
> indicates a modification of some kind. Did you make any changes to the 
> FS source?
>
> -MC



From ledoktre at meanie.us  Wed Feb 17 16:13:07 2010
From: ledoktre at meanie.us (Doc)
Date: Wed, 17 Feb 2010 18:13:07 -0600
Subject: [Freeswitch-users] One more thing..
In-Reply-To: <87f2f3b91002171129v59518b2bg9673212cff957a6c@mail.gmail.com>
References: <4B7B5EE9.4020705@meanie.us>	<191c3a031002170535i12c82731r5829417387dd3228@mail.gmail.com>	<191c3a031002170536g297a816ch2fa94eb2dcfbd16d@mail.gmail.com>	<2CEBE2C8-3982-49B6-A06F-9406C1D1F123@freeswitch.org>	<4B7C253F.9090905@meanie.us>	<5A00F97E-41C9-43AD-8CFA-1C8B87E5EECC@freeswitch.org>	<4B7C2DC3.20502@meanie.us>
	<87f2f3b91002171129v59518b2bg9673212cff957a6c@mail.gmail.com>
Message-ID: <4B7C8613.606@meanie.us>

I have updated to "FreeSWITCH Version 1.0.trunk (16679)" per your 
instructions.  My issues with this request all seem to be handled now.  
I had one or two things yet in my other posting, but I will respond 
there.  THANK YOU !!

Michael Collins wrote:
> Are you able to update your FS to the latest trunk? ("make current" in 
> the FS src dir) Also, I notice that you are on 16619M - the "M" 
> indicates a modification of some kind. Did you make any changes to the 
> FS source?
>
> -MC
>
> On Wed, Feb 17, 2010 at 9:56 AM, Doc <ledoktre at meanie.us 
> <mailto:ledoktre at meanie.us>> wrote:
>
>     I did try changing the rtp time in my Sipura (advanced, sip, RTP
>     Packet
>     Size) from 0.030 to 0.020, and the error went away.  I was able to
>     then
>     test (and succeed) dialing from secondary skype user to skypiax, and
>     have the call bridged automatically to one of my Sipura SPA-1001's.  I
>     could speak into the phone, and hear it come through skype (on my
>     laptop) no problem.  When I would talk on the laptop, I would get
>     garble
>     back on the phone, but that could be a sound issue on my laptop (my
>     skype sometimes does this with Pulse audio on my laptop, Ubuntu
>     Karmic).
>
>     The interesting thing it did do, however, was eventually (within a
>     minute?) It threw a couple of errors :
>
>     2010-02-17 11:47:34.007909 [ERR] mod_skypiax.c:826 rev
>     16619M[(nil)|37     ][ERRORA  826  ][interface1][-1, 5,21] EXIT?
>     sent=-1
>     2010-02-17 11:47:34.027776 [ERR] mod_skypiax.c:826 rev
>     16619M[(nil)|37     ][ERRORA  826  ][interface1][-1, 5,21] EXIT?
>     sent=-1
>     2010-02-17 11:47:34.047643 [ERR] mod_skypiax.c:826 rev
>     16619M[(nil)|37     ][ERRORA  826  ][interface1][-1, 5,21] EXIT?
>     sent=-1
>
>     And on a subsequent test, I received no audio (the above error rolling
>     on the console), and within a matter of seconds, FS crashed with
>     this :
>
>     Segmentation fault (core dumped)
>
>     I have another request open on the group actually for the above error,
>     so I think it'd make sense to discuss further there (rather than
>     duplicating material).
>
>     Thanks!  At any rate, the error for the missing codec, etc, seemed
>     to be
>     gone once I updated the RTP time in my ATA.  Hope this helps someone!!
>
>     Brian West wrote:
>     > change the rtp time to 0.020 from 0.030
>     >
>     > /b
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   



From ledoktre at meanie.us  Wed Feb 17 16:19:07 2010
From: ledoktre at meanie.us (Doc)
Date: Wed, 17 Feb 2010 18:19:07 -0600
Subject: [Freeswitch-users] One more thing..
In-Reply-To: <191c3a031002171123r416f6413wbecddcea498b5616@mail.gmail.com>
References: <4B7B5EE9.4020705@meanie.us>	<191c3a031002170535i12c82731r5829417387dd3228@mail.gmail.com>	<191c3a031002170536g297a816ch2fa94eb2dcfbd16d@mail.gmail.com>	<2CEBE2C8-3982-49B6-A06F-9406C1D1F123@freeswitch.org>	<4B7C253F.9090905@meanie.us>	<5A00F97E-41C9-43AD-8CFA-1C8B87E5EECC@freeswitch.org>	<4B7C2DC3.20502@meanie.us>
	<191c3a031002171123r416f6413wbecddcea498b5616@mail.gmail.com>
Message-ID: <4B7C877B.1080303@meanie.us>

I have updated to "FreeSWITCH Version 1.0.trunk (16679)" which seems to 
have cleared up part # 2 of my question (issues with the errors being 
thrown on the screen during a call, and then subsequent calls failing).

About the DTMF issues, for fun, I got onto Skype chat.  I was told by 
the technician there that Skype does not support DTMF on SkypeIn -OR- 
Skype for SIP.  He said sorry, but you won't get anywhere.  I was also 
told they are not doing any blocking, that the reason its not working is 
probably due to all the transcoding that is taking place has degraded 
the signal too much.  I guess I might try and see if there is a module 
available for a voice driven IVR? (any suggestions?)

And on the rest of my first posting, any thoughts on why I must push a # 
sign when dialing an extension, etc, in FS?  If I don't, I have to wait 
like 30 seconds.  Also, the errors when I run startskype,sh (I am 
thinking must be no big deal) listed, are they any problem :

expected keysym, got XF86KbdLightOnOff: line 70 of pc
expected keysym, got XF86KbdBrightnessDown: line 71 of pc
expected keysym, got XF86KbdBrightnessUp: line 72 of pc
Could not init font path element /usr/share/fonts/X11/cyrillic, removing 
from list!
Could not init font path element 
/var/lib/defoma/x-ttcidfont-conf.d/dirs/TrueType, removing from list!
ALSA lib pcm.c:2211:(snd_pcm_open_noupdate) Unknown PCM null
ALSA lib pcm.c:2211:(snd_pcm_open_noupdate) Unknown PCM null

Thanks again you guys for all your help!

Doc

Anthony Minessale wrote:
> keep updating, the maintainer of mod_skypeiax is adding new patches 
> every few hours.
> You can also join irc on irc.freenode.net <http://irc.freenode.net> 
> #freeswitch and #freeswitch-dev to interact with him live.
>
>
> One hint around here when we ask if you updated we work in 1 minute 
> increments.



From anthony.minessale at gmail.com  Wed Feb 17 16:37:17 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 17 Feb 2010 18:37:17 -0600
Subject: [Freeswitch-users] One more thing..
In-Reply-To: <191c3a031002171636h7232cc81kd54c22532ba04b28@mail.gmail.com>
References: <4B7B5EE9.4020705@meanie.us>
	<191c3a031002170536g297a816ch2fa94eb2dcfbd16d@mail.gmail.com>
	<2CEBE2C8-3982-49B6-A06F-9406C1D1F123@freeswitch.org>
	<4B7C253F.9090905@meanie.us>
	<5A00F97E-41C9-43AD-8CFA-1C8B87E5EECC@freeswitch.org>
	<4B7C2DC3.20502@meanie.us>
	<191c3a031002171123r416f6413wbecddcea498b5616@mail.gmail.com>
	<4B7C877B.1080303@meanie.us>
	<191c3a031002171634j33f70843j40ce8a3d60044d0b@mail.gmail.com>
	<191c3a031002171636h7232cc81kd54c22532ba04b28@mail.gmail.com>
Message-ID: <191c3a031002171637t5c1a2339jadacf8cd89b6fe52@mail.gmail.com>

One hint is to add keywords to the subject like "problems with skypeiax" so
the guy will see your issue he doesn't read every mail looking for skype
issues.  Even better, open a new issue on jira.freeswitch.org under skypeiax
related category......

On Feb 17, 2010 6:23 PM, "Doc" <ledoktre at meanie.us> wrote:

I have updated to "FreeSWITCH Version 1.0.trunk (16679)" which seems to
have cleared up part # 2 of my question (issues with the errors being
thrown on the screen during a call, and then subsequent calls failing).

About the DTMF issues, for fun, I got onto Skype chat.  I was told by
the technician there that Skype does not support DTMF on SkypeIn -OR-
Skype for SIP.  He said sorry, but you won't get anywhere.  I was also
told they are not doing any blocking, that the reason its not working is
probably due to all the transcoding that is taking place has degraded
the signal too much.  I guess I might try and see if there is a module
available for a voice driven IVR? (any suggestions?)

And on the rest of my first posting, any thoughts on why I must push a #
sign when dialing an extension, etc, in FS?  If I don't, I have to wait
like 30 seconds.  Also, the errors when I run startskype,sh (I am
thinking must be no big deal) listed, are they any problem :

expected keysym, got XF86KbdLightOnOff: line 70 of pc
expected keysym, got XF86KbdBrightnessDown: line 71 of pc
expected keysym, got XF86KbdBrightnessUp: line 72 of pc
Could not init font path element /usr/share/fonts/X11/cyrillic, removing
from list!
Could not init font path element
/var/lib/defoma/x-ttcidfont-conf.d/dirs/TrueType, removing from list!
ALSA lib pcm.c:2211:(snd_pcm_open_noupdate) Unknown PCM null
ALSA lib pcm.c:2211:(snd_pcm_open_noupdate) Unknown PCM null

Thanks again you guys for all your help!

Doc


Anthony Minessale wrote:
> keep updating, the maintainer of mod_skypeiax is adding new patches
> e...
> You can also join irc on irc.freenode.net <http://irc.freenode.net>

> #freeswitch and #freeswitch-dev to interact with him live.
>
>
> One hint around here when we ask ...

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists...
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100217/8591a34b/attachment-0002.html 

From troy at tlainvestments.com  Wed Feb 17 16:44:36 2010
From: troy at tlainvestments.com (Troy Anderson)
Date: Wed, 17 Feb 2010 17:44:36 -0700
Subject: [Freeswitch-users] tone_detect timeout
Message-ID: <EC840323-2A76-426E-A386-8DA111195768@tlainvestments.com>

In the wiki about tone_detect, the docs state that the timeout is in seconds (e.g. +2 for 2 seconds from now), but all the examples have +5000, suggesting that it may really be milliseconds.  I'd be happy to update the wiki if someone could say if it's seconds or millisconds or otherwise.

I tried tracing it back in the code, but got lost looking for the definition of switch_media_bug_t!

http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_tone_detect

Thanks!
Troy


From wangdq.no1 at gmail.com  Wed Feb 17 19:17:26 2010
From: wangdq.no1 at gmail.com (daqiang wang)
Date: Thu, 18 Feb 2010 11:17:26 +0800
Subject: [Freeswitch-users] how to use mod_erlang ?
Message-ID: <ecd05ce31002171917w2713f8detcee3f7e36039e9f2@mail.gmail.com>

hello !
I test mod_erlang from : http://wiki.freeswitch.org/wiki/Mod_erlang_event
but when I input :
 >

{foo, freeswitch at localhost } ! {api, status, ""}.
I received
  (test at wangdq-laptop)2>
=ERROR REPORT==== 18-Feb-2010::11:10:13 ===
Error in process <0.41.0> on node 'test at wangdq-laptop' with exit
value: {badarg,[{erlang,list_to_existing_atom,["freeswitch at wangdq-laptop"]},{dist_util,recv_challenge,1},{dist_util,handshake_we_started,1}]}

and at freeswitch console :
   [NOTICE] mod_erlang_event.c:1720 Ignorable error in ei_accept -
probable bad client version, bad cookie or bad nodename


and I think the mod_erlang config file is :


<configuration name="erlang_event.conf" description="Erlang Socket Client">
  <settings>
    <param name="listen-ip" value="0.0.0.0"/>
    <param name="listen-port" value="8031"/>
    <param name="cookie" value="ClueCon"/>
    <param name="shortname" value="true"/>
    <!-- in additon to cookie, optionally restrict by ACL -->
    <param name="apply-inbound-acl" value="lan"/>
    <!-- alternative is "binary" -->
    <param name="encoding" value="string"/>
    <!-- provide compatability with previous OTP release (use with care) -->
    <!--<param name="compat-rel" value="12"/> -->
  </settings>
</configuration>
~


 Thanks !
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100218/055ca0c0/attachment-0002.html 

From mike at jerris.com  Wed Feb 17 21:10:54 2010
From: mike at jerris.com (Michael Jerris)
Date: Thu, 18 Feb 2010 00:10:54 -0500
Subject: [Freeswitch-users] ivr from mysql
In-Reply-To: <7c74f5761002151525p7cc690ces632d46c6ab5e236c@mail.gmail.com>
References: <7c74f5761002150824i41a2ed5did008f36cbb4499a4@mail.gmail.com>
	<87f2f3b91002151002v6bbc3e7fgb67dd6d3e72a915@mail.gmail.com>
	<7c74f5761002151425g512fa685n13c58d8f4b894740@mail.gmail.com>
	<0FD30AAA-C2E2-4228-A09B-DB09F4E26811@avgs.ca>
	<7c74f5761002151525p7cc690ces632d46c6ab5e236c@mail.gmail.com>
Message-ID: <E9EBF591-5DEC-4FF3-B2D8-20DDA54B4FF5@jerris.com>

an example is available here :   http://svn.freeswitch.org/svn/freeswitch/trunk/conf/ivr_menus/demo_ivr.xml

Mike

On Feb 15, 2010, at 6:25 PM, michal kalinowski wrote:
> Could you insert several examples here?




From mike at jerris.com  Wed Feb 17 21:13:08 2010
From: mike at jerris.com (Michael Jerris)
Date: Thu, 18 Feb 2010 00:13:08 -0500
Subject: [Freeswitch-users] Sending message notifications with openzap
In-Reply-To: <e749d5a76cc31b4a3bb2459aaf8b78d9@thom.fr.eu.org>
References: <e749d5a76cc31b4a3bb2459aaf8b78d9@thom.fr.eu.org>
Message-ID: <1B676793-9D83-4C8D-A9C4-78891F8C70B5@jerris.com>

This all uses the event system.  Take a look at how we handle the events in mod_sofia, it should be pretty easy to do the same in mod_openzap.

Mike

On Feb 16, 2010, at 4:33 AM, Fran?ois Legal wrote:
> I was wondering whether or not a facility was available to send message waiting indicator (from mpd_voicemail) using openzap.
> 
> I know this feature is not available in openzap (I'm in the process of coding it for analog channels) but as message indication can be sent via mod_sofia, I wonder if a frame is available in FS core.
> 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100218/0fd81b01/attachment-0002.html 

From scottferri09 at gmail.com  Thu Feb 18 00:12:04 2010
From: scottferri09 at gmail.com (Scott Fernandez)
Date: Thu, 18 Feb 2010 13:42:04 +0530
Subject: [Freeswitch-users] Establishing a Call from .Net based
	application
In-Reply-To: <6E8D2069C08AA84A83D336E996AE4C67032C9D5346@mse17be1.mse17.exchange.ms>
References: <874941.17255.qm@web33502.mail.mud.yahoo.com>
	<df97b9c51001310545n54195cadr9faa3db46917ca9e@mail.gmail.com>
	<367751821002010737o4ca83476j62c96cb60c8a456a@mail.gmail.com>
	<df97b9c51002130752h48aae119ja80797ab8a3f5b2d@mail.gmail.com>
	<6E8D2069C08AA84A83D336E996AE4C67032C9D5346@mse17be1.mse17.exchange.ms>
Message-ID: <df97b9c51002180012wc4d9421ye6e81ae6199321c8@mail.gmail.com>

Hi Diego & Michael,

Thanks for your reply and support.

However, I have some clarifications required from both of you.

1. Here is the question for Diego,

Simple Example:

using FreeSWITCH;
using FreeSWITCH.Native;

namespace BITS.Ivr.Bp.Server
{
 public class CIV_BPFSProcess : FreeSWITCH.IAppPlugin
{
 public void Run(AppContext context)
 {
  //answer call
  context.Session.Answer();
  //sleep 2 seconds
  context.Session.sleep(2000, 1);
  //hangup call
  context.Session.Hangup("
NORMAL_CLEARING");
 }
 }
}

I understand that the concept of your example code. However, would like to
know as to how would my .NET C# know the IP address of Freeswitch to talk to
it as there is no indication for that?. If not here, where would we need to
reference the IP address of FS in .NET code?

I guess the IP address of FS needs to be mentioned in the Target section of
the below web.config file in .NET. If I am right, how to specify the IP
address over here. If I am wrong, please let me know where do we need to
mention the IP address of FS.

    <configuration>
            <dllmap dll="mod_managed.dll" target="mod_managed.so"/>
    </configuration>


2. Here is the question for Michael,

You mentioned that "mod_managed.so will be in your freeswitch mod directory".
This is very clear and what is mod_managed.dll in my .NET application and
the purpose of it?

Thanks for all your help.

Regards,
Scott.



On Sun, Feb 14, 2010 at 1:15 AM, Michael Giagnocavo <mgg at giagnocavo.net>wrote:

>
> 2. There is a configuration settings required to Map the "DLL" to ".so"
> object in CentOS.
> Now, the question is which DLL and .so file to be made available and where?
>
> ?
>
> If you are experiencing NullReferenceExceptions with any plugin run through
> the dialplan, make sure you have included the appropriate entry in your
> dllmap <http://mono-project.com/Config_DllMap> configuration:
>
>  <dllmap dll="mod_managed.dll" target="mod_managed.so" os="!windows"/>
>
> ?
>
> mod_managed.so will be in your freeswitch mod directory.
>
>
> All I need is to initiate a call from .NET application and then it should
> talk to mod_managed module and establish a call. Secondly, I need to know
> the status of the call such as Ringing, Active, Hangup etc.
>
> To initiate a call, try ManagedSession.Originate.
>
> -Michael
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100218/3fadf0d3/attachment-0002.html 

From mcampbellsmith at gmail.com  Thu Feb 18 00:22:54 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Thu, 18 Feb 2010 19:22:54 +1100
Subject: [Freeswitch-users] Forbidden using UDP, works with TCP/TLS
In-Reply-To: <147993F8-9C9E-4D96-9AFE-5EEE4E8F93F0@freeswitch.org>
References: <33c87fa31002101824h21e3bf7lb13b68e8d4fd36a2@mail.gmail.com>
	<33c87fa31002130343k7a01e9abt9c33360399f246f8@mail.gmail.com>
	<C37AB6F3-F712-41B6-9D6D-7A8AE3C6CABB@freeswitch.org>
	<33c87fa31002131336j7b05db7aof6ba6e3439aebc41@mail.gmail.com>
	<9D702362-BB5F-4C0F-9EE7-DABA3B926C36@freeswitch.org>
	<33c87fa31002141708s72cb7246h5372ecea40be615d@mail.gmail.com>
	<1888B1CA-D0A8-407C-9C44-B86E0F1925D6@freeswitch.org>
	<33c87fa31002142017v49d8e7c5nf572d086ec89282b@mail.gmail.com>
	<33c87fa31002142109l5fcabed3ydba21b865ed6cbbb@mail.gmail.com>
	<147993F8-9C9E-4D96-9AFE-5EEE4E8F93F0@freeswitch.org>
Message-ID: <33c87fa31002180022g2e91c8efs65f52bb4275cb59@mail.gmail.com>

Thanks for looking at that Brian.  If the token is not used, I assume
this is not the reason for FS rejecting the Registration attempt?

Also, when is stale=true set in the WWW-Authentication?  I notice that
for this device, I do not see stale=true, but for all my other
devices, I see stale=true (at least from the logs I've taken today).


On Tue, Feb 16, 2010 at 2:03 AM, Brian West <brian at freeswitch.org> wrote:
> Ok looks like the token is not used at all in digest auth. ?This is the
> first time I have seen a device send back something other than auth or
> auth-int.
> /b
> On Feb 14, 2010, at 11:09 PM, Mark Campbell-Smith wrote:
>
> A little more testing. ???I noticed that the Authorization field
> differs when TCP or UDP:
>
> UDP (fails)
> Digest username=\"2010\", realm=\"mydns.dyndns.org\",
> nonce=\"e5f119c6-19e9-11df-bd09-773b7a755f78\",
> uri=\"sip:mydns.dyndns.org:5060\",
> response=\"e37be3e49c159d4f98e8bd04b36f2bd7\", algorithm=MD5,
> qop=\"1fffcc9f\"
>
> TCP (works)
> Digest
> username=\"2010\",realm=\"mydns.dyndns.org\",nonce=\"5d9e75c2-19ea-11df-bd0b-773b7a755f78\",uri=\"sip:mydns.dyndns.org:5060\",algorithm=MD5,response=\"45ba55d3fbafcbf2bc2aa6418656ecc2\",qop=auth,nc=00000001,cnonce=\"3a650454\"
>
> Is qop = 1fffcc9f valid in SIP? ?Does a cnonce need to be included also?
>
> FS sends qop=auth in the Unauthorized response.
>
> Thanks
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From brian at microcomaustralia.com.au  Thu Feb 18 00:46:38 2010
From: brian at microcomaustralia.com.au (Brian May)
Date: Thu, 18 Feb 2010 19:46:38 +1100
Subject: [Freeswitch-users] building for Lenny
Message-ID: <3c5cf5261002180046vb0015e7t787599c9903aedc@mail.gmail.com>

Hello,

What is the most recent instructions?

According to <http://wiki.freeswitch.org/wiki/Installation_Guide>, I
should install the freeswitch-sounds-music-8000 package, but no sounds
packages are created in the build process.

Thanks
-- 
Brian May <brian at microcomaustralia.com.au>



From mcampbellsmith at gmail.com  Thu Feb 18 00:56:47 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Thu, 18 Feb 2010 19:56:47 +1100
Subject: [Freeswitch-users] building for Lenny
In-Reply-To: <3c5cf5261002180046vb0015e7t787599c9903aedc@mail.gmail.com>
References: <3c5cf5261002180046vb0015e7t787599c9903aedc@mail.gmail.com>
Message-ID: <33c87fa31002180056g51ef8496qa9a35746d0fe75ec@mail.gmail.com>

I think you need to download the gzip file from http://files.freeswitch.org/

latest.freeswitch.org does not contain sound files as far as I'm aware ..

On Thu, Feb 18, 2010 at 7:46 PM, Brian May
<brian at microcomaustralia.com.au> wrote:
> Hello,
>
> What is the most recent instructions?
>
> According to <http://wiki.freeswitch.org/wiki/Installation_Guide>, I
> should install the freeswitch-sounds-music-8000 package, but no sounds
> packages are created in the build process.
>
> Thanks
> --
> Brian May <brian at microcomaustralia.com.au>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From brian at microcomaustralia.com.au  Thu Feb 18 01:10:04 2010
From: brian at microcomaustralia.com.au (Brian May)
Date: Thu, 18 Feb 2010 20:10:04 +1100
Subject: [Freeswitch-users] building for Lenny
In-Reply-To: <33c87fa31002180056g51ef8496qa9a35746d0fe75ec@mail.gmail.com>
References: <3c5cf5261002180046vb0015e7t787599c9903aedc@mail.gmail.com>
	<33c87fa31002180056g51ef8496qa9a35746d0fe75ec@mail.gmail.com>
Message-ID: <3c5cf5261002180110he0ae815uc713f77f289ac625@mail.gmail.com>

On 18 February 2010 19:56, Mark Campbell-Smith <mcampbellsmith at gmail.com> wrote:
> I think you need to download the gzip file from http://files.freeswitch.org/
>
> latest.freeswitch.org does not contain sound files as far as I'm aware ..

Does this mean I shouldn't be trying to use the debian packages? *.deb
files would certainly make it easier to compile it on a fast computer
and then transfer to my net5501.

Unfortunately, the website seems have a lot of old information,
including references to obsolete Ubuntu Hardy packages.

I see a thread from late last year that suggests there should be
prebuilt packages, everything I can find so far seems very old
however.
-- 
Brian May <brian at microcomaustralia.com.au>



From devel at thom.fr.eu.org  Thu Feb 18 01:50:30 2010
From: devel at thom.fr.eu.org (=?UTF-8?Q?Fran=C3=A7ois_Legal?=)
Date: Thu, 18 Feb 2010 10:50:30 +0100
Subject: [Freeswitch-users] Sending message notifications with openzap
In-Reply-To: <1B676793-9D83-4C8D-A9C4-78891F8C70B5@jerris.com>
References: <e749d5a76cc31b4a3bb2459aaf8b78d9@thom.fr.eu.org>
	<1B676793-9D83-4C8D-A9C4-78891F8C70B5@jerris.com>
Message-ID: <0ae2b96926e5340a50e659891efb26cc@thom.fr.eu.org>



Thanks, 

yes I could see that this was handled with events. 

Could you
please give more details (that does not seem obvious to me while looking at
sofia_presence.c) 

When and why is the event triggered ? 

What
information do I get with the event ? 

Then I have a design question, is
it mandatory to (as this is done in mod_sofia but I guess for a lot of
reasons) process the event in a separate thread ? 

Thanks for the
information (and sorry if my questions are not relevant, I don't know the
event system at all). 

Fran?ois 

On Thu, 18 Feb 2010 00:13:08 -0500,
Michael Jerris  wrote: This all uses the event system. Take a look at how
we handle the events in mod_sofia, it should be pretty easy to do the same
in mod_openzap.  Mike 
  On Feb 16, 2010, at 4:33 AM, Fran?ois Legal wrote:


I was wondering whether or not a facility was available to send message
waiting indicator (from mpd_voicemail) using openzap. 

I know this feature
is not available in openzap (I'm in the process of coding it for analog
channels) but as message indication can be sent via mod_sofia, I wonder if
a frame is available in FS core.     

 
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100218/a4daf1bd/attachment-0002.html 

From mgg at giagnocavo.net  Thu Feb 18 03:12:29 2010
From: mgg at giagnocavo.net (Michael Giagnocavo)
Date: Thu, 18 Feb 2010 06:12:29 -0500
Subject: [Freeswitch-users] Establishing a Call from .Net
	based	application
In-Reply-To: <df97b9c51002180012wc4d9421ye6e81ae6199321c8@mail.gmail.com>
References: <874941.17255.qm@web33502.mail.mud.yahoo.com>
	<df97b9c51001310545n54195cadr9faa3db46917ca9e@mail.gmail.com>
	<367751821002010737o4ca83476j62c96cb60c8a456a@mail.gmail.com>
	<df97b9c51002130752h48aae119ja80797ab8a3f5b2d@mail.gmail.com>
	<6E8D2069C08AA84A83D336E996AE4C67032C9D5346@mse17be1.mse17.exchange.ms>
	<df97b9c51002180012wc4d9421ye6e81ae6199321c8@mail.gmail.com>
Message-ID: <6E8D2069C08AA84A83D336E996AE4C67032C9D5BAD@mse17be1.mse17.exchange.ms>

I'm not sure what the FreeSWITCH APIs are to figure out what IP Sofia SIP has bound to. Whatever it is, you'd call the same thing in C#. What do you want to do with the API?

mod_managed.dll or .so is the FreeSWITCH native code module that loads the CLR or Mono into the FreeSWITCH process and loads FreeSWITCH.Managed.dll. The managed DLL contains the bulk of the managed-unmanaged interop code (.NET definitions of all the FS C functions).

-Michael

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Scott Fernandez
Sent: Thursday, February 18, 2010 1:12 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Establishing a Call from .Net based application

Hi Diego & Michael,

Thanks for your reply and support.

However, I have some clarifications required from both of you.

1. Here is the question for Diego,

Simple Example:

using FreeSWITCH;
using FreeSWITCH.Native;

namespace BITS.Ivr.Bp.Server
{
 public class CIV_BPFSProcess : FreeSWITCH.IAppPlugin
{
 public void Run(AppContext context)
 {
  //answer call
  context.Session.Answer();
  //sleep 2 seconds
  context.Session.sleep(2000, 1);
  //hangup call
  context.Session.Hangup("
NORMAL_CLEARING");
 }
 }
}

I understand that the concept of your example code. However, would like to know as to how would my .NET C# know the IP address of Freeswitch to talk to it as there is no indication for that?. If not here, where would we need to reference the IP address of FS in .NET code?

I guess the IP address of FS needs to be mentioned in the Target section of the below web.config file in .NET. If I am right, how to specify the IP address over here. If I am wrong, please let me know where do we need to mention the IP address of FS.

    <configuration>
            <dllmap dll="mod_managed.dll" target="mod_managed.so"/>
    </configuration>


2. Here is the question for Michael,

You mentioned that "mod_managed.so will be in your freeswitch mod directory". This is very clear and what is mod_managed.dll in my .NET application and the purpose of it?

Thanks for all your help.

Regards,
Scott.


On Sun, Feb 14, 2010 at 1:15 AM, Michael Giagnocavo <mgg at giagnocavo.net<mailto:mgg at giagnocavo.net>> wrote:

2. There is a configuration settings required to Map the "DLL" to ".so" object in CentOS.
Now, the question is which DLL and .so file to be made available and where?
"
If you are experiencing NullReferenceExceptions with any plugin run through the dialplan, make sure you have included the appropriate entry in your dllmap<http://mono-project.com/Config_DllMap> configuration:
 <dllmap dll="mod_managed.dll" target="mod_managed.so" os="!windows"/>
"
mod_managed.so will be in your freeswitch mod directory.

All I need is to initiate a call from .NET application and then it should talk to mod_managed module and establish a call. Secondly, I need to know the status of the call such as Ringing, Active, Hangup etc.
To initiate a call, try ManagedSession.Originate.
-Michael

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100218/adeb933c/attachment-0002.html 

From matt at webcontracts.co.uk  Thu Feb 18 04:02:50 2010
From: matt at webcontracts.co.uk (Matthew Law)
Date: Thu, 18 Feb 2010 12:02:50 -0000
Subject: [Freeswitch-users] building for Lenny
In-Reply-To: <3c5cf5261002180110he0ae815uc713f77f289ac625@mail.gmail.com>
References: <3c5cf5261002180046vb0015e7t787599c9903aedc@mail.gmail.com>
	<33c87fa31002180056g51ef8496qa9a35746d0fe75ec@mail.gmail.com>
	<3c5cf5261002180110he0ae815uc713f77f289ac625@mail.gmail.com>
Message-ID: <a9f4977a70b3fcf1b30413ea562ca962.squirrel@www.webcontracts.co.uk>


On Thu, February 18, 2010 9:10 am, Brian May wrote:
> On 18 February 2010 19:56, Mark Campbell-Smith <mcampbellsmith at gmail.com>
> wrote:
>> I think you need to download the gzip file from
>> http://files.freeswitch.org/
>>
>> latest.freeswitch.org does not contain sound files as far as I'm aware
>> ..
>
> Does this mean I shouldn't be trying to use the debian packages? *.deb
> files would certainly make it easier to compile it on a fast computer
> and then transfer to my net5501.
>
> Unfortunately, the website seems have a lot of old information,
> including references to obsolete Ubuntu Hardy packages.
>
> I see a thread from late last year that suggests there should be
> prebuilt packages, everything I can find so far seems very old
> however.

Brian,

I had similar problems recently and decided to compile it from svn trunk. 
The dependencies for the build are all available in apt, so I installed
those, checked out trunk and configured it with a prefix of
/usr/local/freeswitch.

I got the impression it is self contained, so you should be able to tar up
the entire /usr/local/freeswitch dir and scp it over - I stand to be
corrected on that, though.

Good luck with it,

Matt.





From scott.torr.fs at letterboxes.org  Thu Feb 18 04:35:52 2010
From: scott.torr.fs at letterboxes.org (Scott Torr)
Date: Thu, 18 Feb 2010 23:35:52 +1100
Subject: [Freeswitch-users] One more thing..
In-Reply-To: <4B7C877B.1080303@meanie.us>
References: <4B7B5EE9.4020705@meanie.us>
	<191c3a031002170535i12c82731r5829417387dd3228@mail.gmail.com>
	<191c3a031002170536g297a816ch2fa94eb2dcfbd16d@mail.gmail.com>
	<2CEBE2C8-3982-49B6-A06F-9406C1D1F123@freeswitch.org>
	<4B7C253F.9090905@meanie.us>
	<5A00F97E-41C9-43AD-8CFA-1C8B87E5EECC@freeswitch.org>
	<4B7C2DC3.20502@meanie.us><191c3a031002171123r416f6413wbecddcea498b5616@mail.gmail.com>
	<4B7C877B.1080303@meanie.us>
Message-ID: <1266496552.11533.1360626205@webmail.messagingengine.com>

On Wed, 17 Feb 2010 18:19 -0600, "Doc" <ledoktre at meanie.us> wrote:
> About the DTMF issues, for fun, I got onto Skype chat.  I was told by 
> the technician there that Skype does not support DTMF on SkypeIn -OR- 
> Skype for SIP.  He said sorry, but you won't get anywhere. 



Hi Doc,

I can confirm that out-of-band DTMF signaling works for the Skype SIP
product using a SkypeIn PSTN number.



>From the Skype support site.
https://support.skype.com/en/faq/FA10292/Do-you-support-DTMF-with-Skype-for-SIP?frompage=search&q=dtmf

<quote>
Do you support DTMF with Skype for SIP?

Yes. Skype supports out-of-band DTMF signaling in accordance with the
RFC 2833 standard. Details of RFC 2833 are located on the IETF website.

We do *not* support in-band DTMF signaling. 
</quote>


Regards,
Scott Torr


regards



From vetali100 at gmail.com  Thu Feb 18 04:52:46 2010
From: vetali100 at gmail.com (Vitalii Colosov)
Date: Thu, 18 Feb 2010 14:52:46 +0200
Subject: [Freeswitch-users] Other party does not hear voice when
	connecting with X-Lite
In-Reply-To: <3e6d7b0c1002150050g3ccd23aas2e7f05f230bff4a7@mail.gmail.com>
References: <3e6d7b0c1002150050g3ccd23aas2e7f05f230bff4a7@mail.gmail.com>
Message-ID: <a858e7691002180452n6ef91ed7k3d9b2f7246d773f1@mail.gmail.com>

Thanks a lot for your advices and sorry for confusion, my description was
not consistent, agree...

We are testing X-Lite client from 4 different machines and using 2 different
FS servers and getting different results - sometimes everything works,
sometimes only signalling works - no voice, and sometimes even signalling
does not work.
YATE client always works without any such issues...

When we will come to some sort of understanding, I will try to share the
results.
If we will stuck, I will try to get some debug information and continue
bothering the professionals. :)

 BTW, I tried to change the codecs in X-Lite, same result.
It is for sure related to our network configuration + X-Lite's way of
sending SIP data, looks like...

Regards,
Vitalii





2010/2/15 Steven Brown <steve at justfone.com>

> I had the same problem with XLite / Freeswitch  a while back that I never
> fully understood,  however the problem vanished when I disabled all codecs
> on Xlite except G711 uLaw, as I say, no idea what was going on but this
> might be worth trying.
>
> Steve
>
>
> Message: 1
> Date: Sun, 14 Feb 2010 09:04:09 -0600
> From: Anthony Minessale <anthony.minessale at gmail.com>
> Subject: Re: [Freeswitch-users] Other party does not hear voice when
>        connecting with X-Lite
> To: freeswitch-users at lists.freeswitch.org
> Message-ID:
>        <191c3a031002140704g705bfc73rd8dd103f3d846062 at mail.gmail.com
> >
> Content-Type: text/plain; charset="iso-8859-1"
>
>
> You need to describe this again its too confusing now.
> List each device, freeswitch, the phones and which ip and combo of addrs it
> uses with the topology clearly stated.
> Your attempt to simplify your explanation is actually making it harder to
> follow.
> Also consider a debug/sip trace as well.
>
> Include
> sofia status profile default.
>
> Then capture a test call after entering these commands.
>
> console loglevel debug.
> sofa profile internal siptrace on
>
> On Feb 14, 2010 4:12 AM, "Vitalii Colosov" <vetali100 at gmail.com> wrote:
>
> No, it is done on the different PCs...
>
> Sorry, when I started the topic, I have described the problem how it is
> visible from PC of my friend.
> Then I tried to reproduce the same on my own PC, and you are right...I was
> not able to hear anything as well, not only both party wasn't.
> Also, from my PC I was NOT able to hear guitar on test number "9999".
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100218/9d8e2527/attachment-0002.html 

From brian at freeswitch.org  Thu Feb 18 05:04:13 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 18 Feb 2010 07:04:13 -0600
Subject: [Freeswitch-users] Forbidden using UDP, works with TCP/TLS
In-Reply-To: <33c87fa31002180022g2e91c8efs65f52bb4275cb59@mail.gmail.com>
References: <33c87fa31002101824h21e3bf7lb13b68e8d4fd36a2@mail.gmail.com>
	<33c87fa31002130343k7a01e9abt9c33360399f246f8@mail.gmail.com>
	<C37AB6F3-F712-41B6-9D6D-7A8AE3C6CABB@freeswitch.org>
	<33c87fa31002131336j7b05db7aof6ba6e3439aebc41@mail.gmail.com>
	<9D702362-BB5F-4C0F-9EE7-DABA3B926C36@freeswitch.org>
	<33c87fa31002141708s72cb7246h5372ecea40be615d@mail.gmail.com>
	<1888B1CA-D0A8-407C-9C44-B86E0F1925D6@freeswitch.org>
	<33c87fa31002142017v49d8e7c5nf572d086ec89282b@mail.gmail.com>
	<33c87fa31002142109l5fcabed3ydba21b865ed6cbbb@mail.gmail.com>
	<147993F8-9C9E-4D96-9AFE-5EEE4E8F93F0@freeswitch.org>
	<33c87fa31002180022g2e91c8efs65f52bb4275cb59@mail.gmail.com>
Message-ID: <7EA7C626-D317-4011-BF6E-A7411C765EC1@freeswitch.org>

I think your device is broken.

/b

On Feb 18, 2010, at 2:22 AM, Mark Campbell-Smith wrote:

> Thanks for looking at that Brian.  If the token is not used, I assume
> this is not the reason for FS rejecting the Registration attempt?
> 
> Also, when is stale=true set in the WWW-Authentication?  I notice that
> for this device, I do not see stale=true, but for all my other
> devices, I see stale=true (at least from the logs I've taken today).




From testeador01 at gmail.com  Thu Feb 18 05:27:15 2010
From: testeador01 at gmail.com (Milena)
Date: Thu, 18 Feb 2010 08:27:15 -0500
Subject: [Freeswitch-users] Greetings and a couple of questions
In-Reply-To: <4B7C261C.1040802@meanie.us>
References: <4B7B5A46.3090304@meanie.us>
	<1266408503.10430.1360421639@webmail.messagingengine.com>
	<4B7C261C.1040802@meanie.us>
Message-ID: <c29b930b1002180527l5f8c3f40icd905072f302f651@mail.gmail.com>

Hello!

For ASR menus check this out:
http://wiki.freeswitch.org/wiki/Mod_pocketsphinx
<http://wiki.freeswitch.org/wiki/Mod_pocketsphinx>I wouldn't say it is
perfect but it is nice, on the other hand there is this dialplan tool that
makes fs detect inband dtmf:
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf

<http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf>hope it
helps.

-Milena

2010/2/17 Doc <ledoktre at meanie.us>

> Once you mentioned the in-band versus out-of-band DTMF, it made sense.
> I've been reading on it since your post, and I'm going to try a few
> things.  Wonder if there is any app for voice controlled IVR (so many
> systems seem to support it these days, it would be a nice way to
> circumvent the DTMF issue... :-) ).
>
> Any ideas on the rest of my points?   -Thanks,
>
> Scott Torr wrote:
> > On Tue, 16 Feb 2010 20:53 -0600, "Doc" <ledoktre at meanie.us> wrote:
> >
> >> 1) I am able to see a call come in, and it gets routed to the sample IVR
> >> to start.  The first thing off is that when I dial from PSTN ->
> >> Skype-In, it does not let me push any buttons.  If I launch a second
> >> skype client, and dial the skype user on FS directly, it works fine.
> >> Any ideas?
> >>
> >
> > Hi Doc,
> >
> > When you dial in from the PSTN the 'push button' events are present as
> > "in band" audio tones.
> > By default the sample IVR only works on "out of band" DTMF events.
> >
> > This is why when you call directly from another skype client the 'push
> > button' events are detected because they are passed as "out of band"
> > signaling.
> >
> >
> > Now,
> > In the dial plan you can tell FS to listen for "in band" audio DTMF
> > tones using <action application="start_dtmf" />
> >
> > However,
> > This currently does not work during a skype call for some reason?
> > http://jira.freeswitch.org/browse/MODSKYPIAX-66
> >
> >
> > A work around,
> > is to sign up for the "Skype SIP Beta" product where the 'push button'
> > events are sent to FS "out of band".
> >
> > This conversion is done at the PSTN --> Skype gateway by dedicated DTMF
> > tone detection hardware.
> >
> >
> > Skype has either made a business decision, or a technical over sight to
> > pass DTMF events 'out of band' for only addition fee products.
> >
> >
> > It has also been reported in New Zealand that even the 'in band' tones
> > where present one day and actually filtered out the next.
> > This seems extreme, but either through deliberate action or a technology
> > change this is what was reported on one blog.
> >
> > It is unclear to me if this was a technical limitation or a blunt
> > business decisions, but a audio sample showed the audio tones missing?
> >
> >
> > In any case you would not want to rely on 'In band' DTMF' tones when
> > passed through 'lossy' codecs anyway.
> >
> > Best to stick with 'out of band' signaling for reliability.
> >
> >
> > regards,
> > Scott Torr
> >
> >
> >
> >
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100218/ee00b5f2/attachment-0002.html 

From mike at jerris.com  Thu Feb 18 08:02:33 2010
From: mike at jerris.com (Michael Jerris)
Date: Thu, 18 Feb 2010 11:02:33 -0500
Subject: [Freeswitch-users] Sending message notifications with openzap
In-Reply-To: <0ae2b96926e5340a50e659891efb26cc@thom.fr.eu.org>
References: <e749d5a76cc31b4a3bb2459aaf8b78d9@thom.fr.eu.org>
	<1B676793-9D83-4C8D-A9C4-78891F8C70B5@jerris.com>
	<0ae2b96926e5340a50e659891efb26cc@thom.fr.eu.org>
Message-ID: <738F6BDF-8A79-4918-A125-C6CE9445C57B@jerris.com>

its 2 way, an event is sent out to request mwi, and another in hte other direction (that mod_voicemail sends) that triggers the notify in sip.  Looking in mod_sofia where we handle the registration request, you will see there where we send the request, and look in mod_voicemail, or the event handler in sofia for the other direction.

Mike
On Feb 18, 2010, at 4:50 AM, Fran?ois Legal wrote:

> Thanks,
> 
>  
> yes I could see that this was handled with events.
> 
>  
> Could you please give more details (that does not seem obvious to me while looking at sofia_presence.c)
> 
> When and why is the event triggered ?
> 
> What information do I get with the event ?
> 
>  
> Then I have a design question, is it mandatory to (as this is done in mod_sofia but I guess for a lot of reasons) process the event in a separate thread ?
> 
>  
> Thanks for the information (and sorry if my questions are not relevant, I don't know the event system at all).
> 
>  
> Fran?ois
> 
>  
> On Thu, 18 Feb 2010 00:13:08 -0500, Michael Jerris <mike at jerris.com> wrote:
> 
> This all uses the event system.  Take a look at how we handle the events in mod_sofia, it should be pretty easy to do the same in mod_openzap.
> Mike
> 
> On Feb 16, 2010, at 4:33 AM, Fran?ois Legal wrote:
> I was wondering whether or not a facility was available to send message waiting indicator (from mpd_voicemail) using openzap.
> 
> I know this feature is not available in openzap (I'm in the process of coding it for analog channels) but as message indication can be sent via mod_sofia, I wonder if a frame is available in FS core.
> 
>  
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100218/71ead5f9/attachment-0002.html 

From anthony.minessale at gmail.com  Thu Feb 18 08:03:14 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 18 Feb 2010 10:03:14 -0600
Subject: [Freeswitch-users] Forbidden using UDP, works with TCP/TLS
In-Reply-To: <7EA7C626-D317-4011-BF6E-A7411C765EC1@freeswitch.org>
References: <33c87fa31002101824h21e3bf7lb13b68e8d4fd36a2@mail.gmail.com>
	<33c87fa31002131336j7b05db7aof6ba6e3439aebc41@mail.gmail.com>
	<9D702362-BB5F-4C0F-9EE7-DABA3B926C36@freeswitch.org>
	<33c87fa31002141708s72cb7246h5372ecea40be615d@mail.gmail.com>
	<1888B1CA-D0A8-407C-9C44-B86E0F1925D6@freeswitch.org>
	<33c87fa31002142017v49d8e7c5nf572d086ec89282b@mail.gmail.com>
	<33c87fa31002142109l5fcabed3ydba21b865ed6cbbb@mail.gmail.com>
	<147993F8-9C9E-4D96-9AFE-5EEE4E8F93F0@freeswitch.org>
	<33c87fa31002180022g2e91c8efs65f52bb4275cb59@mail.gmail.com>
	<7EA7C626-D317-4011-BF6E-A7411C765EC1@freeswitch.org>
Message-ID: <191c3a031002180803u5d862c9blf181d90988c30415@mail.gmail.com>

everyone chip in and send him a new phone so this thread can go away!

On Thu, Feb 18, 2010 at 7:04 AM, Brian West <brian at freeswitch.org> wrote:

> I think your device is broken.
>
> /b
>
> On Feb 18, 2010, at 2:22 AM, Mark Campbell-Smith wrote:
>
> > Thanks for looking at that Brian.  If the token is not used, I assume
> > this is not the reason for FS rejecting the Registration attempt?
> >
> > Also, when is stale=true set in the WWW-Authentication?  I notice that
> > for this device, I do not see stale=true, but for all my other
> > devices, I see stale=true (at least from the logs I've taken today).
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100218/3869949c/attachment-0002.html 

From anthony.minessale at gmail.com  Thu Feb 18 08:05:18 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 18 Feb 2010 10:05:18 -0600
Subject: [Freeswitch-users] Other party does not hear voice when
	connecting with X-Lite
In-Reply-To: <a858e7691002180452n6ef91ed7k3d9b2f7246d773f1@mail.gmail.com>
References: <3e6d7b0c1002150050g3ccd23aas2e7f05f230bff4a7@mail.gmail.com>
	<a858e7691002180452n6ef91ed7k3d9b2f7246d773f1@mail.gmail.com>
Message-ID: <191c3a031002180805m512655a7s3543d3a2744453bc@mail.gmail.com>

maybe you need x-heavy aka eyebeam that has more options.
Counterpath likes to hold required features in SIP hostage for money for
some reason.


On Thu, Feb 18, 2010 at 6:52 AM, Vitalii Colosov <vetali100 at gmail.com>wrote:

> Thanks a lot for your advices and sorry for confusion, my description was
> not consistent, agree...
>
> We are testing X-Lite client from 4 different machines and using 2
> different FS servers and getting different results - sometimes everything
> works, sometimes only signalling works - no voice, and sometimes even
> signalling does not work.
> YATE client always works without any such issues...
>
> When we will come to some sort of understanding, I will try to share the
> results.
> If we will stuck, I will try to get some debug information and continue
> bothering the professionals. :)
>
>  BTW, I tried to change the codecs in X-Lite, same result.
> It is for sure related to our network configuration + X-Lite's way of
> sending SIP data, looks like...
>
> Regards,
> Vitalii
>
>
>
>
>
> 2010/2/15 Steven Brown <steve at justfone.com>
>
>> I had the same problem with XLite / Freeswitch  a while back that I never
>> fully understood,  however the problem vanished when I disabled all codecs
>> on Xlite except G711 uLaw, as I say, no idea what was going on but this
>> might be worth trying.
>>
>> Steve
>>
>>
>> Message: 1
>> Date: Sun, 14 Feb 2010 09:04:09 -0600
>> From: Anthony Minessale <anthony.minessale at gmail.com>
>> Subject: Re: [Freeswitch-users] Other party does not hear voice when
>>        connecting with X-Lite
>> To: freeswitch-users at lists.freeswitch.org
>> Message-ID:
>>        <191c3a031002140704g705bfc73rd8dd103f3d846062 at mail.gmail.com
>> >
>> Content-Type: text/plain; charset="iso-8859-1"
>>
>>
>> You need to describe this again its too confusing now.
>> List each device, freeswitch, the phones and which ip and combo of addrs
>> it
>> uses with the topology clearly stated.
>> Your attempt to simplify your explanation is actually making it harder to
>> follow.
>> Also consider a debug/sip trace as well.
>>
>> Include
>> sofia status profile default.
>>
>> Then capture a test call after entering these commands.
>>
>> console loglevel debug.
>> sofa profile internal siptrace on
>>
>> On Feb 14, 2010 4:12 AM, "Vitalii Colosov" <vetali100 at gmail.com> wrote:
>>
>> No, it is done on the different PCs...
>>
>> Sorry, when I started the topic, I have described the problem how it is
>> visible from PC of my friend.
>> Then I tried to reproduce the same on my own PC, and you are right...I was
>> not able to hear anything as well, not only both party wasn't.
>> Also, from my PC I was NOT able to hear guitar on test number "9999".
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100218/884c3eec/attachment-0002.html 

From frank at carmickle.com  Thu Feb 18 08:17:41 2010
From: frank at carmickle.com (Frank Carmickle)
Date: Thu, 18 Feb 2010 11:17:41 -0500
Subject: [Freeswitch-users] building for Lenny
In-Reply-To: <a9f4977a70b3fcf1b30413ea562ca962.squirrel@www.webcontracts.co.uk>
References: <3c5cf5261002180046vb0015e7t787599c9903aedc@mail.gmail.com>
	<33c87fa31002180056g51ef8496qa9a35746d0fe75ec@mail.gmail.com>
	<3c5cf5261002180110he0ae815uc713f77f289ac625@mail.gmail.com>
	<a9f4977a70b3fcf1b30413ea562ca962.squirrel@www.webcontracts.co.uk>
Message-ID: <20100218161741.GC4236@base.carmickle.com>

On Thu, Feb 18, Matthew Law wrote:
> 
> On Thu, February 18, 2010 9:10 am, Brian May wrote:
> > On 18 February 2010 19:56, Mark Campbell-Smith <mcampbellsmith at gmail.com>
> > wrote:
> >> I think you need to download the gzip file from
> >> http://files.freeswitch.org/
> >>
> >> latest.freeswitch.org does not contain sound files as far as I'm aware
> >> ..
> >
> > Does this mean I shouldn't be trying to use the debian packages? *.deb
> > files would certainly make it easier to compile it on a fast computer
> > and then transfer to my net5501.
> >
> > Unfortunately, the website seems have a lot of old information,
> > including references to obsolete Ubuntu Hardy packages.

Yes, the wiki is quite out of date.  There are a few of us working on getting an apt repository for freeswitch packages.  Sorry it's taking so long.  You can build the debs from what's in tree now.  The only bit is that sounds are not included.  You will have to get them from files.freeswitch.org.  Once we have an apt repo all of this will become much less painless..

> >
> > I see a thread from late last year that suggests there should be
> > prebuilt packages, everything I can find so far seems very old
> > however.
> 
> Brian,
> 
> I had similar problems recently and decided to compile it from svn trunk. 
> The dependencies for the build are all available in apt, so I installed
> those, checked out trunk and configured it with a prefix of
> /usr/local/freeswitch.

Like I said you can and should build debs from svn.  As far as I see it there is no reason to not build debs.

> 
> I got the impression it is self contained, so you should be able to tar up
> the entire /usr/local/freeswitch dir and scp it over - I stand to be
> corrected on that, though.

That's true but then you lose the convenience of doing upgrades with the package management.

--FC



From infos at madovsky.org  Thu Feb 18 08:36:48 2010
From: infos at madovsky.org (Madovsky)
Date: Thu, 18 Feb 2010 11:36:48 -0500
Subject: [Freeswitch-users] register sipphone to an outside network
	registrar within freeswitch
References: <33c87fa31002101824h21e3bf7lb13b68e8d4fd36a2@mail.gmail.com><33c87fa31002130343k7a01e9abt9c33360399f246f8@mail.gmail.com><C37AB6F3-F712-41B6-9D6D-7A8AE3C6CABB@freeswitch.org><33c87fa31002131336j7b05db7aof6ba6e3439aebc41@mail.gmail.com><9D702362-BB5F-4C0F-9EE7-DABA3B926C36@freeswitch.org><33c87fa31002141708s72cb7246h5372ecea40be615d@mail.gmail.com><1888B1CA-D0A8-407C-9C44-B86E0F1925D6@freeswitch.org><33c87fa31002142017v49d8e7c5nf572d086ec89282b@mail.gmail.com><33c87fa31002142109l5fcabed3ydba21b865ed6cbbb@mail.gmail.com><147993F8-9C9E-4D96-9AFE-5EEE4E8F93F0@freeswitch.org><33c87fa31002180022g2e91c8efs65f52bb4275cb59@mail.gmail.com>
	<7EA7C626-D317-4011-BF6E-A7411C765EC1@freeswitch.org>
Message-ID: <55649042222E4C6481C086FC4D9B2AF0@MOBILEE1705>

Hi,

I googled hours without real success.
Is it possible to use a softphone in a local network and login
into a registrar outside it within freeswitch as proxy and call
and receive calls as normal ?
I'd like to use this config because I need to check codec transcoding.

Thank you

Franck



From brian at freeswitch.org  Thu Feb 18 08:44:16 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 18 Feb 2010 10:44:16 -0600
Subject: [Freeswitch-users] register sipphone to an outside network
	registrar within freeswitch
In-Reply-To: <55649042222E4C6481C086FC4D9B2AF0@MOBILEE1705>
References: <33c87fa31002101824h21e3bf7lb13b68e8d4fd36a2@mail.gmail.com><33c87fa31002130343k7a01e9abt9c33360399f246f8@mail.gmail.com><C37AB6F3-F712-41B6-9D6D-7A8AE3C6CABB@freeswitch.org><33c87fa31002131336j7b05db7aof6ba6e3439aebc41@mail.gmail.com><9D702362-BB5F-4C0F-9EE7-DABA3B926C36@freeswitch.org><33c87fa31002141708s72cb7246h5372ecea40be615d@mail.gmail.com><1888B1CA-D0A8-407C-9C44-B86E0F1925D6@freeswitch.org><33c87fa31002142017v49d8e7c5nf572d086ec89282b@mail.gmail.com><33c87fa31002142109l5fcabed3ydba21b865ed6cbbb@mail.gmail.com><147993F8-9C9E-4D96-9AFE-5EEE4E8F93F0@freeswitch.org><33c87fa31002180022g2e91c8efs65f52bb4275cb59@mail.gmail.com>
	<7EA7C626-D317-4011-BF6E-A7411C765EC1@freeswitch.org>
	<55649042222E4C6481C086FC4D9B2AF0@MOBILEE1705>
Message-ID: <7D69E8BC-3875-48AA-8CB5-70B7C207D845@freeswitch.org>

I have to say this first.... Please DO NOT hijack threads.  Click "new message", type the address and then your message and then press send.  What you did is click reply... change the subject and then replace the body. That is how you hijack a thread.

(the archive also lists them hijacked too.)

Yes its possible.  In fact the default config works like this.

/b

On Feb 18, 2010, at 10:36 AM, Madovsky wrote:

> Hi,
> 
> I googled hours without real success.
> Is it possible to use a softphone in a local network and login
> into a registrar outside it within freeswitch as proxy and call
> and receive calls as normal ?
> I'd like to use this config because I need to check codec transcoding.
> 
> Thank you
> 
> Franck




From infos at madovsky.org  Thu Feb 18 08:55:30 2010
From: infos at madovsky.org (Madovsky)
Date: Thu, 18 Feb 2010 11:55:30 -0500
Subject: [Freeswitch-users] register sipphone to an outside
	networkregistrar within freeswitch
References: <33c87fa31002101824h21e3bf7lb13b68e8d4fd36a2@mail.gmail.com><33c87fa31002130343k7a01e9abt9c33360399f246f8@mail.gmail.com><C37AB6F3-F712-41B6-9D6D-7A8AE3C6CABB@freeswitch.org><33c87fa31002131336j7b05db7aof6ba6e3439aebc41@mail.gmail.com><9D702362-BB5F-4C0F-9EE7-DABA3B926C36@freeswitch.org><33c87fa31002141708s72cb7246h5372ecea40be615d@mail.gmail.com><1888B1CA-D0A8-407C-9C44-B86E0F1925D6@freeswitch.org><33c87fa31002142017v49d8e7c5nf572d086ec89282b@mail.gmail.com><33c87fa31002142109l5fcabed3ydba21b865ed6cbbb@mail.gmail.com><147993F8-9C9E-4D96-9AFE-5EEE4E8F93F0@freeswitch.org><33c87fa31002180022g2e91c8efs65f52bb4275cb59@mail.gmail.com><7EA7C626-D317-4011-BF6E-A7411C765EC1@freeswitch.org><55649042222E4C6481C086FC4D9B2AF0@MOBILEE1705>
	<7D69E8BC-3875-48AA-8CB5-70B7C207D845@freeswitch.org>
Message-ID: <74FC05481EED41E2BB1737C8E2D931EC@MOBILEE1705>


----- Original Message ----- 
From: "Brian West" <brian at freeswitch.org>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Thursday, February 18, 2010 11:44 AM
Subject: Re: [Freeswitch-users] register sipphone to an outside 
networkregistrar within freeswitch


>I have to say this first.... Please DO NOT hijack threads.  Click "new 
>message", type the address and then your message and then press send.  What 
>you did is click reply... change the subject and then replace the body. 
>That is how you hijack a thread.
>
> (the archive also lists them hijacked too.)
>
> Yes its possible.  In fact the default config works like this.
>
> /b
>
> On Feb 18, 2010, at 10:36 AM, Madovsky wrote:
>
>> Hi,
>>
>> I googled hours without real success.
>> Is it possible to use a softphone in a local network and login
>> into a registrar outside it within freeswitch as proxy and call
>> and receive calls as normal ?
>> I'd like to use this config because I need to check codec transcoding.
>>
>> Thank you
>>
>> Franck
>
>

I know sorry I forgot to remove RE:





From brian at freeswitch.org  Thu Feb 18 09:01:02 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 18 Feb 2010 11:01:02 -0600
Subject: [Freeswitch-users] register sipphone to an outside
	networkregistrar within freeswitch
In-Reply-To: <74FC05481EED41E2BB1737C8E2D931EC@MOBILEE1705>
References: <33c87fa31002101824h21e3bf7lb13b68e8d4fd36a2@mail.gmail.com><33c87fa31002130343k7a01e9abt9c33360399f246f8@mail.gmail.com><C37AB6F3-F712-41B6-9D6D-7A8AE3C6CABB@freeswitch.org><33c87fa31002131336j7b05db7aof6ba6e3439aebc41@mail.gmail.com><9D702362-BB5F-4C0F-9EE7-DABA3B926C36@freeswitch.org><33c87fa31002141708s72cb7246h5372ecea40be615d@mail.gmail.com><1888B1CA-D0A8-407C-9C44-B86E0F1925D6@freeswitch.org><33c87fa31002142017v49d8e7c5nf572d086ec89282b@mail.gmail.com><33c87fa31002142109l5fcabed3ydba21b865ed6cbbb@mail.gmail.com><147993F8-9C9E-4D96-9AFE-5EEE4E8F93F0@freeswitch.org><33c87fa31002180022g2e91c8efs65f52bb4275cb59@mail.gmail.com><7EA7C626-D317-4011-BF6E-A7411C765EC1@freeswitch.org><55649042222E4C6481C086FC4D9B2AF0@MOBILEE1705>
	<7D69E8BC-3875-48AA-8CB5-70B7C207D845@freeswitch.org>
	<74FC05481EED41E2BB1737C8E2D931EC@MOBILEE1705>
Message-ID: <D4F504FD-DCBA-402C-8A25-72595CB4459D@freeswitch.org>

No that still hijacks the thread :P  YOU MUST click new message.  Because their are headers that are reflected back to the list server.

/b

On Feb 18, 2010, at 10:55 AM, Madovsky wrote:

> I know sorry I forgot to remove RE:




From brian at freeswitch.org  Thu Feb 18 09:23:42 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 18 Feb 2010 11:23:42 -0600
Subject: [Freeswitch-users] Testers
Message-ID: <8BDBC6EB-00D4-48B7-9234-1067E01DEE3D@freeswitch.org>

Here is what I need:

Testers for our G729 installer for linux 32bit and 64bit installs.

I have tested CentOS 5x, Debian (excluding 4.0 unsupported)

Please contact me off list so I can issue you a temp. license and a make sure it all works.

Thanks,
Brian




From infos at madovsky.org  Thu Feb 18 09:34:19 2010
From: infos at madovsky.org (Madovsky)
Date: Thu, 18 Feb 2010 12:34:19 -0500
Subject: [Freeswitch-users] freeswitch config expert wanted
Message-ID: <3E855EBF921248F28F231F1CDCF9B8E0@MOBILEE1705>

Hi again,

as I'm a researcher and work on a (personal) project since 5 years now,
I decided to add a "light" VOIP functionality to my project, but
after 3 weeks of learning, I decided to stop protocols headdick and waste of my time
(as I never wanted to be in the VOIP dev world), it's definetly not my cup of tea.
So, if anybody in this emailist wants to help me (I pay of course) to configure freeeswitch
as I'm expecting for (I'm sure my config request is really easy but I have not the right informations and example until now)
it would be great, and maybe I will take my retirement more early... ;)

Thanks

Franck Chionna
infos

at
madovsky
dot


org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100218/ffd485b4/attachment-0002.html 

From devel at thom.fr.eu.org  Thu Feb 18 09:36:31 2010
From: devel at thom.fr.eu.org (=?UTF-8?Q?Fran=C3=A7ois_Legal?=)
Date: Thu, 18 Feb 2010 18:36:31 +0100
Subject: [Freeswitch-users] Sending message notifications with openzap
In-Reply-To: <738F6BDF-8A79-4918-A125-C6CE9445C57B@jerris.com>
References: <e749d5a76cc31b4a3bb2459aaf8b78d9@thom.fr.eu.org>
	<1B676793-9D83-4C8D-A9C4-78891F8C70B5@jerris.com>
	<0ae2b96926e5340a50e659891efb26cc@thom.fr.eu.org>
	<738F6BDF-8A79-4918-A125-C6CE9445C57B@jerris.com>
Message-ID: <68be1f4017938f41a48f7a619a3e667b@thom.fr.eu.org>


I not sure to understand this process, so please correct me if I'm
wrong.


A module, to receive the event notification, does not only have to
bind
to the event (with switch_event_bind or switch_event_bind_removable)
but
also have to send the event (using switch_event_create
and
switch_event_fire) for registration ? 

What is the
variable
mod_sofia_globals.mwi_node used for ? Is it used to queue the
events until
they are processed ? If yes, is it mandatory to queue the
events in openzap
? 

As a side question, I was wondering how to associate
a mailbox (and/or
a user) with an openzap channel. I guess I need that kind
of association to
bring MWI to openzap. I thought I could use the
MWI-Account (something like
MWI-Account = openzap/x/y). Would it be a
correct way to go ? 

Thanks

Fran?ois 

On Thu, 18 Feb 2010 11:02:33
-0500, Michael Jerris  wrote: its 2 way, an event is sent out to request
mwi, and another in hte other direction (that mod_voicemail sends) that
triggers the notify in sip. Looking in mod_sofia where we handle the
registration request, you will see there where we send the request, and
look in mod_voicemail, or the event handler in sofia for the other
direction.  Mike
  On Feb 18, 2010, at 4:50 AM, Fran?ois Legal wrote:


Thanks,  

yes I could see that this was handled with events.  

Could
you please give more details (that does not seem obvious to me while
looking at sofia_presence.c) 

When and why is the event triggered ? 

What
information do I get with the event ?  

Then I have a design question, is
it mandatory to (as this is done in mod_sofia but I guess for a lot of
reasons) process the event in a separate thread ?  

Thanks for the
information (and sorry if my questions are not relevant, I don't know the
event system at all).  

Fran?ois  

On Thu, 18 Feb 2010 00:13:08 -0500,
Michael Jerris  wrote: This all uses the event system. Take a look at how
we handle the events in mod_sofia, it should be pretty easy to do the same
in mod_openzap. Mike 
  On Feb 16, 2010, at 4:33 AM, Fran?ois Legal wrote: 


I was wondering whether or not a facility was available to send message
waiting indicator (from mpd_voicemail) using openzap. 

I know this feature
is not available in openzap (I'm in the process of coding it for analog
channels) but as message indication can be sent via mod_sofia, I wonder if
a frame is available in FS core.      

_______________________________________________
FreeSWITCH-users mailing
list
FreeSWITCH-users at lists.freeswitch.org
[2]
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




Links:
------
[1] mailto:mike at jerris.com
[2]
mailto:FreeSWITCH-users at lists.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100218/651a60fd/attachment-0002.html 

From msc at freeswitch.org  Thu Feb 18 09:43:37 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 18 Feb 2010 09:43:37 -0800
Subject: [Freeswitch-users] tone_detect timeout
In-Reply-To: <EC840323-2A76-426E-A386-8DA111195768@tlainvestments.com>
References: <EC840323-2A76-426E-A386-8DA111195768@tlainvestments.com>
Message-ID: <87f2f3b91002180943u5ef06108rcec1f8d4665783fe@mail.gmail.com>

On Wed, Feb 17, 2010 at 4:44 PM, Troy Anderson <troy at tlainvestments.com>wrote:

> In the wiki about tone_detect, the docs state that the timeout is in
> seconds (e.g. +2 for 2 seconds from now), but all the examples have +5000,
> suggesting that it may really be milliseconds.  I'd be happy to update the
> wiki if someone could say if it's seconds or millisconds or otherwise.
>
> I tried tracing it back in the code, but got lost looking for the
> definition of switch_media_bug_t!
>
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_tone_detect
>
> Thanks!
> Troy
>

Troy,

Good catch. It is definitely milliseconds. Please update the wiki. You get a
gold star for taking the initiative. BTW, it's okay to make the change if
you're reasonably certain it's correct and then update us here. It's easy to
undo a wiki edit.

-MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100218/b27b5cfa/attachment-0002.html 

From frank at carmickle.com  Thu Feb 18 09:49:05 2010
From: frank at carmickle.com (Frank Carmickle)
Date: Thu, 18 Feb 2010 12:49:05 -0500
Subject: [Freeswitch-users] freeswitch config expert wanted
In-Reply-To: <3E855EBF921248F28F231F1CDCF9B8E0@MOBILEE1705>
References: <3E855EBF921248F28F231F1CDCF9B8E0@MOBILEE1705>
Message-ID: <20100218174905.GD4236@base.carmickle.com>

On Thu, Feb 18, Madovsky wrote:
> Hi again,
> 
> as I'm a researcher and work on a (personal) project since 5 years now,
> I decided to add a "light" VOIP functionality to my project, but
> after 3 weeks of learning, I decided to stop protocols headdick and waste of my time
> (as I never wanted to be in the VOIP dev world), it's definetly not my cup of tea.
> So, if anybody in this emailist wants to help me (I pay of course) to configure freeeswitch
> as I'm expecting for (I'm sure my config request is really easy but I have not the right informations and example until now)
> it would be great, and maybe I will take my retirement more early... ;)

What is it that you require?  If it would help you to speak on the phone my number is +1 (315) 703-1608.  Feel free to call at any time.  If I do not answer please leave a message.

Regards
--Frank



From msc at freeswitch.org  Thu Feb 18 09:58:15 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 18 Feb 2010 09:58:15 -0800
Subject: [Freeswitch-users] freeswitch config expert wanted
In-Reply-To: <20100218174905.GD4236@base.carmickle.com>
References: <3E855EBF921248F28F231F1CDCF9B8E0@MOBILEE1705>
	<20100218174905.GD4236@base.carmickle.com>
Message-ID: <87f2f3b91002180958w4a3c71ddt789052f45ad95af2@mail.gmail.com>

On Thu, Feb 18, 2010 at 9:49 AM, Frank Carmickle <frank at carmickle.com>wrote:

> On Thu, Feb 18, Madovsky wrote:
> > Hi again,
> >
> > as I'm a researcher and work on a (personal) project since 5 years now,
> > I decided to add a "light" VOIP functionality to my project, but
> > after 3 weeks of learning, I decided to stop protocols headdick and waste
> of my time
> > (as I never wanted to be in the VOIP dev world), it's definetly not my
> cup of tea.
> > So, if anybody in this emailist wants to help me (I pay of course) to
> configure freeeswitch
> > as I'm expecting for (I'm sure my config request is really easy but I
> have not the right informations and example until now)
> > it would be great, and maybe I will take my retirement more early... ;)
>
> What is it that you require?  If it would help you to speak on the phone my
> number is +1 (315) 703-1608.  Feel free to call at any time.  If I do not
> answer please leave a message.
>
> Regards
> --Frank
>

Additionally, we have a community conference call tomorrow, so if you would
like to join us and talk about your project that would be okay. We have an
official agenda that we discuss:

http://wiki.freeswitch.org/wiki/FS_weekly_2010_02_19

And after the agenda people take turns asking questions and talking about
subjects of interest.
-MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100218/1809e2f7/attachment-0002.html 

From troy at tlainvestments.com  Thu Feb 18 10:07:11 2010
From: troy at tlainvestments.com (Troy Anderson)
Date: Thu, 18 Feb 2010 11:07:11 -0700
Subject: [Freeswitch-users] tone_detect timeout
In-Reply-To: <87f2f3b91002180943u5ef06108rcec1f8d4665783fe@mail.gmail.com>
References: <EC840323-2A76-426E-A386-8DA111195768@tlainvestments.com>
	<87f2f3b91002180943u5ef06108rcec1f8d4665783fe@mail.gmail.com>
Message-ID: <758D6E07-90B3-4117-97AD-3CA899B4CAE5@tlainvestments.com>

Changed to indicate milliseconds for relative and seconds for the absolute option as I assume the absolute option is for a unix timestamp which is, indeed, seconds.

-Troy


On Feb 18, 2010, at 10:43 AM, Michael Collins wrote:

> 
> 
> On Wed, Feb 17, 2010 at 4:44 PM, Troy Anderson <troy at tlainvestments.com> wrote:
> In the wiki about tone_detect, the docs state that the timeout is in seconds (e.g. +2 for 2 seconds from now), but all the examples have +5000, suggesting that it may really be milliseconds.  I'd be happy to update the wiki if someone could say if it's seconds or millisconds or otherwise.
> 
> I tried tracing it back in the code, but got lost looking for the definition of switch_media_bug_t!
> 
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_tone_detect
> 
> Thanks!
> Troy
> 
> Troy,
> 
> Good catch. It is definitely milliseconds. Please update the wiki. You get a gold star for taking the initiative. BTW, it's okay to make the change if you're reasonably certain it's correct and then update us here. It's easy to undo a wiki edit.
> 
> -MC
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100218/60395ef5/attachment-0002.html 

From msc at freeswitch.org  Thu Feb 18 10:13:00 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 18 Feb 2010 10:13:00 -0800
Subject: [Freeswitch-users] tone_detect timeout
In-Reply-To: <758D6E07-90B3-4117-97AD-3CA899B4CAE5@tlainvestments.com>
References: <EC840323-2A76-426E-A386-8DA111195768@tlainvestments.com>
	<87f2f3b91002180943u5ef06108rcec1f8d4665783fe@mail.gmail.com>
	<758D6E07-90B3-4117-97AD-3CA899B4CAE5@tlainvestments.com>
Message-ID: <87f2f3b91002181013w7bec493cm37946405ec46ee5f@mail.gmail.com>

On Thu, Feb 18, 2010 at 10:07 AM, Troy Anderson <troy at tlainvestments.com>wrote:

> Changed to indicate milliseconds for relative and seconds for the absolute
> option as I assume the absolute option is for a unix timestamp which is,
> indeed, seconds.
>
> -Troy
>
> Another gold star 4 U! Thanks for pitching in. We really appreciate it when
folks lend a hand documenting the stuff they know. Keep up the good work.
-MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100218/fc015664/attachment-0002.html 

From max.bridgewater at gmail.com  Thu Feb 18 10:26:37 2010
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Thu, 18 Feb 2010 10:26:37 -0800
Subject: [Freeswitch-users] Skypiax snd-dummy, One way audio
Message-ID: <c55d46951002181026n49b1dbcl8cfa868f902a8a0d@mail.gmail.com>

Hi Skypiax Lovers,

I'm trying to get Skypiax running on CentOS5.3 but I keep having
problems with snd-dummy. My problem is that I have sound only in one
direction (inbound). I suspect the sound devices are not set properly.
In particular, I can't find "hw:dummy"  in the Skype options (Static
Build 2.1).

Here are the sound devices I see in Skype options:
a) Dummy, Dummy, PCM Default Audio 0default:CARD=Dummy)
b) HDA Intel, ALC662 Analog (hw:0,0)
c) Dummy, Dummy PCM (hw:1,0)
d) hdmi (unlnown)

Here is what i notice:

1) modprobe snd-dummy returns nothing; implying to me that the module
was loaded properly.
2) lsmod | grep "snd" returns:

snd_dummy              15553  0
snd_hda_intel         343537  0
snd_hwdep              12869  1 snd_hda_intel
snd_seq_dummy           7877  0
snd_seq_oss            32577  0
snd_seq_midi_event     11073  1 snd_seq_oss
snd_seq                49585  5 snd_seq_dummy,snd_seq_oss,snd_seq_midi_event
snd_seq_device         11725  3 snd_seq_dummy,snd_seq_oss,snd_seq
snd_pcm_oss            42817  0
snd_mixer_oss          19009  1 snd_pcm_oss
snd_pcm                72133  3 snd_dummy,snd_hda_intel,snd_pcm_oss
snd_timer              24517  2 snd_seq,snd_pcm
snd                    55237  10
snd_dummy,snd_hda_intel,snd_hwdep,snd_seq_oss,snd_seq,snd_seq_device,snd_pcm_oss,snd_mixer_oss,snd_pcm,snd_timer
soundcore              11553  1 snd
snd_page_alloc         14281  2 snd_hda_intel,snd_pcm

Any idea?
Thanks,
Max.



From gmaruzz at celliax.org  Thu Feb 18 10:52:02 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Thu, 18 Feb 2010 19:52:02 +0100
Subject: [Freeswitch-users] Skypiax snd-dummy, One way audio
In-Reply-To: <c55d46951002181026n49b1dbcl8cfa868f902a8a0d@mail.gmail.com>
References: <c55d46951002181026n49b1dbcl8cfa868f902a8a0d@mail.gmail.com>
Message-ID: <7b197bef1002181052x37a01b75q17624efb6fc5fca4@mail.gmail.com>

Hi Max,

on centos 5.3 is working well, but it will consume much cpu time. If
you are planning to use few skype channels, then ok.
If you want to "scale", you must use a kernel at 100HZ, there is a
.config file in the mod_skypiax sources in the subdirectory kernel,
and instructions in the wiki page.

You can keep your same centos running, just with the new compiled kernel.

That said, for few channels will be ok.

You have to use the last static build of beta skype (.2.1.0.81).

In your case you will use the:
c) Dummy, Dummy PCM (hw:1,0)
device.

check with dmesg if the module is running properly, it will add a line
were it tells on how many HZ is running. If there is not that line,
the custom snd-dummy is not loaded.

Also, you can test with
aplay -l
if it show you a soundcard with 127 subdevices, you're ok.

Please svn update mod_skypiax directory 'cause last change was few
minutes ago ;).

Let know if you still encounter problems, you can find me in IRC
(irc.freenode.net #freeswitch) as gmaruzz, if you need,

ciao for now,

-giovanni


On Thu, Feb 18, 2010 at 7:26 PM, Max Bridgewater
<max.bridgewater at gmail.com> wrote:
> Hi Skypiax Lovers,
>
> I'm trying to get Skypiax running on CentOS5.3 but I keep having
> problems with snd-dummy. My problem is that I have sound only in one
> direction (inbound). I suspect the sound devices are not set properly.
> In particular, I can't find "hw:dummy" ?in the Skype options (Static
> Build 2.1).
>
> Here are the sound devices I see in Skype options:
> a) Dummy, Dummy, PCM Default Audio 0default:CARD=Dummy)
> b) HDA Intel, ALC662 Analog (hw:0,0)
> c) Dummy, Dummy PCM (hw:1,0)
> d) hdmi (unlnown)
>
> Here is what i notice:
>
> 1) modprobe snd-dummy returns nothing; implying to me that the module
> was loaded properly.
> 2) lsmod | grep "snd" returns:
>
> snd_dummy ? ? ? ? ? ? ?15553 ?0
> snd_hda_intel ? ? ? ? 343537 ?0
> snd_hwdep ? ? ? ? ? ? ?12869 ?1 snd_hda_intel
> snd_seq_dummy ? ? ? ? ? 7877 ?0
> snd_seq_oss ? ? ? ? ? ?32577 ?0
> snd_seq_midi_event ? ? 11073 ?1 snd_seq_oss
> snd_seq ? ? ? ? ? ? ? ?49585 ?5 snd_seq_dummy,snd_seq_oss,snd_seq_midi_event
> snd_seq_device ? ? ? ? 11725 ?3 snd_seq_dummy,snd_seq_oss,snd_seq
> snd_pcm_oss ? ? ? ? ? ?42817 ?0
> snd_mixer_oss ? ? ? ? ?19009 ?1 snd_pcm_oss
> snd_pcm ? ? ? ? ? ? ? ?72133 ?3 snd_dummy,snd_hda_intel,snd_pcm_oss
> snd_timer ? ? ? ? ? ? ?24517 ?2 snd_seq,snd_pcm
> snd ? ? ? ? ? ? ? ? ? ?55237 ?10
> snd_dummy,snd_hda_intel,snd_hwdep,snd_seq_oss,snd_seq,snd_seq_device,snd_pcm_oss,snd_mixer_oss,snd_pcm,snd_timer
> soundcore ? ? ? ? ? ? ?11553 ?1 snd
> snd_page_alloc ? ? ? ? 14281 ?2 snd_hda_intel,snd_pcm
>
> Any idea?
> Thanks,
> Max.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618



From andrew at hijacked.us  Thu Feb 18 11:01:29 2010
From: andrew at hijacked.us (Andrew Thompson)
Date: Thu, 18 Feb 2010 14:01:29 -0500
Subject: [Freeswitch-users] how to use mod_erlang ?
In-Reply-To: <ecd05ce31002171917w2713f8detcee3f7e36039e9f2@mail.gmail.com>
References: <ecd05ce31002171917w2713f8detcee3f7e36039e9f2@mail.gmail.com>
Message-ID: <20100218190129.GD8518@hijacked.us>

On Thu, Feb 18, 2010 at 11:17:26AM +0800, daqiang wang wrote:
> hello !
> I test mod_erlang from : http://wiki.freeswitch.org/wiki/Mod_erlang_event
> but when I input :
>  >
> 
> {foo, freeswitch at localhost } ! {api, status, ""}.
> I received
>   (test at wangdq-laptop)2>
> =ERROR REPORT==== 18-Feb-2010::11:10:13 ===
> Error in process <0.41.0> on node 'test at wangdq-laptop' with exit
> value: {badarg,[{erlang,list_to_existing_atom,["freeswitch at wangdq-laptop"]},{dist_util,recv_challenge,1},{dist_util,handshake_we_started,1}]}
>

Try {foo, 'freeswitch at wangdq-laptop' } ! {api, status, ""}. instead.

Andrew



From Prometheus001 at gmx.net  Thu Feb 18 11:27:16 2010
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Thu, 18 Feb 2010 20:27:16 +0100
Subject: [Freeswitch-users] more than 1 profile for sofia/user ?
Message-ID: <4B7D9494.8050208@gmx.net>

Hello,

in the standard setup - if a phone is registering to port 5060 - it is
bound to the "internal" profile. And I can dial it via sofia/user/xxxx then.

However due to NAT issues I would like to have to 2 seperate profiles
for SIP phones. For example I have a "local" profile for all devices
inside the LAN (e.g. Pattons und in future: local phones) and another
"internal" profile which allows also external phones via
external-xxx-ip. That way I would like to ensure that local phones have
nothing to do with natted adresses and that external phones can register
via external IPs.

Question How do I manage that I can register a phone to the "local"
profile and being able to dial that phone via sofia/user/xxxxx?

Or do I think too complicated and there is simply nothing special to do?

Best regards
Peter



From anthony.minessale at gmail.com  Thu Feb 18 11:40:28 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 18 Feb 2010 13:40:28 -0600
Subject: [Freeswitch-users] more than 1 profile for sofia/user ?
In-Reply-To: <4B7D9494.8050208@gmx.net>
References: <4B7D9494.8050208@gmx.net>
Message-ID: <191c3a031002181140y44135106yb34e6a6f73a04773@mail.gmail.com>

edit the dial-string for that user in the directory xml to try the extension
on both profile at once

On Thu, Feb 18, 2010 at 1:27 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:

> Hello,
>
> in the standard setup - if a phone is registering to port 5060 - it is
> bound to the "internal" profile. And I can dial it via sofia/user/xxxx
> then.
>
> However due to NAT issues I would like to have to 2 seperate profiles
> for SIP phones. For example I have a "local" profile for all devices
> inside the LAN (e.g. Pattons und in future: local phones) and another
> "internal" profile which allows also external phones via
> external-xxx-ip. That way I would like to ensure that local phones have
> nothing to do with natted adresses and that external phones can register
> via external IPs.
>
> Question How do I manage that I can register a phone to the "local"
> profile and being able to dial that phone via sofia/user/xxxxx?
>
> Or do I think too complicated and there is simply nothing special to do?
>
> Best regards
> Peter
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100218/a7b1f7d9/attachment-0002.html 

From jerry.richards at teotech.com  Thu Feb 18 12:26:28 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Thu, 18 Feb 2010 12:26:28 -0800
Subject: [Freeswitch-users] Presence PUBLISH Not Updating
	AfterSoftphoneOffLine Then Available
In-Reply-To: <191c3a031002162139p57c94046y660fb8d01fd76f46@mail.gmail.com>
References: <31FEE9072F104AE18B26812978FCA341@greyhawk.tonecommander.com><45963EB5-DFE5-43FA-AFCC-857E1E719E2D@jerris.com><191c3a031002091557jfc9b486lb832ea538883bce9@mail.gmail.com><68A4C011A72B4ABA91A185F32480EB99@greyhawk.tonecommander.com>
	<191c3a031002162139p57c94046y660fb8d01fd76f46@mail.gmail.com>
Message-ID: <E69C19EDE6F541198E27BE0883CB773D@greyhawk.tonecommander.com>

Okay, you have made some good suggestions.  I will look into this further on
my end (I think I might know the cause).  If I find it to be an FS bug, I
will open a Jira Issue.  Yes, these are Bria (CounterPath) phones, but these
are phones that I'm using and they are popular, and as far as I know,
faithful to the SIP RFCs, so I think it will make FS more robust.  I haven't
use the IRC much in the past, but I can try to login there when I have
issues in the future.
 
Sorry for my slow response.  I have a lot to do at the moment and sometimes
I must do some SIP research.  By the way, I think Freeswitch is a great
design and you all are doing a great job with this project.
 
Thanks And Best Regards,
Jerry
 


  _____  

From: Anthony Minessale [mailto:anthony.minessale at gmail.com] 
Sent: Tuesday, February 16, 2010 9:40 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Presence PUBLISH Not Updating
AfterSoftphoneOffLine Then Available



You see one case at the top where it sends a notify and more where it doesnt
.  You have the sql stmts right there (they are not errs just logging in red
so they are obvious) run them manually and figure out why there are no
matches.  No subscriptions maybe?

Its beginning to sound like a broken record with so many bria isssues, its a
new software afterall and not free like we are, why must we support it so
much?

Also if you are actually concerned with this issue, maybe you can come back
sooner than once every week or 2 weeks.  We quickly lose track of threads
like this that linger for a month, that's what jira is for....

Maybe you can stop by irc or keep an eye on your email client so we can
confirm what you are doing wrong or if we have an interop with bria, a pay
softphone none of us have a copy of........




On Feb 16, 2010 11:18 AM, "Jerry Richards" <jerry.richards at teotech.com>
wrote:


I got version freeswitch-1.0.5-20100215-0400, built it, and ran it, and I am
seeing the same issue.  That is, once I set the Bria softphone status to
'Appear Offline', FS does not forward presence states until resubscription
time (i.e. tens of minutes later).  I posted a trace at
http://pastebin.freeswitch.org/12164.  At line 359 of the trace, FS is
logging an ERR at sofia_presence.c:662.  Here is the scenario:
 
1) Set Bria softphone presence state to 'Appear Offline'
2) Subscibing softphones reflect offline status

3) Set Bria softphone presence state to 'Available'
4) *** Subscibing softphones do not get status update ***
 

Thanks And Best Regards,
Jerry
 


  _____  

From: Anthony Minessale [mailto:anthony.minessale at gmail.com] 
Sent: Tuesday, February 09, 2010 3:58 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Presence PUBLISH Not Updating AfterSoftphone
OffLine Then Available



> he means update to trunk first then try it again obviously.
>
>
> On Tue, Feb 9, 2010 at 3:10 PM, ...


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org





-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100218/ee19c628/attachment-0002.html 

From mrene_lists at avgs.ca  Thu Feb 18 13:10:53 2010
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Thu, 18 Feb 2010 16:10:53 -0500
Subject: [Freeswitch-users] Establishing a Call from .Net based
	application
In-Reply-To: <6E8D2069C08AA84A83D336E996AE4C67032C9D5BAD@mse17be1.mse17.exchange.ms>
References: <874941.17255.qm@web33502.mail.mud.yahoo.com>
	<df97b9c51001310545n54195cadr9faa3db46917ca9e@mail.gmail.com>
	<367751821002010737o4ca83476j62c96cb60c8a456a@mail.gmail.com>
	<df97b9c51002130752h48aae119ja80797ab8a3f5b2d@mail.gmail.com>
	<6E8D2069C08AA84A83D336E996AE4C67032C9D5346@mse17be1.mse17.exchange.ms>
	<df97b9c51002180012wc4d9421ye6e81ae6199321c8@mail.gmail.com>
	<6E8D2069C08AA84A83D336E996AE4C67032C9D5BAD@mse17be1.mse17.exchange.ms>
Message-ID: <9ABD2E73-F7EB-4B2E-9306-8BBC6718F215@avgs.ca>

parse "sofia xmlstatus"

<profile>
<name>external</name>
<type>profile</type>
<data>sip:mod_sofia at 192.168.0.9:5080</data>
<state>RUNNING (0)</state>
</profile>


Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 18-Feb-10, at 6:12 AM, Michael Giagnocavo wrote:

> I?m not sure what the FreeSWITCH APIs are to figure out what IP  
> Sofia SIP has bound to. Whatever it is, you?d call the same thing in  
> C#. What do you want to do with the API?
>
> mod_managed.dll or .so is the FreeSWITCH native code module that  
> loads the CLR or Mono into the FreeSWITCH process and loads  
> FreeSWITCH.Managed.dll. The managed DLL contains the bulk of the  
> managed-unmanaged interop code (.NET definitions of all the FS C  
> functions).
>
> -Michael
>
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org 
> ] On Behalf Of Scott Fernandez
> Sent: Thursday, February 18, 2010 1:12 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Establishing a Call from .Net based  
> application
>
> Hi Diego & Michael,
>
> Thanks for your reply and support.
>
> However, I have some clarifications required from both of you.
>
> 1. Here is the question for Diego,
>
> Simple Example:
>
> using FreeSWITCH;
> using FreeSWITCH.Native;
>
> namespace BITS.Ivr.Bp.Server
> {
>  public class CIV_BPFSProcess : FreeSWITCH.IAppPlugin
> {
>  public void Run(AppContext context)
>  {
>   //answer call
>   context.Session.Answer();
>   //sleep 2 seconds
>   context.Session.sleep(2000, 1);
>   //hangup call
>   context.Session.Hangup("
> NORMAL_CLEARING");
>  }
>  }
> }
>
> I understand that the concept of your example code. However, would  
> like to know as to how would my .NET C# know the IP address of  
> Freeswitch to talk to it as there is no indication for that?. If not  
> here, where would we need to reference the IP address of FS in .NET  
> code?
>
> I guess the IP address of FS needs to be mentioned in the Target  
> section of the below web.config file in .NET. If I am right, how to  
> specify the IP address over here. If I am wrong, please let me know  
> where do we need to mention the IP address of FS.
>
>     <configuration>
>             <dllmap dll="mod_managed.dll" target="mod_managed.so"/>
>     </configuration>
>
>
> 2. Here is the question for Michael,
>
> You mentioned that "mod_managed.so will be in your freeswitch mod  
> directory". This is very clear and what is mod_managed.dll in  
> my .NET application and the purpose of it?
>
> Thanks for all your help.
>
> Regards,
> Scott.
>
>
>
> On Sun, Feb 14, 2010 at 1:15 AM, Michael Giagnocavo <mgg at giagnocavo.net 
> > wrote:
>
> 2. There is a configuration settings required to Map the "DLL" to  
> ".so" object in CentOS.
> Now, the question is which DLL and .so file to be made available and  
> where?
>
> ?
> If you are experiencing NullReferenceExceptions with any plugin run  
> through the dialplan, make sure you have included the appropriate  
> entry in your dllmap configuration:
>  <dllmap dll="mod_managed.dll" target="mod_managed.so" os="!windows"/>
> ?
>
> mod_managed.so will be in your freeswitch mod directory.
>
>
> All I need is to initiate a call from .NET application and then it  
> should talk to mod_managed module and establish a call. Secondly, I  
> need to know the status of the call such as Ringing, Active, Hangup  
> etc.
>
> To initiate a call, try ManagedSession.Originate.
> -Michael
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100218/43223e9c/attachment-0002.html 

From Prometheus001 at gmx.net  Thu Feb 18 13:14:48 2010
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Thu, 18 Feb 2010 22:14:48 +0100
Subject: [Freeswitch-users] more than 1 profile for sofia/user ?
In-Reply-To: <191c3a031002181140y44135106yb34e6a6f73a04773@mail.gmail.com>
References: <4B7D9494.8050208@gmx.net>
	<191c3a031002181140y44135106yb34e6a6f73a04773@mail.gmail.com>
Message-ID: <4B7DADC8.1060405@gmx.net>

Any idea how to do this?

currently I have
{presence_id=${dialed_user}@${dialed_domain},transfer_fallback_extension=${dialed_user}}${sofia_contact(${dialed_user}@${dialed_domain})}


Best regards
Peter

Anthony Minessale schrieb:
> edit the dial-string for that user in the directory xml to try the
> extension on both profile at once
>
> On Thu, Feb 18, 2010 at 1:27 PM, Peter P GMX <Prometheus001 at gmx.net
> <mailto:Prometheus001 at gmx.net>> wrote:
>
>     Hello,
>
>     in the standard setup - if a phone is registering to port 5060 - it is
>     bound to the "internal" profile. And I can dial it via
>     sofia/user/xxxx then.
>
>     However due to NAT issues I would like to have to 2 seperate profiles
>     for SIP phones. For example I have a "local" profile for all devices
>     inside the LAN (e.g. Pattons und in future: local phones) and another
>     "internal" profile which allows also external phones via
>     external-xxx-ip. That way I would like to ensure that local phones
>     have
>     nothing to do with natted adresses and that external phones can
>     register
>     via external IPs.
>
>     Question How do I manage that I can register a phone to the "local"
>     profile and being able to dial that phone via sofia/user/xxxxx?
>
>     Or do I think too complicated and there is simply nothing special
>     to do?
>
>     Best regards
>     Peter
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> <mailto:MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> <mailto:sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> <http://iax:guest at conference.freeswitch.org/888>
> googletalk:conf+888 at conference.freeswitch.org
> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
> ------------------------------------------------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   



From anthony.minessale at gmail.com  Thu Feb 18 13:53:28 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 18 Feb 2010 15:53:28 -0600
Subject: [Freeswitch-users] more than 1 profile for sofia/user ?
In-Reply-To: <4B7DADC8.1060405@gmx.net>
References: <4B7D9494.8050208@gmx.net>
	<191c3a031002181140y44135106yb34e6a6f73a04773@mail.gmail.com>
	<4B7DADC8.1060405@gmx.net>
Message-ID: <191c3a031002181353r2dbf15c6h206509f9b1399148@mail.gmail.com>

add on a , then another dial string to reflect the other profile too

On Thu, Feb 18, 2010 at 3:14 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:

> Any idea how to do this?
>
> currently I have
> {presence_id=${dialed_user}@
> ${dialed_domain},transfer_fallback_extension=${dialed_user}}${sofia_contact(${dialed_user}@
> ${dialed_domain})}
>
>
> Best regards
> Peter
>
> Anthony Minessale schrieb:
> > edit the dial-string for that user in the directory xml to try the
> > extension on both profile at once
> >
> > On Thu, Feb 18, 2010 at 1:27 PM, Peter P GMX <Prometheus001 at gmx.net
> > <mailto:Prometheus001 at gmx.net>> wrote:
> >
> >     Hello,
> >
> >     in the standard setup - if a phone is registering to port 5060 - it
> is
> >     bound to the "internal" profile. And I can dial it via
> >     sofia/user/xxxx then.
> >
> >     However due to NAT issues I would like to have to 2 seperate profiles
> >     for SIP phones. For example I have a "local" profile for all devices
> >     inside the LAN (e.g. Pattons und in future: local phones) and another
> >     "internal" profile which allows also external phones via
> >     external-xxx-ip. That way I would like to ensure that local phones
> >     have
> >     nothing to do with natted adresses and that external phones can
> >     register
> >     via external IPs.
> >
> >     Question How do I manage that I can register a phone to the "local"
> >     profile and being able to dial that phone via sofia/user/xxxxx?
> >
> >     Or do I think too complicated and there is simply nothing special
> >     to do?
> >
> >     Best regards
> >     Peter
> >
> >     _______________________________________________
> >     FreeSWITCH-users mailing list
> >     FreeSWITCH-users at lists.freeswitch.org
> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>
> >     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >     UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >     http://www.freeswitch.org
> >
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> > Twitter: http://twitter.com/FreeSWITCH_wire
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> > <mailto:MSN%3Aanthony_minessale at hotmail.com<MSN%253Aanthony_minessale at hotmail.com>
> >
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> > <mailto:PAYPAL%3Aanthony.minessale at gmail.com<PAYPAL%253Aanthony.minessale at gmail.com>
> >
> > IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> > <mailto:sip%3A888 at conference.freeswitch.org<sip%253A888 at conference.freeswitch.org>
> >
> > iax:guest at conference.freeswitch.org/888
> > <http://iax:guest at conference.freeswitch.org/888>
> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org<googletalk%253Aconf%252B888 at conference.freeswitch.org>
> >
> > pstn:+19193869900
> > ------------------------------------------------------------------------
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100218/4545016c/attachment-0002.html 

From lloyd.aloysius at gmail.com  Thu Feb 18 14:10:24 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Thu, 18 Feb 2010 17:10:24 -0500
Subject: [Freeswitch-users] IVR greeting - first two words missing
Message-ID: <8a19bf2e1002181410x44b40216j374a30a5675fd162@mail.gmail.com>

Hi All,

I setup a simple IVR. Here is the script.

<menu name="test_ivr"
          greet-long="test/test-ivr.wav"
          greet-short="tset/test-ivr.wav"
          invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"
          exit-sound="voicemail/vm-goodbye.wav"
          timeout ="10000"
                  inter-digit-timeout="2000"
                  max-failures="3">
 <entry action="menu-exec-app" digits="/^(20[0-9])$/" param="transfer $1 XML
default"/>
 <entry action="menu-exec-app" digits="9999" param="transfer 9999 XML
default"/>
</menu>
  </menus>
</configuration>

*Dial Plan*

<action application="answer"/>
<action application="ivr" data="test_ivr"/>

Every time when I reach the IVR . I am getting first one or two words
missing( or may be not clear). How can I fix this issue.

Thanks,
Lloyd
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100218/72232ec7/attachment-0002.html 

From Prometheus001 at gmx.net  Thu Feb 18 14:41:06 2010
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Thu, 18 Feb 2010 23:41:06 +0100
Subject: [Freeswitch-users] more than 1 profile for sofia/user ?
In-Reply-To: <191c3a031002181353r2dbf15c6h206509f9b1399148@mail.gmail.com>
References: <4B7D9494.8050208@gmx.net>	<191c3a031002181140y44135106yb34e6a6f73a04773@mail.gmail.com>	<4B7DADC8.1060405@gmx.net>
	<191c3a031002181353r2dbf15c6h206509f9b1399148@mail.gmail.com>
Message-ID: <4B7DC202.7090409@gmx.net>

Hello Anthony,

>add on a , then another dial string to reflect the other profile too
I really tried to understand this, but
can you give me an example?

Best regards
Peter

Anthony Minessale schrieb:
> add on a , then another dial string to reflect the other profile too
>
> On Thu, Feb 18, 2010 at 3:14 PM, Peter P GMX <Prometheus001 at gmx.net
> <mailto:Prometheus001 at gmx.net>> wrote:
>
>     Any idea how to do this?
>
>     currently I have
>     {presence_id=${dialed_user}@${dialed_domain},transfer_fallback_extension=${dialed_user}}${sofia_contact(${dialed_user}@${dialed_domain})}
>
>
>     Best regards
>     Peter
>
>     Anthony Minessale schrieb:
>     > edit the dial-string for that user in the directory xml to try the
>     > extension on both profile at once
>     >
>     > On Thu, Feb 18, 2010 at 1:27 PM, Peter P GMX
>     <Prometheus001 at gmx.net <mailto:Prometheus001 at gmx.net>
>     > <mailto:Prometheus001 at gmx.net <mailto:Prometheus001 at gmx.net>>>
>     wrote:
>     >
>     >     Hello,
>     >
>     >     in the standard setup - if a phone is registering to port
>     5060 - it is
>     >     bound to the "internal" profile. And I can dial it via
>     >     sofia/user/xxxx then.
>     >
>     >     However due to NAT issues I would like to have to 2 seperate
>     profiles
>     >     for SIP phones. For example I have a "local" profile for all
>     devices
>     >     inside the LAN (e.g. Pattons und in future: local phones)
>     and another
>     >     "internal" profile which allows also external phones via
>     >     external-xxx-ip. That way I would like to ensure that local
>     phones
>     >     have
>     >     nothing to do with natted adresses and that external phones can
>     >     register
>     >     via external IPs.
>     >
>     >     Question How do I manage that I can register a phone to the
>     "local"
>     >     profile and being able to dial that phone via sofia/user/xxxxx?
>     >
>     >     Or do I think too complicated and there is simply nothing
>     special
>     >     to do?
>     >
>     >     Best regards
>     >     Peter
>     >
>     >     _______________________________________________
>     >     FreeSWITCH-users mailing list
>     >     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     >     <mailto:FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>>
>     >     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     >    
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     >     http://www.freeswitch.org
>     >
>     >
>     >
>     >
>     > --
>     > Anthony Minessale II
>     >
>     > FreeSWITCH http://www.freeswitch.org/
>     > ClueCon http://www.cluecon.com/
>     > Twitter: http://twitter.com/FreeSWITCH_wire
>     >
>     > AIM: anthm
>     > MSN:anthony_minessale at hotmail.com
>     <mailto:MSN%3Aanthony_minessale at hotmail.com>
>     > <mailto:MSN%3Aanthony_minessale at hotmail.com
>     <mailto:MSN%253Aanthony_minessale at hotmail.com>>
>     > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>     <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
>     > <mailto:PAYPAL%3Aanthony.minessale at gmail.com
>     <mailto:PAYPAL%253Aanthony.minessale at gmail.com>>
>     > IRC: irc.freenode.net <http://irc.freenode.net>
>     <http://irc.freenode.net> #freeswitch
>     >
>     > FreeSWITCH Developer Conference
>     > sip:888 at conference.freeswitch.org
>     <mailto:sip%3A888 at conference.freeswitch.org>
>     > <mailto:sip%3A888 at conference.freeswitch.org
>     <mailto:sip%253A888 at conference.freeswitch.org>>
>     > iax:guest at conference.freeswitch.org/888
>     <http://iax:guest at conference.freeswitch.org/888>
>     > <http://iax:guest at conference.freeswitch.org/888>
>     > googletalk:conf+888 at conference.freeswitch.org
>     <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
>     > <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org
>     <mailto:googletalk%253Aconf%252B888 at conference.freeswitch.org>>
>     > pstn:+19193869900
>     >
>     ------------------------------------------------------------------------
>     >
>     > _______________________________________________
>     > FreeSWITCH-users mailing list
>     > FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     >
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     > http://www.freeswitch.org
>     >
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> <mailto:MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> <mailto:sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> <http://iax:guest at conference.freeswitch.org/888>
> googletalk:conf+888 at conference.freeswitch.org
> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
> ------------------------------------------------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   



From valentin.doroga at pronexus.com  Thu Feb 18 14:48:06 2010
From: valentin.doroga at pronexus.com (Valentin Doroga)
Date: Thu, 18 Feb 2010 17:48:06 -0500
Subject: [Freeswitch-users] IVR greeting - first two words missing
In-Reply-To: <8a19bf2e1002181410x44b40216j374a30a5675fd162@mail.gmail.com>
Message-ID: <mailman.16.1269221164.3324.freeswitch-users@lists.freeswitch.org>

          greet-long="test/test-ivr.wav"
          greet-short="tset/test-ivr.wav"
 
test or tset?

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Aloysius
Lloyd
Sent: Thursday, February 18, 2010 5:10 PM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] IVR greeting - first two words missing


Hi All, 

I setup a simple IVR. Here is the script. 

<menu name="test_ivr"
          greet-long="test/test-ivr.wav"
          greet-short="tset/test-ivr.wav"
          invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"
          exit-sound="voicemail/vm-goodbye.wav"
          timeout ="10000"
                  inter-digit-timeout="2000"
                  max-failures="3">
 <entry action="menu-exec-app" digits="/^(20[0-9])$/" param="transfer $1 XML
default"/>
 <entry action="menu-exec-app" digits="9999" param="transfer 9999 XML
default"/>
</menu>
  </menus>
</configuration>

Dial Plan

<action application="answer"/>
<action application="ivr" data="test_ivr"/>

Every time when I reach the IVR . I am getting first one or two words
missing( or may be not clear). How can I fix this issue.


Thanks,
Lloyd 
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100218/7e7b1f78/attachment.html 

From iamcanadian at myfastmail.com  Thu Feb 18 14:50:09 2010
From: iamcanadian at myfastmail.com (Edward Stevenson)
Date: Thu, 18 Feb 2010 14:50:09 -0800 (PST)
Subject: [Freeswitch-users]  Voicemail quality
Message-ID: <27251411.post@talk.nabble.com>


I have V1.0.4 running on a production server.  It's working quite well,
except for voicemail retrieval over a satellite internet connection.  Voice
calls over satellite sound fine, other than the 600ms delay.  I'm using the
Howler G729 module for G729 transcoding.

I've noticed that in voice calls, the bandwidth over the satellite is a
steady 24 kbps in both directions.  When accessing voicemail, the bandwidth
fluctuates with voice in the call.  What I mean by that is, pauses in audio
in the voicemail, or in the IVR, cause the bandwidth of the call to drop off
momentarily.  It's almost like it's using a variable bit rate.  The audio
sounds like there's packet loss.  Pops and garbled speach.  This is not
noticable over a landline.  I'm using the built in voicemail.

Anyone else had any similar issues?
-- 
View this message in context: http://old.nabble.com/Voicemail-quality-tp27251411p27251411.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From msc at freeswitch.org  Thu Feb 18 15:22:02 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 18 Feb 2010 15:22:02 -0800
Subject: [Freeswitch-users] IVR greeting - first two words missing
In-Reply-To: <8a19bf2e1002181410x44b40216j374a30a5675fd162@mail.gmail.com>
References: <8a19bf2e1002181410x44b40216j374a30a5675fd162@mail.gmail.com>
Message-ID: <87f2f3b91002181522te296581u12527a2a9cdf1d44@mail.gmail.com>

On Thu, Feb 18, 2010 at 2:10 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:

> Hi All,
>
> I setup a simple IVR. Here is the script.
>
> <menu name="test_ivr"
>           greet-long="test/test-ivr.wav"
>           greet-short="tset/test-ivr.wav"
>           invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"
>           exit-sound="voicemail/vm-goodbye.wav"
>           timeout ="10000"
>                   inter-digit-timeout="2000"
>                   max-failures="3">
>  <entry action="menu-exec-app" digits="/^(20[0-9])$/" param="transfer $1
> XML default"/>
>  <entry action="menu-exec-app" digits="9999" param="transfer 9999 XML
> default"/>
> </menu>
>   </menus>
> </configuration>
>
> *Dial Plan*
>
> <action application="answer"/>
> <action application="ivr" data="test_ivr"/>
>
> Every time when I reach the IVR . I am getting first one or two words
> missing( or may be not clear). How can I fix this issue.
>
> Thanks,
> Lloyd
>

put a sleep after the answer:
<action application="sleep" data="1000"/>

You may have to tinker with the exact time, like maybe 1500 or 2000.
-MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100218/f3925d67/attachment-0002.html 

From dftoro at yahoo.com  Thu Feb 18 15:34:44 2010
From: dftoro at yahoo.com (Diego Toro)
Date: Thu, 18 Feb 2010 15:34:44 -0800 (PST)
Subject: [Freeswitch-users] Establishing a Call from .Net based
	application
In-Reply-To: <6E8D2069C08AA84A83D336E996AE4C67032C9D5BAD@mse17be1.mse17.exchange.ms>
Message-ID: <922191.64755.qm@web33507.mail.mud.yahoo.com>

The managed module is loaded as a module during the startup of FreeSWITCH if set in modules.conf.xml or through the command "load mod_managed" must keep in mind that there is a directory "mod/managed. As mod_managed is loaded into FreeSWITCH process to take control of the call must be running FreeSWITCH. So to "talk" with FreeSWITCH is not necessary to know the IP, the IP depends on the profile you've defined in the configuration of the module sofia. If you need the local address of the box running FreeSWITCH try expand variable $${local_ip_v4} which is assigned automatically by FreeSWITCH.

Being more clear, when you use mod_managed including <action application="managed" data="yourclassname"/>
 in a dialplan already have way to run your C# code.
 
Now, if you need is to have control of the call to answer, originate, etc, without the application run inside FreeSWITCH process, you can use managed ESL (see examples in libs/esl/managed) this library allows your code using events "talk" with FreeSWITCH.

Diego Toro
http://lacarretade.blogspot.com/


--- On Thu, 2/18/10, Michael Giagnocavo <mgg at giagnocavo.net> wrote:

> From: Michael Giagnocavo <mgg at giagnocavo.net>
> Subject: Re: [Freeswitch-users] Establishing a Call from .Net based application
> To: "freeswitch-users at lists.freeswitch.org" <freeswitch-users at lists.freeswitch.org>
> Date: Thursday, February 18, 2010, 6:12 AM
> I?m not sure what the
> FreeSWITCH APIs are to figure out what IP Sofia SIP has
> bound to. Whatever it is, you?d call the same thing in
> C#. What do you want to do with the API? ?mod_managed.dll or .so is the
> FreeSWITCH native code module that loads the CLR or Mono
> into the FreeSWITCH process and loads
> FreeSWITCH.Managed.dll. The managed DLL contains the bulk of
> the managed-unmanaged interop code (.NET definitions of all
> the FS C functions).  ?-Michael ?From:
> freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On
> Behalf Of Scott Fernandez
> Sent: Thursday, February 18, 2010 1:12 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Establishing a Call
> from .Net based application
> ?Hi Diego & Michael,
> 
> Thanks for your reply and support.
> 
> However, I have some clarifications required from both of
> you.
> 
> 1. Here is the question for Diego,
> 
> Simple Example:
> 
> using FreeSWITCH;
> using FreeSWITCH.Native;
> 
> namespace BITS.Ivr.Bp.Server
> {
> ?public class CIV_BPFSProcess : FreeSWITCH.IAppPlugin
> {
> ?public void Run(AppContext context)
> ?{
> ? //answer call
> ? context.Session.Answer();
> ? //sleep 2 seconds
> ? context.Session.sleep(2000, 1);
> ? //hangup call
> ? context.Session.Hangup("NORMAL_CLEARING");
> ?}
> ?}
> }
> I understand that the concept of your example code.
> However, would like to know as to how would my .NET C# know the
> IP address of Freeswitch to talk to it as there is no
> indication for that?. If not here, where would we need to
> reference the IP address of FS in .NET code?
> 
> I guess the IP address of FS needs to be mentioned in the
> Target section of the below web.config file in .NET. If I am
> right, how to specify the IP address over here. If I am
> wrong, please let me know where do we need to mention the IP
> address of FS.
> 
> ??? <configuration>
> ???????????
> <dllmap dll="mod_managed.dll"
> target="mod_managed.so"/>
> ??? </configuration>
> 
> 
> 2. Here is the question for Michael,
> 
> You mentioned that "mod_managed.so will
> be in your freeswitch mod directory". This is
> very clear and what is mod_managed.dll in my .NET
> application and the purpose of it?
> 
> Thanks for all your help.
> 
> Regards,
> Scott.
> 
> 
> On Sun, Feb 14, 2010 at 1:15
> AM, Michael Giagnocavo <mgg at giagnocavo.net>
> wrote:
> 2. There is a configuration settings required to Map the
> "DLL" to ".so" object in CentOS.
> Now, the question is which DLL and .so file to be made
> available and where??If you are
> experiencing NullReferenceExceptions with any plugin run
> through the dialplan, make sure you have included the
> appropriate entry in your dllmap
> configuration: ?<dllmap dll="mod_managed.dll"
> target="mod_managed.so"
> os="!windows"/>?mod_managed.so will
> be in your freeswitch mod directory.
> All I need is to initiate a call from .NET application and
> then it should talk to mod_managed module and establish a
> call. Secondly, I need to know the status of the call such
> as Ringing, Active, Hangup etc. To initiate a
> call, try ManagedSession.Originate.-Michael
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org ?
> -----Inline Attachment Follows-----
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 


      



From brian at microcomaustralia.com.au  Thu Feb 18 15:42:47 2010
From: brian at microcomaustralia.com.au (Brian May)
Date: Fri, 19 Feb 2010 10:42:47 +1100
Subject: [Freeswitch-users] building for Lenny
In-Reply-To: <20100218161741.GC4236@base.carmickle.com>
References: <3c5cf5261002180046vb0015e7t787599c9903aedc@mail.gmail.com>
	<33c87fa31002180056g51ef8496qa9a35746d0fe75ec@mail.gmail.com>
	<3c5cf5261002180110he0ae815uc713f77f289ac625@mail.gmail.com>
	<a9f4977a70b3fcf1b30413ea562ca962.squirrel@www.webcontracts.co.uk>
	<20100218161741.GC4236@base.carmickle.com>
Message-ID: <3c5cf5261002181542y43adb7a7ofe2a15c38baf0327@mail.gmail.com>

On 19 February 2010 03:17, Frank Carmickle <frank at carmickle.com> wrote:
> Like I said you can and should build debs from svn. ?As far as
> I see it there is no reason to not build debs.

Unfortunately, that didn't create any of the packages for the sound
files, and I can't see where to get a deb package for the sound files
that really does contain the sound files.



Also I get errors when trying to start it up, not sure how many of
these I can ignore are warnings and how many are because I am doing it
wrong:

voyage:~# /opt/freeswitch/bin/freeswitch
Error: stacksize 4194303 is too large: run ulimit -s 240 or run
/opt/freeswitch/bin/freeswitch -waste.
auto-adjusting stack size for optimal performance...
2010-02-19 10:59:41.203000 [INFO] switch_event.c:580 Activate Eventing Engine.
2010-02-19 10:59:41.208000 [DEBUG] switch_event.c:568 Create event
dispatch thread 0
2010-02-19 10:59:41.428000 [ERR] switch_xml.c:1297 Couldnt open
/opt/freeswitch/conf/autoload_configs/../ivr_menus/*.xml (No such file
or directory)
Error including
/opt/freeswitch/conf/autoload_configs/../ivr_menus/*.xml (No such file
or directory)
2010-02-19 10:59:42.376000 [ERR] switch_xml.c:1297 Couldnt open
/opt/freeswitch/conf/lang/de/*.xml (No such file or directory)
Error including /opt/freeswitch/conf/lang/de/*.xml (No such file or directory)
2010-02-19 10:59:42.526000 [ERR] switch_xml.c:1297 Couldnt open
/opt/freeswitch/conf/lang/en/dir/sounds.xml (No such file or
directory)
Error including /opt/freeswitch/conf/lang/en/dir/sounds.xml (No such
file or directory)
2010-02-19 10:59:42.528000 [ERR] switch_xml.c:1297 Couldnt open
/opt/freeswitch/conf/lang/fr/*.xml (No such file or directory)
Error including /opt/freeswitch/conf/lang/fr/*.xml (No such file or directory)
2010-02-19 10:59:42.529000 [ERR] switch_xml.c:1297 Couldnt open
/opt/freeswitch/conf/lang/ru/*.xml (No such file or directory)
Error including /opt/freeswitch/conf/lang/ru/*.xml (No such file or directory)
2010-02-19 10:59:42.609000 [INFO] switch_nat.c:409 Scanning for NAT
2010-02-19 10:59:42.611000 [DEBUG] switch_nat.c:166 Checking for PMP 1/5
2010-02-19 10:59:42.614000 [ERR] switch_nat.c:197 Error checking for
PMP [general error]
2010-02-19 10:59:42.614000 [DEBUG] switch_nat.c:414 Checking for UPnP
2010-02-19 10:59:54.619000 [INFO] switch_nat.c:429 No PMP or UPnP NAT
devices detected!
2010-02-19 10:59:54.623000 [INFO] switch_core_sqldb.c:1248 Opening DB
2010-02-19 10:59:54.627000 [ERR] switch_core_sqldb.c:404 SQL ERR [no
such table: channels]
drop table channels
2010-02-19 10:59:54.628000 [ERR] switch_core_sqldb.c:404 SQL ERR [no
such table: calls]
drop table calls
2010-02-19 10:59:54.630000 [ERR] switch_core_sqldb.c:404 SQL ERR [no
such table: interfaces]
drop table interfaces
2010-02-19 10:59:54.631000 [ERR] switch_core_sqldb.c:404 SQL ERR [no
such table: tasks]
drop table tasks
2010-02-19 10:59:54.634000 [DEBUG] switch_core_sqldb.c:765 SQL ERR [no
such table: aliases]
[select hostname from aliases]
Auto Generating Table!
2010-02-19 10:59:54.636000 [DEBUG] switch_core_sqldb.c:772 SQL ERR [no
such table: aliases]
[CREATE TABLE aliases (
   sticky  INTEGER,
   alias  VARCHAR(128),
   command  VARCHAR(4096),
   hostname VARCHAR(256)
);
]
2010-02-19 10:59:54.644000 [DEBUG] switch_core_sqldb.c:765 SQL ERR [no
such table: nat]
[select hostname from nat]
Auto Generating Table!
2010-02-19 10:59:54.646000 [DEBUG] switch_core_sqldb.c:772 SQL ERR [no
such table: nat]
[CREATE TABLE nat (
   sticky  INTEGER,
	port	INTEGER,
	proto	INTEGER,
   hostname VARCHAR(256)
);
]

Am I expected to setup a SQL database to get this working? Or did it
just setup one automatically?

2010-02-19 10:59:59.684000 [CRIT] switch_loadable_module.c:882 Error
Loading module /opt/freeswitch/mod/mod_voipcodecs.so
**libjpeg.so.62: cannot open shared object file: No such file or directory**
2010-02-19 10:59:59.686000 [CRIT] switch_loadable_module.c:882 Error
Loading module /opt/freeswitch/mod/mod_g723_1.so
**/opt/freeswitch/mod/mod_g723_1.so: cannot open shared object file:
No such file or directory**
2010-02-19 10:59:59.687000 [CRIT] switch_loadable_module.c:882 Error
Loading module /opt/freeswitch/mod/mod_g729.so
**/opt/freeswitch/mod/mod_g729.so: cannot open shared object file: No
such file or directory**
2010-02-19 10:59:59.688000 [CRIT] switch_loadable_module.c:882 Error
Loading module /opt/freeswitch/mod/mod_amr.so
**/opt/freeswitch/mod/mod_amr.so: cannot open shared object file: No
such file or directory**
2010-02-19 10:59:59.835000 [CRIT] switch_loadable_module.c:882 Error
Loading module /opt/freeswitch/mod/mod_file_string.so
**/opt/freeswitch/mod/mod_file_string.so: cannot open shared object
file: No such file or directory**
2010-02-19 11:00:00.143000 [CRIT] switch_loadable_module.c:882 Error
Loading module /opt/freeswitch/mod/mod_say_ru.so
**/opt/freeswitch/mod/mod_say_ru.so: cannot open shared object file:
No such file or directory**

suspect I don't really need to worry about some of these. I assume
there is a config file somewhere where I can disable these options.


Ok, as a really pathetic question, now I have started it, how do I stop it?

freeswitch at voyage> halt
Unknown Command: halt
freeswitch at voyage> quit
Unknown Command: quit
freeswitch at voyage> exit
Unknown Command: exit
freeswitch at voyage> bye
Unknown Command: bye

-- 
Brian May <brian at microcomaustralia.com.au>



From iamcanadian at myfastmail.com  Thu Feb 18 17:21:58 2010
From: iamcanadian at myfastmail.com (Edward Stevenson)
Date: Thu, 18 Feb 2010 17:21:58 -0800 (PST)
Subject: [Freeswitch-users]  voivemail quality
Message-ID: <27648642.post@talk.nabble.com>


I have V1.0.4 running on a test/production server.  It's working quite well,
except for voicemail retrieval over a satellite internet connection.  Voice
calls over satellite sound fine, other than the 600ms delay.  I'm using the
Howler Tech G729 module for G729 transcoding.

I've noticed that in voice calls, the bandwidth over the satellite is a
steady 24 kbps in both directions.  When accessing voicemail, the bandwidth
fluctuates with voice in the call.  What I mean by that is, pauses in audio
in the voicemail, or in the IVR, cause the bandwidth of the call to drop off
momentarily.  It's almost like it's using a variable bit rate.  The audio
sounds like there's packet loss.  Pops and garbled speach.  This is not
noticable over a land internet connection.  I'm using the built in
voicemail.

If I change the phone's codec to G711, the call is perfectly clear.  Perhaps
the Howler module doesn't like transcoding from the L16 codec that
Freeswitch seems to use to play the wav file?

Anyone else had any similar issues?
-- 
View this message in context: http://old.nabble.com/voivemail-quality-tp27648642p27648642.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From iamcanadian at myfastmail.com  Thu Feb 18 17:22:48 2010
From: iamcanadian at myfastmail.com (Edward Stevenson)
Date: Thu, 18 Feb 2010 17:22:48 -0800 (PST)
Subject: [Freeswitch-users]  voivemail quality
Message-ID: <27648642.post@talk.nabble.com>


I have V1.0.4 running on a test/production server.  It's working quite well,
except for voicemail retrieval over a satellite internet connection.  Voice
calls over satellite sound fine, other than the 600ms delay.  I'm using the
Howler Tech G729 module for G729 transcoding.

I've noticed that in voice calls, the bandwidth over the satellite is a
steady 24 kbps in both directions.  When accessing voicemail, the bandwidth
fluctuates with voice in the call.  What I mean by that is, pauses in audio
in the voicemail, or in the IVR, cause the bandwidth of the call to drop off
momentarily.  It's almost like it's using a variable bit rate.  The audio
sounds like there's packet loss.  Pops and garbled speach.  This is not
noticable over a land internet connection.  I'm using the built in
voicemail.

If I change the phone's codec to G711, the call is perfectly clear.  Perhaps
the Howler module doesn't like transcoding from the L16 codec that
Freeswitch seems to use to play the wav file?

Anyone else had any similar issues?
-- 
View this message in context: http://old.nabble.com/voivemail-quality-tp27648642p27648642.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From lon at kickasspixels.com  Thu Feb 18 17:41:02 2010
From: lon at kickasspixels.com (Lon Baker)
Date: Thu, 18 Feb 2010 17:41:02 -0800
Subject: [Freeswitch-users] 1.0.4 vs. trunk vs. bugs
Message-ID: <5d3e0dc61002181741r57b91f86h4810de63c1a3d46d@mail.gmail.com>

Guys,

With all due respect I want to suggest a policy change. A while ago it
was announced that bug reports against 1.0.4 would not be accepted and
the solution was to work off the trunk or the latest nightly build.

It seems more reasonable to have a release/production branch that can
be depended on for production use. This branch would only accept
reports and fixes for critical bugs.

The development branch is where feature requests and non-critical bugs
reports would be filed for the next production release.

The current process leaves a gap between production ready and
development code that may become greater over time.

Just a thought.

Lon



From jason at jasonjgw.net  Thu Feb 18 17:51:18 2010
From: jason at jasonjgw.net (Jason White)
Date: Fri, 19 Feb 2010 12:51:18 +1100
Subject: [Freeswitch-users] 1.0.4 vs. trunk vs. bugs
In-Reply-To: <5d3e0dc61002181741r57b91f86h4810de63c1a3d46d@mail.gmail.com>
References: <5d3e0dc61002181741r57b91f86h4810de63c1a3d46d@mail.gmail.com>
Message-ID: <20100219015118.GA12983@jdc.jasonjgw.net>

Lon Baker <lon at kickasspixels.com> wrote:
 
> The development branch is where feature requests and non-critical bugs
> reports would be filed for the next production release.
> 
> The current process leaves a gap between production ready and
> development code that may become greater over time.

Did you read the statements by FreeSWITCH developers indicating that the svn
trunk is usually more stable than "released" versions, and that this is at
least partly due to a lack of testers/testing prior to release?

A change of policy isn't going to address those underlying problems.

For the record, I don't favour the proposed change.




From wangdq.no1 at gmail.com  Thu Feb 18 17:54:12 2010
From: wangdq.no1 at gmail.com (daqiang wang)
Date: Fri, 19 Feb 2010 09:54:12 +0800
Subject: [Freeswitch-users] how to use mod_erlang ?
In-Reply-To: <20100218190129.GD8518@hijacked.us>
References: <ecd05ce31002171917w2713f8detcee3f7e36039e9f2@mail.gmail.com>
	<20100218190129.GD8518@hijacked.us>
Message-ID: <ecd05ce31002181754x338f6b4bt5e3609cd74b28f3b@mail.gmail.com>

ok, thank you very much!


2010/2/19 Andrew Thompson <andrew at hijacked.us>

> On Thu, Feb 18, 2010 at 11:17:26AM +0800, daqiang wang wrote:
> > hello !
> > I test mod_erlang from :
> http://wiki.freeswitch.org/wiki/Mod_erlang_event
> > but when I input :
> >  >
> >
> > {foo, freeswitch at localhost } ! {api, status, ""}.
> > I received
> >   (test at wangdq-laptop)2>
> > =ERROR REPORT==== 18-Feb-2010::11:10:13 ===
> > Error in process <0.41.0> on node 'test at wangdq-laptop' with exit
> > value: {badarg,[{erlang,list_to_existing_atom,["freeswitch at wangdq-laptop
> "]},{dist_util,recv_challenge,1},{dist_util,handshake_we_started,1}]}
> >
>
> Try {foo, 'freeswitch at wangdq-laptop' } ! {api, status, ""}. instead.
>
> Andrew
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100219/f5d92703/attachment-0002.html 

From technical at ttnc.co.uk  Thu Feb 18 17:57:55 2010
From: technical at ttnc.co.uk (TTNC - Technical)
Date: Fri, 19 Feb 2010 01:57:55 +0000
Subject: [Freeswitch-users] mod_fax undefined symbol
Message-ID: <6FCE44FF-70C1-412D-B722-4C8D359381BA@ttnc.co.uk>

Hi Guys

I'm having trouble getting mod_fax to load. Running on Debian testing (squeeze). Everything is installed as per - http://wiki.freeswitch.org/wiki/Installation_Guide. (dpkg-buildpackage etc)

When trying to load the fax module I get:

2010-02-19 01:48:17.554935 [CRIT] switch_loadable_module.c:882 Error Loading module /opt/freeswitch/mod/mod_fax.so
**/opt/freeswitch/mod/mod_fax.so: undefined symbol: TIFFDefaultStripSize**

And when a fax is sent, I'm getting:

 2010-02-19 01:04:57.355330 [ERR] switch_core_session.c:1490 Invalid Application rxfax 

I guess because mod_fax isn't loaded.

I've got libtiff4 and libtiff4-dev installed:

ii  libtiff4                             3.9.2-2                  Tag Image File Format (TIFF) library
ii  libtiff4-dev                         3.9.2-2                  Tag Image File Format library (TIFF), development files
ii  libtiffxx0c2                         3.9.2-2                  Tag Image File Format (TIFF) library -- C++ interface

Just tried updating to the latest svn trunk (16700M) and it hasn't made any difference.

From googling, it suggests that it could be because the module is complied against a different one currently running on the system, however I'm not sure how this can be the case, there is only the one version installed. 

Any suggestions as to what I can try?

Any help appreciated

Russ


From rupa at rupa.com  Thu Feb 18 18:01:17 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Thu, 18 Feb 2010 20:01:17 -0600
Subject: [Freeswitch-users] voivemail quality
In-Reply-To: <27648642.post@talk.nabble.com>
References: <27648642.post@talk.nabble.com>
Message-ID: <b39273531002181801kd81a6b2x3327be4887563eae@mail.gmail.com>

Have you asked Howler about this?  This is not a support channel for
commercial software that doesn't participate or contribute in the community.

On Thu, Feb 18, 2010 at 7:21 PM, Edward Stevenson <
iamcanadian at myfastmail.com> wrote:

>
> I have V1.0.4 running on a test/production server.  It's working quite
> well,
> except for voicemail retrieval over a satellite internet connection.  Voice
> calls over satellite sound fine, other than the 600ms delay.  I'm using the
> Howler Tech G729 module for G729 transcoding.
>
> I've noticed that in voice calls, the bandwidth over the satellite is a
> steady 24 kbps in both directions.  When accessing voicemail, the bandwidth
> fluctuates with voice in the call.  What I mean by that is, pauses in audio
> in the voicemail, or in the IVR, cause the bandwidth of the call to drop
> off
> momentarily.  It's almost like it's using a variable bit rate.  The audio
> sounds like there's packet loss.  Pops and garbled speach.  This is not
> noticable over a land internet connection.  I'm using the built in
> voicemail.
>
> If I change the phone's codec to G711, the call is perfectly clear.
>  Perhaps
> the Howler module doesn't like transcoding from the L16 codec that
> Freeswitch seems to use to play the wav file?
>
> Anyone else had any similar issues?
> --
> View this message in context:
> http://old.nabble.com/voivemail-quality-tp27648642p27648642.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
-Rupa
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100218/664ab29f/attachment-0002.html 

From jaybinks at gmail.com  Thu Feb 18 18:14:00 2010
From: jaybinks at gmail.com (jay binks)
Date: Fri, 19 Feb 2010 12:14:00 +1000
Subject: [Freeswitch-users] voivemail quality
In-Reply-To: <b39273531002181801kd81a6b2x3327be4887563eae@mail.gmail.com>
References: <27648642.post@talk.nabble.com>
	<b39273531002181801kd81a6b2x3327be4887563eae@mail.gmail.com>
Message-ID: <ec2ce17e1002181814l26038ddat19923218eb014ff0@mail.gmail.com>

I might suggest you try with G711 , ilbc, speex ..
see if it works with these ( and not which do / dont work )

then post back, I think Rupa's point is that it may be a codec issue.
be helpful ( either way ) to prove that it is / isnt this 3rd party codec.

Jay

On Fri, Feb 19, 2010 at 12:01 PM, Rupa Schomaker <rupa at rupa.com> wrote:

> Have you asked Howler about this?  This is not a support channel for
> commercial software that doesn't participate or contribute in the community.
>
>
> On Thu, Feb 18, 2010 at 7:21 PM, Edward Stevenson <
> iamcanadian at myfastmail.com> wrote:
>
>>
>> I have V1.0.4 running on a test/production server.  It's working quite
>> well,
>> except for voicemail retrieval over a satellite internet connection.
>>  Voice
>> calls over satellite sound fine, other than the 600ms delay.  I'm using
>> the
>> Howler Tech G729 module for G729 transcoding.
>>
>> I've noticed that in voice calls, the bandwidth over the satellite is a
>> steady 24 kbps in both directions.  When accessing voicemail, the
>> bandwidth
>> fluctuates with voice in the call.  What I mean by that is, pauses in
>> audio
>> in the voicemail, or in the IVR, cause the bandwidth of the call to drop
>> off
>> momentarily.  It's almost like it's using a variable bit rate.  The audio
>> sounds like there's packet loss.  Pops and garbled speach.  This is not
>> noticable over a land internet connection.  I'm using the built in
>> voicemail.
>>
>> If I change the phone's codec to G711, the call is perfectly clear.
>>  Perhaps
>> the Howler module doesn't like transcoding from the L16 codec that
>> Freeswitch seems to use to play the wav file?
>>
>> Anyone else had any similar issues?
>> --
>> View this message in context:
>> http://old.nabble.com/voivemail-quality-tp27648642p27648642.html
>> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Sincerely

Jay
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100219/4e62cc29/attachment-0002.html 

From lon at kickasspixels.com  Thu Feb 18 18:28:30 2010
From: lon at kickasspixels.com (Lon Baker)
Date: Thu, 18 Feb 2010 18:28:30 -0800
Subject: [Freeswitch-users] 1.0.4 vs. trunk vs. bugs
In-Reply-To: <20100219015118.GA12983@jdc.jasonjgw.net>
References: <5d3e0dc61002181741r57b91f86h4810de63c1a3d46d@mail.gmail.com>
	<20100219015118.GA12983@jdc.jasonjgw.net>
Message-ID: <DE4923F9-EEC2-4009-9A0D-D62C1CF3BD0F@kickasspixels.com>

Jason,

Yes. I saw that. But, quite a few times I have found the trunk unstable or broken. Its why I'm proposing the changes.

To be clear, I'm not in anyway impugning the quality or expertise of the dev team. They are doing amazing work! 

I'm looking down the road. As the project grows and FS becomes a critical piece of any company's infrastructure a clear distinction between "stable" and "development" branches are pretty standard in project the scope of FS.

Lon

On Feb 18, 2010, at 5:51 PM, Jason White wrote:

> Lon Baker <lon at kickasspixels.com> wrote:
> 
>> The development branch is where feature requests and non-critical bugs
>> reports would be filed for the next production release.
>> 
>> The current process leaves a gap between production ready and
>> development code that may become greater over time.
> 
> Did you read the statements by FreeSWITCH developers indicating that the svn
> trunk is usually more stable than "released" versions, and that this is at
> least partly due to a lack of testers/testing prior to release?
> 
> A change of policy isn't going to address those underlying problems.
> 
> For the record, I don't favour the proposed change.
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From brian at freeswitch.org  Thu Feb 18 18:35:06 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 18 Feb 2010 20:35:06 -0600
Subject: [Freeswitch-users] 1.0.4 vs. trunk vs. bugs
In-Reply-To: <DE4923F9-EEC2-4009-9A0D-D62C1CF3BD0F@kickasspixels.com>
References: <5d3e0dc61002181741r57b91f86h4810de63c1a3d46d@mail.gmail.com>
	<20100219015118.GA12983@jdc.jasonjgw.net>
	<DE4923F9-EEC2-4009-9A0D-D62C1CF3BD0F@kickasspixels.com>
Message-ID: <865D1CDE-736D-4ED6-B7D6-94384143BCDA@freeswitch.org>

In the grand scheme trunk isn't unstable or broken for long... this is just downright false.  If it was why aren't you opening bugs?

We have had a few snafu's but again they don't stay long.

/b


On Feb 18, 2010, at 8:28 PM, Lon Baker wrote:

> Yes. I saw that. But, quite a few times I have found the trunk unstable or broken. Its why I'm proposing the changes.




From jason at jasonjgw.net  Thu Feb 18 18:53:11 2010
From: jason at jasonjgw.net (Jason White)
Date: Fri, 19 Feb 2010 13:53:11 +1100
Subject: [Freeswitch-users] 1.0.4 vs. trunk vs. bugs
In-Reply-To: <DE4923F9-EEC2-4009-9A0D-D62C1CF3BD0F@kickasspixels.com>
References: <5d3e0dc61002181741r57b91f86h4810de63c1a3d46d@mail.gmail.com>
	<20100219015118.GA12983@jdc.jasonjgw.net>
	<DE4923F9-EEC2-4009-9A0D-D62C1CF3BD0F@kickasspixels.com>
Message-ID: <20100219025311.GA13893@jdc.jasonjgw.net>

Lon Baker <lon at kickasspixels.com> wrote:
 
> I'm looking down the road. As the project grows and FS becomes a critical
> piece of any company's infrastructure a clear distinction between "stable"
> and "development" branches are pretty standard in project the scope of FS.

I would rather that the developers spend time fixing bugs and implementing new
features instead of backporting changes to a "stable" branch that may be very
outdated.

One way of managing this is the Linux kernel's model, with short
development/release cycles, where there is only one branch. (There are
short-lived "stable" branches as in 2.6.32.1, 2.6.32.2 etc., but my
understanding is that bug fixes destined for those branches must already have
been applied to the development branch destined for the next release; this
minimizes back-porting of fixes.)

I'm sure there are other models. My essential point is that, so far, the
FreeSWITCH developers have not chosen to maintain long-lived "stable"
branches, that there are alternatives to this approach, and that it has its
downside, especially regarding the extra time/effort/work associated with
maintaining it.




From lon at kickasspixels.com  Thu Feb 18 18:53:16 2010
From: lon at kickasspixels.com (Lon Baker)
Date: Thu, 18 Feb 2010 18:53:16 -0800
Subject: [Freeswitch-users] 1.0.4 vs. trunk vs. bugs
In-Reply-To: <865D1CDE-736D-4ED6-B7D6-94384143BCDA@freeswitch.org>
References: <5d3e0dc61002181741r57b91f86h4810de63c1a3d46d@mail.gmail.com> 
	<20100219015118.GA12983@jdc.jasonjgw.net>
	<DE4923F9-EEC2-4009-9A0D-D62C1CF3BD0F@kickasspixels.com> 
	<865D1CDE-736D-4ED6-B7D6-94384143BCDA@freeswitch.org>
Message-ID: <5d3e0dc61002181853u2b5d7022ga79d41adaeb7ae68@mail.gmail.com>

I'll drop it. You're right they don't last long. I didn't open bugs
because I assumed the trunk is being worked in, so breakage is
something I expect and anywhere from minutes to a few hours later they
are gone.

Lon

On Thu, Feb 18, 2010 at 6:35 PM, Brian West <brian at freeswitch.org> wrote:
> In the grand scheme trunk isn't unstable or broken for long... this is just downright false. ?If it was why aren't you opening bugs?
>
> We have had a few snafu's but again they don't stay long.
>
> /b
>
>
> On Feb 18, 2010, at 8:28 PM, Lon Baker wrote:
>
>> Yes. I saw that. But, quite a few times I have found the trunk unstable or broken. Its why I'm proposing the changes.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From lon at kickasspixels.com  Thu Feb 18 19:08:09 2010
From: lon at kickasspixels.com (Lon Baker)
Date: Thu, 18 Feb 2010 19:08:09 -0800
Subject: [Freeswitch-users] 1.0.4 vs. trunk vs. bugs
In-Reply-To: <20100219025311.GA13893@jdc.jasonjgw.net>
References: <5d3e0dc61002181741r57b91f86h4810de63c1a3d46d@mail.gmail.com> 
	<20100219015118.GA12983@jdc.jasonjgw.net>
	<DE4923F9-EEC2-4009-9A0D-D62C1CF3BD0F@kickasspixels.com> 
	<20100219025311.GA13893@jdc.jasonjgw.net>
Message-ID: <5d3e0dc61002181908p5534c03fw5a258f165c9d3119@mail.gmail.com>

Jason,

I agree with you. Guess its just the anticipation for 1.0.5 to be
released and clients that only allow "released" versions to be
deployed.

Lon



From jmesquita at freeswitch.org  Thu Feb 18 19:29:58 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Fri, 19 Feb 2010 00:29:58 -0300
Subject: [Freeswitch-users] more than 1 profile for sofia/user ?
In-Reply-To: <4B7DC202.7090409@gmx.net>
References: <4B7D9494.8050208@gmx.net>
	<191c3a031002181140y44135106yb34e6a6f73a04773@mail.gmail.com>
	<4B7DADC8.1060405@gmx.net>
	<191c3a031002181353r2dbf15c6h206509f9b1399148@mail.gmail.com>
	<4B7DC202.7090409@gmx.net>
Message-ID: <ebbc2d551002181929m72a17b7bj1fa1cff45b02607b@mail.gmail.com>

I would:

{presence_id=${dialed_user}@
${dialed_domain},transfer_fallback_extension=${dialed_user}}${sofia_contact(${dialed_user}@
${dialed_domain}),sofia/other_profile/${dialed_user}}

You could toy with that a bit. The dialstring is really just an origination
string that is generated by the user/<extension> ...

Hope that clears it up a bit.

Regards,
Jo?o Mesquita



On Thu, Feb 18, 2010 at 7:41 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:

> Hello Anthony,
>
> >add on a , then another dial string to reflect the other profile too
> I really tried to understand this, but
> can you give me an example?
>
> Best regards
> Peter
>
> Anthony Minessale schrieb:
> > add on a , then another dial string to reflect the other profile too
> >
> > On Thu, Feb 18, 2010 at 3:14 PM, Peter P GMX <Prometheus001 at gmx.net
> > <mailto:Prometheus001 at gmx.net>> wrote:
> >
> >     Any idea how to do this?
> >
> >     currently I have
> >     {presence_id=${dialed_user}@
> ${dialed_domain},transfer_fallback_extension=${dialed_user}}${sofia_contact(${dialed_user}@
> ${dialed_domain})}
> >
> >
> >     Best regards
> >     Peter
> >
> >     Anthony Minessale schrieb:
> >     > edit the dial-string for that user in the directory xml to try the
> >     > extension on both profile at once
> >     >
> >     > On Thu, Feb 18, 2010 at 1:27 PM, Peter P GMX
> >     <Prometheus001 at gmx.net <mailto:Prometheus001 at gmx.net>
> >     > <mailto:Prometheus001 at gmx.net <mailto:Prometheus001 at gmx.net>>>
> >     wrote:
> >     >
> >     >     Hello,
> >     >
> >     >     in the standard setup - if a phone is registering to port
> >     5060 - it is
> >     >     bound to the "internal" profile. And I can dial it via
> >     >     sofia/user/xxxx then.
> >     >
> >     >     However due to NAT issues I would like to have to 2 seperate
> >     profiles
> >     >     for SIP phones. For example I have a "local" profile for all
> >     devices
> >     >     inside the LAN (e.g. Pattons und in future: local phones)
> >     and another
> >     >     "internal" profile which allows also external phones via
> >     >     external-xxx-ip. That way I would like to ensure that local
> >     phones
> >     >     have
> >     >     nothing to do with natted adresses and that external phones can
> >     >     register
> >     >     via external IPs.
> >     >
> >     >     Question How do I manage that I can register a phone to the
> >     "local"
> >     >     profile and being able to dial that phone via sofia/user/xxxxx?
> >     >
> >     >     Or do I think too complicated and there is simply nothing
> >     special
> >     >     to do?
> >     >
> >     >     Best regards
> >     >     Peter
> >     >
> >     >     _______________________________________________
> >     >     FreeSWITCH-users mailing list
> >     >     FreeSWITCH-users at lists.freeswitch.org
> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>
> >     >     <mailto:FreeSWITCH-users at lists.freeswitch.org
> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>>
> >     >     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >     >
> >     UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >     >     http://www.freeswitch.org
> >     >
> >     >
> >     >
> >     >
> >     > --
> >     > Anthony Minessale II
> >     >
> >     > FreeSWITCH http://www.freeswitch.org/
> >     > ClueCon http://www.cluecon.com/
> >     > Twitter: http://twitter.com/FreeSWITCH_wire
> >     >
> >     > AIM: anthm
> >     > MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
> >     <mailto:MSN%3Aanthony_minessale at hotmail.com<MSN%253Aanthony_minessale at hotmail.com>
> >
> >     > <mailto:MSN%3Aanthony_minessale at hotmail.com<MSN%253Aanthony_minessale at hotmail.com>
> >     <mailto:MSN%253Aanthony_minessale at hotmail.com<MSN%25253Aanthony_minessale at hotmail.com>
> >>
> >     > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >     <mailto:PAYPAL%3Aanthony.minessale at gmail.com<PAYPAL%253Aanthony.minessale at gmail.com>
> >
> >     > <mailto:PAYPAL%3Aanthony.minessale at gmail.com<PAYPAL%253Aanthony.minessale at gmail.com>
> >     <mailto:PAYPAL%253Aanthony.minessale at gmail.com<PAYPAL%25253Aanthony.minessale at gmail.com>
> >>
> >     > IRC: irc.freenode.net <http://irc.freenode.net>
> >     <http://irc.freenode.net> #freeswitch
> >     >
> >     > FreeSWITCH Developer Conference
> >     > sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> >     <mailto:sip%3A888 at conference.freeswitch.org<sip%253A888 at conference.freeswitch.org>
> >
> >     > <mailto:sip%3A888 at conference.freeswitch.org<sip%253A888 at conference.freeswitch.org>
> >     <mailto:sip%253A888 at conference.freeswitch.org<sip%25253A888 at conference.freeswitch.org>
> >>
> >     > iax:guest at conference.freeswitch.org/888
> >     <http://iax:guest at conference.freeswitch.org/888>
> >     > <http://iax:guest at conference.freeswitch.org/888>
> >     > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >     <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org<googletalk%253Aconf%252B888 at conference.freeswitch.org>
> >
> >     > <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org<googletalk%253Aconf%252B888 at conference.freeswitch.org>
> >     <mailto:googletalk%253Aconf%252B888 at conference.freeswitch.org<googletalk%25253Aconf%25252B888 at conference.freeswitch.org>
> >>
> >     > pstn:+19193869900
> >     >
> >
> ------------------------------------------------------------------------
> >     >
> >     > _______________________________________________
> >     > FreeSWITCH-users mailing list
> >     > FreeSWITCH-users at lists.freeswitch.org
> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>
> >     > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >     >
> >     UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >     > http://www.freeswitch.org
> >     >
> >
> >     _______________________________________________
> >     FreeSWITCH-users mailing list
> >     FreeSWITCH-users at lists.freeswitch.org
> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>
> >     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >     UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >     http://www.freeswitch.org
> >
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> > Twitter: http://twitter.com/FreeSWITCH_wire
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> > <mailto:MSN%3Aanthony_minessale at hotmail.com<MSN%253Aanthony_minessale at hotmail.com>
> >
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> > <mailto:PAYPAL%3Aanthony.minessale at gmail.com<PAYPAL%253Aanthony.minessale at gmail.com>
> >
> > IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> > <mailto:sip%3A888 at conference.freeswitch.org<sip%253A888 at conference.freeswitch.org>
> >
> > iax:guest at conference.freeswitch.org/888
> > <http://iax:guest at conference.freeswitch.org/888>
> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org<googletalk%253Aconf%252B888 at conference.freeswitch.org>
> >
> > pstn:+19193869900
> > ------------------------------------------------------------------------
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100219/690877ed/attachment-0002.html 

From brian at freeswitch.org  Thu Feb 18 19:34:07 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 18 Feb 2010 21:34:07 -0600
Subject: [Freeswitch-users] 1.0.4 vs. trunk vs. bugs
In-Reply-To: <20100219025311.GA13893@jdc.jasonjgw.net>
References: <5d3e0dc61002181741r57b91f86h4810de63c1a3d46d@mail.gmail.com>
	<20100219015118.GA12983@jdc.jasonjgw.net>
	<DE4923F9-EEC2-4009-9A0D-D62C1CF3BD0F@kickasspixels.com>
	<20100219025311.GA13893@jdc.jasonjgw.net>
Message-ID: <98C2329F-D453-493E-9CC0-AA17DC023B7A@freeswitch.org>

The one thing I'm going to stop doing is helping anyone unless they write docs.  I can't count the times I have helped someone with the promise they would write docs.  They get what they want and move on... its selfish and rude.  The same goes for a stable branch... it seems its wanted/demanded but again nobody steps up to help with it.  I have been at this for over 14 hours today and yet still have a stack of stuff to do that keeps growing.

I thank each and everyone that helps out... I truly enjoy having you all involved.

Thanks,
Brian


On Feb 18, 2010, at 8:53 PM, Jason White wrote:

> I'm sure there are other models. My essential point is that, so far, the
> FreeSWITCH developers have not chosen to maintain long-lived "stable"
> branches, that there are alternatives to this approach, and that it has its
> downside, especially regarding the extra time/effort/work associated with
> maintaining it.




From brian at freeswitch.org  Thu Feb 18 19:36:57 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 18 Feb 2010 21:36:57 -0600
Subject: [Freeswitch-users] 1.0.4 vs. trunk vs. bugs
In-Reply-To: <5d3e0dc61002181853u2b5d7022ga79d41adaeb7ae68@mail.gmail.com>
References: <5d3e0dc61002181741r57b91f86h4810de63c1a3d46d@mail.gmail.com>
	<20100219015118.GA12983@jdc.jasonjgw.net>
	<DE4923F9-EEC2-4009-9A0D-D62C1CF3BD0F@kickasspixels.com>
	<865D1CDE-736D-4ED6-B7D6-94384143BCDA@freeswitch.org>
	<5d3e0dc61002181853u2b5d7022ga79d41adaeb7ae68@mail.gmail.com>
Message-ID: <32F19ACC-1D6B-4C94-BB2E-CB53FD143DC2@freeswitch.org>

We usually don't go to bed if its not fixed... I recall many times I was up at 2am dialing digits and testing while Anthony was coding the fix bugs... its part of what we do and shows just how dedicated we are to this project and how picky we are about the code.

/b

On Feb 18, 2010, at 8:53 PM, Lon Baker wrote:

> I'll drop it. You're right they don't last long. I didn't open bugs
> because I assumed the trunk is being worked in, so breakage is
> something I expect and anywhere from minutes to a few hours later they
> are gone.
> 
> Lon




From lon at kickasspixels.com  Thu Feb 18 20:00:09 2010
From: lon at kickasspixels.com (Kickass Pixels)
Date: Thu, 18 Feb 2010 20:00:09 -0800
Subject: [Freeswitch-users] 1.0.4 vs. trunk vs. bugs
In-Reply-To: <32F19ACC-1D6B-4C94-BB2E-CB53FD143DC2@freeswitch.org>
References: <5d3e0dc61002181741r57b91f86h4810de63c1a3d46d@mail.gmail.com>
	<20100219015118.GA12983@jdc.jasonjgw.net>
	<DE4923F9-EEC2-4009-9A0D-D62C1CF3BD0F@kickasspixels.com>
	<865D1CDE-736D-4ED6-B7D6-94384143BCDA@freeswitch.org>
	<5d3e0dc61002181853u2b5d7022ga79d41adaeb7ae68@mail.gmail.com>
	<32F19ACC-1D6B-4C94-BB2E-CB53FD143DC2@freeswitch.org>
Message-ID: <A10ABF8D-6900-49DE-9020-A7753C6A98BF@kickasspixels.com>

Brian,

Didn't intend to step on any toes. I do open bug reports, have  
submitted patches and have someone on my team who does contribute to  
the docs/wiki.

If the team wants to maintain a stable trunk I will try to find a team  
member to help manage it. If there is a philosophical reason for not  
doing it, I'm fine the current process. Whatever helps the dev team.

You guys do a great job and its a pleasure to work with freeswitch.

Lon

On Feb 18, 2010, at 7:36 PM, Brian West <brian at freeswitch.org> wrote:

> We usually don't go to bed if its not fixed... I recall many times I  
> was up at 2am dialing digits and testing while Anthony was coding  
> the fix bugs... its part of what we do and shows just how dedicated  
> we are to this project and how picky we are about the code.
>
> /b
>
> On Feb 18, 2010, at 8:53 PM, Lon Baker wrote:
>
>> I'll drop it. You're right they don't last long. I didn't open bugs
>> because I assumed the trunk is being worked in, so breakage is
>> something I expect and anywhere from minutes to a few hours later  
>> they
>> are gone.
>>
>> Lon
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org



From infos at madovsky.org  Thu Feb 18 20:08:30 2010
From: infos at madovsky.org (Madovsky)
Date: Thu, 18 Feb 2010 23:08:30 -0500
Subject: [Freeswitch-users] 1.0.4 vs. trunk vs. bugs
References: <5d3e0dc61002181741r57b91f86h4810de63c1a3d46d@mail.gmail.com><20100219015118.GA12983@jdc.jasonjgw.net><DE4923F9-EEC2-4009-9A0D-D62C1CF3BD0F@kickasspixels.com><20100219025311.GA13893@jdc.jasonjgw.net>
	<98C2329F-D453-493E-9CC0-AA17DC023B7A@freeswitch.org>
Message-ID: <FA6FB3864EE44C9581F0FD2A28738084@MOBILEE1705>


----- Original Message ----- 
From: "Brian West" <brian at freeswitch.org>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Thursday, February 18, 2010 10:34 PM
Subject: Re: [Freeswitch-users] 1.0.4 vs. trunk vs. bugs


> The one thing I'm going to stop doing is helping anyone unless they write 
> docs.  I can't count the times I have helped someone with the promise they 
> would write docs.  They get what they want and move on... its selfish and 
> rude.  The same goes for a stable branch... it seems its wanted/demanded 
> but again nobody steps up to help with it.  I have been at this for over 
> 14 hours today and yet still have a stack of stuff to do that keeps 
> growing.
>
> I thank each and everyone that helps out... I truly enjoy having you all 
> involved.
>
> Thanks,
> Brian
>
>
> On Feb 18, 2010, at 8:53 PM, Jason White wrote:
>
>> I'm sure there are other models. My essential point is that, so far, the
>> FreeSWITCH developers have not chosen to maintain long-lived "stable"
>> branches, that there are alternatives to this approach, and that it has 
>> its
>> downside, especially regarding the extra time/effort/work associated with
>> maintaining it.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


This kind of guy is called "leechers", since the internet exists there are 
leechers...
Don't be scared Brian, you are not alone in the same case,  i work also in 
programming and research 15 hours / days
since 8 years.... but I can say that Freeswitch has a very interesting wiki 
doc rather than other app ... 




From dave at 3c.co.uk  Thu Feb 18 20:10:05 2010
From: dave at 3c.co.uk (David Knell)
Date: Thu, 18 Feb 2010 21:10:05 -0700
Subject: [Freeswitch-users] 1.0.4 vs. trunk vs. bugs
In-Reply-To: <20100219015118.GA12983@jdc.jasonjgw.net>
References: <5d3e0dc61002181741r57b91f86h4810de63c1a3d46d@mail.gmail.com>
	<20100219015118.GA12983@jdc.jasonjgw.net>
Message-ID: <1266552605.7684.11.camel@local.freepabx.com>


> Lon Baker <lon at kickasspixels.com> wrote:
>  
> > The development branch is where feature requests and non-critical bugs
> > reports would be filed for the next production release.
> > 
> > The current process leaves a gap between production ready and
> > development code that may become greater over time.

Going against the grain here, I agree with you.  The current way of
doing things is, in my opinion, not well thought through - there's no
reason to tag and release versions if the answer to any issue is 'make
current', and support is not available unless that's been done.  Far
better to either have meaningful releases with stable and devel
branches, or not to have releases at all.

--Dave





From brian at freeswitch.org  Thu Feb 18 20:09:54 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 18 Feb 2010 22:09:54 -0600
Subject: [Freeswitch-users] 1.0.4 vs. trunk vs. bugs
In-Reply-To: <A10ABF8D-6900-49DE-9020-A7753C6A98BF@kickasspixels.com>
References: <5d3e0dc61002181741r57b91f86h4810de63c1a3d46d@mail.gmail.com>
	<20100219015118.GA12983@jdc.jasonjgw.net>
	<DE4923F9-EEC2-4009-9A0D-D62C1CF3BD0F@kickasspixels.com>
	<865D1CDE-736D-4ED6-B7D6-94384143BCDA@freeswitch.org>
	<5d3e0dc61002181853u2b5d7022ga79d41adaeb7ae68@mail.gmail.com>
	<32F19ACC-1D6B-4C94-BB2E-CB53FD143DC2@freeswitch.org>
	<A10ABF8D-6900-49DE-9020-A7753C6A98BF@kickasspixels.com>
Message-ID: <B4813D53-115C-4261-99EE-67CF06D2B945@freeswitch.org>


On Feb 18, 2010, at 10:00 PM, Kickass Pixels wrote:

> Brian,
> 
> Didn't intend to step on any toes. I do open bug reports, have  
> submitted patches and have someone on my team who does contribute to  
> the docs/wiki.

I thank them... I have to keep a close eye on the wiki because those viagra spammers hit it every now and again.  :P  So I know when and who does work pretty much real time... thanks to handy RSS feed of recent changes ;)

> If the team wants to maintain a stable trunk I will try to find a team  
> member to help manage it. If there is a philosophical reason for not  
> doing it, I'm fine the current process. Whatever helps the dev team.

If you want to assemble a team to help manage a "stable" branch then we'll allow you to do so.  I even welcome it if you want to help with this.  Its the only reason we aren't able to do it we don't have enough bodies to keep everything going.  ;)


> You guys do a great job and its a pleasure to work with freeswitch.

You too... keep up the good work... its these kinds of exchanges that bring about great ideas and change.

> 
> Lon




From dave at 3c.co.uk  Thu Feb 18 20:14:40 2010
From: dave at 3c.co.uk (David Knell)
Date: Thu, 18 Feb 2010 21:14:40 -0700
Subject: [Freeswitch-users] voivemail quality
In-Reply-To: <b39273531002181801kd81a6b2x3327be4887563eae@mail.gmail.com>
References: <27648642.post@talk.nabble.com>
	<b39273531002181801kd81a6b2x3327be4887563eae@mail.gmail.com>
Message-ID: <1266552880.7684.17.camel@local.freepabx.com>

Rupa -

Howler filled a big hole that we've been promised is going to be filled
for the last two years, but said filling is still not available.  That's
a significant risk and contribution on their part.

And the request is as to whether others have experienced similar
problems, which strikes me as completely reasonable on a users mailing
list.  If you can't help Edward, then I'd suggest you keep quiet.

--Dave

Rupa wrote:
> Have you asked Howler about this?  This is not a support channel for
> commercial software that doesn't participate or contribute in the
> community.
> 
> On Thu, Feb 18, 2010 at 7:21 PM, Edward Stevenson
> <iamcanadian at myfastmail.com> wrote:
>         
>         
>         I have V1.0.4 running on a test/production server.  It's
>         working quite well,
>         except for voicemail retrieval over a satellite internet
>         connection.  Voice
>         calls over satellite sound fine, other than the 600ms delay.
>          I'm using the
>         Howler Tech G729 module for G729 transcoding.
>         
>         I've noticed that in voice calls, the bandwidth over the
>         satellite is a
>         steady 24 kbps in both directions.  When accessing voicemail,
>         the bandwidth
>         fluctuates with voice in the call.  What I mean by that is,
>         pauses in audio
>         in the voicemail, or in the IVR, cause the bandwidth of the
>         call to drop off
>         momentarily.  It's almost like it's using a variable bit
>         rate.  The audio
>         sounds like there's packet loss.  Pops and garbled speach.
>          This is not
>         noticable over a land internet connection.  I'm using the
>         built in
>         voicemail.
>         
>         If I change the phone's codec to G711, the call is perfectly
>         clear.  Perhaps
>         the Howler module doesn't like transcoding from the L16 codec
>         that
>         Freeswitch seems to use to play the wav file?
>         
>         Anyone else had any similar issues?
>         --
>         View this message in context:
>         http://old.nabble.com/voivemail-quality-tp27648642p27648642.html
>         Sent from the Freeswitch-users mailing list archive at
>         Nabble.com.
>         
>         
>         _______________________________________________
>         FreeSWITCH-users mailing list
>         FreeSWITCH-users at lists.freeswitch.org
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         http://www.freeswitch.org
>         
> 
> 
> 
> -- 
> -Rupa
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From brian at freeswitch.org  Thu Feb 18 20:16:40 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 18 Feb 2010 22:16:40 -0600
Subject: [Freeswitch-users] 1.0.4 vs. trunk vs. bugs
In-Reply-To: <1266552605.7684.11.camel@local.freepabx.com>
References: <5d3e0dc61002181741r57b91f86h4810de63c1a3d46d@mail.gmail.com>
	<20100219015118.GA12983@jdc.jasonjgw.net>
	<1266552605.7684.11.camel@local.freepabx.com>
Message-ID: <616F57B5-182A-49A9-9825-2E5B72F5DB05@freeswitch.org>

OK so I can sign you up for the stable team?  ;)  As per my previous email i'm 100% sure we would do a stable release if we had people tending to issues.  The only problem is you would have to be on IRC tending to issues because if tony sees someone asking about a problem he'll be diving in to fix it before they can say "I have this one".  This also means working in a similar manner we do already.  Our process is very chaotic at times but it has served us well so far.  

The goal is to leave Anthony alone so he can move forward and let the stable team manage the jira's and issues on the list related to stable.

/b

On Feb 18, 2010, at 10:10 PM, David Knell wrote:

>> 
>> Lon Baker <lon at kickasspixels.com> wrote:
>> 
>>> The development branch is where feature requests and non-critical bugs
>>> reports would be filed for the next production release.
>>> 
>>> The current process leaves a gap between production ready and
>>> development code that may become greater over time.
> 
> Going against the grain here, I agree with you.  The current way of
> doing things is, in my opinion, not well thought through - there's no
> reason to tag and release versions if the answer to any issue is 'make
> current', and support is not available unless that's been done.  Far
> better to either have meaningful releases with stable and devel
> branches, or not to have releases at all.
> 
> --Dave

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100218/70f07be6/attachment-0002.html 

From brian at freeswitch.org  Thu Feb 18 20:21:11 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 18 Feb 2010 22:21:11 -0600
Subject: [Freeswitch-users] voivemail quality
In-Reply-To: <1266552880.7684.17.camel@local.freepabx.com>
References: <27648642.post@talk.nabble.com>
	<b39273531002181801kd81a6b2x3327be4887563eae@mail.gmail.com>
	<1266552880.7684.17.camel@local.freepabx.com>
Message-ID: <05F7797B-510C-431B-B78B-D1AB58A81B37@freeswitch.org>

We have g729 ready and tested... We just have to flip the switch and write a check.  I have personally been working on this.  The installer and testing of the license server and various registration details isn't a trivial task.  

I have been asking for testers over the past few weeks to flush out issues so we can lower our support overhead as our resources are already taxed to their limits.  I thank everyone that has been involved in testing.. our first g729 release should be next week for the G729  codec at $10.00 per channel direct from our website.

Thanks,
Brian

On Feb 18, 2010, at 10:14 PM, David Knell wrote:

> Rupa -
> 
> Howler filled a big hole that we've been promised is going to be filled
> for the last two years, but said filling is still not available.  That's
> a significant risk and contribution on their part.
> 
> And the request is as to whether others have experienced similar
> problems, which strikes me as completely reasonable on a users mailing
> list.  If you can't help Edward, then I'd suggest you keep quiet.
> 
> --Dave




From jaybinks at gmail.com  Thu Feb 18 20:26:24 2010
From: jaybinks at gmail.com (jay binks)
Date: Fri, 19 Feb 2010 14:26:24 +1000
Subject: [Freeswitch-users] voivemail quality
In-Reply-To: <1266552880.7684.17.camel@local.freepabx.com>
References: <27648642.post@talk.nabble.com>
	<b39273531002181801kd81a6b2x3327be4887563eae@mail.gmail.com> 
	<1266552880.7684.17.camel@local.freepabx.com>
Message-ID: <ec2ce17e1002182026m5986ab85t71006d608af73f88@mail.gmail.com>

I think suggesting the user test with other codecs is very helpful..

your attacking rupa  for making suggestions about where the issue is
and what to do in order to resolve it how does that help ??

pot, kettle, black ???

J

On Fri, Feb 19, 2010 at 2:14 PM, David Knell <dave at 3c.co.uk> wrote:

> Rupa -
>
> Howler filled a big hole that we've been promised is going to be filled
> for the last two years, but said filling is still not available.  That's
> a significant risk and contribution on their part.
>
> And the request is as to whether others have experienced similar
> problems, which strikes me as completely reasonable on a users mailing
> list.  If you can't help Edward, then I'd suggest you keep quiet.
>
> --Dave
>
> Rupa wrote:
> > Have you asked Howler about this?  This is not a support channel for
> > commercial software that doesn't participate or contribute in the
> > community.
> >
> > On Thu, Feb 18, 2010 at 7:21 PM, Edward Stevenson
> > <iamcanadian at myfastmail.com> wrote:
> >
> >
> >         I have V1.0.4 running on a test/production server.  It's
> >         working quite well,
> >         except for voicemail retrieval over a satellite internet
> >         connection.  Voice
> >         calls over satellite sound fine, other than the 600ms delay.
> >          I'm using the
> >         Howler Tech G729 module for G729 transcoding.
> >
> >         I've noticed that in voice calls, the bandwidth over the
> >         satellite is a
> >         steady 24 kbps in both directions.  When accessing voicemail,
> >         the bandwidth
> >         fluctuates with voice in the call.  What I mean by that is,
> >         pauses in audio
> >         in the voicemail, or in the IVR, cause the bandwidth of the
> >         call to drop off
> >         momentarily.  It's almost like it's using a variable bit
> >         rate.  The audio
> >         sounds like there's packet loss.  Pops and garbled speach.
> >          This is not
> >         noticable over a land internet connection.  I'm using the
> >         built in
> >         voicemail.
> >
> >         If I change the phone's codec to G711, the call is perfectly
> >         clear.  Perhaps
> >         the Howler module doesn't like transcoding from the L16 codec
> >         that
> >         Freeswitch seems to use to play the wav file?
> >
> >         Anyone else had any similar issues?
> >         --
> >         View this message in context:
> >         http://old.nabble.com/voivemail-quality-tp27648642p27648642.html
> >         Sent from the Freeswitch-users mailing list archive at
> >         Nabble.com.
> >
> >
> >         _______________________________________________
> >         FreeSWITCH-users mailing list
> >         FreeSWITCH-users at lists.freeswitch.org
> >         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >         UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >         http://www.freeswitch.org
> >
> >
> >
> >
> > --
> > -Rupa
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely

Jay
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100219/260cddfc/attachment-0002.html 

From brian at freeswitch.org  Thu Feb 18 20:32:13 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 18 Feb 2010 22:32:13 -0600
Subject: [Freeswitch-users] voivemail quality
In-Reply-To: <ec2ce17e1002182026m5986ab85t71006d608af73f88@mail.gmail.com>
References: <27648642.post@talk.nabble.com>
	<b39273531002181801kd81a6b2x3327be4887563eae@mail.gmail.com>
	<1266552880.7684.17.camel@local.freepabx.com>
	<ec2ce17e1002182026m5986ab85t71006d608af73f88@mail.gmail.com>
Message-ID: <C8ECE002-AA2B-4E3B-AEEA-0FE92222F48C@freeswitch.org>

NOW NOW everyone just needs to get their panties out of a bunch.  All was helpful... We are all here to accomplish the same thing and by working together more we can accomplish these tasks.

After this friday Meeting our weekly meetings will be moved to wednesdays.  We have a few things to go over and some of them are all related to what happens when 1.0.5 is tagged.

/b

On Feb 18, 2010, at 10:26 PM, jay binks wrote:

> I think suggesting the user test with other codecs is very helpful..
> 
> your attacking rupa  for making suggestions about where the issue is
> and what to do in order to resolve it how does that help ??
> 
> pot, kettle, black ???
> 
> J




From dave at 3c.co.uk  Thu Feb 18 20:53:29 2010
From: dave at 3c.co.uk (David Knell)
Date: Thu, 18 Feb 2010 21:53:29 -0700
Subject: [Freeswitch-users] voivemail quality
In-Reply-To: <ec2ce17e1002182026m5986ab85t71006d608af73f88@mail.gmail.com>
References: <27648642.post@talk.nabble.com>
	<b39273531002181801kd81a6b2x3327be4887563eae@mail.gmail.com>
	<1266552880.7684.17.camel@local.freepabx.com>
	<ec2ce17e1002182026m5986ab85t71006d608af73f88@mail.gmail.com>
Message-ID: <1266555209.7684.23.camel@local.freepabx.com>


On Fri, 2010-02-19 at 14:26 +1000, jay binks wrote:
> I think suggesting the user test with other codecs is very helpful..

Edward already said that things worked with G.711 - that ought to be
enough testing with other codecs.

> your attacking rupa  for making suggestions about where the issue is
> and what to do in order to resolve it how does that help ??

It doesn't, directly.  And, had Rupa simply asked if the OP had raised
the issue with Howler - which is a completely valid thing to do,
particularly with the evidence in front of him - that'd have been one
thing.  But he didn't.

> pot, kettle, black ???

How so, exactly?

--Dave

> 
> 
> J
> 
> On Fri, Feb 19, 2010 at 2:14 PM, David Knell <dave at 3c.co.uk> wrote:
>         Rupa -
>         
>         Howler filled a big hole that we've been promised is going to
>         be filled
>         for the last two years, but said filling is still not
>         available.  That's
>         a significant risk and contribution on their part.
>         
>         And the request is as to whether others have experienced
>         similar
>         problems, which strikes me as completely reasonable on a users
>         mailing
>         list.  If you can't help Edward, then I'd suggest you keep
>         quiet.
>         
>         --Dave
>         
>         
>         Rupa wrote:
>         > Have you asked Howler about this?  This is not a support
>         channel for
>         > commercial software that doesn't participate or contribute
>         in the
>         > community.
>         >
>         > On Thu, Feb 18, 2010 at 7:21 PM, Edward Stevenson
>         > <iamcanadian at myfastmail.com> wrote:
>         >
>         >
>         >         I have V1.0.4 running on a test/production server.
>          It's
>         >         working quite well,
>         >         except for voicemail retrieval over a satellite
>         internet
>         >         connection.  Voice
>         >         calls over satellite sound fine, other than the
>         600ms delay.
>         >          I'm using the
>         >         Howler Tech G729 module for G729 transcoding.
>         >
>         >         I've noticed that in voice calls, the bandwidth over
>         the
>         >         satellite is a
>         >         steady 24 kbps in both directions.  When accessing
>         voicemail,
>         >         the bandwidth
>         >         fluctuates with voice in the call.  What I mean by
>         that is,
>         >         pauses in audio
>         >         in the voicemail, or in the IVR, cause the bandwidth
>         of the
>         >         call to drop off
>         >         momentarily.  It's almost like it's using a variable
>         bit
>         >         rate.  The audio
>         >         sounds like there's packet loss.  Pops and garbled
>         speach.
>         >          This is not
>         >         noticable over a land internet connection.  I'm
>         using the
>         >         built in
>         >         voicemail.
>         >
>         >         If I change the phone's codec to G711, the call is
>         perfectly
>         >         clear.  Perhaps
>         >         the Howler module doesn't like transcoding from the
>         L16 codec
>         >         that
>         >         Freeswitch seems to use to play the wav file?
>         >
>         >         Anyone else had any similar issues?
>         >         --
>         >         View this message in context:
>         >
>         http://old.nabble.com/voivemail-quality-tp27648642p27648642.html
>         >         Sent from the Freeswitch-users mailing list archive
>         at
>         >         Nabble.com.
>         >
>         >
>         >         _______________________________________________
>         >         FreeSWITCH-users mailing list
>         >         FreeSWITCH-users at lists.freeswitch.org
>         >
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         >
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         >         http://www.freeswitch.org
>         >
>         >
>         >
>         >
>         > --
>         > -Rupa
>         > _______________________________________________
>         > FreeSWITCH-users mailing list
>         > FreeSWITCH-users at lists.freeswitch.org
>         >
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         >
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         > http://www.freeswitch.org
>         
>         
>         _______________________________________________
>         FreeSWITCH-users mailing list
>         FreeSWITCH-users at lists.freeswitch.org
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         http://www.freeswitch.org
>         
> 
> 
> 
> -- 
> Sincerely
> 
> Jay
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From yehavi.bourvine at gmail.com  Thu Feb 18 21:01:27 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Fri, 19 Feb 2010 07:01:27 +0200
Subject: [Freeswitch-users] Announcement: FreeSWITCH Add Broadsoft SCA
	Support
In-Reply-To: <ADC8B813-4161-481C-877E-93FC012858B4@freeswitch.org>
References: <87f2f3b91001111742k691966cfs82515db77c13d4d4@mail.gmail.com>
	<d361ca7b1002090838j6c42dcd2of8affb5f281900fd@mail.gmail.com>
	<767B677E-0D19-43A9-89DE-D96EC4A715EC@freeswitch.org>
	<8D062B6E-4691-4BA6-B038-0EC6E8642DDA@jerris.com>
	<9D328485-D4C3-4B14-808A-CB63BBF06DC1@freeswitch.org>
	<d361ca7b1002170549g608d6e16vb2eb3f087fff9752@mail.gmail.com>
	<191c3a031002170556p5dfd4c55le3c279f194808d92@mail.gmail.com>
	<191c3a031002170557g5d8c9df9xf40d1b74b9a6f19d@mail.gmail.com>
	<d361ca7b1002170700x15630ab2tabc6f3d91ebc3303@mail.gmail.com>
	<ADC8B813-4161-481C-877E-93FC012858B4@freeswitch.org>
Message-ID: <d361ca7b1002182101p60b4e8f8hf4dc95e1d2251e30@mail.gmail.com>

Thanks Brian. It now works better, but not fully (using 16659M).

What happens is:

   - When one of the Polycoms seize the line it is ok - the other phone gets
   notification and the extension status is "in use".
   - When one of the Polycom phones initiates a call - all is ok:
      - The other side sees that the extension is in use.
      - When it is put to hold all phones who share this extension see it
      and can pick the call.
   -  When a call arrives, both ring; the one that did not answer gets only
   a cancel mesage *without *any further notification that the extension is
   in use by the other phone.

                         Thanks! __Yehavi:

2010/2/17 Brian West <brian at freeswitch.org>

> Step 1. Enable manage-shared-appearance=true
>
> Step 2. Now in the phone's config Configure the phone as usually, set the
> line shared and DO NOT set the third party name.
>
> Step 3. Reboot
>
> It should work.
>
> I wish someone that has this working would write some wiki docs these
> threads about it not working are getting rather old when I know for a fact
> they work fine.
>
> The gateway info missing is a gateway you have configured getting a notify.
>  It has nothing to do with SCA.
>
> /b
>
> On Feb 17, 2010, at 9:00 AM, Yehavi Bourvine wrote:
>
> > .
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100219/cd5b84dc/attachment-0002.html 

From brian at freeswitch.org  Thu Feb 18 21:09:35 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 18 Feb 2010 23:09:35 -0600
Subject: [Freeswitch-users] Announcement: FreeSWITCH Add Broadsoft SCA
	Support
In-Reply-To: <d361ca7b1002182101p60b4e8f8hf4dc95e1d2251e30@mail.gmail.com>
References: <87f2f3b91001111742k691966cfs82515db77c13d4d4@mail.gmail.com>
	<d361ca7b1002090838j6c42dcd2of8affb5f281900fd@mail.gmail.com>
	<767B677E-0D19-43A9-89DE-D96EC4A715EC@freeswitch.org>
	<8D062B6E-4691-4BA6-B038-0EC6E8642DDA@jerris.com>
	<9D328485-D4C3-4B14-808A-CB63BBF06DC1@freeswitch.org>
	<d361ca7b1002170549g608d6e16vb2eb3f087fff9752@mail.gmail.com>
	<191c3a031002170556p5dfd4c55le3c279f194808d92@mail.gmail.com>
	<191c3a031002170557g5d8c9df9xf40d1b74b9a6f19d@mail.gmail.com>
	<d361ca7b1002170700x15630ab2tabc6f3d91ebc3303@mail.gmail.com>
	<ADC8B813-4161-481C-877E-93FC012858B4@freeswitch.org>
	<d361ca7b1002182101p60b4e8f8hf4dc95e1d2251e30@mail.gmail.com>
Message-ID: <C577F89F-4621-4DE9-A62B-BD40D1235538@freeswitch.org>

That last bit is wrong... I need sip traces.

/b

On Feb 18, 2010, at 11:01 PM, Yehavi Bourvine wrote:

> Thanks Brian. It now works better, but not fully (using 16659M).
>  
> What happens is:
> When one of the Polycoms seize the line it is ok - the other phone gets notification and the extension status is "in use".
> When one of the Polycom phones initiates a call - all is ok:
> The other side sees that the extension is in use.
> When it is put to hold all phones who share this extension see it and can pick the call.
>  When a call arrives, both ring; the one that did not answer gets only a cancel mesage without any further notification that the extension is in use by the other phone.
>                          Thanks! __Yehavi:

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100218/b2902689/attachment-0002.html 

From yehavi.bourvine at gmail.com  Thu Feb 18 21:41:43 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Fri, 19 Feb 2010 07:41:43 +0200
Subject: [Freeswitch-users] SQL ERR: library routine called out of
	sequence
In-Reply-To: <191c3a031002161230h2b6e6f9wf3f65ddb71f08297@mail.gmail.com>
References: <d361ca7b1002160656w7d616f6dn5e80087ebde94dbe@mail.gmail.com>
	<191c3a031002160705w4050c3cdk2bf2922e9aff5ba3@mail.gmail.com>
	<d361ca7b1002160727ucb6e0c8id62e803757333213@mail.gmail.com>
	<191c3a031002160755q569fe2ednfefb01b6d9cf7f0e@mail.gmail.com>
	<d361ca7b1002161059w1ef9484dr87e69a33aa2b62fc@mail.gmail.com>
	<191c3a031002161110s637a48a2r717ae94557fe9e9a@mail.gmail.com>
	<d361ca7b1002161130k3eb2d88by3100b30a9d29bd51@mail.gmail.com>
	<191c3a031002161230h2b6e6f9wf3f65ddb71f08297@mail.gmail.com>
Message-ID: <d361ca7b1002182141p53e3884fs2a9993d4739a1f91@mail.gmail.com>

Unfortunately it did not help. I still get these error messages.

Just for the record, I have two systems:

   - Production one which is running Fedora-10 and exibits this problem.
   - Test system which is running Fedora-12 and does not exibit this
   phenomenon; however, there is almost zero traffic on this system.


I plan in upgrading the production system to Fedora-12, but it is not that
trivial...

                   Thanks, __Yehavi:

 2010/2/16 Anthony Minessale <anthony.minessale at gmail.com>

> Strange, even on abusive testing we have not seen this problem.
>
> please update to latest trunk.
> There was only one change I can think of that may cause your issue and I
> added a patch for it.
> If it persists try setting the sql-in-transactions profile param to false.
>
>
>
>
>
> On Tue, Feb 16, 2010 at 1:30 PM, Yehavi Bourvine <
> yehavi.bourvine at gmail.com> wrote:
>
>>  Most of the queries are ok, only some fail, thus it doesn't look like
>> permission problem. Furthermore, under 1.0.5pre10 it works for months.
>>
>> Might it be thread unsafe function calls? I've found the following while
>> searching the WEB:
>>
>> *According to the MSDN docs, System.Timers.Timer operates in a thread
>> pool.  If that's the case, your code is breaking the "connections cannot be
>> shared across threads" rule for SQLit*
>>
>> Although it quotes MSDN, it might be related to Linux as well.
>>
>>                     Thanks, __Yehavi:
>>
>>  2010/2/16 Anthony Minessale <anthony.minessale at gmail.com>
>>
>>> That sounds about right.
>>>
>>> That error usually has something to do with using db calls on a closed
>>> file or something along those lines.
>>> Maybe you have a permission problem on the directory where the db files
>>> are?
>>>
>>>
>>>
>>> On Tue, Feb 16, 2010 at 12:59 PM, Yehavi Bourvine <
>>> yehavi.bourvine at gmail.com> wrote:
>>>
>>>>  The OS is Fedora-10 (soon to be upgraded to 12).
>>>>
>>>> What I do when I want to test a new version:
>>>>
>>>>    - Download the latest one into a fresh directory
>>>>    - bootstrap.sh, configure and make
>>>>    - stop Freeswitch, delete everything in lib, mod, bin ,db
>>>>    - make install and run it.
>>>>
>>>>
>>>> Is there additional place to clean?
>>>>
>>>>                         Thanks! __Yehavi:
>>>>
>>>>  2010/2/16 Anthony Minessale <anthony.minessale at gmail.com>
>>>>
>>>>> you may want to do a clean wipe of all files related to FS then.
>>>>> you clearly have some problem with legacy something or other because we
>>>>> don't see that on dozens of dev boxes.
>>>>>
>>>>> What os is it?
>>>>>
>>>>>
>>>>>
>>>>> On Tue, Feb 16, 2010 at 9:27 AM, Yehavi Bourvine <
>>>>> yehavi.bourvine at gmail.com> wrote:
>>>>>
>>>>>>  Tried this, but it didn't help. I delete these DB files before any
>>>>>> upgrade just to be sure.
>>>>>>
>>>>>>                   Thanks! __Yehavi:
>>>>>>
>>>>>>  2010/2/16 Anthony Minessale <anthony.minessale at gmail.com>
>>>>>>
>>>>>>> try removing all the .db files from /usr/local/freeswitch/db
>>>>>>>
>>>>>>>
>>>>>>>   On Tue, Feb 16, 2010 at 8:56 AM, Yehavi Bourvine <
>>>>>>> yehavi.bourvine at gmail.com> wrote:
>>>>>>>
>>>>>>>>   Hello,
>>>>>>>>
>>>>>>>>   After upgrading from 1.0.5pre10 to 1.0.5-20100216-0400 we started
>>>>>>>> getting the above errors (I append bellow two samples). It seems Freeswitch
>>>>>>>> fails to read a database using Sqlite.
>>>>>>>> Anyone have seen this?
>>>>>>>>
>>>>>>>> Other details: Fedora 10, SQlite 3.5.9.
>>>>>>>> We also do SQLite quesries during call setup via LUA from CoreDB. Is
>>>>>>>> it an SQLite problem?
>>>>>>>>
>>>>>>>>                          Thanks! __Yehavi:
>>>>>>>>
>>>>>>>> The samples:
>>>>>>>> 2010-02-16 16:22:13.762679 [ERR] switch_core_sqldb.c:404 SQL ERR
>>>>>>>> [library routin
>>>>>>>> e called out of sequence]
>>>>>>>> delete from sip_dialogs where call_id='
>>>>>>>> 8656841832142-120172129116107 at 10.64.1.2'
>>>>>>>>
>>>>>>>> 2010-02-16 16:42:00.802442 [ERR] switch_core_sqldb.c:722 SQL ERR:
>>>>>>>> [select call_i
>>>>>>>>
>>>>>>>> d,sip_user,sip_host,contact,status,rpid,expires,user_agent,server_user,server_ho
>>>>>>>>
>>>>>>>> st,profile_name,hostname,network_ip,network_port,sip_username,sip_realm,mwi_user
>>>>>>>> ,mwi_host from sip_registrations where profile_name='phones' and
>>>>>>>> contact like '%
>>>>>>>> 80635%'] library routine called out of sequence
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> --
>>>>>>> Anthony Minessale II
>>>>>>>
>>>>>>> FreeSWITCH http://www.freeswitch.org/
>>>>>>> ClueCon http://www.cluecon.com/
>>>>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>>>>
>>>>>>> AIM: anthm
>>>>>>> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
>>>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>>>>> IRC: irc.freenode.net #freeswitch
>>>>>>>
>>>>>>> FreeSWITCH Developer Conference
>>>>>>> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>>>>>>> iax:guest at conference.freeswitch.org/888
>>>>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>>>>> pstn:+19193869900
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> Anthony Minessale II
>>>>>
>>>>> FreeSWITCH http://www.freeswitch.org/
>>>>> ClueCon http://www.cluecon.com/
>>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>>
>>>>> AIM: anthm
>>>>> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>>> IRC: irc.freenode.net #freeswitch
>>>>>
>>>>> FreeSWITCH Developer Conference
>>>>> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>>>>> iax:guest at conference.freeswitch.org/888
>>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>>> pstn:+19193869900
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>> iax:guest at conference.freeswitch.org/888
>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> pstn:+19193869900
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100219/3124d299/attachment-0002.html 

From gkuri at ieee.org  Thu Feb 18 21:44:31 2010
From: gkuri at ieee.org (Gabriel Kuri)
Date: Thu, 18 Feb 2010 21:44:31 -0800
Subject: [Freeswitch-users] Announcement: FreeSWITCH Add Broadsoft SCA
	Support
In-Reply-To: <d361ca7b1002182101p60b4e8f8hf4dc95e1d2251e30@mail.gmail.com>
References: <87f2f3b91001111742k691966cfs82515db77c13d4d4@mail.gmail.com>
	<767B677E-0D19-43A9-89DE-D96EC4A715EC@freeswitch.org>
	<8D062B6E-4691-4BA6-B038-0EC6E8642DDA@jerris.com>
	<9D328485-D4C3-4B14-808A-CB63BBF06DC1@freeswitch.org>
	<d361ca7b1002170549g608d6e16vb2eb3f087fff9752@mail.gmail.com>
	<191c3a031002170556p5dfd4c55le3c279f194808d92@mail.gmail.com>
	<191c3a031002170557g5d8c9df9xf40d1b74b9a6f19d@mail.gmail.com>
	<d361ca7b1002170700x15630ab2tabc6f3d91ebc3303@mail.gmail.com>
	<ADC8B813-4161-481C-877E-93FC012858B4@freeswitch.org>
	<d361ca7b1002182101p60b4e8f8hf4dc95e1d2251e30@mail.gmail.com>
Message-ID: <8b1c9cda1002182144m65ccf979w71384a1e7a55e2ea@mail.gmail.com>

>  When a call arrives, both ring; the one that did not answer gets only a
> cancel mesage without any further notification that the extension is in use
> by the other phone.

These are the same exact symptoms I posted about earlier this week,
with the Cisco SPA-5xx series phones. I still have yet to figure out
why this is happening, if you find out what's going on, please post
back the solution, I'd like to know the resolution.

Thanks,
Gabe



On Thu, Feb 18, 2010 at 9:01 PM, Yehavi Bourvine
<yehavi.bourvine at gmail.com> wrote:
> Thanks Brian. It now works better, but not fully (using 16659M).
>
> What happens is:
>
> When one of the Polycoms seize the line it is ok?- the other phone gets
> notification and the extension status is "in use".
> When?one of the Polycom phones initiates a call - all is ok:
>
> The other side sees that the extension is in use.
> When it is put to hold all phones?who share this extension see it and can
> pick the call.
>

>
> ???????????????????????? Thanks! __Yehavi:
>
> 2010/2/17 Brian West <brian at freeswitch.org>
>>
>> Step 1. Enable manage-shared-appearance=true
>>
>> Step 2. Now in the phone's config Configure the phone as usually, set the
>> line shared and DO NOT set the third party name.
>>
>> Step 3. Reboot
>>
>> It should work.
>>
>> I wish someone that has this working would write some wiki docs these
>> threads about it not working are getting rather old when I know for a fact
>> they work fine.
>>
>> The gateway info missing is a gateway you have configured getting a
>> notify. ?It has nothing to do with SCA.
>>
>> /b
>>
>> On Feb 17, 2010, at 9:00 AM, Yehavi Bourvine wrote:
>>
>> > .
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From yehavi.bourvine at gmail.com  Thu Feb 18 21:54:12 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Fri, 19 Feb 2010 07:54:12 +0200
Subject: [Freeswitch-users] Announcement: FreeSWITCH Add Broadsoft SCA
	Support
In-Reply-To: <8b1c9cda1002182144m65ccf979w71384a1e7a55e2ea@mail.gmail.com>
References: <87f2f3b91001111742k691966cfs82515db77c13d4d4@mail.gmail.com>
	<8D062B6E-4691-4BA6-B038-0EC6E8642DDA@jerris.com>
	<9D328485-D4C3-4B14-808A-CB63BBF06DC1@freeswitch.org>
	<d361ca7b1002170549g608d6e16vb2eb3f087fff9752@mail.gmail.com>
	<191c3a031002170556p5dfd4c55le3c279f194808d92@mail.gmail.com>
	<191c3a031002170557g5d8c9df9xf40d1b74b9a6f19d@mail.gmail.com>
	<d361ca7b1002170700x15630ab2tabc6f3d91ebc3303@mail.gmail.com>
	<ADC8B813-4161-481C-877E-93FC012858B4@freeswitch.org>
	<d361ca7b1002182101p60b4e8f8hf4dc95e1d2251e30@mail.gmail.com>
	<8b1c9cda1002182144m65ccf979w71384a1e7a55e2ea@mail.gmail.com>
Message-ID: <d361ca7b1002182154o52352c62h8c0ec02fa66e8444@mail.gmail.com>

Hello Gabe,

  As you can see - Brian is actively investigating it, so you can expect for
some fix soon...

                       Regards, __Yehavi:

2010/2/19 Gabriel Kuri <gkuri at ieee.org>

> >  When a call arrives, both ring; the one that did not answer gets only a
> > cancel mesage without any further notification that the extension is in
> use
> > by the other phone.
>
> These are the same exact symptoms I posted about earlier this week,
> with the Cisco SPA-5xx series phones. I still have yet to figure out
> why this is happening, if you find out what's going on, please post
> back the solution, I'd like to know the resolution.
>
> Thanks,
> Gabe
>
>
>
> On Thu, Feb 18, 2010 at 9:01 PM, Yehavi Bourvine
> <yehavi.bourvine at gmail.com> wrote:
> > Thanks Brian. It now works better, but not fully (using 16659M).
> >
> > What happens is:
> >
> > When one of the Polycoms seize the line it is ok - the other phone gets
> > notification and the extension status is "in use".
> > When one of the Polycom phones initiates a call - all is ok:
> >
> > The other side sees that the extension is in use.
> > When it is put to hold all phones who share this extension see it and can
> > pick the call.
> >
>
> >
>  >                          Thanks! __Yehavi:
> >
> > 2010/2/17 Brian West <brian at freeswitch.org>
> >>
> >> Step 1. Enable manage-shared-appearance=true
> >>
> >> Step 2. Now in the phone's config Configure the phone as usually, set
> the
> >> line shared and DO NOT set the third party name.
> >>
> >> Step 3. Reboot
> >>
> >> It should work.
> >>
> >> I wish someone that has this working would write some wiki docs these
> >> threads about it not working are getting rather old when I know for a
> fact
> >> they work fine.
> >>
> >> The gateway info missing is a gateway you have configured getting a
> >> notify.  It has nothing to do with SCA.
> >>
> >> /b
> >>
> >> On Feb 17, 2010, at 9:00 AM, Yehavi Bourvine wrote:
> >>
> >> > .
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100219/bec80318/attachment-0002.html 

From pmhshz at gmail.com  Thu Feb 18 22:09:11 2010
From: pmhshz at gmail.com (MohammedShehzad)
Date: Fri, 19 Feb 2010 11:39:11 +0530
Subject: [Freeswitch-users] stream a file multicast with mod_esf
In-Reply-To: <85C74329-0D7C-48B2-B7A8-D4AA67C6E3F2@freeswitch.org>
References: <B98B6921358D0C49BD21CD023F1F52C302623B4B@hbghs04a.ww300.siemens.net>
	<B98B6921358D0C49BD21CD023F1F52C302623BE0@hbghs04a.ww300.siemens.net>
	<7C996136-963A-42D2-B0FE-D2729F968E52@freeswitch.org>
	<27071973.post@talk.nabble.com>
	<378CDEC3-5A41-4EF9-927A-311F2531E6AB@freeswitch.org>
	<a792304b1001080725y19ff28camd37425927550f311@mail.gmail.com>
	<191c3a031001081848w9036207yf6a4b473ec233243@mail.gmail.com>
	<85C74329-0D7C-48B2-B7A8-D4AA67C6E3F2@freeswitch.org>
Message-ID: <a792304b1002182209m43ada13cp8ed3d91ad509f0a2@mail.gmail.com>

Hello,
I am now looking into the code from few days to create the RTP Multicast
Listener first.
Is this something similar coding as mod_esf need to do here to listen on the
Multicast IP & Port? Or I need to use combination some rtp related function
of (switch_rtp.c) here?

I think this not proper place to discuss here, let me know where should I
continue further discussion.

-MohammedShehzad
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100219/c679aed0/attachment-0002.html 

From anthony.minessale at gmail.com  Thu Feb 18 22:15:30 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 19 Feb 2010 00:15:30 -0600
Subject: [Freeswitch-users] SQL ERR: library routine called out of
	sequence
In-Reply-To: <191c3a031002182214h614c8c4aladed66943939a3bf@mail.gmail.com>
References: <d361ca7b1002160656w7d616f6dn5e80087ebde94dbe@mail.gmail.com>
	<191c3a031002160705w4050c3cdk2bf2922e9aff5ba3@mail.gmail.com>
	<d361ca7b1002160727ucb6e0c8id62e803757333213@mail.gmail.com>
	<191c3a031002160755q569fe2ednfefb01b6d9cf7f0e@mail.gmail.com>
	<d361ca7b1002161059w1ef9484dr87e69a33aa2b62fc@mail.gmail.com>
	<191c3a031002161110s637a48a2r717ae94557fe9e9a@mail.gmail.com>
	<d361ca7b1002161130k3eb2d88by3100b30a9d29bd51@mail.gmail.com>
	<191c3a031002161230h2b6e6f9wf3f65ddb71f08297@mail.gmail.com>
	<d361ca7b1002182141p53e3884fs2a9993d4739a1f91@mail.gmail.com>
	<191c3a031002182214h614c8c4aladed66943939a3bf@mail.gmail.com>
Message-ID: <191c3a031002182215t7e253924rca5de90e013ec49b@mail.gmail.com>

There is no evidence of this on any box I have used.
Do you have selinux on maybe.
The best I can do now is declare we do not support your current
configuration.

On Feb 18, 2010 11:48 PM, "Yehavi Bourvine" <yehavi.bourvine at gmail.com>
wrote:

Unfortunately it did not help. I still get these error messages.

Just for the record, I have two systems:

   - Production one which is running Fedora-10 and exibits this problem.
   - Test system which is running Fedora-12 and does not exibit this
   phenomenon; however, there is almost zero traffic on this system.


I plan in upgrading the production system to Fedora-12, but it is not that
trivial...



                   Thanks, __Yehavi:

2010/2/16 Anthony Minessale <anthony.minessale at gmail.com>...

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100219/e7179a62/attachment-0002.html 

From lloyd.aloysius at gmail.com  Thu Feb 18 22:19:00 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Fri, 19 Feb 2010 01:19:00 -0500
Subject: [Freeswitch-users] IVR greeting - first two words missing
In-Reply-To: <87f2f3b91002181522te296581u12527a2a9cdf1d44@mail.gmail.com>
References: <8a19bf2e1002181410x44b40216j374a30a5675fd162@mail.gmail.com>
	<87f2f3b91002181522te296581u12527a2a9cdf1d44@mail.gmail.com>
Message-ID: <8a19bf2e1002182219i189664f2vfac02744331d5f7@mail.gmail.com>

Thank you Michael. It is now working perfect.

Lloyd


On Thu, Feb 18, 2010 at 6:22 PM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Thu, Feb 18, 2010 at 2:10 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:
>
>> Hi All,
>>
>> I setup a simple IVR. Here is the script.
>>
>> <menu name="test_ivr"
>>           greet-long="test/test-ivr.wav"
>>           greet-short="tset/test-ivr.wav"
>>           invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"
>>           exit-sound="voicemail/vm-goodbye.wav"
>>           timeout ="10000"
>>                   inter-digit-timeout="2000"
>>                   max-failures="3">
>>  <entry action="menu-exec-app" digits="/^(20[0-9])$/" param="transfer $1
>> XML default"/>
>>  <entry action="menu-exec-app" digits="9999" param="transfer 9999 XML
>> default"/>
>> </menu>
>>   </menus>
>> </configuration>
>>
>> *Dial Plan*
>>
>> <action application="answer"/>
>> <action application="ivr" data="test_ivr"/>
>>
>> Every time when I reach the IVR . I am getting first one or two words
>> missing( or may be not clear). How can I fix this issue.
>>
>> Thanks,
>> Lloyd
>>
>
> put a sleep after the answer:
> <action application="sleep" data="1000"/>
>
> You may have to tinker with the exact time, like maybe 1500 or 2000.
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100219/7932c2a5/attachment-0002.html 

From yehavi.bourvine at gmail.com  Thu Feb 18 22:19:23 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Fri, 19 Feb 2010 08:19:23 +0200
Subject: [Freeswitch-users] SQL ERR: library routine called out of
	sequence
In-Reply-To: <d361ca7b1002182141p53e3884fs2a9993d4739a1f91@mail.gmail.com>
References: <d361ca7b1002160656w7d616f6dn5e80087ebde94dbe@mail.gmail.com>
	<191c3a031002160705w4050c3cdk2bf2922e9aff5ba3@mail.gmail.com>
	<d361ca7b1002160727ucb6e0c8id62e803757333213@mail.gmail.com>
	<191c3a031002160755q569fe2ednfefb01b6d9cf7f0e@mail.gmail.com>
	<d361ca7b1002161059w1ef9484dr87e69a33aa2b62fc@mail.gmail.com>
	<191c3a031002161110s637a48a2r717ae94557fe9e9a@mail.gmail.com>
	<d361ca7b1002161130k3eb2d88by3100b30a9d29bd51@mail.gmail.com>
	<191c3a031002161230h2b6e6f9wf3f65ddb71f08297@mail.gmail.com>
	<d361ca7b1002182141p53e3884fs2a9993d4739a1f91@mail.gmail.com>
Message-ID: <d361ca7b1002182219k3a26ad6dx40c02538ccb2da1@mail.gmail.com>

Correction: It happens also on my test system (Fedora 12, Kernel 2.6.32.8,
Pentium-III, 1GHz).

                     Thanks, __Yehavi:

2010/2/19 Yehavi Bourvine <yehavi.bourvine at gmail.com>

>  Unfortunately it did not help. I still get these error messages.
>
> Just for the record, I have two systems:
>
>    - Production one which is running Fedora-10 and exibits this problem.
>    - Test system which is running Fedora-12 and does not exibit this
>    phenomenon; however, there is almost zero traffic on this system.
>
>
> I plan in upgrading the production system to Fedora-12, but it is not that
> trivial...
>
>                    Thanks, __Yehavi:
>
>  2010/2/16 Anthony Minessale <anthony.minessale at gmail.com>
>
>> Strange, even on abusive testing we have not seen this problem.
>>
>> please update to latest trunk.
>> There was only one change I can think of that may cause your issue and I
>> added a patch for it.
>> If it persists try setting the sql-in-transactions profile param to false.
>>
>>
>>
>>
>>
>>
>> On Tue, Feb 16, 2010 at 1:30 PM, Yehavi Bourvine <
>> yehavi.bourvine at gmail.com> wrote:
>>
>>>  Most of the queries are ok, only some fail, thus it doesn't look like
>>> permission problem. Furthermore, under 1.0.5pre10 it works for months.
>>>
>>> Might it be thread unsafe function calls? I've found the following while
>>> searching the WEB:
>>>
>>> *According to the MSDN docs, System.Timers.Timer operates in a thread
>>> pool.  If that's the case, your code is breaking the "connections cannot be
>>> shared across threads" rule for SQLit*
>>>
>>> Although it quotes MSDN, it might be related to Linux as well.
>>>
>>>                     Thanks, __Yehavi:
>>>
>>>  2010/2/16 Anthony Minessale <anthony.minessale at gmail.com>
>>>
>>>> That sounds about right.
>>>>
>>>> That error usually has something to do with using db calls on a closed
>>>> file or something along those lines.
>>>> Maybe you have a permission problem on the directory where the db files
>>>> are?
>>>>
>>>>
>>>>
>>>> On Tue, Feb 16, 2010 at 12:59 PM, Yehavi Bourvine <
>>>> yehavi.bourvine at gmail.com> wrote:
>>>>
>>>>>  The OS is Fedora-10 (soon to be upgraded to 12).
>>>>>
>>>>> What I do when I want to test a new version:
>>>>>
>>>>>    - Download the latest one into a fresh directory
>>>>>    - bootstrap.sh, configure and make
>>>>>    - stop Freeswitch, delete everything in lib, mod, bin ,db
>>>>>    - make install and run it.
>>>>>
>>>>>
>>>>> Is there additional place to clean?
>>>>>
>>>>>                         Thanks! __Yehavi:
>>>>>
>>>>>  2010/2/16 Anthony Minessale <anthony.minessale at gmail.com>
>>>>>
>>>>>> you may want to do a clean wipe of all files related to FS then.
>>>>>> you clearly have some problem with legacy something or other because
>>>>>> we don't see that on dozens of dev boxes.
>>>>>>
>>>>>> What os is it?
>>>>>>
>>>>>>
>>>>>>
>>>>>> On Tue, Feb 16, 2010 at 9:27 AM, Yehavi Bourvine <
>>>>>> yehavi.bourvine at gmail.com> wrote:
>>>>>>
>>>>>>>  Tried this, but it didn't help. I delete these DB files before any
>>>>>>> upgrade just to be sure.
>>>>>>>
>>>>>>>                   Thanks! __Yehavi:
>>>>>>>
>>>>>>>  2010/2/16 Anthony Minessale <anthony.minessale at gmail.com>
>>>>>>>
>>>>>>>> try removing all the .db files from /usr/local/freeswitch/db
>>>>>>>>
>>>>>>>>
>>>>>>>>   On Tue, Feb 16, 2010 at 8:56 AM, Yehavi Bourvine <
>>>>>>>> yehavi.bourvine at gmail.com> wrote:
>>>>>>>>
>>>>>>>>>   Hello,
>>>>>>>>>
>>>>>>>>>   After upgrading from 1.0.5pre10 to 1.0.5-20100216-0400 we started
>>>>>>>>> getting the above errors (I append bellow two samples). It seems Freeswitch
>>>>>>>>> fails to read a database using Sqlite.
>>>>>>>>> Anyone have seen this?
>>>>>>>>>
>>>>>>>>> Other details: Fedora 10, SQlite 3.5.9.
>>>>>>>>> We also do SQLite quesries during call setup via LUA from CoreDB.
>>>>>>>>> Is it an SQLite problem?
>>>>>>>>>
>>>>>>>>>                          Thanks! __Yehavi:
>>>>>>>>>
>>>>>>>>> The samples:
>>>>>>>>> 2010-02-16 16:22:13.762679 [ERR] switch_core_sqldb.c:404 SQL ERR
>>>>>>>>> [library routin
>>>>>>>>> e called out of sequence]
>>>>>>>>> delete from sip_dialogs where call_id='
>>>>>>>>> 8656841832142-120172129116107 at 10.64.1.2'
>>>>>>>>>
>>>>>>>>> 2010-02-16 16:42:00.802442 [ERR] switch_core_sqldb.c:722 SQL ERR:
>>>>>>>>> [select call_i
>>>>>>>>>
>>>>>>>>> d,sip_user,sip_host,contact,status,rpid,expires,user_agent,server_user,server_ho
>>>>>>>>>
>>>>>>>>> st,profile_name,hostname,network_ip,network_port,sip_username,sip_realm,mwi_user
>>>>>>>>> ,mwi_host from sip_registrations where profile_name='phones' and
>>>>>>>>> contact like '%
>>>>>>>>> 80635%'] library routine called out of sequence
>>>>>>>>>
>>>>>>>>> _______________________________________________
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>
>>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> --
>>>>>>>> Anthony Minessale II
>>>>>>>>
>>>>>>>> FreeSWITCH http://www.freeswitch.org/
>>>>>>>> ClueCon http://www.cluecon.com/
>>>>>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>>>>>
>>>>>>>> AIM: anthm
>>>>>>>> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
>>>>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>>>>>> IRC: irc.freenode.net #freeswitch
>>>>>>>>
>>>>>>>> FreeSWITCH Developer Conference
>>>>>>>> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>>>>>>>> iax:guest at conference.freeswitch.org/888
>>>>>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>>>>>> pstn:+19193869900
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>> Anthony Minessale II
>>>>>>
>>>>>> FreeSWITCH http://www.freeswitch.org/
>>>>>> ClueCon http://www.cluecon.com/
>>>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>>>
>>>>>> AIM: anthm
>>>>>> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
>>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>>>> IRC: irc.freenode.net #freeswitch
>>>>>>
>>>>>> FreeSWITCH Developer Conference
>>>>>> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>>>>>> iax:guest at conference.freeswitch.org/888
>>>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>>>> pstn:+19193869900
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>>
>>>> --
>>>> Anthony Minessale II
>>>>
>>>> FreeSWITCH http://www.freeswitch.org/
>>>> ClueCon http://www.cluecon.com/
>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>
>>>> AIM: anthm
>>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>> IRC: irc.freenode.net #freeswitch
>>>>
>>>> FreeSWITCH Developer Conference
>>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>>> iax:guest at conference.freeswitch.org/888
>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>> pstn:+19193869900
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100219/e6a6f60c/attachment-0002.html 

From anthony.minessale at gmail.com  Thu Feb 18 22:21:43 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 19 Feb 2010 00:21:43 -0600
Subject: [Freeswitch-users] voivemail quality
In-Reply-To: <191c3a031002182219l1dc9d55dj6bfb065b9edc0263@mail.gmail.com>
References: <27648642.post@talk.nabble.com>
	<b39273531002181801kd81a6b2x3327be4887563eae@mail.gmail.com>
	<1266552880.7684.17.camel@local.freepabx.com>
	<ec2ce17e1002182026m5986ab85t71006d608af73f88@mail.gmail.com>
	<1266555209.7684.23.camel@local.freepabx.com>
	<191c3a031002182219l1dc9d55dj6bfb065b9edc0263@mail.gmail.com>
Message-ID: <191c3a031002182221rf5fac8v76f611294b4881ad@mail.gmail.com>

Our g729 would have been done a lot sooner if we did not spend all our time
making sure there is no audio hiccups on a 600ms latent connection.  Yes we
made it possible and now we have g729 too nonetheless.  The only thing
howler filled is their wallet.  Let's see them as a gold sponsor of cluecon
if they are doing us such a big favor........

On Feb 18, 2010 10:58 PM, "David Knell" <dave at 3c.co.uk> wrote:


On Fri, 2010-02-19 at 14:26 +1000, jay binks wrote:
> I think suggesting the user test with other c...
Edward already said that things worked with G.711 - that ought to be
enough testing with other codecs.


> your attacking rupa for making suggestions about where the issue is
> and what to do in order to...
It doesn't, directly.  And, had Rupa simply asked if the OP had raised
the issue with Howler - which is a completely valid thing to do,
particularly with the evidence in front of him - that'd have been one
thing.  But he didn't.

> pot, kettle, black ???

How so, exactly?

--Dave


>
>
> J
>
> On Fri, Feb 19, 2010 at 2:14 PM, David Knell <dave at 3c.co.uk> wrote:
> Rupa -...
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100219/5316e654/attachment-0002.html 

From anthony.minessale at gmail.com  Thu Feb 18 22:23:20 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 19 Feb 2010 00:23:20 -0600
Subject: [Freeswitch-users] 1.0.4 vs. trunk vs. bugs
In-Reply-To: <616F57B5-182A-49A9-9825-2E5B72F5DB05@freeswitch.org>
References: <5d3e0dc61002181741r57b91f86h4810de63c1a3d46d@mail.gmail.com>
	<20100219015118.GA12983@jdc.jasonjgw.net>
	<1266552605.7684.11.camel@local.freepabx.com>
	<616F57B5-182A-49A9-9825-2E5B72F5DB05@freeswitch.org>
Message-ID: <191c3a031002182223o66a16cdcjc5e43b637a5b91be@mail.gmail.com>

How many more ungrateful complaints can we get in one night.... sigh

On Feb 18, 2010 10:21 PM, "Brian West" <brian at freeswitch.org> wrote:

OK so I can sign you up for the stable team?  ;)  As per my previous email
i'm 100% sure we would do a stable release if we had people tending to
issues.  The only problem is you would have to be on IRC tending to issues
because if tony sees someone asking about a problem he'll be diving in to
fix it before they can say "I have this one".  This also means working in a
similar manner we do already.  Our process is very chaotic at times but it
has served us well so far.

The goal is to leave Anthony alone so he can move forward and let the stable
team manage the jira's and issues on the list related to stable.

/b



On Feb 18, 2010, at 10:10 PM, David Knell wrote:

>>
>> Lon Baker <lon at kickasspixels.com> wrote:
>...

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100219/632ca546/attachment-0002.html 

From mike at jerris.com  Thu Feb 18 23:01:39 2010
From: mike at jerris.com (Michael Jerris)
Date: Fri, 19 Feb 2010 02:01:39 -0500
Subject: [Freeswitch-users] Presence PUBLISH Not Updating
	AfterSoftphoneOffLine Then Available
In-Reply-To: <E69C19EDE6F541198E27BE0883CB773D@greyhawk.tonecommander.com>
References: <31FEE9072F104AE18B26812978FCA341@greyhawk.tonecommander.com><45963EB5-DFE5-43FA-AFCC-857E1E719E2D@jerris.com><191c3a031002091557jfc9b486lb832ea538883bce9@mail.gmail.com><68A4C011A72B4ABA91A185F32480EB99@greyhawk.tonecommander.com>
	<191c3a031002162139p57c94046y660fb8d01fd76f46@mail.gmail.com>
	<E69C19EDE6F541198E27BE0883CB773D@greyhawk.tonecommander.com>
Message-ID: <8E59FC36-2CDB-4B77-8736-2319D883889F@jerris.com>



On Feb 18, 2010, at 3:26 PM, "Jerry Richards" <jerry.richards at teotech.com 
 > wrote:

> Yes, these are Bria (CounterPath) phones, but these are phones that  
> I'm using and they are popular, and as far as I know, faithful to  
> the SIP RFCs,

Hahahha

/me falls over laughing 
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100219/0196b4a8/attachment-0002.html 

From mike at jerris.com  Thu Feb 18 23:10:39 2010
From: mike at jerris.com (Michael Jerris)
Date: Fri, 19 Feb 2010 02:10:39 -0500
Subject: [Freeswitch-users] building for Lenny
In-Reply-To: <3c5cf5261002181542y43adb7a7ofe2a15c38baf0327@mail.gmail.com>
References: <3c5cf5261002180046vb0015e7t787599c9903aedc@mail.gmail.com>
	<33c87fa31002180056g51ef8496qa9a35746d0fe75ec@mail.gmail.com>
	<3c5cf5261002180110he0ae815uc713f77f289ac625@mail.gmail.com>
	<a9f4977a70b3fcf1b30413ea562ca962.squirrel@www.webcontracts.co.uk>
	<20100218161741.GC4236@base.carmickle.com>
	<3c5cf5261002181542y43adb7a7ofe2a15c38baf0327@mail.gmail.com>
Message-ID: <4C13BCBF-1DA0-47AE-A749-738AD6D8FE48@jerris.com>

Please create me a bug on http://jira.freeswitch.org for this issue.

On Feb 18, 2010, at 6:42 PM, Brian May  
<brian at microcomaustralia.com.au> wrote:

> On 19 February 2010 03:17, Frank Carmickle <frank at carmickle.com>  
> wrote:
>> Like I said you can and should build debs from svn.  As far as
>> I see it there is no reason to not build debs.
>
> Unfortunately, that didn't create any of the packages for the sound
> files, and I can't see where to get a deb package for the sound files
> that really does contain the sound files.
>
>
>
> Also I get errors when trying to start it up, not sure how many of
> these I can ignore are warnings and how many are because I am doing it
> wrong:
>
> voyage:~# /opt/freeswitch/bin/freeswitch
> Error: stacksize 4194303 is too large: run ulimit -s 240 or run
> /opt/freeswitch/bin/freeswitch -waste.
> auto-adjusting stack size for optimal performance...
> 2010-02-19 10:59:41.203000 [INFO] switch_event.c:580 Activate  
> Eventing Engine.
> 2010-02-19 10:59:41.208000 [DEBUG] switch_event.c:568 Create event
> dispatch thread 0
> 2010-02-19 10:59:41.428000 [ERR] switch_xml.c:1297 Couldnt open
> /opt/freeswitch/conf/autoload_configs/../ivr_menus/*.xml (No such file
> or directory)
> Error including
> /opt/freeswitch/conf/autoload_configs/../ivr_menus/*.xml (No such file
> or directory)
> 2010-02-19 10:59:42.376000 [ERR] switch_xml.c:1297 Couldnt open
> /opt/freeswitch/conf/lang/de/*.xml (No such file or directory)
> Error including /opt/freeswitch/conf/lang/de/*.xml (No such file or  
> directory)
> 2010-02-19 10:59:42.526000 [ERR] switch_xml.c:1297 Couldnt open
> /opt/freeswitch/conf/lang/en/dir/sounds.xml (No such file or
> directory)
> Error including /opt/freeswitch/conf/lang/en/dir/sounds.xml (No such
> file or directory)
> 2010-02-19 10:59:42.528000 [ERR] switch_xml.c:1297 Couldnt open
> /opt/freeswitch/conf/lang/fr/*.xml (No such file or directory)
> Error including /opt/freeswitch/conf/lang/fr/*.xml (No such file or  
> directory)
> 2010-02-19 10:59:42.529000 [ERR] switch_xml.c:1297 Couldnt open
> /opt/freeswitch/conf/lang/ru/*.xml (No such file or directory)
> Error including /opt/freeswitch/conf/lang/ru/*.xml (No such file or  
> directory)
> 2010-02-19 10:59:42.609000 [INFO] switch_nat.c:409 Scanning for NAT
> 2010-02-19 10:59:42.611000 [DEBUG] switch_nat.c:166 Checking for PMP  
> 1/5
> 2010-02-19 10:59:42.614000 [ERR] switch_nat.c:197 Error checking for
> PMP [general error]
> 2010-02-19 10:59:42.614000 [DEBUG] switch_nat.c:414 Checking for UPnP
> 2010-02-19 10:59:54.619000 [INFO] switch_nat.c:429 No PMP or UPnP NAT
> devices detected!
> 2010-02-19 10:59:54.623000 [INFO] switch_core_sqldb.c:1248 Opening DB
> 2010-02-19 10:59:54.627000 [ERR] switch_core_sqldb.c:404 SQL ERR [no
> such table: channels]
> drop table channels
> 2010-02-19 10:59:54.628000 [ERR] switch_core_sqldb.c:404 SQL ERR [no
> such table: calls]
> drop table calls
> 2010-02-19 10:59:54.630000 [ERR] switch_core_sqldb.c:404 SQL ERR [no
> such table: interfaces]
> drop table interfaces
> 2010-02-19 10:59:54.631000 [ERR] switch_core_sqldb.c:404 SQL ERR [no
> such table: tasks]
> drop table tasks
> 2010-02-19 10:59:54.634000 [DEBUG] switch_core_sqldb.c:765 SQL ERR [no
> such table: aliases]
> [select hostname from aliases]
> Auto Generating Table!
> 2010-02-19 10:59:54.636000 [DEBUG] switch_core_sqldb.c:772 SQL ERR [no
> such table: aliases]
> [CREATE TABLE aliases (
>   sticky  INTEGER,
>   alias  VARCHAR(128),
>   command  VARCHAR(4096),
>   hostname VARCHAR(256)
> );
> ]
> 2010-02-19 10:59:54.644000 [DEBUG] switch_core_sqldb.c:765 SQL ERR [no
> such table: nat]
> [select hostname from nat]
> Auto Generating Table!
> 2010-02-19 10:59:54.646000 [DEBUG] switch_core_sqldb.c:772 SQL ERR [no
> such table: nat]
> [CREATE TABLE nat (
>   sticky  INTEGER,
>    port    INTEGER,
>    proto    INTEGER,
>   hostname VARCHAR(256)
> );
> ]
>
> Am I expected to setup a SQL database to get this working? Or did it
> just setup one automatically?
>
> 2010-02-19 10:59:59.684000 [CRIT] switch_loadable_module.c:882 Error
> Loading module /opt/freeswitch/mod/mod_voipcodecs.so
> **libjpeg.so.62: cannot open shared object file: No such file or  
> directory**
> 2010-02-19 10:59:59.686000 [CRIT] switch_loadable_module.c:882 Error
> Loading module /opt/freeswitch/mod/mod_g723_1.so
> **/opt/freeswitch/mod/mod_g723_1.so: cannot open shared object file:
> No such file or directory**
> 2010-02-19 10:59:59.687000 [CRIT] switch_loadable_module.c:882 Error
> Loading module /opt/freeswitch/mod/mod_g729.so
> **/opt/freeswitch/mod/mod_g729.so: cannot open shared object file: No
> such file or directory**
> 2010-02-19 10:59:59.688000 [CRIT] switch_loadable_module.c:882 Error
> Loading module /opt/freeswitch/mod/mod_amr.so
> **/opt/freeswitch/mod/mod_amr.so: cannot open shared object file: No
> such file or directory**
> 2010-02-19 10:59:59.835000 [CRIT] switch_loadable_module.c:882 Error
> Loading module /opt/freeswitch/mod/mod_file_string.so
> **/opt/freeswitch/mod/mod_file_string.so: cannot open shared object
> file: No such file or directory**
> 2010-02-19 11:00:00.143000 [CRIT] switch_loadable_module.c:882 Error
> Loading module /opt/freeswitch/mod/mod_say_ru.so
> **/opt/freeswitch/mod/mod_say_ru.so: cannot open shared object file:
> No such file or directory**
>
> suspect I don't really need to worry about some of these. I assume
> there is a config file somewhere where I can disable these options.
>
>
> Ok, as a really pathetic question, now I have started it, how do I  
> stop it?
>
> freeswitch at voyage> halt
> Unknown Command: halt
> freeswitch at voyage> quit
> Unknown Command: quit
> freeswitch at voyage> exit
> Unknown Command: exit
> freeswitch at voyage> bye
> Unknown Command: bye
>
> -- 
> Brian May <brian at microcomaustralia.com.au>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org



From mrene_lists at avgs.ca  Thu Feb 18 23:14:11 2010
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Fri, 19 Feb 2010 02:14:11 -0500
Subject: [Freeswitch-users] building for Lenny
In-Reply-To: <4C13BCBF-1DA0-47AE-A749-738AD6D8FE48@jerris.com>
References: <3c5cf5261002180046vb0015e7t787599c9903aedc@mail.gmail.com>
	<33c87fa31002180056g51ef8496qa9a35746d0fe75ec@mail.gmail.com>
	<3c5cf5261002180110he0ae815uc713f77f289ac625@mail.gmail.com>
	<a9f4977a70b3fcf1b30413ea562ca962.squirrel@www.webcontracts.co.uk>
	<20100218161741.GC4236@base.carmickle.com>
	<3c5cf5261002181542y43adb7a7ofe2a15c38baf0327@mail.gmail.com>
	<4C13BCBF-1DA0-47AE-A749-738AD6D8FE48@jerris.com>
Message-ID: <155182E9-FD25-4BE2-A5EE-389E3D64B9DC@avgs.ca>

You can stop it with "..." or "fsctl shutdown"

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 19-Feb-10, at 2:10 AM, Michael Jerris wrote:

> Please create me a bug on http://jira.freeswitch.org for this issue.
>
> On Feb 18, 2010, at 6:42 PM, Brian May
> <brian at microcomaustralia.com.au> wrote:
>
>> On 19 February 2010 03:17, Frank Carmickle <frank at carmickle.com>
>> wrote:
>>> Like I said you can and should build debs from svn.  As far as
>>> I see it there is no reason to not build debs.
>>
>> Unfortunately, that didn't create any of the packages for the sound
>> files, and I can't see where to get a deb package for the sound files
>> that really does contain the sound files.
>>
>>
>>
>> Also I get errors when trying to start it up, not sure how many of
>> these I can ignore are warnings and how many are because I am doing  
>> it
>> wrong:
>>
>> voyage:~# /opt/freeswitch/bin/freeswitch
>> Error: stacksize 4194303 is too large: run ulimit -s 240 or run
>> /opt/freeswitch/bin/freeswitch -waste.
>> auto-adjusting stack size for optimal performance...
>> 2010-02-19 10:59:41.203000 [INFO] switch_event.c:580 Activate
>> Eventing Engine.
>> 2010-02-19 10:59:41.208000 [DEBUG] switch_event.c:568 Create event
>> dispatch thread 0
>> 2010-02-19 10:59:41.428000 [ERR] switch_xml.c:1297 Couldnt open
>> /opt/freeswitch/conf/autoload_configs/../ivr_menus/*.xml (No such  
>> file
>> or directory)
>> Error including
>> /opt/freeswitch/conf/autoload_configs/../ivr_menus/*.xml (No such  
>> file
>> or directory)
>> 2010-02-19 10:59:42.376000 [ERR] switch_xml.c:1297 Couldnt open
>> /opt/freeswitch/conf/lang/de/*.xml (No such file or directory)
>> Error including /opt/freeswitch/conf/lang/de/*.xml (No such file or
>> directory)
>> 2010-02-19 10:59:42.526000 [ERR] switch_xml.c:1297 Couldnt open
>> /opt/freeswitch/conf/lang/en/dir/sounds.xml (No such file or
>> directory)
>> Error including /opt/freeswitch/conf/lang/en/dir/sounds.xml (No such
>> file or directory)
>> 2010-02-19 10:59:42.528000 [ERR] switch_xml.c:1297 Couldnt open
>> /opt/freeswitch/conf/lang/fr/*.xml (No such file or directory)
>> Error including /opt/freeswitch/conf/lang/fr/*.xml (No such file or
>> directory)
>> 2010-02-19 10:59:42.529000 [ERR] switch_xml.c:1297 Couldnt open
>> /opt/freeswitch/conf/lang/ru/*.xml (No such file or directory)
>> Error including /opt/freeswitch/conf/lang/ru/*.xml (No such file or
>> directory)
>> 2010-02-19 10:59:42.609000 [INFO] switch_nat.c:409 Scanning for NAT
>> 2010-02-19 10:59:42.611000 [DEBUG] switch_nat.c:166 Checking for PMP
>> 1/5
>> 2010-02-19 10:59:42.614000 [ERR] switch_nat.c:197 Error checking for
>> PMP [general error]
>> 2010-02-19 10:59:42.614000 [DEBUG] switch_nat.c:414 Checking for UPnP
>> 2010-02-19 10:59:54.619000 [INFO] switch_nat.c:429 No PMP or UPnP NAT
>> devices detected!
>> 2010-02-19 10:59:54.623000 [INFO] switch_core_sqldb.c:1248 Opening DB
>> 2010-02-19 10:59:54.627000 [ERR] switch_core_sqldb.c:404 SQL ERR [no
>> such table: channels]
>> drop table channels
>> 2010-02-19 10:59:54.628000 [ERR] switch_core_sqldb.c:404 SQL ERR [no
>> such table: calls]
>> drop table calls
>> 2010-02-19 10:59:54.630000 [ERR] switch_core_sqldb.c:404 SQL ERR [no
>> such table: interfaces]
>> drop table interfaces
>> 2010-02-19 10:59:54.631000 [ERR] switch_core_sqldb.c:404 SQL ERR [no
>> such table: tasks]
>> drop table tasks
>> 2010-02-19 10:59:54.634000 [DEBUG] switch_core_sqldb.c:765 SQL ERR  
>> [no
>> such table: aliases]
>> [select hostname from aliases]
>> Auto Generating Table!
>> 2010-02-19 10:59:54.636000 [DEBUG] switch_core_sqldb.c:772 SQL ERR  
>> [no
>> such table: aliases]
>> [CREATE TABLE aliases (
>>  sticky  INTEGER,
>>  alias  VARCHAR(128),
>>  command  VARCHAR(4096),
>>  hostname VARCHAR(256)
>> );
>> ]
>> 2010-02-19 10:59:54.644000 [DEBUG] switch_core_sqldb.c:765 SQL ERR  
>> [no
>> such table: nat]
>> [select hostname from nat]
>> Auto Generating Table!
>> 2010-02-19 10:59:54.646000 [DEBUG] switch_core_sqldb.c:772 SQL ERR  
>> [no
>> such table: nat]
>> [CREATE TABLE nat (
>>  sticky  INTEGER,
>>   port    INTEGER,
>>   proto    INTEGER,
>>  hostname VARCHAR(256)
>> );
>> ]
>>
>> Am I expected to setup a SQL database to get this working? Or did it
>> just setup one automatically?
>>
>> 2010-02-19 10:59:59.684000 [CRIT] switch_loadable_module.c:882 Error
>> Loading module /opt/freeswitch/mod/mod_voipcodecs.so
>> **libjpeg.so.62: cannot open shared object file: No such file or
>> directory**
>> 2010-02-19 10:59:59.686000 [CRIT] switch_loadable_module.c:882 Error
>> Loading module /opt/freeswitch/mod/mod_g723_1.so
>> **/opt/freeswitch/mod/mod_g723_1.so: cannot open shared object file:
>> No such file or directory**
>> 2010-02-19 10:59:59.687000 [CRIT] switch_loadable_module.c:882 Error
>> Loading module /opt/freeswitch/mod/mod_g729.so
>> **/opt/freeswitch/mod/mod_g729.so: cannot open shared object file: No
>> such file or directory**
>> 2010-02-19 10:59:59.688000 [CRIT] switch_loadable_module.c:882 Error
>> Loading module /opt/freeswitch/mod/mod_amr.so
>> **/opt/freeswitch/mod/mod_amr.so: cannot open shared object file: No
>> such file or directory**
>> 2010-02-19 10:59:59.835000 [CRIT] switch_loadable_module.c:882 Error
>> Loading module /opt/freeswitch/mod/mod_file_string.so
>> **/opt/freeswitch/mod/mod_file_string.so: cannot open shared object
>> file: No such file or directory**
>> 2010-02-19 11:00:00.143000 [CRIT] switch_loadable_module.c:882 Error
>> Loading module /opt/freeswitch/mod/mod_say_ru.so
>> **/opt/freeswitch/mod/mod_say_ru.so: cannot open shared object file:
>> No such file or directory**
>>
>> suspect I don't really need to worry about some of these. I assume
>> there is a config file somewhere where I can disable these options.
>>
>>
>> Ok, as a really pathetic question, now I have started it, how do I
>> stop it?
>>
>> freeswitch at voyage> halt
>> Unknown Command: halt
>> freeswitch at voyage> quit
>> Unknown Command: quit
>> freeswitch at voyage> exit
>> Unknown Command: exit
>> freeswitch at voyage> bye
>> Unknown Command: bye
>>
>> -- 
>> Brian May <brian at microcomaustralia.com.au>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>> users
>> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From mike at jerris.com  Thu Feb 18 23:17:52 2010
From: mike at jerris.com (Michael Jerris)
Date: Fri, 19 Feb 2010 02:17:52 -0500
Subject: [Freeswitch-users] 1.0.4 vs. trunk vs. bugs
In-Reply-To: <616F57B5-182A-49A9-9825-2E5B72F5DB05@freeswitch.org>
References: <5d3e0dc61002181741r57b91f86h4810de63c1a3d46d@mail.gmail.com>
	<20100219015118.GA12983@jdc.jasonjgw.net>
	<1266552605.7684.11.camel@local.freepabx.com>
	<616F57B5-182A-49A9-9825-2E5B72F5DB05@freeswitch.org>
Message-ID: <7716C919-BCD1-44FA-9ED2-5268F4C42100@jerris.com>

This seems a good time to note that we are still looking for  
volunteers to assist in maintaining a stable branch.  I can not do  
this without additional volunteer resources.  We have asked several  
times recently to fairly silent response.  If anyone is interested in  
assisting with this effort, please contact  me offlist and we can  
discuss further.

Mike

On Feb 18, 2010, at 11:16 PM, Brian West <brian at freeswitch.org> wrote:

> OK so I can sign you up for the stable team?  ;)  As per my previous  
> email i'm 100% sure we would do a stable release if we had people  
> tending to issues.  The only problem is you would have to be on IRC  
> tending to issues because if tony sees someone asking about a  
> problem he'll be diving in to fix it before they can say "I have  
> this one".  This also means working in a similar manner we do  
> already.  Our process is very chaotic at times but it has served us  
> well so far.
>
> The goal is to leave Anthony alone so he can move forward and let  
> the stable team manage the jira's and issues on the list related to  
> stable.
>
> /b
>
> On Feb 18, 2010, at 10:10 PM, David Knell wrote:
>
>>>
>>> Lon Baker <lon at kickasspixels.com> wrote:
>>>
>>>> The development branch is where feature requests and non-critical  
>>>> bugs
>>>> reports would be filed for the next production release.
>>>>
>>>> The current process leaves a gap between production ready and
>>>> development code that may become greater over time.
>>
>> Going against the grain here, I agree with you.  The current way of
>> doing things is, in my opinion, not well thought through - there's no
>> reason to tag and release versions if the answer to any issue is  
>> 'make
>> current', and support is not available unless that's been done.  Far
>> better to either have meaningful releases with stable and devel
>> branches, or not to have releases at all.
>>
>> --Dave
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100219/c53d7751/attachment-0002.html 

From mike at jerris.com  Thu Feb 18 23:22:26 2010
From: mike at jerris.com (Michael Jerris)
Date: Fri, 19 Feb 2010 02:22:26 -0500
Subject: [Freeswitch-users] mod_fax undefined symbol
In-Reply-To: <6FCE44FF-70C1-412D-B722-4C8D359381BA@ttnc.co.uk>
References: <6FCE44FF-70C1-412D-B722-4C8D359381BA@ttnc.co.uk>
Message-ID: <0F7476C7-FA7E-441F-9E17-EB2D23CD6055@jerris.com>

Please open a bug on http://jira.freeswitch.org for this issue.  As a  
note, we use our own copy of libtiff, statically linked to the  
module.  Is this recent trunk or something older?

On Feb 18, 2010, at 8:57 PM, TTNC - Technical <technical at ttnc.co.uk>  
wrote:

> Hi Guys
>
> I'm having trouble getting mod_fax to load. Running on Debian  
> testing (squeeze). Everything is installed as per - http://wiki.freeswitch.org/wiki/Installation_Guide 
> . (dpkg-buildpackage etc)
>
> When trying to load the fax module I get:
>
> 2010-02-19 01:48:17.554935 [CRIT] switch_loadable_module.c:882 Error  
> Loading module /opt/freeswitch/mod/mod_fax.so
> **/opt/freeswitch/mod/mod_fax.so: undefined symbol:  
> TIFFDefaultStripSize**
>
> And when a fax is sent, I'm getting:
>
> 2010-02-19 01:04:57.355330 [ERR] switch_core_session.c:1490 Invalid  
> Application rxfax
>
> I guess because mod_fax isn't loaded.
>
> I've got libtiff4 and libtiff4-dev installed:
>
> ii  libtiff4                             3.9.2-2                   
> Tag Image File Format (TIFF) library
> ii  libtiff4-dev                         3.9.2-2                   
> Tag Image File Format library (TIFF), development files
> ii  libtiffxx0c2                         3.9.2-2                   
> Tag Image File Format (TIFF) library -- C++ interface
>
> Just tried updating to the latest svn trunk (16700M) and it hasn't  
> made any difference.
>
>> From googling, it suggests that it could be because the module is  
>> complied against a different one currently running on the system,  
>> however I'm not sure how this can be the case, there is only the  
>> one version installed.
>
> Any suggestions as to what I can try?
>
> Any help appreciated
>
> Russ
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org



From mike at jerris.com  Thu Feb 18 23:24:37 2010
From: mike at jerris.com (Michael Jerris)
Date: Fri, 19 Feb 2010 02:24:37 -0500
Subject: [Freeswitch-users] Announcement: FreeSWITCH Add Broadsoft SCA
	Support
In-Reply-To: <d361ca7b1002182154o52352c62h8c0ec02fa66e8444@mail.gmail.com>
References: <87f2f3b91001111742k691966cfs82515db77c13d4d4@mail.gmail.com>
	<8D062B6E-4691-4BA6-B038-0EC6E8642DDA@jerris.com>
	<9D328485-D4C3-4B14-808A-CB63BBF06DC1@freeswitch.org>
	<d361ca7b1002170549g608d6e16vb2eb3f087fff9752@mail.gmail.com>
	<191c3a031002170556p5dfd4c55le3c279f194808d92@mail.gmail.com>
	<191c3a031002170557g5d8c9df9xf40d1b74b9a6f19d@mail.gmail.com>
	<d361ca7b1002170700x15630ab2tabc6f3d91ebc3303@mail.gmail.com>
	<ADC8B813-4161-481C-877E-93FC012858B4@freeswitch.org>
	<d361ca7b1002182101p60b4e8f8hf4dc95e1d2251e30@mail.gmail.com>
	<8b1c9cda1002182144m65ccf979w71384a1e7a55e2ea@mail.gmail.com>
	<d361ca7b1002182154o52352c62h8c0ec02fa66e8444@mail.gmail.com>
Message-ID: <2F87B95C-EA23-4072-B845-3F7AE0617BE3@jerris.com>

If this issue is not already on jira could you please make sure it  
gets added?

Mike

On Feb 19, 2010, at 12:54 AM, Yehavi Bourvine  
<yehavi.bourvine at gmail.com> wrote:

> Hello Gabe,
>
>   As you can see - Brian is actively investigating it, so you can  
> expect for some fix soon...
>
>                        Regards, __Yehavi:
>
> 2010/2/19 Gabriel Kuri <gkuri at ieee.org>
> >  When a call arrives, both ring; the one that did not answer gets  
> only a
> > cancel mesage without any further notification that the extension  
> is in use
> > by the other phone.
>
> These are the same exact symptoms I posted about earlier this week,
> with the Cisco SPA-5xx series phones. I still have yet to figure out
> why this is happening, if you find out what's going on, please post
> back the solution, I'd like to know the resolution.
>
> Thanks,
> Gabe
>
>
>
> On Thu, Feb 18, 2010 at 9:01 PM, Yehavi Bourvine
> <yehavi.bourvine at gmail.com> wrote:
> > Thanks Brian. It now works better, but not fully (using 16659M).
> >
> > What happens is:
> >
> > When one of the Polycoms seize the line it is ok - the other phone  
> gets
> > notification and the extension status is "in use".
> > When one of the Polycom phones initiates a call - all is ok:
> >
> > The other side sees that the extension is in use.
> > When it is put to hold all phones who share this extension see it  
> and can
> > pick the call.
> >
>
> >
> >                          Thanks! __Yehavi:
> >
> > 2010/2/17 Brian West <brian at freeswitch.org>
> >>
> >> Step 1. Enable manage-shared-appearance=true
> >>
> >> Step 2. Now in the phone's config Configure the phone as usually,  
> set the
> >> line shared and DO NOT set the third party name.
> >>
> >> Step 3. Reboot
> >>
> >> It should work.
> >>
> >> I wish someone that has this working would write some wiki docs  
> these
> >> threads about it not working are getting rather old when I know  
> for a fact
> >> they work fine.
> >>
> >> The gateway info missing is a gateway you have configured getting a
> >> notify.  It has nothing to do with SCA.
> >>
> >> /b
> >>
> >> On Feb 17, 2010, at 9:00 AM, Yehavi Bourvine wrote:
> >>
> >> > .
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/ 
> freeswitch-users
> >> http://www.freeswitch.org
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100219/4c863452/attachment-0002.html 

From mike at jerris.com  Thu Feb 18 23:29:18 2010
From: mike at jerris.com (Michael Jerris)
Date: Fri, 19 Feb 2010 02:29:18 -0500
Subject: [Freeswitch-users] stream a file multicast with mod_esf
In-Reply-To: <a792304b1002182209m43ada13cp8ed3d91ad509f0a2@mail.gmail.com>
References: <B98B6921358D0C49BD21CD023F1F52C302623B4B@hbghs04a.ww300.siemens.net>
	<B98B6921358D0C49BD21CD023F1F52C302623BE0@hbghs04a.ww300.siemens.net>
	<7C996136-963A-42D2-B0FE-D2729F968E52@freeswitch.org>
	<27071973.post@talk.nabble.com>
	<378CDEC3-5A41-4EF9-927A-311F2531E6AB@freeswitch.org>
	<a792304b1001080725y19ff28camd37425927550f311@mail.gmail.com>
	<191c3a031001081848w9036207yf6a4b473ec233243@mail.gmail.com>
	<85C74329-0D7C-48B2-B7A8-D4AA67C6E3F2@freeswitch.org>
	<a792304b1002182209m43ada13cp8ed3d91ad509f0a2@mail.gmail.com>
Message-ID: <DBA4E291-9377-489A-841A-1A3D83880DF2@jerris.com>

Listening on multicast is noting special for multicast, it is just  
like reading any other udp socket

Mike

On Feb 19, 2010, at 1:09 AM, MohammedShehzad <pmhshz at gmail.com> wrote:

> Hello,
> I am now looking into the code from few days to create the RTP  
> Multicast Listener first.
> Is this something similar coding as mod_esf need to do here to  
> listen on the Multicast IP & Port? Or I need to use combination some  
> rtp related function of (switch_rtp.c) here?
>
> I think this not proper place to discuss here, let me know where  
> should I continue further discussion.
>
> -MohammedShehzad
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org



From mike at jerris.com  Thu Feb 18 23:30:56 2010
From: mike at jerris.com (Michael Jerris)
Date: Fri, 19 Feb 2010 02:30:56 -0500
Subject: [Freeswitch-users] building for Lenny
In-Reply-To: <155182E9-FD25-4BE2-A5EE-389E3D64B9DC@avgs.ca>
References: <3c5cf5261002180046vb0015e7t787599c9903aedc@mail.gmail.com>
	<33c87fa31002180056g51ef8496qa9a35746d0fe75ec@mail.gmail.com>
	<3c5cf5261002180110he0ae815uc713f77f289ac625@mail.gmail.com>
	<a9f4977a70b3fcf1b30413ea562ca962.squirrel@www.webcontracts.co.uk>
	<20100218161741.GC4236@base.carmickle.com>
	<3c5cf5261002181542y43adb7a7ofe2a15c38baf0327@mail.gmail.com>
	<4C13BCBF-1DA0-47AE-A749-738AD6D8FE48@jerris.com>
	<155182E9-FD25-4BE2-A5EE-389E3D64B9DC@avgs.ca>
Message-ID: <D35A6E61-BBBF-49F5-9EF1-FAFCECCF679A@jerris.com>

There is also a "help" command

Mike

On Feb 19, 2010, at 2:14 AM, Mathieu Rene <mrene_lists at avgs.ca> wrote:

> You can stop it with "..." or "fsctl shutdown"
>
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>
>
>
>
> On 19-Feb-10, at 2:10 AM, Michael Jerris wrote:
>
>> Please create me a bug on http://jira.freeswitch.org for this issue.
>>
>> On Feb 18, 2010, at 6:42 PM, Brian May
>> <brian at microcomaustralia.com.au> wrote:
>>
>>> On 19 February 2010 03:17, Frank Carmickle <frank at carmickle.com>
>>> wrote:
>>>> Like I said you can and should build debs from svn.  As far as
>>>> I see it there is no reason to not build debs.
>>>
>>> Unfortunately, that didn't create any of the packages for the sound
>>> files, and I can't see where to get a deb package for the sound  
>>> files
>>> that really does contain the sound files.
>>>
>>>
>>>
>>> Also I get errors when trying to start it up, not sure how many of
>>> these I can ignore are warnings and how many are because I am doing
>>> it
>>> wrong:
>>>
>>> voyage:~# /opt/freeswitch/bin/freeswitch
>>> Error: stacksize 4194303 is too large: run ulimit -s 240 or run
>>> /opt/freeswitch/bin/freeswitch -waste.
>>> auto-adjusting stack size for optimal performance...
>>> 2010-02-19 10:59:41.203000 [INFO] switch_event.c:580 Activate
>>> Eventing Engine.
>>> 2010-02-19 10:59:41.208000 [DEBUG] switch_event.c:568 Create event
>>> dispatch thread 0
>>> 2010-02-19 10:59:41.428000 [ERR] switch_xml.c:1297 Couldnt open
>>> /opt/freeswitch/conf/autoload_configs/../ivr_menus/*.xml (No such
>>> file
>>> or directory)
>>> Error including
>>> /opt/freeswitch/conf/autoload_configs/../ivr_menus/*.xml (No such
>>> file
>>> or directory)
>>> 2010-02-19 10:59:42.376000 [ERR] switch_xml.c:1297 Couldnt open
>>> /opt/freeswitch/conf/lang/de/*.xml (No such file or directory)
>>> Error including /opt/freeswitch/conf/lang/de/*.xml (No such file or
>>> directory)
>>> 2010-02-19 10:59:42.526000 [ERR] switch_xml.c:1297 Couldnt open
>>> /opt/freeswitch/conf/lang/en/dir/sounds.xml (No such file or
>>> directory)
>>> Error including /opt/freeswitch/conf/lang/en/dir/sounds.xml (No such
>>> file or directory)
>>> 2010-02-19 10:59:42.528000 [ERR] switch_xml.c:1297 Couldnt open
>>> /opt/freeswitch/conf/lang/fr/*.xml (No such file or directory)
>>> Error including /opt/freeswitch/conf/lang/fr/*.xml (No such file or
>>> directory)
>>> 2010-02-19 10:59:42.529000 [ERR] switch_xml.c:1297 Couldnt open
>>> /opt/freeswitch/conf/lang/ru/*.xml (No such file or directory)
>>> Error including /opt/freeswitch/conf/lang/ru/*.xml (No such file or
>>> directory)
>>> 2010-02-19 10:59:42.609000 [INFO] switch_nat.c:409 Scanning for NAT
>>> 2010-02-19 10:59:42.611000 [DEBUG] switch_nat.c:166 Checking for PMP
>>> 1/5
>>> 2010-02-19 10:59:42.614000 [ERR] switch_nat.c:197 Error checking for
>>> PMP [general error]
>>> 2010-02-19 10:59:42.614000 [DEBUG] switch_nat.c:414 Checking for  
>>> UPnP
>>> 2010-02-19 10:59:54.619000 [INFO] switch_nat.c:429 No PMP or UPnP  
>>> NAT
>>> devices detected!
>>> 2010-02-19 10:59:54.623000 [INFO] switch_core_sqldb.c:1248 Opening  
>>> DB
>>> 2010-02-19 10:59:54.627000 [ERR] switch_core_sqldb.c:404 SQL ERR [no
>>> such table: channels]
>>> drop table channels
>>> 2010-02-19 10:59:54.628000 [ERR] switch_core_sqldb.c:404 SQL ERR [no
>>> such table: calls]
>>> drop table calls
>>> 2010-02-19 10:59:54.630000 [ERR] switch_core_sqldb.c:404 SQL ERR [no
>>> such table: interfaces]
>>> drop table interfaces
>>> 2010-02-19 10:59:54.631000 [ERR] switch_core_sqldb.c:404 SQL ERR [no
>>> such table: tasks]
>>> drop table tasks
>>> 2010-02-19 10:59:54.634000 [DEBUG] switch_core_sqldb.c:765 SQL ERR
>>> [no
>>> such table: aliases]
>>> [select hostname from aliases]
>>> Auto Generating Table!
>>> 2010-02-19 10:59:54.636000 [DEBUG] switch_core_sqldb.c:772 SQL ERR
>>> [no
>>> such table: aliases]
>>> [CREATE TABLE aliases (
>>> sticky  INTEGER,
>>> alias  VARCHAR(128),
>>> command  VARCHAR(4096),
>>> hostname VARCHAR(256)
>>> );
>>> ]
>>> 2010-02-19 10:59:54.644000 [DEBUG] switch_core_sqldb.c:765 SQL ERR
>>> [no
>>> such table: nat]
>>> [select hostname from nat]
>>> Auto Generating Table!
>>> 2010-02-19 10:59:54.646000 [DEBUG] switch_core_sqldb.c:772 SQL ERR
>>> [no
>>> such table: nat]
>>> [CREATE TABLE nat (
>>> sticky  INTEGER,
>>>  port    INTEGER,
>>>  proto    INTEGER,
>>> hostname VARCHAR(256)
>>> );
>>> ]
>>>
>>> Am I expected to setup a SQL database to get this working? Or did it
>>> just setup one automatically?
>>>
>>> 2010-02-19 10:59:59.684000 [CRIT] switch_loadable_module.c:882 Error
>>> Loading module /opt/freeswitch/mod/mod_voipcodecs.so
>>> **libjpeg.so.62: cannot open shared object file: No such file or
>>> directory**
>>> 2010-02-19 10:59:59.686000 [CRIT] switch_loadable_module.c:882 Error
>>> Loading module /opt/freeswitch/mod/mod_g723_1.so
>>> **/opt/freeswitch/mod/mod_g723_1.so: cannot open shared object file:
>>> No such file or directory**
>>> 2010-02-19 10:59:59.687000 [CRIT] switch_loadable_module.c:882 Error
>>> Loading module /opt/freeswitch/mod/mod_g729.so
>>> **/opt/freeswitch/mod/mod_g729.so: cannot open shared object file:  
>>> No
>>> such file or directory**
>>> 2010-02-19 10:59:59.688000 [CRIT] switch_loadable_module.c:882 Error
>>> Loading module /opt/freeswitch/mod/mod_amr.so
>>> **/opt/freeswitch/mod/mod_amr.so: cannot open shared object file: No
>>> such file or directory**
>>> 2010-02-19 10:59:59.835000 [CRIT] switch_loadable_module.c:882 Error
>>> Loading module /opt/freeswitch/mod/mod_file_string.so
>>> **/opt/freeswitch/mod/mod_file_string.so: cannot open shared object
>>> file: No such file or directory**
>>> 2010-02-19 11:00:00.143000 [CRIT] switch_loadable_module.c:882 Error
>>> Loading module /opt/freeswitch/mod/mod_say_ru.so
>>> **/opt/freeswitch/mod/mod_say_ru.so: cannot open shared object file:
>>> No such file or directory**
>>>
>>> suspect I don't really need to worry about some of these. I assume
>>> there is a config file somewhere where I can disable these options.
>>>
>>>
>>> Ok, as a really pathetic question, now I have started it, how do I
>>> stop it?
>>>
>>> freeswitch at voyage> halt
>>> Unknown Command: halt
>>> freeswitch at voyage> quit
>>> Unknown Command: quit
>>> freeswitch at voyage> exit
>>> Unknown Command: exit
>>> freeswitch at voyage> bye
>>> Unknown Command: bye
>>>
>>> -- 
>>> Brian May <brian at microcomaustralia.com.au>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>>> users
>>> http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>> users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org



From yehavi.bourvine at gmail.com  Thu Feb 18 23:46:04 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Fri, 19 Feb 2010 09:46:04 +0200
Subject: [Freeswitch-users] Announcement: FreeSWITCH Add Broadsoft SCA
	Support
In-Reply-To: <2F87B95C-EA23-4072-B845-3F7AE0617BE3@jerris.com>
References: <87f2f3b91001111742k691966cfs82515db77c13d4d4@mail.gmail.com>
	<d361ca7b1002170549g608d6e16vb2eb3f087fff9752@mail.gmail.com>
	<191c3a031002170556p5dfd4c55le3c279f194808d92@mail.gmail.com>
	<191c3a031002170557g5d8c9df9xf40d1b74b9a6f19d@mail.gmail.com>
	<d361ca7b1002170700x15630ab2tabc6f3d91ebc3303@mail.gmail.com>
	<ADC8B813-4161-481C-877E-93FC012858B4@freeswitch.org>
	<d361ca7b1002182101p60b4e8f8hf4dc95e1d2251e30@mail.gmail.com>
	<8b1c9cda1002182144m65ccf979w71384a1e7a55e2ea@mail.gmail.com>
	<d361ca7b1002182154o52352c62h8c0ec02fa66e8444@mail.gmail.com>
	<2F87B95C-EA23-4072-B845-3F7AE0617BE3@jerris.com>
Message-ID: <d361ca7b1002182346k71fd602fh6e6725834a6de528@mail.gmail.com>

A jira issue has been created:
*MODSOFIA-61*<http://jira.freeswitch.org/browse/MODSOFIA-61>
.

                  Thanks, __Yehavi:

 2010/2/19 Michael Jerris <mike at jerris.com>

>  If this issue is not already on jira could you please make sure it gets
> added?
>
> Mike
>
>
> On Feb 19, 2010, at 12:54 AM, Yehavi Bourvine <yehavi.bourvine at gmail.com>
> wrote:
>
>     Hello Gabe,
>
>   As you can see - Brian is actively investigating it, so you can expect
> for some fix soon...
>
>                        Regards, __Yehavi:
>
> 2010/2/19 Gabriel Kuri <gkuri at ieee.org>
>
>> >  When a call arrives, both ring; the one that did not answer gets only a
>> > cancel mesage without any further notification that the extension is in
>> use
>> > by the other phone.
>>
>> These are the same exact symptoms I posted about earlier this week,
>> with the Cisco SPA-5xx series phones. I still have yet to figure out
>> why this is happening, if you find out what's going on, please post
>> back the solution, I'd like to know the resolution.
>>
>> Thanks,
>> Gabe
>>
>>
>>
>> On Thu, Feb 18, 2010 at 9:01 PM, Yehavi Bourvine
>> <yehavi.bourvine at gmail.com> wrote:
>> > Thanks Brian. It now works better, but not fully (using 16659M).
>> >
>> > What happens is:
>> >
>> > When one of the Polycoms seize the line it is ok - the other phone gets
>> > notification and the extension status is "in use".
>> > When one of the Polycom phones initiates a call - all is ok:
>> >
>> > The other side sees that the extension is in use.
>> > When it is put to hold all phones who share this extension see it and
>> can
>> > pick the call.
>> >
>>
>> >
>>  >                          Thanks! __Yehavi:
>> >
>> > 2010/2/17 Brian West <brian at freeswitch.org>
>> >>
>> >> Step 1. Enable manage-shared-appearance=true
>> >>
>> >> Step 2. Now in the phone's config Configure the phone as usually, set
>> the
>> >> line shared and DO NOT set the third party name.
>> >>
>> >> Step 3. Reboot
>> >>
>> >> It should work.
>> >>
>> >> I wish someone that has this working would write some wiki docs these
>> >> threads about it not working are getting rather old when I know for a
>> fact
>> >> they work fine.
>> >>
>> >> The gateway info missing is a gateway you have configured getting a
>> >> notify.  It has nothing to do with SCA.
>> >>
>> >> /b
>> >>
>> >> On Feb 17, 2010, at 9:00 AM, Yehavi Bourvine wrote:
>> >>
>> >> > .
>> >>
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>  _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100219/07c5c86b/attachment-0002.html 

From pmhshz at gmail.com  Fri Feb 19 00:02:43 2010
From: pmhshz at gmail.com (MohammedShehzad)
Date: Fri, 19 Feb 2010 13:32:43 +0530
Subject: [Freeswitch-users] stream a file multicast with mod_esf
In-Reply-To: <DBA4E291-9377-489A-841A-1A3D83880DF2@jerris.com>
References: <B98B6921358D0C49BD21CD023F1F52C302623B4B@hbghs04a.ww300.siemens.net>
	<B98B6921358D0C49BD21CD023F1F52C302623BE0@hbghs04a.ww300.siemens.net>
	<7C996136-963A-42D2-B0FE-D2729F968E52@freeswitch.org>
	<27071973.post@talk.nabble.com>
	<378CDEC3-5A41-4EF9-927A-311F2531E6AB@freeswitch.org>
	<a792304b1001080725y19ff28camd37425927550f311@mail.gmail.com>
	<191c3a031001081848w9036207yf6a4b473ec233243@mail.gmail.com>
	<85C74329-0D7C-48B2-B7A8-D4AA67C6E3F2@freeswitch.org>
	<a792304b1002182209m43ada13cp8ed3d91ad509f0a2@mail.gmail.com>
	<DBA4E291-9377-489A-841A-1A3D83880DF2@jerris.com>
Message-ID: <a792304b1002190002u50ec6912g1905ffc63a119174@mail.gmail.com>

On Fri, Feb 19, 2010 at 12:59 PM, Michael Jerris <mike at jerris.com> wrote:

> Listening on multicast is noting special for multicast, it is just
> like reading any other udp socket
>
> Mike
>
> Correct, but I have to play those audio stream back to caller taking care
of the audio codec  and other things, do anybody have any idea in that part?
Please let me know that.
-- 

-MohammedShehzad
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100219/6e6998d6/attachment-0002.html 

From technical at ttnc.co.uk  Fri Feb 19 01:26:25 2010
From: technical at ttnc.co.uk (TTNC - Technical)
Date: Fri, 19 Feb 2010 09:26:25 +0000
Subject: [Freeswitch-users] mod_fax undefined symbol
In-Reply-To: <0F7476C7-FA7E-441F-9E17-EB2D23CD6055@jerris.com>
References: <6FCE44FF-70C1-412D-B722-4C8D359381BA@ttnc.co.uk>
	<0F7476C7-FA7E-441F-9E17-EB2D23CD6055@jerris.com>
Message-ID: <EBF5F462-9A4F-4699-95FB-6947A9D2D55C@ttnc.co.uk>

Hi Michel

I'm running the latest trunk I believe, r16700.

I've opened a bug report: http://jira.freeswitch.org/browse/FSMOD-37

I had libtiff4-dev installed as per the Debian setup instructions on the wiki, wasn't sure whether it was important as I could see tiff-3.8.2 in the libs folder in FreeSWITCH.

If you could come back to me or take a look at the bug report at your earliest convenience, as it's affecting service for us, it'd be appreciated.

Thanks

Russ


On 19 Feb 2010, at 07:22, Michael Jerris wrote:

> Please open a bug on http://jira.freeswitch.org for this issue.  As a  
> note, we use our own copy of libtiff, statically linked to the  
> module.  Is this recent trunk or something older?
> 
> On Feb 18, 2010, at 8:57 PM, TTNC - Technical <technical at ttnc.co.uk>  
> wrote:
> 
>> Hi Guys
>> 
>> I'm having trouble getting mod_fax to load. Running on Debian  
>> testing (squeeze). Everything is installed as per - http://wiki.freeswitch.org/wiki/Installation_Guide 
>> . (dpkg-buildpackage etc)
>> 
>> When trying to load the fax module I get:
>> 
>> 2010-02-19 01:48:17.554935 [CRIT] switch_loadable_module.c:882 Error  
>> Loading module /opt/freeswitch/mod/mod_fax.so
>> **/opt/freeswitch/mod/mod_fax.so: undefined symbol:  
>> TIFFDefaultStripSize**
>> 
>> And when a fax is sent, I'm getting:
>> 
>> 2010-02-19 01:04:57.355330 [ERR] switch_core_session.c:1490 Invalid  
>> Application rxfax
>> 
>> I guess because mod_fax isn't loaded.
>> 
>> I've got libtiff4 and libtiff4-dev installed:
>> 
>> ii  libtiff4                             3.9.2-2                   
>> Tag Image File Format (TIFF) library
>> ii  libtiff4-dev                         3.9.2-2                   
>> Tag Image File Format library (TIFF), development files
>> ii  libtiffxx0c2                         3.9.2-2                   
>> Tag Image File Format (TIFF) library -- C++ interface
>> 
>> Just tried updating to the latest svn trunk (16700M) and it hasn't  
>> made any difference.
>> 
>>> From googling, it suggests that it could be because the module is  
>>> complied against a different one currently running on the system,  
>>> however I'm not sure how this can be the case, there is only the  
>>> one version installed.
>> 
>> Any suggestions as to what I can try?
>> 
>> Any help appreciated
>> 
>> Russ
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>> users
>> http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From gmaruzz at celliax.org  Fri Feb 19 01:46:57 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Fri, 19 Feb 2010 10:46:57 +0100
Subject: [Freeswitch-users] mod_fax undefined symbol
In-Reply-To: <EBF5F462-9A4F-4699-95FB-6947A9D2D55C@ttnc.co.uk>
References: <6FCE44FF-70C1-412D-B722-4C8D359381BA@ttnc.co.uk> 
	<0F7476C7-FA7E-441F-9E17-EB2D23CD6055@jerris.com>
	<EBF5F462-9A4F-4699-95FB-6947A9D2D55C@ttnc.co.uk>
Message-ID: <7b197bef1002190146x53492b85va88e4025c59e37f3@mail.gmail.com>

On Fri, Feb 19, 2010 at 10:26 AM, TTNC - Technical <technical at ttnc.co.uk> wrote:
> I'm running the latest trunk I believe, r16700.
>
> I've opened a bug report: http://jira.freeswitch.org/browse/FSMOD-37
>
> I had libtiff4-dev installed as per the Debian setup instructions on the wiki, wasn't sure whether it was important as I could see tiff-3.8.2 in the libs folder in FreeSWITCH.

FS mod_fax uses FS provided libtiff.

If you have not modified the Makefile, mod_fax is built like that, to
use the FS provided libtiff.

Have you modified the Makefile?

-giovanni


>
> If you could come back to me or take a look at the bug report at your earliest convenience, as it's affecting service for us, it'd be appreciated.
>
> Thanks
>
> Russ
>
>
> On 19 Feb 2010, at 07:22, Michael Jerris wrote:
>
>> Please open a bug on http://jira.freeswitch.org for this issue. ?As a
>> note, we use our own copy of libtiff, statically linked to the
>> module. ?Is this recent trunk or something older?
>>
>> On Feb 18, 2010, at 8:57 PM, TTNC - Technical <technical at ttnc.co.uk>
>> wrote:
>>
>>> Hi Guys
>>>
>>> I'm having trouble getting mod_fax to load. Running on Debian
>>> testing (squeeze). Everything is installed as per - http://wiki.freeswitch.org/wiki/Installation_Guide
>>> . (dpkg-buildpackage etc)
>>>
>>> When trying to load the fax module I get:
>>>
>>> 2010-02-19 01:48:17.554935 [CRIT] switch_loadable_module.c:882 Error
>>> Loading module /opt/freeswitch/mod/mod_fax.so
>>> **/opt/freeswitch/mod/mod_fax.so: undefined symbol:
>>> TIFFDefaultStripSize**
>>>
>>> And when a fax is sent, I'm getting:
>>>
>>> 2010-02-19 01:04:57.355330 [ERR] switch_core_session.c:1490 Invalid
>>> Application rxfax
>>>
>>> I guess because mod_fax isn't loaded.
>>>
>>> I've got libtiff4 and libtiff4-dev installed:
>>>
>>> ii ?libtiff4 ? ? ? ? ? ? ? ? ? ? ? ? ? ? 3.9.2-2
>>> Tag Image File Format (TIFF) library
>>> ii ?libtiff4-dev ? ? ? ? ? ? ? ? ? ? ? ? 3.9.2-2
>>> Tag Image File Format library (TIFF), development files
>>> ii ?libtiffxx0c2 ? ? ? ? ? ? ? ? ? ? ? ? 3.9.2-2
>>> Tag Image File Format (TIFF) library -- C++ interface
>>>
>>> Just tried updating to the latest svn trunk (16700M) and it hasn't
>>> made any difference.
>>>
>>>> From googling, it suggests that it could be because the module is
>>>> complied against a different one currently running on the system,
>>>> however I'm not sure how this can be the case, there is only the
>>>> one version installed.
>>>
>>> Any suggestions as to what I can try?
>>>
>>> Any help appreciated
>>>
>>> Russ
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>>> users
>>> http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618



From technical at ttnc.co.uk  Fri Feb 19 02:12:32 2010
From: technical at ttnc.co.uk (TTNC - Technical)
Date: Fri, 19 Feb 2010 10:12:32 +0000
Subject: [Freeswitch-users] mod_fax undefined symbol
In-Reply-To: <7b197bef1002190146x53492b85va88e4025c59e37f3@mail.gmail.com>
References: <6FCE44FF-70C1-412D-B722-4C8D359381BA@ttnc.co.uk>
	<0F7476C7-FA7E-441F-9E17-EB2D23CD6055@jerris.com>
	<EBF5F462-9A4F-4699-95FB-6947A9D2D55C@ttnc.co.uk>
	<7b197bef1002190146x53492b85va88e4025c59e37f3@mail.gmail.com>
Message-ID: <496A4C3D-D185-4A86-BCA7-E4F5C05DA929@ttnc.co.uk>


On 19 Feb 2010, at 09:46, Giovanni Maruzzelli wrote:

> On Fri, Feb 19, 2010 at 10:26 AM, TTNC - Technical <technical at ttnc.co.uk> wrote:
>> I'm running the latest trunk I believe, r16700.
>> 
>> I've opened a bug report: http://jira.freeswitch.org/browse/FSMOD-37
>> 
>> I had libtiff4-dev installed as per the Debian setup instructions on the wiki, wasn't sure whether it was important as I could see tiff-3.8.2 in the libs folder in FreeSWITCH.
> 
> FS mod_fax uses FS provided libtiff.
> 
> If you have not modified the Makefile, mod_fax is built like that, to
> use the FS provided libtiff.
> 
> Have you modified the Makefile?
> 
> -giovanni

I haven't touched the make file, I followed the Debian install guide: http://wiki.freeswitch.org/wiki/Installation_Guide#Debian_Linux

Could it be that the Debian version of libtiff4-dev is conflicting with the FreeSWITCH static one? Doesn't seem likely...


From technical at ttnc.co.uk  Fri Feb 19 02:26:02 2010
From: technical at ttnc.co.uk (TTNC - Technical)
Date: Fri, 19 Feb 2010 10:26:02 +0000
Subject: [Freeswitch-users] mod_fax undefined symbol
In-Reply-To: <496A4C3D-D185-4A86-BCA7-E4F5C05DA929@ttnc.co.uk>
References: <6FCE44FF-70C1-412D-B722-4C8D359381BA@ttnc.co.uk>
	<0F7476C7-FA7E-441F-9E17-EB2D23CD6055@jerris.com>
	<EBF5F462-9A4F-4699-95FB-6947A9D2D55C@ttnc.co.uk>
	<7b197bef1002190146x53492b85va88e4025c59e37f3@mail.gmail.com>
	<496A4C3D-D185-4A86-BCA7-E4F5C05DA929@ttnc.co.uk>
Message-ID: <8D506E69-CB30-40BD-8AB9-9524C18E4503@ttnc.co.uk>

On 19 Feb 2010, at 10:12, TTNC - Technical wrote:

> On 19 Feb 2010, at 09:46, Giovanni Maruzzelli wrote:
> 
>> FS mod_fax uses FS provided libtiff.
>> 
>> If you have not modified the Makefile, mod_fax is built like that, to
>> use the FS provided libtiff.
>> 
>> Have you modified the Makefile?
>> 
>> -giovanni
> 
> I haven't touched the make file, I followed the Debian install guide: http://wiki.freeswitch.org/wiki/Installation_Guide#Debian_Linux
> 
> Could it be that the Debian version of libtiff4-dev is conflicting with the FreeSWITCH static one? Doesn't seem likely...


Just as another note - I've tried an install the 'freeswitch' way rather than the 'Debian' one.

dpkg-buildpackage in FreeSWITCH root requires the Debian package libtiff4-dev to be installed before it'll work. I've uninstall all Debian libtiff* packages and gone into the FreeSWITCH root and done a 'make && make install'.

Same problem persists:

2010-02-19 10:23:01.277590 [CRIT] switch_loadable_module.c:882 Error Loading module /opt/freeswitch/mod/mod_fax.so
**/opt/freeswitch/mod/mod_fax.so: undefined symbol: TIFFDefaultStripSize**

I guess that sort of shows it's an internal problem with the FreeSWITCH libtiff or mod_fax rather than the Debian package?

I'll update the bug report with this.

Russ
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100219/ebba2d99/attachment-0002.html 

From jason at jasonjgw.net  Fri Feb 19 02:31:28 2010
From: jason at jasonjgw.net (Jason White)
Date: Fri, 19 Feb 2010 21:31:28 +1100
Subject: [Freeswitch-users] mod_fax undefined symbol
In-Reply-To: <496A4C3D-D185-4A86-BCA7-E4F5C05DA929@ttnc.co.uk>
References: <6FCE44FF-70C1-412D-B722-4C8D359381BA@ttnc.co.uk>
	<0F7476C7-FA7E-441F-9E17-EB2D23CD6055@jerris.com>
	<EBF5F462-9A4F-4699-95FB-6947A9D2D55C@ttnc.co.uk>
	<7b197bef1002190146x53492b85va88e4025c59e37f3@mail.gmail.com>
	<496A4C3D-D185-4A86-BCA7-E4F5C05DA929@ttnc.co.uk>
Message-ID: <20100219103128.GA30809@jdc.jasonjgw.net>

TTNC - Technical <technical at ttnc.co.uk> wrote:
 
> I haven't touched the make file, I followed the Debian install guide:
> http://wiki.freeswitch.org/wiki/Installation_Guide#Debian_Linux
> 
> Could it be that the Debian version of libtiff4-dev is conflicting with the
> FreeSWITCH static one? Doesn't seem likely...

It isn't likely. You could always remove that package and recompile.

I can confirm this bug in my Debian build of R16654.

(I hadn't noticed it before, since I don't need and therefore don't load
mod_fax).




From gmaruzz at celliax.org  Fri Feb 19 02:34:19 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Fri, 19 Feb 2010 11:34:19 +0100
Subject: [Freeswitch-users] mod_fax undefined symbol
In-Reply-To: <8D506E69-CB30-40BD-8AB9-9524C18E4503@ttnc.co.uk>
References: <6FCE44FF-70C1-412D-B722-4C8D359381BA@ttnc.co.uk> 
	<0F7476C7-FA7E-441F-9E17-EB2D23CD6055@jerris.com>
	<EBF5F462-9A4F-4699-95FB-6947A9D2D55C@ttnc.co.uk> 
	<7b197bef1002190146x53492b85va88e4025c59e37f3@mail.gmail.com> 
	<496A4C3D-D185-4A86-BCA7-E4F5C05DA929@ttnc.co.uk>
	<8D506E69-CB30-40BD-8AB9-9524C18E4503@ttnc.co.uk>
Message-ID: <7b197bef1002190234m58c7e8cbyd945af9ee0f5a3a@mail.gmail.com>

On Fri, Feb 19, 2010 at 11:26 AM, TTNC - Technical <technical at ttnc.co.uk> wrote:
> Just as another note - I've tried an install the 'freeswitch' way rather
> than the 'Debian' one.
> dpkg-buildpackage in FreeSWITCH root requires the Debian package
> libtiff4-dev to be installed before it'll work. I've uninstall all Debian
> libtiff* packages and gone into the FreeSWITCH root and done a 'make && make
> install'.
> Same problem persists:
> 2010-02-19 10:23:01.277590 [CRIT] switch_loadable_module.c:882 Error Loading
> module /opt/freeswitch/mod/mod_fax.so
> **/opt/freeswitch/mod/mod_fax.so: undefined symbol: TIFFDefaultStripSize**
> I guess that sort of shows it's an internal problem with the FreeSWITCH
> libtiff or mod_fax rather than the Debian package?
> I'll update the bug report with this.

try first with a fresh checkout from svn (not the one you are using
now, checkout in another dir), then ./bootstrap.sh and ./configure,
then make && make install

So you'll be sure of the results for the report.

-gm

> Russ
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618



From gmaruzz at celliax.org  Fri Feb 19 02:41:23 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Fri, 19 Feb 2010 11:41:23 +0100
Subject: [Freeswitch-users] mod_fax undefined symbol
In-Reply-To: <20100219103128.GA30809@jdc.jasonjgw.net>
References: <6FCE44FF-70C1-412D-B722-4C8D359381BA@ttnc.co.uk> 
	<0F7476C7-FA7E-441F-9E17-EB2D23CD6055@jerris.com>
	<EBF5F462-9A4F-4699-95FB-6947A9D2D55C@ttnc.co.uk> 
	<7b197bef1002190146x53492b85va88e4025c59e37f3@mail.gmail.com> 
	<496A4C3D-D185-4A86-BCA7-E4F5C05DA929@ttnc.co.uk>
	<20100219103128.GA30809@jdc.jasonjgw.net>
Message-ID: <7b197bef1002190241k7e061323nb663b14800e78232@mail.gmail.com>

On Fri, Feb 19, 2010 at 11:31 AM, Jason White <jason at jasonjgw.net> wrote:
>
> I can confirm this bug in my Debian build of R16654.
>
> (I hadn't noticed it before, since I don't need and therefore don't load
> mod_fax).

oooops, I can confirm it too, in a non debian (as in "normal") build.

>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618



From ivdreg at gmail.com  Fri Feb 19 02:42:27 2010
From: ivdreg at gmail.com (ivdreg ivdreg)
Date: Fri, 19 Feb 2010 12:42:27 +0200
Subject: [Freeswitch-users] SDP With T.38 in INVITE Problem
Message-ID: <b437a0e1002190242y2aa64696j91e8bc76f7ae2a92@mail.gmail.com>

Hi all,

Dose someone have a problem that if there T.38 in coming from gateway
FreeSwitch drops the call because of media error ? As I see from log only
T.38 port is zero and SDP has also media port. Is it possible to configure
FS to do not break a call but if media is OK.

2010-02-19 12:26:14.305244 [NOTICE] switch_channel.c:666 New Channel
sofia/backend/XXXXXXXXXX at 10.10.1.110:7065[6cd9f634-411d-df11-99ca-003048bb99cc]
2010-02-19 12:26:14.305244 [DEBUG] mod_sofia.c:3369 (sofia/backend/
XXXXXXXXXX at 10.10.1.110:7065) State Change CS_NEW -> CS_INIT
2010-02-19 12:26:14.305244 [DEBUG] switch_core_session.c:1018 Send signal
sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [BREAK]
2010-02-19 12:26:14.305244 [DEBUG] switch_core_state_machine.c:314
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) Running State Change CS_INIT
2010-02-19 12:26:14.305244 [DEBUG] switch_core_state_machine.c:338
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State INIT
2010-02-19 12:26:14.305244 [DEBUG] mod_sofia.c:83 sofia/backend/
XXXXXXXXXX at 10.10.1.110:7065 SOFIA INIT
2010-02-19 12:26:14.305244 [DEBUG] sofia_glue.c:1377 sofia/backend/
XXXXXXXXXX at 10.10.1.110:7065 Patched SDP
---
v=0
o=- 3779949069 5166785777234550622 IN IP4 206.132.232.212
s=session
t=0 0
m=audio 21108 RTP/AVP 18 4 8 0
c=IN IP4 10.10.1.110
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
m=image 21108 udptl t38
c=IN IP4 10.10.1.110
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxUdpEC:t38UDPRedundancy
a=T38FaxRateManagement:transferredTCF

+++
v=0
o=- 3779949069 5166785777234550622 IN IP4 206.132.232.212
s=session
t=0 0
m=audio 17058 RTP/AVP 18 4 8 0
c=IN IP4 10.10.1.100
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
m=image 17058 udptl t38
c=IN IP4 10.10.1.100
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxUdpEC:t38UDPRedundancy
a=T38FaxRateManagement:transferredTCF

2010-02-19 12:26:14.305244 [DEBUG] mod_sofia.c:117 (sofia/backend/
XXXXXXXXXX at 10.10.1.110:7065) State Change CS_INIT -> CS_ROUTING
......
2010-02-19 12:26:21.255201 [DEBUG] sofia.c:4135 Remote SDP:
v=0
o=FreeSWITCH 1266548331 1266548332 IN IP4 10.10.1.110
s=FreeSWITCH
c=IN IP4 10.10.1.110
t=0 0
*m=audio 26850 RTP/AVP 8*
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
*m=image 0 udptl 19*

2010-02-19 12:26:21.255201 [DEBUG] sofia.c:4124 Channel sofia/backend/
XXXXXXXXXX at 10.10.1.110:7065 entering state [ready][200]
2010-02-19 12:26:21.255201 [DEBUG] switch_channel.c:2285 Send signal
sofia/backend/YYYYYYYYYY at 10.10.1.110 [BREAK]
2010-02-19 12:26:21.255201 [NOTICE] sofia.c:4668 Channel [sofia/backend/
XXXXXXXXXX at 10.10.1.110:7065] has been answered
2010-02-19 12:26:21.255201 [DEBUG] sofia_glue.c:2288 Set Codec
sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 PROXY/8000 20 ms 160 samples
*2010-02-19 12:26:21.255201 [DEBUG] sofia_glue.c:2571 PROXY AUDIO RTP
[sofia/backend/XXXXXXXXXX at 10.10.1.110:7065]
10.10.1.100:17058->10.10.1.110:0codec: 0 ms: 20
2010-02-19 12:26:21.255201 [ERR] sofia_glue.c:2879 AUDIO RTP REPORTS ERROR:
[Missing remote port]
2010-02-19 12:26:21.255201 [NOTICE] sofia_glue.c:2880 Hangup sofia/backend/
XXXXXXXXXX at 10.10.1.110:7065 [CS_CONSUME_MEDIA] [DESTINATION_OUT_OF_ORDER]*
2010-02-19 12:26:21.255201 [DEBUG] switch_channel.c:2063 Send signal
sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [KILL]
2010-02-19 12:26:21.255201 [DEBUG] switch_core_session.c:1018 Send signal
sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [BREAK]
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:314
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) Running State Change CS_HANGUP
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:499
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State HANGUP
2010-02-19 12:26:21.255201 [DEBUG] mod_sofia.c:411 Channel sofia/backend/
XXXXXXXXXX at 10.10.1.110:7065 hanging up, cause: DESTINATION_OUT_OF_ORDER
2010-02-19 12:26:21.255201 [DEBUG] mod_sofia.c:454 Sending BYE to
sofia/backend/XXXXXXXXXX at 10.10.1.110:7065
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:46
sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 Standard HANGUP, cause:
DESTINATION_OUT_OF_ORDER
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:499
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State HANGUP going to sleep
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:333
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State Change CS_HANGUP ->
CS_REPORTING
2010-02-19 12:26:21.255201 [DEBUG] switch_core_session.c:1018 Send signal
sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [BREAK]
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:314
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) Running State Change
CS_REPORTING
2010-02-19 12:26:21.255201 [DEBUG] switch_ivr_originate.c:3185 Originate
Resulted in Error Cause: 27 [DESTINATION_OUT_OF_ORDER]
2010-02-19 12:26:21.255201 [INFO] mod_dptools.c:2353 Originate Failed.
Cause: DESTINATION_OUT_OF_ORDER
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:590
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State REPORTING


Thanks
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100219/bcc1fd79/attachment-0002.html 

From gmaruzz at celliax.org  Fri Feb 19 02:44:03 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Fri, 19 Feb 2010 11:44:03 +0100
Subject: [Freeswitch-users] mod_fax undefined symbol
In-Reply-To: <7b197bef1002190241k7e061323nb663b14800e78232@mail.gmail.com>
References: <6FCE44FF-70C1-412D-B722-4C8D359381BA@ttnc.co.uk> 
	<0F7476C7-FA7E-441F-9E17-EB2D23CD6055@jerris.com>
	<EBF5F462-9A4F-4699-95FB-6947A9D2D55C@ttnc.co.uk> 
	<7b197bef1002190146x53492b85va88e4025c59e37f3@mail.gmail.com> 
	<496A4C3D-D185-4A86-BCA7-E4F5C05DA929@ttnc.co.uk>
	<20100219103128.GA30809@jdc.jasonjgw.net> 
	<7b197bef1002190241k7e061323nb663b14800e78232@mail.gmail.com>
Message-ID: <7b197bef1002190244m18aca4cfjb6a1c64b71e2a58@mail.gmail.com>

On Fri, Feb 19, 2010 at 11:41 AM, Giovanni Maruzzelli
<gmaruzz at celliax.org> wrote:
>
> oooops, I can confirm it too, in a non debian (as in "normal") build.

Double oooops, I had not built it ;)

I now built it, and loads flawlessly. So, "normal" as in non-debian,
build of mod_fax are OK



>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618



From technical at ttnc.co.uk  Fri Feb 19 02:58:46 2010
From: technical at ttnc.co.uk (TTNC - Technical)
Date: Fri, 19 Feb 2010 10:58:46 +0000
Subject: [Freeswitch-users] mod_fax undefined symbol
In-Reply-To: <7b197bef1002190241k7e061323nb663b14800e78232@mail.gmail.com>
References: <6FCE44FF-70C1-412D-B722-4C8D359381BA@ttnc.co.uk>
	<0F7476C7-FA7E-441F-9E17-EB2D23CD6055@jerris.com>
	<EBF5F462-9A4F-4699-95FB-6947A9D2D55C@ttnc.co.uk>
	<7b197bef1002190146x53492b85va88e4025c59e37f3@mail.gmail.com>
	<496A4C3D-D185-4A86-BCA7-E4F5C05DA929@ttnc.co.uk>
	<20100219103128.GA30809@jdc.jasonjgw.net>
	<7b197bef1002190241k7e061323nb663b14800e78232@mail.gmail.com>
Message-ID: <B05DBE65-CF57-4E13-9CE2-E490128F1C41@ttnc.co.uk>


On 19 Feb 2010, at 10:41, Giovanni Maruzzelli wrote:

> On Fri, Feb 19, 2010 at 11:31 AM, Jason White <jason at jasonjgw.net> wrote:
>> 
>> I can confirm this bug in my Debian build of R16654.
>> 
>> (I hadn't noticed it before, since I don't need and therefore don't load
>> mod_fax).
> 
> oooops, I can confirm it too, in a non debian (as in "normal") build.
> 

Hmm... that's a bit odd. You would have thought that if FreeSWITCH is compiled statically against it's own libtiff - then anything Debian-centric shouldn't affect it?


From gmaruzz at celliax.org  Fri Feb 19 03:08:25 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Fri, 19 Feb 2010 12:08:25 +0100
Subject: [Freeswitch-users] mod_fax undefined symbol
In-Reply-To: <B05DBE65-CF57-4E13-9CE2-E490128F1C41@ttnc.co.uk>
References: <6FCE44FF-70C1-412D-B722-4C8D359381BA@ttnc.co.uk> 
	<0F7476C7-FA7E-441F-9E17-EB2D23CD6055@jerris.com>
	<EBF5F462-9A4F-4699-95FB-6947A9D2D55C@ttnc.co.uk> 
	<7b197bef1002190146x53492b85va88e4025c59e37f3@mail.gmail.com> 
	<496A4C3D-D185-4A86-BCA7-E4F5C05DA929@ttnc.co.uk>
	<20100219103128.GA30809@jdc.jasonjgw.net> 
	<7b197bef1002190241k7e061323nb663b14800e78232@mail.gmail.com> 
	<B05DBE65-CF57-4E13-9CE2-E490128F1C41@ttnc.co.uk>
Message-ID: <7b197bef1002190308u375c16f0p9ed83d7e8e19cf89@mail.gmail.com>

On Fri, Feb 19, 2010 at 11:58 AM, TTNC - Technical <technical at ttnc.co.uk> wrote:
> Hmm... that's a bit odd. You would have thought that if FreeSWITCH is compiled statically against it's own libtiff - then anything Debian-centric shouldn't affect it?

yes, definitely

probably in your debian build libtiff is not compiled, or mod_fax is
not linked (statically) with it

"normal" build, as in non-debian, works fine
-gm


> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618



From technical at ttnc.co.uk  Fri Feb 19 04:04:21 2010
From: technical at ttnc.co.uk (TTNC - Technical)
Date: Fri, 19 Feb 2010 12:04:21 +0000
Subject: [Freeswitch-users] mod_fax undefined symbol
In-Reply-To: <7b197bef1002190234m58c7e8cbyd945af9ee0f5a3a@mail.gmail.com>
References: <6FCE44FF-70C1-412D-B722-4C8D359381BA@ttnc.co.uk>
	<0F7476C7-FA7E-441F-9E17-EB2D23CD6055@jerris.com>
	<EBF5F462-9A4F-4699-95FB-6947A9D2D55C@ttnc.co.uk>
	<7b197bef1002190146x53492b85va88e4025c59e37f3@mail.gmail.com>
	<496A4C3D-D185-4A86-BCA7-E4F5C05DA929@ttnc.co.uk>
	<8D506E69-CB30-40BD-8AB9-9524C18E4503@ttnc.co.uk>
	<7b197bef1002190234m58c7e8cbyd945af9ee0f5a3a@mail.gmail.com>
Message-ID: <D04CFC8A-4909-4E32-A480-567E712AE63A@ttnc.co.uk>

On 19 Feb 2010, at 10:34, Giovanni Maruzzelli wrote:

> On Fri, Feb 19, 2010 at 11:26 AM, TTNC - Technical <technical at ttnc.co.uk> wrote:
> 
> try first with a fresh checkout from svn (not the one you are using
> now, checkout in another dir), then ./bootstrap.sh and ./configure,
> then make && make install
> 
> So you'll be sure of the results for the report.
> 
> -gm

Totally clean build as suggested, checked out a new source tree, built the 'freeswitch' way:

./bootstrap.sh
./configure --prefix=/opt/freeswitch
make
make install

Same problem persists. 

Guess will have to wait to see what Michael can find out! :)

Russ




From rupa at rupa.com  Fri Feb 19 06:21:16 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Fri, 19 Feb 2010 08:21:16 -0600
Subject: [Freeswitch-users] 1.0.4 vs. trunk vs. bugs
In-Reply-To: <7716C919-BCD1-44FA-9ED2-5268F4C42100@jerris.com>
References: <5d3e0dc61002181741r57b91f86h4810de63c1a3d46d@mail.gmail.com>
	<20100219015118.GA12983@jdc.jasonjgw.net>
	<1266552605.7684.11.camel@local.freepabx.com>
	<616F57B5-182A-49A9-9825-2E5B72F5DB05@freeswitch.org>
	<7716C919-BCD1-44FA-9ED2-5268F4C42100@jerris.com>
Message-ID: <b39273531002190621l656f44a2s8b05bc4162aa5182@mail.gmail.com>

I would caution that maintaining a stable branch is going to be quite
challenging.  All commits against trunk will fall into 3 categories:

1) clearly bug fix
2) clearly new feature
3) a mix of the two

1 can probably be easily back ported and 2 would be not but what of 3?  We
don't split patches/commits based on a clear split between 1 and 2 so it
would be the job of the stable maintainer to figure it out, split it up and
then commit just the bug fix part.

I would argue that the churn in the stable branch would be sufficient to
make it a moving target just like trunk, just slower moving and one step
removed from tony ensuring everything is up to his standard.

I would also argue that at some point this project will clearly go from
"balls to the wall" development like now (lots of bug fixes and new features
all the time) to something more sane as it matures.  At some point going to
stable/dev might make sense.

Another thought.  Look at how the linux kernel is developed now.  There is
linus's branch which is essentially unstable.  It is the vendor's (distro)
job to pick a line in the sand and keep that kernel rev stable.  There is
help from people that have stepped up and maintain a "stable" kernel branch,
but that has nothing to do with the mainline development.

I can appreciate the pain that some people have with dealing with production
systems where you want stability above all else.  In reality you don't want
stability, you don't want surprising behavioral changes.  You want code that
doesn't change except in those areas that fix bugs in components that you
use.  But your component set and mine are different.  Once you are accepting
bug fixes for all components, the set that changes can churn quite a bit.

Anyway, just some food for thought.

I know that if I had to double commit (or at least consider
double committing)  every piece of code I'd get frustrated quickly.

On Fri, Feb 19, 2010 at 1:17 AM, Michael Jerris <mike at jerris.com> wrote:

> This seems a good time to note that we are still looking for volunteers to
> assist in maintaining a stable branch.  I can not do this without additional
> volunteer resources.  We have asked several times recently to fairly silent
> response.  If anyone is interested in assisting with this effort, please
> contact  me offlist and we can discuss further.
>
> Mike
>
> On Feb 18, 2010, at 11:16 PM, Brian West <brian at freeswitch.org> wrote:
>
> OK so I can sign you up for the stable team?  ;)  As per my previous email
> i'm 100% sure we would do a stable release if we had people tending to
> issues.  The only problem is you would have to be on IRC tending to issues
> because if tony sees someone asking about a problem he'll be diving in to
> fix it before they can say "I have this one".  This also means working in a
> similar manner we do already.  Our process is very chaotic at times but it
> has served us well so far.
>
> The goal is to leave Anthony alone so he can move forward and let the
> stable team manage the jira's and issues on the list related to stable.
>
> /b
>
> On Feb 18, 2010, at 10:10 PM, David Knell wrote:
>
>
> Lon Baker < <lon at kickasspixels.com>lon at kickasspixels.com> wrote:
>
>
> The development branch is where feature requests and non-critical bugs
>
> reports would be filed for the next production release.
>
>
> The current process leaves a gap between production ready and
>
> development code that may become greater over time.
>
>
> Going against the grain here, I agree with you.  The current way of
> doing things is, in my opinion, not well thought through - there's no
> reason to tag and release versions if the answer to any issue is 'make
> current', and support is not available unless that's been done.  Far
> better to either have meaningful releases with stable and devel
> branches, or not to have releases at all.
>
> --Dave
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100219/95e5af0a/attachment-0002.html 

From mgg at giagnocavo.net  Fri Feb 19 06:23:16 2010
From: mgg at giagnocavo.net (Michael Giagnocavo)
Date: Fri, 19 Feb 2010 09:23:16 -0500
Subject: [Freeswitch-users] 1.0.4 vs. trunk vs. bugs
In-Reply-To: <7716C919-BCD1-44FA-9ED2-5268F4C42100@jerris.com>
References: <5d3e0dc61002181741r57b91f86h4810de63c1a3d46d@mail.gmail.com>
	<20100219015118.GA12983@jdc.jasonjgw.net>
	<1266552605.7684.11.camel@local.freepabx.com>
	<616F57B5-182A-49A9-9825-2E5B72F5DB05@freeswitch.org>
	<7716C919-BCD1-44FA-9ED2-5268F4C42100@jerris.com>
Message-ID: <6E8D2069C08AA84A83D336E996AE4C67032C9D5E98@mse17be1.mse17.exchange.ms>

Maybe something that could work is a simple way for people to record the SVN number and config they?re using. It might encourage people to not get so stuck up on a released version number. And once they?re more comfortable with SVN, perhaps they?ll try head more often.

-Michael

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris
Sent: Friday, February 19, 2010 12:18 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] 1.0.4 vs. trunk vs. bugs

This seems a good time to note that we are still looking for volunteers to assist in maintaining a stable branch.  I can not do this without additional volunteer resources.  We have asked several times recently to fairly silent response.  If anyone is interested in assisting with this effort, please contact  me offlist and we can discuss further.

Mike

On Feb 18, 2010, at 11:16 PM, Brian West <brian at freeswitch.org<mailto:brian at freeswitch.org>> wrote:
OK so I can sign you up for the stable team?  ;)  As per my previous email i'm 100% sure we would do a stable release if we had people tending to issues.  The only problem is you would have to be on IRC tending to issues because if tony sees someone asking about a problem he'll be diving in to fix it before they can say "I have this one".  This also means working in a similar manner we do already.  Our process is very chaotic at times but it has served us well so far.

The goal is to leave Anthony alone so he can move forward and let the stable team manage the jira's and issues on the list related to stable.

/b

On Feb 18, 2010, at 10:10 PM, David Knell wrote:



Lon Baker <lon at kickasspixels.com<mailto:lon at kickasspixels.com>> wrote:

The development branch is where feature requests and non-critical bugs
reports would be filed for the next production release.

The current process leaves a gap between production ready and
development code that may become greater over time.

Going against the grain here, I agree with you.  The current way of
doing things is, in my opinion, not well thought through - there's no
reason to tag and release versions if the answer to any issue is 'make
current', and support is not available unless that's been done.  Far
better to either have meaningful releases with stable and devel
branches, or not to have releases at all.

--Dave

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100219/28ca4ed7/attachment-0002.html 

From mike at jerris.com  Fri Feb 19 06:37:25 2010
From: mike at jerris.com (Michael Jerris)
Date: Fri, 19 Feb 2010 09:37:25 -0500
Subject: [Freeswitch-users] mod_fax undefined symbol
In-Reply-To: <D04CFC8A-4909-4E32-A480-567E712AE63A@ttnc.co.uk>
References: <6FCE44FF-70C1-412D-B722-4C8D359381BA@ttnc.co.uk>
	<0F7476C7-FA7E-441F-9E17-EB2D23CD6055@jerris.com>
	<EBF5F462-9A4F-4699-95FB-6947A9D2D55C@ttnc.co.uk>
	<7b197bef1002190146x53492b85va88e4025c59e37f3@mail.gmail.com>
	<496A4C3D-D185-4A86-BCA7-E4F5C05DA929@ttnc.co.uk>
	<8D506E69-CB30-40BD-8AB9-9524C18E4503@ttnc.co.uk>
	<7b197bef1002190234m58c7e8cbyd945af9ee0f5a3a@mail.gmail.com>
	<D04CFC8A-4909-4E32-A480-567E712AE63A@ttnc.co.uk>
Message-ID: <2547643E-D259-46CE-919D-51393A4151BC@jerris.com>

My only thought without actually looking is this could be related to  
libjpeg being there during build but not when running.  Do you have  
jpeg devel package but not lib installed?

Mike

On Feb 19, 2010, at 7:04 AM, TTNC - Technical <technical at ttnc.co.uk>  
wrote:

> On 19 Feb 2010, at 10:34, Giovanni Maruzzelli wrote:
>
>> On Fri, Feb 19, 2010 at 11:26 AM, TTNC - Technical <technical at ttnc.co.uk 
>> > wrote:
>>
>> try first with a fresh checkout from svn (not the one you are using
>> now, checkout in another dir), then ./bootstrap.sh and ./configure,
>> then make && make install
>>
>> So you'll be sure of the results for the report.
>>
>> -gm
>
> Totally clean build as suggested, checked out a new source tree,  
> built the 'freeswitch' way:
>
> ./bootstrap.sh
> ./configure --prefix=/opt/freeswitch
> make
> make install
>
> Same problem persists.
>
> Guess will have to wait to see what Michael can find out! :)
>
> Russ
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org



From helmut.kuper at ewetel.de  Fri Feb 19 07:08:04 2010
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Fri, 19 Feb 2010 16:08:04 +0100
Subject: [Freeswitch-users] Question about sofia_contact
Message-ID: <4B7EA954.30402@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hello,

I try to setup a FS sofia sip-profile which allows me to have multiple
sip-profiles but one registration database. So I set the following
parameters:

<param name="challenge-realm" value="auto_from"/>
<param name="force-register-domain" value="$${domain}"/>
<param name="force-subscription-domain" value="$${domain}"/>
<param name="force-register-db-domain" value="$${domain}"/>

where domain is set to "mydomain". "sofia status profile internal"
delivers the following:


Call-ID:        3c26705038e5-vwlg8u5q9cwe
User:           2701 at mydomain
Contact:        <sip:2701 at 85.16.245.208:1024;line=l2wksedo>
Agent:          snom370/8.2.22
Status:         Registered(UDP)(unknown) EXP(2010-02-19 16:13:31)
Host:           ippbx-prod-node0
IP:             85.16.245.208
Port:           1024
Auth-User:      2701
Auth-Realm:     mydomain
MWI-Account:    2701 at mydomain



sofia_contact internal/2701 at mydomain delivers this:
error/user_not_registered

The Phone is fully functional.

I use SVN trunk 16601

regards
Helmut
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.4.7 (MingW32)

iD8DBQFLfqlT4tZeNddg3dwRAgk1AJ4gtXoVLn8anWF6BX2nijuvApoJiwCbBYyT
jo6HuOqH62g8Mmia5PEXII8=
=fDxi
-----END PGP SIGNATURE-----



From helmut.kuper at ewetel.de  Fri Feb 19 07:26:52 2010
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Fri, 19 Feb 2010 16:26:52 +0100
Subject: [Freeswitch-users] Question about sofia_contact
In-Reply-To: <4B7EA954.30402@ewetel.de>
References: <4B7EA954.30402@ewetel.de>
Message-ID: <4B7EADBC.1040001@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hi,

an update:
The corresponding select statement looks for sip_user="2701" and
sip_host="internal" in registration table.
This delivers of course no result because 2701 is registered with
sip_host="mydomain".


Hm any workaround or am I going in a wrong direction?


regards
Helmut


On 19.02.2010 16:08, Helmut Kuper wrote:
> Hello,
> 
> I try to setup a FS sofia sip-profile which allows me to have multiple
> sip-profiles but one registration database. So I set the following
> parameters:
> 
> <param name="challenge-realm" value="auto_from"/>
> <param name="force-register-domain" value="$${domain}"/>
> <param name="force-subscription-domain" value="$${domain}"/>
> <param name="force-register-db-domain" value="$${domain}"/>
> 
> where domain is set to "mydomain". "sofia status profile internal"
> delivers the following:
> 
> 
> Call-ID:        3c26705038e5-vwlg8u5q9cwe
> User:           2701 at mydomain
> Contact:        <sip:2701 at 85.16.245.208:1024;line=l2wksedo>
> Agent:          snom370/8.2.22
> Status:         Registered(UDP)(unknown) EXP(2010-02-19 16:13:31)
> Host:           ippbx-prod-node0
> IP:             85.16.245.208
> Port:           1024
> Auth-User:      2701
> Auth-Realm:     mydomain
> MWI-Account:    2701 at mydomain
> 
> 
> 
> sofia_contact internal/2701 at mydomain delivers this:
> error/user_not_registered
> 
> The Phone is fully functional.
> 
> I use SVN trunk 16601
> 
> regards
> Helmut
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.4.7 (MingW32)

iD8DBQFLfq274tZeNddg3dwRAl/pAJ96PIV5s/sZTbcJ/Pq4v/VirYteygCfXsML
j/Wdb5ApZx+x0q0uilvqkEU=
=NdFk
-----END PGP SIGNATURE-----



From technical at ttnc.co.uk  Fri Feb 19 07:30:09 2010
From: technical at ttnc.co.uk (TTNC - Technical)
Date: Fri, 19 Feb 2010 15:30:09 +0000
Subject: [Freeswitch-users] mod_fax undefined symbol
In-Reply-To: <2547643E-D259-46CE-919D-51393A4151BC@jerris.com>
References: <6FCE44FF-70C1-412D-B722-4C8D359381BA@ttnc.co.uk>
	<0F7476C7-FA7E-441F-9E17-EB2D23CD6055@jerris.com>
	<EBF5F462-9A4F-4699-95FB-6947A9D2D55C@ttnc.co.uk>
	<7b197bef1002190146x53492b85va88e4025c59e37f3@mail.gmail.com>
	<496A4C3D-D185-4A86-BCA7-E4F5C05DA929@ttnc.co.uk>
	<8D506E69-CB30-40BD-8AB9-9524C18E4503@ttnc.co.uk>
	<7b197bef1002190234m58c7e8cbyd945af9ee0f5a3a@mail.gmail.com>
	<D04CFC8A-4909-4E32-A480-567E712AE63A@ttnc.co.uk>
	<2547643E-D259-46CE-919D-51393A4151BC@jerris.com>
Message-ID: <CAC92986-7EA7-446B-8A95-50BF2FC25FCB@ttnc.co.uk>

Hi Mike

libjpeg installed, both -dev library:

voipin1:~# dpkg -l | grep jpeg
ii  libjpeg62                            6b-15                    The Independent JPEG Group's JPEG runtime library
ii  libjpeg62-dev                        6b-15                    Development files for the IJG JPEG library

/usr/lib/libjpeg.a
/usr/lib/libjpeg.la
/usr/lib/libjpeg.so
/usr/lib/libjpeg.so.62
/usr/lib/libjpeg.so.62.0.0
/usr/include/jpeglib.h
/usr/include/jpegint.h
/usr/include/jconfig.h
/usr/include/jerror.h
/usr/include/jmorecfg.h

Any other ideas? :-/


On 19 Feb 2010, at 14:37, Michael Jerris wrote:

> My only thought without actually looking is this could be related to  
> libjpeg being there during build but not when running.  Do you have  
> jpeg devel package but not lib installed?
> 
> Mike
> 
> On Feb 19, 2010, at 7:04 AM, TTNC - Technical <technical at ttnc.co.uk>  
> wrote:
> 
>> On 19 Feb 2010, at 10:34, Giovanni Maruzzelli wrote:
>> 
>>> On Fri, Feb 19, 2010 at 11:26 AM, TTNC - Technical <technical at ttnc.co.uk 
>>>> wrote:
>>> 
>>> try first with a fresh checkout from svn (not the one you are using
>>> now, checkout in another dir), then ./bootstrap.sh and ./configure,
>>> then make && make install
>>> 
>>> So you'll be sure of the results for the report.
>>> 
>>> -gm
>> 
>> Totally clean build as suggested, checked out a new source tree,  
>> built the 'freeswitch' way:
>> 
>> ./bootstrap.sh
>> ./configure --prefix=/opt/freeswitch
>> make
>> make install
>> 
>> Same problem persists.
>> 
>> Guess will have to wait to see what Michael can find out! :)
>> 
>> Russ
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>> users
>> http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From anthony.minessale at gmail.com  Fri Feb 19 07:32:14 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 19 Feb 2010 09:32:14 -0600
Subject: [Freeswitch-users] 1.0.4 vs. trunk vs. bugs
In-Reply-To: <b39273531002190621l656f44a2s8b05bc4162aa5182@mail.gmail.com>
References: <5d3e0dc61002181741r57b91f86h4810de63c1a3d46d@mail.gmail.com>
	<20100219015118.GA12983@jdc.jasonjgw.net>
	<1266552605.7684.11.camel@local.freepabx.com>
	<616F57B5-182A-49A9-9825-2E5B72F5DB05@freeswitch.org>
	<7716C919-BCD1-44FA-9ED2-5268F4C42100@jerris.com>
	<b39273531002190621l656f44a2s8b05bc4162aa5182@mail.gmail.com>
Message-ID: <191c3a031002190732i3ff1598eob746072b5a55219d@mail.gmail.com>

Here's the deal.

This is a community project and its public but it only has a small group
individuals who are "all-in" committed to the project.  I am the one who
started the project and who had to spend a solid 2 years completely alone
working towards my goals before others even showed interest.

Now we are growing very fast, we have a lot of feedback and we listen to it
regardless of our position on the subject and I will decide and make any
policy that I choose and feel is the best interest of this project.

It's reasonable that someone who is using the software wants to have
wonderful stable stepping stones to migrate towards the future on.  It's
also customary that most software, even when you pay a premium price for it,
does not meet those standards every time.  If you do find software that has
these graceful releases, they probably have a lot of people getting paid to
work diligently on it.  There are also many successful open source projects
with shiny revision numbers and packaged up with a bow but that is because
they have a dedicated team of people.

So, we don't have that long list of people.  We invited people to do it and
we had nobody step up.  So, this is what i am planning to do:

We are going to move our development to a branch, work on it from there and
push them down to trunk when we think its the best time.  This might not
always end up perfect but this is what we are going to do.  The actual
release versions are still just fancy road signs in a long journey towards
perfection.  We are still on 1.0 for almost 2 years with 5 micro release
that contain a 12 page change log each time.  We are as careful as we can be
about releases and we have no time to try to back-port patches to 6 month
old code with more than 2000 revisions in between them.

When we feel we are happy with 1.0 we will then branch to 1.1 and all this
stuff everyone wants, *IF* we get enough volunteers by that time to dedicate
their time to maintaining it.  If not we will make the best of what we
have........

This is the final word on this subject, feel free to quote me.





On Fri, Feb 19, 2010 at 8:21 AM, Rupa Schomaker <rupa at rupa.com> wrote:

> I would caution that maintaining a stable branch is going to be quite
> challenging.  All commits against trunk will fall into 3 categories:
>
> 1) clearly bug fix
> 2) clearly new feature
> 3) a mix of the two
>
> 1 can probably be easily back ported and 2 would be not but what of 3?  We
> don't split patches/commits based on a clear split between 1 and 2 so it
> would be the job of the stable maintainer to figure it out, split it up and
> then commit just the bug fix part.
>
> I would argue that the churn in the stable branch would be sufficient to
> make it a moving target just like trunk, just slower moving and one step
> removed from tony ensuring everything is up to his standard.
>
> I would also argue that at some point this project will clearly go from
> "balls to the wall" development like now (lots of bug fixes and new features
> all the time) to something more sane as it matures.  At some point going to
> stable/dev might make sense.
>
> Another thought.  Look at how the linux kernel is developed now.  There is
> linus's branch which is essentially unstable.  It is the vendor's (distro)
> job to pick a line in the sand and keep that kernel rev stable.  There is
> help from people that have stepped up and maintain a "stable" kernel branch,
> but that has nothing to do with the mainline development.
>
> I can appreciate the pain that some people have with dealing with
> production systems where you want stability above all else.  In reality you
> don't want stability, you don't want surprising behavioral changes.  You
> want code that doesn't change except in those areas that fix bugs in
> components that you use.  But your component set and mine are different.
>  Once you are accepting bug fixes for all components, the set that changes
> can churn quite a bit.
>
> Anyway, just some food for thought.
>
> I know that if I had to double commit (or at least consider
> double committing)  every piece of code I'd get frustrated quickly.
>
>
> On Fri, Feb 19, 2010 at 1:17 AM, Michael Jerris <mike at jerris.com> wrote:
>
>> This seems a good time to note that we are still looking for volunteers to
>> assist in maintaining a stable branch.  I can not do this without additional
>> volunteer resources.  We have asked several times recently to fairly silent
>> response.  If anyone is interested in assisting with this effort, please
>> contact  me offlist and we can discuss further.
>>
>> Mike
>>
>> On Feb 18, 2010, at 11:16 PM, Brian West <brian at freeswitch.org> wrote:
>>
>> OK so I can sign you up for the stable team?  ;)  As per my previous email
>> i'm 100% sure we would do a stable release if we had people tending to
>> issues.  The only problem is you would have to be on IRC tending to issues
>> because if tony sees someone asking about a problem he'll be diving in to
>> fix it before they can say "I have this one".  This also means working in a
>> similar manner we do already.  Our process is very chaotic at times but it
>> has served us well so far.
>>
>> The goal is to leave Anthony alone so he can move forward and let the
>> stable team manage the jira's and issues on the list related to stable.
>>
>> /b
>>
>> On Feb 18, 2010, at 10:10 PM, David Knell wrote:
>>
>>
>> Lon Baker < <lon at kickasspixels.com>lon at kickasspixels.com> wrote:
>>
>>
>> The development branch is where feature requests and non-critical bugs
>>
>> reports would be filed for the next production release.
>>
>>
>> The current process leaves a gap between production ready and
>>
>> development code that may become greater over time.
>>
>>
>> Going against the grain here, I agree with you.  The current way of
>> doing things is, in my opinion, not well thought through - there's no
>> reason to tag and release versions if the answer to any issue is 'make
>> current', and support is not available unless that's been done.  Far
>> better to either have meaningful releases with stable and devel
>> branches, or not to have releases at all.
>>
>> --Dave
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100219/67e469b1/attachment-0002.html 

From testeador01 at gmail.com  Fri Feb 19 07:41:16 2010
From: testeador01 at gmail.com (Milena)
Date: Fri, 19 Feb 2010 10:41:16 -0500
Subject: [Freeswitch-users] 1.0.4 vs. trunk vs. bugs
In-Reply-To: <6E8D2069C08AA84A83D336E996AE4C67032C9D5E98@mse17be1.mse17.exchange.ms>
References: <5d3e0dc61002181741r57b91f86h4810de63c1a3d46d@mail.gmail.com>
	<20100219015118.GA12983@jdc.jasonjgw.net>
	<1266552605.7684.11.camel@local.freepabx.com>
	<616F57B5-182A-49A9-9825-2E5B72F5DB05@freeswitch.org>
	<7716C919-BCD1-44FA-9ED2-5268F4C42100@jerris.com>
	<6E8D2069C08AA84A83D336E996AE4C67032C9D5E98@mse17be1.mse17.exchange.ms>
Message-ID: <c29b930b1002190741k215db689v2fc20a0574db3659@mail.gmail.com>

there's no reason to tag and release versions if the answer to any
issue is 'make current', and support is not available unless that's
been done.  Far better to either have meaningful releases with  stable
and devel branches, or not to have releases at all.


Well, the response on a stable version about a bug report that's been
already fixed would be: "apply this patch and retest, come back to us
if problem persists"

Isn't that exactly what make current does?
This way of developing may be not broken-trunk-proof, but from your
comment I think an easier way to approach Lon's suggestion would be
making a more understandable way for simple users (not fs devs) to
check what modules have been altered from their version to the latest,
in order to check if an issue has been solved or not, as of those
clients that only want "released versions", that's the hard part.

ps: hmmm Anthony's message makes my post outdated but i spent a lil
bit of thought on it so i post anyways ^^

-Milena

2010/2/19 Michael Giagnocavo <mgg at giagnocavo.net>

> Maybe something that could work is a simple way for people to record the
> SVN number and config they?re using. It might encourage people to not get so
> stuck up on a released version number. And once they?re more comfortable
> with SVN, perhaps they?ll try head more often.
>
>
>
> -Michael
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Jerris
> *Sent:* Friday, February 19, 2010 12:18 AM
>
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] 1.0.4 vs. trunk vs. bugs
>
>
>
> This seems a good time to note that we are still looking for volunteers to
> assist in maintaining a stable branch.  I can not do this without additional
> volunteer resources.  We have asked several times recently to fairly silent
> response.  If anyone is interested in assisting with this effort, please
> contact  me offlist and we can discuss further.
>
>
>
> Mike
>
>
> On Feb 18, 2010, at 11:16 PM, Brian West <brian at freeswitch.org> wrote:
>
> OK so I can sign you up for the stable team?  ;)  As per my previous email
> i'm 100% sure we would do a stable release if we had people tending to
> issues.  The only problem is you would have to be on IRC tending to issues
> because if tony sees someone asking about a problem he'll be diving in to
> fix it before they can say "I have this one".  This also means working in a
> similar manner we do already.  Our process is very chaotic at times but it
> has served us well so far.
>
>
>
> The goal is to leave Anthony alone so he can move forward and let the
> stable team manage the jira's and issues on the list related to stable.
>
>
>
> /b
>
>
>
> On Feb 18, 2010, at 10:10 PM, David Knell wrote:
>
>
>
>
> Lon Baker <lon at kickasspixels.com> wrote:
>
>
>
> The development branch is where feature requests and non-critical bugs
>
> reports would be filed for the next production release.
>
>
>
> The current process leaves a gap between production ready and
>
> development code that may become greater over time.
>
>
> Going against the grain here, I agree with you.  The current way of
> doing things is, in my opinion, not well thought through - there's no
> reason to tag and release versions if the answer to any issue is 'make
> current', and support is not available unless that's been done.  Far
> better to either have meaningful releases with stable and devel
> branches, or not to have releases at all.
>
> --Dave
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100219/c8ea5705/attachment-0002.html 

From jalsot at gmail.com  Fri Feb 19 08:12:27 2010
From: jalsot at gmail.com (Tamas)
Date: Fri, 19 Feb 2010 17:12:27 +0100
Subject: [Freeswitch-users] mod_fax undefined symbol
In-Reply-To: <CAC92986-7EA7-446B-8A95-50BF2FC25FCB@ttnc.co.uk>
References: <6FCE44FF-70C1-412D-B722-4C8D359381BA@ttnc.co.uk>	<0F7476C7-FA7E-441F-9E17-EB2D23CD6055@jerris.com>	<EBF5F462-9A4F-4699-95FB-6947A9D2D55C@ttnc.co.uk>	<7b197bef1002190146x53492b85va88e4025c59e37f3@mail.gmail.com>	<496A4C3D-D185-4A86-BCA7-E4F5C05DA929@ttnc.co.uk>	<8D506E69-CB30-40BD-8AB9-9524C18E4503@ttnc.co.uk>	<7b197bef1002190234m58c7e8cbyd945af9ee0f5a3a@mail.gmail.com>	<D04CFC8A-4909-4E32-A480-567E712AE63A@ttnc.co.uk>	<2547643E-D259-46CE-919D-51393A4151BC@jerris.com>
	<CAC92986-7EA7-446B-8A95-50BF2FC25FCB@ttnc.co.uk>
Message-ID: <4B7EB86B.9010002@gmail.com>

An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100219/397af4cf/attachment-0002.html 

From t.mahe at telemaque.fr  Fri Feb 19 08:19:37 2010
From: t.mahe at telemaque.fr (=?ISO-8859-1?Q?Tristan_Mah=E9?=)
Date: Fri, 19 Feb 2010 17:19:37 +0100
Subject: [Freeswitch-users] 1.0.4 vs. trunk vs. bugs
In-Reply-To: <191c3a031002190732i3ff1598eob746072b5a55219d@mail.gmail.com>
References: <5d3e0dc61002181741r57b91f86h4810de63c1a3d46d@mail.gmail.com>	<20100219015118.GA12983@jdc.jasonjgw.net>	<1266552605.7684.11.camel@local.freepabx.com>	<616F57B5-182A-49A9-9825-2E5B72F5DB05@freeswitch.org>	<7716C919-BCD1-44FA-9ED2-5268F4C42100@jerris.com>	<b39273531002190621l656f44a2s8b05bc4162aa5182@mail.gmail.com>
	<191c3a031002190732i3ff1598eob746072b5a55219d@mail.gmail.com>
Message-ID: <4B7EBA19.8000507@telemaque.fr>

Hi Anthony,

As I said to Mike offlist, You can count on me for some help, not full
time badly, as I only have very little free time ( yep you
did/freeswitch is a much too awesome piece of code ;) ),:
backports/automated tests/some resources like boxes, I already said on
IRC that you just have to ask...

I follow carefully every commit to the project ( viva fs-trunk ML ), so
helping on identifying bugfixes and backporting them ( or at least
keeping track of what should be backported ) is a task I can help on...

For my part, I must say that I LOVE the 'make current' way, when I need
stable, I just stick to a specific svn rev I've fully tested.
I understand the need for a stable release for some people in the other
hand...

My 2cents on this subject:
As long as you and the people involved don't focus on maintaining a
'stable' branch, which would obviously slow down future improvements,
and keeps working like you did for all these years, adding a 'test field
branch' for instant commits is great to ensure trunk don't get broken
the time you fix it. This is a great compromise, with few overhead of work.

Regarding the labelling versions, it might also be quite a nightmare:
When do you branch ? new features ? new behaviour in code ? That would
require strict labelling rules that might add you some overhead of work
again, slowing things down.
Hope a team will arise for those people needing labels, that would carry
this work, and let you follow your journey on trunk without bothering
with theses issues, I would definitively give a hand to help on this if
needed...


Regards,

Gled

Anthony Minessale a ?crit :
> Here's the deal.
> 
> This is a community project and its public but it only has a small group
> individuals who are "all-in" committed to the project.  I am the one who
> started the project and who had to spend a solid 2 years completely
> alone working towards my goals before others even showed interest. 
> 
> Now we are growing very fast, we have a lot of feedback and we listen to
> it regardless of our position on the subject and I will decide and make
> any policy that I choose and feel is the best interest of this project.
> 
> It's reasonable that someone who is using the software wants to have
> wonderful stable stepping stones to migrate towards the future on.  It's
> also customary that most software, even when you pay a premium price for
> it, does not meet those standards every time.  If you do find software
> that has these graceful releases, they probably have a lot of people
> getting paid to work diligently on it.  There are also many successful
> open source projects with shiny revision numbers and packaged up with a
> bow but that is because they have a dedicated team of people.
> 
> So, we don't have that long list of people.  We invited people to do it
> and we had nobody step up.  So, this is what i am planning to do:
> 
> We are going to move our development to a branch, work on it from there
> and push them down to trunk when we think its the best time.  This might
> not always end up perfect but this is what we are going to do.  The
> actual release versions are still just fancy road signs in a long
> journey towards perfection.  We are still on 1.0 for almost 2 years with
> 5 micro release that contain a 12 page change log each time.  We are as
> careful as we can be about releases and we have no time to try to
> back-port patches to 6 month old code with more than 2000 revisions in
> between them.
> 
> When we feel we are happy with 1.0 we will then branch to 1.1 and all
> this stuff everyone wants, *IF* we get enough volunteers by that time to
> dedicate their time to maintaining it.  If not we will make the best of
> what we have........
> 
> This is the final word on this subject, feel free to quote me.
> 
> 
> 
> 
> 
> On Fri, Feb 19, 2010 at 8:21 AM, Rupa Schomaker <rupa at rupa.com
> <mailto:rupa at rupa.com>> wrote:
> 
>     I would caution that maintaining a stable branch is going to be
>     quite challenging.  All commits against trunk will fall into 3
>     categories:
> 
>     1) clearly bug fix
>     2) clearly new feature
>     3) a mix of the two
> 
>     1 can probably be easily back ported and 2 would be not but what of
>     3?  We don't split patches/commits based on a clear split between 1
>     and 2 so it would be the job of the stable maintainer to figure it
>     out, split it up and then commit just the bug fix part.
> 
>     I would argue that the churn in the stable branch would be
>     sufficient to make it a moving target just like trunk, just slower
>     moving and one step removed from tony ensuring everything is up to
>     his standard.
> 
>     I would also argue that at some point this project will clearly go
>     from "balls to the wall" development like now (lots of bug fixes and
>     new features all the time) to something more sane as it matures.  At
>     some point going to stable/dev might make sense.
> 
>     Another thought.  Look at how the linux kernel is developed now.
>      There is linus's branch which is essentially unstable.  It is the
>     vendor's (distro) job to pick a line in the sand and keep that
>     kernel rev stable.  There is help from people that have stepped up
>     and maintain a "stable" kernel branch, but that has nothing to do
>     with the mainline development.
> 
>     I can appreciate the pain that some people have with dealing with
>     production systems where you want stability above all else.  In
>     reality you don't want stability, you don't want
>     surprising behavioral changes.  You want code that doesn't change
>     except in those areas that fix bugs in components that you use.  But
>     your component set and mine are different.  Once you are accepting
>     bug fixes for all components, the set that changes can churn quite a
>     bit.  
> 
>     Anyway, just some food for thought.
> 
>     I know that if I had to double commit (or at least consider
>     double committing)  every piece of code I'd get frustrated quickly.
> 
> 
>     On Fri, Feb 19, 2010 at 1:17 AM, Michael Jerris <mike at jerris.com
>     <mailto:mike at jerris.com>> wrote:
> 
>         This seems a good time to note that we are still looking for
>         volunteers to assist in maintaining a stable branch.  I can not
>         do this without additional volunteer resources.  We have asked
>         several times recently to fairly silent response.  If anyone is
>         interested in assisting with this effort, please contact  me
>         offlist and we can discuss further.
> 
>         Mike
> 
>         On Feb 18, 2010, at 11:16 PM, Brian West <brian at freeswitch.org
>         <mailto:brian at freeswitch.org>> wrote:
> 
>>         OK so I can sign you up for the stable team?  ;)  As per my
>>         previous email i'm 100% sure we would do a stable release if
>>         we had people tending to issues.  The only problem is you
>>         would have to be on IRC tending to issues because if tony sees
>>         someone asking about a problem he'll be diving in to fix it
>>         before they can say "I have this one".  This also means
>>         working in a similar manner we do already.  Our process is
>>         very chaotic at times but it has served us well so far.  
>>
>>         The goal is to leave Anthony alone so he can move forward and
>>         let the stable team manage the jira's and issues on the list
>>         related to stable.
>>
>>         /b
>>
>>         On Feb 18, 2010, at 10:10 PM, David Knell wrote:
>>
>>>>
>>>>         Lon Baker <
>>>>         <mailto:lon at kickasspixels.com>lon at kickasspixels.com
>>>>         <mailto:lon at kickasspixels.com>> wrote:
>>>>
>>>>>         The development branch is where feature requests and
>>>>>         non-critical bugs
>>>>>         reports would be filed for the next production release.
>>>>>
>>>>>         The current process leaves a gap between production ready and
>>>>>         development code that may become greater over time.
>>>
>>>         Going against the grain here, I agree with you.  The current
>>>         way of
>>>         doing things is, in my opinion, not well thought through -
>>>         there's no
>>>         reason to tag and release versions if the answer to any issue
>>>         is 'make
>>>         current', and support is not available unless that's been
>>>         done.  Far
>>>         better to either have meaningful releases with stable and devel
>>>         branches, or not to have releases at all.
>>>
>>>         --Dave
>>
>>         _______________________________________________
>>         FreeSWITCH-users mailing list
>>         FreeSWITCH-users at lists.freeswitch.org
>>         <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>         http://www.freeswitch.org
> 
>         _______________________________________________
>         FreeSWITCH-users mailing list
>         FreeSWITCH-users at lists.freeswitch.org
>         <mailto:FreeSWITCH-users at lists.freeswitch.org>
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         http://www.freeswitch.org
> 
> 
> 
> 
>     -- 
>     -Rupa
> 
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
> 
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> <mailto:MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> <mailto:sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> <http://iax:guest at conference.freeswitch.org/888>
> googletalk:conf+888 at conference.freeswitch.org
> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From msc at freeswitch.org  Fri Feb 19 09:07:29 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 19 Feb 2010 09:07:29 -0800
Subject: [Freeswitch-users] FreeSWITCH Conference Call Starting Shortly!
Message-ID: <87f2f3b91002190907p19860ef7m461cc1957031d80e@mail.gmail.com>

Come on in!

http://wiki.freeswitch.org/wiki/FS_weekly_2010_02_19

-Michael
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100219/00657107/attachment-0002.html 

From anthony.minessale at gmail.com  Fri Feb 19 09:08:37 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 19 Feb 2010 11:08:37 -0600
Subject: [Freeswitch-users] Announcement: FreeSWITCH Add Broadsoft SCA
	Support
In-Reply-To: <d361ca7b1002182346k71fd602fh6e6725834a6de528@mail.gmail.com>
References: <87f2f3b91001111742k691966cfs82515db77c13d4d4@mail.gmail.com>
	<191c3a031002170556p5dfd4c55le3c279f194808d92@mail.gmail.com>
	<191c3a031002170557g5d8c9df9xf40d1b74b9a6f19d@mail.gmail.com>
	<d361ca7b1002170700x15630ab2tabc6f3d91ebc3303@mail.gmail.com>
	<ADC8B813-4161-481C-877E-93FC012858B4@freeswitch.org>
	<d361ca7b1002182101p60b4e8f8hf4dc95e1d2251e30@mail.gmail.com>
	<8b1c9cda1002182144m65ccf979w71384a1e7a55e2ea@mail.gmail.com>
	<d361ca7b1002182154o52352c62h8c0ec02fa66e8444@mail.gmail.com>
	<2F87B95C-EA23-4072-B845-3F7AE0617BE3@jerris.com>
	<d361ca7b1002182346k71fd602fh6e6725834a6de528@mail.gmail.com>
Message-ID: <191c3a031002190908r3922efccqc6d223a5fabd458b@mail.gmail.com>

go see my comments on that bug note.
be prepared to give us ssh access and call or irc so we can can see you
reproducing it.

If you are not on the latest firmware on all the phones, we will not
continue with this process.



On Fri, Feb 19, 2010 at 1:46 AM, Yehavi Bourvine
<yehavi.bourvine at gmail.com>wrote:

> A jira issue has been created: *MODSOFIA-61*<http://jira.freeswitch.org/browse/MODSOFIA-61>
> .
>
>                   Thanks, __Yehavi:
>
>  2010/2/19 Michael Jerris <mike at jerris.com>
>
>>  If this issue is not already on jira could you please make sure it gets
>> added?
>>
>> Mike
>>
>>
>> On Feb 19, 2010, at 12:54 AM, Yehavi Bourvine <yehavi.bourvine at gmail.com>
>> wrote:
>>
>>     Hello Gabe,
>>
>>   As you can see - Brian is actively investigating it, so you can expect
>> for some fix soon...
>>
>>                        Regards, __Yehavi:
>>
>> 2010/2/19 Gabriel Kuri < <gkuri at ieee.org>gkuri at ieee.org>
>>
>>> >  When a call arrives, both ring; the one that did not answer gets only
>>> a
>>> > cancel mesage without any further notification that the extension is in
>>> use
>>> > by the other phone.
>>>
>>> These are the same exact symptoms I posted about earlier this week,
>>> with the Cisco SPA-5xx series phones. I still have yet to figure out
>>> why this is happening, if you find out what's going on, please post
>>> back the solution, I'd like to know the resolution.
>>>
>>> Thanks,
>>> Gabe
>>>
>>>
>>>
>>> On Thu, Feb 18, 2010 at 9:01 PM, Yehavi Bourvine
>>> < <yehavi.bourvine at gmail.com>yehavi.bourvine at gmail.com> wrote:
>>> > Thanks Brian. It now works better, but not fully (using 16659M).
>>> >
>>> > What happens is:
>>> >
>>> > When one of the Polycoms seize the line it is ok - the other phone gets
>>> > notification and the extension status is "in use".
>>> > When one of the Polycom phones initiates a call - all is ok:
>>> >
>>> > The other side sees that the extension is in use.
>>> > When it is put to hold all phones who share this extension see it and
>>> can
>>> > pick the call.
>>> >
>>>
>>> >
>>>  >                          Thanks! __Yehavi:
>>> >
>>> > 2010/2/17 Brian West < <brian at freeswitch.org>brian at freeswitch.org>
>>> >>
>>> >> Step 1. Enable manage-shared-appearance=true
>>> >>
>>> >> Step 2. Now in the phone's config Configure the phone as usually, set
>>> the
>>> >> line shared and DO NOT set the third party name.
>>> >>
>>> >> Step 3. Reboot
>>> >>
>>> >> It should work.
>>> >>
>>> >> I wish someone that has this working would write some wiki docs these
>>> >> threads about it not working are getting rather old when I know for a
>>> fact
>>> >> they work fine.
>>> >>
>>> >> The gateway info missing is a gateway you have configured getting a
>>> >> notify.  It has nothing to do with SCA.
>>> >>
>>> >> /b
>>> >>
>>> >> On Feb 17, 2010, at 9:00 AM, Yehavi Bourvine wrote:
>>> >>
>>> >> > .
>>> >>
>>> >>
>>> >> _______________________________________________
>>> >> FreeSWITCH-users mailing list
>>> >> <FreeSWITCH-users at lists.freeswitch.org>
>>> FreeSWITCH-users at lists.freeswitch.org
>>> >> <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> <http://www.freeswitch.org/>http://www.freeswitch.org
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > <FreeSWITCH-users at lists.freeswitch.org>
>>> FreeSWITCH-users at lists.freeswitch.org
>>> > <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > <http://www.freeswitch.org/>http://www.freeswitch.org
>>> >
>>> >
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>> FreeSWITCH-users at lists.freeswitch.org
>>> <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>  <http://www.freeswitch.org/>http://www.freeswitch.org
>>>
>>  _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100219/ef0dac4c/attachment-0002.html 

From errotan at gmail.com  Fri Feb 19 09:25:50 2010
From: errotan at gmail.com (=?iso-8859-1?q?Pusk=E1s_Zsolt?=)
Date: Fri, 19 Feb 2010 18:25:50 +0100
Subject: [Freeswitch-users] mod_fax undefined symbol
In-Reply-To: <7b197bef1002190308u375c16f0p9ed83d7e8e19cf89@mail.gmail.com>
References: <6FCE44FF-70C1-412D-B722-4C8D359381BA@ttnc.co.uk>
	<B05DBE65-CF57-4E13-9CE2-E490128F1C41@ttnc.co.uk>
	<7b197bef1002190308u375c16f0p9ed83d7e8e19cf89@mail.gmail.com>
Message-ID: <201002191825.50957.errotan@gmail.com>

2010. febru?r 19. 12.08.25 Giovanni Maruzzelli d?tummal ezt ?rta:
> On Fri, Feb 19, 2010 at 11:58 AM, TTNC - Technical <technical at ttnc.co.uk> 
wrote:
> > Hmm... that's a bit odd. You would have thought that if FreeSWITCH is
> > compiled statically against it's own libtiff - then anything
> > Debian-centric shouldn't affect it?
> 
> yes, definitely
> 
> probably in your debian build libtiff is not compiled, or mod_fax is
> not linked (statically) with it
> 
> "normal" build, as in non-debian, works fine
> -gm
> 
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 

Just compiled svn16700 on Debian "lenny" x86, mod_fax loads and works 
perfectly. I have an ongoing compile on another machine (amd64) if It don't 
works i will send a mail (in 1 hour) otherwise consider it working.




From technical at ttnc.co.uk  Fri Feb 19 09:41:22 2010
From: technical at ttnc.co.uk (TTNC - Technical)
Date: Fri, 19 Feb 2010 17:41:22 +0000
Subject: [Freeswitch-users] mod_fax undefined symbol
In-Reply-To: <4B7EB86B.9010002@gmail.com>
References: <6FCE44FF-70C1-412D-B722-4C8D359381BA@ttnc.co.uk>	<0F7476C7-FA7E-441F-9E17-EB2D23CD6055@jerris.com>	<EBF5F462-9A4F-4699-95FB-6947A9D2D55C@ttnc.co.uk>	<7b197bef1002190146x53492b85va88e4025c59e37f3@mail.gmail.com>	<496A4C3D-D185-4A86-BCA7-E4F5C05DA929@ttnc.co.uk>	<8D506E69-CB30-40BD-8AB9-9524C18E4503@ttnc.co.uk>	<7b197bef1002190234m58c7e8cbyd945af9ee0f5a3a@mail.gmail.com>	<D04CFC8A-4909-4E32-A480-567E712AE63A@ttnc.co.uk>	<2547643E-D259-46CE-919D-51393A4151BC@jerris.com>
	<CAC92986-7EA7-446B-8A95-50BF2FC25FCB@ttnc.co.uk>
	<4B7EB86B.9010002@gmail.com>
Message-ID: <743BF178-CB48-41EC-A16B-31188ECE9496@ttnc.co.uk>


On 19 Feb 2010, at 16:12, Tamas wrote:

> Hi,
> 
> What does 'ldd mod_fax.so' say?

voipin2:/opt/freeswitch/mod# ldd mod_fax.so 
	linux-gate.so.1 =>  (0xb8015000)
	libm.so.6 => /lib/i686/cmov/libm.so.6 (0xb7f75000)
	libfreeswitch.so.1 => /opt/freeswitch/lib/libfreeswitch.so.1 (0xb7da1000)
	libpthread.so.0 => /lib/i686/cmov/libpthread.so.0 (0xb7d87000)
	libc.so.6 => /lib/i686/cmov/libc.so.6 (0xb7c40000)
	/lib/ld-linux.so.2 (0xb8016000)
	libssl.so.0.9.8 => /usr/lib/i686/cmov/libssl.so.0.9.8 (0xb7bf9000)
	libcrypto.so.0.9.8 => /usr/lib/i686/cmov/libcrypto.so.0.9.8 (0xb7aa2000)
	libncurses.so.5 => /lib/libncurses.so.5 (0xb7a6a000)
	libstdc++.so.6 => /usr/lib/libstdc++.so.6 (0xb7978000)
	libgcc_s.so.1 => /lib/libgcc_s.so.1 (0xb795b000)
	libodbc.so.1 => /usr/lib/libodbc.so.1 (0xb78f9000)
	libdl.so.2 => /lib/i686/cmov/libdl.so.2 (0xb78f5000)
	libz.so.1 => /usr/lib/libz.so.1 (0xb78e1000)
	libltdl.so.7 => /usr/lib/libltdl.so.7 (0xb78d8000)

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100219/000d8e27/attachment-0002.html 

From technical at ttnc.co.uk  Fri Feb 19 09:44:32 2010
From: technical at ttnc.co.uk (TTNC - Technical)
Date: Fri, 19 Feb 2010 17:44:32 +0000
Subject: [Freeswitch-users] mod_fax undefined symbol
In-Reply-To: <201002191825.50957.errotan@gmail.com>
References: <6FCE44FF-70C1-412D-B722-4C8D359381BA@ttnc.co.uk>
	<B05DBE65-CF57-4E13-9CE2-E490128F1C41@ttnc.co.uk>
	<7b197bef1002190308u375c16f0p9ed83d7e8e19cf89@mail.gmail.com>
	<201002191825.50957.errotan@gmail.com>
Message-ID: <AB3AD9C8-C162-4DB5-8095-97CA83EDDE57@ttnc.co.uk>


On 19 Feb 2010, at 17:25, Pusk?s Zsolt wrote:
> Just compiled svn16700 on Debian "lenny" x86, mod_fax loads and works 
> perfectly. I have an ongoing compile on another machine (amd64) if It don't 
> works i will send a mail (in 1 hour) otherwise consider it working.
> 

How did you compile it? Using dpkg-buildpackage or via make/make install?

Do you have any debian versions of libtiff4(-dev) installed?
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100219/49daefa6/attachment-0002.html 

From yehavi.bourvine at gmail.com  Fri Feb 19 09:50:42 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Fri, 19 Feb 2010 19:50:42 +0200
Subject: [Freeswitch-users] Announcement: FreeSWITCH Add Broadsoft SCA
	Support
In-Reply-To: <191c3a031002190908r3922efccqc6d223a5fabd458b@mail.gmail.com>
References: <87f2f3b91001111742k691966cfs82515db77c13d4d4@mail.gmail.com>
	<191c3a031002170557g5d8c9df9xf40d1b74b9a6f19d@mail.gmail.com>
	<d361ca7b1002170700x15630ab2tabc6f3d91ebc3303@mail.gmail.com>
	<ADC8B813-4161-481C-877E-93FC012858B4@freeswitch.org>
	<d361ca7b1002182101p60b4e8f8hf4dc95e1d2251e30@mail.gmail.com>
	<8b1c9cda1002182144m65ccf979w71384a1e7a55e2ea@mail.gmail.com>
	<d361ca7b1002182154o52352c62h8c0ec02fa66e8444@mail.gmail.com>
	<2F87B95C-EA23-4072-B845-3F7AE0617BE3@jerris.com>
	<d361ca7b1002182346k71fd602fh6e6725834a6de528@mail.gmail.com>
	<191c3a031002190908r3922efccqc6d223a5fabd458b@mail.gmail.com>
Message-ID: <d361ca7b1002190950t7b4da3ccy245c1667f82092e0@mail.gmail.com>

I see  now that Polycom released newer versions of firmware for the phones
recently. On Sunday's mornning I'll download them and retest with the latest
FreeSwitch snapshot.

                   __Yehavi:

 2010/2/19 Anthony Minessale <anthony.minessale at gmail.com>

> go see my comments on that bug note.
> be prepared to give us ssh access and call or irc so we can can see you
> reproducing it.
>
> If you are not on the latest firmware on all the phones, we will not
> continue with this process.
>
>
>
>
> On Fri, Feb 19, 2010 at 1:46 AM, Yehavi Bourvine <
> yehavi.bourvine at gmail.com> wrote:
>
>>  A jira issue has been created: *MODSOFIA-61*<http://jira.freeswitch.org/browse/MODSOFIA-61>
>> .
>>
>>                   Thanks, __Yehavi:
>>
>>  2010/2/19 Michael Jerris <mike at jerris.com>
>>
>>>  If this issue is not already on jira could you please make sure it gets
>>> added?
>>>
>>> Mike
>>>
>>>
>>> On Feb 19, 2010, at 12:54 AM, Yehavi Bourvine <yehavi.bourvine at gmail.com>
>>> wrote:
>>>
>>>     Hello Gabe,
>>>
>>>   As you can see - Brian is actively investigating it, so you can expect
>>> for some fix soon...
>>>
>>>                        Regards, __Yehavi:
>>>
>>> 2010/2/19 Gabriel Kuri < <gkuri at ieee.org>gkuri at ieee.org>
>>>
>>>> >  When a call arrives, both ring; the one that did not answer gets only
>>>> a
>>>> > cancel mesage without any further notification that the extension is
>>>> in use
>>>> > by the other phone.
>>>>
>>>> These are the same exact symptoms I posted about earlier this week,
>>>> with the Cisco SPA-5xx series phones. I still have yet to figure out
>>>> why this is happening, if you find out what's going on, please post
>>>> back the solution, I'd like to know the resolution.
>>>>
>>>> Thanks,
>>>> Gabe
>>>>
>>>>
>>>>
>>>> On Thu, Feb 18, 2010 at 9:01 PM, Yehavi Bourvine
>>>> < <yehavi.bourvine at gmail.com>yehavi.bourvine at gmail.com> wrote:
>>>> > Thanks Brian. It now works better, but not fully (using 16659M).
>>>> >
>>>> > What happens is:
>>>> >
>>>> > When one of the Polycoms seize the line it is ok - the other phone
>>>> gets
>>>> > notification and the extension status is "in use".
>>>> > When one of the Polycom phones initiates a call - all is ok:
>>>> >
>>>> > The other side sees that the extension is in use.
>>>> > When it is put to hold all phones who share this extension see it and
>>>> can
>>>> > pick the call.
>>>> >
>>>>
>>>> >
>>>>  >                          Thanks! __Yehavi:
>>>> >
>>>> > 2010/2/17 Brian West < <brian at freeswitch.org>brian at freeswitch.org>
>>>> >>
>>>> >> Step 1. Enable manage-shared-appearance=true
>>>> >>
>>>> >> Step 2. Now in the phone's config Configure the phone as usually, set
>>>> the
>>>> >> line shared and DO NOT set the third party name.
>>>> >>
>>>> >> Step 3. Reboot
>>>> >>
>>>> >> It should work.
>>>> >>
>>>> >> I wish someone that has this working would write some wiki docs these
>>>> >> threads about it not working are getting rather old when I know for a
>>>> fact
>>>> >> they work fine.
>>>> >>
>>>> >> The gateway info missing is a gateway you have configured getting a
>>>> >> notify.  It has nothing to do with SCA.
>>>> >>
>>>> >> /b
>>>> >>
>>>> >> On Feb 17, 2010, at 9:00 AM, Yehavi Bourvine wrote:
>>>> >>
>>>> >> > .
>>>> >>
>>>> >>
>>>> >> _______________________________________________
>>>> >> FreeSWITCH-users mailing list
>>>> >> <FreeSWITCH-users at lists.freeswitch.org>
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> >> <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> >> <http://www.freeswitch.org/>http://www.freeswitch.org
>>>> >
>>>> > _______________________________________________
>>>> > FreeSWITCH-users mailing list
>>>> > <FreeSWITCH-users at lists.freeswitch.org>
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> > <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> > UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> > <http://www.freeswitch.org/>http://www.freeswitch.org
>>>> >
>>>> >
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> <FreeSWITCH-users at lists.freeswitch.org>
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> <http://www.freeswitch.org/>http://www.freeswitch.org
>>>>
>>>  _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100219/bd79ca33/attachment-0002.html 

From errotan at gmail.com  Fri Feb 19 10:04:39 2010
From: errotan at gmail.com (=?utf-8?q?Pusk=C3=A1s_Zsolt?=)
Date: Fri, 19 Feb 2010 19:04:39 +0100
Subject: [Freeswitch-users] mod_fax undefined symbol
In-Reply-To: <AB3AD9C8-C162-4DB5-8095-97CA83EDDE57@ttnc.co.uk>
References: <6FCE44FF-70C1-412D-B722-4C8D359381BA@ttnc.co.uk>
	<201002191825.50957.errotan@gmail.com>
	<AB3AD9C8-C162-4DB5-8095-97CA83EDDE57@ttnc.co.uk>
Message-ID: <201002191904.39081.errotan@gmail.com>

2010. febru?r 19. 18.44.32 TTNC - Technical d?tummal ezt ?rta:
> On 19 Feb 2010, at 17:25, Pusk?s Zsolt wrote:
> > Just compiled svn16700 on Debian "lenny" x86, mod_fax loads and works
> > perfectly. I have an ongoing compile on another machine (amd64) if It
> > don't works i will send a mail (in 1 hour) otherwise consider it working.
> 
> How did you compile it? Using dpkg-buildpackage or via make/make install?
> 
> Do you have any debian versions of libtiff4(-dev) installed?
> 

Okay i can confirm that mod_fax works on Debian "lenny" amd64 but it don't work 
on Debian "testing,squeeze" amd64.

2010-02-19 18:49:14.610297 [CRIT] switch_loadable_module.c:882 Error Loading 
module /usr/local/freeswitch/mod/mod_fax.so
**/usr/local/freeswitch/mod/mod_fax.so: undefined symbol: 
TIFFDefaultStripSize**

I haven't tried to compile mod_fax on testing before so i don't know what is 
causeing the problem :(

# ldd mod_fax.so
        linux-vdso.so.1 =>  (0x00007fff106f6000)
        libm.so.6 => /lib/libm.so.6 (0x00007f506b711000)
        libfreeswitch.so.1 => /usr/local/freeswitch/lib/libfreeswitch.so.1 
(0x00007f506b345000)
        libpthread.so.0 => /lib/libpthread.so.0 (0x00007f506b128000)
        libc.so.6 => /lib/libc.so.6 (0x00007f506add4000)
        libssl.so.0.9.8 => /usr/lib/libssl.so.0.9.8 (0x00007f506ab82000)
        libcrypto.so.0.9.8 => /usr/lib/libcrypto.so.0.9.8 (0x00007f506a7e2000)
        libncurses.so.5 => /lib/libncurses.so.5 (0x00007f506a59d000)
        libstdc++.so.6 => /usr/lib/libstdc++.so.6 (0x00007f506a28d000)
        libgcc_s.so.1 => /lib/libgcc_s.so.1 (0x00007f506a076000)
        libodbc.so.1 => /usr/lib/libodbc.so.1 (0x00007f5069e17000)
        /lib64/ld-linux-x86-64.so.2 (0x00007f506bc31000)
        libdl.so.2 => /lib/libdl.so.2 (0x00007f5069c13000)
        libz.so.1 => /usr/lib/libz.so.1 (0x00007f50699fb000)
        libltdl.so.7 => /usr/lib/libltdl.so.7 (0x00007f50697f2000)

Recently in debian "testing" libtiff4 and libjpeg is upgraded:
libtiff 3.9.2-3+b1
libjpeg62 6b-16.1
libjeg8 8-2.1

Q&A:
Q: How did you compile it? Using dpkg-buildpackage or via make/make install?
A: svn-clean ./bootsrap ./configure make etc.

Q: Do you have any debian versions of libtiff4(-dev) installed?
A: Yes:3.8.2-11.2

I open a jira for this.



From scottferri09 at gmail.com  Fri Feb 19 12:01:34 2010
From: scottferri09 at gmail.com (Scott Fernandez)
Date: Sat, 20 Feb 2010 01:31:34 +0530
Subject: [Freeswitch-users] Establishing a Call from .Net based
	application
In-Reply-To: <922191.64755.qm@web33507.mail.mud.yahoo.com>
References: <6E8D2069C08AA84A83D336E996AE4C67032C9D5BAD@mse17be1.mse17.exchange.ms>
	<922191.64755.qm@web33507.mail.mud.yahoo.com>
Message-ID: <df97b9c51002191201n16183ebdk858fe50a0d479f0b@mail.gmail.com>

Thanks a lot Deigo and Michael.

Will work on this and let you know if I face any problems.

Thanks,
Scott

On Fri, Feb 19, 2010 at 5:04 AM, Diego Toro <dftoro at yahoo.com> wrote:

> The managed module is loaded as a module during the startup of FreeSWITCH
> if set in modules.conf.xml or through the command "load mod_managed" must
> keep in mind that there is a directory "mod/managed. As mod_managed is
> loaded into FreeSWITCH process to take control of the call must be running
> FreeSWITCH. So to "talk" with FreeSWITCH is not necessary to know the IP,
> the IP depends on the profile you've defined in the configuration of the
> module sofia. If you need the local address of the box running FreeSWITCH
> try expand variable $${local_ip_v4} which is assigned automatically by
> FreeSWITCH.
>
> Being more clear, when you use mod_managed including <action
> application="managed" data="yourclassname"/>
>  in a dialplan already have way to run your C# code.
>
> Now, if you need is to have control of the call to answer, originate, etc,
> without the application run inside FreeSWITCH process, you can use managed
> ESL (see examples in libs/esl/managed) this library allows your code using
> events "talk" with FreeSWITCH.
>
> Diego Toro
> http://lacarretade.blogspot.com/
>
>
> --- On Thu, 2/18/10, Michael Giagnocavo <mgg at giagnocavo.net> wrote:
>
> > From: Michael Giagnocavo <mgg at giagnocavo.net>
> > Subject: Re: [Freeswitch-users] Establishing a Call from .Net based
> application
> > To: "freeswitch-users at lists.freeswitch.org" <
> freeswitch-users at lists.freeswitch.org>
> > Date: Thursday, February 18, 2010, 6:12 AM
> > I?m not sure what the
> > FreeSWITCH APIs are to figure out what IP Sofia SIP has
> > bound to. Whatever it is, you?d call the same thing in
> > C#. What do you want to do with the API?  mod_managed.dll or .so is the
> > FreeSWITCH native code module that loads the CLR or Mono
> > into the FreeSWITCH process and loads
> > FreeSWITCH.Managed.dll. The managed DLL contains the bulk of
> > the managed-unmanaged interop code (.NET definitions of all
> > the FS C functions).   -Michael  From:
> > freeswitch-users-bounces at lists.freeswitch.org
> > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On
> > Behalf Of Scott Fernandez
> > Sent: Thursday, February 18, 2010 1:12 AM
> > To: freeswitch-users at lists.freeswitch.org
> > Subject: Re: [Freeswitch-users] Establishing a Call
> > from .Net based application
> >  Hi Diego & Michael,
> >
> > Thanks for your reply and support.
> >
> > However, I have some clarifications required from both of
> > you.
> >
> > 1. Here is the question for Diego,
> >
> > Simple Example:
> >
> > using FreeSWITCH;
> > using FreeSWITCH.Native;
> >
> > namespace BITS.Ivr.Bp.Server
> > {
> >  public class CIV_BPFSProcess : FreeSWITCH.IAppPlugin
> > {
> >  public void Run(AppContext context)
> >  {
> >   //answer call
> >   context.Session.Answer();
> >   //sleep 2 seconds
> >   context.Session.sleep(2000, 1);
> >   //hangup call
> >   context.Session.Hangup("NORMAL_CLEARING");
> >  }
> >  }
> > }
> > I understand that the concept of your example code.
> > However, would like to know as to how would my .NET C# know the
> > IP address of Freeswitch to talk to it as there is no
> > indication for that?. If not here, where would we need to
> > reference the IP address of FS in .NET code?
> >
> > I guess the IP address of FS needs to be mentioned in the
> > Target section of the below web.config file in .NET. If I am
> > right, how to specify the IP address over here. If I am
> > wrong, please let me know where do we need to mention the IP
> > address of FS.
> >
> >     <configuration>
> >
> > <dllmap dll="mod_managed.dll"
> > target="mod_managed.so"/>
> >     </configuration>
> >
> >
> > 2. Here is the question for Michael,
> >
> > You mentioned that "mod_managed.so will
> > be in your freeswitch mod directory". This is
> > very clear and what is mod_managed.dll in my .NET
> > application and the purpose of it?
> >
> > Thanks for all your help.
> >
> > Regards,
> > Scott.
> >
> >
> > On Sun, Feb 14, 2010 at 1:15
> > AM, Michael Giagnocavo <mgg at giagnocavo.net>
> > wrote:
> > 2. There is a configuration settings required to Map the
> > "DLL" to ".so" object in CentOS.
> > Now, the question is which DLL and .so file to be made
> > available and where??If you are
> > experiencing NullReferenceExceptions with any plugin run
> > through the dialplan, make sure you have included the
> > appropriate entry in your dllmap
> > configuration:  <dllmap dll="mod_managed.dll"
> > target="mod_managed.so"
> > os="!windows"/>?mod_managed.so will
> > be in your freeswitch mod directory.
> > All I need is to initiate a call from .NET application and
> > then it should talk to mod_managed module and establish a
> > call. Secondly, I need to know the status of the call such
> > as Ringing, Active, Hangup etc. To initiate a
> > call, try ManagedSession.Originate.-Michael
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> > -----Inline Attachment Follows-----
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100220/dcb92b1b/attachment-0002.html 

From ledoktre at meanie.us  Fri Feb 19 13:30:42 2010
From: ledoktre at meanie.us (Doc)
Date: Fri, 19 Feb 2010 15:30:42 -0600
Subject: [Freeswitch-users] Skypiax- how to know which card to use?
Message-ID: <4B7F0302.3060303@meanie.us>

Hey guys,

I am attempting to setup Skypiax and Freeswitch on Ubuntu Hardy.  I've 
had a couple of previous problems which updating to SVN trunk release 
seemed to resolve, but I've got one lingering one to ask about.

When a call comes in to my SkypeIn number (PSTN that rings my Skype), FS 
picks up and rings the desired extension.  No problems there.  What 
happens though is, the person that is on the extension in the office can 
hear the calling party just fine, but the calling party hears stuttered 
choppy sound.

I've already tested my internet connection, and through QOS have 
assigned my test box 512K up and 512K down, so on one test call 
bandwidth should not be an issue.

What do you think causes the stuttering on my audio for the calling party?

Do you think it might be a codec?  I have everything setup right now to 
use g711u.  I'd like to use something smaller and higher quality, but 
I'm not sure what is a better option.

I had made some phone calls as listed above and they tested out just 
fine, but in that same time frame I also recompiled Alsa drivers with 
the Skypiax dummy file, so do you think there might be something in that 
causing the stutter?

Looking for some advice.

Thanks,

Doc



From brian at microcomaustralia.com.au  Fri Feb 19 14:38:36 2010
From: brian at microcomaustralia.com.au (Brian May)
Date: Sat, 20 Feb 2010 09:38:36 +1100
Subject: [Freeswitch-users] building for Lenny
In-Reply-To: <4C13BCBF-1DA0-47AE-A749-738AD6D8FE48@jerris.com>
References: <3c5cf5261002180046vb0015e7t787599c9903aedc@mail.gmail.com>
	<33c87fa31002180056g51ef8496qa9a35746d0fe75ec@mail.gmail.com>
	<3c5cf5261002180110he0ae815uc713f77f289ac625@mail.gmail.com>
	<a9f4977a70b3fcf1b30413ea562ca962.squirrel@www.webcontracts.co.uk>
	<20100218161741.GC4236@base.carmickle.com>
	<3c5cf5261002181542y43adb7a7ofe2a15c38baf0327@mail.gmail.com>
	<4C13BCBF-1DA0-47AE-A749-738AD6D8FE48@jerris.com>
Message-ID: <3c5cf5261002191438o47aabde9tb588fe3ef23f27bc@mail.gmail.com>

On 19 February 2010 18:10, Michael Jerris <mike at jerris.com> wrote:
> Please create me a bug on http://jira.freeswitch.org for this issue.

Ok, hope I did that right. Now done.

Thanks
-- 
Brian May <brian at microcomaustralia.com.au>



From gmaruzz at celliax.org  Fri Feb 19 14:41:41 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Fri, 19 Feb 2010 23:41:41 +0100
Subject: [Freeswitch-users] Skypiax- how to know which card to use?
In-Reply-To: <4B7F0302.3060303@meanie.us>
References: <4B7F0302.3060303@meanie.us>
Message-ID: <7b197bef1002191441u455e8d8an4eb03c460254c6d3@mail.gmail.com>

On Fri, Feb 19, 2010 at 10:30 PM, Doc <ledoktre at meanie.us> wrote:
>
> I am attempting to setup Skypiax and Freeswitch on Ubuntu Hardy. ?I've
> had a couple of previous problems which updating to SVN trunk release
> seemed to resolve, but I've got one lingering one to ask about.

Hi Doc,

before to delve in the troubleshooting, I have to say that I'm
modifying the audio skypiax code in svn, so maybe it's just my fault
;).

please be patient for a little while, I hope to have done with it in a
couple days.

I'll announce to the mailing list when done.

In the mean time, at least one good news for you user of SkypeIn
service: a new feature of mod_skypiax is intended to recognize the
DTMFs coming from SkypeIn, so the incoming calls will be able to use
ivr, voicemail, etc.

>
> When a call comes in to my SkypeIn number (PSTN that rings my Skype), FS
> picks up and rings the desired extension. ?No problems there. ?What
> happens though is, the person that is on the extension in the office can
> hear the calling party just fine, but the calling party hears stuttered
> choppy sound.
> What do you think causes the stuttering on my audio for the calling party?
>
> Do you think it might be a codec? ?I have everything setup right now to
> use g711u. ?I'd like to use something smaller and higher quality, but
> I'm not sure what is a better option.

I would exclude is a codec problem

>
> I had made some phone calls as listed above and they tested out just
> fine, but in that same time frame I also recompiled Alsa drivers with
> the Skypiax dummy file, so do you think there might be something in that
> causing the stutter?

more probably is some change I made to the code, please be patient for
a while, couple days.

-giovanni


-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618



From ledoktre at meanie.us  Fri Feb 19 14:47:41 2010
From: ledoktre at meanie.us (Doc)
Date: Fri, 19 Feb 2010 16:47:41 -0600
Subject: [Freeswitch-users] Skypiax- how to know which card to use?
In-Reply-To: <7b197bef1002191441u455e8d8an4eb03c460254c6d3@mail.gmail.com>
References: <4B7F0302.3060303@meanie.us>
	<7b197bef1002191441u455e8d8an4eb03c460254c6d3@mail.gmail.com>
Message-ID: <4B7F150D.2040204@meanie.us>

Giovanni,

Awesome to hear from the creator of Skypiax.  You might be right, maybe 
it is the code in SVN.  I had recently updated SVN (at the time that I 
re-compiled with the skypiax dummy file).  I will look forward to an 
update on this.  I am also excited to hear that you have some new things 
coming that will allow for DTMF and IVR's, etc.  Yay!

Thanks for your reply,

Doc

> Hi Doc,
>
> before to delve in the troubleshooting, I have to say that I'm
> modifying the audio skypiax code in svn, so maybe it's just my fault
> ;).
>
> please be patient for a little while, I hope to have done with it in a
> couple days.
>
> I'll announce to the mailing list when done.
>
> In the mean time, at least one good news for you user of SkypeIn
> service: a new feature of mod_skypiax is intended to recognize the
> DTMFs coming from SkypeIn, so the incoming calls will be able to use
> ivr, voicemail, etc.
>
>   
>> When a call comes in to my SkypeIn number (PSTN that rings my Skype), FS
>> picks up and rings the desired extension.  No problems there.  What
>> happens though is, the person that is on the extension in the office can
>> hear the calling party just fine, but the calling party hears stuttered
>> choppy sound.
>> What do you think causes the stuttering on my audio for the calling party?
>>
>> Do you think it might be a codec?  I have everything setup right now to
>> use g711u.  I'd like to use something smaller and higher quality, but
>> I'm not sure what is a better option.
>>     
>
> I would exclude is a codec problem
>
>   
>> I had made some phone calls as listed above and they tested out just
>> fine, but in that same time frame I also recompiled Alsa drivers with
>> the Skypiax dummy file, so do you think there might be something in that
>> causing the stutter?
>>     
>
> more probably is some change I made to the code, please be patient for
> a while, couple days.
>
> -giovanni
>
>
>   



From spiritonly at gmail.com  Fri Feb 19 17:19:50 2010
From: spiritonly at gmail.com (=?UTF-8?B?5p2o5rGf6aqF?=)
Date: Sat, 20 Feb 2010 09:19:50 +0800
Subject: [Freeswitch-users] How to exchange media when I developed new
	endpoint module?
In-Reply-To: <ebbc2d551002091932u52dbf1b6jfecf2dfdb43b8a7c@mail.gmail.com>
References: <93b0f8ce1002091914q581f12fvd1de01e2181f0f3a@mail.gmail.com>
	<ebbc2d551002091932u52dbf1b6jfecf2dfdb43b8a7c@mail.gmail.com>
Message-ID: <93b0f8ce1002191719j71a7de5j374de41836ee2923@mail.gmail.com>

I have read mod_loopback and other endpoint modules, I found that it use
'switch_ivr_uuid_bridge' to bridge two session in mod_loopback, but in
others there are not 'switch_ivr_uuid_bridge' or bridge functions. So what
is different and which endpoint module should I consult?


2010/2/10 Jo?o Mesquita <jmesquita at freeswitch.org>

> You should look at read_frame and write_frame implementations of other
> endpoint modules.
>
> This should pretty much tell you how things work...
>
> Jo?o Mesquita
>
>
> On Wed, Feb 10, 2010 at 1:14 AM, ??? <spiritonly at gmail.com> wrote:
>
>> Hi,
>>     I am developping a new endpoint module, now I can make an inbound call
>> and execute IVR.
>> When I make an outbound call and bridge the inbound leg and outbound leg,
>> I receive remote alerting and pickup remote phone but there isn't
>> any voice exchange.
>>     So how to exchange media next?
>> ----------------------------------------------------------------------
>> gtalk:    spiritonly at gmail.com
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100220/c6050d87/attachment-0002.html 

From spiritonly at gmail.com  Fri Feb 19 17:22:34 2010
From: spiritonly at gmail.com (=?UTF-8?B?5p2o5rGf6aqF?=)
Date: Sat, 20 Feb 2010 09:22:34 +0800
Subject: [Freeswitch-users] How to exchange media when I developed new
	endpoint module?
In-Reply-To: <4021C303-1AC3-48F2-A17B-63C02C1C54D4@freeswitch.org>
References: <93b0f8ce1002091914q581f12fvd1de01e2181f0f3a@mail.gmail.com>
	<ebbc2d551002091932u52dbf1b6jfecf2dfdb43b8a7c@mail.gmail.com>
	<4021C303-1AC3-48F2-A17B-63C02C1C54D4@freeswitch.org>
Message-ID: <93b0f8ce1002191722h5782b787qe56a06dd0cb94e76@mail.gmail.com>

Do you know mod_khomp? You can found it in FS wiki. I am developing an
endpoint module like it.
So you can give me some advice to bridge two session?

On Wed, Feb 10, 2010 at 11:44 AM, Brian West <brian at freeswitch.org> wrote:

> But the bigger question is what protocol are you doing that you have to
> create your own endpoint module?
>
> /b
>
> On Feb 9, 2010, at 9:32 PM, Jo?o Mesquita wrote:
>
> > You should look at read_frame and write_frame implementations of other
> endpoint modules.
> >
> > This should pretty much tell you how things work...
> >
> > Jo?o Mesquita
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100220/e7583a54/attachment-0002.html 

From gamar at center.com  Fri Feb 19 06:07:33 2010
From: gamar at center.com (Gilbert Amar)
Date: Fri, 19 Feb 2010 15:07:33 +0100
Subject: [Freeswitch-users] Troubles bridging calls with mod_opal
Message-ID: <F9E933AA84A243688042045839DDEDFE@gamar>

Hello,

I am having troubles to make FS/opal works. Please do not ask me to use SIP 
instead of H323 it is out of scope.
I am trying here to bridge an incoming h323 call to another h323 party.
The result is that most of the time (sometime it actually works) the bridge 
is done but the callee cannot be heard by the caller.

If I dial directly the callee without FS/opal it works.
If I dial from a sip client like xlite then it works.

Here is our setup:
 FreeSWITCH Version 1.0.trunk (16659M) (at 2010-02-19 12:57 UTC)
 FS runs on a CentOs 5.2
 The calling party is a soft OpenPhone
 The called party is a Swissvoice IP10S


I try to enable/disable several codecs with no success.
Any idea where I should look ?

Please find attach the FS log and the Openphone log


Thanks
Gilbert 
-------------- next part --------------
A non-text attachment was scrubbed...
Name: ko.zip
Type: application/octet-stream
Size: 45635 bytes
Desc: not available
Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100219/5f0df6c2/attachment-0002.obj 

From gamar at center.com  Fri Feb 19 09:53:32 2010
From: gamar at center.com (Gilbert Amar)
Date: Fri, 19 Feb 2010 18:53:32 +0100
Subject: [Freeswitch-users] Troubles bridging calls with mod_opal
Message-ID: <91C2F3BC25F34E7AB6DA0441D390EA96@gamar>

Hello,

I am having troubles to make FS/opal works. Please do not ask me to use SIP 
instead of H323 it is out of scope.
I am trying here to bridge an incoming h323 call to another h323 party.
The result is that most of the time (sometime it actually works) the bridge 
is done but the callee cannot be heard by the caller.

If I dial directly the callee without FS/opal it works.
If I dial from a sip client like xlite then it works.

Here is our setup:
 FreeSWITCH Version 1.0.trunk (16659M) (at 2010-02-19 12:57 UTC)
 FS runs on a CentOs 5.2
 The calling party is a soft OpenPhone
 The called party is a Swissvoice IP10S


I try to enable/disable several codecs with no success.
Any idea where I should look ?

Please find attach the FS log and the Openphone log


Thanks
Gilbert 
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100219/3842edb1/attachment-0002.html 
-------------- next part --------------
A non-text attachment was scrubbed...
Name: ko.zip
Type: application/octet-stream
Size: 45635 bytes
Desc: not available
Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100219/3842edb1/attachment-0002.obj 

From mike at jerris.com  Fri Feb 19 18:00:31 2010
From: mike at jerris.com (Michael Jerris)
Date: Fri, 19 Feb 2010 21:00:31 -0500
Subject: [Freeswitch-users] mod_fax undefined symbol
In-Reply-To: <201002191904.39081.errotan@gmail.com>
References: <6FCE44FF-70C1-412D-B722-4C8D359381BA@ttnc.co.uk>
	<201002191825.50957.errotan@gmail.com>
	<AB3AD9C8-C162-4DB5-8095-97CA83EDDE57@ttnc.co.uk>
	<201002191904.39081.errotan@gmail.com>
Message-ID: <E6568E57-A3E2-4650-A8CD-70B9C2846AF4@jerris.com>

replying with more details on jira.


On Feb 19, 2010, at 1:04 PM, Pusk?s Zsolt wrote:

> 2010. febru?r 19. 18.44.32 TTNC - Technical d?tummal ezt ?rta:
>> On 19 Feb 2010, at 17:25, Pusk?s Zsolt wrote:
>>> Just compiled svn16700 on Debian "lenny" x86, mod_fax loads and works
>>> perfectly. I have an ongoing compile on another machine (amd64) if It
>>> don't works i will send a mail (in 1 hour) otherwise consider it working.
>> 
>> How did you compile it? Using dpkg-buildpackage or via make/make install?
>> 
>> Do you have any debian versions of libtiff4(-dev) installed?
>> 
> 
> Okay i can confirm that mod_fax works on Debian "lenny" amd64 but it don't work 
> on Debian "testing,squeeze" amd64.
> 
> 2010-02-19 18:49:14.610297 [CRIT] switch_loadable_module.c:882 Error Loading 
> module /usr/local/freeswitch/mod/mod_fax.so
> **/usr/local/freeswitch/mod/mod_fax.so: undefined symbol: 
> TIFFDefaultStripSize**
> 
> I haven't tried to compile mod_fax on testing before so i don't know what is 
> causeing the problem :(
> 
> # ldd mod_fax.so
>        linux-vdso.so.1 =>  (0x00007fff106f6000)
>        libm.so.6 => /lib/libm.so.6 (0x00007f506b711000)
>        libfreeswitch.so.1 => /usr/local/freeswitch/lib/libfreeswitch.so.1 
> (0x00007f506b345000)
>        libpthread.so.0 => /lib/libpthread.so.0 (0x00007f506b128000)
>        libc.so.6 => /lib/libc.so.6 (0x00007f506add4000)
>        libssl.so.0.9.8 => /usr/lib/libssl.so.0.9.8 (0x00007f506ab82000)
>        libcrypto.so.0.9.8 => /usr/lib/libcrypto.so.0.9.8 (0x00007f506a7e2000)
>        libncurses.so.5 => /lib/libncurses.so.5 (0x00007f506a59d000)
>        libstdc++.so.6 => /usr/lib/libstdc++.so.6 (0x00007f506a28d000)
>        libgcc_s.so.1 => /lib/libgcc_s.so.1 (0x00007f506a076000)
>        libodbc.so.1 => /usr/lib/libodbc.so.1 (0x00007f5069e17000)
>        /lib64/ld-linux-x86-64.so.2 (0x00007f506bc31000)
>        libdl.so.2 => /lib/libdl.so.2 (0x00007f5069c13000)
>        libz.so.1 => /usr/lib/libz.so.1 (0x00007f50699fb000)
>        libltdl.so.7 => /usr/lib/libltdl.so.7 (0x00007f50697f2000)
> 
> Recently in debian "testing" libtiff4 and libjpeg is upgraded:
> libtiff 3.9.2-3+b1
> libjpeg62 6b-16.1
> libjeg8 8-2.1
> 
> Q&A:
> Q: How did you compile it? Using dpkg-buildpackage or via make/make install?
> A: svn-clean ./bootsrap ./configure make etc.
> 
> Q: Do you have any debian versions of libtiff4(-dev) installed?
> A: Yes:3.8.2-11.2




From infos at madovsky.org  Fri Feb 19 21:08:25 2010
From: infos at madovsky.org (Madovsky)
Date: Sat, 20 Feb 2010 00:08:25 -0500
Subject: [Freeswitch-users] codecs transcoding
Message-ID: <DA78008AFA4A414480CDB572EE602D7D@MOBILEE1705>

Hello,

is it need proxy_media on true to transcode codecs ?

Thanks

Farnck
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100220/4ba33482/attachment-0002.html 

From jason at jasonjgw.net  Fri Feb 19 21:42:43 2010
From: jason at jasonjgw.net (Jason White)
Date: Sat, 20 Feb 2010 16:42:43 +1100
Subject: [Freeswitch-users] Troubles bridging calls with mod_opal
In-Reply-To: <91C2F3BC25F34E7AB6DA0441D390EA96@gamar>
References: <91C2F3BC25F34E7AB6DA0441D390EA96@gamar>
Message-ID: <20100220054243.GA4160@jdc.jasonjgw.net>

Gilbert Amar <gamar at center.com> wrote:
> I am having troubles to make FS/opal works. 

My impression is that more work is being devoted to mod_h323 than to mod_opal
to provide H323 support.

Try mod_h323 and see if it solves your problem.
If not, there are H323 users on the list who might be able to help. (I don't
use H323 personally.)




From pmhshz at gmail.com  Fri Feb 19 21:50:13 2010
From: pmhshz at gmail.com (MohammedShehzad)
Date: Sat, 20 Feb 2010 11:20:13 +0530
Subject: [Freeswitch-users] stream a file multicast with mod_esf
In-Reply-To: <a792304b1002190002u50ec6912g1905ffc63a119174@mail.gmail.com>
References: <B98B6921358D0C49BD21CD023F1F52C302623B4B@hbghs04a.ww300.siemens.net>
	<7C996136-963A-42D2-B0FE-D2729F968E52@freeswitch.org>
	<27071973.post@talk.nabble.com>
	<378CDEC3-5A41-4EF9-927A-311F2531E6AB@freeswitch.org>
	<a792304b1001080725y19ff28camd37425927550f311@mail.gmail.com>
	<191c3a031001081848w9036207yf6a4b473ec233243@mail.gmail.com>
	<85C74329-0D7C-48B2-B7A8-D4AA67C6E3F2@freeswitch.org>
	<a792304b1002182209m43ada13cp8ed3d91ad509f0a2@mail.gmail.com>
	<DBA4E291-9377-489A-841A-1A3D83880DF2@jerris.com>
	<a792304b1002190002u50ec6912g1905ffc63a119174@mail.gmail.com>
Message-ID: <a792304b1002192150n44cc500fnd5b2ecb7cff9dd1a@mail.gmail.com>

On Fri, Feb 19, 2010 at 1:32 PM, MohammedShehzad <pmhshz at gmail.com> wrote:

>
>
> On Fri, Feb 19, 2010 at 12:59 PM, Michael Jerris <mike at jerris.com> wrote:
>
>> Listening on multicast is noting special for multicast, it is just
>> like reading any other udp socket
>>
>> Mike
>>
>> Correct, but I have to play those audio stream back to caller taking care
> of the audio codec  and other things, do anybody have any idea in that part?
> Please let me know that.
> --
>
> -MohammedShehzad
>

I am able to receive the play the multicasted RAW PCMU RTP (modified the
skel of format provided by brian), so that caller can hear the multicast
which done by other Freeswitch server using mod_esf application, but when i
change the caller's codec from PCMU to something else, it breaks.

-- 

-MohammedShehzad
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100220/4cd80cf6/attachment-0002.html 

From brian at microcomaustralia.com.au  Fri Feb 19 21:57:21 2010
From: brian at microcomaustralia.com.au (Brian May)
Date: Sat, 20 Feb 2010 16:57:21 +1100
Subject: [Freeswitch-users] openzap TDM400 card
Message-ID: <3c5cf5261002192157t25de5798xc10ab2a7755e67d0@mail.gmail.com>

Hello,

I think I have the config correct, and not confused FXO/FXS anywhere.

voyage:/opt/freeswitch/conf# ztcfg  -vv

Zaptel Version: 1.4.11
Echo Canceller: MG2
Configuration
======================


Channel map:

Channel 01: FXO Kewlstart (Default) (Slaves: 01)
Channel 02: FXO Kewlstart (Default) (Slaves: 02)
Channel 03: FXS Kewlstart (Default) (Slaves: 03)
Channel 04: FXS Kewlstart (Default) (Slaves: 04)

4 channels to configure.

voyage:/opt/freeswitch/conf# cat openzap.conf
[span zt FXO1]
name => OpenZAP-FXO1
number => 1
fxo-channel => 1

[span zt FXO2]
name => OpenZAP-FXO2
number => 2
fxo-channel => 2

[span zt FXS1]
name => OpenZAP-FXS1
number => 3
fxs-channel => 3

[span zt FXS2]
name => OpenZAP-FXS2
number => 4
fxs-channel => 4

voyage:/opt/freeswitch/conf# cat autoload_configs/openzap.conf.xml
<configuration name="openzap.conf" description="OpenZAP Configuration">
  <settings>
    <param name="debug" value="0"/>
    <!--<param name="hold-music" value="$${moh_uri}"/>-->
    <!--<param name="enable-analog-option" value="call-swap"/>-->
    <!--<param name="enable-analog-option" value="3-way"/>-->
  </settings>
  <!-- one entry here per openzap span -->
  <analog_spans>
    <span id="1">
      <param name="tonegroup" value="au"/>
      <param name="digit-timeout" value="2000"/>
      <param name="max-digits" value="11"/>
      <param name="dialplan" value="XML"/>
      <param name="context" value="default"/>
      <param name="enable-analog-option" value="3-way"/>
<!--      <param name="moh" value="$${base_dir}/sounds/Dardedisco.wav"/> -->
    </span>
    <span id="2">
      <param name="tonegroup" value="au"/>
      <param name="digit-timeout" value="2000"/>
      <param name="max-digits" value="11"/>
      <param name="dialplan" value="XML"/>
      <param name="context" value="default"/>
      <param name="enable-analog-option" value="3-way"/>
<!--      <param name="moh" value="$${base_dir}/sounds/Dardedisco.wav"/> -->
    </span>
    <span id="3">
      <param name="tonegroup" value="au"/>
      <param name="digit-timeout" value="2000"/>
      <param name="max-digits" value="11"/>
      <param name="dialplan" value="XML"/>
      <param name="context" value="default"/>
    </span>
    <span id="4">
      <param name="tonegroup" value="au"/>
      <param name="digit-timeout" value="2000"/>
      <param name="max-digits" value="11"/>
      <param name="dialplan" value="XML"/>
      <param name="context" value="default"/>
    </span>
  </analog_spans>
</configuration>


freeswitch at voyage> reload mod_openzap
2010-02-20 16:45:53.312813 [NOTICE] switch_loadable_module.c:464
Deleting Endpoint 'openzap'
2010-02-20 16:45:53.312813 [NOTICE] switch_loadable_module.c:545
Deleting Application 'disable_ec'
2010-02-20 16:45:53.312813 [NOTICE] switch_loadable_module.c:572
Deleting API Function 'oz'
2010-02-20 16:45:53.312813 [CONSOLE] switch_loadable_module.c:1277
Stopping: mod_openzap
2010-02-20 16:45:53.312813 [INFO] zap_io.c:269 Closing channel zt:1:1 fd:36
2010-02-20 16:45:53.312813 [INFO] zap_io.c:269 Closing channel zt:2:1 fd:40
2010-02-20 16:45:53.312813 [INFO] zap_io.c:269 Closing channel zt:3:1 fd:41
2010-02-20 16:45:53.312813 [INFO] zap_io.c:269 Closing channel zt:4:1 fd:42
2010-02-20 16:45:53.612813 [ERR] ozmod_analog.c:951 Failure Polling
event! [no matching descriptor]
2010-02-20 16:45:54.162813 [ERR] ozmod_analog.c:951 Failure Polling
event! [no matching descriptor]
2010-02-20 16:45:54.162813 [ERR] ozmod_analog.c:951 Failure Polling
event! [no matching descriptor]
2010-02-20 16:45:54.162813 [ERR] ozmod_analog.c:951 Failure Polling
event! [no matching descriptor]
2010-02-20 16:45:54.322813 [INFO] zap_io.c:2694 Unloading
/opt/freeswitch/mod/ozmod_analog.so
2010-02-20 16:45:54.322813 [INFO] zap_io.c:2679 Unloading IO zt
2010-02-20 16:45:54.322813 [INFO] zap_io.c:2694 Unloading
/opt/freeswitch/mod/ozmod_zt.so
2010-02-20 16:45:54.322813 [CONSOLE] switch_loadable_module.c:1297
mod_openzap unloaded.
2010-02-20 16:45:54.322813 [NOTICE] zap_io.c:2778 Modules configured: 1
2010-02-20 16:45:54.322813 [NOTICE] ozmod_zt.c:1161 Using Zaptel control device
2010-02-20 16:45:54.322813 [INFO] zap_io.c:2579 Loading IO from
/opt/freeswitch/mod/ozmod_zt.so [zt]
2010-02-20 16:45:54.322813 [INFO] ozmod_zt.c:556 Setting rxgain val to 0.000000
2010-02-20 16:45:54.322813 [INFO] ozmod_zt.c:565 Setting txgain val to 0.000000
2010-02-20 16:45:54.322813 [INFO] zap_io.c:2379 auto-loaded 'zt'
2010-02-20 16:45:54.322813 [WARNING] ozmod_zt.c:333 this ioctl fails
on older zaptel but is harmless if you used ztcfg
[device /dev/zap/channel chan 1 fd 27 (Invalid argument)]
2010-02-20 16:45:54.322813 [INFO] ozmod_zt.c:385 configuring device
/dev/zap/channel channel 1 as OpenZAP device 1:1 fd:36
2010-02-20 16:45:54.322813 [WARNING] ozmod_zt.c:333 this ioctl fails
on older zaptel but is harmless if you used ztcfg
[device /dev/zap/channel chan 2 fd 27 (Invalid argument)]
2010-02-20 16:45:54.322813 [INFO] ozmod_zt.c:385 configuring device
/dev/zap/channel channel 2 as OpenZAP device 2:1 fd:40
2010-02-20 16:45:54.322813 [WARNING] ozmod_zt.c:333 this ioctl fails
on older zaptel but is harmless if you used ztcfg
[device /dev/zap/channel chan 3 fd 27 (Invalid argument)]
2010-02-20 16:45:54.322813 [INFO] ozmod_zt.c:385 configuring device
/dev/zap/channel channel 3 as OpenZAP device 3:1 fd:41
2010-02-20 16:45:54.322813 [WARNING] ozmod_zt.c:333 this ioctl fails
on older zaptel but is harmless if you used ztcfg
[device /dev/zap/channel chan 4 fd 27 (Invalid argument)]
2010-02-20 16:45:54.322813 [INFO] ozmod_zt.c:385 configuring device
/dev/zap/channel channel 4 as OpenZAP device 4:1 fd:42
2010-02-20 16:45:54.322813 [INFO] zap_io.c:2502 Configured 4 channel(s)
2010-02-20 16:45:54.322813 [INFO] zap_io.c:2596 Loading SIG from
/opt/freeswitch/mod/ozmod_analog.so
2010-02-20 16:45:54.322813 [INFO] zap_io.c:2712 auto-loaded 'analog'
2010-02-20 16:45:54.358813 [CONSOLE] switch_loadable_module.c:900
Successfully Loaded [mod_openzap]
2010-02-20 16:45:54.362813 [NOTICE] switch_loadable_module.c:144
Adding Endpoint 'openzap'
2010-02-20 16:45:54.362813 [NOTICE] switch_loadable_module.c:250
Adding Application 'disable_ec'
2010-02-20 16:45:54.362813 [NOTICE] switch_loadable_module.c:272
Adding API Function 'oz'

+OK module unloaded
+OK module loaded

freeswitch at voyage> oz list

+OK
span: 1 (FXO1)
type: analog
chan_count: 1
dialplan: XML
context: default
dial_regex:
fail_dial_regex:
hold_music:
analog_options 3way
+OK
span: 2 (FXO2)
type: analog
chan_count: 1
dialplan: XML
context: default
dial_regex:
fail_dial_regex:
hold_music:
analog_options 3way
+OK
span: 3 (FXS1)
type: analog
chan_count: 1
dialplan: XML
context: default
dial_regex:
fail_dial_regex:
hold_music:
analog_options none
+OK
span: 4 (FXS2)
type: analog
chan_count: 1
dialplan: XML
context: default
dial_regex:
fail_dial_regex:
hold_music:
analog_options none


Yet, when I lift the handset on port 1 or port 2 I don't get a dial tone :-(

Instead I get this message:

2010-02-20 16:52:26.332813 [INFO] ozmod_zt.c:640 Setting echo cancel
to 64 taps for 2:1
2010-02-20 16:52:26.332813 [ERR] zap_io.c:1599 I/O backend does not
support command 24!

Am I doing something wrong?
-- 
Brian May <brian at microcomaustralia.com.au>



From brian at microcomaustralia.com.au  Sat Feb 20 01:48:36 2010
From: brian at microcomaustralia.com.au (Brian May)
Date: Sat, 20 Feb 2010 20:48:36 +1100
Subject: [Freeswitch-users] openzap TDM400 card
In-Reply-To: <3c5cf5261002192157t25de5798xc10ab2a7755e67d0@mail.gmail.com>
References: <3c5cf5261002192157t25de5798xc10ab2a7755e67d0@mail.gmail.com>
Message-ID: <3c5cf5261002200148k1bb3a1f7mf97fabccfbd1c178@mail.gmail.com>

On 20 February 2010 16:57, Brian May <brian at microcomaustralia.com.au> wrote:
> I think I have the config correct, and not confused FXO/FXS anywhere.

I was confused. The messages after the incoming call gave it away.

2010-02-20 20:42:52.339813 [ERR] ozmod_analog.c:798 Cannot get a
RING_START event on a non-fxo channel, please check your config.
2010-02-20 20:42:52.339813 [WARNING] ozmod_analog.c:799 Why bother
changing state on 3:1 from DOWN to DOWN
2010-02-20 20:42:53.189813 [INFO] ozmod_zt.c:640 Setting echo cancel
to 64 taps for 3:1
2010-02-20 20:42:55.189813 [ERR] ozmod_analog.c:798 Cannot get a
RING_START event on a non-fxo channel, please check your config.
2010-02-20 20:42:56.449813 [INFO] ozmod_zt.c:640 Setting echo cancel
to 64 taps for 3:1
2010-02-20 20:42:58.239813 [ERR] ozmod_analog.c:798 Cannot get a
RING_START event on a non-fxo channel, please check your config.
2010-02-20 20:42:59.499813 [INFO] ozmod_zt.c:640 Setting echo cancel
to 64 taps for 3:1
2010-02-20 20:43:01.299813 [ERR] ozmod_analog.c:798 Cannot get a
RING_START event on a non-fxo channel, please check your config.
2010-02-20 20:43:02.569813 [INFO] ozmod_zt.c:640 Setting echo cancel
to 64 taps for 3:1
2010-02-20 20:43:04.379813 [ERR] ozmod_analog.c:798 Cannot get a
RING_START event on a non-fxo channel, please check your config.
2010-02-20 20:43:05.659813 [INFO] ozmod_zt.c:640 Setting echo cancel
to 64 taps for 3:1

It appears that a port listed as fxoks in zaptel.conf becomes fxs in
openzap.conf, and a port listed as fxsks becomes fxo in openzap.conf -
it would be nice if this were documented somewhere...


Working configuration:


voyage:/opt/freeswitch/conf# cat /etc/zaptel.conf
# Autogenerated by /usr/sbin/genzaptelconf -- do not hand edit
# Zaptel Configuration File
#
# This file is parsed by the Zaptel Configurator, ztcfg
#

# It must be in the module loading order


# Span 1: WCTDM/0 "Wildcard TDM400P REV I Board 1" (MASTER)
fxoks=1
fxoks=2
fxsks=3
fxsks=4

# Global data

loadzone	= au
defaultzone	= au
voyage:/opt/freeswitch/conf# vim openzap.conf
voyage:/opt/freeswitch/conf# cat /etc/zaptel.conf
# Autogenerated by /usr/sbin/genzaptelconf -- do not hand edit
# Zaptel Configuration File
#
# This file is parsed by the Zaptel Configurator, ztcfg
#

# It must be in the module loading order


# Span 1: WCTDM/0 "Wildcard TDM400P REV I Board 1" (MASTER)
fxoks=1
fxoks=2
fxsks=3
fxsks=4

# Global data

loadzone	= au
defaultzone	= au
voyage:/opt/freeswitch/conf# cat openzap.conf
[span zt FXS1]
name => OpenZAP-FXS1
number => 1
fxs-channel => 1

[span zt FXS2]
name => OpenZAP-FXS2
number => 2
fxs-channel => 2

[span zt FXO1]
name => OpenZAP-FXO1
number => 3
fxo-channel => 3

[span zt FXO2]
name => OpenZAP-FXO2
number => 4
fxo-channel => 4


-- 
Brian May <brian at microcomaustralia.com.au>



From Russell.Mosemann at cune.org  Sat Feb 20 03:25:06 2010
From: Russell.Mosemann at cune.org (Russell Mosemann)
Date: Sat, 20 Feb 2010 05:25:06 -0600
Subject: [Freeswitch-users] openzap TDM400 card
In-Reply-To: <3c5cf5261002200148k1bb3a1f7mf97fabccfbd1c178@mail.gmail.com>
References: <3c5cf5261002192157t25de5798xc10ab2a7755e67d0@mail.gmail.com>
	<3c5cf5261002200148k1bb3a1f7mf97fabccfbd1c178@mail.gmail.com>
Message-ID: <C8B78F95DB344ADDA010C52AE0AC8B66@cune.pri>

> It appears that a port listed as fxoks in zaptel.conf becomes fxs in
> openzap.conf, and a port listed as fxsks becomes fxo in openzap.conf -
> it would be nice if this were documented somewhere...

I believe it is, with the specific error you mention, "Cannot get a RING_START event on a non-fxo channel".

http://wiki.freeswitch.org/wiki/OpenZAP#Symptom:_.22Why_bother_changing_state_on_1:1_from_UP_to_UP.22_or_.22Cannot_get_a_RING_START_event_on_a_non-fxo_channel.22

If you don't think it is clear, please add more details to the wiki so that others will be able to more easily solve this problem when it happens.

--
Russell Mosemann




From rob4manhere at gmail.com  Sat Feb 20 05:48:48 2010
From: rob4manhere at gmail.com (Rob Forman)
Date: Sat, 20 Feb 2010 07:48:48 -0600
Subject: [Freeswitch-users] codecs transcoding
In-Reply-To: <DA78008AFA4A414480CDB572EE602D7D@MOBILEE1705>
References: <DA78008AFA4A414480CDB572EE602D7D@MOBILEE1705>
Message-ID: <73A3511B-EC7F-465B-BA51-9A7067303AFF@gmail.com>

No, proxy media is the opposite.  Its for staying in the middle of the  
RTP stream, yet *not* transcoding or processing the packets.

http://wiki.freeswitch.org/wiki/Proxy_Media

If you want to transcode, and both are supported codecs, just bridge  
the two channels.

Rob

On Feb 19, 2010, at 11:08 PM, Madovsky wrote:

> Hello,
>
> is it need proxy_media on true to transcode codecs ?
>
> Thanks
>
> Farnck
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100220/0112f237/attachment-0002.html 

From infos at madovsky.org  Sat Feb 20 08:26:13 2010
From: infos at madovsky.org (Madovsky)
Date: Sat, 20 Feb 2010 11:26:13 -0500
Subject: [Freeswitch-users] codecs transcoding
References: <DA78008AFA4A414480CDB572EE602D7D@MOBILEE1705>
	<73A3511B-EC7F-465B-BA51-9A7067303AFF@gmail.com>
Message-ID: <789ED5715DFF4072A38363B654C5AD19@MOBILEE1705>


  ----- Original Message ----- 
  From: Rob Forman 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Saturday, February 20, 2010 8:48 AM
  Subject: Re: [Freeswitch-users] codecs transcoding


  No, proxy media is the opposite.  Its for staying in the middle of the RTP stream, yet *not* transcoding or processing the packets.


  http://wiki.freeswitch.org/wiki/Proxy_Media


  If you want to transcode, and both are supported codecs, just bridge the two channels.


  Rob


  On Feb 19, 2010, at 11:08 PM, Madovsky wrote:


    Hello,

    is it need proxy_media on true to transcode codecs ?

    Thanks

    Farnck
    _______________________________________________
    FreeSWITCH-users mailing list
    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    http://www.freeswitch.org





------------------------------------------------------------------------------


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  http://www.freeswitch.org

  Thanks Rob, I got it yesterday.

  Franck
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100220/9f5b8eaf/attachment-0002.html 

From mbsip at gazeta.pl  Sat Feb 20 09:48:44 2010
From: mbsip at gazeta.pl (Maciej Bylica)
Date: Sat, 20 Feb 2010 18:48:44 +0100
Subject: [Freeswitch-users] LUA script providing dynamic directory
	information
Message-ID: <28f27f5d1002200948j33d818e1h9fc8c8f99a506656@mail.gmail.com>

Hello,

I am trying to use mod_lua to provide dynamic directory information
(binding in mod_lua.conf.xml)
Here is my script.
#!/usr/local/bin/lua
-- load driver
require "luasql.odbc"
-- create environment object
env = luasql.odbc();
-- connect to data source
conn = env:connect("freeswitch","root");
-- reset our table
if ( conn ~= nil ) then
cur = conn:execute(string.format("SELECT email from VM where
called_num='48112223344'"));

if ( cur ~= nil ) then
row = cur:fetch({}, "a");
if ( row ~= nil ) then

freeswitch.consoleLog("info", " Email fetched from DB is = "..
row.email .."\n");

cur:close(); conn:close(); env:close();

mydialplan = [[
<?xml version="1.0" encoding="UTF-8" standalone="no"?>
<document type="freeswitch/xml">
  <section name="directory">
    <domain name="10.10.10.1">
      <user id="48112223344">
        <params>
          <param name="vm-email-all-messages" value="true"/>
          <param name="vm-attach-file" value="true"/>
          <param name="vm-disk-quota" value="0"/>
          <param name="vm-email-all-messages" value="true"/>
          <param name="vm-mailto" value=".. row.email .."/>
          <param name="vm-keep-local-after-email" value="false"/>
        </params>
      </user>
    </domain>
  </section>
</document>
]]

XML_STRING = mydialplan
end
end
end


I've encountered a problem how to pass row.email gathered from DB
directly to XML (vm-mailto).
As you can see below configuration I have does not work properly.

2010-02-20 20:37:03.267071 [DEBUG] mod_voicemail.c:2358 Deliver VM to
48112223344 at 10.10.10.1
2010-02-20 20:37:03.276533 [INFO] switch_cpp.cpp:1129  Email fetched
from DB is = hereis at MyEmaill.com
2010-02-20 20:37:03.435902 [DEBUG] switch_utils.c:631 Emailed file
[/tmp/mail.1266694623f261] to [.. row.email ..]
2010-02-20 20:37:03.435902 [DEBUG] mod_voicemail.c:2526 Sending
message to .. row.email ..

I tried with:
<param name="vm-mailto" value=.. row.email ../>
<param name="vm-mailto" value="".. row.email ..""/>
Both of them do not produce any "Sending message" output at all.


Any thoughts?
Thanks in advance.
Maciej



From msc at freeswitch.org  Sat Feb 20 10:10:21 2010
From: msc at freeswitch.org (Michael S Collins)
Date: Sat, 20 Feb 2010 10:10:21 -0800
Subject: [Freeswitch-users] openzap TDM400 card
In-Reply-To: <3c5cf5261002192157t25de5798xc10ab2a7755e67d0@mail.gmail.com>
References: <3c5cf5261002192157t25de5798xc10ab2a7755e67d0@mail.gmail.com>
Message-ID: <E94B1537-7943-4328-B090-AEF26C781673@freeswitch.org>

Ports 1 and 2 are FXO which need a phone line. Try port 3 or 4. Also  
pastebin your zaptel.conf file.

-MC

Sent from my iPhone

On Feb 19, 2010, at 9:57 PM, Brian May  
<brian at microcomaustralia.com.au> wrote:

> Hello,
>
> I think I have the config correct, and not confused FXO/FXS anywhere.
>
> voyage:/opt/freeswitch/conf# ztcfg  -vv
>
> Zaptel Version: 1.4.11
> Echo Canceller: MG2
> Configuration
> ======================
>
>
> Channel map:
>
> Channel 01: FXO Kewlstart (Default) (Slaves: 01)
> Channel 02: FXO Kewlstart (Default) (Slaves: 02)
> Channel 03: FXS Kewlstart (Default) (Slaves: 03)
> Channel 04: FXS Kewlstart (Default) (Slaves: 04)
>
> 4 channels to configure.
>
> voyage:/opt/freeswitch/conf# cat openzap.conf
> [span zt FXO1]
> name => OpenZAP-FXO1
> number => 1
> fxo-channel => 1
>
> [span zt FXO2]
> name => OpenZAP-FXO2
> number => 2
> fxo-channel => 2
>
> [span zt FXS1]
> name => OpenZAP-FXS1
> number => 3
> fxs-channel => 3
>
> [span zt FXS2]
> name => OpenZAP-FXS2
> number => 4
> fxs-channel => 4
>
> voyage:/opt/freeswitch/conf# cat autoload_configs/openzap.conf.xml
> <configuration name="openzap.conf" description="OpenZAP  
> Configuration">
>  <settings>
>    <param name="debug" value="0"/>
>    <!--<param name="hold-music" value="$${moh_uri}"/>-->
>    <!--<param name="enable-analog-option" value="call-swap"/>-->
>    <!--<param name="enable-analog-option" value="3-way"/>-->
>  </settings>
>  <!-- one entry here per openzap span -->
>  <analog_spans>
>    <span id="1">
>      <param name="tonegroup" value="au"/>
>      <param name="digit-timeout" value="2000"/>
>      <param name="max-digits" value="11"/>
>      <param name="dialplan" value="XML"/>
>      <param name="context" value="default"/>
>      <param name="enable-analog-option" value="3-way"/>
> <!--      <param name="moh" value="$${base_dir}/sounds/ 
> Dardedisco.wav"/> -->
>    </span>
>    <span id="2">
>      <param name="tonegroup" value="au"/>
>      <param name="digit-timeout" value="2000"/>
>      <param name="max-digits" value="11"/>
>      <param name="dialplan" value="XML"/>
>      <param name="context" value="default"/>
>      <param name="enable-analog-option" value="3-way"/>
> <!--      <param name="moh" value="$${base_dir}/sounds/ 
> Dardedisco.wav"/> -->
>    </span>
>    <span id="3">
>      <param name="tonegroup" value="au"/>
>      <param name="digit-timeout" value="2000"/>
>      <param name="max-digits" value="11"/>
>      <param name="dialplan" value="XML"/>
>      <param name="context" value="default"/>
>    </span>
>    <span id="4">
>      <param name="tonegroup" value="au"/>
>      <param name="digit-timeout" value="2000"/>
>      <param name="max-digits" value="11"/>
>      <param name="dialplan" value="XML"/>
>      <param name="context" value="default"/>
>    </span>
>  </analog_spans>
> </configuration>
>
>
> freeswitch at voyage> reload mod_openzap
> 2010-02-20 16:45:53.312813 [NOTICE] switch_loadable_module.c:464
> Deleting Endpoint 'openzap'
> 2010-02-20 16:45:53.312813 [NOTICE] switch_loadable_module.c:545
> Deleting Application 'disable_ec'
> 2010-02-20 16:45:53.312813 [NOTICE] switch_loadable_module.c:572
> Deleting API Function 'oz'
> 2010-02-20 16:45:53.312813 [CONSOLE] switch_loadable_module.c:1277
> Stopping: mod_openzap
> 2010-02-20 16:45:53.312813 [INFO] zap_io.c:269 Closing channel zt: 
> 1:1 fd:36
> 2010-02-20 16:45:53.312813 [INFO] zap_io.c:269 Closing channel zt: 
> 2:1 fd:40
> 2010-02-20 16:45:53.312813 [INFO] zap_io.c:269 Closing channel zt: 
> 3:1 fd:41
> 2010-02-20 16:45:53.312813 [INFO] zap_io.c:269 Closing channel zt: 
> 4:1 fd:42
> 2010-02-20 16:45:53.612813 [ERR] ozmod_analog.c:951 Failure Polling
> event! [no matching descriptor]
> 2010-02-20 16:45:54.162813 [ERR] ozmod_analog.c:951 Failure Polling
> event! [no matching descriptor]
> 2010-02-20 16:45:54.162813 [ERR] ozmod_analog.c:951 Failure Polling
> event! [no matching descriptor]
> 2010-02-20 16:45:54.162813 [ERR] ozmod_analog.c:951 Failure Polling
> event! [no matching descriptor]
> 2010-02-20 16:45:54.322813 [INFO] zap_io.c:2694 Unloading
> /opt/freeswitch/mod/ozmod_analog.so
> 2010-02-20 16:45:54.322813 [INFO] zap_io.c:2679 Unloading IO zt
> 2010-02-20 16:45:54.322813 [INFO] zap_io.c:2694 Unloading
> /opt/freeswitch/mod/ozmod_zt.so
> 2010-02-20 16:45:54.322813 [CONSOLE] switch_loadable_module.c:1297
> mod_openzap unloaded.
> 2010-02-20 16:45:54.322813 [NOTICE] zap_io.c:2778 Modules  
> configured: 1
> 2010-02-20 16:45:54.322813 [NOTICE] ozmod_zt.c:1161 Using Zaptel  
> control device
> 2010-02-20 16:45:54.322813 [INFO] zap_io.c:2579 Loading IO from
> /opt/freeswitch/mod/ozmod_zt.so [zt]
> 2010-02-20 16:45:54.322813 [INFO] ozmod_zt.c:556 Setting rxgain val  
> to 0.000000
> 2010-02-20 16:45:54.322813 [INFO] ozmod_zt.c:565 Setting txgain val  
> to 0.000000
> 2010-02-20 16:45:54.322813 [INFO] zap_io.c:2379 auto-loaded 'zt'
> 2010-02-20 16:45:54.322813 [WARNING] ozmod_zt.c:333 this ioctl fails
> on older zaptel but is harmless if you used ztcfg
> [device /dev/zap/channel chan 1 fd 27 (Invalid argument)]
> 2010-02-20 16:45:54.322813 [INFO] ozmod_zt.c:385 configuring device
> /dev/zap/channel channel 1 as OpenZAP device 1:1 fd:36
> 2010-02-20 16:45:54.322813 [WARNING] ozmod_zt.c:333 this ioctl fails
> on older zaptel but is harmless if you used ztcfg
> [device /dev/zap/channel chan 2 fd 27 (Invalid argument)]
> 2010-02-20 16:45:54.322813 [INFO] ozmod_zt.c:385 configuring device
> /dev/zap/channel channel 2 as OpenZAP device 2:1 fd:40
> 2010-02-20 16:45:54.322813 [WARNING] ozmod_zt.c:333 this ioctl fails
> on older zaptel but is harmless if you used ztcfg
> [device /dev/zap/channel chan 3 fd 27 (Invalid argument)]
> 2010-02-20 16:45:54.322813 [INFO] ozmod_zt.c:385 configuring device
> /dev/zap/channel channel 3 as OpenZAP device 3:1 fd:41
> 2010-02-20 16:45:54.322813 [WARNING] ozmod_zt.c:333 this ioctl fails
> on older zaptel but is harmless if you used ztcfg
> [device /dev/zap/channel chan 4 fd 27 (Invalid argument)]
> 2010-02-20 16:45:54.322813 [INFO] ozmod_zt.c:385 configuring device
> /dev/zap/channel channel 4 as OpenZAP device 4:1 fd:42
> 2010-02-20 16:45:54.322813 [INFO] zap_io.c:2502 Configured 4 channel 
> (s)
> 2010-02-20 16:45:54.322813 [INFO] zap_io.c:2596 Loading SIG from
> /opt/freeswitch/mod/ozmod_analog.so
> 2010-02-20 16:45:54.322813 [INFO] zap_io.c:2712 auto-loaded 'analog'
> 2010-02-20 16:45:54.358813 [CONSOLE] switch_loadable_module.c:900
> Successfully Loaded [mod_openzap]
> 2010-02-20 16:45:54.362813 [NOTICE] switch_loadable_module.c:144
> Adding Endpoint 'openzap'
> 2010-02-20 16:45:54.362813 [NOTICE] switch_loadable_module.c:250
> Adding Application 'disable_ec'
> 2010-02-20 16:45:54.362813 [NOTICE] switch_loadable_module.c:272
> Adding API Function 'oz'
>
> +OK module unloaded
> +OK module loaded
>
> freeswitch at voyage> oz list
>
> +OK
> span: 1 (FXO1)
> type: analog
> chan_count: 1
> dialplan: XML
> context: default
> dial_regex:
> fail_dial_regex:
> hold_music:
> analog_options 3way
> +OK
> span: 2 (FXO2)
> type: analog
> chan_count: 1
> dialplan: XML
> context: default
> dial_regex:
> fail_dial_regex:
> hold_music:
> analog_options 3way
> +OK
> span: 3 (FXS1)
> type: analog
> chan_count: 1
> dialplan: XML
> context: default
> dial_regex:
> fail_dial_regex:
> hold_music:
> analog_options none
> +OK
> span: 4 (FXS2)
> type: analog
> chan_count: 1
> dialplan: XML
> context: default
> dial_regex:
> fail_dial_regex:
> hold_music:
> analog_options none
>
>
> Yet, when I lift the handset on port 1 or port 2 I don't get a dial  
> tone :-(
>
> Instead I get this message:
>
> 2010-02-20 16:52:26.332813 [INFO] ozmod_zt.c:640 Setting echo cancel
> to 64 taps for 2:1
> 2010-02-20 16:52:26.332813 [ERR] zap_io.c:1599 I/O backend does not
> support command 24!
>
> Am I doing something wrong?
> -- 
> Brian May <brian at microcomaustralia.com.au>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org



From infos at madovsky.org  Sat Feb 20 10:36:01 2010
From: infos at madovsky.org (Madovsky)
Date: Sat, 20 Feb 2010 13:36:01 -0500
Subject: [Freeswitch-users] outbound calls
Message-ID: <C9E83EA051BC49D8AB9425D5D1225C27@MOBILEE1705>

Hello,

I'm able to transcode a cal between 2 local legs,
but when a local user call an oubound call,
the call hangs up saying "not acceptable here",
so it doesn't transcode.

Any idea ?

Thanks

Franck
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100220/73f2ebd5/attachment-0002.html 

From ederwander at gmail.com  Sat Feb 20 10:51:40 2010
From: ederwander at gmail.com (Eder Souza)
Date: Sat, 20 Feb 2010 16:51:40 -0200
Subject: [Freeswitch-users] openzap TDM400 card
In-Reply-To: <E94B1537-7943-4328-B090-AEF26C781673@freeswitch.org>
References: <3c5cf5261002192157t25de5798xc10ab2a7755e67d0@mail.gmail.com>
	<E94B1537-7943-4328-B090-AEF26C781673@freeswitch.org>
Message-ID: <622bedea1002201051y4de06dcdh9c1c1681e05c0d3b@mail.gmail.com>

Try set  loadzone  and defaultzone for your country in zaptel.conf

then do this:

ztcfg -vv

and

zttol -vvv  (see if status of your card is OK)


Att

Eng Eder de Souza




On Sat, Feb 20, 2010 at 4:10 PM, Michael S Collins <msc at freeswitch.org>wrote:

> Ports 1 and 2 are FXO which need a phone line. Try port 3 or 4. Also
> pastebin your zaptel.conf file.
>
> -MC
>
> Sent from my iPhone
>
> On Feb 19, 2010, at 9:57 PM, Brian May
> <brian at microcomaustralia.com.au> wrote:
>
> > Hello,
> >
> > I think I have the config correct, and not confused FXO/FXS anywhere.
> >
> > voyage:/opt/freeswitch/conf# ztcfg  -vv
> >
> > Zaptel Version: 1.4.11
> > Echo Canceller: MG2
> > Configuration
> > ======================
> >
> >
> > Channel map:
> >
> > Channel 01: FXO Kewlstart (Default) (Slaves: 01)
> > Channel 02: FXO Kewlstart (Default) (Slaves: 02)
> > Channel 03: FXS Kewlstart (Default) (Slaves: 03)
> > Channel 04: FXS Kewlstart (Default) (Slaves: 04)
> >
> > 4 channels to configure.
> >
> > voyage:/opt/freeswitch/conf# cat openzap.conf
> > [span zt FXO1]
> > name => OpenZAP-FXO1
> > number => 1
> > fxo-channel => 1
> >
> > [span zt FXO2]
> > name => OpenZAP-FXO2
> > number => 2
> > fxo-channel => 2
> >
> > [span zt FXS1]
> > name => OpenZAP-FXS1
> > number => 3
> > fxs-channel => 3
> >
> > [span zt FXS2]
> > name => OpenZAP-FXS2
> > number => 4
> > fxs-channel => 4
> >
> > voyage:/opt/freeswitch/conf# cat autoload_configs/openzap.conf.xml
> > <configuration name="openzap.conf" description="OpenZAP
> > Configuration">
> >  <settings>
> >    <param name="debug" value="0"/>
> >    <!--<param name="hold-music" value="$${moh_uri}"/>-->
> >    <!--<param name="enable-analog-option" value="call-swap"/>-->
> >    <!--<param name="enable-analog-option" value="3-way"/>-->
> >  </settings>
> >  <!-- one entry here per openzap span -->
> >  <analog_spans>
> >    <span id="1">
> >      <param name="tonegroup" value="au"/>
> >      <param name="digit-timeout" value="2000"/>
> >      <param name="max-digits" value="11"/>
> >      <param name="dialplan" value="XML"/>
> >      <param name="context" value="default"/>
> >      <param name="enable-analog-option" value="3-way"/>
> > <!--      <param name="moh" value="$${base_dir}/sounds/
> > Dardedisco.wav"/> -->
> >    </span>
> >    <span id="2">
> >      <param name="tonegroup" value="au"/>
> >      <param name="digit-timeout" value="2000"/>
> >      <param name="max-digits" value="11"/>
> >      <param name="dialplan" value="XML"/>
> >      <param name="context" value="default"/>
> >      <param name="enable-analog-option" value="3-way"/>
> > <!--      <param name="moh" value="$${base_dir}/sounds/
> > Dardedisco.wav"/> -->
> >    </span>
> >    <span id="3">
> >      <param name="tonegroup" value="au"/>
> >      <param name="digit-timeout" value="2000"/>
> >      <param name="max-digits" value="11"/>
> >      <param name="dialplan" value="XML"/>
> >      <param name="context" value="default"/>
> >    </span>
> >    <span id="4">
> >      <param name="tonegroup" value="au"/>
> >      <param name="digit-timeout" value="2000"/>
> >      <param name="max-digits" value="11"/>
> >      <param name="dialplan" value="XML"/>
> >      <param name="context" value="default"/>
> >    </span>
> >  </analog_spans>
> > </configuration>
> >
> >
> > freeswitch at voyage> reload mod_openzap
> > 2010-02-20 16:45:53.312813 [NOTICE] switch_loadable_module.c:464
> > Deleting Endpoint 'openzap'
> > 2010-02-20 16:45:53.312813 [NOTICE] switch_loadable_module.c:545
> > Deleting Application 'disable_ec'
> > 2010-02-20 16:45:53.312813 [NOTICE] switch_loadable_module.c:572
> > Deleting API Function 'oz'
> > 2010-02-20 16:45:53.312813 [CONSOLE] switch_loadable_module.c:1277
> > Stopping: mod_openzap
> > 2010-02-20 16:45:53.312813 [INFO] zap_io.c:269 Closing channel zt:
> > 1:1 fd:36
> > 2010-02-20 16:45:53.312813 [INFO] zap_io.c:269 Closing channel zt:
> > 2:1 fd:40
> > 2010-02-20 16:45:53.312813 [INFO] zap_io.c:269 Closing channel zt:
> > 3:1 fd:41
> > 2010-02-20 16:45:53.312813 [INFO] zap_io.c:269 Closing channel zt:
> > 4:1 fd:42
> > 2010-02-20 16:45:53.612813 [ERR] ozmod_analog.c:951 Failure Polling
> > event! [no matching descriptor]
> > 2010-02-20 16:45:54.162813 [ERR] ozmod_analog.c:951 Failure Polling
> > event! [no matching descriptor]
> > 2010-02-20 16:45:54.162813 [ERR] ozmod_analog.c:951 Failure Polling
> > event! [no matching descriptor]
> > 2010-02-20 16:45:54.162813 [ERR] ozmod_analog.c:951 Failure Polling
> > event! [no matching descriptor]
> > 2010-02-20 16:45:54.322813 [INFO] zap_io.c:2694 Unloading
> > /opt/freeswitch/mod/ozmod_analog.so
> > 2010-02-20 16:45:54.322813 [INFO] zap_io.c:2679 Unloading IO zt
> > 2010-02-20 16:45:54.322813 [INFO] zap_io.c:2694 Unloading
> > /opt/freeswitch/mod/ozmod_zt.so
> > 2010-02-20 16:45:54.322813 [CONSOLE] switch_loadable_module.c:1297
> > mod_openzap unloaded.
> > 2010-02-20 16:45:54.322813 [NOTICE] zap_io.c:2778 Modules
> > configured: 1
> > 2010-02-20 16:45:54.322813 [NOTICE] ozmod_zt.c:1161 Using Zaptel
> > control device
> > 2010-02-20 16:45:54.322813 [INFO] zap_io.c:2579 Loading IO from
> > /opt/freeswitch/mod/ozmod_zt.so [zt]
> > 2010-02-20 16:45:54.322813 [INFO] ozmod_zt.c:556 Setting rxgain val
> > to 0.000000
> > 2010-02-20 16:45:54.322813 [INFO] ozmod_zt.c:565 Setting txgain val
> > to 0.000000
> > 2010-02-20 16:45:54.322813 [INFO] zap_io.c:2379 auto-loaded 'zt'
> > 2010-02-20 16:45:54.322813 [WARNING] ozmod_zt.c:333 this ioctl fails
> > on older zaptel but is harmless if you used ztcfg
> > [device /dev/zap/channel chan 1 fd 27 (Invalid argument)]
> > 2010-02-20 16:45:54.322813 [INFO] ozmod_zt.c:385 configuring device
> > /dev/zap/channel channel 1 as OpenZAP device 1:1 fd:36
> > 2010-02-20 16:45:54.322813 [WARNING] ozmod_zt.c:333 this ioctl fails
> > on older zaptel but is harmless if you used ztcfg
> > [device /dev/zap/channel chan 2 fd 27 (Invalid argument)]
> > 2010-02-20 16:45:54.322813 [INFO] ozmod_zt.c:385 configuring device
> > /dev/zap/channel channel 2 as OpenZAP device 2:1 fd:40
> > 2010-02-20 16:45:54.322813 [WARNING] ozmod_zt.c:333 this ioctl fails
> > on older zaptel but is harmless if you used ztcfg
> > [device /dev/zap/channel chan 3 fd 27 (Invalid argument)]
> > 2010-02-20 16:45:54.322813 [INFO] ozmod_zt.c:385 configuring device
> > /dev/zap/channel channel 3 as OpenZAP device 3:1 fd:41
> > 2010-02-20 16:45:54.322813 [WARNING] ozmod_zt.c:333 this ioctl fails
> > on older zaptel but is harmless if you used ztcfg
> > [device /dev/zap/channel chan 4 fd 27 (Invalid argument)]
> > 2010-02-20 16:45:54.322813 [INFO] ozmod_zt.c:385 configuring device
> > /dev/zap/channel channel 4 as OpenZAP device 4:1 fd:42
> > 2010-02-20 16:45:54.322813 [INFO] zap_io.c:2502 Configured 4 channel
> > (s)
> > 2010-02-20 16:45:54.322813 [INFO] zap_io.c:2596 Loading SIG from
> > /opt/freeswitch/mod/ozmod_analog.so
> > 2010-02-20 16:45:54.322813 [INFO] zap_io.c:2712 auto-loaded 'analog'
> > 2010-02-20 16:45:54.358813 [CONSOLE] switch_loadable_module.c:900
> > Successfully Loaded [mod_openzap]
> > 2010-02-20 16:45:54.362813 [NOTICE] switch_loadable_module.c:144
> > Adding Endpoint 'openzap'
> > 2010-02-20 16:45:54.362813 [NOTICE] switch_loadable_module.c:250
> > Adding Application 'disable_ec'
> > 2010-02-20 16:45:54.362813 [NOTICE] switch_loadable_module.c:272
> > Adding API Function 'oz'
> >
> > +OK module unloaded
> > +OK module loaded
> >
> > freeswitch at voyage> oz list
> >
> > +OK
> > span: 1 (FXO1)
> > type: analog
> > chan_count: 1
> > dialplan: XML
> > context: default
> > dial_regex:
> > fail_dial_regex:
> > hold_music:
> > analog_options 3way
> > +OK
> > span: 2 (FXO2)
> > type: analog
> > chan_count: 1
> > dialplan: XML
> > context: default
> > dial_regex:
> > fail_dial_regex:
> > hold_music:
> > analog_options 3way
> > +OK
> > span: 3 (FXS1)
> > type: analog
> > chan_count: 1
> > dialplan: XML
> > context: default
> > dial_regex:
> > fail_dial_regex:
> > hold_music:
> > analog_options none
> > +OK
> > span: 4 (FXS2)
> > type: analog
> > chan_count: 1
> > dialplan: XML
> > context: default
> > dial_regex:
> > fail_dial_regex:
> > hold_music:
> > analog_options none
> >
> >
> > Yet, when I lift the handset on port 1 or port 2 I don't get a dial
> > tone :-(
> >
> > Instead I get this message:
> >
> > 2010-02-20 16:52:26.332813 [INFO] ozmod_zt.c:640 Setting echo cancel
> > to 64 taps for 2:1
> > 2010-02-20 16:52:26.332813 [ERR] zap_io.c:1599 I/O backend does not
> > support command 24!
> >
> > Am I doing something wrong?
> > --
> > Brian May <brian at microcomaustralia.com.au>
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> > users
> > http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100220/8757e35f/attachment-0002.html 

From jmesquita at freeswitch.org  Sat Feb 20 13:27:00 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Sat, 20 Feb 2010 19:27:00 -0200
Subject: [Freeswitch-users] How to exchange media when I developed new
	endpoint module?
In-Reply-To: <93b0f8ce1002191722h5782b787qe56a06dd0cb94e76@mail.gmail.com>
References: <93b0f8ce1002091914q581f12fvd1de01e2181f0f3a@mail.gmail.com>
	<ebbc2d551002091932u52dbf1b6jfecf2dfdb43b8a7c@mail.gmail.com>
	<4021C303-1AC3-48F2-A17B-63C02C1C54D4@freeswitch.org>
	<93b0f8ce1002191722h5782b787qe56a06dd0cb94e76@mail.gmail.com>
Message-ID: <ebbc2d551002201327o7a7bad26g48cac48f610aed55@mail.gmail.com>

I developed the current implementation of mod_khomp. I wouldn't take it as
an example for anything since there has been no activity there for the past
4 months. If you care to share a snippet of your code, maybe we can help
better.


JM


On Fri, Feb 19, 2010 at 11:22 PM, ??? <spiritonly at gmail.com> wrote:

> Do you know mod_khomp? You can found it in FS wiki. I am developing an
> endpoint module like it.
> So you can give me some advice to bridge two session?
>
>
> On Wed, Feb 10, 2010 at 11:44 AM, Brian West <brian at freeswitch.org> wrote:
>
>> But the bigger question is what protocol are you doing that you have to
>> create your own endpoint module?
>>
>> /b
>>
>> On Feb 9, 2010, at 9:32 PM, Jo?o Mesquita wrote:
>>
>> > You should look at read_frame and write_frame implementations of other
>> endpoint modules.
>> >
>> > This should pretty much tell you how things work...
>> >
>> > Jo?o Mesquita
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100220/86320003/attachment-0002.html 

From dave at 3c.co.uk  Sat Feb 20 13:39:56 2010
From: dave at 3c.co.uk (David Knell)
Date: Sat, 20 Feb 2010 14:39:56 -0700
Subject: [Freeswitch-users] LUA script providing dynamic
	directoryinformation
References: <28f27f5d1002200948j33d818e1h9fc8c8f99a506656@mail.gmail.com>
Message-ID: <A80748C5894146E594F157A9811E1011@DELL9>

Hi Maciej,

Your problem is that Lua won't automatically substitute variables inside a 
string - which is why you're just seeing ..row.email.. passed directly 
through.

If you replace that line in the source with something like
<param name="vm-mailto" value="]] .. row.email .. [["/>

the it's more likely to work.

Cheers --

Dave

----- Original Message ----- 
From: "Maciej Bylica" <mbsip at gazeta.pl>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Saturday, February 20, 2010 10:48 AM
Subject: [Freeswitch-users] LUA script providing dynamic 
directoryinformation


> Hello,
>
> I am trying to use mod_lua to provide dynamic directory information
> (binding in mod_lua.conf.xml)
> Here is my script.
> #!/usr/local/bin/lua
> -- load driver
> require "luasql.odbc"
> -- create environment object
> env = luasql.odbc();
> -- connect to data source
> conn = env:connect("freeswitch","root");
> -- reset our table
> if ( conn ~= nil ) then
> cur = conn:execute(string.format("SELECT email from VM where
> called_num='48112223344'"));
>
> if ( cur ~= nil ) then
> row = cur:fetch({}, "a");
> if ( row ~= nil ) then
>
> freeswitch.consoleLog("info", " Email fetched from DB is = "..
> row.email .."\n");
>
> cur:close(); conn:close(); env:close();
>
> mydialplan = [[
> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
> <document type="freeswitch/xml">
>  <section name="directory">
>    <domain name="10.10.10.1">
>      <user id="48112223344">
>        <params>
>          <param name="vm-email-all-messages" value="true"/>
>          <param name="vm-attach-file" value="true"/>
>          <param name="vm-disk-quota" value="0"/>
>          <param name="vm-email-all-messages" value="true"/>
>          <param name="vm-mailto" value=".. row.email .."/>
>          <param name="vm-keep-local-after-email" value="false"/>
>        </params>
>      </user>
>    </domain>
>  </section>
> </document>
> ]]
>
> XML_STRING = mydialplan
> end
> end
> end
>
>
> I've encountered a problem how to pass row.email gathered from DB
> directly to XML (vm-mailto).
> As you can see below configuration I have does not work properly.
>
> 2010-02-20 20:37:03.267071 [DEBUG] mod_voicemail.c:2358 Deliver VM to
> 48112223344 at 10.10.10.1
> 2010-02-20 20:37:03.276533 [INFO] switch_cpp.cpp:1129  Email fetched
> from DB is = hereis at MyEmaill.com
> 2010-02-20 20:37:03.435902 [DEBUG] switch_utils.c:631 Emailed file
> [/tmp/mail.1266694623f261] to [.. row.email ..]
> 2010-02-20 20:37:03.435902 [DEBUG] mod_voicemail.c:2526 Sending
> message to .. row.email ..
>
> I tried with:
> <param name="vm-mailto" value=.. row.email ../>
> <param name="vm-mailto" value="".. row.email ..""/>
> Both of them do not produce any "Sending message" output at all.
>
>
> Any thoughts?
> Thanks in advance.
> Maciej
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> 




From brian at microcomaustralia.com.au  Sat Feb 20 14:08:05 2010
From: brian at microcomaustralia.com.au (Brian May)
Date: Sun, 21 Feb 2010 09:08:05 +1100
Subject: [Freeswitch-users] openzap TDM400 card
In-Reply-To: <C8B78F95DB344ADDA010C52AE0AC8B66@cune.pri>
References: <3c5cf5261002192157t25de5798xc10ab2a7755e67d0@mail.gmail.com>
	<3c5cf5261002200148k1bb3a1f7mf97fabccfbd1c178@mail.gmail.com>
	<C8B78F95DB344ADDA010C52AE0AC8B66@cune.pri>
Message-ID: <3c5cf5261002201408v63c150d3v419493ab5e4e2089@mail.gmail.com>

On 20 February 2010 22:25, Russell Mosemann <Russell.Mosemann at cune.org> wrote:
> I believe it is, with the specific error you mention, "Cannot get a RING_START event on a non-fxo channel".

That error was clear. I was able to solve the issue when I saw that,
without having to look up the documentation in fact.

Only I spent ages before I received this error, trying to debug the
other ports with the phone lines connected. These ports didn't give
any errors, except for stuff I can ignore. Based on the other
responses I have got here, it appears other people were confused too.

Lets see if I have got this correct:

Ports 1 & 2 are FXS, but use FXO signalling. So, yes I do plug a phone
into these.
Ports 3 & 4 are FXO, but use FXS signalling. So I plug the phone line
into these.


I will update the wiki, but there is still one part I don't
understand, see the example at the end here:

<http://wiki.freeswitch.org/wiki/Openzap.conf.xml_Examples>

I assume ports 1&2 (with 3-way and moh set) are the FXS ports that use
the FXO signalling? i.e. the same numbering as what I have?

Do I really need to set moh for every line in this file?


Looking at the openzap part of the wiki, the only thing I see related
is "I didn't understand why we have to swap the channel numbers for
FXS and FXO." There is also a link to
http://unixtoys.ca/wordpress/2008/09/freeswitch-and-tdm-hardware-pots-fxofxs/
which looks like it would have helped but unfortunately the link
appears to be broken. Am going to try and update these issues now.
-- 
Brian May <brian at microcomaustralia.com.au>



From mbsip at gazeta.pl  Sat Feb 20 14:39:25 2010
From: mbsip at gazeta.pl (Maciej Bylica)
Date: Sat, 20 Feb 2010 23:39:25 +0100
Subject: [Freeswitch-users] LUA script providing dynamic
	directoryinformation
In-Reply-To: <A80748C5894146E594F157A9811E1011@DELL9>
References: <28f27f5d1002200948j33d818e1h9fc8c8f99a506656@mail.gmail.com>
	<A80748C5894146E594F157A9811E1011@DELL9>
Message-ID: <28f27f5d1002201439j30d4b88cn14ad07ae83067ed5@mail.gmail.com>

Hi Dave,

Yesssss it works :P
I have been struggling with this for hours, so I appreciate Your help.

Thx,
Maciej.

> Hi Maciej,
>
> Your problem is that Lua won't automatically substitute variables inside a
> string - which is why you're just seeing ..row.email.. passed directly
> through.
>
> If you replace that line in the source with something like
> <param name="vm-mailto" value="]] .. row.email .. [["/>
>
> the it's more likely to work.
>
> Cheers --
>
> Dave
>
> ----- Original Message -----
> From: "Maciej Bylica" <mbsip at gazeta.pl>
> To: <freeswitch-users at lists.freeswitch.org>
> Sent: Saturday, February 20, 2010 10:48 AM
> Subject: [Freeswitch-users] LUA script providing dynamic
> directoryinformation
>
>
>> Hello,
>>
>> I am trying to use mod_lua to provide dynamic directory information
>> (binding in mod_lua.conf.xml)
>> Here is my script.
>> #!/usr/local/bin/lua
>> -- load driver
>> require "luasql.odbc"
>> -- create environment object
>> env = luasql.odbc();
>> -- connect to data source
>> conn = env:connect("freeswitch","root");
>> -- reset our table
>> if ( conn ~= nil ) then
>> cur = conn:execute(string.format("SELECT email from VM where
>> called_num='48112223344'"));
>>
>> if ( cur ~= nil ) then
>> row = cur:fetch({}, "a");
>> if ( row ~= nil ) then
>>
>> freeswitch.consoleLog("info", " Email fetched from DB is = "..
>> row.email .."\n");
>>
>> cur:close(); conn:close(); env:close();
>>
>> mydialplan = [[
>> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
>> <document type="freeswitch/xml">
>> ?<section name="directory">
>> ? ?<domain name="10.10.10.1">
>> ? ? ?<user id="48112223344">
>> ? ? ? ?<params>
>> ? ? ? ? ?<param name="vm-email-all-messages" value="true"/>
>> ? ? ? ? ?<param name="vm-attach-file" value="true"/>
>> ? ? ? ? ?<param name="vm-disk-quota" value="0"/>
>> ? ? ? ? ?<param name="vm-email-all-messages" value="true"/>
>> ? ? ? ? ?<param name="vm-mailto" value=".. row.email .."/>
>> ? ? ? ? ?<param name="vm-keep-local-after-email" value="false"/>
>> ? ? ? ?</params>
>> ? ? ?</user>
>> ? ?</domain>
>> ?</section>
>> </document>
>> ]]
>>
>> XML_STRING = mydialplan
>> end
>> end
>> end
>>
>>
>> I've encountered a problem how to pass row.email gathered from DB
>> directly to XML (vm-mailto).
>> As you can see below configuration I have does not work properly.
>>
>> 2010-02-20 20:37:03.267071 [DEBUG] mod_voicemail.c:2358 Deliver VM to
>> 48112223344 at 10.10.10.1
>> 2010-02-20 20:37:03.276533 [INFO] switch_cpp.cpp:1129 ?Email fetched
>> from DB is = hereis at MyEmaill.com
>> 2010-02-20 20:37:03.435902 [DEBUG] switch_utils.c:631 Emailed file
>> [/tmp/mail.1266694623f261] to [.. row.email ..]
>> 2010-02-20 20:37:03.435902 [DEBUG] mod_voicemail.c:2526 Sending
>> message to .. row.email ..
>>
>> I tried with:
>> <param name="vm-mailto" value=.. row.email ../>
>> <param name="vm-mailto" value="".. row.email ..""/>
>> Both of them do not produce any "Sending message" output at all.
>>
>>
>> Any thoughts?
>> Thanks in advance.
>> Maciej
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From brian at microcomaustralia.com.au  Sat Feb 20 16:32:46 2010
From: brian at microcomaustralia.com.au (Brian May)
Date: Sun, 21 Feb 2010 11:32:46 +1100
Subject: [Freeswitch-users] building for Lenny
In-Reply-To: <3c5cf5261002191438o47aabde9tb588fe3ef23f27bc@mail.gmail.com>
References: <3c5cf5261002180046vb0015e7t787599c9903aedc@mail.gmail.com>
	<33c87fa31002180056g51ef8496qa9a35746d0fe75ec@mail.gmail.com>
	<3c5cf5261002180110he0ae815uc713f77f289ac625@mail.gmail.com>
	<a9f4977a70b3fcf1b30413ea562ca962.squirrel@www.webcontracts.co.uk>
	<20100218161741.GC4236@base.carmickle.com>
	<3c5cf5261002181542y43adb7a7ofe2a15c38baf0327@mail.gmail.com>
	<4C13BCBF-1DA0-47AE-A749-738AD6D8FE48@jerris.com>
	<3c5cf5261002191438o47aabde9tb588fe3ef23f27bc@mail.gmail.com>
Message-ID: <3c5cf5261002201632j33256190ve36b667db928eedb@mail.gmail.com>

Just noticed the latest efforts for getting this packaged and into
Debian proper are described here:

http://bugs.debian.org/cgi-bin/bugreport.cgi?bug=389591
-- 
Brian May <brian at microcomaustralia.com.au>



From Prometheus001 at gmx.net  Sat Feb 20 16:44:22 2010
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Sun, 21 Feb 2010 01:44:22 +0100
Subject: [Freeswitch-users] more than 1 profile for sofia/user ?
In-Reply-To: <ebbc2d551002181929m72a17b7bj1fa1cff45b02607b@mail.gmail.com>
References: <4B7D9494.8050208@gmx.net>	<191c3a031002181140y44135106yb34e6a6f73a04773@mail.gmail.com>	<4B7DADC8.1060405@gmx.net>	<191c3a031002181353r2dbf15c6h206509f9b1399148@mail.gmail.com>	<4B7DC202.7090409@gmx.net>
	<ebbc2d551002181929m72a17b7bj1fa1cff45b02607b@mail.gmail.com>
Message-ID: <4B8081E6.8060806@gmx.net>

Thanks Jo?o, this solved my problem.

Just for the records how it works:

    * created a new profile "internalnat" as a copy of "internal"
    * modified ports 5060 and 5061 to 5065 and 5066
    * added parameters external-rtp-ip and external-sip-ip with external IPs
    * modified the dialstring as proposed below
    * Phone registers, phone can dial and can be called.

Freeswitch rocks!

Best regards
Peter

Jo?o Mesquita schrieb:
> I would:
>
> {presence_id=${dialed_user}@${dialed_domain},transfer_fallback_extension=${dialed_user}}${sofia_contact(${dialed_user}@${dialed_domain}),sofia/other_profile/${dialed_user}}
>
> You could toy with that a bit. The dialstring is really just an
> origination string that is generated by the user/<extension> ...
>
> Hope that clears it up a bit.
>
> Regards,
> Jo?o Mesquita
>
>
>
> On Thu, Feb 18, 2010 at 7:41 PM, Peter P GMX <Prometheus001 at gmx.net
> <mailto:Prometheus001 at gmx.net>> wrote:
>
>     Hello Anthony,
>
>     >add on a , then another dial string to reflect the other profile too
>     I really tried to understand this, but
>     can you give me an example?
>
>     Best regards
>     Peter
>
>     Anthony Minessale schrieb:
>     > add on a , then another dial string to reflect the other profile too
>     >
>     > On Thu, Feb 18, 2010 at 3:14 PM, Peter P GMX
>     <Prometheus001 at gmx.net <mailto:Prometheus001 at gmx.net>
>     > <mailto:Prometheus001 at gmx.net <mailto:Prometheus001 at gmx.net>>>
>     wrote:
>     >
>     >     Any idea how to do this?
>     >
>     >     currently I have
>     >    
>     {presence_id=${dialed_user}@${dialed_domain},transfer_fallback_extension=${dialed_user}}${sofia_contact(${dialed_user}@${dialed_domain})}
>     >
>     >
>     >     Best regards
>     >     Peter
>     >
>     >     Anthony Minessale schrieb:
>     >     > edit the dial-string for that user in the directory xml to
>     try the
>     >     > extension on both profile at once
>     >     >
>     >     > On Thu, Feb 18, 2010 at 1:27 PM, Peter P GMX
>     >     <Prometheus001 at gmx.net <mailto:Prometheus001 at gmx.net>
>     <mailto:Prometheus001 at gmx.net <mailto:Prometheus001 at gmx.net>>
>     >     > <mailto:Prometheus001 at gmx.net
>     <mailto:Prometheus001 at gmx.net> <mailto:Prometheus001 at gmx.net
>     <mailto:Prometheus001 at gmx.net>>>>
>     >     wrote:
>     >     >
>     >     >     Hello,
>     >     >
>     >     >     in the standard setup - if a phone is registering to port
>     >     5060 - it is
>     >     >     bound to the "internal" profile. And I can dial it via
>     >     >     sofia/user/xxxx then.
>     >     >
>     >     >     However due to NAT issues I would like to have to 2
>     seperate
>     >     profiles
>     >     >     for SIP phones. For example I have a "local" profile
>     for all
>     >     devices
>     >     >     inside the LAN (e.g. Pattons und in future: local phones)
>     >     and another
>     >     >     "internal" profile which allows also external phones via
>     >     >     external-xxx-ip. That way I would like to ensure that
>     local
>     >     phones
>     >     >     have
>     >     >     nothing to do with natted adresses and that external
>     phones can
>     >     >     register
>     >     >     via external IPs.
>     >     >
>     >     >     Question How do I manage that I can register a phone
>     to the
>     >     "local"
>     >     >     profile and being able to dial that phone via
>     sofia/user/xxxxx?
>     >     >
>     >     >     Or do I think too complicated and there is simply nothing
>     >     special
>     >     >     to do?
>     >     >
>     >     >     Best regards
>     >     >     Peter
>     >     >
>     >     >     _______________________________________________
>     >     >     FreeSWITCH-users mailing list
>     >     >     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     >     <mailto:FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>>
>     >     >     <mailto:FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     >     <mailto:FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>>>
>     >     >    
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     >     >
>     >    
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     >     >     http://www.freeswitch.org
>     >     >
>     >     >
>     >     >
>     >     >
>     >     > --
>     >     > Anthony Minessale II
>     >     >
>     >     > FreeSWITCH http://www.freeswitch.org/
>     >     > ClueCon http://www.cluecon.com/
>     >     > Twitter: http://twitter.com/FreeSWITCH_wire
>     >     >
>     >     > AIM: anthm
>     >     > MSN:anthony_minessale at hotmail.com
>     <mailto:MSN%3Aanthony_minessale at hotmail.com>
>     >     <mailto:MSN%3Aanthony_minessale at hotmail.com
>     <mailto:MSN%253Aanthony_minessale at hotmail.com>>
>     >     > <mailto:MSN%3Aanthony_minessale at hotmail.com
>     <mailto:MSN%253Aanthony_minessale at hotmail.com>
>     >     <mailto:MSN%253Aanthony_minessale at hotmail.com
>     <mailto:MSN%25253Aanthony_minessale at hotmail.com>>>
>     >     > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>     <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
>     >     <mailto:PAYPAL%3Aanthony.minessale at gmail.com
>     <mailto:PAYPAL%253Aanthony.minessale at gmail.com>>
>     >     > <mailto:PAYPAL%3Aanthony.minessale at gmail.com
>     <mailto:PAYPAL%253Aanthony.minessale at gmail.com>
>     >     <mailto:PAYPAL%253Aanthony.minessale at gmail.com
>     <mailto:PAYPAL%25253Aanthony.minessale at gmail.com>>>
>     >     > IRC: irc.freenode.net <http://irc.freenode.net>
>     <http://irc.freenode.net>
>     >     <http://irc.freenode.net> #freeswitch
>     >     >
>     >     > FreeSWITCH Developer Conference
>     >     > sip:888 at conference.freeswitch.org
>     <mailto:sip%3A888 at conference.freeswitch.org>
>     >     <mailto:sip%3A888 at conference.freeswitch.org
>     <mailto:sip%253A888 at conference.freeswitch.org>>
>     >     > <mailto:sip%3A888 at conference.freeswitch.org
>     <mailto:sip%253A888 at conference.freeswitch.org>
>     >     <mailto:sip%253A888 at conference.freeswitch.org
>     <mailto:sip%25253A888 at conference.freeswitch.org>>>
>     >     > iax:guest at conference.freeswitch.org/888
>     <http://iax:guest at conference.freeswitch.org/888>
>     >     <http://iax:guest at conference.freeswitch.org/888>
>     >     > <http://iax:guest at conference.freeswitch.org/888>
>     >     > googletalk:conf+888 at conference.freeswitch.org
>     <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
>     >     <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org
>     <mailto:googletalk%253Aconf%252B888 at conference.freeswitch.org>>
>     >     > <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org
>     <mailto:googletalk%253Aconf%252B888 at conference.freeswitch.org>
>     >    
>     <mailto:googletalk%253Aconf%252B888 at conference.freeswitch.org
>     <mailto:googletalk%25253Aconf%25252B888 at conference.freeswitch.org>>>
>     >     > pstn:+19193869900
>     >     >
>     >    
>     ------------------------------------------------------------------------
>     >     >
>     >     > _______________________________________________
>     >     > FreeSWITCH-users mailing list
>     >     > FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     >     <mailto:FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>>
>     >     > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     >     >
>     >    
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     >     > http://www.freeswitch.org
>     >     >
>     >
>     >     _______________________________________________
>     >     FreeSWITCH-users mailing list
>     >     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     >     <mailto:FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>>
>     >     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     >    
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     >     http://www.freeswitch.org
>     >
>     >
>     >
>     >
>     > --
>     > Anthony Minessale II
>     >
>     > FreeSWITCH http://www.freeswitch.org/
>     > ClueCon http://www.cluecon.com/
>     > Twitter: http://twitter.com/FreeSWITCH_wire
>     >
>     > AIM: anthm
>     > MSN:anthony_minessale at hotmail.com
>     <mailto:MSN%3Aanthony_minessale at hotmail.com>
>     > <mailto:MSN%3Aanthony_minessale at hotmail.com
>     <mailto:MSN%253Aanthony_minessale at hotmail.com>>
>     > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>     <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
>     > <mailto:PAYPAL%3Aanthony.minessale at gmail.com
>     <mailto:PAYPAL%253Aanthony.minessale at gmail.com>>
>     > IRC: irc.freenode.net <http://irc.freenode.net>
>     <http://irc.freenode.net> #freeswitch
>     >
>     > FreeSWITCH Developer Conference
>     > sip:888 at conference.freeswitch.org
>     <mailto:sip%3A888 at conference.freeswitch.org>
>     > <mailto:sip%3A888 at conference.freeswitch.org
>     <mailto:sip%253A888 at conference.freeswitch.org>>
>     > iax:guest at conference.freeswitch.org/888
>     <http://iax:guest at conference.freeswitch.org/888>
>     > <http://iax:guest at conference.freeswitch.org/888>
>     > googletalk:conf+888 at conference.freeswitch.org
>     <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
>     > <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org
>     <mailto:googletalk%253Aconf%252B888 at conference.freeswitch.org>>
>     > pstn:+19193869900
>     >
>     ------------------------------------------------------------------------
>     >
>     > _______________________________________________
>     > FreeSWITCH-users mailing list
>     > FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     >
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     > http://www.freeswitch.org
>     >
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   



From will.traenkle at yahoo.com  Fri Feb 19 22:26:18 2010
From: will.traenkle at yahoo.com (William Traenkle)
Date: Fri, 19 Feb 2010 22:26:18 -0800 (PST)
Subject: [Freeswitch-users] freeswitch.serial
Message-ID: <921026.87533.qm@web57602.mail.re1.yahoo.com>

A couple of quick questions:

1) what is the freeswitch.serial file under the &base_dir/conf directory?

2) can I change its location?

Thanks,

-Will


      
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100219/60761da9/attachment-0002.html 

From mike at jerris.com  Sat Feb 20 17:47:17 2010
From: mike at jerris.com (Michael Jerris)
Date: Sat, 20 Feb 2010 20:47:17 -0500
Subject: [Freeswitch-users] stream a file multicast with mod_esf
In-Reply-To: <a792304b1002192150n44cc500fnd5b2ecb7cff9dd1a@mail.gmail.com>
References: <B98B6921358D0C49BD21CD023F1F52C302623B4B@hbghs04a.ww300.siemens.net>
	<7C996136-963A-42D2-B0FE-D2729F968E52@freeswitch.org>
	<27071973.post@talk.nabble.com>
	<378CDEC3-5A41-4EF9-927A-311F2531E6AB@freeswitch.org>
	<a792304b1001080725y19ff28camd37425927550f311@mail.gmail.com>
	<191c3a031001081848w9036207yf6a4b473ec233243@mail.gmail.com>
	<85C74329-0D7C-48B2-B7A8-D4AA67C6E3F2@freeswitch.org>
	<a792304b1002182209m43ada13cp8ed3d91ad509f0a2@mail.gmail.com>
	<DBA4E291-9377-489A-841A-1A3D83880DF2@jerris.com>
	<a792304b1002190002u50ec6912g1905ffc63a119174@mail.gmail.com>
	<a792304b1002192150n44cc500fnd5b2ecb7cff9dd1a@mail.gmail.com>
Message-ID: <795CD2F3-1395-4C08-A0E7-F5A7BA85F938@jerris.com>

You will need to create the codec for what you need, I think it is hardcoded in there to PCMU at the moment, correct?  This will of course need to match the stream its reading.

Mike

On Feb 20, 2010, at 12:50 AM, MohammedShehzad wrote:

> 
> 
> On Fri, Feb 19, 2010 at 1:32 PM, MohammedShehzad <pmhshz at gmail.com> wrote:
> 
> 
> On Fri, Feb 19, 2010 at 12:59 PM, Michael Jerris <mike at jerris.com> wrote:
> Listening on multicast is noting special for multicast, it is just
> like reading any other udp socket
> 
> Mike
> 
> Correct, but I have to play those audio stream back to caller taking care of the audio codec  and other things, do anybody have any idea in that part? Please let me know that.
> -- 
> 
> -MohammedShehzad
> 
> I am able to receive the play the multicasted RAW PCMU RTP (modified the skel of format provided by brian), so that caller can hear the multicast which done by other Freeswitch server using mod_esf application, but when i change the caller's codec from PCMU to something else, it breaks.

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100220/753626b2/attachment-0002.html 

From mike at jerris.com  Sat Feb 20 17:48:39 2010
From: mike at jerris.com (Michael Jerris)
Date: Sat, 20 Feb 2010 20:48:39 -0500
Subject: [Freeswitch-users] outbound calls
In-Reply-To: <C9E83EA051BC49D8AB9425D5D1225C27@MOBILEE1705>
References: <C9E83EA051BC49D8AB9425D5D1225C27@MOBILEE1705>
Message-ID: <F0495877-E2B4-4830-AAA0-889D95CC2EEA@jerris.com>

Sounds like the remote end does not like the codecs we are offering.  There are some vars to adjust this, by default we only offer the 1 codec chosen with the a leg of the call.

Mike

On Feb 20, 2010, at 1:36 PM, Madovsky wrote:

> Hello,
>  
> I'm able to transcode a cal between 2 local legs,
> but when a local user call an oubound call,
> the call hangs up saying "not acceptable here",
> so it doesn't transcode.
>  

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100220/b7289bac/attachment-0002.html 

From mike at jerris.com  Sat Feb 20 17:50:37 2010
From: mike at jerris.com (Michael Jerris)
Date: Sat, 20 Feb 2010 20:50:37 -0500
Subject: [Freeswitch-users] How to exchange media when I developed new
	endpoint module?
In-Reply-To: <ebbc2d551002201327o7a7bad26g48cac48f610aed55@mail.gmail.com>
References: <93b0f8ce1002091914q581f12fvd1de01e2181f0f3a@mail.gmail.com>
	<ebbc2d551002091932u52dbf1b6jfecf2dfdb43b8a7c@mail.gmail.com>
	<4021C303-1AC3-48F2-A17B-63C02C1C54D4@freeswitch.org>
	<93b0f8ce1002191722h5782b787qe56a06dd0cb94e76@mail.gmail.com>
	<ebbc2d551002201327o7a7bad26g48cac48f610aed55@mail.gmail.com>
Message-ID: <77784B66-AF0B-47FE-9DDB-D274F22E65E9@jerris.com>

If you are developing a module for hardware and you do not already have and want to use code for all the signaling, (pri, analog, etc) then take a look down at openzap.  This has all that for you already, and should be fairly trivial to implement a new piece of hardware.

Mike

On Feb 20, 2010, at 4:27 PM, Jo?o Mesquita wrote:

> I developed the current implementation of mod_khomp. I wouldn't take it as an example for anything since there has been no activity there for the past 4 months. If you care to share a snippet of your code, maybe we can help better.
> 
> On Fri, Feb 19, 2010 at 11:22 PM, ??? <spiritonly at gmail.com> wrote:
> Do you know mod_khomp? You can found it in FS wiki. I am developing an endpoint module like it.
> So you can give me some advice to bridge two session?
> 
> 
> On Wed, Feb 10, 2010 at 11:44 AM, Brian West <brian at freeswitch.org> wrote:
> But the bigger question is what protocol are you doing that you have to create your own endpoint module?
> 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100220/c514a77a/attachment-0002.html 

From rupa at rupa.com  Sat Feb 20 17:53:31 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Sat, 20 Feb 2010 19:53:31 -0600
Subject: [Freeswitch-users] freeswitch.serial
In-Reply-To: <921026.87533.qm@web57602.mail.re1.yahoo.com>
References: <921026.87533.qm@web57602.mail.re1.yahoo.com>
Message-ID: <b39273531002201753x16c7be04n1a73d181141bee64@mail.gmail.com>

1) The only current user of it is zrtp

2) it is configured to be in the conf directory, can't move it without
changing src.

On Sat, Feb 20, 2010 at 12:26 AM, William Traenkle
<will.traenkle at yahoo.com>wrote:

> A couple of quick questions:
>
> 1) what is the freeswitch.serial file under the &base_dir/conf directory?
>
> 2) can I change its location?
>
> Thanks,
>
> -Will
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100220/ab86ecb2/attachment-0002.html 

From mike at jerris.com  Sat Feb 20 17:53:58 2010
From: mike at jerris.com (Michael Jerris)
Date: Sat, 20 Feb 2010 20:53:58 -0500
Subject: [Freeswitch-users] freeswitch.serial
In-Reply-To: <921026.87533.qm@web57602.mail.re1.yahoo.com>
References: <921026.87533.qm@web57602.mail.re1.yahoo.com>
Message-ID: <BCB79C3E-4EFB-4FFA-B4D9-B4CB69A8F947@jerris.com>

On Feb 20, 2010, at 1:26 AM, William Traenkle wrote:

> A couple of quick questions:
> 
> 1) what is the freeswitch.serial file under the &base_dir/conf directory?

It is a unique number for that instance of FreeSWITCH, it is mostly used for zrtp

> 2) can I change its location?

No


Mike
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100220/345f450e/attachment-0002.html 

From infos at madovsky.org  Sat Feb 20 17:57:44 2010
From: infos at madovsky.org (Madovsky)
Date: Sat, 20 Feb 2010 20:57:44 -0500
Subject: [Freeswitch-users] outbound calls
References: <C9E83EA051BC49D8AB9425D5D1225C27@MOBILEE1705>
	<F0495877-E2B4-4830-AAA0-889D95CC2EEA@jerris.com>
Message-ID: <634B5DC983EC4267993346BB172A4FD1@MOBILEE1705>

yes I know it is codec problem,
but what vars it needs to force transcoding when
B leg doesn't match any A leg  codec ?
in vars.xml example I can see only global_codecs_prefs and outbound_codecs prefs correctly set

Thanks
  ----- OriginaleMessage ----- 
  From: Michael Jerris 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Saturday, February 20, 2010 8:48 PM
  Subject: Re: [Freeswitch-users] outbound callsh


  Sounds like the remote end does not like the codecs we are offering.  There are some vars to adjust this, by default we only offer the 1 codec chosen with the a leg of the call.


  Mike


  On Feb 20, 2010, at 1:36 PM, Madovsky wrote:


    Hello,

    I'm able to transcode a cal between 2 local legs,
    but when a local user call an oubound call,
    the call hangs up saying "not acceptable here",
    so it doesn't transcode.





------------------------------------------------------------------------------


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100220/ebd19340/attachment-0002.html 

From frank at carmickle.com  Sat Feb 20 18:00:38 2010
From: frank at carmickle.com (Frank Carmickle)
Date: Sat, 20 Feb 2010 21:00:38 -0500
Subject: [Freeswitch-users] outbound calls
In-Reply-To: <C9E83EA051BC49D8AB9425D5D1225C27@MOBILEE1705>
References: <C9E83EA051BC49D8AB9425D5D1225C27@MOBILEE1705>
Message-ID: <20100221013429.GA9832@base.carmickle.com>

On Sat, Feb 20, Madovsky wrote:
> Hello,
> 
> I'm able to transcode a cal between 2 local legs,
> but when a local user call an oubound call,
> the call hangs up saying "not acceptable here",
> so it doesn't transcode.
> 
> Any idea ?

What are your outbound_codec_prefs set to in your vars.xml?

--FC



From spiritonly at gmail.com  Sat Feb 20 18:12:52 2010
From: spiritonly at gmail.com (=?UTF-8?B?5p2o5rGf6aqF?=)
Date: Sun, 21 Feb 2010 10:12:52 +0800
Subject: [Freeswitch-users] How to exchange media when I developed new
	endpoint module?
In-Reply-To: <ebbc2d551002201327o7a7bad26g48cac48f610aed55@mail.gmail.com>
References: <93b0f8ce1002091914q581f12fvd1de01e2181f0f3a@mail.gmail.com>
	<ebbc2d551002091932u52dbf1b6jfecf2dfdb43b8a7c@mail.gmail.com>
	<4021C303-1AC3-48F2-A17B-63C02C1C54D4@freeswitch.org>
	<93b0f8ce1002191722h5782b787qe56a06dd0cb94e76@mail.gmail.com>
	<ebbc2d551002201327o7a7bad26g48cac48f610aed55@mail.gmail.com>
Message-ID: <93b0f8ce1002201812x24cf3853u920bdba87f938b43@mail.gmail.com>

Oh, My God! You are the developer of mod_khomp! I read your blog, checked
out mod_khomp from google code, and found it had not update for a long time.
So what is your roadmap about mod_khomp?

I share my snippet in FS pastebin, you can find it with '
http://pastebin.freeswitch.org/12192 '.

My MSN is spiritonly at live.cn.
My Gtalk is spiritonly at gmail.com.
Hope we can keep connected.


2010/2/21 Jo?o Mesquita <jmesquita at freeswitch.org>

> I developed the current implementation of mod_khomp. I wouldn't take it as
> an example for anything since there has been no activity there for the past
> 4 months. If you care to share a snippet of your code, maybe we can help
> better.
>
>
> JM
>
>
>
> On Fri, Feb 19, 2010 at 11:22 PM, ??? <spiritonly at gmail.com> wrote:
>
>> Do you know mod_khomp? You can found it in FS wiki. I am developing an
>> endpoint module like it.
>> So you can give me some advice to bridge two session?
>>
>>
>> On Wed, Feb 10, 2010 at 11:44 AM, Brian West <brian at freeswitch.org>wrote:
>>
>>> But the bigger question is what protocol are you doing that you have to
>>> create your own endpoint module?
>>>
>>> /b
>>>
>>> On Feb 9, 2010, at 9:32 PM, Jo?o Mesquita wrote:
>>>
>>> > You should look at read_frame and write_frame implementations of other
>>> endpoint modules.
>>> >
>>> > This should pretty much tell you how things work...
>>> >
>>> > Jo?o Mesquita
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100221/e9a0626b/attachment-0002.html 

From frank at carmickle.com  Sat Feb 20 18:15:35 2010
From: frank at carmickle.com (Frank Carmickle)
Date: Sat, 20 Feb 2010 21:15:35 -0500
Subject: [Freeswitch-users] outbound calls
In-Reply-To: <634B5DC983EC4267993346BB172A4FD1@MOBILEE1705>
References: <C9E83EA051BC49D8AB9425D5D1225C27@MOBILEE1705>
	<F0495877-E2B4-4830-AAA0-889D95CC2EEA@jerris.com>
	<634B5DC983EC4267993346BB172A4FD1@MOBILEE1705>
Message-ID: <20100221021535.GB9832@base.carmickle.com>

On Sat, Feb 20, Madovsky wrote:
> yes I know it is codec problem,
> but what vars it needs to force transcoding when
> B leg doesn't match any A leg  codec ?
> in vars.xml example I can see only global_codecs_prefs and outbound_codecs prefs correctly set

You know the codec you want to match on the B leg so in the dialplan

>From the wiki http://wiki.freeswitch.org/wiki/Codec_Negotiation
<action application="bridge" data="{absolute_codec_string='PCMA,PCMU'}sofia/gateway/mygateway/mynumber"/>
 
HTH
--FC



From infos at madovsky.org  Sat Feb 20 18:19:47 2010
From: infos at madovsky.org (Madovsky)
Date: Sat, 20 Feb 2010 21:19:47 -0500
Subject: [Freeswitch-users] outbound calls
References: <C9E83EA051BC49D8AB9425D5D1225C27@MOBILEE1705>
	<20100221013429.GA9832@base.carmickle.com>
Message-ID: <235403598AE24079969DAAE9A61E823B@MOBILEE1705>


----- Original Message ----- 
From: "Frank Carmickle" <frank at carmickle.com>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Saturday, February 20, 2010 9:00 PM
Subject: Re: [Freeswitch-users] outbound calls


> On Sat, Feb 20, Madovsky wrote:
>> Hello,
>>
>> I'm able to transcode a cal between 2 local legs,
>> but when a local user call an oubound call,
>> the call hangs up saying "not acceptable here",
>> so it doesn't transcode.
>>
>> Any idea ?
>
> What are your outbound_codec_prefs set to in your vars.xml?
>
> --FC
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

Hi Frank,

global
<X-PRE-PROCESS cmd="set" 
data="global_codec_prefs=speex at 16000k,PCMU,PCMA,GSM"/>

outbound

<X-PRE-PROCESS cmd="set" 
data="outbound_codec_prefs=speex at 16000k,PCMU,PCMA,GSM"/>


Thanks

Franck 




From brian at microcomaustralia.com.au  Sat Feb 20 18:21:34 2010
From: brian at microcomaustralia.com.au (Brian May)
Date: Sun, 21 Feb 2010 13:21:34 +1100
Subject: [Freeswitch-users] list all users
Message-ID: <3c5cf5261002201821q7b9f024bt7ab32b10946a5370@mail.gmail.com>

Hello,

How do I get a list of all users? Including users that are not registered yet?

(note: I have already changed the domain to microcomaustralia.com.au)

I have tried:

freeswitch at voyage> user_exists brian at microcomaustralia.com.au
false

i believe this user should be defined in
conf/directory/default/brian.xml, however it is like nothing in
conf/directory/default.xml is being read. I want to prove if this is
the case or not.

When I try to register I get this error:

2010-02-21 13:09:02.568081 [WARNING] sofia_reg.c:1019 SIP auth failure
(REGISTER) on sofia profile 'internal' for
[brian at microcomaustralia.com.au] from ip 192.168.87.14

which I think might be because it can't find the matching user entry.

Thanks
-- 
Brian May <brian at microcomaustralia.com.au>



From infos at madovsky.org  Sat Feb 20 18:36:41 2010
From: infos at madovsky.org (Madovsky)
Date: Sat, 20 Feb 2010 21:36:41 -0500
Subject: [Freeswitch-users] outbound calls
References: <C9E83EA051BC49D8AB9425D5D1225C27@MOBILEE1705><F0495877-E2B4-4830-AAA0-889D95CC2EEA@jerris.com><634B5DC983EC4267993346BB172A4FD1@MOBILEE1705>
	<20100221021535.GB9832@base.carmickle.com>
Message-ID: <849A848F7F7142639B883B5F8471B5AB@MOBILEE1705>


----- Original Message ----- 
From: "Frank Carmickle" <frank at carmickle.com>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Saturday, February 20, 2010 9:15 PM
Subject: Re: [Freeswitch-users] outbound calls


> On Sat, Feb 20, Madovsky wrote:
>> yes I know it is codec problem,
>> but what vars it needs to force transcoding when
>> B leg doesn't match any A leg  codec ?
>> in vars.xml example I can see only global_codecs_prefs and 
>> outbound_codecs prefs correctly set
>
> You know the codec you want to match on the B leg so in the dialplan
>
>>From the wiki http://wiki.freeswitch.org/wiki/Codec_Negotiation
> <action application="bridge" 
> data="{absolute_codec_string='PCMA,PCMU'}sofia/gateway/mygateway/mynumber"/>
>
> HTH
> --FC
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

No, in fact I don't know which codec will come from B leg,
but in my test I did the codec of B leg matches one codec in the outbound 
list but doesn't  transcode and
the call fails. but if A leg and B leg are local so it transcodes 
correctly...

Thanks 




From jason at jasonjgw.net  Sat Feb 20 18:37:41 2010
From: jason at jasonjgw.net (Jason White)
Date: Sun, 21 Feb 2010 13:37:41 +1100
Subject: [Freeswitch-users] list all users
In-Reply-To: <3c5cf5261002201821q7b9f024bt7ab32b10946a5370@mail.gmail.com>
References: <3c5cf5261002201821q7b9f024bt7ab32b10946a5370@mail.gmail.com>
Message-ID: <20100221023741.GA15005@jdc.jasonjgw.net>

Brian May <brian at microcomaustralia.com.au> wrote:
> (note: I have already changed the domain to microcomaustralia.com.au)
> 
> I have tried:
> 
> freeswitch at voyage> user_exists brian at microcomaustralia.com.au
> false
> 
> i believe this user should be defined in
> conf/directory/default/brian.xml, however it is like nothing in
> conf/directory/default.xml is being read. I want to prove if this is
> the case or not.

Have a look at /opt/freeswitch/log/freeswitch.xml.fsxml
This is the compilation of all the XML files, and it's the file which is
actually consulted to do the configuration lookup. If your user entry is in
there, then you know that FreeSWITCH has it.

Also, make sure the user-agent is configured to register to the domain, not to
an IP address.




From infos at madovsky.org  Sat Feb 20 18:40:57 2010
From: infos at madovsky.org (Madovsky)
Date: Sat, 20 Feb 2010 21:40:57 -0500
Subject: [Freeswitch-users] outbound calls
References: <C9E83EA051BC49D8AB9425D5D1225C27@MOBILEE1705><F0495877-E2B4-4830-AAA0-889D95CC2EEA@jerris.com><634B5DC983EC4267993346BB172A4FD1@MOBILEE1705>
	<20100221021535.GB9832@base.carmickle.com>
Message-ID: <9954CEB3764E4A11BEBE0DC14B2C968A@MOBILEE1705>


----- Original Message ----- 
From: "Frank Carmickle" <frank at carmickle.com>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Saturday, February 20, 2010 9:15 PM
Subject: Re: [Freeswitch-users] outbound calls


> On Sat, Feb 20, Madovsky wrote:
>> yes I know it is codec problem,
>> but what vars it needs to force transcoding when
>> B leg doesn't match any A leg  codec ?
>> in vars.xml example I can see only global_codecs_prefs and 
>> outbound_codecs prefs correctly set
>
> You know the codec you want to match on the B leg so in the dialplan
>
>>From the wiki http://wiki.freeswitch.org/wiki/Codec_Negotiation
> <action application="bridge" 
> data="{absolute_codec_string='PCMA,PCMU'}sofia/gateway/mygateway/mynumber"/>
>
> HTH
> --FC
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

I retried your suggestion (that I already did 3 days ago)
but no work, it's the same 




From mcampbellsmith at gmail.com  Sat Feb 20 19:08:45 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Sun, 21 Feb 2010 14:08:45 +1100
Subject: [Freeswitch-users] list all users
In-Reply-To: <3c5cf5261002201821q7b9f024bt7ab32b10946a5370@mail.gmail.com>
References: <3c5cf5261002201821q7b9f024bt7ab32b10946a5370@mail.gmail.com>
Message-ID: <33c87fa31002201908x55e43e62g32b340872df9863d@mail.gmail.com>

Try

user_exists id <domain>

Where <domain> is the output from the command 'eval ${domain}'

On Sun, Feb 21, 2010 at 1:21 PM, Brian May
<brian at microcomaustralia.com.au> wrote:
> Hello,
>
> How do I get a list of all users? Including users that are not registered yet?
>
> (note: I have already changed the domain to microcomaustralia.com.au)
>
> I have tried:
>
> freeswitch at voyage> user_exists brian at microcomaustralia.com.au
> false
>
> i believe this user should be defined in
> conf/directory/default/brian.xml, however it is like nothing in
> conf/directory/default.xml is being read. I want to prove if this is
> the case or not.
>
> When I try to register I get this error:
>
> 2010-02-21 13:09:02.568081 [WARNING] sofia_reg.c:1019 SIP auth failure
> (REGISTER) on sofia profile 'internal' for
> [brian at microcomaustralia.com.au] from ip 192.168.87.14
>
> which I think might be because it can't find the matching user entry.
>
> Thanks
> --
> Brian May <brian at microcomaustralia.com.au>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From frank at carmickle.com  Sat Feb 20 19:10:02 2010
From: frank at carmickle.com (Frank Carmickle)
Date: Sat, 20 Feb 2010 22:10:02 -0500
Subject: [Freeswitch-users] outbound calls
In-Reply-To: <9954CEB3764E4A11BEBE0DC14B2C968A@MOBILEE1705>
References: <20100221021535.GB9832@base.carmickle.com>
	<9954CEB3764E4A11BEBE0DC14B2C968A@MOBILEE1705>
Message-ID: <20100221031001.GC9832@base.carmickle.com>

On Sat, Feb 20, Madovsky wrote:
> 
> ----- Original Message ----- 
> From: "Frank Carmickle" <frank at carmickle.com>
> To: <freeswitch-users at lists.freeswitch.org>
> Sent: Saturday, February 20, 2010 9:15 PM
> Subject: Re: [Freeswitch-users] outbound calls
> 
> 
> > On Sat, Feb 20, Madovsky wrote:
> >> yes I know it is codec problem,
> >> but what vars it needs to force transcoding when
> >> B leg doesn't match any A leg  codec ?
> >> in vars.xml example I can see only global_codecs_prefs and 
> >> outbound_codecs prefs correctly set
> >
> > You know the codec you want to match on the B leg so in the dialplan
> >
> >>From the wiki http://wiki.freeswitch.org/wiki/Codec_Negotiation
> > <action application="bridge" 
> > data="{absolute_codec_string='PCMA,PCMU'}sofia/gateway/mygateway/mynumber"/>
> >
> > HTH
> > --FC
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 
> I retried your suggestion (that I already did 3 days ago)
> but no work, it's the same 

Is the external phone using the external profile?  Do the different profiles have differing codec settings?

--FC



From infos at madovsky.org  Sat Feb 20 19:17:56 2010
From: infos at madovsky.org (Madovsky)
Date: Sat, 20 Feb 2010 22:17:56 -0500
Subject: [Freeswitch-users] outbound calls
References: <C9E83EA051BC49D8AB9425D5D1225C27@MOBILEE1705><F0495877-E2B4-4830-AAA0-889D95CC2EEA@jerris.com><634B5DC983EC4267993346BB172A4FD1@MOBILEE1705>
	<20100221021535.GB9832@base.carmickle.com>
Message-ID: <793742B9BB1D4EB3990A6A7DA72A1A4B@MOBILEE1705>


----- Original Message ----- 
From: "Frank Carmickle" <frank at carmickle.com>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Saturday, February 20, 2010 9:15 PM
Subject: Re: [Freeswitch-users] outbound calls


> On Sat, Feb 20, Madovsky wrote:
>> yes I know it is codec problem,
>> but what vars it needs to force transcoding when
>> B leg doesn't match any A leg  codec ?
>> in vars.xml example I can see only global_codecs_prefs and 
>> outbound_codecs prefs correctly set
>
> You know the codec you want to match on the B leg so in the dialplan
>
>>From the wiki http://wiki.freeswitch.org/wiki/Codec_Negotiation
> <action application="bridge" 
> data="{absolute_codec_string='PCMA,PCMU'}sofia/gateway/mygateway/mynumber"/>
>
> HTH
> --FC
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

Ok now I can call from an external sip account to a local FS sip account and 
it
transcodes correctly (I made the same mistake as hundred users I imagine 
like I forgot that the exterrnal port is 5080, so I changed to 5060). but if 
I do the contrary (internal A leg to ext B leg) it doesn't .
is outbound-late-negociation exists ?




From infos at madovsky.org  Sat Feb 20 19:22:32 2010
From: infos at madovsky.org (Madovsky)
Date: Sat, 20 Feb 2010 22:22:32 -0500
Subject: [Freeswitch-users] outbound calls
References: <20100221021535.GB9832@base.carmickle.com><9954CEB3764E4A11BEBE0DC14B2C968A@MOBILEE1705>
	<20100221031001.GC9832@base.carmickle.com>
Message-ID: <74283D2E3F924DC5800BAFBAE2BF7F57@MOBILEE1705>


----- Original Message ----- 
From: "Frank Carmickle" <frank at carmickle.com>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Saturday, February 20, 2010 10:10 PM
Subject: Re: [Freeswitch-users] outbound calls


> On Sat, Feb 20, Madovsky wrote:
>>
>> ----- Original Message ----- 
>> From: "Frank Carmickle" <frank at carmickle.com>
>> To: <freeswitch-users at lists.freeswitch.org>
>> Sent: Saturday, February 20, 2010 9:15 PM
>> Subject: Re: [Freeswitch-users] outbound calls
>>
>>
>> > On Sat, Feb 20, Madovsky wrote:
>> >> yes I know it is codec problem,
>> >> but what vars it needs to force transcoding when
>> >> B leg doesn't match any A leg  codec ?
>> >> in vars.xml example I can see only global_codecs_prefs and
>> >> outbound_codecs prefs correctly set
>> >
>> > You know the codec you want to match on the B leg so in the dialplan
>> >
>> >>From the wiki http://wiki.freeswitch.org/wiki/Codec_Negotiation
>> > <action application="bridge"
>> > data="{absolute_codec_string='PCMA,PCMU'}sofia/gateway/mygateway/mynumber"/>
>> >
>> > HTH
>> > --FC
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>> I retried your suggestion (that I already did 3 days ago)
>> but no work, it's the same
>
> Is the external phone using the external profile?  Do the different 
> profiles have differing codec settings?
>
> --FC
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


FIrst it was the same vars, now I succeed to transcode from a call from Bleg 
to Aleg in listing
the only Aleg codec available ...





From brian at microcomaustralia.com.au  Sat Feb 20 20:00:15 2010
From: brian at microcomaustralia.com.au (Brian May)
Date: Sun, 21 Feb 2010 15:00:15 +1100
Subject: [Freeswitch-users] list all users
In-Reply-To: <33c87fa31002201908x55e43e62g32b340872df9863d@mail.gmail.com>
References: <3c5cf5261002201821q7b9f024bt7ab32b10946a5370@mail.gmail.com>
	<33c87fa31002201908x55e43e62g32b340872df9863d@mail.gmail.com>
Message-ID: <3c5cf5261002202000h12b9d6b2ybefd5422fd4f9ee5@mail.gmail.com>

Ok, I solved one problem. Yes, the user really was there, the problem
was I followed the wiki documentation for generating a password hash:

echo "username:domain:password" | openssl dgst -md5

This won't work because echo will append a new line; the correct version is:

echo -n "username:domain:password" | openssl dgst -md5

I have updated the wiki. Now the client will register, and can make
outgoing calls.

Incoming calls don't work however, I get the messages:

2010-02-21 14:45:51.400081 [ERR] switch_ivr_originate.c:2387 Cannot
create outgoing channel of type [error] cause: [USER_NOT_REGISTERED]
2010-02-21 14:45:51.410081 [ERR] switch_ivr_originate.c:2387 Cannot
create outgoing channel of type [user] cause: [USER_NOT_REGISTERED]

As per one suggestion (as far as I can tell this shouldn't be
required) I tried changing this (in dialplan/default.xml):

<action application="bridge" data="user/${dialed_extension}@${domain_name}"/>

with:

<action application="bridge"
data="sofia/internal/${dialed_extension}@${domain_name}"/>

However that isn't using the registered IP address for the
brian at microcomaustralia.com.au user; rather it does a DNS lookup for
$(domain_name) and tries to contact that address instead, this is
wrong; that
DNS address resolves to my webserver, not the jabber client.


On 21 February 2010 14:08, Mark Campbell-Smith <mcampbellsmith at gmail.com> wrote:
> Try
>
> user_exists id <domain>
>
> Where <domain> is the output from the command 'eval ${domain}'

Curiously that still doesn't work, even though I registered:

freeswitch at voyage> eval ${domain}

microcomaustralia.com.au
freeswitch at voyage>
freeswitch at voyage> user_exists id brian at microcomaustralia.com.au

false


freeswitch at voyage> sofia status profile microcomaustralia.com.au

=================================================================================================
Name             	microcomaustralia.com.au
Domain Name      	N/A
Alias Of         	internal
Auto-NAT         	false
DBName           	sofia_reg_internal
Pres Hosts       	
Dialplan         	XML
Context          	public
Challenge Realm  	auto_from
RTP-IP           	192.168.86.4
SIP-IP           	192.168.86.4
URL              	sip:mod_sofia at 192.168.86.4:5060
BIND-URL         	sip:mod_sofia at 192.168.86.4:5060
HOLD-MUSIC       	local_stream://moh
OUTBOUND-PROXY   	N/A
CODECS IN        	G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM
CODECS OUT       	G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM
TEL-EVENT        	101
DTMF-MODE        	rfc2833
CNG              	13
SESSION-TO       	0
MAX-DIALOG       	0
NOMEDIA          	false
LATE-NEG         	false
PROXY-MEDIA      	false
AGGRESSIVENAT    	false
STUN-ENABLED     	true
STUN-AUTO-DISABLE	false
CALLS-IN         	0
FAILED-CALLS-IN  	0
CALLS-OUT        	4
FAILED-CALLS-OUT 	4

Registrations:
=================================================================================================
Call-ID:    	advktkbqoyvqmwm at andean.pri
User:       	brian at microcomaustralia.com.au
Contact:    	"Brian May"
<sip:brian at 192.168.87.14;fs_nat=yes;fs_path=sip%3Abrian%40192.168.87.14%3A5060>
Agent:      	Twinkle/1.4.2
Status:     	Registered(UDP-NAT)(unknown) EXP(2010-02-21 16:07:39)
Host:       	voyage
IP:         	192.168.87.14
Port:       	5060
Auth-User:  	brian
Auth-Realm: 	microcomaustralia.com.au
MWI-Account:	brian at microcomaustralia.com.au

=================================================================================================

-- 
Brian May <brian at microcomaustralia.com.au>



From jason at jasonjgw.net  Sat Feb 20 20:15:59 2010
From: jason at jasonjgw.net (Jason White)
Date: Sun, 21 Feb 2010 15:15:59 +1100
Subject: [Freeswitch-users] list all users
In-Reply-To: <3c5cf5261002202000h12b9d6b2ybefd5422fd4f9ee5@mail.gmail.com>
References: <3c5cf5261002201821q7b9f024bt7ab32b10946a5370@mail.gmail.com>
	<33c87fa31002201908x55e43e62g32b340872df9863d@mail.gmail.com>
	<3c5cf5261002202000h12b9d6b2ybefd5422fd4f9ee5@mail.gmail.com>
Message-ID: <20100221041559.GA21387@jdc.jasonjgw.net>

Brian May <brian at microcomaustralia.com.au> wrote:
 
> Incoming calls don't work however, I get the messages:
> 
> 2010-02-21 14:45:51.400081 [ERR] switch_ivr_originate.c:2387 Cannot
> create outgoing channel of type [error] cause: [USER_NOT_REGISTERED]
> 2010-02-21 14:45:51.410081 [ERR] switch_ivr_originate.c:2387 Cannot
> create outgoing channel of type [user] cause: [USER_NOT_REGISTERED]

what does sofia_contact username at microcomaustralia.com.au
show?

Maybe turning on debug logging at this point would help.
/log debug




From brian at microcomaustralia.com.au  Sat Feb 20 20:30:03 2010
From: brian at microcomaustralia.com.au (Brian May)
Date: Sun, 21 Feb 2010 15:30:03 +1100
Subject: [Freeswitch-users] list all users
In-Reply-To: <20100221041559.GA21387@jdc.jasonjgw.net>
References: <3c5cf5261002201821q7b9f024bt7ab32b10946a5370@mail.gmail.com>
	<33c87fa31002201908x55e43e62g32b340872df9863d@mail.gmail.com>
	<3c5cf5261002202000h12b9d6b2ybefd5422fd4f9ee5@mail.gmail.com>
	<20100221041559.GA21387@jdc.jasonjgw.net>
Message-ID: <3c5cf5261002202030l74935bbr1013547a315aaf6a@mail.gmail.com>

On 21 February 2010 15:15, Jason White <jason at jasonjgw.net> wrote:
> what does sofia_contact username at microcomaustralia.com.au
> show?

freeswitch at voyage> sofia_contact brian at microcomaustralia.com.au

sofia/internal/sip:brian at 192.168.87.14;fs_nat=yes;fs_path=sip%3Abrian%40192.168.87.14%3A5060


looks good to me...


> Maybe turning on debug logging at this point would help.
> /log debug


freeswitch at voyage> /log debug
Unknown Command: /log debug


Only debug I have found is:

freeswitch at voyage> sofia loglevel all 9

Sofia log level for component [all] has been set to [9]


however that doesn't display any extra information for this test case.
-- 
Brian May <brian at microcomaustralia.com.au>



From jason at jasonjgw.net  Sat Feb 20 21:40:30 2010
From: jason at jasonjgw.net (Jason White)
Date: Sun, 21 Feb 2010 16:40:30 +1100
Subject: [Freeswitch-users] list all users
In-Reply-To: <3c5cf5261002202030l74935bbr1013547a315aaf6a@mail.gmail.com>
References: <3c5cf5261002201821q7b9f024bt7ab32b10946a5370@mail.gmail.com>
	<33c87fa31002201908x55e43e62g32b340872df9863d@mail.gmail.com>
	<3c5cf5261002202000h12b9d6b2ybefd5422fd4f9ee5@mail.gmail.com>
	<20100221041559.GA21387@jdc.jasonjgw.net>
	<3c5cf5261002202030l74935bbr1013547a315aaf6a@mail.gmail.com>
Message-ID: <20100221054030.GA25531@jdc.jasonjgw.net>

Brian May <brian at microcomaustralia.com.au> wrote:
 
> freeswitch at voyage> sofia_contact brian at microcomaustralia.com.au
> 
> sofia/internal/sip:brian at 192.168.87.14;fs_nat=yes;fs_path=sip%3Abrian%40192.168.87.14%3A5060
> 
> 
> looks good to me...

It does.
> 
> 
> > Maybe turning on debug logging at this point would help.
> > /log debug
> 
> 
> freeswitch at voyage> /log debug
> Unknown Command: /log debug

I was assuming you were running fs_cli with FreeSWITCH in daemon mode, as is
usual. At the console it's a different command to enable debugging, but I
never run FreeSWITCH that way unless I'm trying to deal with a segfault or
other startup issue.




From brian at microcomaustralia.com.au  Sat Feb 20 21:46:41 2010
From: brian at microcomaustralia.com.au (Brian May)
Date: Sun, 21 Feb 2010 16:46:41 +1100
Subject: [Freeswitch-users] list all users
In-Reply-To: <3c5cf5261002202030l74935bbr1013547a315aaf6a@mail.gmail.com>
References: <3c5cf5261002201821q7b9f024bt7ab32b10946a5370@mail.gmail.com>
	<33c87fa31002201908x55e43e62g32b340872df9863d@mail.gmail.com>
	<3c5cf5261002202000h12b9d6b2ybefd5422fd4f9ee5@mail.gmail.com>
	<20100221041559.GA21387@jdc.jasonjgw.net>
	<3c5cf5261002202030l74935bbr1013547a315aaf6a@mail.gmail.com>
Message-ID: <3c5cf5261002202146sd1dc48drb274951b4be687c8@mail.gmail.com>

On 21 February 2010 15:30, Brian May <brian at microcomaustralia.com.au> wrote:
> freeswitch at voyage> sofia_contact brian at microcomaustralia.com.au
>
> sofia/internal/sip:brian at 192.168.87.14;fs_nat=yes;fs_path=sip%3Abrian%40192.168.87.14%3A5060

Just managed to disable NAT. Just in case. No NAT in use here.

freeswitch at voyage> sofia_contact brian at microcomaustralia.com.au
sofia/internal/sip:brian at 192.168.87.14


I know it is locating my user record, because the voice message I get
is "The person at the extension b-r-i-a-n is not available" when I
dialled the extension of 1000.

However, for some reason it doesn't seem to locate the registration record:

=== cut ===
freeswitch at voyage> sofia status profile internal reg brian


Registrations:
=================================================================================================
Call-ID:    	advktkbqoyvqmwm at andean.pri
User:       	brian at microcomaustralia.com.au
Contact:    	"Brian May" <sip:brian at 192.168.87.14>
Agent:      	Twinkle/1.4.2
Status:     	Registered(UDP)(unknown) EXP(2010-02-21 18:38:40)
Host:       	voyage
IP:         	192.168.87.14
Port:       	5060
Auth-User:  	brian
Auth-Realm: 	microcomaustralia.com.au
MWI-Account:	brian at microcomaustralia.com.au

=================================================================================================

freeswitch at voyage> sofia status profile internal reg
brian at microcomaustralia.com.au


Registrations:
=================================================================================================
=================================================================================================

=== cut ===


Ok then, afraid I can't explain this behaviour. Am sure 2nd command
was working recently...
-- 
Brian May <brian at microcomaustralia.com.au>



From brian at microcomaustralia.com.au  Sat Feb 20 21:56:25 2010
From: brian at microcomaustralia.com.au (Brian May)
Date: Sun, 21 Feb 2010 16:56:25 +1100
Subject: [Freeswitch-users] list all users
In-Reply-To: <20100221054030.GA25531@jdc.jasonjgw.net>
References: <3c5cf5261002201821q7b9f024bt7ab32b10946a5370@mail.gmail.com>
	<33c87fa31002201908x55e43e62g32b340872df9863d@mail.gmail.com>
	<3c5cf5261002202000h12b9d6b2ybefd5422fd4f9ee5@mail.gmail.com>
	<20100221041559.GA21387@jdc.jasonjgw.net>
	<3c5cf5261002202030l74935bbr1013547a315aaf6a@mail.gmail.com>
	<20100221054030.GA25531@jdc.jasonjgw.net>
Message-ID: <3c5cf5261002202156u2f2e455jbf1e864c738b77a3@mail.gmail.com>

On 21 February 2010 16:40, Jason White <jason at jasonjgw.net> wrote:
> I was assuming you were running fs_cli with FreeSWITCH in daemon mode, as is
> usual. At the console it's a different command to enable debugging, but I
> never run FreeSWITCH that way unless I'm trying to deal with a segfault or
> other startup issue.

Yes, I should do that too. Seems to confirm what I already know,
nothing new though :-(

EXECUTE OpenZAP/1:1/1000 bridge(user/1000 at microcomaustralia.com.au)
2010-02-21 16:59:56.790081 [DEBUG] switch_ivr_originate.c:1859
variable string 0 = [presence_id=1000 at microcomaustralia.com.au]
2010-02-21 16:59:56.790081 [ERR] switch_ivr_originate.c:2387 Cannot
create outgoing channel of type [error] cause: [USER_NOT_REGISTERED]
2010-02-21 16:59:56.790081 [DEBUG] switch_ivr_originate.c:3185
Originate Resulted in Error Cause: 606 [USER_NOT_REGISTERED]
2010-02-21 16:59:56.800081 [ERR] switch_ivr_originate.c:2387 Cannot
create outgoing channel of type [user] cause: [USER_NOT_REGISTERED]
2010-02-21 16:59:56.800081 [DEBUG] switch_ivr_originate.c:3185
Originate Resulted in Error Cause: 606 [USER_NOT_REGISTERED]
2010-02-21 16:59:56.800081 [INFO] mod_dptools.c:2353 Originate Failed.
 Cause: USER_NOT_REGISTERED

-- 
Brian May <brian at microcomaustralia.com.au>



From brian at microcomaustralia.com.au  Sat Feb 20 22:28:54 2010
From: brian at microcomaustralia.com.au (Brian May)
Date: Sun, 21 Feb 2010 17:28:54 +1100
Subject: [Freeswitch-users] list all users
In-Reply-To: <3c5cf5261002202156u2f2e455jbf1e864c738b77a3@mail.gmail.com>
References: <3c5cf5261002201821q7b9f024bt7ab32b10946a5370@mail.gmail.com>
	<33c87fa31002201908x55e43e62g32b340872df9863d@mail.gmail.com>
	<3c5cf5261002202000h12b9d6b2ybefd5422fd4f9ee5@mail.gmail.com>
	<20100221041559.GA21387@jdc.jasonjgw.net>
	<3c5cf5261002202030l74935bbr1013547a315aaf6a@mail.gmail.com>
	<20100221054030.GA25531@jdc.jasonjgw.net>
	<3c5cf5261002202156u2f2e455jbf1e864c738b77a3@mail.gmail.com>
Message-ID: <3c5cf5261002202228n7605936uf517fc608b38d2a2@mail.gmail.com>

Ok, not sure what best practise is here...

Previously, I had defined the user as such:

<user id="brian" number-alias="1000">

This seems to work a lot better if I just register it as the number:

<user id="1000">
-- 
Brian May <brian at microcomaustralia.com.au>



From jason at jasonjgw.net  Sat Feb 20 22:45:20 2010
From: jason at jasonjgw.net (Jason White)
Date: Sun, 21 Feb 2010 17:45:20 +1100
Subject: [Freeswitch-users] list all users
In-Reply-To: <3c5cf5261002202156u2f2e455jbf1e864c738b77a3@mail.gmail.com>
References: <3c5cf5261002201821q7b9f024bt7ab32b10946a5370@mail.gmail.com>
	<33c87fa31002201908x55e43e62g32b340872df9863d@mail.gmail.com>
	<3c5cf5261002202000h12b9d6b2ybefd5422fd4f9ee5@mail.gmail.com>
	<20100221041559.GA21387@jdc.jasonjgw.net>
	<3c5cf5261002202030l74935bbr1013547a315aaf6a@mail.gmail.com>
	<20100221054030.GA25531@jdc.jasonjgw.net>
	<3c5cf5261002202156u2f2e455jbf1e864c738b77a3@mail.gmail.com>
Message-ID: <20100221064520.GA26593@jdc.jasonjgw.net>

Brian May <brian at microcomaustralia.com.au> wrote:
 
> EXECUTE OpenZAP/1:1/1000 bridge(user/1000 at microcomaustralia.com.au)
> 2010-02-21 16:59:56.790081 [DEBUG] switch_ivr_originate.c:1859
> variable string 0 = [presence_id=1000 at microcomaustralia.com.au]
> 2010-02-21 16:59:56.790081 [ERR] switch_ivr_originate.c:2387 Cannot
> create outgoing channel of type [error] cause: [USER_NOT_REGISTERED]

I think it's looking for 1000 rather than the user name, which of course isn't
registered.

In the extension for the user in your dial-plan, you could always write:
<action application="bridge"
data="$sofia_contact(brian at microcomaustralia.com.au)"/>
or you could use a variable instead of writing the domain name explicitly, of
course, which would be better practice.




From jason at jasonjgw.net  Sat Feb 20 22:51:38 2010
From: jason at jasonjgw.net (Jason White)
Date: Sun, 21 Feb 2010 17:51:38 +1100
Subject: [Freeswitch-users] list all users
In-Reply-To: <20100221064520.GA26593@jdc.jasonjgw.net>
References: <3c5cf5261002201821q7b9f024bt7ab32b10946a5370@mail.gmail.com>
	<33c87fa31002201908x55e43e62g32b340872df9863d@mail.gmail.com>
	<3c5cf5261002202000h12b9d6b2ybefd5422fd4f9ee5@mail.gmail.com>
	<20100221041559.GA21387@jdc.jasonjgw.net>
	<3c5cf5261002202030l74935bbr1013547a315aaf6a@mail.gmail.com>
	<20100221054030.GA25531@jdc.jasonjgw.net>
	<3c5cf5261002202156u2f2e455jbf1e864c738b77a3@mail.gmail.com>
	<20100221064520.GA26593@jdc.jasonjgw.net>
Message-ID: <20100221065138.GA26756@jdc.jasonjgw.net>

Jason White <jason at jasonjgw.net> wrote:
 
> In the extension for the user in your dial-plan, you could always write:
> <action application="bridge"
> data="$sofia_contact(brian at microcomaustralia.com.au)"/>
> or you could use a variable instead of writing the domain name explicitly, of
> course, which would be better practice.

Sorry - there should have been braces in the above:
${sofia_contact(user at domain)}
it's the same function you called from the console in response to my earlier
post, but this time executed from the dial-plan to perform the lookup.




From brian at microcomaustralia.com.au  Sun Feb 21 00:30:39 2010
From: brian at microcomaustralia.com.au (Brian May)
Date: Sun, 21 Feb 2010 19:30:39 +1100
Subject: [Freeswitch-users] outgoing dialplans for australia
Message-ID: <3c5cf5261002210030n12bd743fh69f0c862eae0db8@mail.gmail.com>

Hello again!

Ok, after spending several hours trying to debug why incoming calls
didn't work, for two separate issues (2nd time I accidentally set "Do
not disturb" on my VOIP client. Duh!), I am feeling slightly lazy...

Anyone got a set of rules for outgoing calls (Australia) that they are
willing to share?

Even if I don't use it directly, it may help me understand this new
XML based syntax (I am only really familiar with the Asterisk dialplan
syntax).

As a specific example of something I am unsure of, is it possible to
have it try and dial using a sip provider, and if that fails try the
zap port? I don't want it to fall back to the zap port though if the
person doesn't answer or is engaged, only if there is a problem with
the SIP connection (e.g. Internet connection down).

Thanks.
-- 
Brian May <brian at microcomaustralia.com.au>



From brian at microcomaustralia.com.au  Sun Feb 21 00:50:48 2010
From: brian at microcomaustralia.com.au (Brian May)
Date: Sun, 21 Feb 2010 19:50:48 +1100
Subject: [Freeswitch-users] groups
Message-ID: <3c5cf5261002210050s235142d6o3f8be42146399c57@mail.gmail.com>

Hello,

Can I add a analogue FXS port to a group that is then accessed using
group_call(...)?

If so, how? I can only see support for SIP members, as defined in
directory/default.xml

Thanks
-- 
Brian May <brian at microcomaustralia.com.au>



From jason at jasonjgw.net  Sun Feb 21 01:10:11 2010
From: jason at jasonjgw.net (Jason White)
Date: Sun, 21 Feb 2010 20:10:11 +1100
Subject: [Freeswitch-users] outgoing dialplans for australia
In-Reply-To: <3c5cf5261002210030n12bd743fh69f0c862eae0db8@mail.gmail.com>
References: <3c5cf5261002210030n12bd743fh69f0c862eae0db8@mail.gmail.com>
Message-ID: <20100221091011.GA28429@jdc.jasonjgw.net>

Brian May <brian at microcomaustralia.com.au> wrote:
 
> As a specific example of something I am unsure of, is it possible to
> have it try and dial using a sip provider, and if that fails try the
> zap port? I don't want it to fall back to the zap port though if the
> person doesn't answer or is engaged, only if there is a problem with
> the SIP connection (e.g. Internet connection down).

http://wiki.freeswitch.org/wiki/Extension_Status_Example
is similar to what you want, except that instead of invoking a Javascript
application if the call fails (as in the example), you'll need to test the
value of originate_disposition to decide how to handle the call if the bridge
is unsuccessful and it falls through to the next action in the dial-plan.

(I'm assuming based on that page that originate_disposition is the correct
variable in this kind of scenario.)

Someone (perhaps you after you've implemented and debugged it) should document
the result on the wiki, as I'm sure this is a common use case, but there is no
documentation currently other than the above-mentioned page.





From Russell.Mosemann at cune.org  Sun Feb 21 03:55:28 2010
From: Russell.Mosemann at cune.org (Russell Mosemann)
Date: Sun, 21 Feb 2010 05:55:28 -0600
Subject: [Freeswitch-users] openzap TDM400 card
In-Reply-To: <3c5cf5261002201408v63c150d3v419493ab5e4e2089@mail.gmail.com>
References: <3c5cf5261002192157t25de5798xc10ab2a7755e67d0@mail.gmail.com><3c5cf5261002200148k1bb3a1f7mf97fabccfbd1c178@mail.gmail.com><C8B78F95DB344ADDA010C52AE0AC8B66@cune.pri>
	<3c5cf5261002201408v63c150d3v419493ab5e4e2089@mail.gmail.com>
Message-ID: <5C49F61510CA4156A574A1CCFA743B94@cune.pri>

Brian May wrote:

> Ports 1 & 2 are FXS, but use FXO signalling. So, yes I do plug a phone
> into these.
> Ports 3 & 4 are FXO, but use FXS signalling. So I plug the phone line
> into these.

Maybe a better way to think about it is that Zaptel is looking at the remote end, because it is interested in what signals might be received. FS is looking at the local end, because that is what FS has to manage. If Zaptel is expecting the remote end to be FXO (office phone), then FS is managing an FXS (subscriber signaling).

--
Russell Mosemann




From technical at ttnc.co.uk  Sun Feb 21 05:15:53 2010
From: technical at ttnc.co.uk (TTNC - Technical)
Date: Sun, 21 Feb 2010 13:15:53 +0000
Subject: [Freeswitch-users] mod_fax undefined symbol
In-Reply-To: <E6568E57-A3E2-4650-A8CD-70B9C2846AF4@jerris.com>
References: <6FCE44FF-70C1-412D-B722-4C8D359381BA@ttnc.co.uk>
	<201002191825.50957.errotan@gmail.com>
	<AB3AD9C8-C162-4DB5-8095-97CA83EDDE57@ttnc.co.uk>
	<201002191904.39081.errotan@gmail.com>
	<E6568E57-A3E2-4650-A8CD-70B9C2846AF4@jerris.com>
Message-ID: <EFB211AD-9B65-4CCE-9C5E-BC0A9ED55B7D@ttnc.co.uk>

Hi Guys

Without meaning to hassle (I know I am), has there been any progress with this bug yet? Anything else I can do to assist?

Please let me know.

Thanks

Russ

On 20 Feb 2010, at 02:00, Michael Jerris wrote:

> replying with more details on jira.
> 
> 
> On Feb 19, 2010, at 1:04 PM, Pusk?s Zsolt wrote:
> 
>> 2010. febru?r 19. 18.44.32 TTNC - Technical d?tummal ezt ?rta:
>>> On 19 Feb 2010, at 17:25, Pusk?s Zsolt wrote:
>>>> Just compiled svn16700 on Debian "lenny" x86, mod_fax loads and works
>>>> perfectly. I have an ongoing compile on another machine (amd64) if It
>>>> don't works i will send a mail (in 1 hour) otherwise consider it working.
>>> 
>>> How did you compile it? Using dpkg-buildpackage or via make/make install?
>>> 
>>> Do you have any debian versions of libtiff4(-dev) installed?
>>> 
>> 
>> Okay i can confirm that mod_fax works on Debian "lenny" amd64 but it don't work 
>> on Debian "testing,squeeze" amd64.
>> 
>> 2010-02-19 18:49:14.610297 [CRIT] switch_loadable_module.c:882 Error Loading 
>> module /usr/local/freeswitch/mod/mod_fax.so
>> **/usr/local/freeswitch/mod/mod_fax.so: undefined symbol: 
>> TIFFDefaultStripSize**
>> 
>> I haven't tried to compile mod_fax on testing before so i don't know what is 
>> causeing the problem :(
>> 
>> # ldd mod_fax.so
>>       linux-vdso.so.1 =>  (0x00007fff106f6000)
>>       libm.so.6 => /lib/libm.so.6 (0x00007f506b711000)
>>       libfreeswitch.so.1 => /usr/local/freeswitch/lib/libfreeswitch.so.1 
>> (0x00007f506b345000)
>>       libpthread.so.0 => /lib/libpthread.so.0 (0x00007f506b128000)
>>       libc.so.6 => /lib/libc.so.6 (0x00007f506add4000)
>>       libssl.so.0.9.8 => /usr/lib/libssl.so.0.9.8 (0x00007f506ab82000)
>>       libcrypto.so.0.9.8 => /usr/lib/libcrypto.so.0.9.8 (0x00007f506a7e2000)
>>       libncurses.so.5 => /lib/libncurses.so.5 (0x00007f506a59d000)
>>       libstdc++.so.6 => /usr/lib/libstdc++.so.6 (0x00007f506a28d000)
>>       libgcc_s.so.1 => /lib/libgcc_s.so.1 (0x00007f506a076000)
>>       libodbc.so.1 => /usr/lib/libodbc.so.1 (0x00007f5069e17000)
>>       /lib64/ld-linux-x86-64.so.2 (0x00007f506bc31000)
>>       libdl.so.2 => /lib/libdl.so.2 (0x00007f5069c13000)
>>       libz.so.1 => /usr/lib/libz.so.1 (0x00007f50699fb000)
>>       libltdl.so.7 => /usr/lib/libltdl.so.7 (0x00007f50697f2000)
>> 
>> Recently in debian "testing" libtiff4 and libjpeg is upgraded:
>> libtiff 3.9.2-3+b1
>> libjpeg62 6b-16.1
>> libjeg8 8-2.1
>> 
>> Q&A:
>> Q: How did you compile it? Using dpkg-buildpackage or via make/make install?
>> A: svn-clean ./bootsrap ./configure make etc.
>> 
>> Q: Do you have any debian versions of libtiff4(-dev) installed?
>> A: Yes:3.8.2-11.2
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From steveu at coppice.org  Sun Feb 21 05:47:17 2010
From: steveu at coppice.org (Steve Underwood)
Date: Sun, 21 Feb 2010 21:47:17 +0800
Subject: [Freeswitch-users] mod_fax undefined symbol
In-Reply-To: <EFB211AD-9B65-4CCE-9C5E-BC0A9ED55B7D@ttnc.co.uk>
References: <6FCE44FF-70C1-412D-B722-4C8D359381BA@ttnc.co.uk>	<201002191825.50957.errotan@gmail.com>	<AB3AD9C8-C162-4DB5-8095-97CA83EDDE57@ttnc.co.uk>	<201002191904.39081.errotan@gmail.com>	<E6568E57-A3E2-4650-A8CD-70B9C2846AF4@jerris.com>
	<EFB211AD-9B65-4CCE-9C5E-BC0A9ED55B7D@ttnc.co.uk>
Message-ID: <4B813965.1030704@coppice.org>

Hi Russ,

The only place in FS where TIFFDefaulyStripSize is used is in the file 
t4_rx.c, and you probably won't actually be calling it. Try commenting 
out that line, and see if there are any other stumbling blocks. Often 
there is a mass of errors, and the system just tells you about them one 
by one.

Steve

On 02/21/2010 09:15 PM, TTNC - Technical wrote:
> Hi Guys
>
> Without meaning to hassle (I know I am), has there been any progress with this bug yet? Anything else I can do to assist?
>
> Please let me know.
>
> Thanks
>
> Russ
>
> On 20 Feb 2010, at 02:00, Michael Jerris wrote:
>
>    
>> replying with more details on jira.
>>
>>
>> On Feb 19, 2010, at 1:04 PM, Pusk?s Zsolt wrote:
>>
>>      
>>> 2010. febru?r 19. 18.44.32 TTNC - Technical d?tummal ezt ?rta:
>>>        
>>>> On 19 Feb 2010, at 17:25, Pusk?s Zsolt wrote:
>>>>          
>>>>> Just compiled svn16700 on Debian "lenny" x86, mod_fax loads and works
>>>>> perfectly. I have an ongoing compile on another machine (amd64) if It
>>>>> don't works i will send a mail (in 1 hour) otherwise consider it working.
>>>>>            
>>>> How did you compile it? Using dpkg-buildpackage or via make/make install?
>>>>
>>>> Do you have any debian versions of libtiff4(-dev) installed?
>>>>
>>>>          
>>> Okay i can confirm that mod_fax works on Debian "lenny" amd64 but it don't work
>>> on Debian "testing,squeeze" amd64.
>>>
>>> 2010-02-19 18:49:14.610297 [CRIT] switch_loadable_module.c:882 Error Loading
>>> module /usr/local/freeswitch/mod/mod_fax.so
>>> **/usr/local/freeswitch/mod/mod_fax.so: undefined symbol:
>>> TIFFDefaultStripSize**
>>>
>>> I haven't tried to compile mod_fax on testing before so i don't know what is
>>> causeing the problem :(
>>>
>>> # ldd mod_fax.so
>>>        linux-vdso.so.1 =>   (0x00007fff106f6000)
>>>        libm.so.6 =>  /lib/libm.so.6 (0x00007f506b711000)
>>>        libfreeswitch.so.1 =>  /usr/local/freeswitch/lib/libfreeswitch.so.1
>>> (0x00007f506b345000)
>>>        libpthread.so.0 =>  /lib/libpthread.so.0 (0x00007f506b128000)
>>>        libc.so.6 =>  /lib/libc.so.6 (0x00007f506add4000)
>>>        libssl.so.0.9.8 =>  /usr/lib/libssl.so.0.9.8 (0x00007f506ab82000)
>>>        libcrypto.so.0.9.8 =>  /usr/lib/libcrypto.so.0.9.8 (0x00007f506a7e2000)
>>>        libncurses.so.5 =>  /lib/libncurses.so.5 (0x00007f506a59d000)
>>>        libstdc++.so.6 =>  /usr/lib/libstdc++.so.6 (0x00007f506a28d000)
>>>        libgcc_s.so.1 =>  /lib/libgcc_s.so.1 (0x00007f506a076000)
>>>        libodbc.so.1 =>  /usr/lib/libodbc.so.1 (0x00007f5069e17000)
>>>        /lib64/ld-linux-x86-64.so.2 (0x00007f506bc31000)
>>>        libdl.so.2 =>  /lib/libdl.so.2 (0x00007f5069c13000)
>>>        libz.so.1 =>  /usr/lib/libz.so.1 (0x00007f50699fb000)
>>>        libltdl.so.7 =>  /usr/lib/libltdl.so.7 (0x00007f50697f2000)
>>>
>>> Recently in debian "testing" libtiff4 and libjpeg is upgraded:
>>> libtiff 3.9.2-3+b1
>>> libjpeg62 6b-16.1
>>> libjeg8 8-2.1
>>>
>>> Q&A:
>>> Q: How did you compile it? Using dpkg-buildpackage or via make/make install?
>>> A: svn-clean ./bootsrap ./configure make etc.
>>>
>>> Q: Do you have any debian versions of libtiff4(-dev) installed?
>>> A: Yes:3.8.2-11.2
>>>        
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>      
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>    




From technical at ttnc.co.uk  Sun Feb 21 06:12:28 2010
From: technical at ttnc.co.uk (TTNC - Technical)
Date: Sun, 21 Feb 2010 14:12:28 +0000
Subject: [Freeswitch-users] mod_fax undefined symbol
In-Reply-To: <4B813965.1030704@coppice.org>
References: <6FCE44FF-70C1-412D-B722-4C8D359381BA@ttnc.co.uk>	<201002191825.50957.errotan@gmail.com>	<AB3AD9C8-C162-4DB5-8095-97CA83EDDE57@ttnc.co.uk>	<201002191904.39081.errotan@gmail.com>	<E6568E57-A3E2-4650-A8CD-70B9C2846AF4@jerris.com>
	<EFB211AD-9B65-4CCE-9C5E-BC0A9ED55B7D@ttnc.co.uk>
	<4B813965.1030704@coppice.org>
Message-ID: <1656F554-374C-40A7-8E67-A311CB9BEC19@ttnc.co.uk>

Hi Steve

Just tried that, changed:

TIFFSetField(t->tiff_file,
                     TIFFTAG_ROWSPERSTRIP,
                     TIFFDefaultStripSize(t->tiff_file, 0));
to:

TIFFSetField(t->tiff_file,
                     TIFFTAG_ROWSPERSTRIP,
                     0);

Compiled OK, but then got the following error when trying to 'load mod_fax':

2010-02-21 14:08:30.242671 [CRIT] switch_loadable_module.c:882 Error Loading module /opt/freeswitch/mod/mod_fax.so
**/opt/freeswitch/mod/mod_fax.so: undefined symbol: TIFFSetDirectory**

It seems TIFFSetDirectory appears in quite a few places throughout t4_rx.c along with t4.c - so I doubt going through and commenting them out will really work?

Hopefully this maybe of some use to anyone looking into the problem though?

Anything else I can do to help then please let me know.

Russ


On 21 Feb 2010, at 13:47, Steve Underwood wrote:

> Hi Russ,
> 
> The only place in FS where TIFFDefaulyStripSize is used is in the file 
> t4_rx.c, and you probably won't actually be calling it. Try commenting 
> out that line, and see if there are any other stumbling blocks. Often 
> there is a mass of errors, and the system just tells you about them one 
> by one.
> 
> Steve
> 
> On 02/21/2010 09:15 PM, TTNC - Technical wrote:
>> Hi Guys
>> 
>> Without meaning to hassle (I know I am), has there been any progress with this bug yet? Anything else I can do to assist?
>> 
>> Please let me know.
>> 
>> Thanks
>> 
>> Russ
>> 
>> On 20 Feb 2010, at 02:00, Michael Jerris wrote:
>> 
>> 
>>> replying with more details on jira.
>>> 
>>> 
>>> On Feb 19, 2010, at 1:04 PM, Pusk?s Zsolt wrote:
>>> 
>>> 
>>>> 2010. febru?r 19. 18.44.32 TTNC - Technical d?tummal ezt ?rta:
>>>> 
>>>>> On 19 Feb 2010, at 17:25, Pusk?s Zsolt wrote:
>>>>> 
>>>>>> Just compiled svn16700 on Debian "lenny" x86, mod_fax loads and works
>>>>>> perfectly. I have an ongoing compile on another machine (amd64) if It
>>>>>> don't works i will send a mail (in 1 hour) otherwise consider it working.
>>>>>> 
>>>>> How did you compile it? Using dpkg-buildpackage or via make/make install?
>>>>> 
>>>>> Do you have any debian versions of libtiff4(-dev) installed?
>>>>> 
>>>>> 
>>>> Okay i can confirm that mod_fax works on Debian "lenny" amd64 but it don't work
>>>> on Debian "testing,squeeze" amd64.
>>>> 
>>>> 2010-02-19 18:49:14.610297 [CRIT] switch_loadable_module.c:882 Error Loading
>>>> module /usr/local/freeswitch/mod/mod_fax.so
>>>> **/usr/local/freeswitch/mod/mod_fax.so: undefined symbol:
>>>> TIFFDefaultStripSize**
>>>> 
>>>> I haven't tried to compile mod_fax on testing before so i don't know what is
>>>> causeing the problem :(
>>>> 
>>>> # ldd mod_fax.so
>>>>       linux-vdso.so.1 =>   (0x00007fff106f6000)
>>>>       libm.so.6 =>  /lib/libm.so.6 (0x00007f506b711000)
>>>>       libfreeswitch.so.1 =>  /usr/local/freeswitch/lib/libfreeswitch.so.1
>>>> (0x00007f506b345000)
>>>>       libpthread.so.0 =>  /lib/libpthread.so.0 (0x00007f506b128000)
>>>>       libc.so.6 =>  /lib/libc.so.6 (0x00007f506add4000)
>>>>       libssl.so.0.9.8 =>  /usr/lib/libssl.so.0.9.8 (0x00007f506ab82000)
>>>>       libcrypto.so.0.9.8 =>  /usr/lib/libcrypto.so.0.9.8 (0x00007f506a7e2000)
>>>>       libncurses.so.5 =>  /lib/libncurses.so.5 (0x00007f506a59d000)
>>>>       libstdc++.so.6 =>  /usr/lib/libstdc++.so.6 (0x00007f506a28d000)
>>>>       libgcc_s.so.1 =>  /lib/libgcc_s.so.1 (0x00007f506a076000)
>>>>       libodbc.so.1 =>  /usr/lib/libodbc.so.1 (0x00007f5069e17000)
>>>>       /lib64/ld-linux-x86-64.so.2 (0x00007f506bc31000)
>>>>       libdl.so.2 =>  /lib/libdl.so.2 (0x00007f5069c13000)
>>>>       libz.so.1 =>  /usr/lib/libz.so.1 (0x00007f50699fb000)
>>>>       libltdl.so.7 =>  /usr/lib/libltdl.so.7 (0x00007f50697f2000)
>>>> 
>>>> Recently in debian "testing" libtiff4 and libjpeg is upgraded:
>>>> libtiff 3.9.2-3+b1
>>>> libjpeg62 6b-16.1
>>>> libjeg8 8-2.1
>>>> 
>>>> Q&A:
>>>> Q: How did you compile it? Using dpkg-buildpackage or via make/make install?
>>>> A: svn-clean ./bootsrap ./configure make etc.
>>>> 
>>>> Q: Do you have any debian versions of libtiff4(-dev) installed?
>>>> A: Yes:3.8.2-11.2
>>>> 
>>> 
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From steveu at coppice.org  Sun Feb 21 06:28:48 2010
From: steveu at coppice.org (Steve Underwood)
Date: Sun, 21 Feb 2010 22:28:48 +0800
Subject: [Freeswitch-users] mod_fax undefined symbol
In-Reply-To: <1656F554-374C-40A7-8E67-A311CB9BEC19@ttnc.co.uk>
References: <6FCE44FF-70C1-412D-B722-4C8D359381BA@ttnc.co.uk>	<201002191825.50957.errotan@gmail.com>	<AB3AD9C8-C162-4DB5-8095-97CA83EDDE57@ttnc.co.uk>	<201002191904.39081.errotan@gmail.com>	<E6568E57-A3E2-4650-A8CD-70B9C2846AF4@jerris.com>	<EFB211AD-9B65-4CCE-9C5E-BC0A9ED55B7D@ttnc.co.uk>	<4B813965.1030704@coppice.org>
	<1656F554-374C-40A7-8E67-A311CB9BEC19@ttnc.co.uk>
Message-ID: <4B814320.7040306@coppice.org>

Hi Russ,

It sounds like this is nothing to do with the libtiff version, but that 
you just don't have libtiff there at all. Having no familiarity with 
Debian, I'll leave further analysis to someone else.

Regards,
Steve

On 02/21/2010 10:12 PM, TTNC - Technical wrote:
> Hi Steve
>
> Just tried that, changed:
>
> TIFFSetField(t->tiff_file,
>                       TIFFTAG_ROWSPERSTRIP,
>                       TIFFDefaultStripSize(t->tiff_file, 0));
> to:
>
> TIFFSetField(t->tiff_file,
>                       TIFFTAG_ROWSPERSTRIP,
>                       0);
>
> Compiled OK, but then got the following error when trying to 'load mod_fax':
>
> 2010-02-21 14:08:30.242671 [CRIT] switch_loadable_module.c:882 Error Loading module /opt/freeswitch/mod/mod_fax.so
> **/opt/freeswitch/mod/mod_fax.so: undefined symbol: TIFFSetDirectory**
>
> It seems TIFFSetDirectory appears in quite a few places throughout t4_rx.c along with t4.c - so I doubt going through and commenting them out will really work?
>
> Hopefully this maybe of some use to anyone looking into the problem though?
>
> Anything else I can do to help then please let me know.
>
> Russ
>
>
> On 21 Feb 2010, at 13:47, Steve Underwood wrote:
>
>    
>> Hi Russ,
>>
>> The only place in FS where TIFFDefaulyStripSize is used is in the file
>> t4_rx.c, and you probably won't actually be calling it. Try commenting
>> out that line, and see if there are any other stumbling blocks. Often
>> there is a mass of errors, and the system just tells you about them one
>> by one.
>>
>> Steve
>>
>> On 02/21/2010 09:15 PM, TTNC - Technical wrote:
>>      
>>> Hi Guys
>>>
>>> Without meaning to hassle (I know I am), has there been any progress with this bug yet? Anything else I can do to assist?
>>>
>>> Please let me know.
>>>
>>> Thanks
>>>
>>> Russ
>>>
>>> On 20 Feb 2010, at 02:00, Michael Jerris wrote:
>>>
>>>
>>>        
>>>> replying with more details on jira.
>>>>
>>>>
>>>> On Feb 19, 2010, at 1:04 PM, Pusk?s Zsolt wrote:
>>>>
>>>>
>>>>          
>>>>> 2010. febru?r 19. 18.44.32 TTNC - Technical d?tummal ezt ?rta:
>>>>>
>>>>>            
>>>>>> On 19 Feb 2010, at 17:25, Pusk?s Zsolt wrote:
>>>>>>
>>>>>>              
>>>>>>> Just compiled svn16700 on Debian "lenny" x86, mod_fax loads and works
>>>>>>> perfectly. I have an ongoing compile on another machine (amd64) if It
>>>>>>> don't works i will send a mail (in 1 hour) otherwise consider it working.
>>>>>>>
>>>>>>>                
>>>>>> How did you compile it? Using dpkg-buildpackage or via make/make install?
>>>>>>
>>>>>> Do you have any debian versions of libtiff4(-dev) installed?
>>>>>>
>>>>>>
>>>>>>              
>>>>> Okay i can confirm that mod_fax works on Debian "lenny" amd64 but it don't work
>>>>> on Debian "testing,squeeze" amd64.
>>>>>
>>>>> 2010-02-19 18:49:14.610297 [CRIT] switch_loadable_module.c:882 Error Loading
>>>>> module /usr/local/freeswitch/mod/mod_fax.so
>>>>> **/usr/local/freeswitch/mod/mod_fax.so: undefined symbol:
>>>>> TIFFDefaultStripSize**
>>>>>
>>>>> I haven't tried to compile mod_fax on testing before so i don't know what is
>>>>> causeing the problem :(
>>>>>
>>>>> # ldd mod_fax.so
>>>>>        linux-vdso.so.1 =>    (0x00007fff106f6000)
>>>>>        libm.so.6 =>   /lib/libm.so.6 (0x00007f506b711000)
>>>>>        libfreeswitch.so.1 =>   /usr/local/freeswitch/lib/libfreeswitch.so.1
>>>>> (0x00007f506b345000)
>>>>>        libpthread.so.0 =>   /lib/libpthread.so.0 (0x00007f506b128000)
>>>>>        libc.so.6 =>   /lib/libc.so.6 (0x00007f506add4000)
>>>>>        libssl.so.0.9.8 =>   /usr/lib/libssl.so.0.9.8 (0x00007f506ab82000)
>>>>>        libcrypto.so.0.9.8 =>   /usr/lib/libcrypto.so.0.9.8 (0x00007f506a7e2000)
>>>>>        libncurses.so.5 =>   /lib/libncurses.so.5 (0x00007f506a59d000)
>>>>>        libstdc++.so.6 =>   /usr/lib/libstdc++.so.6 (0x00007f506a28d000)
>>>>>        libgcc_s.so.1 =>   /lib/libgcc_s.so.1 (0x00007f506a076000)
>>>>>        libodbc.so.1 =>   /usr/lib/libodbc.so.1 (0x00007f5069e17000)
>>>>>        /lib64/ld-linux-x86-64.so.2 (0x00007f506bc31000)
>>>>>        libdl.so.2 =>   /lib/libdl.so.2 (0x00007f5069c13000)
>>>>>        libz.so.1 =>   /usr/lib/libz.so.1 (0x00007f50699fb000)
>>>>>        libltdl.so.7 =>   /usr/lib/libltdl.so.7 (0x00007f50697f2000)
>>>>>
>>>>> Recently in debian "testing" libtiff4 and libjpeg is upgraded:
>>>>> libtiff 3.9.2-3+b1
>>>>> libjpeg62 6b-16.1
>>>>> libjeg8 8-2.1
>>>>>
>>>>> Q&A:
>>>>> Q: How did you compile it? Using dpkg-buildpackage or via make/make install?
>>>>> A: svn-clean ./bootsrap ./configure make etc.
>>>>>
>>>>> Q: Do you have any debian versions of libtiff4(-dev) installed?
>>>>> A: Yes:3.8.2-11.2
>>>>>
>>>>>            




From moizchinoy at gmail.com  Sun Feb 21 08:00:59 2010
From: moizchinoy at gmail.com (Moiz Chinoy)
Date: Sun, 21 Feb 2010 20:00:59 +0400
Subject: [Freeswitch-users] Dingaling - external rtp supported?
In-Reply-To: <191c3a031002171136q7424e9bai34379abb542daf93@mail.gmail.com>
References: <29b888f81002160835y6e290319ub0d04c088caa1b79@mail.gmail.com>
	<DEA39AD1-0EE9-473A-B78C-CDDF3F100578@freeswitch.org>
	<29b888f81002170318p63622f3fhc8d26ad0e85602d2@mail.gmail.com>
	<191c3a031002170538r3b029515i8a000812a70ad558@mail.gmail.com>
	<29b888f81002170741y3d13cb57ycb4436a372c4eded@mail.gmail.com>
	<191c3a031002171136q7424e9bai34379abb542daf93@mail.gmail.com>
Message-ID: <29b888f81002210800v617bca9dk4bf1abf1bd791ee0@mail.gmail.com>

Guys,

To make things simple gtalk client is entirely on different network.

Call comes from outside through external Sip profile.

If gtalk answers the call after 3-4 rings both parties can hear each other.
If gtalk answers the call after 2 rings both parties no one can hear each other.
If gtalk answers the call immediately FS crashes.

Attached is the screen shot of the error...

Here is the FS log...
--------------------------------
http://pastebin.freeswitch.org/12197

External Sip Profile has following lines:
---------------------------------------------------------
    <param name="rtp-ip" value="$${local_ip_v4}"/>
    <param name="sip-ip" value="$${local_ip_v4}"/>
    <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
    <param name="ext-sip-ip" value="$${external_sip_ip}"/>

Jingle Client.xml has following lines:
-----------------------------------------------------
    <param name="rtp-ip" value="$${bind_server_ip}"/>
    <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
    <param name="disable-rtp-auto-adjust" value="true"/>
    <!--<param name="avatar" value="/path/to/tiny.jpg"/>-->
    <!--<param name="candidate-acl" value="wan.auto"/>-->
    <!--<param name="local-network-acl" value="localnet.auto"/>-->

Vars.xml has following lines:
-------------------------------------------
<X-PRE-PROCESS cmd="set" data="external_rtp_ip=stun:stun.freeswitch.org"/>
<X-PRE-PROCESS cmd="set" data="external_sip_ip=stun:stun.freeswitch.org"/>


Please advise me how can I provide more of the required data.

On Wed, Feb 17, 2010 at 11:36 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> you cant combine stun and gtalk and boxes in the same lan very easily if you
> do need to do that you will need to mess with
>
> <param name="candidate-acl" value="wan.auto"/>
> <param name="local-network-acl" value="localnet.auto"/>
>
>
>
>
>
> On Wed, Feb 17, 2010 at 9:41 AM, Moiz Chinoy <moizchinoy at gmail.com> wrote:
>>
>> Guys I am unable to produce the crash but now both parties cannot hear
>> each other!
>>
>> Vars.xml has following lines:
>> ?<X-PRE-PROCESS cmd="set"
>> data="external_rtp_ip=stun:stun.freeswitch.org"/>
>> ?<X-PRE-PROCESS cmd="set"
>> data="external_sip_ip=stun:stun.freeswitch.org"/>
>>
>> Jingle Client.xml has following lines:
>> ? ?<param name="rtp-ip" value="$${bind_server_ip}"/>
>> ? ?<!-- <param name="ext-rtp-ip" value="auto-nat"/> -->
>> ? ?<param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
>> ? ?<param name="disable-rtp-auto-adjust" value="true"/>
>> ? ?<param name="local-network-acl" value="localnet.auto"/>
>>
>> On Wed, Feb 17, 2010 at 5:38 PM, Anthony Minessale
>> <anthony.minessale at gmail.com> wrote:
>> > Obtain a stack trace from the crash.
>> >
>> > On Feb 17, 2010 5:26 AM, "Moiz Chinoy" <moizchinoy at gmail.com> wrote:
>> >
>> > Hi,
>> >
>> > FS rev: 16673
>> > Platform: Windows
>> >
>> > More details:
>> >
>> > FS is behind NAT and machine is running a VPN connection.
>> >
>> > FS and GTalk client on the same machine:
>> >
>> > --------------------------------------------------------------------------------------------------
>> > jingle profile client.xml has following line:
>> > <param name="rtp-ip" value="$${bind_server_ip}"/>
>> >
>> > External SIP call is successfully bridged to GTalk client.
>> >
>> >
>> > FS and GTalk client on the different machine:
>> >
>> > --------------------------------------------------------------------------------------------------
>> > jingle profile client.xml has following lines:
>> > <param name="rtp-ip" value="$${bind_server_ip}"/>
>> >
>> > <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
>> >
>> > As soon as external SIP call land and I try to bridge the call to
>> > GTalk client, FS crashes.
>> >
>> >
>> > NAT Details:
>> > ---------------------------
>> > I think my NAT does not support UpNP or PMP. The reason I say it
>> > because when FS starts following message is displayed:
>> >
>> > 2010-02-17 09:15:42.809674 [INFO] switch_nat.c:409 Scanning for NAT
>> > 2010-02-17 09:15:42.809674 [ERR] switch_nat.c:197 Error checking for
>> > PMP [init failed]
>> > 2010-02-17 09:15:42.809674 [DEBUG] switch_nat.c:414 Checking for UPnP
>> > 2010-02-17 09:15:56.825388 [DEBUG] switch_nat.c:114 No
>> > InternetGatewayDevice, using first entry as default
>> > (http://192.168.16.17:50144/).
>> > 2010-02-17 09:15:56.887889 [INFO] switch_nat.c:429 No PMP or UPnP NAT
>> > devices detected!
>> >
>> >
>> >
>> > On Tue, Feb 16, 2010 at 8:41 PM, Brian West <brian at freeswitch.org>
>> > wrote:
>> >> can you please update...
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>>
>>
>> --
>> Regards,
>> Moiz Chinoy.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Regards,
Moiz Chinoy.
-------------- next part --------------
A non-text attachment was scrubbed...
Name: mutex_error.JPG
Type: image/jpeg
Size: 33069 bytes
Desc: not available
Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100221/d942434c/attachment-0002.jpe 

From infos at madovsky.org  Sun Feb 21 09:43:34 2010
From: infos at madovsky.org (Madovsky)
Date: Sun, 21 Feb 2010 12:43:34 -0500
Subject: [Freeswitch-users] codec negociations
Message-ID: <E16A52EDBA634BDB98A00F938612E526@MOBILEE1705>

Hi,

is it possible for FS to take the callee codec from an interant call to external as reference ?
example:
userA (with only GSM) from FS calls whoeverUser at whateverdomain, this whoeverUser has only PCMU codec,
so FS has a list of GSM,PCMU,PCMA and so starts to transcode from PCMU to GSM.

I succeed to do the contrary (it transcodes from external to internal phone)

Thanks

Franck
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100221/04a36de7/attachment-0002.html 

From oseslija at gmail.com  Sun Feb 21 10:03:05 2010
From: oseslija at gmail.com (Ognjen Seslija)
Date: Sun, 21 Feb 2010 19:03:05 +0100
Subject: [Freeswitch-users] ASTPP For FreeSWITCH
In-Reply-To: <4B799839.2090008@aleph-com.net>
References: <87f2f3b91002150959q6accece5od67b262fb38ec41e@mail.gmail.com>
	<4B799839.2090008@aleph-com.net>
Message-ID: <4468a6771002211003o41fdb86en72b9c7a669a17ab1@mail.gmail.com>

I installed it. Don't see the way to switch from current asterisk menus to
FS ones.

Is there a irc channel to ask questions?

Regards,
Ognjen

On Mon, Feb 15, 2010 at 7:53 PM, Darren Wiebe <darren at aleph-com.net> wrote:

>  I will comment.  We've been using ASTPP for rating freeswitch cdrs for
> some time already.  It provides lcr from a database as well as sip user
> management.  It uses the mod_xml_curl and mod_xml_cdr modules for routing as
> well as realtime rating.  It also has an application that can listen to
> freeswitch and rate calls in realtime that way.  I patched a couple of bugs
> earlier this morning and I would not say that it's bug free but it's
> certainly in testing.
>
> Darren Wiebe
> darren at aleph-com.net
>
>
>
> On 02/15/2010 10:59 AM, Michael Collins wrote:
>
> Hey all,
>
> Here's a quick story <http://www.freeswitch.org/node/239> about ASTPP and
> FreeSWITCH. If you are using ASTPP with FreeSWITCH please let me know how it
> works. I didn't see any information on our wiki about ASTPP. If ASTPP is
> viable then we should document it as best we can.
>
> Thanks!
> -Michael
>
>
> _______________________________________________
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100221/ecadfb4b/attachment-0002.html 

From darren at aleph-com.net  Sun Feb 21 11:37:44 2010
From: darren at aleph-com.net (Darren Wiebe)
Date: Sun, 21 Feb 2010 12:37:44 -0700
Subject: [Freeswitch-users] ASTPP For FreeSWITCH
In-Reply-To: <4468a6771002211003o41fdb86en72b9c7a669a17ab1@mail.gmail.com>
References: <87f2f3b91002150959q6accece5od67b262fb38ec41e@mail.gmail.com>	<4B799839.2090008@aleph-com.net>
	<4468a6771002211003o41fdb86en72b9c7a669a17ab1@mail.gmail.com>
Message-ID: <4B818B88.3030806@aleph-com.net>

Look in System->Configuration.  Set "users_dids_rt" to 0 and 
"users_dids_freeswitch" to 1.  There's not currently an irc channel but 
that's a good idea.

Darren Wiebe
darren at aleph-com.net


On 21/02/2010 11:03 AM, Ognjen Seslija wrote:
> I installed it. Don't see the way to switch from current asterisk 
> menus to FS ones.
>
> Is there a irc channel to ask questions?
>
> Regards,
> Ognjen
>
> On Mon, Feb 15, 2010 at 7:53 PM, Darren Wiebe <darren at aleph-com.net 
> <mailto:darren at aleph-com.net>> wrote:
>
>     I will comment.  We've been using ASTPP for rating freeswitch cdrs
>     for some time already.  It provides lcr from a database as well as
>     sip user management.  It uses the mod_xml_curl and mod_xml_cdr
>     modules for routing as well as realtime rating.  It also has an
>     application that can listen to freeswitch and rate calls in
>     realtime that way.  I patched a couple of bugs earlier this
>     morning and I would not say that it's bug free but it's certainly
>     in testing.
>
>     Darren Wiebe
>     darren at aleph-com.net <mailto:darren at aleph-com.net>
>
>
>
>     On 02/15/2010 10:59 AM, Michael Collins wrote:
>>     Hey all,
>>
>>     Here's a quick story <http://www.freeswitch.org/node/239> about
>>     ASTPP and FreeSWITCH. If you are using ASTPP with FreeSWITCH
>>     please let me know how it works. I didn't see any information on
>>     our wiki about ASTPP. If ASTPP is viable then we should document
>>     it as best we can.
>>
>>     Thanks!
>>     -Michael
>>
>>
>>     _______________________________________________
>>     FreeSWITCH-users mailing list
>>     FreeSWITCH-users at lists.freeswitch.org  <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     http://www.freeswitch.org
>>        
>
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>    

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100221/6bd70949/attachment-0002.html 

From oseslija at gmail.com  Sun Feb 21 12:04:31 2010
From: oseslija at gmail.com (Ognjen Seslija)
Date: Sun, 21 Feb 2010 21:04:31 +0100
Subject: [Freeswitch-users] ASTPP For FreeSWITCH
In-Reply-To: <4B818B88.3030806@aleph-com.net>
References: <87f2f3b91002150959q6accece5od67b262fb38ec41e@mail.gmail.com>
	<4B799839.2090008@aleph-com.net>
	<4468a6771002211003o41fdb86en72b9c7a669a17ab1@mail.gmail.com>
	<4B818B88.3030806@aleph-com.net>
Message-ID: <4468a6771002211204y1aeb09b0tbe7980da9b15da19@mail.gmail.com>

Thanks.

On Sun, Feb 21, 2010 at 8:37 PM, Darren Wiebe <darren at aleph-com.net> wrote:

>  Look in System->Configuration.  Set "users_dids_rt" to 0 and
> "users_dids_freeswitch" to 1.  There's not currently an irc channel but
> that's a good idea.
>
>
> Darren Wiebe
> darren at aleph-com.net
>
>
> On 21/02/2010 11:03 AM, Ognjen Seslija wrote:
>
> I installed it. Don't see the way to switch from current asterisk menus to
> FS ones.
>
> Is there a irc channel to ask questions?
>
> Regards,
> Ognjen
>
> On Mon, Feb 15, 2010 at 7:53 PM, Darren Wiebe <darren at aleph-com.net>wrote:
>
>>  I will comment.  We've been using ASTPP for rating freeswitch cdrs for
>> some time already.  It provides lcr from a database as well as sip user
>> management.  It uses the mod_xml_curl and mod_xml_cdr modules for routing as
>> well as realtime rating.  It also has an application that can listen to
>> freeswitch and rate calls in realtime that way.  I patched a couple of bugs
>> earlier this morning and I would not say that it's bug free but it's
>> certainly in testing.
>>
>> Darren Wiebe
>> darren at aleph-com.net
>>
>>
>>
>> On 02/15/2010 10:59 AM, Michael Collins wrote:
>>
>>  Hey all,
>>
>> Here's a quick story <http://www.freeswitch.org/node/239> about ASTPP and
>> FreeSWITCH. If you are using ASTPP with FreeSWITCH please let me know how it
>> works. I didn't see any information on our wiki about ASTPP. If ASTPP is
>> viable then we should document it as best we can.
>>
>> Thanks!
>> -Michael
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100221/0226f863/attachment-0002.html 

From technical at ttnc.co.uk  Sun Feb 21 13:25:05 2010
From: technical at ttnc.co.uk (TTNC - Technical)
Date: Sun, 21 Feb 2010 21:25:05 +0000
Subject: [Freeswitch-users] mod_fax undefined symbol
In-Reply-To: <201002191904.39081.errotan@gmail.com>
References: <6FCE44FF-70C1-412D-B722-4C8D359381BA@ttnc.co.uk>
	<201002191825.50957.errotan@gmail.com>
	<AB3AD9C8-C162-4DB5-8095-97CA83EDDE57@ttnc.co.uk>
	<201002191904.39081.errotan@gmail.com>
Message-ID: <41F9D585-2116-4219-9AF7-E8E944D43362@ttnc.co.uk>

Out of interest, I downgraded my versions of libtiff and libjpeg to the versions shipped with Lenny:

voipin1:/opt# dpkg -l | egrep 'libtiff|libjpeg'
ii  libjpeg62                            6b-14                    The Independent JPEG Group's JPEG runtime library
ii  libjpeg62-dev                        6b-14                    Development files for the IJG JPEG library
ii  libtiff4                             3.8.2-11.2               Tag Image File Format (TIFF) library
ii  libtiff4-dev                         3.8.2-11.2               Tag Image File Format library (TIFF), development files
ii  libtiffxx0c2                         3.8.2-11.2               Tag Image File Format (TIFF) library -- C++ interface

Everything else stayed at the 'squeeze' version.

Still didn't make any different, **/opt/freeswitch/mod/mod_fax.so: undefined symbol: TIFFDefaultStripSize**

I'm guessing that points to it being a problem outside of these packages and somewhere else in Debian?

Russ


On 19 Feb 2010, at 18:04, Pusk?s Zsolt wrote:

> 2010. febru?r 19. 18.44.32 TTNC - Technical d?tummal ezt ?rta:
>> On 19 Feb 2010, at 17:25, Pusk?s Zsolt wrote:
>>> Just compiled svn16700 on Debian "lenny" x86, mod_fax loads and works
>>> perfectly. I have an ongoing compile on another machine (amd64) if It
>>> don't works i will send a mail (in 1 hour) otherwise consider it working.
>> 
>> How did you compile it? Using dpkg-buildpackage or via make/make install?
>> 
>> Do you have any debian versions of libtiff4(-dev) installed?
>> 
> 
> Okay i can confirm that mod_fax works on Debian "lenny" amd64 but it don't work 
> on Debian "testing,squeeze" amd64.
> 
> 2010-02-19 18:49:14.610297 [CRIT] switch_loadable_module.c:882 Error Loading 
> module /usr/local/freeswitch/mod/mod_fax.so
> **/usr/local/freeswitch/mod/mod_fax.so: undefined symbol: 
> TIFFDefaultStripSize**
> 
> I haven't tried to compile mod_fax on testing before so i don't know what is 
> causeing the problem :(
> 
> # ldd mod_fax.so
>        linux-vdso.so.1 =>  (0x00007fff106f6000)
>        libm.so.6 => /lib/libm.so.6 (0x00007f506b711000)
>        libfreeswitch.so.1 => /usr/local/freeswitch/lib/libfreeswitch.so.1 
> (0x00007f506b345000)
>        libpthread.so.0 => /lib/libpthread.so.0 (0x00007f506b128000)
>        libc.so.6 => /lib/libc.so.6 (0x00007f506add4000)
>        libssl.so.0.9.8 => /usr/lib/libssl.so.0.9.8 (0x00007f506ab82000)
>        libcrypto.so.0.9.8 => /usr/lib/libcrypto.so.0.9.8 (0x00007f506a7e2000)
>        libncurses.so.5 => /lib/libncurses.so.5 (0x00007f506a59d000)
>        libstdc++.so.6 => /usr/lib/libstdc++.so.6 (0x00007f506a28d000)
>        libgcc_s.so.1 => /lib/libgcc_s.so.1 (0x00007f506a076000)
>        libodbc.so.1 => /usr/lib/libodbc.so.1 (0x00007f5069e17000)
>        /lib64/ld-linux-x86-64.so.2 (0x00007f506bc31000)
>        libdl.so.2 => /lib/libdl.so.2 (0x00007f5069c13000)
>        libz.so.1 => /usr/lib/libz.so.1 (0x00007f50699fb000)
>        libltdl.so.7 => /usr/lib/libltdl.so.7 (0x00007f50697f2000)
> 
> Recently in debian "testing" libtiff4 and libjpeg is upgraded:
> libtiff 3.9.2-3+b1
> libjpeg62 6b-16.1
> libjeg8 8-2.1
> 
> Q&A:
> Q: How did you compile it? Using dpkg-buildpackage or via make/make install?
> A: svn-clean ./bootsrap ./configure make etc.
> 
> Q: Do you have any debian versions of libtiff4(-dev) installed?
> A: Yes:3.8.2-11.2
> 
> I open a jira for this.
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From matt at webcontracts.co.uk  Sun Feb 21 15:26:08 2010
From: matt at webcontracts.co.uk (Matthew Law)
Date: Sun, 21 Feb 2010 23:26:08 -0000
Subject: [Freeswitch-users] Dialplan question
Message-ID: <2dac3814a041c79208af8a433b279566.squirrel@www.webcontracts.co.uk>

I have FS installed and I can make outgoing calls through my SIP provider.
 I can also call other extensions (FS is running on a small Xen domU on
the internet), but I am having problems getting the dialplan for incoming
calls to work.

What I want to do is have incoming calls on my number ring all extensions,
e.g. 1000 - 1005 for 10 seconds and then go to voicemail for extension
1000.  If there are no logged-on users, then it should go straight to
voicemail.  Rather than bite off too much, I thought I would try and get a
very basic setup working and take it from there...

At the moment it goes straight to voicemail for extension 1000 even if
1000 is logged in.  Here are the dialplan files I have (everything else is
default from the trunk install):

dialplan/public/00_inbound_did.xml:

<include>
    <extension name="08433xxxxxxx">
        <condition field="destination_number" expression="^(0843xxxxxxx)$">
            <action application="transfer" data="$1 XML default"/>
        </condition>
    </extension>
</include>

dialplan/default/12_voiptalk.xml:

<include>
  <extension name="public_did">
    <condition field="destination_number" expression="^(0843xxxxxxx)$">
      <action application="transfer" data="1000 XML default"/>
    </condition>
  </extension>
  <extension name="outgoing">
    <condition field="destination_number" expression="^9([0-9]+)$">
      <action application="bridge" data="sofia/gateway/voiptalk/$1" />
    </condition>
  </extension>
</include>

I would be very grateful if someone could tell me where I am going wrong. 
I've been looking at various FS wiki pages for hours as well as the
example configs and can't seem to make any headway.  My other question is
what command should I be run after changing the dialplan? is it just
'reloadxml'?


Many thanks,

Matt.




From brian at freeswitch.org  Sun Feb 21 15:32:18 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 21 Feb 2010 17:32:18 -0600
Subject: [Freeswitch-users] Dialplan question
In-Reply-To: <2dac3814a041c79208af8a433b279566.squirrel@www.webcontracts.co.uk>
References: <2dac3814a041c79208af8a433b279566.squirrel@www.webcontracts.co.uk>
Message-ID: <FD94E01E-580B-457D-A26C-1641A680A76A@freeswitch.org>

What do your logs say??? Press F8 and make a call.  Then check the green lines.. maybe its not matching the right thing.

/b

On Feb 21, 2010, at 5:26 PM, Matthew Law wrote:

> I would be very grateful if someone could tell me where I am going wrong. 
> I've been looking at various FS wiki pages for hours as well as the
> example configs and can't seem to make any headway.  My other question is
> what command should I be run after changing the dialplan? is it just
> 'reloadxml'?




From brian at microcomaustralia.com.au  Sun Feb 21 15:44:03 2010
From: brian at microcomaustralia.com.au (Brian May)
Date: Mon, 22 Feb 2010 10:44:03 +1100
Subject: [Freeswitch-users] list all users
In-Reply-To: <3c5cf5261002202000h12b9d6b2ybefd5422fd4f9ee5@mail.gmail.com>
References: <3c5cf5261002201821q7b9f024bt7ab32b10946a5370@mail.gmail.com>
	<33c87fa31002201908x55e43e62g32b340872df9863d@mail.gmail.com>
	<3c5cf5261002202000h12b9d6b2ybefd5422fd4f9ee5@mail.gmail.com>
Message-ID: <3c5cf5261002211544m79832c1cwfe12001dc8b411e9@mail.gmail.com>

On 21 February 2010 15:00, Brian May <brian at microcomaustralia.com.au> wrote:
> As per one suggestion (as far as I can tell this shouldn't be
> required) I tried changing this (in dialplan/default.xml):
>
> <action application="bridge" data="user/${dialed_extension}@${domain_name}"/>
>
> with:
>
> <action application="bridge" data="sofia/internal/${dialed_extension}@${domain_name}"/>

Just noticed why this change didn't work. According to the example in
http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML I needed to
use

<action application="bridge"
data="sofia/internal/${dialed_extension}%${domain_name}"/>

as "The % behind the username tells FS to lookup the user in it's
local sip_registration database" and "If you don't want to bridge a
call to a local registered user, but to a SIP URI, use the @ instead"

disclaimer: not tested! However this matches exactly was was happening.
-- 
Brian May <brian at microcomaustralia.com.au>



From brian at microcomaustralia.com.au  Sun Feb 21 16:51:53 2010
From: brian at microcomaustralia.com.au (Brian May)
Date: Mon, 22 Feb 2010 11:51:53 +1100
Subject: [Freeswitch-users] outgoing dialplans for australia
In-Reply-To: <20100221091011.GA28429@jdc.jasonjgw.net>
References: <3c5cf5261002210030n12bd743fh69f0c862eae0db8@mail.gmail.com>
	<20100221091011.GA28429@jdc.jasonjgw.net>
Message-ID: <3c5cf5261002211651v7962c0afj30e055778b59b1e7@mail.gmail.com>

On 21 February 2010 20:10, Jason White <jason at jasonjgw.net> wrote:
> http://wiki.freeswitch.org/wiki/Extension_Status_Example

Actually suspect the solution might be even simpler. e.g something like:

<action application="set" data="hangup_after_bridge=true"/>
<action application="set"
data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,TIMEOUT,NO_ROUTE_DESTINATION,NORMAL_CIRCUIT_CONGESTION,NETWORK_OUT_OF_ORDER,etc"/>
<action application="bridge" data="sofia/.../$1"/>
<action application="bridge" data="openzap/.../$1"/>
error tone if all else fails

Only I am not in a position to test it just yet, and not sure yet how
to generate the error tone if everything fails either.
-- 
Brian May <brian at microcomaustralia.com.au>



From jason at jasonjgw.net  Sun Feb 21 17:13:29 2010
From: jason at jasonjgw.net (Jason White)
Date: Mon, 22 Feb 2010 12:13:29 +1100
Subject: [Freeswitch-users] outgoing dialplans for australia
In-Reply-To: <3c5cf5261002211651v7962c0afj30e055778b59b1e7@mail.gmail.com>
References: <3c5cf5261002210030n12bd743fh69f0c862eae0db8@mail.gmail.com>
	<20100221091011.GA28429@jdc.jasonjgw.net>
	<3c5cf5261002211651v7962c0afj30e055778b59b1e7@mail.gmail.com>
Message-ID: <20100222011329.GA14787@jdc.jasonjgw.net>

Brian May <brian at microcomaustralia.com.au> wrote:
> On 21 February 2010 20:10, Jason White <jason at jasonjgw.net> wrote:
> > http://wiki.freeswitch.org/wiki/Extension_Status_Example
> 
> Actually suspect the solution might be even simpler. e.g something like:
> 
> <action application="set" data="hangup_after_bridge=true"/>
> <action application="set"
> data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,TIMEOUT,NO_ROUTE_DESTINATION,NORMAL_CIRCUIT_CONGESTION,NETWORK_OUT_OF_ORDER,etc"/>
> <action application="bridge" data="sofia/.../$1"/>
> <action application="bridge" data="openzap/.../$1"/>
> error tone if all else fails

That looks promising.




From brian at microcomaustralia.com.au  Sun Feb 21 18:13:01 2010
From: brian at microcomaustralia.com.au (Brian May)
Date: Mon, 22 Feb 2010 13:13:01 +1100
Subject: [Freeswitch-users] variable substitutions
Message-ID: <3c5cf5261002211813m1835b2b6oc246518a057b578c@mail.gmail.com>

In the sample dialplan, I see the syntax ${...} and the syntax
$${...}. Are both these correct? Using eval suggests that the later
simply prefix the result with a dollar sign, I am not sure if this
intended...

examples - copied from random non-sequential lines in default.xml:

        <action application="set" data="ringback=${us-ring}"/>
        <action application="bridge"
data="user/${dialed_extension}@${domain_name}"/>

and for the $${...} syntax:

        <action application="set" data="transfer_ringback=$${hold_music}"/>
        <action application="set" data="ringback=$${uk-ring}"/>
        <action application="bridge" data="sofia/$${domain}/2000 at bkw.org"/>

Maybe both mean the same thing?
-- 
Brian May <brian at microcomaustralia.com.au>



From jason at jasonjgw.net  Sun Feb 21 18:24:02 2010
From: jason at jasonjgw.net (Jason White)
Date: Mon, 22 Feb 2010 13:24:02 +1100
Subject: [Freeswitch-users] variable substitutions
In-Reply-To: <3c5cf5261002211813m1835b2b6oc246518a057b578c@mail.gmail.com>
References: <3c5cf5261002211813m1835b2b6oc246518a057b578c@mail.gmail.com>
Message-ID: <20100222022402.GA15634@jdc.jasonjgw.net>

Brian May <brian at microcomaustralia.com.au> wrote:
> In the sample dialplan, I see the syntax ${...} and the syntax
> $${...}. Are both these correct? 

Yes. $$ is a preprocessor variable which is expanded when the XML
configuration is parsed. There is a wiki page on the subject.




From Russell.Mosemann at cune.org  Sun Feb 21 18:31:39 2010
From: Russell.Mosemann at cune.org (Russell Mosemann)
Date: Sun, 21 Feb 2010 20:31:39 -0600
Subject: [Freeswitch-users] variable substitutions
In-Reply-To: <3c5cf5261002211813m1835b2b6oc246518a057b578c@mail.gmail.com>
References: <3c5cf5261002211813m1835b2b6oc246518a057b578c@mail.gmail.com>
Message-ID: <6E9F8E83BBE74F2B8DF116B86AE05EDD@cune.pri>

Brian May wrote:

> In the sample dialplan, I see the syntax ${...} and the syntax
> $${...}. Are both these correct?

http://wiki.freeswitch.org/wiki/Channel_Variables#.24.7Bvariable.7D_vs._.24.24.7Bvariable.7D

--
Russell Mosemann




From brian at microcomaustralia.com.au  Sun Feb 21 18:41:35 2010
From: brian at microcomaustralia.com.au (Brian May)
Date: Mon, 22 Feb 2010 13:41:35 +1100
Subject: [Freeswitch-users] variable substitutions
In-Reply-To: <20100222022402.GA15634@jdc.jasonjgw.net>
References: <3c5cf5261002211813m1835b2b6oc246518a057b578c@mail.gmail.com>
	<20100222022402.GA15634@jdc.jasonjgw.net>
Message-ID: <3c5cf5261002211841p2d2626d7j6cfba3b063e05e5c@mail.gmail.com>

On 22 February 2010 13:24, Jason White <jason at jasonjgw.net> wrote:
> Yes. $$ is a preprocessor variable which is expanded when the XML
> configuration is parsed. There is a wiki page on the subject.

Oh, Ok. This looks like a good reference:

http://wiki.freeswitch.org/wiki/Channel_Variables

Thanks.
-- 
Brian May <brian at microcomaustralia.com.au>



From brian at microcomaustralia.com.au  Sun Feb 21 18:52:30 2010
From: brian at microcomaustralia.com.au (Brian May)
Date: Mon, 22 Feb 2010 13:52:30 +1100
Subject: [Freeswitch-users] altering callerid
Message-ID: <3c5cf5261002211852k577d4d78x32e43c0b4d29a1c4@mail.gmail.com>

Ok, now for something maybe a little bit dodgy. How do I alter the
callerid for an incoming call?

Example, the SIP provider I use provides callerid in the format
613XXXXXXXX, however some of the analogue phones have fixed width
displays and cannot display this long number correctly, as such, I
would like to change that to 03XXXXXXXX - compatible with what the
local telephone company uses.

Is this possible?

In asterisk I used:

exten => number/_61NXXXXXXXX,1,Set(CALLERID(num)=0${CALLERID(num):2})
exten => number/_X.,1,Set(CALLERID(num)=+${CALLERID(num)})

(disclaimer - 2nd line not tested with analogue phones)
-- 
Brian May <brian at microcomaustralia.com.au>



From brian at freeswitch.org  Sun Feb 21 18:58:30 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 21 Feb 2010 20:58:30 -0600
Subject: [Freeswitch-users] altering callerid
In-Reply-To: <3c5cf5261002211852k577d4d78x32e43c0b4d29a1c4@mail.gmail.com>
References: <3c5cf5261002211852k577d4d78x32e43c0b4d29a1c4@mail.gmail.com>
Message-ID: <8C9F2C93-DDF3-4283-8015-82062AF794CD@freeswitch.org>

read the variables page ;)

/b

On Feb 21, 2010, at 8:52 PM, Brian May wrote:

> Is this possible?




From brian at microcomaustralia.com.au  Sun Feb 21 19:34:52 2010
From: brian at microcomaustralia.com.au (Brian May)
Date: Mon, 22 Feb 2010 14:34:52 +1100
Subject: [Freeswitch-users] altering callerid
In-Reply-To: <8C9F2C93-DDF3-4283-8015-82062AF794CD@freeswitch.org>
References: <3c5cf5261002211852k577d4d78x32e43c0b4d29a1c4@mail.gmail.com>
	<8C9F2C93-DDF3-4283-8015-82062AF794CD@freeswitch.org>
Message-ID: <3c5cf5261002211934t12f79a5dh5564003af0f4b74d@mail.gmail.com>

On 22 February 2010 13:58, Brian West <brian at freeswitch.org> wrote:
> read the variables page ;)

Which variables? For caller_id_number it says "Practically it is read
only." What does this mean?
-- 
Brian May <brian at microcomaustralia.com.au>



From brian at freeswitch.org  Sun Feb 21 19:39:14 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 21 Feb 2010 21:39:14 -0600
Subject: [Freeswitch-users] altering callerid
In-Reply-To: <3c5cf5261002211934t12f79a5dh5564003af0f4b74d@mail.gmail.com>
References: <3c5cf5261002211852k577d4d78x32e43c0b4d29a1c4@mail.gmail.com>
	<8C9F2C93-DDF3-4283-8015-82062AF794CD@freeswitch.org>
	<3c5cf5261002211934t12f79a5dh5564003af0f4b74d@mail.gmail.com>
Message-ID: <973F4A98-1461-483A-BB9D-8C1746C7DFE8@freeswitch.org>

effective_caller_id_*
origination_caller_id_*

/b

On Feb 21, 2010, at 9:34 PM, Brian May wrote:

> Which variables? For caller_id_number it says "Practically it is read
> only." What does this mean?




From rupa at rupa.com  Sun Feb 21 19:46:01 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Sun, 21 Feb 2010 21:46:01 -0600
Subject: [Freeswitch-users] altering callerid
In-Reply-To: <3c5cf5261002211934t12f79a5dh5564003af0f4b74d@mail.gmail.com>
References: <3c5cf5261002211852k577d4d78x32e43c0b4d29a1c4@mail.gmail.com>
	<8C9F2C93-DDF3-4283-8015-82062AF794CD@freeswitch.org>
	<3c5cf5261002211934t12f79a5dh5564003af0f4b74d@mail.gmail.com>
Message-ID: <b39273531002211946y5361be9cvce49b9aaa8d10d26@mail.gmail.com>

caller_id is slightly tricky.  Use the dptool set_profile_var
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_set_profile_var.
 Though that is under documented.

But really, if you are just bridging then do what Brian said.  That is the
"appropriate" method in most cases.

On Sun, Feb 21, 2010 at 9:34 PM, Brian May
<brian at microcomaustralia.com.au>wrote:

> On 22 February 2010 13:58, Brian West <brian at freeswitch.org> wrote:
> > read the variables page ;)
>
> Which variables? For caller_id_number it says "Practically it is read
> only." What does this mean?
> --
> Brian May <brian at microcomaustralia.com.au>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
-Rupa
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100221/92374dcd/attachment-0002.html 

From brian at microcomaustralia.com.au  Sun Feb 21 19:50:15 2010
From: brian at microcomaustralia.com.au (Brian May)
Date: Mon, 22 Feb 2010 14:50:15 +1100
Subject: [Freeswitch-users] altering callerid
In-Reply-To: <973F4A98-1461-483A-BB9D-8C1746C7DFE8@freeswitch.org>
References: <3c5cf5261002211852k577d4d78x32e43c0b4d29a1c4@mail.gmail.com>
	<8C9F2C93-DDF3-4283-8015-82062AF794CD@freeswitch.org>
	<3c5cf5261002211934t12f79a5dh5564003af0f4b74d@mail.gmail.com>
	<973F4A98-1461-483A-BB9D-8C1746C7DFE8@freeswitch.org>
Message-ID: <3c5cf5261002211950t1bbbb34i20e6509ed345a1a5@mail.gmail.com>

On 22 February 2010 14:39, Brian West <brian at freeswitch.org> wrote:
> effective_caller_id_*
> origination_caller_id_*

How do I generate the new number? I need to be able to test if it
starts with '61', and if so, replace the first two digits with '0',
otherwise just prefix the number with '+'.
-- 
Brian May <brian at microcomaustralia.com.au>



From Russell.Mosemann at cune.org  Sun Feb 21 19:57:47 2010
From: Russell.Mosemann at cune.org (Russell Mosemann)
Date: Sun, 21 Feb 2010 21:57:47 -0600
Subject: [Freeswitch-users] altering callerid
In-Reply-To: <3c5cf5261002211950t1bbbb34i20e6509ed345a1a5@mail.gmail.com>
References: <3c5cf5261002211852k577d4d78x32e43c0b4d29a1c4@mail.gmail.com><8C9F2C93-DDF3-4283-8015-82062AF794CD@freeswitch.org><3c5cf5261002211934t12f79a5dh5564003af0f4b74d@mail.gmail.com><973F4A98-1461-483A-BB9D-8C1746C7DFE8@freeswitch.org>
	<3c5cf5261002211950t1bbbb34i20e6509ed345a1a5@mail.gmail.com>
Message-ID: <B93F97C3D14E4BEA833E14D920F5FCFC@cune.pri>

Brian May asked:
> How do I generate the new number? I need to be able to test if it
> starts with '61', and if so, replace the first two digits with '0',
> otherwise just prefix the number with '+'.

http://wiki.freeswitch.org/wiki/Regular_Expression
http://wiki.freeswitch.org/wiki/Dialplan_XML

--
Russell Mosemann




From freeswitch at cartissolutions.com  Sun Feb 21 20:00:05 2010
From: freeswitch at cartissolutions.com (Yossi Neiman)
Date: Sun, 21 Feb 2010 22:00:05 -0600
Subject: [Freeswitch-users] Doxygen help
Message-ID: <4B820145.2090109@cartissolutions.com>

I am not sure how many folks make use of the Doxygen documentation.  I 
know I do all the time.  I find that it provides a nice conceptual view 
of FreeSWITCH's API, which can make it very easy to find the functions 
and data types needed for writing modules for FreeSWITCH.

A person by the name of Mohammad Shahzad (apologies if I misspelled it) 
started to do a lot of work on revamping the Doxygen configuration for 
FreeSWITCH a few months back.  The problem with the work that he did was 
that he used Doxygen 1.6.x specific configuration parameters that are 
not understood by the Doxygen 1.4.x tree (which is what we currently use 
in FreeSWITCH) or even the 1.5.x tree which is what is in Slackware.  In 
my discussions with Michael Collins we have decided that it might not 
hurt to go ahead and move the Doxygen version forward to match the 1.6.x 
tree.

We are looking for somebody who has interest in working with the Doxygen 
configuration to continue the work that Mohammad had started and to help 
the project out.  If you are such a person, please contact me off-list 
and I can provide further information.

Thanks!

Yossi Neiman



From Russell.Mosemann at cune.org  Sun Feb 21 20:05:59 2010
From: Russell.Mosemann at cune.org (Russell Mosemann)
Date: Sun, 21 Feb 2010 22:05:59 -0600
Subject: [Freeswitch-users] altering callerid
In-Reply-To: <B93F97C3D14E4BEA833E14D920F5FCFC@cune.pri>
References: <3c5cf5261002211852k577d4d78x32e43c0b4d29a1c4@mail.gmail.com><8C9F2C93-DDF3-4283-8015-82062AF794CD@freeswitch.org><3c5cf5261002211934t12f79a5dh5564003af0f4b74d@mail.gmail.com><973F4A98-1461-483A-BB9D-8C1746C7DFE8@freeswitch.org><3c5cf5261002211950t1bbbb34i20e6509ed345a1a5@mail.gmail.com>
	<B93F97C3D14E4BEA833E14D920F5FCFC@cune.pri>
Message-ID: <69D702BA87A54ABA93281A8DA621BB12@cune.pri>

> Brian May asked:
> > How do I generate the new number? I need to be able to test if it
> > starts with '61', and if so, replace the first two digits with '0',
> > otherwise just prefix the number with '+'.
> 
> http://wiki.freeswitch.org/wiki/Regular_Expression
> http://wiki.freeswitch.org/wiki/Dialplan_XML

And

http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_set

--
Russell Mosemann




From jason at jasonjgw.net  Sun Feb 21 20:28:27 2010
From: jason at jasonjgw.net (Jason White)
Date: Mon, 22 Feb 2010 15:28:27 +1100
Subject: [Freeswitch-users] altering callerid
In-Reply-To: <69D702BA87A54ABA93281A8DA621BB12@cune.pri>
References: <3c5cf5261002211852k577d4d78x32e43c0b4d29a1c4@mail.gmail.com>
	<8C9F2C93-DDF3-4283-8015-82062AF794CD@freeswitch.org>
	<3c5cf5261002211934t12f79a5dh5564003af0f4b74d@mail.gmail.com>
	<973F4A98-1461-483A-BB9D-8C1746C7DFE8@freeswitch.org>
	<3c5cf5261002211950t1bbbb34i20e6509ed345a1a5@mail.gmail.com>
	<B93F97C3D14E4BEA833E14D920F5FCFC@cune.pri>
	<69D702BA87A54ABA93281A8DA621BB12@cune.pri>
Message-ID: <20100222042827.GA21444@jdc.jasonjgw.net>

Russell Mosemann <Russell.Mosemann at cune.org> wrote:
 > 
> > http://wiki.freeswitch.org/wiki/Regular_Expression
> > http://wiki.freeswitch.org/wiki/Dialplan_XML
> 
> And
> 
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_set

And as a further hint, regular expressions and variable substitutions ($1 $2
etc.).




From brian at microcomaustralia.com.au  Sun Feb 21 20:37:10 2010
From: brian at microcomaustralia.com.au (Brian May)
Date: Mon, 22 Feb 2010 15:37:10 +1100
Subject: [Freeswitch-users] altering callerid
In-Reply-To: <69D702BA87A54ABA93281A8DA621BB12@cune.pri>
References: <3c5cf5261002211852k577d4d78x32e43c0b4d29a1c4@mail.gmail.com>
	<8C9F2C93-DDF3-4283-8015-82062AF794CD@freeswitch.org>
	<3c5cf5261002211934t12f79a5dh5564003af0f4b74d@mail.gmail.com>
	<973F4A98-1461-483A-BB9D-8C1746C7DFE8@freeswitch.org>
	<3c5cf5261002211950t1bbbb34i20e6509ed345a1a5@mail.gmail.com>
	<B93F97C3D14E4BEA833E14D920F5FCFC@cune.pri>
	<69D702BA87A54ABA93281A8DA621BB12@cune.pri>
Message-ID: <3c5cf5261002212037l3576f9d7s33e32c310799aaf7@mail.gmail.com>

On 22 February 2010 15:05, Russell Mosemann >>
>> http://wiki.freeswitch.org/wiki/Regular_Expression
>> http://wiki.freeswitch.org/wiki/Dialplan_XML
>
> And
>
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_set

So would something like this work?

  <extension name="oztel_cid" continue="true">
    <condition field="destination_number" expression="^(12345678)$"/>
    <condition field="caller_id_number"
expression="^61([2-9]\d\d\d\d\d\d\d\d)$" break="on-true">
      <action application="set" data="effective_caller_id_number=0$1"/>
    </condition>
    <condition field="caller_id_number" expression="^(\d+)$" break="on-true">
      <action application="set" data="effective_caller_id_number=$1"/>
    </condition>
  </extension>
  <extension name="oztel_cid">
    <condition field="destination_number" expression="^(12345678)$">
      <action application="set" data="domain_name=$${domain}"/>
      <action application="transfer" data="2000 XML default"/>
    </condition>
  </extension>

-- 
Brian May <brian at microcomaustralia.com.au>



From brian at freeswitch.org  Sun Feb 21 20:43:02 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 21 Feb 2010 22:43:02 -0600
Subject: [Freeswitch-users] altering callerid
In-Reply-To: <3c5cf5261002212037l3576f9d7s33e32c310799aaf7@mail.gmail.com>
References: <3c5cf5261002211852k577d4d78x32e43c0b4d29a1c4@mail.gmail.com>
	<8C9F2C93-DDF3-4283-8015-82062AF794CD@freeswitch.org>
	<3c5cf5261002211934t12f79a5dh5564003af0f4b74d@mail.gmail.com>
	<973F4A98-1461-483A-BB9D-8C1746C7DFE8@freeswitch.org>
	<3c5cf5261002211950t1bbbb34i20e6509ed345a1a5@mail.gmail.com>
	<B93F97C3D14E4BEA833E14D920F5FCFC@cune.pri>
	<69D702BA87A54ABA93281A8DA621BB12@cune.pri>
	<3c5cf5261002212037l3576f9d7s33e32c310799aaf7@mail.gmail.com>
Message-ID: <C7134664-A3C3-4B9A-BCBA-745EE3B1817A@freeswitch.org>

regex lesson:

^61([2-9]\d{8})$

/b


On Feb 21, 2010, at 10:37 PM, Brian May wrote:

> "^61([2-9]\d\d\d\d\d\d\d\d)$"




From brian at microcomaustralia.com.au  Sun Feb 21 20:57:31 2010
From: brian at microcomaustralia.com.au (Brian May)
Date: Mon, 22 Feb 2010 15:57:31 +1100
Subject: [Freeswitch-users] altering callerid
In-Reply-To: <C7134664-A3C3-4B9A-BCBA-745EE3B1817A@freeswitch.org>
References: <3c5cf5261002211852k577d4d78x32e43c0b4d29a1c4@mail.gmail.com>
	<8C9F2C93-DDF3-4283-8015-82062AF794CD@freeswitch.org>
	<3c5cf5261002211934t12f79a5dh5564003af0f4b74d@mail.gmail.com>
	<973F4A98-1461-483A-BB9D-8C1746C7DFE8@freeswitch.org>
	<3c5cf5261002211950t1bbbb34i20e6509ed345a1a5@mail.gmail.com>
	<B93F97C3D14E4BEA833E14D920F5FCFC@cune.pri>
	<69D702BA87A54ABA93281A8DA621BB12@cune.pri>
	<3c5cf5261002212037l3576f9d7s33e32c310799aaf7@mail.gmail.com>
	<C7134664-A3C3-4B9A-BCBA-745EE3B1817A@freeswitch.org>
Message-ID: <3c5cf5261002212057s6fab56a7o403a3e84763d1ad3@mail.gmail.com>

On 22 February 2010 15:43, Brian West <brian at freeswitch.org> wrote:
> regex lesson:
>
> ^61([2-9]\d{8})$

Much better. Thanks for this.
-- 
Brian May <brian at microcomaustralia.com.au>



From feeswitch.ml at hez.ca  Sun Feb 21 12:11:58 2010
From: feeswitch.ml at hez.ca (Hez Ronningen)
Date: Sun, 21 Feb 2010 12:11:58 -0800
Subject: [Freeswitch-users] dingaling module failing to load with gnutls
	error
Message-ID: <C1FC75D9-29E0-4233-8613-2E23551045B6@hez.ca>

Hello,

Installed freeswitch on ubuntu and enabled the dingaling module but when it boots I get the following error

2010-02-21 11:59:55.213568 [CRIT] switch_loadable_module.c:882 Error Loading module /opt/freeswitch/mod/mod_dingaling.so
**/opt/freeswitch/mod/mod_dingaling.so: undefined symbol: gnutls_global_init**

I have the following libraries installed

ii  libgnutls-dev                        2.8.3-2                                                the GNU TLS library - development files
ii  libgnutls26                          2.8.3-2                                                the GNU TLS library - runtime library


Is there a library I am missing or an incompatibility?

I've checked around on the web and the mailing list archives and no one else seems to have run in to this problem.

Any help is much appreciated,
Hez


From rperry at madisonip.com  Sun Feb 21 21:14:21 2010
From: rperry at madisonip.com (Ryan Perry)
Date: Sun, 21 Feb 2010 23:14:21 -0600
Subject: [Freeswitch-users] Using FS with Asterisk as a PBX
Message-ID: <ea35b6851002212114kd90a5b5h8ebde347d7b0feae@mail.gmail.com>

I'm new to FS.  I am trying to get started with implementing a phone system
to manage 12+ small companies.  I'd planned to use Asterisk, but I've come
to understand the problems with it on a larger scale.  My question is will I
avoid potential problems by using FS to manage 12 Asterisk PBXs?  OR is it
advantageous to use FS's PBX abilities?

Thanks for your opinions and expertise.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100221/67ec9dce/attachment-0002.html 

From mike at jerris.com  Sun Feb 21 22:12:45 2010
From: mike at jerris.com (Michael Jerris)
Date: Mon, 22 Feb 2010 01:12:45 -0500
Subject: [Freeswitch-users] groups
In-Reply-To: <3c5cf5261002210050s235142d6o3f8be42146399c57@mail.gmail.com>
References: <3c5cf5261002210050s235142d6o3f8be42146399c57@mail.gmail.com>
Message-ID: <F7769DC5-6497-4031-BF3E-8E69288D3C43@jerris.com>

You can use any endpoints you want in a group.  This is defined by the dial string for each user.

On Feb 21, 2010, at 3:50 AM, Brian May wrote:

> Hello,
> 
> Can I add a analogue FXS port to a group that is then accessed using
> group_call(...)?
> 
> If so, how? I can only see support for SIP members, as defined in
> directory/default.xml




From mike at jerris.com  Sun Feb 21 22:21:35 2010
From: mike at jerris.com (Michael Jerris)
Date: Mon, 22 Feb 2010 01:21:35 -0500
Subject: [Freeswitch-users] altering callerid
In-Reply-To: <b39273531002211946y5361be9cvce49b9aaa8d10d26@mail.gmail.com>
References: <3c5cf5261002211852k577d4d78x32e43c0b4d29a1c4@mail.gmail.com>
	<8C9F2C93-DDF3-4283-8015-82062AF794CD@freeswitch.org>
	<3c5cf5261002211934t12f79a5dh5564003af0f4b74d@mail.gmail.com>
	<b39273531002211946y5361be9cvce49b9aaa8d10d26@mail.gmail.com>
Message-ID: <A8955E1D-72FF-4385-AE73-3240F8492653@jerris.com>

As a note, this method overwrites what is in the caller profile.  The vars are really the right way to do this.  If you ever find yourself actually using set profile var you are almost definitely doing the wrong thing unless you are 100% sure you are not.  This function was originally left completely undocumented because you should not be using it.  If any documentation is added for this other than completely removing documentation, it should be a bold warning that says you should not under any circumstances use this.

Mike

On Feb 21, 2010, at 10:46 PM, Rupa Schomaker wrote:

> caller_id is slightly tricky.  Use the dptool set_profile_var http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_set_profile_var.  Though that is under documented.  
> 
> But really, if you are just bridging then do what Brian said.  That is the "appropriate" method in most cases.
> 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100222/3f981eb1/attachment-0002.html 

From mike at jerris.com  Sun Feb 21 22:32:40 2010
From: mike at jerris.com (Michael Jerris)
Date: Mon, 22 Feb 2010 01:32:40 -0500
Subject: [Freeswitch-users] Using FS with Asterisk as a PBX
In-Reply-To: <ea35b6851002212114kd90a5b5h8ebde347d7b0feae@mail.gmail.com>
References: <ea35b6851002212114kd90a5b5h8ebde347d7b0feae@mail.gmail.com>
Message-ID: <27ED0182-15CC-41AB-BA9D-864C29B11BDD@jerris.com>

I don't understand the question.  FreeSWITCH does not provide any functionality to manage asterisk instances.

Mike

On Feb 22, 2010, at 12:14 AM, Ryan Perry wrote:

> I'm new to FS.  I am trying to get started with implementing a phone system to manage 12+ small companies.  I'd planned to use Asterisk, but I've come to understand the problems with it on a larger scale.  My question is will I avoid potential problems by using FS to manage 12 Asterisk PBXs?  OR is it advantageous to use FS's PBX abilities?




From pmhshz at gmail.com  Sun Feb 21 22:47:20 2010
From: pmhshz at gmail.com (MohammedShehzad)
Date: Mon, 22 Feb 2010 12:17:20 +0530
Subject: [Freeswitch-users] stream a file multicast with mod_esf
In-Reply-To: <795CD2F3-1395-4C08-A0E7-F5A7BA85F938@jerris.com>
References: <B98B6921358D0C49BD21CD023F1F52C302623B4B@hbghs04a.ww300.siemens.net>
	<378CDEC3-5A41-4EF9-927A-311F2531E6AB@freeswitch.org>
	<a792304b1001080725y19ff28camd37425927550f311@mail.gmail.com>
	<191c3a031001081848w9036207yf6a4b473ec233243@mail.gmail.com>
	<85C74329-0D7C-48B2-B7A8-D4AA67C6E3F2@freeswitch.org>
	<a792304b1002182209m43ada13cp8ed3d91ad509f0a2@mail.gmail.com>
	<DBA4E291-9377-489A-841A-1A3D83880DF2@jerris.com>
	<a792304b1002190002u50ec6912g1905ffc63a119174@mail.gmail.com>
	<a792304b1002192150n44cc500fnd5b2ecb7cff9dd1a@mail.gmail.com>
	<795CD2F3-1395-4C08-A0E7-F5A7BA85F938@jerris.com>
Message-ID: <a792304b1002212247q7cc27ecy5e74368acd68b99c@mail.gmail.com>

Yes, PCMU is hardcoded currently from multicaster. I looked into mod_sndfile
for decoding PCMU to other codec, but it seems that module is using
libsndfile, which reads sound file directly and decode them to L16. If
something similar to libsndfile is available, which work on stream instead
of file io, then it would surely work.

I don't know how exactly Freeswitch's codec structures & functions work, I
am sure decoding can be done by using that, but don't know how to use them.

On Sun, Feb 21, 2010 at 7:17 AM, Michael Jerris <mike at jerris.com> wrote:

> You will need to create the codec for what you need, I think it is
> hardcoded in there to PCMU at the moment, correct?  This will of course need
> to match the stream its reading.
>
> Mike
>
> On Feb 20, 2010, at 12:50 AM, MohammedShehzad wrote:
>
>
>
> On Fri, Feb 19, 2010 at 1:32 PM, MohammedShehzad <pmhshz at gmail.com> wrote:
>
>>
>>
>> On Fri, Feb 19, 2010 at 12:59 PM, Michael Jerris <mike at jerris.com> wrote:
>>
>>> Listening on multicast is noting special for multicast, it is just
>>> like reading any other udp socket
>>>
>>> Mike
>>>
>>> Correct, but I have to play those audio stream back to caller taking care
>> of the audio codec  and other things, do anybody have any idea in that part?
>> Please let me know that.
>>  --
>>
>> -MohammedShehzad
>>
>
> I am able to receive the play the multicasted RAW PCMU RTP (modified the
> skel of format provided by brian), so that caller can hear the multicast
> which done by other Freeswitch server using mod_esf application, but when i
> change the caller's codec from PCMU to something else, it breaks.
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 

-MohammedShehzad
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100222/99c3fef0/attachment-0002.html 

From mike at jerris.com  Sun Feb 21 23:19:23 2010
From: mike at jerris.com (Michael Jerris)
Date: Mon, 22 Feb 2010 02:19:23 -0500
Subject: [Freeswitch-users] stream a file multicast with mod_esf
In-Reply-To: <a792304b1002212247q7cc27ecy5e74368acd68b99c@mail.gmail.com>
References: <B98B6921358D0C49BD21CD023F1F52C302623B4B@hbghs04a.ww300.siemens.net>
	<378CDEC3-5A41-4EF9-927A-311F2531E6AB@freeswitch.org>
	<a792304b1001080725y19ff28camd37425927550f311@mail.gmail.com>
	<191c3a031001081848w9036207yf6a4b473ec233243@mail.gmail.com>
	<85C74329-0D7C-48B2-B7A8-D4AA67C6E3F2@freeswitch.org>
	<a792304b1002182209m43ada13cp8ed3d91ad509f0a2@mail.gmail.com>
	<DBA4E291-9377-489A-841A-1A3D83880DF2@jerris.com>
	<a792304b1002190002u50ec6912g1905ffc63a119174@mail.gmail.com>
	<a792304b1002192150n44cc500fnd5b2ecb7cff9dd1a@mail.gmail.com>
	<795CD2F3-1395-4C08-A0E7-F5A7BA85F938@jerris.com>
	<a792304b1002212247q7cc27ecy5e74368acd68b99c@mail.gmail.com>
Message-ID: <8A3C6D64-9215-47A5-8FCD-7A328770772D@jerris.com>

You would just change the PCMU to whatever codec you want.  This should have nothing to do with file io, take a look at the line that has PCMU hardcoded, thats all you should need to change.

Mike

On Feb 22, 2010, at 1:47 AM, MohammedShehzad wrote:

> Yes, PCMU is hardcoded currently from multicaster. I looked into mod_sndfile for decoding PCMU to other codec, but it seems that module is using libsndfile, which reads sound file directly and decode them to L16. If something similar to libsndfile is available, which work on stream instead of file io, then it would surely work.
> 
> I don't know how exactly Freeswitch's codec structures & functions work, I am sure decoding can be done by using that, but don't know how to use them.
> 
> On Sun, Feb 21, 2010 at 7:17 AM, Michael Jerris <mike at jerris.com> wrote:
> You will need to create the codec for what you need, I think it is hardcoded in there to PCMU at the moment, correct?  This will of course need to match the stream its reading.
> 
> Mike
> 
> On Feb 20, 2010, at 12:50 AM, MohammedShehzad wrote:
> 
>> 
>> 
>> On Fri, Feb 19, 2010 at 1:32 PM, MohammedShehzad <pmhshz at gmail.com> wrote:
>> 
>> 
>> On Fri, Feb 19, 2010 at 12:59 PM, Michael Jerris <mike at jerris.com> wrote:
>> Listening on multicast is noting special for multicast, it is just
>> like reading any other udp socket
>> 
>> Mike
>> 
>> Correct, but I have to play those audio stream back to caller taking care of the audio codec  and other things, do anybody have any idea in that part? Please let me know that.
>> -- 
>> 
>> -MohammedShehzad
>> 
>> I am able to receive the play the multicasted RAW PCMU RTP (modified the skel of format provided by brian), so that caller can hear the multicast which done by other Freeswitch server using mod_esf application, but when i change the caller's codec from PCMU to something else, it breaks.
> 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100222/7489f43d/attachment-0002.html 

From gamar at center.com  Mon Feb 22 00:52:08 2010
From: gamar at center.com (Gilbert Amar)
Date: Mon, 22 Feb 2010 09:52:08 +0100
Subject: [Freeswitch-users] Freeswitch and mod_opal
Message-ID: <4A424236C3C44A8FBED67E818845BCAD@gamar>

Hello,


I try Freeswitch and mod_opal on CenTos and on Windows XP
Calling FS IVR from Openphone or a regular H323 phone works
But on those two platforms I could not bridge two calls using h323. There is 
always a mute or deaf leg.
I also try to build mod_h323 with no success.
Did anyone have tried this on the svn trunk and succeeded.
If yes I will be glad to know how you did and what parameters you choose 
regarding faststart, h245 tunneling, codecs, etc.


Gilbert




From helmut.kuper at ewetel.de  Mon Feb 22 01:09:19 2010
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Mon, 22 Feb 2010 10:09:19 +0100
Subject: [Freeswitch-users] Question about sofia_contact
In-Reply-To: <4B7EADBC.1040001@ewetel.de>
References: <4B7EA954.30402@ewetel.de> <4B7EADBC.1040001@ewetel.de>
Message-ID: <4B8249BF.3090708@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hello,

has anybody an idea?

regards
helmut


On 19.02.2010 16:26, Helmut Kuper wrote:
> Hi,
> 
> an update:
> The corresponding select statement looks for sip_user="2701" and
> sip_host="internal" in registration table.
> This delivers of course no result because 2701 is registered with
> sip_host="mydomain".
> 
> 
> Hm any workaround or am I going in a wrong direction?
> 
> 
> regards
> Helmut
> 
> 
> On 19.02.2010 16:08, Helmut Kuper wrote:
>> Hello,
> 
>> I try to setup a FS sofia sip-profile which allows me to have multiple
>> sip-profiles but one registration database. So I set the following
>> parameters:
> 
>> <param name="challenge-realm" value="auto_from"/>
>> <param name="force-register-domain" value="$${domain}"/>
>> <param name="force-subscription-domain" value="$${domain}"/>
>> <param name="force-register-db-domain" value="$${domain}"/>
> 
>> where domain is set to "mydomain". "sofia status profile internal"
>> delivers the following:
> 
> 
>> Call-ID:        3c26705038e5-vwlg8u5q9cwe
>> User:           2701 at mydomain
>> Contact:        <sip:2701 at 85.16.245.208:1024;line=l2wksedo>
>> Agent:          snom370/8.2.22
>> Status:         Registered(UDP)(unknown) EXP(2010-02-19 16:13:31)
>> Host:           ippbx-prod-node0
>> IP:             85.16.245.208
>> Port:           1024
>> Auth-User:      2701
>> Auth-Realm:     mydomain
>> MWI-Account:    2701 at mydomain
> 
> 
> 
>> sofia_contact internal/2701 at mydomain delivers this:
>> error/user_not_registered
> 
>> The Phone is fully functional.
> 
>> I use SVN trunk 16601
> 
>> regards
>> Helmut

-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.4.7 (MingW32)

iD8DBQFLgkm/4tZeNddg3dwRApfoAKCiX8fX/WNrZ7GXRrBJA54+VTThmACfT0d3
fBzQlyVObkJaLHxJbfUjZG4=
=uhHR
-----END PGP SIGNATURE-----



From oseslija at gmail.com  Mon Feb 22 01:12:47 2010
From: oseslija at gmail.com (Ognjen Seslija)
Date: Mon, 22 Feb 2010 10:12:47 +0100
Subject: [Freeswitch-users] Using FS with Asterisk as a PBX
In-Reply-To: <ea35b6851002212114kd90a5b5h8ebde347d7b0feae@mail.gmail.com>
References: <ea35b6851002212114kd90a5b5h8ebde347d7b0feae@mail.gmail.com>
Message-ID: <4468a6771002220112k2b656607g4638602582fb5cc1@mail.gmail.com>

You can safely replace those asterisks with a single FS configured for
multitenant/multidomain PBXes.

Ognjen

On Mon, Feb 22, 2010 at 6:14 AM, Ryan Perry <rperry at madisonip.com> wrote:

> I'm new to FS.  I am trying to get started with implementing a phone system
> to manage 12+ small companies.  I'd planned to use Asterisk, but I've come
> to understand the problems with it on a larger scale.  My question is will I
> avoid potential problems by using FS to manage 12 Asterisk PBXs?  OR is it
> advantageous to use FS's PBX abilities?
>
> Thanks for your opinions and expertise.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100222/98b545b5/attachment-0002.html 

From shaheryarkh at googlemail.com  Mon Feb 22 02:00:21 2010
From: shaheryarkh at googlemail.com (Muhammad Shahzad)
Date: Mon, 22 Feb 2010 15:00:21 +0500
Subject: [Freeswitch-users] Doxygen help
In-Reply-To: <4B820145.2090109@cartissolutions.com>
References: <4B820145.2090109@cartissolutions.com>
Message-ID: <b16156851002220200u3736acbdre1a1b82db0e82651@mail.gmail.com>

Hi,

Its a quite embarrassing to admit that i could completed that work, not even
able to submit what was done to FS trunk due to my own stupidity. Actually,
we are having major power crisis in the country for last 1 years and most of
power cuts are so long that even best UPS systems can't ensure 24/7 up time
for our servers. About 3 months ago the development server i was working on
for FS Docs was hit by this problem and its hard disk crashed, destroying
all my hard work on FS Docs and mod_msn. It was my mistake that i didn't
arrange for proper backup for my work due to shortage of resources and
didn't inform the FS developers community about the loss.

I want to restart this work but don't want to take the lead on this due to
issues described above. If you can arrange a common server for FS docs
development then i can submit my work to you which you can test on this dev
server before committing it to FS Trunk.

Second option is that i work on a single file, test it on my laptop and
immediately submit it to FS trunk. But this could cause problem for end
users (developers using FS doxygen docs) as many hyper links connected to
undocumented files won't work till their docs are done and uploaded.

We also need to decide on,

1. What Doxygen version to use? i use Archlinux which has the latest 1.6.x
version, I can downgrade it to 1.4.x.
2. What format of documentation to adopt? I think currently we have HTML
format only, but Doxygen can also generate PDF and CHM formats.

Thank you.


On Mon, Feb 22, 2010 at 9:00 AM, Yossi Neiman <
freeswitch at cartissolutions.com> wrote:

> I am not sure how many folks make use of the Doxygen documentation.  I
> know I do all the time.  I find that it provides a nice conceptual view
> of FreeSWITCH's API, which can make it very easy to find the functions
> and data types needed for writing modules for FreeSWITCH.
>
> A person by the name of Mohammad Shahzad (apologies if I misspelled it)
> started to do a lot of work on revamping the Doxygen configuration for
> FreeSWITCH a few months back.  The problem with the work that he did was
> that he used Doxygen 1.6.x specific configuration parameters that are
> not understood by the Doxygen 1.4.x tree (which is what we currently use
> in FreeSWITCH) or even the 1.5.x tree which is what is in Slackware.  In
> my discussions with Michael Collins we have decided that it might not
> hurt to go ahead and move the Doxygen version forward to match the 1.6.x
> tree.
>
> We are looking for somebody who has interest in working with the Doxygen
> configuration to continue the work that Mohammad had started and to help
> the project out.  If you are such a person, please contact me off-list
> and I can provide further information.
>
> Thanks!
>
> Yossi Neiman
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Muhammad Shahzad
-----------------------------------
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +92 334 422 40 88
MSN: shari_786pk at hotmail.com
Email: shaheryarkh at googlemail.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100222/c7b55b40/attachment-0002.html 

From pmhshz at gmail.com  Mon Feb 22 03:56:55 2010
From: pmhshz at gmail.com (MohammedShehzad)
Date: Mon, 22 Feb 2010 17:26:55 +0530
Subject: [Freeswitch-users] stream a file multicast with mod_esf
In-Reply-To: <8A3C6D64-9215-47A5-8FCD-7A328770772D@jerris.com>
References: <B98B6921358D0C49BD21CD023F1F52C302623B4B@hbghs04a.ww300.siemens.net>
	<191c3a031001081848w9036207yf6a4b473ec233243@mail.gmail.com>
	<85C74329-0D7C-48B2-B7A8-D4AA67C6E3F2@freeswitch.org>
	<a792304b1002182209m43ada13cp8ed3d91ad509f0a2@mail.gmail.com>
	<DBA4E291-9377-489A-841A-1A3D83880DF2@jerris.com>
	<a792304b1002190002u50ec6912g1905ffc63a119174@mail.gmail.com>
	<a792304b1002192150n44cc500fnd5b2ecb7cff9dd1a@mail.gmail.com>
	<795CD2F3-1395-4C08-A0E7-F5A7BA85F938@jerris.com>
	<a792304b1002212247q7cc27ecy5e74368acd68b99c@mail.gmail.com>
	<8A3C6D64-9215-47A5-8FCD-7A328770772D@jerris.com>
Message-ID: <a792304b1002220356t18b1fb5exfdeb02a75e425976@mail.gmail.com>

Actually I don't want to change anything from Multicaster, I am talking
about the changes required on Listener side (the format module I am going to
develo). I think I should discuss further on developer's mailing list,
please let me know you ideas there.

Thanks for your response.

On Mon, Feb 22, 2010 at 12:49 PM, Michael Jerris <mike at jerris.com> wrote:

> You would just change the PCMU to whatever codec you want.  This should
> have nothing to do with file io, take a look at the line that has PCMU
> hardcoded, thats all you should need to change.
>
> Mike
>
> On Feb 22, 2010, at 1:47 AM, MohammedShehzad wrote:
>
> Yes, PCMU is hardcoded currently from multicaster. I looked into
> mod_sndfile for decoding PCMU to other codec, but it seems that module is
> using libsndfile, which reads sound file directly and decode them to L16. If
> something similar to libsndfile is available, which work on stream instead
> of file io, then it would surely work.
>
> I don't know how exactly Freeswitch's codec structures & functions work, I
> am sure decoding can be done by using that, but don't know how to use them.
>
> On Sun, Feb 21, 2010 at 7:17 AM, Michael Jerris <mike at jerris.com> wrote:
>
>> You will need to create the codec for what you need, I think it is
>> hardcoded in there to PCMU at the moment, correct?  This will of course need
>> to match the stream its reading.
>>
>> Mike
>>
>> On Feb 20, 2010, at 12:50 AM, MohammedShehzad wrote:
>>
>>
>>
>> On Fri, Feb 19, 2010 at 1:32 PM, MohammedShehzad <pmhshz at gmail.com>wrote:
>>
>>>
>>>
>>> On Fri, Feb 19, 2010 at 12:59 PM, Michael Jerris <mike at jerris.com>wrote:
>>>
>>>> Listening on multicast is noting special for multicast, it is just
>>>> like reading any other udp socket
>>>>
>>>> Mike
>>>>
>>>> Correct, but I have to play those audio stream back to caller taking
>>> care of the audio codec  and other things, do anybody have any idea in that
>>> part? Please let me know that.
>>>  --
>>>
>>> -MohammedShehzad
>>>
>>
>> I am able to receive the play the multicasted RAW PCMU RTP (modified the
>> skel of format provided by brian), so that caller can hear the multicast
>> which done by other Freeswitch server using mod_esf application, but when i
>> change the caller's codec from PCMU to something else, it breaks.
>>
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 

-MohammedShehzad
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100222/7da0738f/attachment-0002.html 

From tculjaga at gmail.com  Mon Feb 22 04:33:34 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Mon, 22 Feb 2010 13:33:34 +0100
Subject: [Freeswitch-users] Freeswitch and mod_opal
In-Reply-To: <4A424236C3C44A8FBED67E818845BCAD@gamar>
References: <4A424236C3C44A8FBED67E818845BCAD@gamar>
Message-ID: <65d96fc81002220433j25e428d9m2ddff5c0b40ef344@mail.gmail.com>

mod_h323

you need to build h323plus & ptlib before you build the module itself.

check http://wiki.freeswitch.org/wiki/Mod_h323 it works out of the box.


regarding you call setup, faststart = true, h245tuneling = true, h245InSetup
= false

It has to work.... if not post the logs here.

T.



On Mon, Feb 22, 2010 at 9:52 AM, Gilbert Amar <gamar at center.com> wrote:

> Hello,
>
>
> I try Freeswitch and mod_opal on CenTos and on Windows XP
> Calling FS IVR from Openphone or a regular H323 phone works
> But on those two platforms I could not bridge two calls using h323. There
> is
> always a mute or deaf leg.
> I also try to build mod_h323 with no success.
> Did anyone have tried this on the svn trunk and succeeded.
> If yes I will be glad to know how you did and what parameters you choose
> regarding faststart, h245 tunneling, codecs, etc.
>
>
> Gilbert
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100222/1970fca9/attachment-0002.html 

From ivdreg at gmail.com  Mon Feb 22 04:48:57 2010
From: ivdreg at gmail.com (ivdreg ivdreg)
Date: Mon, 22 Feb 2010 14:48:57 +0200
Subject: [Freeswitch-users] SDP With T.38 in INVITE Problem
In-Reply-To: <b437a0e1002190242y2aa64696j91e8bc76f7ae2a92@mail.gmail.com>
References: <b437a0e1002190242y2aa64696j91e8bc76f7ae2a92@mail.gmail.com>
Message-ID: <b437a0e1002220448g3e3bda9fpa283e93a11d319f1@mail.gmail.com>

Hi All,

Actually while seeking the solution in internet I see some people having
this problem with sofia library. I'm not sure that SIP reply in this case
contains a valid SDP (I think that teminating endpoint is broken) but in my
opinion if we have at least one valid media type in SDP (video, audio, image
...) call must be established. Can someone comment and/or help me with this
issue.

Regards.

2010/2/19 ivdreg ivdreg <ivdreg at gmail.com>

> Hi all,
>
> Dose someone have a problem that if there T.38 in coming from gateway
> FreeSwitch drops the call because of media error ? As I see from log only
> T.38 port is zero and SDP has also media port. Is it possible to configure
> FS to do not break a call but if media is OK.
>
> 2010-02-19 12:26:14.305244 [NOTICE] switch_channel.c:666 New Channel
> sofia/backend/XXXXXXXXXX at 10.10.1.110:7065[6cd9f634-411d-df11-99ca-003048bb99cc]
> 2010-02-19 12:26:14.305244 [DEBUG] mod_sofia.c:3369 (sofia/backend/
> XXXXXXXXXX at 10.10.1.110:7065) State Change CS_NEW -> CS_INIT
> 2010-02-19 12:26:14.305244 [DEBUG] switch_core_session.c:1018 Send signal
> sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [BREAK]
> 2010-02-19 12:26:14.305244 [DEBUG] switch_core_state_machine.c:314
> (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) Running State Change CS_INIT
> 2010-02-19 12:26:14.305244 [DEBUG] switch_core_state_machine.c:338
> (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State INIT
> 2010-02-19 12:26:14.305244 [DEBUG] mod_sofia.c:83 sofia/backend/
> XXXXXXXXXX at 10.10.1.110:7065 SOFIA INIT
> 2010-02-19 12:26:14.305244 [DEBUG] sofia_glue.c:1377 sofia/backend/
> XXXXXXXXXX at 10.10.1.110:7065 Patched SDP
> ---
> v=0
> o=- 3779949069 5166785777234550622 IN IP4 206.132.232.212
> s=session
> t=0 0
> m=audio 21108 RTP/AVP 18 4 8 0
> c=IN IP4 10.10.1.110
> a=rtpmap:18 G729/8000
> a=rtpmap:4 G723/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:0 PCMU/8000
> m=image 21108 udptl t38
> c=IN IP4 10.10.1.110
> a=T38FaxVersion:0
> a=T38MaxBitRate:14400
> a=T38FaxUdpEC:t38UDPRedundancy
> a=T38FaxRateManagement:transferredTCF
>
> +++
> v=0
> o=- 3779949069 5166785777234550622 IN IP4 206.132.232.212
> s=session
> t=0 0
> m=audio 17058 RTP/AVP 18 4 8 0
> c=IN IP4 10.10.1.100
> a=rtpmap:18 G729/8000
> a=rtpmap:4 G723/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:0 PCMU/8000
> m=image 17058 udptl t38
> c=IN IP4 10.10.1.100
> a=T38FaxVersion:0
> a=T38MaxBitRate:14400
> a=T38FaxUdpEC:t38UDPRedundancy
> a=T38FaxRateManagement:transferredTCF
>
> 2010-02-19 12:26:14.305244 [DEBUG] mod_sofia.c:117 (sofia/backend/
> XXXXXXXXXX at 10.10.1.110:7065) State Change CS_INIT -> CS_ROUTING
> ......
> 2010-02-19 12:26:21.255201 [DEBUG] sofia.c:4135 Remote SDP:
> v=0
> o=FreeSWITCH 1266548331 1266548332 IN IP4 10.10.1.110
> s=FreeSWITCH
> c=IN IP4 10.10.1.110
> t=0 0
> *m=audio 26850 RTP/AVP 8*
> a=rtpmap:8 PCMA/8000
> a=silenceSupp:off - - - -
> a=ptime:20
> *m=image 0 udptl 19*
>
> 2010-02-19 12:26:21.255201 [DEBUG] sofia.c:4124 Channel sofia/backend/
> XXXXXXXXXX at 10.10.1.110:7065 entering state [ready][200]
> 2010-02-19 12:26:21.255201 [DEBUG] switch_channel.c:2285 Send signal
> sofia/backend/YYYYYYYYYY at 10.10.1.110 [BREAK]
> 2010-02-19 12:26:21.255201 [NOTICE] sofia.c:4668 Channel [sofia/backend/
> XXXXXXXXXX at 10.10.1.110:7065] has been answered
> 2010-02-19 12:26:21.255201 [DEBUG] sofia_glue.c:2288 Set Codec
> sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 PROXY/8000 20 ms 160 samples
> *2010-02-19 12:26:21.255201 [DEBUG] sofia_glue.c:2571 PROXY AUDIO RTP
> [sofia/backend/XXXXXXXXXX at 10.10.1.110:7065] 10.10.1.100:17058->
> 10.10.1.110:0 codec: 0 ms: 20
> 2010-02-19 12:26:21.255201 [ERR] sofia_glue.c:2879 AUDIO RTP REPORTS ERROR:
> [Missing remote port]
> 2010-02-19 12:26:21.255201 [NOTICE] sofia_glue.c:2880 Hangup sofia/backend/
> XXXXXXXXXX at 10.10.1.110:7065 [CS_CONSUME_MEDIA] [DESTINATION_OUT_OF_ORDER]*
> 2010-02-19 12:26:21.255201 [DEBUG] switch_channel.c:2063 Send signal
> sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [KILL]
> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_session.c:1018 Send signal
> sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [BREAK]
> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:314
> (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) Running State Change CS_HANGUP
> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:499
> (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State HANGUP
> 2010-02-19 12:26:21.255201 [DEBUG] mod_sofia.c:411 Channel sofia/backend/
> XXXXXXXXXX at 10.10.1.110:7065 hanging up, cause: DESTINATION_OUT_OF_ORDER
> 2010-02-19 12:26:21.255201 [DEBUG] mod_sofia.c:454 Sending BYE to
> sofia/backend/XXXXXXXXXX at 10.10.1.110:7065
> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:46
> sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 Standard HANGUP, cause:
> DESTINATION_OUT_OF_ORDER
> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:499
> (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State HANGUP going to sleep
> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:333
> (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State Change CS_HANGUP ->
> CS_REPORTING
> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_session.c:1018 Send signal
> sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [BREAK]
> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:314
> (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) Running State Change
> CS_REPORTING
> 2010-02-19 12:26:21.255201 [DEBUG] switch_ivr_originate.c:3185 Originate
> Resulted in Error Cause: 27 [DESTINATION_OUT_OF_ORDER]
> 2010-02-19 12:26:21.255201 [INFO] mod_dptools.c:2353 Originate Failed.
> Cause: DESTINATION_OUT_OF_ORDER
> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:590
> (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State REPORTING
>
>
> Thanks
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100222/38410808/attachment-0002.html 

From dftoro at yahoo.com  Mon Feb 22 05:11:29 2010
From: dftoro at yahoo.com (Diego Toro)
Date: Mon, 22 Feb 2010 05:11:29 -0800 (PST)
Subject: [Freeswitch-users] Dialplan question
In-Reply-To: <2dac3814a041c79208af8a433b279566.squirrel@www.webcontracts.co.uk>
Message-ID: <240053.46379.qm@web33504.mail.mud.yahoo.com>

Hi, check <extension name="Local_Extension"> on conf/dialplan/default.xml, You can get an idea of how to do what you need.


Diego Toro
http://lacarretade.blogspot.com/


--- On Sun, 2/21/10, Matthew Law <matt at webcontracts.co.uk> wrote:

> From: Matthew Law <matt at webcontracts.co.uk>
> Subject: [Freeswitch-users] Dialplan question
> To: freeswitch-users at lists.freeswitch.org
> Date: Sunday, February 21, 2010, 6:26 PM
> I have FS installed and I can make
> outgoing calls through my SIP provider.
>  I can also call other extensions (FS is running on a small
> Xen domU on
> the internet), but I am having problems getting the
> dialplan for incoming
> calls to work.
> 
> What I want to do is have incoming calls on my number ring
> all extensions,
> e.g. 1000 - 1005 for 10 seconds and then go to voicemail
> for extension
> 1000.? If there are no logged-on users, then it should
> go straight to
> voicemail.? Rather than bite off too much, I thought I
> would try and get a
> very basic setup working and take it from there...
> 
> At the moment it goes straight to voicemail for extension
> 1000 even if
> 1000 is logged in.? Here are the dialplan files I have
> (everything else is
> default from the trunk install):
> 
> dialplan/public/00_inbound_did.xml:
> 
> <include>
> ? ? <extension name="08433xxxxxxx">
> ? ? ? ? <condition
> field="destination_number" expression="^(0843xxxxxxx)$">
> ? ? ? ? ? ? <action
> application="transfer" data="$1 XML default"/>
> ? ? ? ? </condition>
> ? ? </extension>
> </include>
> 
> dialplan/default/12_voiptalk.xml:
> 
> <include>
> ? <extension name="public_did">
> ? ? <condition field="destination_number"
> expression="^(0843xxxxxxx)$">
> ? ? ? <action application="transfer"
> data="1000 XML default"/>
> ? ? </condition>
> ? </extension>
> ? <extension name="outgoing">
> ? ? <condition field="destination_number"
> expression="^9([0-9]+)$">
> ? ? ? <action application="bridge"
> data="sofia/gateway/voiptalk/$1" />
> ? ? </condition>
> ? </extension>
> </include>
> 
> I would be very grateful if someone could tell me where I
> am going wrong. 
> I've been looking at various FS wiki pages for hours as
> well as the
> example configs and can't seem to make any headway.?
> My other question is
> what command should I be run after changing the dialplan?
> is it just
> 'reloadxml'?
> 
> 
> Many thanks,
> 
> Matt.
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 


      



From gamar at center.com  Mon Feb 22 05:30:22 2010
From: gamar at center.com (Gilbert Amar)
Date: Mon, 22 Feb 2010 14:30:22 +0100
Subject: [Freeswitch-users] Freeswitch and mod_opal
Message-ID: <BDB26458EDF74CF6A382375739AF86BD@gamar>

Thank you Tihomir


We tried to build the svn trunk of mod_h323 and succeded but got a core dump
a the first call hanging up.

>mod_h323
>you need to build h323plus & ptlib before you build the module itself.
>check http://wiki.freeswitch.org/wiki/Mod_h323 it works out of the box.
>regarding you call setup, faststart = true, h245tuneling = true,
>h245InSetup = false
>It has to work.... if not post the logs here.

>T.




From matt at webcontracts.co.uk  Mon Feb 22 06:16:48 2010
From: matt at webcontracts.co.uk (Matthew Law)
Date: Mon, 22 Feb 2010 14:16:48 -0000
Subject: [Freeswitch-users] Dialplan question
In-Reply-To: <240053.46379.qm@web33504.mail.mud.yahoo.com>
References: <240053.46379.qm@web33504.mail.mud.yahoo.com>
Message-ID: <a8f8fa302ad239a2d6c8332838785448.squirrel@www.webcontracts.co.uk>

On Mon, February 22, 2010 1:11 pm, Diego Toro wrote:
> Hi, check <extension name="Local_Extension"> on conf/dialplan/default.xml,
> You can get an idea of how to do what you need.
>
>
> Diego Toro
> http://lacarretade.blogspot.com/

Thank you all.

It seems it was working in it's current configuration, I just had to
figure out how to get FS to reload the internal config.

Matt.






From mike at jerris.com  Mon Feb 22 06:45:08 2010
From: mike at jerris.com (Michael Jerris)
Date: Mon, 22 Feb 2010 09:45:08 -0500
Subject: [Freeswitch-users] SDP With T.38 in INVITE Problem
In-Reply-To: <b437a0e1002220448g3e3bda9fpa283e93a11d319f1@mail.gmail.com>
References: <b437a0e1002190242y2aa64696j91e8bc76f7ae2a92@mail.gmail.com>
	<b437a0e1002220448g3e3bda9fpa283e93a11d319f1@mail.gmail.com>
Message-ID: <24736FFE-BEEF-4D34-BBA7-77B6B6AE0EBD@jerris.com>

If the endpoint does not correctly follow the sdp o/a model its not going to work.  This is not a "problem" with the sofia library, this is intended behavior and what we are supposed to do.  What is the device?

Mike

On Feb 22, 2010, at 7:48 AM, ivdreg ivdreg wrote:

> Hi All,
> 
> Actually while seeking the solution in internet I see some people having this problem with sofia library. I'm not sure that SIP reply in this case contains a valid SDP (I think that teminating endpoint is broken) but in my opinion if we have at least one valid media type in SDP (video, audio, image ...) call must be established. Can someone comment and/or help me with this issue.
> 
> Regards.
> 
> 2010/2/19 ivdreg ivdreg <ivdreg at gmail.com>
> Hi all,
> 
> Dose someone have a problem that if there T.38 in coming from gateway FreeSwitch drops the call because of media error ? As I see from log only T.38 port is zero and SDP has also media port. Is it possible to configure FS to do not break a call but if media is OK.
> 
> 2010-02-19 12:26:14.305244 [NOTICE] switch_channel.c:666 New Channel sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [6cd9f634-411d-df11-99ca-003048bb99cc]
> 2010-02-19 12:26:14.305244 [DEBUG] mod_sofia.c:3369 (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State Change CS_NEW -> CS_INIT
> 2010-02-19 12:26:14.305244 [DEBUG] switch_core_session.c:1018 Send signal sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [BREAK]
> 2010-02-19 12:26:14.305244 [DEBUG] switch_core_state_machine.c:314 (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) Running State Change CS_INIT
> 2010-02-19 12:26:14.305244 [DEBUG] switch_core_state_machine.c:338 (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State INIT
> 2010-02-19 12:26:14.305244 [DEBUG] mod_sofia.c:83 sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 SOFIA INIT
> 2010-02-19 12:26:14.305244 [DEBUG] sofia_glue.c:1377 sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 Patched SDP
> ---
> v=0
> o=- 3779949069 5166785777234550622 IN IP4 206.132.232.212
> s=session
> t=0 0
> m=audio 21108 RTP/AVP 18 4 8 0
> c=IN IP4 10.10.1.110
> a=rtpmap:18 G729/8000
> a=rtpmap:4 G723/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:0 PCMU/8000
> m=image 21108 udptl t38
> c=IN IP4 10.10.1.110
> a=T38FaxVersion:0
> a=T38MaxBitRate:14400
> a=T38FaxUdpEC:t38UDPRedundancy
> a=T38FaxRateManagement:transferredTCF
> 
> +++
> v=0
> o=- 3779949069 5166785777234550622 IN IP4 206.132.232.212
> s=session
> t=0 0
> m=audio 17058 RTP/AVP 18 4 8 0
> c=IN IP4 10.10.1.100
> a=rtpmap:18 G729/8000
> a=rtpmap:4 G723/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:0 PCMU/8000
> m=image 17058 udptl t38
> c=IN IP4 10.10.1.100
> a=T38FaxVersion:0
> a=T38MaxBitRate:14400
> a=T38FaxUdpEC:t38UDPRedundancy
> a=T38FaxRateManagement:transferredTCF
> 
> 2010-02-19 12:26:14.305244 [DEBUG] mod_sofia.c:117 (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State Change CS_INIT -> CS_ROUTING
> ......
> 2010-02-19 12:26:21.255201 [DEBUG] sofia.c:4135 Remote SDP:
> v=0
> o=FreeSWITCH 1266548331 1266548332 IN IP4 10.10.1.110
> s=FreeSWITCH
> c=IN IP4 10.10.1.110
> t=0 0
> m=audio 26850 RTP/AVP 8
> a=rtpmap:8 PCMA/8000
> a=silenceSupp:off - - - -
> a=ptime:20
> m=image 0 udptl 19
> 
> 2010-02-19 12:26:21.255201 [DEBUG] sofia.c:4124 Channel sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 entering state [ready][200]
> 2010-02-19 12:26:21.255201 [DEBUG] switch_channel.c:2285 Send signal sofia/backend/YYYYYYYYYY at 10.10.1.110 [BREAK]
> 2010-02-19 12:26:21.255201 [NOTICE] sofia.c:4668 Channel [sofia/backend/XXXXXXXXXX at 10.10.1.110:7065] has been answered
> 2010-02-19 12:26:21.255201 [DEBUG] sofia_glue.c:2288 Set Codec sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 PROXY/8000 20 ms 160 samples
> 2010-02-19 12:26:21.255201 [DEBUG] sofia_glue.c:2571 PROXY AUDIO RTP [sofia/backend/XXXXXXXXXX at 10.10.1.110:7065] 10.10.1.100:17058->10.10.1.110:0 codec: 0 ms: 20
> 2010-02-19 12:26:21.255201 [ERR] sofia_glue.c:2879 AUDIO RTP REPORTS ERROR: [Missing remote port]
> 2010-02-19 12:26:21.255201 [NOTICE] sofia_glue.c:2880 Hangup sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [CS_CONSUME_MEDIA] [DESTINATION_OUT_OF_ORDER]
> 2010-02-19 12:26:21.255201 [DEBUG] switch_channel.c:2063 Send signal sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [KILL]
> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_session.c:1018 Send signal sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [BREAK]
> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:314 (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) Running State Change CS_HANGUP
> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:499 (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State HANGUP
> 2010-02-19 12:26:21.255201 [DEBUG] mod_sofia.c:411 Channel sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 hanging up, cause: DESTINATION_OUT_OF_ORDER
> 2010-02-19 12:26:21.255201 [DEBUG] mod_sofia.c:454 Sending BYE to sofia/backend/XXXXXXXXXX at 10.10.1.110:7065
> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:46 sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 Standard HANGUP, cause: DESTINATION_OUT_OF_ORDER
> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:499 (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State HANGUP going to sleep
> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:333 (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State Change CS_HANGUP -> CS_REPORTING
> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_session.c:1018 Send signal sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [BREAK]
> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:314 (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) Running State Change CS_REPORTING
> 2010-02-19 12:26:21.255201 [DEBUG] switch_ivr_originate.c:3185 Originate Resulted in Error Cause: 27 [DESTINATION_OUT_OF_ORDER]
> 2010-02-19 12:26:21.255201 [INFO] mod_dptools.c:2353 Originate Failed.  Cause: DESTINATION_OUT_OF_ORDER
> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:590 (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State REPORTING
> 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100222/10e660e2/attachment-0002.html 

From phunk000 at hotmail.com  Mon Feb 22 06:19:13 2010
From: phunk000 at hotmail.com (phunk000)
Date: Mon, 22 Feb 2010 06:19:13 -0800 (PST)
Subject: [Freeswitch-users] mod_nibblebill
Message-ID: <1266848353835-4612298.post@n2.nabble.com>


Hello there!  I am using the fusionPBX freeSWITCH installation.  I am
attempting to setup the mod_nibblebill module to keep track of billing using
a database.  I have installed ODBC, mod_spidermonkey, and nibblebill appears
to be working, except for the blank line after "last successful billing time
was____" based on the freeSWITCH log as follows:


2010-02-22 08:46:55.535673 [DEBUG] switch_ivr_bridge.c:131
sofia/internal/sip:3008 at 192.168.15.176:42866 receive message [DISPLAY]
2010-02-22 08:46:55.555674 [DEBUG] switch_ivr_bridge.c:131
sofia/internal/3007 at 192.168.15.177 receive message [DISPLAY]
2010-02-22 08:47:23.735442 [DEBUG] mod_nibblebill.c:529 Received request via
SESSION_HEARTBEAT!
2010-02-22 08:47:23.735442 [DEBUG] mod_nibblebill.c:397 Attempting to bill
at $1.0 per minute to account 3007
2010-02-22 08:47:23.735442 [INFO] mod_nibblebill.c:447 Beginning new billing
on 68cde83c-26f4-44e3-8060-4188a106ff51
2010-02-22 08:47:23.735442 [DEBUG] mod_nibblebill.c:455 28 seconds passed
since last bill time of 2010-02-22 08:46:55
2010-02-22 08:47:23.735442 [DEBUG] mod_nibblebill.c:461 Billing $0.474338 to
3007 (Call: 68cde83c-26f4-44e3-8060-4188a106ff51 / 0.000000 so far)
2010-02-22 08:47:42.701637 [DEBUG] switch_core_sqldb.c:111 Dropping idle DB
connection db="core";user="";pass="";thread="3070618512"
2010-02-22 08:47:42.701637 [DEBUG] switch_core_sqldb.c:111 Dropping idle DB
connection db="sofia_reg_internal-ipv6";user="";pass="";thread="3070618512"
2010-02-22 08:47:42.701637 [DEBUG] switch_core_sqldb.c:111 Dropping idle DB
connection db="sofia_reg_external";user="";pass="";thread="3070372752"
2010-02-22 08:47:53.875324 [DEBUG] mod_nibblebill.c:529 Received request via
SESSION_HEARTBEAT!
2010-02-22 08:47:53.875324 [DEBUG] mod_nibblebill.c:397 Attempting to bill
at $1.0 per minute to account 3007
2010-02-22 08:47:53.875324 [DEBUG] mod_nibblebill.c:449 Last successful
billing time was 
2010-02-22 08:47:53.875324 [DEBUG] mod_nibblebill.c:455 30 seconds passed
since last bill time of 2010-02-22 08:47:23
2010-02-22 08:47:53.875324 [DEBUG] mod_nibblebill.c:461 Billing $0.502331 to
3007 (Call: 68cde83c-26f4-44e3-8060-4188a106ff51 / 0.474338 so far)
2010-02-22 08:48:23.995212 [DEBUG] mod_nibblebill.c:529 Received request via
SESSION_HEARTBEAT!
2010-02-22 08:48:23.995212 [DEBUG] mod_nibblebill.c:397 Attempting to bill
at $1.0 per minute to account 3007
2010-02-22 08:48:23.995212 [DEBUG] mod_nibblebill.c:449 Last successful
billing time was 
2010-02-22 08:48:23.995212 [DEBUG] mod_nibblebill.c:455 30 seconds passed
since last bill time of 2010-02-22 08:47:53
2010-02-22 08:48:23.995212 [DEBUG] mod_nibblebill.c:461 Billing $0.501998 to
3007 (Call: 68cde83c-26f4-44e3-8060-4188a106ff51 / 0.976670 so far)
2010-02-22 08:48:54.115099 [DEBUG] mod_nibblebill.c:529 Received request via
SESSION_HEARTBEAT!
2010-02-22 08:48:54.115099 [DEBUG] mod_nibblebill.c:397 Attempting to bill
at $1.0 per minute to account 3007
2010-02-22 08:48:54.115099 [DEBUG] mod_nibblebill.c:449 Last successful
billing time was 
2010-02-22 08:48:54.115099 [DEBUG] mod_nibblebill.c:455 30 seconds passed
since last bill time of 2010-02-22 08:48:23
2010-02-22 08:48:54.115099 [DEBUG] mod_nibblebill.c:461 Billing $0.501998 to
3007 (Call: 68cde83c-26f4-44e3-8060-4188a106ff51 / 1.478668 so far)
2010-02-22 08:49:12.271231 [NOTICE] sofia.c:329 Hangup
sofia/internal/3007 at 192.168.15.177 [CS_EXECUTE] [NORMAL_CLEARING]

Any help in regards of how to get nibblebill to access the database properly
would be great.  Again I have installed ODBC and created the accounts table
with the required fields.  Thanks a ton


-----
Todd who is to learn
-- 
View this message in context: http://n2.nabble.com/mod-nibblebill-tp4612298p4612298.html
Sent from the freeswitch-users mailing list archive at Nabble.com.



From ivdreg at gmail.com  Mon Feb 22 07:11:26 2010
From: ivdreg at gmail.com (ivdreg ivdreg)
Date: Mon, 22 Feb 2010 17:11:26 +0200
Subject: [Freeswitch-users] SDP With T.38 in INVITE Problem
In-Reply-To: <24736FFE-BEEF-4D34-BBA7-77B6B6AE0EBD@jerris.com>
References: <b437a0e1002190242y2aa64696j91e8bc76f7ae2a92@mail.gmail.com>
	<b437a0e1002220448g3e3bda9fpa283e93a11d319f1@mail.gmail.com>
	<24736FFE-BEEF-4D34-BBA7-77B6B6AE0EBD@jerris.com>
Message-ID: <b437a0e1002220711u1bfd138l8a67b4e0b024d856@mail.gmail.com>

Hi Michael,

This happens when ONLY IF initial INVITE is coming with T.38 from a GW (this
is ITSP equipment and I don't know vendor) to our SIP subscribers with ATCOM
ATA and IP Phone. We use now in production YATE for terminating and
originating GWs to ITSPs and FS as main routing logic (backend). We want to
switch YATE to FS for a GW also but we faced this problem. This not happens
if initial INVITE nave no T.38 offered and later re-INVITE with T.38 with
valid SDP port.

Thanks

2010/2/22 Michael Jerris <mike at jerris.com>

> If the endpoint does not correctly follow the sdp o/a model its not going
> to work.  This is not a "problem" with the sofia library, this is intended
> behavior and what we are supposed to do.  What is the device?
>
> Mike
>
> On Feb 22, 2010, at 7:48 AM, ivdreg ivdreg wrote:
>
> Hi All,
>
> Actually while seeking the solution in internet I see some people having
> this problem with sofia library. I'm not sure that SIP reply in this case
> contains a valid SDP (I think that teminating endpoint is broken) but in my
> opinion if we have at least one valid media type in SDP (video, audio, image
> ...) call must be established. Can someone comment and/or help me with this
> issue.
>
> Regards.
>
> 2010/2/19 ivdreg ivdreg <ivdreg at gmail.com>
>
>> Hi all,
>>
>> Dose someone have a problem that if there T.38 in coming from gateway
>> FreeSwitch drops the call because of media error ? As I see from log only
>> T.38 port is zero and SDP has also media port. Is it possible to configure
>> FS to do not break a call but if media is OK.
>>
>> 2010-02-19 12:26:14.305244 [NOTICE] switch_channel.c:666 New Channel
>> sofia/backend/XXXXXXXXXX at 10.10.1.110:7065[6cd9f634-411d-df11-99ca-003048bb99cc]
>> 2010-02-19 12:26:14.305244 [DEBUG] mod_sofia.c:3369 (sofia/backend/
>> XXXXXXXXXX at 10.10.1.110:7065) State Change CS_NEW -> CS_INIT
>> 2010-02-19 12:26:14.305244 [DEBUG] switch_core_session.c:1018 Send signal
>> sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [BREAK]
>> 2010-02-19 12:26:14.305244 [DEBUG] switch_core_state_machine.c:314
>> (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) Running State Change CS_INIT
>> 2010-02-19 12:26:14.305244 [DEBUG] switch_core_state_machine.c:338
>> (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State INIT
>> 2010-02-19 12:26:14.305244 [DEBUG] mod_sofia.c:83 sofia/backend/
>> XXXXXXXXXX at 10.10.1.110:7065 SOFIA INIT
>> 2010-02-19 12:26:14.305244 [DEBUG] sofia_glue.c:1377 sofia/backend/
>> XXXXXXXXXX at 10.10.1.110:7065 Patched SDP
>> ---
>> v=0
>> o=- 3779949069 5166785777234550622 IN IP4 206.132.232.212
>> s=session
>> t=0 0
>> m=audio 21108 RTP/AVP 18 4 8 0
>> c=IN IP4 10.10.1.110
>> a=rtpmap:18 G729/8000
>> a=rtpmap:4 G723/8000
>> a=rtpmap:8 PCMA/8000
>> a=rtpmap:0 PCMU/8000
>> m=image 21108 udptl t38
>> c=IN IP4 10.10.1.110
>> a=T38FaxVersion:0
>> a=T38MaxBitRate:14400
>> a=T38FaxUdpEC:t38UDPRedundancy
>> a=T38FaxRateManagement:transferredTCF
>>
>> +++
>> v=0
>> o=- 3779949069 5166785777234550622 IN IP4 206.132.232.212
>> s=session
>> t=0 0
>> m=audio 17058 RTP/AVP 18 4 8 0
>> c=IN IP4 10.10.1.100
>> a=rtpmap:18 G729/8000
>> a=rtpmap:4 G723/8000
>> a=rtpmap:8 PCMA/8000
>> a=rtpmap:0 PCMU/8000
>> m=image 17058 udptl t38
>> c=IN IP4 10.10.1.100
>> a=T38FaxVersion:0
>> a=T38MaxBitRate:14400
>> a=T38FaxUdpEC:t38UDPRedundancy
>> a=T38FaxRateManagement:transferredTCF
>>
>> 2010-02-19 12:26:14.305244 [DEBUG] mod_sofia.c:117 (sofia/backend/
>> XXXXXXXXXX at 10.10.1.110:7065) State Change CS_INIT -> CS_ROUTING
>> ......
>> 2010-02-19 12:26:21.255201 [DEBUG] sofia.c:4135 Remote SDP:
>> v=0
>> o=FreeSWITCH 1266548331 1266548332 IN IP4 10.10.1.110
>> s=FreeSWITCH
>> c=IN IP4 10.10.1.110
>> t=0 0
>> *m=audio 26850 RTP/AVP 8*
>> a=rtpmap:8 PCMA/8000
>> a=silenceSupp:off - - - -
>> a=ptime:20
>> *m=image 0 udptl 19*
>>
>> 2010-02-19 12:26:21.255201 [DEBUG] sofia.c:4124 Channel sofia/backend/
>> XXXXXXXXXX at 10.10.1.110:7065 entering state [ready][200]
>> 2010-02-19 12:26:21.255201 [DEBUG] switch_channel.c:2285 Send signal
>> sofia/backend/YYYYYYYYYY at 10.10.1.110 [BREAK]
>> 2010-02-19 12:26:21.255201 [NOTICE] sofia.c:4668 Channel [sofia/backend/
>> XXXXXXXXXX at 10.10.1.110:7065] has been answered
>> 2010-02-19 12:26:21.255201 [DEBUG] sofia_glue.c:2288 Set Codec
>> sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 PROXY/8000 20 ms 160 samples
>> *2010-02-19 12:26:21.255201 [DEBUG] sofia_glue.c:2571 PROXY AUDIO RTP
>> [sofia/backend/XXXXXXXXXX at 10.10.1.110:7065] 10.10.1.100:17058->
>> 10.10.1.110:0 codec: 0 ms: 20
>> 2010-02-19 12:26:21.255201 [ERR] sofia_glue.c:2879 AUDIO RTP REPORTS
>> ERROR: [Missing remote port]
>> 2010-02-19 12:26:21.255201 [NOTICE] sofia_glue.c:2880 Hangup
>> sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [CS_CONSUME_MEDIA]
>> [DESTINATION_OUT_OF_ORDER]*
>> 2010-02-19 12:26:21.255201 [DEBUG] switch_channel.c:2063 Send signal
>> sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [KILL]
>> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_session.c:1018 Send signal
>> sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [BREAK]
>> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:314
>> (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) Running State Change
>> CS_HANGUP
>> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:499
>> (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State HANGUP
>> 2010-02-19 12:26:21.255201 [DEBUG] mod_sofia.c:411 Channel sofia/backend/
>> XXXXXXXXXX at 10.10.1.110:7065 hanging up, cause: DESTINATION_OUT_OF_ORDER
>> 2010-02-19 12:26:21.255201 [DEBUG] mod_sofia.c:454 Sending BYE to
>> sofia/backend/XXXXXXXXXX at 10.10.1.110:7065
>> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:46
>> sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 Standard HANGUP, cause:
>> DESTINATION_OUT_OF_ORDER
>> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:499
>> (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State HANGUP going to sleep
>> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:333
>> (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State Change CS_HANGUP ->
>> CS_REPORTING
>> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_session.c:1018 Send signal
>> sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [BREAK]
>> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:314
>> (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) Running State Change
>> CS_REPORTING
>> 2010-02-19 12:26:21.255201 [DEBUG] switch_ivr_originate.c:3185 Originate
>> Resulted in Error Cause: 27 [DESTINATION_OUT_OF_ORDER]
>> 2010-02-19 12:26:21.255201 [INFO] mod_dptools.c:2353 Originate Failed.
>> Cause: DESTINATION_OUT_OF_ORDER
>> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:590
>> (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State REPORTING
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100222/94b5f65c/attachment-0002.html 

From tculjaga at gmail.com  Mon Feb 22 07:23:22 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Mon, 22 Feb 2010 16:23:22 +0100
Subject: [Freeswitch-users] Freeswitch and mod_opal
In-Reply-To: <BDB26458EDF74CF6A382375739AF86BD@gamar>
References: <BDB26458EDF74CF6A382375739AF86BD@gamar>
Message-ID: <65d96fc81002220723o7186d29dr89edeb1be7ead647@mail.gmail.com>

On Mon, Feb 22, 2010 at 2:30 PM, Gilbert Amar <gamar at center.com> wrote:

> Thank you Tihomir
>
>
> We tried to build the svn trunk of mod_h323 and succeded but got a core
> dump
> a the first call hanging up.
>


strange, what h323plus and ptlib version are you using ?

can you post some logs on pastebin and add a link here ?


>
> >mod_h323
> >you need to build h323plus & ptlib before you build the module itself.
> >check http://wiki.freeswitch.org/wiki/Mod_h323 it works out of the box.
> >regarding you call setup, faststart = true, h245tuneling = true,
> >h245InSetup = false
> >It has to work.... if not post the logs here.
>
> >T.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100222/306ec349/attachment-0002.html 

From mike at jerris.com  Mon Feb 22 07:29:22 2010
From: mike at jerris.com (Michael Jerris)
Date: Mon, 22 Feb 2010 10:29:22 -0500
Subject: [Freeswitch-users] SDP With T.38 in INVITE Problem
In-Reply-To: <b437a0e1002220711u1bfd138l8a67b4e0b024d856@mail.gmail.com>
References: <b437a0e1002190242y2aa64696j91e8bc76f7ae2a92@mail.gmail.com>
	<b437a0e1002220448g3e3bda9fpa283e93a11d319f1@mail.gmail.com>
	<24736FFE-BEEF-4D34-BBA7-77B6B6AE0EBD@jerris.com>
	<b437a0e1002220711u1bfd138l8a67b4e0b024d856@mail.gmail.com>
Message-ID: <93769c21002220729q5bf36eadk80726aa8c00e3bd@mail.gmail.com>

if you want to see what is going on, crank up the debug in freeswitch and
sofia and you should see exactly what is going on.

Mike

On Mon, Feb 22, 2010 at 10:11 AM, ivdreg ivdreg <ivdreg at gmail.com> wrote:

> Hi Michael,
>
> This happens when ONLY IF initial INVITE is coming with T.38 from a GW
> (this is ITSP equipment and I don't know vendor) to our SIP subscribers with
> ATCOM ATA and IP Phone. We use now in production YATE for terminating and
> originating GWs to ITSPs and FS as main routing logic (backend). We want to
> switch YATE to FS for a GW also but we faced this problem. This not happens
> if initial INVITE nave no T.38 offered and later re-INVITE with T.38 with
> valid SDP port.
>
> Thanks
>
> 2010/2/22 Michael Jerris <mike at jerris.com>
>
>> If the endpoint does not correctly follow the sdp o/a model its not going
>> to work.  This is not a "problem" with the sofia library, this is intended
>> behavior and what we are supposed to do.  What is the device?
>>
>> Mike
>>
>> On Feb 22, 2010, at 7:48 AM, ivdreg ivdreg wrote:
>>
>> Hi All,
>>
>> Actually while seeking the solution in internet I see some people having
>> this problem with sofia library. I'm not sure that SIP reply in this case
>> contains a valid SDP (I think that teminating endpoint is broken) but in my
>> opinion if we have at least one valid media type in SDP (video, audio, image
>> ...) call must be established. Can someone comment and/or help me with this
>> issue.
>>
>> Regards.
>>
>> 2010/2/19 ivdreg ivdreg <ivdreg at gmail.com>
>>
>>> Hi all,
>>>
>>> Dose someone have a problem that if there T.38 in coming from gateway
>>> FreeSwitch drops the call because of media error ? As I see from log only
>>> T.38 port is zero and SDP has also media port. Is it possible to configure
>>> FS to do not break a call but if media is OK.
>>>
>>> 2010-02-19 12:26:14.305244 [NOTICE] switch_channel.c:666 New Channel
>>> sofia/backend/XXXXXXXXXX at 10.10.1.110:7065[6cd9f634-411d-df11-99ca-003048bb99cc]
>>> 2010-02-19 12:26:14.305244 [DEBUG] mod_sofia.c:3369 (sofia/backend/
>>> XXXXXXXXXX at 10.10.1.110:7065) State Change CS_NEW -> CS_INIT
>>> 2010-02-19 12:26:14.305244 [DEBUG] switch_core_session.c:1018 Send signal
>>> sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [BREAK]
>>> 2010-02-19 12:26:14.305244 [DEBUG] switch_core_state_machine.c:314
>>> (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) Running State Change CS_INIT
>>> 2010-02-19 12:26:14.305244 [DEBUG] switch_core_state_machine.c:338
>>> (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State INIT
>>> 2010-02-19 12:26:14.305244 [DEBUG] mod_sofia.c:83 sofia/backend/
>>> XXXXXXXXXX at 10.10.1.110:7065 SOFIA INIT
>>> 2010-02-19 12:26:14.305244 [DEBUG] sofia_glue.c:1377 sofia/backend/
>>> XXXXXXXXXX at 10.10.1.110:7065 Patched SDP
>>> ---
>>> v=0
>>> o=- 3779949069 5166785777234550622 IN IP4 206.132.232.212
>>> s=session
>>> t=0 0
>>> m=audio 21108 RTP/AVP 18 4 8 0
>>> c=IN IP4 10.10.1.110
>>> a=rtpmap:18 G729/8000
>>> a=rtpmap:4 G723/8000
>>> a=rtpmap:8 PCMA/8000
>>> a=rtpmap:0 PCMU/8000
>>> m=image 21108 udptl t38
>>> c=IN IP4 10.10.1.110
>>> a=T38FaxVersion:0
>>> a=T38MaxBitRate:14400
>>> a=T38FaxUdpEC:t38UDPRedundancy
>>> a=T38FaxRateManagement:transferredTCF
>>>
>>> +++
>>> v=0
>>> o=- 3779949069 5166785777234550622 IN IP4 206.132.232.212
>>> s=session
>>> t=0 0
>>> m=audio 17058 RTP/AVP 18 4 8 0
>>> c=IN IP4 10.10.1.100
>>> a=rtpmap:18 G729/8000
>>> a=rtpmap:4 G723/8000
>>> a=rtpmap:8 PCMA/8000
>>> a=rtpmap:0 PCMU/8000
>>> m=image 17058 udptl t38
>>> c=IN IP4 10.10.1.100
>>> a=T38FaxVersion:0
>>> a=T38MaxBitRate:14400
>>> a=T38FaxUdpEC:t38UDPRedundancy
>>> a=T38FaxRateManagement:transferredTCF
>>>
>>> 2010-02-19 12:26:14.305244 [DEBUG] mod_sofia.c:117 (sofia/backend/
>>> XXXXXXXXXX at 10.10.1.110:7065) State Change CS_INIT -> CS_ROUTING
>>> ......
>>> 2010-02-19 12:26:21.255201 [DEBUG] sofia.c:4135 Remote SDP:
>>> v=0
>>> o=FreeSWITCH 1266548331 1266548332 IN IP4 10.10.1.110
>>> s=FreeSWITCH
>>> c=IN IP4 10.10.1.110
>>> t=0 0
>>> *m=audio 26850 RTP/AVP 8*
>>> a=rtpmap:8 PCMA/8000
>>> a=silenceSupp:off - - - -
>>> a=ptime:20
>>> *m=image 0 udptl 19*
>>>
>>> 2010-02-19 12:26:21.255201 [DEBUG] sofia.c:4124 Channel sofia/backend/
>>> XXXXXXXXXX at 10.10.1.110:7065 entering state [ready][200]
>>> 2010-02-19 12:26:21.255201 [DEBUG] switch_channel.c:2285 Send signal
>>> sofia/backend/YYYYYYYYYY at 10.10.1.110 [BREAK]
>>> 2010-02-19 12:26:21.255201 [NOTICE] sofia.c:4668 Channel [sofia/backend/
>>> XXXXXXXXXX at 10.10.1.110:7065] has been answered
>>> 2010-02-19 12:26:21.255201 [DEBUG] sofia_glue.c:2288 Set Codec
>>> sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 PROXY/8000 20 ms 160 samples
>>> *2010-02-19 12:26:21.255201 [DEBUG] sofia_glue.c:2571 PROXY AUDIO RTP
>>> [sofia/backend/XXXXXXXXXX at 10.10.1.110:7065] 10.10.1.100:17058->
>>> 10.10.1.110:0 codec: 0 ms: 20
>>> 2010-02-19 12:26:21.255201 [ERR] sofia_glue.c:2879 AUDIO RTP REPORTS
>>> ERROR: [Missing remote port]
>>> 2010-02-19 12:26:21.255201 [NOTICE] sofia_glue.c:2880 Hangup
>>> sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [CS_CONSUME_MEDIA]
>>> [DESTINATION_OUT_OF_ORDER]*
>>> 2010-02-19 12:26:21.255201 [DEBUG] switch_channel.c:2063 Send signal
>>> sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [KILL]
>>> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_session.c:1018 Send signal
>>> sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [BREAK]
>>> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:314
>>> (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) Running State Change
>>> CS_HANGUP
>>> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:499
>>> (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State HANGUP
>>> 2010-02-19 12:26:21.255201 [DEBUG] mod_sofia.c:411 Channel sofia/backend/
>>> XXXXXXXXXX at 10.10.1.110:7065 hanging up, cause: DESTINATION_OUT_OF_ORDER
>>> 2010-02-19 12:26:21.255201 [DEBUG] mod_sofia.c:454 Sending BYE to
>>> sofia/backend/XXXXXXXXXX at 10.10.1.110:7065
>>> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:46
>>> sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 Standard HANGUP, cause:
>>> DESTINATION_OUT_OF_ORDER
>>> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:499
>>> (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State HANGUP going to sleep
>>> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:333
>>> (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State Change CS_HANGUP ->
>>> CS_REPORTING
>>> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_session.c:1018 Send signal
>>> sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [BREAK]
>>> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:314
>>> (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) Running State Change
>>> CS_REPORTING
>>> 2010-02-19 12:26:21.255201 [DEBUG] switch_ivr_originate.c:3185 Originate
>>> Resulted in Error Cause: 27 [DESTINATION_OUT_OF_ORDER]
>>> 2010-02-19 12:26:21.255201 [INFO] mod_dptools.c:2353 Originate Failed.
>>> Cause: DESTINATION_OUT_OF_ORDER
>>> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:590
>>> (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State REPORTING
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100222/0825dfa9/attachment-0002.html 

From ederwander at gmail.com  Mon Feb 22 07:58:18 2010
From: ederwander at gmail.com (Eder Souza)
Date: Mon, 22 Feb 2010 12:58:18 -0300
Subject: [Freeswitch-users] Dial String Inject in FreeSwitch
Message-ID: <622bedea1002220758r3fb59040g4a12af7996d289b5@mail.gmail.com>

http://ederwander.wordpress.com/2010/02/22/dial-string-inject-in-freeswitch/

just for yours informations i write this article my test for injections in
freesitch

version of my tests

freeswitch at internal> version
FreeSWITCH Version 1.0.5-20100218-0400 (hacked)
freeswitch at internal>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100222/8c929f9a/attachment-0002.html 

From anthony.minessale at gmail.com  Mon Feb 22 08:19:21 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 22 Feb 2010 10:19:21 -0600
Subject: [Freeswitch-users] Dial String Inject in FreeSwitch
In-Reply-To: <622bedea1002220758r3fb59040g4a12af7996d289b5@mail.gmail.com>
References: <622bedea1002220758r3fb59040g4a12af7996d289b5@mail.gmail.com>
Message-ID: <191c3a031002220819j462ee4c1v3e573124bf161d5b@mail.gmail.com>

Please do not use our project to try to make your blog more popular.

Your example requires you to prepare an intentional specific extension on
the FreeSWITCH custom made for your attack. It?s like saying if you leave
your door wide open at your house and call and tell someone, they can come
and rob you at 8:30.

This extension is also vulnerable ?by virtue of the stupidity of the
composer?

<extension name=?please-hack-me?/>
  <condition>
   <action application=?system? data=?${destination_number}?/>
  </condition>
</extension>

You should not allow tainted data from outside system to be fed directly
into your code. There is a regex system in place to extract legitimate data
from the user tainted input and safeguard against this.





On Mon, Feb 22, 2010 at 9:58 AM, Eder Souza <ederwander at gmail.com> wrote:

>
> http://ederwander.wordpress.com/2010/02/22/dial-string-inject-in-freeswitch/
>
> just for yours informations i write this article my test for injections in
> freesitch
>
> version of my tests
>
> freeswitch at internal> version
> FreeSWITCH Version 1.0.5-20100218-0400 (hacked)
> freeswitch at internal>
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100222/9052a7e0/attachment-0002.html 

From ederwander at gmail.com  Mon Feb 22 08:33:32 2010
From: ederwander at gmail.com (Eder Souza)
Date: Mon, 22 Feb 2010 13:33:32 -0300
Subject: [Freeswitch-users] Dial String Inject in FreeSwitch
In-Reply-To: <191c3a031002220819j462ee4c1v3e573124bf161d5b@mail.gmail.com>
References: <622bedea1002220758r3fb59040g4a12af7996d289b5@mail.gmail.com>
	<191c3a031002220819j462ee4c1v3e573124bf161d5b@mail.gmail.com>
Message-ID: <622bedea1002220833q2a14a9fbh280d0944f5b86fee@mail.gmail.com>

Antony i dont see why ??


this is just one alert for all comunity of danger in the use of regular
expression (.*) or (.*) ...

many peoples can make dialplans witch use of this expressions ...






On Mon, Feb 22, 2010 at 1:19 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> Please do not use our project to try to make your blog more popular.
>
> Your example requires you to prepare an intentional specific extension on
> the FreeSWITCH custom made for your attack. It?s like saying if you leave
> your door wide open at your house and call and tell someone, they can come
> and rob you at 8:30.
>
> This extension is also vulnerable ?by virtue of the stupidity of the
> composer?
>
> <extension name=?please-hack-me?/>
>   <condition>
>    <action application=?system? data=?${destination_number}?/>
>   </condition>
> </extension>
>
> You should not allow tainted data from outside system to be fed directly
> into your code. There is a regex system in place to extract legitimate data
> from the user tainted input and safeguard against this.
>
>
>
>
>
>   On Mon, Feb 22, 2010 at 9:58 AM, Eder Souza <ederwander at gmail.com>wrote:
>
>>
>> http://ederwander.wordpress.com/2010/02/22/dial-string-inject-in-freeswitch/
>>
>> just for yours informations i write this article my test for injections in
>> freesitch
>>
>> version of my tests
>>
>> freeswitch at internal> version
>> FreeSWITCH Version 1.0.5-20100218-0400 (hacked)
>> freeswitch at internal>
>>
>>
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100222/9aa03884/attachment-0002.html 

From brian at freeswitch.org  Mon Feb 22 08:39:37 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 22 Feb 2010 10:39:37 -0600
Subject: [Freeswitch-users] Dial String Inject in FreeSwitch
In-Reply-To: <622bedea1002220833q2a14a9fbh280d0944f5b86fee@mail.gmail.com>
References: <622bedea1002220758r3fb59040g4a12af7996d289b5@mail.gmail.com>
	<191c3a031002220819j462ee4c1v3e573124bf161d5b@mail.gmail.com>
	<622bedea1002220833q2a14a9fbh280d0944f5b86fee@mail.gmail.com>
Message-ID: <B2819BAB-7337-460C-BFF0-E0F7242A6D70@freeswitch.org>

And many people that own guns end up shooting themselves too.

/b

On Feb 22, 2010, at 10:33 AM, Eder Souza wrote:

> many peoples can make dialplans witch use of this expressions ...




From steveu at coppice.org  Mon Feb 22 08:41:39 2010
From: steveu at coppice.org (Steve Underwood)
Date: Tue, 23 Feb 2010 00:41:39 +0800
Subject: [Freeswitch-users] Dial String Inject in FreeSwitch
In-Reply-To: <622bedea1002220758r3fb59040g4a12af7996d289b5@mail.gmail.com>
References: <622bedea1002220758r3fb59040g4a12af7996d289b5@mail.gmail.com>
Message-ID: <4B82B3C3.9090007@coppice.org>

On 02/22/2010 11:58 PM, Eder Souza wrote:
> http://ederwander.wordpress.com/2010/02/22/dial-string-inject-in-freeswitch/
> just for yours informations i write this article my test for 
> injections in freesitch
> version of my tests
> freeswitch at internal <mailto:freeswitch at internal>> version
> FreeSWITCH Version 1.0.5-20100218-0400 (hacked)
> freeswitch at internal <mailto:freeswitch at internal>>
>
Leaving your car unlocked, the keys in the ignition, and a big sign on 
the windscreen saying "THIS CAR IS UNLOCKED" has a comparable effect. 
This is a sleazy way to get page hits. Someone really should create a 
wordpress blacklist.

Steve




From anthony.minessale at gmail.com  Mon Feb 22 08:42:13 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 22 Feb 2010 10:42:13 -0600
Subject: [Freeswitch-users] Question about sofia_contact
In-Reply-To: <4B8249BF.3090708@ewetel.de>
References: <4B7EA954.30402@ewetel.de> <4B7EADBC.1040001@ewetel.de>
	<4B8249BF.3090708@ewetel.de>
Message-ID: <191c3a031002220842j5bec442an5f1ea89cb0e8a6ff@mail.gmail.com>

it's mad at you for asking twice before waiting for a reply, so it's not
working on purpose.

Actually it's mad at you because your domain does not contain a . so it is
assuming you are specifying a profile name as the domain.  if your domain
was mydomain.com instead it would work.


On Mon, Feb 22, 2010 at 3:09 AM, Helmut Kuper <helmut.kuper at ewetel.de>wrote:

> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> Hello,
>
> has anybody an idea?
>
> regards
> helmut
>
>
> On 19.02.2010 16:26, Helmut Kuper wrote:
> > Hi,
> >
> > an update:
> > The corresponding select statement looks for sip_user="2701" and
> > sip_host="internal" in registration table.
> > This delivers of course no result because 2701 is registered with
> > sip_host="mydomain".
> >
> >
> > Hm any workaround or am I going in a wrong direction?
> >
> >
> > regards
> > Helmut
> >
> >
> > On 19.02.2010 16:08, Helmut Kuper wrote:
> >> Hello,
> >
> >> I try to setup a FS sofia sip-profile which allows me to have multiple
> >> sip-profiles but one registration database. So I set the following
> >> parameters:
> >
> >> <param name="challenge-realm" value="auto_from"/>
> >> <param name="force-register-domain" value="$${domain}"/>
> >> <param name="force-subscription-domain" value="$${domain}"/>
> >> <param name="force-register-db-domain" value="$${domain}"/>
> >
> >> where domain is set to "mydomain". "sofia status profile internal"
> >> delivers the following:
> >
> >
> >> Call-ID:        3c26705038e5-vwlg8u5q9cwe
> >> User:           2701 at mydomain
> >> Contact:        <sip:2701 at 85.16.245.208:1024;line=l2wksedo>
> >> Agent:          snom370/8.2.22
> >> Status:         Registered(UDP)(unknown) EXP(2010-02-19 16:13:31)
> >> Host:           ippbx-prod-node0
> >> IP:             85.16.245.208
> >> Port:           1024
> >> Auth-User:      2701
> >> Auth-Realm:     mydomain
> >> MWI-Account:    2701 at mydomain
> >
> >
> >
> >> sofia_contact internal/2701 at mydomain delivers this:
> >> error/user_not_registered
> >
> >> The Phone is fully functional.
> >
> >> I use SVN trunk 16601
> >
> >> regards
> >> Helmut
>
> -----BEGIN PGP SIGNATURE-----
> Version: GnuPG v1.4.7 (MingW32)
>
> iD8DBQFLgkm/4tZeNddg3dwRApfoAKCiX8fX/WNrZ7GXRrBJA54+VTThmACfT0d3
> fBzQlyVObkJaLHxJbfUjZG4=
> =uhHR
> -----END PGP SIGNATURE-----
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100222/fd4e782f/attachment-0002.html 

From anthony.minessale at gmail.com  Mon Feb 22 08:47:43 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 22 Feb 2010 10:47:43 -0600
Subject: [Freeswitch-users] Dial String Inject in FreeSwitch
In-Reply-To: <622bedea1002220833q2a14a9fbh280d0944f5b86fee@mail.gmail.com>
References: <622bedea1002220758r3fb59040g4a12af7996d289b5@mail.gmail.com>
	<191c3a031002220819j462ee4c1v3e573124bf161d5b@mail.gmail.com>
	<622bedea1002220833q2a14a9fbh280d0944f5b86fee@mail.gmail.com>
Message-ID: <191c3a031002220847pa6ca94ey94fa526dad3583d5@mail.gmail.com>

To me it sounds like a way to sound the alarms and bring negative attention.

For instance, if you were sincerely concerned, you could have told us about
your discovery privately first, and we could feature a story on our own site
warning people of this danger and reminding them how to compose extension
properly.

The posting was instead made like a big public announcement calling our
software "imperfect".
Yes it is imperfect, It can't properly detect someone being a moron 100% of
the time but it sure tries it's darndest.




On Mon, Feb 22, 2010 at 10:33 AM, Eder Souza <ederwander at gmail.com> wrote:

> Antony i dont see why ??
>
>
> this is just one alert for all comunity of danger in the use of regular
> expression (.*) or (.*) ...
>
> many peoples can make dialplans witch use of this expressions ...
>
>
>
>
>
>
> On Mon, Feb 22, 2010 at 1:19 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> Please do not use our project to try to make your blog more popular.
>>
>> Your example requires you to prepare an intentional specific extension on
>> the FreeSWITCH custom made for your attack. It?s like saying if you leave
>> your door wide open at your house and call and tell someone, they can come
>> and rob you at 8:30.
>>
>> This extension is also vulnerable ?by virtue of the stupidity of the
>> composer?
>>
>> <extension name=?please-hack-me?/>
>>   <condition>
>>    <action application=?system? data=?${destination_number}?/>
>>   </condition>
>> </extension>
>>
>> You should not allow tainted data from outside system to be fed directly
>> into your code. There is a regex system in place to extract legitimate data
>> from the user tainted input and safeguard against this.
>>
>>
>>
>>
>>
>>   On Mon, Feb 22, 2010 at 9:58 AM, Eder Souza <ederwander at gmail.com>wrote:
>>
>>>
>>> http://ederwander.wordpress.com/2010/02/22/dial-string-inject-in-freeswitch/
>>>
>>> just for yours informations i write this article my test for injections
>>> in freesitch
>>>
>>> version of my tests
>>>
>>> freeswitch at internal> version
>>> FreeSWITCH Version 1.0.5-20100218-0400 (hacked)
>>> freeswitch at internal>
>>>
>>>
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100222/187bcb50/attachment-0002.html 

From ivdreg at gmail.com  Mon Feb 22 08:49:07 2010
From: ivdreg at gmail.com (ivdreg ivdreg)
Date: Mon, 22 Feb 2010 18:49:07 +0200
Subject: [Freeswitch-users] SDP With T.38 in INVITE Problem
In-Reply-To: <93769c21002220729q5bf36eadk80726aa8c00e3bd@mail.gmail.com>
References: <b437a0e1002190242y2aa64696j91e8bc76f7ae2a92@mail.gmail.com>
	<b437a0e1002220448g3e3bda9fpa283e93a11d319f1@mail.gmail.com>
	<24736FFE-BEEF-4D34-BBA7-77B6B6AE0EBD@jerris.com>
	<b437a0e1002220711u1bfd138l8a67b4e0b024d856@mail.gmail.com>
	<93769c21002220729q5bf36eadk80726aa8c00e3bd@mail.gmail.com>
Message-ID: <b437a0e1002220849k16fd87ffl5a7257d80e43b036@mail.gmail.com>

Hi Michael,

As I said in a previous mails I know exactly what is happening.
In working setup:

ITSP ---> YATE (GW - Frontend) ---> FreeSwitch (routing server/xml_curl)
---> YATE (GW - SIP Interop) ---> OpenSIPS ---> Subscriber.

I prefer to change YATE (GW - Frontend) and YATE (GW - SIP Interop) with
FreeSwitch for some reasons. The problem is:

INVITE comes from ITSP and goes to subscriber via OpenSIPS. INIVITE between
FreeSwitch (routing server) and YATE (GW - SIP Interop) contains SDP:
m=audio 21108 RTP/AVP 18 4 8 0
c=IN IP4 10.10.1.110
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
m=image 21108 udptl t38
c=IN IP4 10.10.1.110
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxUdpEC:t38UDPRedundancy
a=T38FaxRateManagement:
transferredTCF

And reply 200 OK contains in SDP:
*m=audio 34788 RTP/AVP 8*
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20

Reply 200 OK SDP between YATE (GW - SIP Interop) and OpenSIPS contains in
SDP:
*m=audio 16330 RTP/AVP 8*
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
*m=image 0 udptl 19*

In this case everything works fine. Line *m=image 0 udptl 19 *is removed by
YATE.
But same test with FreeSWITCH on the place of YATE (GW - SIP Interop) *"m=image
0 udptl 19" *call brakes as you can see in my first mail.

I don't want to compare or discus YATE and FS functionality or something
else. I just see difference in behavior and because I'm not a big expert
don't know witch implementation is more accurate according standards. And
second: Is it impossible for me to upgrade all CPE so only thing I can do is
to fix it on server side. That is because I ask for a help.


Thanks to all.


2010/2/22 Michael Jerris <mike at jerris.com>

> if you want to see what is going on, crank up the debug in freeswitch and
> sofia and you should see exactly what is going on.
>
> Mike
>
>
> On Mon, Feb 22, 2010 at 10:11 AM, ivdreg ivdreg <ivdreg at gmail.com> wrote:
>
>> Hi Michael,
>>
>> This happens when ONLY IF initial INVITE is coming with T.38 from a GW
>> (this is ITSP equipment and I don't know vendor) to our SIP subscribers with
>> ATCOM ATA and IP Phone. We use now in production YATE for terminating and
>> originating GWs to ITSPs and FS as main routing logic (backend). We want to
>> switch YATE to FS for a GW also but we faced this problem. This not happens
>> if initial INVITE nave no T.38 offered and later re-INVITE with T.38 with
>> valid SDP port.
>>
>> Thanks
>>
>> 2010/2/22 Michael Jerris <mike at jerris.com>
>>
>>>  If the endpoint does not correctly follow the sdp o/a model its not
>>> going to work.  This is not a "problem" with the sofia library, this is
>>> intended behavior and what we are supposed to do.  What is the device?
>>>
>>> Mike
>>>
>>> On Feb 22, 2010, at 7:48 AM, ivdreg ivdreg wrote:
>>>
>>> Hi All,
>>>
>>> Actually while seeking the solution in internet I see some people having
>>> this problem with sofia library. I'm not sure that SIP reply in this case
>>> contains a valid SDP (I think that teminating endpoint is broken) but in my
>>> opinion if we have at least one valid media type in SDP (video, audio, image
>>> ...) call must be established. Can someone comment and/or help me with this
>>> issue.
>>>
>>> Regards.
>>>
>>> 2010/2/19 ivdreg ivdreg <ivdreg at gmail.com>
>>>
>>>> Hi all,
>>>>
>>>> Dose someone have a problem that if there T.38 in coming from gateway
>>>> FreeSwitch drops the call because of media error ? As I see from log only
>>>> T.38 port is zero and SDP has also media port. Is it possible to configure
>>>> FS to do not break a call but if media is OK.
>>>>
>>>> 2010-02-19 12:26:14.305244 [NOTICE] switch_channel.c:666 New Channel
>>>> sofia/backend/XXXXXXXXXX at 10.10.1.110:7065[6cd9f634-411d-df11-99ca-003048bb99cc]
>>>> 2010-02-19 12:26:14.305244 [DEBUG] mod_sofia.c:3369 (sofia/backend/
>>>> XXXXXXXXXX at 10.10.1.110:7065) State Change CS_NEW -> CS_INIT
>>>> 2010-02-19 12:26:14.305244 [DEBUG] switch_core_session.c:1018 Send
>>>> signal sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [BREAK]
>>>> 2010-02-19 12:26:14.305244 [DEBUG] switch_core_state_machine.c:314
>>>> (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) Running State Change
>>>> CS_INIT
>>>> 2010-02-19 12:26:14.305244 [DEBUG] switch_core_state_machine.c:338
>>>> (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State INIT
>>>> 2010-02-19 12:26:14.305244 [DEBUG] mod_sofia.c:83 sofia/backend/
>>>> XXXXXXXXXX at 10.10.1.110:7065 SOFIA INIT
>>>> 2010-02-19 12:26:14.305244 [DEBUG] sofia_glue.c:1377 sofia/backend/
>>>> XXXXXXXXXX at 10.10.1.110:7065 Patched SDP
>>>> ---
>>>> v=0
>>>> o=- 3779949069 5166785777234550622 IN IP4 206.132.232.212
>>>> s=session
>>>> t=0 0
>>>> m=audio 21108 RTP/AVP 18 4 8 0
>>>> c=IN IP4 10.10.1.110
>>>> a=rtpmap:18 G729/8000
>>>> a=rtpmap:4 G723/8000
>>>> a=rtpmap:8 PCMA/8000
>>>> a=rtpmap:0 PCMU/8000
>>>> m=image 21108 udptl t38
>>>> c=IN IP4 10.10.1.110
>>>> a=T38FaxVersion:0
>>>> a=T38MaxBitRate:14400
>>>> a=T38FaxUdpEC:t38UDPRedundancy
>>>> a=T38FaxRateManagement:transferredTCF
>>>>
>>>> +++
>>>> v=0
>>>> o=- 3779949069 5166785777234550622 IN IP4 206.132.232.212
>>>> s=session
>>>> t=0 0
>>>> m=audio 17058 RTP/AVP 18 4 8 0
>>>> c=IN IP4 10.10.1.100
>>>> a=rtpmap:18 G729/8000
>>>> a=rtpmap:4 G723/8000
>>>> a=rtpmap:8 PCMA/8000
>>>> a=rtpmap:0 PCMU/8000
>>>> m=image 17058 udptl t38
>>>> c=IN IP4 10.10.1.100
>>>> a=T38FaxVersion:0
>>>> a=T38MaxBitRate:14400
>>>> a=T38FaxUdpEC:t38UDPRedundancy
>>>> a=T38FaxRateManagement:transferredTCF
>>>>
>>>> 2010-02-19 12:26:14.305244 [DEBUG] mod_sofia.c:117 (sofia/backend/
>>>> XXXXXXXXXX at 10.10.1.110:7065) State Change CS_INIT -> CS_ROUTING
>>>> ......
>>>> 2010-02-19 12:26:21.255201 [DEBUG] sofia.c:4135 Remote SDP:
>>>> v=0
>>>> o=FreeSWITCH 1266548331 1266548332 IN IP4 10.10.1.110
>>>> s=FreeSWITCH
>>>> c=IN IP4 10.10.1.110
>>>> t=0 0
>>>> *m=audio 26850 RTP/AVP 8*
>>>> a=rtpmap:8 PCMA/8000
>>>> a=silenceSupp:off - - - -
>>>> a=ptime:20
>>>> *m=image 0 udptl 19*
>>>>
>>>> 2010-02-19 12:26:21.255201 [DEBUG] sofia.c:4124 Channel sofia/backend/
>>>> XXXXXXXXXX at 10.10.1.110:7065 entering state [ready][200]
>>>> 2010-02-19 12:26:21.255201 [DEBUG] switch_channel.c:2285 Send signal
>>>> sofia/backend/YYYYYYYYYY at 10.10.1.110 [BREAK]
>>>> 2010-02-19 12:26:21.255201 [NOTICE] sofia.c:4668 Channel [sofia/backend/
>>>> XXXXXXXXXX at 10.10.1.110:7065] has been answered
>>>> 2010-02-19 12:26:21.255201 [DEBUG] sofia_glue.c:2288 Set Codec
>>>> sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 PROXY/8000 20 ms 160 samples
>>>> *2010-02-19 12:26:21.255201 [DEBUG] sofia_glue.c:2571 PROXY AUDIO RTP
>>>> [sofia/backend/XXXXXXXXXX at 10.10.1.110:7065] 10.10.1.100:17058->
>>>> 10.10.1.110:0 codec: 0 ms: 20
>>>> 2010-02-19 12:26:21.255201 [ERR] sofia_glue.c:2879 AUDIO RTP REPORTS
>>>> ERROR: [Missing remote port]
>>>> 2010-02-19 12:26:21.255201 [NOTICE] sofia_glue.c:2880 Hangup
>>>> sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [CS_CONSUME_MEDIA]
>>>> [DESTINATION_OUT_OF_ORDER]*
>>>> 2010-02-19 12:26:21.255201 [DEBUG] switch_channel.c:2063 Send signal
>>>> sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [KILL]
>>>> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_session.c:1018 Send
>>>> signal sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [BREAK]
>>>> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:314
>>>> (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) Running State Change
>>>> CS_HANGUP
>>>> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:499
>>>> (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State HANGUP
>>>> 2010-02-19 12:26:21.255201 [DEBUG] mod_sofia.c:411 Channel
>>>> sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 hanging up, cause:
>>>> DESTINATION_OUT_OF_ORDER
>>>> 2010-02-19 12:26:21.255201 [DEBUG] mod_sofia.c:454 Sending BYE to
>>>> sofia/backend/XXXXXXXXXX at 10.10.1.110:7065
>>>> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:46
>>>> sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 Standard HANGUP, cause:
>>>> DESTINATION_OUT_OF_ORDER
>>>> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:499
>>>> (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State HANGUP going to sleep
>>>> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:333
>>>> (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State Change CS_HANGUP ->
>>>> CS_REPORTING
>>>> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_session.c:1018 Send
>>>> signal sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [BREAK]
>>>> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:314
>>>> (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) Running State Change
>>>> CS_REPORTING
>>>> 2010-02-19 12:26:21.255201 [DEBUG] switch_ivr_originate.c:3185 Originate
>>>> Resulted in Error Cause: 27 [DESTINATION_OUT_OF_ORDER]
>>>> 2010-02-19 12:26:21.255201 [INFO] mod_dptools.c:2353 Originate Failed.
>>>> Cause: DESTINATION_OUT_OF_ORDER
>>>> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:590
>>>> (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State REPORTING
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100222/e5b81361/attachment-0002.html 

From ederwander at gmail.com  Mon Feb 22 09:09:41 2010
From: ederwander at gmail.com (Eder Souza)
Date: Mon, 22 Feb 2010 14:09:41 -0300
Subject: [Freeswitch-users] Dial String Inject in FreeSwitch
In-Reply-To: <191c3a031002220847pa6ca94ey94fa526dad3583d5@mail.gmail.com>
References: <622bedea1002220758r3fb59040g4a12af7996d289b5@mail.gmail.com>
	<191c3a031002220819j462ee4c1v3e573124bf161d5b@mail.gmail.com>
	<622bedea1002220833q2a14a9fbh280d0944f5b86fee@mail.gmail.com>
	<191c3a031002220847pa6ca94ey94fa526dad3583d5@mail.gmail.com>
Message-ID: <622bedea1002220909x6694fdfdr6dd43f30b056edb2@mail.gmail.com>

i prefer FreeSwitch im left Asterisk

FreeSwitch is Very Very betther then Asterisk in my option !!


my intention is just say dont use (.*), (.+)  or combinations of this
regular expressions, for me FreeSwitch is the betther  !!




On Mon, Feb 22, 2010 at 1:47 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> To me it sounds like a way to sound the alarms and bring negative
> attention.
>
> For instance, if you were sincerely concerned, you could have told us about
> your discovery privately first, and we could feature a story on our own site
> warning people of this danger and reminding them how to compose extension
> properly.
>
> The posting was instead made like a big public announcement calling our
> software "imperfect".
> Yes it is imperfect, It can't properly detect someone being a moron 100% of
> the time but it sure tries it's darndest.
>
>
>
>
>
> On Mon, Feb 22, 2010 at 10:33 AM, Eder Souza <ederwander at gmail.com> wrote:
>
>> Antony i dont see why ??
>>
>>
>> this is just one alert for all comunity of danger in the use of regular
>> expression (.*) or (.*) ...
>>
>> many peoples can make dialplans witch use of this expressions ...
>>
>>
>>
>>
>>
>>
>> On Mon, Feb 22, 2010 at 1:19 PM, Anthony Minessale <
>> anthony.minessale at gmail.com> wrote:
>>
>>> Please do not use our project to try to make your blog more popular.
>>>
>>> Your example requires you to prepare an intentional specific extension on
>>> the FreeSWITCH custom made for your attack. It?s like saying if you leave
>>> your door wide open at your house and call and tell someone, they can come
>>> and rob you at 8:30.
>>>
>>> This extension is also vulnerable ?by virtue of the stupidity of the
>>> composer?
>>>
>>> <extension name=?please-hack-me?/>
>>>   <condition>
>>>    <action application=?system? data=?${destination_number}?/>
>>>   </condition>
>>> </extension>
>>>
>>> You should not allow tainted data from outside system to be fed directly
>>> into your code. There is a regex system in place to extract legitimate data
>>> from the user tainted input and safeguard against this.
>>>
>>>
>>>
>>>
>>>
>>>   On Mon, Feb 22, 2010 at 9:58 AM, Eder Souza <ederwander at gmail.com>wrote:
>>>
>>>>
>>>> http://ederwander.wordpress.com/2010/02/22/dial-string-inject-in-freeswitch/
>>>>
>>>> just for yours informations i write this article my test for injections
>>>> in freesitch
>>>>
>>>> version of my tests
>>>>
>>>> freeswitch at internal> version
>>>> FreeSWITCH Version 1.0.5-20100218-0400 (hacked)
>>>> freeswitch at internal>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>> iax:guest at conference.freeswitch.org/888
>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> pstn:+19193869900
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100222/ec4f6533/attachment-0002.html 

From steveu at coppice.org  Mon Feb 22 09:10:40 2010
From: steveu at coppice.org (Steve Underwood)
Date: Tue, 23 Feb 2010 01:10:40 +0800
Subject: [Freeswitch-users] SDP With T.38 in INVITE Problem
In-Reply-To: <b437a0e1002220849k16fd87ffl5a7257d80e43b036@mail.gmail.com>
References: <b437a0e1002190242y2aa64696j91e8bc76f7ae2a92@mail.gmail.com>	<b437a0e1002220448g3e3bda9fpa283e93a11d319f1@mail.gmail.com>	<24736FFE-BEEF-4D34-BBA7-77B6B6AE0EBD@jerris.com>	<b437a0e1002220711u1bfd138l8a67b4e0b024d856@mail.gmail.com>	<93769c21002220729q5bf36eadk80726aa8c00e3bd@mail.gmail.com>
	<b437a0e1002220849k16fd87ffl5a7257d80e43b036@mail.gmail.com>
Message-ID: <4B82BA90.10709@coppice.org>

Hi Michael,

On 02/23/2010 12:49 AM, ivdreg ivdreg wrote:
> Hi Michael,
>
> As I said in a previous mails I know exactly what is happening.
> In working setup:
>
> ITSP ---> YATE (GW - Frontend) ---> FreeSwitch (routing 
> server/xml_curl) ---> YATE (GW - SIP Interop) ---> OpenSIPS ---> 
> Subscriber.
>
> I prefer to change YATE (GW - Frontend) and YATE (GW - SIP Interop) 
> with FreeSwitch for some reasons. The problem is:
>
> INVITE comes from ITSP and goes to subscriber via OpenSIPS. INIVITE 
> between FreeSwitch (routing server) and YATE (GW - SIP Interop) 
> contains SDP:
> m=audio 21108 RTP/AVP 18 4 8 0
> c=IN IP4 10.10.1.110
> a=rtpmap:18 G729/8000
> a=rtpmap:4 G723/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:0 PCMU/8000
> m=image 21108 udptl t38
> c=IN IP4 10.10.1.110
> a=T38FaxVersion:0
> a=T38MaxBitRate:14400
> a=T38FaxUdpEC:t38UDPRedundancy
> a=T38FaxRateManagement:
> transferredTCF
>
> And reply 200 OK contains in SDP:
> *m=audio 34788 RTP/AVP 8*
> a=rtpmap:8 PCMA/8000
> a=silenceSupp:off - - - -
> a=ptime:20
>
> Reply 200 OK SDP between YATE (GW - SIP Interop) and OpenSIPS contains 
> in SDP:
> *m=audio 16330 RTP/AVP 8*
> a=rtpmap:8 PCMA/8000
> a=silenceSupp:off - - - -
> a=ptime:20
> *m=image 0 udptl 19*
>
> In this case everything works fine. Line *m=image 0 udptl 19 *is 
> removed by YATE.
> But same test with FreeSWITCH on the place of YATE (GW - SIP Interop) 
> *"m=image 0 udptl 19" *call brakes as you can see in my first mail.
>
> I don't want to compare or discus YATE and FS functionality or 
> something else. I just see difference in behavior and because I'm not 
> a big expert don't know witch implementation is more accurate 
> according standards. And second: Is it impossible for me to upgrade 
> all CPE so only thing I can do is to fix it on server side. That is 
> because I ask for a help.
You said the the broken line "m=image 0 udptl 19" goes from OpenSIPS to 
YATE. Do you know if it originates from the OpenSIPS box or the 
subscriber? If it originates from the OpenSIPS box it should be reported 
to them. If its from the subscriber, well...... your chances of getting 
anything fixed are usually small.

Steve




From christian.loeschenkohl at xpirio.com  Mon Feb 22 09:16:43 2010
From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=)
Date: Mon, 22 Feb 2010 18:16:43 +0100
Subject: [Freeswitch-users] sending a sip notify with sendevent
Message-ID: <4B82BBFB.1040804@xpirio.com>

hi

i try to send a sip notify message to a registered sip device
"sofia profile nat flush_inbound_reg 10 at vts.vie.xpirio.net reboot" works, but i need
to send "check-sync;reboot=false" - so the device does a resync and don't do a reboot

my message looks like this

sendevent NOTIFY
profile: nat
event-string: check-sync;reboot=false
user: 10
host: vts.vie.xpirio.net
content-type: application/simple-message-summary

if i listen on the loopback interface i do see

##
T 2010/02/22 18:11:59.083204 127.0.0.1:51840 -> 127.0.0.1:8021 [AP]
sendevent NOTIFY
profile: nat
event-string: check-sync;reboot=false
user: 10
host: vts.vie.xpirio.net
content-type: application/simple-message-summary

##
T 2010/02/22 18:11:59.084032 127.0.0.1:8021 -> 127.0.0.1:51840 [AP]
Content-Type: command/reply
Reply-Text: -ERR invalid

--------
i don't get what it is wrong. i also rechecked the registered user in the sqlite database and this
looks good to me.

no message is send to the user.

we do use multiple domains, so user could also be 10 at somedomain.com - or am i wrong on this?
could somebody please bring some light in this.

we do use trunk rev. 16631

br



-- 
Ing. Christian L?schenkohl
Technische Leitung, Forschung & Entwicklung VoIP

xpirio
Telekommunikation & Service GmbH
Lakeside B04
9020 Klagenfurt
Austria

T  +43 (0) 5 77 11 - 1000
F  +43 (0) 5 77 11 - 1002
E  christian.loeschenkohl at xpirio.com



From m.sobkow at marketelsystems.com  Mon Feb 22 09:21:40 2010
From: m.sobkow at marketelsystems.com (Mark Sobkow)
Date: Mon, 22 Feb 2010 11:21:40 -0600
Subject: [Freeswitch-users] 8000 rate .wav files
Message-ID: <4B82BD24.2030108@marketelsystems.com>

I've got the 8000 sample rate .wav files installed for Freeswitch.  
According to the logs, my SIP phone is connecting with an 8000 rate.  
However, when I try to play_and_get_digits using those sound files, I 
get errors:

2010-02-22 11:15:22.029347 [ERR] switch_core_file.c:122 Invalid file 
format [wav] for 
[/opt/freeswitch/sounds/en/us/callie/conference/8000/conf-pin.wav]!
2010-02-22 11:15:22.029347 [ERR] switch_core_file.c:122 Invalid file 
format [wav] for 
[/opt/freeswitch/sounds/en/us/callie/conference/8000/conf-bad-pin.wav]!

Aren't .wav files supposed to be compatible with all codecs for 
playback?  If not, what do I have to do to convert them to the proper 
formats?  How do I find out what the proper formats are?

-- 
Mark Sobkow
Senior Developer
MarkeTel Multi-Line Dialing Systems LTD.
428 Victoria Ave
Regina, SK S4N-0P6
Toll-Free: 800-289-8616-X533
Local: 306-359-6893-X533
Fax: 306-359-6879
Email: m.sobkow at marketelsystems.com
Web: http://www.marketelsystems.com 




From anthony.minessale at gmail.com  Mon Feb 22 09:24:37 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 22 Feb 2010 11:24:37 -0600
Subject: [Freeswitch-users] Dial String Inject in FreeSwitch
In-Reply-To: <622bedea1002220909x6694fdfdr6dd43f30b056edb2@mail.gmail.com>
References: <622bedea1002220758r3fb59040g4a12af7996d289b5@mail.gmail.com>
	<191c3a031002220819j462ee4c1v3e573124bf161d5b@mail.gmail.com>
	<622bedea1002220833q2a14a9fbh280d0944f5b86fee@mail.gmail.com>
	<191c3a031002220847pa6ca94ey94fa526dad3583d5@mail.gmail.com>
	<622bedea1002220909x6694fdfdr6dd43f30b056edb2@mail.gmail.com>
Message-ID: <191c3a031002220924m465a74b9w7e8b60678d7a3de@mail.gmail.com>

correct,

You could write a CGI for apache too that could let someone figure out how
to download the root password.

By default, nobody should trust the data supplied by the outside user.
FreeSWITCH cannot do this for you or the limitations would impair desired
functionality.

All you have to do is look for a digit sequence in your dial string.
Moreover you need to make sure even then that it's safe to pass this digit
string to the provider.
Here in USA we share the 1 country code with several other countries that
could cost 50 cents to a dollar a minute.
So you are not even safe when you made sure it's a number.



On Mon, Feb 22, 2010 at 11:09 AM, Eder Souza <ederwander at gmail.com> wrote:

> i prefer FreeSwitch im left Asterisk
>
> FreeSwitch is Very Very betther then Asterisk in my option !!
>
>
> my intention is just say dont use (.*), (.+)  or combinations of this
> regular expressions, for me FreeSwitch is the betther  !!
>
>
>
>
> On Mon, Feb 22, 2010 at 1:47 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> To me it sounds like a way to sound the alarms and bring negative
>> attention.
>>
>> For instance, if you were sincerely concerned, you could have told us
>> about your discovery privately first, and we could feature a story on our
>> own site warning people of this danger and reminding them how to compose
>> extension properly.
>>
>> The posting was instead made like a big public announcement calling our
>> software "imperfect".
>> Yes it is imperfect, It can't properly detect someone being a moron 100%
>> of the time but it sure tries it's darndest.
>>
>>
>>
>>
>>
>> On Mon, Feb 22, 2010 at 10:33 AM, Eder Souza <ederwander at gmail.com>wrote:
>>
>>> Antony i dont see why ??
>>>
>>>
>>> this is just one alert for all comunity of danger in the use of regular
>>> expression (.*) or (.*) ...
>>>
>>> many peoples can make dialplans witch use of this expressions ...
>>>
>>>
>>>
>>>
>>>
>>>
>>> On Mon, Feb 22, 2010 at 1:19 PM, Anthony Minessale <
>>> anthony.minessale at gmail.com> wrote:
>>>
>>>> Please do not use our project to try to make your blog more popular.
>>>>
>>>> Your example requires you to prepare an intentional specific extension
>>>> on the FreeSWITCH custom made for your attack. It?s like saying if you leave
>>>> your door wide open at your house and call and tell someone, they can come
>>>> and rob you at 8:30.
>>>>
>>>> This extension is also vulnerable ?by virtue of the stupidity of the
>>>> composer?
>>>>
>>>> <extension name=?please-hack-me?/>
>>>>   <condition>
>>>>    <action application=?system? data=?${destination_number}?/>
>>>>   </condition>
>>>> </extension>
>>>>
>>>> You should not allow tainted data from outside system to be fed directly
>>>> into your code. There is a regex system in place to extract legitimate data
>>>> from the user tainted input and safeguard against this.
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>   On Mon, Feb 22, 2010 at 9:58 AM, Eder Souza <ederwander at gmail.com>wrote:
>>>>
>>>>>
>>>>> http://ederwander.wordpress.com/2010/02/22/dial-string-inject-in-freeswitch/
>>>>>
>>>>> just for yours informations i write this article my test for injections
>>>>> in freesitch
>>>>>
>>>>> version of my tests
>>>>>
>>>>> freeswitch at internal> version
>>>>> FreeSWITCH Version 1.0.5-20100218-0400 (hacked)
>>>>> freeswitch at internal>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>>
>>>> --
>>>> Anthony Minessale II
>>>>
>>>> FreeSWITCH http://www.freeswitch.org/
>>>> ClueCon http://www.cluecon.com/
>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>
>>>> AIM: anthm
>>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>> IRC: irc.freenode.net #freeswitch
>>>>
>>>> FreeSWITCH Developer Conference
>>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>>> iax:guest at conference.freeswitch.org/888
>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>> pstn:+19193869900
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100222/6c7fa474/attachment-0002.html 

From gmaruzz at celliax.org  Mon Feb 22 09:26:00 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Mon, 22 Feb 2010 18:26:00 +0100
Subject: [Freeswitch-users] Dial String Inject in FreeSwitch
In-Reply-To: <622bedea1002220909x6694fdfdr6dd43f30b056edb2@mail.gmail.com>
References: <622bedea1002220758r3fb59040g4a12af7996d289b5@mail.gmail.com> 
	<191c3a031002220819j462ee4c1v3e573124bf161d5b@mail.gmail.com> 
	<622bedea1002220833q2a14a9fbh280d0944f5b86fee@mail.gmail.com> 
	<191c3a031002220847pa6ca94ey94fa526dad3583d5@mail.gmail.com> 
	<622bedea1002220909x6694fdfdr6dd43f30b056edb2@mail.gmail.com>
Message-ID: <7b197bef1002220926k5c659d67y1b2e6b843a97afee@mail.gmail.com>

Eder,

If you fear people can do such *really stupid* things, and this is
nice from you, please add something to the wiki, for example a
paragraph in the dialplan page, or whatever, explaining why this is a
stupid thing.

If you publish a page in your blog, that look like a security alert,
or that you found a security flaw in FS, people will rightly think
that you are just looking for some attention in the search engines,
and to bring viewers to your page.

Also, in doing so, you push non technical people to think there is a
security problem in FS, and this is really a big damage to the
project. Because it is not true, it is just how it look like in your
page.

So, delete that page, and add something to the wiki, if you care about
telling people not to do stupid things.

But please, be aware that your page, the page you published, is really
something that do a damage and put a bad light on a project, and there
is no one reason for doing this.

-giovanni



On Mon, Feb 22, 2010 at 6:09 PM, Eder Souza <ederwander at gmail.com> wrote:
> i prefer FreeSwitch im left Asterisk
>
> FreeSwitch is Very Very betther then Asterisk in my option !!
>
>
> my intention is just say dont use (.*),?(.+)? or combinations of this
> regular expressions, for me FreeSwitch?is the betther??!!
>
>
>
> On Mon, Feb 22, 2010 at 1:47 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>>
>> To me it sounds like a way to sound the alarms and bring negative
>> attention.
>>
>> For instance, if you were sincerely concerned, you could have told us
>> about your discovery privately first, and we could feature a story on our
>> own site warning people of this danger and reminding them how to compose
>> extension properly.
>>
>> The posting was instead made like a big public announcement calling our
>> software "imperfect".
>> Yes it is imperfect, It can't properly detect someone being a moron 100%
>> of the time but it sure tries it's darndest.
>>
>>
>>
>>
>> On Mon, Feb 22, 2010 at 10:33 AM, Eder Souza <ederwander at gmail.com> wrote:
>>>
>>> Antony i dont see why ??
>>>
>>>
>>> this is just one alert for all comunity of danger in the use of regular
>>> expression (.*) or (.*) ...
>>>
>>> many peoples can make dialplans?witch use of this expressions ...
>>>
>>>
>>>
>>>
>>>
>>> On Mon, Feb 22, 2010 at 1:19 PM, Anthony Minessale
>>> <anthony.minessale at gmail.com> wrote:
>>>>
>>>> Please do not use our project to try to make your blog more popular.
>>>>
>>>> Your example requires you to prepare an intentional specific extension
>>>> on the FreeSWITCH custom made for your attack. It?s like saying if you leave
>>>> your door wide open at your house and call and tell someone, they can come
>>>> and rob you at 8:30.
>>>>
>>>> This extension is also vulnerable ?by virtue of the stupidity of the
>>>> composer?
>>>>
>>>> <extension name=?please-hack-me?/>
>>>> ? <condition>
>>>> ?? <action application=?system? data=?${destination_number}?/>
>>>> ? </condition>
>>>> </extension>
>>>>
>>>> You should not allow tainted data from outside system to be fed directly
>>>> into your code. There is a regex system in place to extract legitimate data
>>>> from the user tainted input and safeguard against this.
>>>>
>>>>
>>>>
>>>> On Mon, Feb 22, 2010 at 9:58 AM, Eder Souza <ederwander at gmail.com>
>>>> wrote:
>>>>>
>>>>>
>>>>> http://ederwander.wordpress.com/2010/02/22/dial-string-inject-in-freeswitch/
>>>>>
>>>>> just for yours informations i?write this article my test for injections
>>>>> in freesitch
>>>>>
>>>>> version of my tests
>>>>>
>>>>> freeswitch at internal> version
>>>>> FreeSWITCH Version 1.0.5-20100218-0400 (hacked)
>>>>> freeswitch at internal>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> Anthony Minessale II
>>>>
>>>> FreeSWITCH http://www.freeswitch.org/
>>>> ClueCon http://www.cluecon.com/
>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>
>>>> AIM: anthm
>>>> MSN:anthony_minessale at hotmail.com
>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>>> IRC: irc.freenode.net #freeswitch
>>>>
>>>> FreeSWITCH Developer Conference
>>>> sip:888 at conference.freeswitch.org
>>>> iax:guest at conference.freeswitch.org/888
>>>> googletalk:conf+888 at conference.freeswitch.org
>>>> pstn:+19193869900
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618



From ivdreg at gmail.com  Mon Feb 22 09:28:43 2010
From: ivdreg at gmail.com (ivdreg ivdreg)
Date: Mon, 22 Feb 2010 19:28:43 +0200
Subject: [Freeswitch-users] SDP With T.38 in INVITE Problem
In-Reply-To: <4B82BA90.10709@coppice.org>
References: <b437a0e1002190242y2aa64696j91e8bc76f7ae2a92@mail.gmail.com>
	<b437a0e1002220448g3e3bda9fpa283e93a11d319f1@mail.gmail.com>
	<24736FFE-BEEF-4D34-BBA7-77B6B6AE0EBD@jerris.com>
	<b437a0e1002220711u1bfd138l8a67b4e0b024d856@mail.gmail.com>
	<93769c21002220729q5bf36eadk80726aa8c00e3bd@mail.gmail.com>
	<b437a0e1002220849k16fd87ffl5a7257d80e43b036@mail.gmail.com>
	<4B82BA90.10709@coppice.org>
Message-ID: <b437a0e1002220928o2df6b326m6b928d78e025293e@mail.gmail.com>

It comes form subscriber (ATCOM CPEs). As you know OpenSIPS is just a proxy
so cannot generate or rewrite SDP (generally speaking). Yes, I cannot fix
CPEs only server :(

Regards

2010/2/22 Steve Underwood <steveu at coppice.org>

> Hi Michael,
>
> On 02/23/2010 12:49 AM, ivdreg ivdreg wrote:
> > Hi Michael,
> >
> > As I said in a previous mails I know exactly what is happening.
> > In working setup:
> >
> > ITSP ---> YATE (GW - Frontend) ---> FreeSwitch (routing
> > server/xml_curl) ---> YATE (GW - SIP Interop) ---> OpenSIPS --->
> > Subscriber.
> >
> > I prefer to change YATE (GW - Frontend) and YATE (GW - SIP Interop)
> > with FreeSwitch for some reasons. The problem is:
> >
> > INVITE comes from ITSP and goes to subscriber via OpenSIPS. INIVITE
> > between FreeSwitch (routing server) and YATE (GW - SIP Interop)
> > contains SDP:
> > m=audio 21108 RTP/AVP 18 4 8 0
> > c=IN IP4 10.10.1.110
> > a=rtpmap:18 G729/8000
> > a=rtpmap:4 G723/8000
> > a=rtpmap:8 PCMA/8000
> > a=rtpmap:0 PCMU/8000
> > m=image 21108 udptl t38
> > c=IN IP4 10.10.1.110
> > a=T38FaxVersion:0
> > a=T38MaxBitRate:14400
> > a=T38FaxUdpEC:t38UDPRedundancy
> > a=T38FaxRateManagement:
> > transferredTCF
> >
> > And reply 200 OK contains in SDP:
> > *m=audio 34788 RTP/AVP 8*
> > a=rtpmap:8 PCMA/8000
> > a=silenceSupp:off - - - -
> > a=ptime:20
> >
> > Reply 200 OK SDP between YATE (GW - SIP Interop) and OpenSIPS contains
> > in SDP:
> > *m=audio 16330 RTP/AVP 8*
> > a=rtpmap:8 PCMA/8000
> > a=silenceSupp:off - - - -
> > a=ptime:20
> > *m=image 0 udptl 19*
> >
> > In this case everything works fine. Line *m=image 0 udptl 19 *is
> > removed by YATE.
> > But same test with FreeSWITCH on the place of YATE (GW - SIP Interop)
> > *"m=image 0 udptl 19" *call brakes as you can see in my first mail.
> >
> > I don't want to compare or discus YATE and FS functionality or
> > something else. I just see difference in behavior and because I'm not
> > a big expert don't know witch implementation is more accurate
> > according standards. And second: Is it impossible for me to upgrade
> > all CPE so only thing I can do is to fix it on server side. That is
> > because I ask for a help.
> You said the the broken line "m=image 0 udptl 19" goes from OpenSIPS to
> YATE. Do you know if it originates from the OpenSIPS box or the
> subscriber? If it originates from the OpenSIPS box it should be reported
> to them. If its from the subscriber, well...... your chances of getting
> anything fixed are usually small.
>
> Steve
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100222/a5f50c8b/attachment-0002.html 

From mike at jerris.com  Mon Feb 22 09:29:22 2010
From: mike at jerris.com (Michael Jerris)
Date: Mon, 22 Feb 2010 12:29:22 -0500
Subject: [Freeswitch-users] SDP With T.38 in INVITE Problem
In-Reply-To: <4B82BA90.10709@coppice.org>
References: <b437a0e1002190242y2aa64696j91e8bc76f7ae2a92@mail.gmail.com>
	<b437a0e1002220448g3e3bda9fpa283e93a11d319f1@mail.gmail.com>
	<24736FFE-BEEF-4D34-BBA7-77B6B6AE0EBD@jerris.com>
	<b437a0e1002220711u1bfd138l8a67b4e0b024d856@mail.gmail.com>
	<93769c21002220729q5bf36eadk80726aa8c00e3bd@mail.gmail.com>
	<b437a0e1002220849k16fd87ffl5a7257d80e43b036@mail.gmail.com>
	<4B82BA90.10709@coppice.org>
Message-ID: <93769c21002220929t1dbba5bcm3d9200f68a9e1800@mail.gmail.com>

the port 0 with PT of 19 is sofia rejecting the sdp becuase we don't support
it.

On Mon, Feb 22, 2010 at 12:10 PM, Steve Underwood <steveu at coppice.org>wrote:

> Hi Michael,
>
> On 02/23/2010 12:49 AM, ivdreg ivdreg wrote:
> > Hi Michael,
> >
> > As I said in a previous mails I know exactly what is happening.
> > In working setup:
> >
> > ITSP ---> YATE (GW - Frontend) ---> FreeSwitch (routing
> > server/xml_curl) ---> YATE (GW - SIP Interop) ---> OpenSIPS --->
> > Subscriber.
> >
> > I prefer to change YATE (GW - Frontend) and YATE (GW - SIP Interop)
> > with FreeSwitch for some reasons. The problem is:
> >
> > INVITE comes from ITSP and goes to subscriber via OpenSIPS. INIVITE
> > between FreeSwitch (routing server) and YATE (GW - SIP Interop)
> > contains SDP:
> > m=audio 21108 RTP/AVP 18 4 8 0
> > c=IN IP4 10.10.1.110
> > a=rtpmap:18 G729/8000
> > a=rtpmap:4 G723/8000
> > a=rtpmap:8 PCMA/8000
> > a=rtpmap:0 PCMU/8000
> > m=image 21108 udptl t38
> > c=IN IP4 10.10.1.110
> > a=T38FaxVersion:0
> > a=T38MaxBitRate:14400
> > a=T38FaxUdpEC:t38UDPRedundancy
> > a=T38FaxRateManagement:
> > transferredTCF
> >
> > And reply 200 OK contains in SDP:
> > *m=audio 34788 RTP/AVP 8*
> > a=rtpmap:8 PCMA/8000
> > a=silenceSupp:off - - - -
> > a=ptime:20
> >
> > Reply 200 OK SDP between YATE (GW - SIP Interop) and OpenSIPS contains
> > in SDP:
> > *m=audio 16330 RTP/AVP 8*
> > a=rtpmap:8 PCMA/8000
> > a=silenceSupp:off - - - -
> > a=ptime:20
> > *m=image 0 udptl 19*
> >
> > In this case everything works fine. Line *m=image 0 udptl 19 *is
> > removed by YATE.
> > But same test with FreeSWITCH on the place of YATE (GW - SIP Interop)
> > *"m=image 0 udptl 19" *call brakes as you can see in my first mail.
> >
> > I don't want to compare or discus YATE and FS functionality or
> > something else. I just see difference in behavior and because I'm not
> > a big expert don't know witch implementation is more accurate
> > according standards. And second: Is it impossible for me to upgrade
> > all CPE so only thing I can do is to fix it on server side. That is
> > because I ask for a help.
> You said the the broken line "m=image 0 udptl 19" goes from OpenSIPS to
> YATE. Do you know if it originates from the OpenSIPS box or the
> subscriber? If it originates from the OpenSIPS box it should be reported
> to them. If its from the subscriber, well...... your chances of getting
> anything fixed are usually small.
>
> Steve
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100222/8f3184e4/attachment-0002.html 

From mike at jerris.com  Mon Feb 22 09:32:45 2010
From: mike at jerris.com (Michael Jerris)
Date: Mon, 22 Feb 2010 12:32:45 -0500
Subject: [Freeswitch-users] Dial String Inject in FreeSwitch
In-Reply-To: <622bedea1002220909x6694fdfdr6dd43f30b056edb2@mail.gmail.com>
References: <622bedea1002220758r3fb59040g4a12af7996d289b5@mail.gmail.com>
	<191c3a031002220819j462ee4c1v3e573124bf161d5b@mail.gmail.com>
	<622bedea1002220833q2a14a9fbh280d0944f5b86fee@mail.gmail.com>
	<191c3a031002220847pa6ca94ey94fa526dad3583d5@mail.gmail.com>
	<622bedea1002220909x6694fdfdr6dd43f30b056edb2@mail.gmail.com>
Message-ID: <93769c21002220932l40e26365u33b034636dc44949@mail.gmail.com>

a good response to this would be to put appropriate notes on the wiki about
what is good practice in this respect.  Perhaps a patch to the default
configs to add notes with an extra warning would be good as well.

Mike

On Mon, Feb 22, 2010 at 12:09 PM, Eder Souza <ederwander at gmail.com> wrote:

> i prefer FreeSwitch im left Asterisk
>
> FreeSwitch is Very Very betther then Asterisk in my option !!
>
>
> my intention is just say dont use (.*), (.+)  or combinations of this
> regular expressions, for me FreeSwitch is the betther  !!
>
> On Mon, Feb 22, 2010 at 1:47 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> To me it sounds like a way to sound the alarms and bring negative
>> attention.
>>
>> For instance, if you were sincerely concerned, you could have told us
>> about your discovery privately first, and we could feature a story on our
>> own site warning people of this danger and reminding them how to compose
>> extension properly.
>>
>> The posting was instead made like a big public announcement calling our
>> software "imperfect".
>> Yes it is imperfect, It can't properly detect someone being a moron 100%
>> of the time but it sure tries it's darndest.
>>
>>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100222/ed2d65da/attachment-0002.html 

From null at invalid.name  Mon Feb 22 09:35:53 2010
From: null at invalid.name (Dan Lane)
Date: Mon, 22 Feb 2010 17:35:53 +0000
Subject: [Freeswitch-users] Dial String Inject in FreeSwitch
In-Reply-To: <622bedea1002220758r3fb59040g4a12af7996d289b5@mail.gmail.com>
References: <622bedea1002220758r3fb59040g4a12af7996d289b5@mail.gmail.com>
Message-ID: <b042b23b1002220935k9e9c281jefc8202e21159b20@mail.gmail.com>

On Mon, Feb 22, 2010 at 3:58 PM, Eder Souza <ederwander at gmail.com> wrote:
> http://ederwander.wordpress.com/2010/02/22/dial-string-inject-in-freeswitch/
>
> just for yours informations i?write this article my test for injections in
> freesitch
>
> version of my tests

I look forward to next week's blog post where you reveal that "rm -rf
/" results in unexpected data loss.



From null at invalid.name  Mon Feb 22 09:39:20 2010
From: null at invalid.name (Dan Lane)
Date: Mon, 22 Feb 2010 17:39:20 +0000
Subject: [Freeswitch-users] 8000 rate .wav files
In-Reply-To: <4B82BD24.2030108@marketelsystems.com>
References: <4B82BD24.2030108@marketelsystems.com>
Message-ID: <b042b23b1002220939v42633482j1084100413f8c52f@mail.gmail.com>

On Mon, Feb 22, 2010 at 5:21 PM, Mark Sobkow
<m.sobkow at marketelsystems.com> wrote:
> I've got the 8000 sample rate .wav files installed for Freeswitch.
> According to the logs, my SIP phone is connecting with an 8000 rate.
> However, when I try to play_and_get_digits using those sound files, I
> get errors:
>
> 2010-02-22 11:15:22.029347 [ERR] switch_core_file.c:122 Invalid file
> format [wav] for
> [/opt/freeswitch/sounds/en/us/callie/conference/8000/conf-pin.wav]!
> 2010-02-22 11:15:22.029347 [ERR] switch_core_file.c:122 Invalid file
> format [wav] for
> [/opt/freeswitch/sounds/en/us/callie/conference/8000/conf-bad-pin.wav]!
>
> Aren't .wav files supposed to be compatible with all codecs for
> playback? ?If not, what do I have to do to convert them to the proper
> formats? ?How do I find out what the proper formats are?

Try resampling the file using sox... something like the following
should do the trick:

sox input.wav -r 8000 -c 1 -s -w output.wav resample -ql



From ederwander at gmail.com  Mon Feb 22 09:39:43 2010
From: ederwander at gmail.com (Eder Souza)
Date: Mon, 22 Feb 2010 14:39:43 -0300
Subject: [Freeswitch-users] Dial String Inject in FreeSwitch
In-Reply-To: <7b197bef1002220926k5c659d67y1b2e6b843a97afee@mail.gmail.com>
References: <622bedea1002220758r3fb59040g4a12af7996d289b5@mail.gmail.com>
	<191c3a031002220819j462ee4c1v3e573124bf161d5b@mail.gmail.com>
	<622bedea1002220833q2a14a9fbh280d0944f5b86fee@mail.gmail.com>
	<191c3a031002220847pa6ca94ey94fa526dad3583d5@mail.gmail.com>
	<622bedea1002220909x6694fdfdr6dd43f30b056edb2@mail.gmail.com>
	<7b197bef1002220926k5c659d67y1b2e6b843a97afee@mail.gmail.com>
Message-ID: <622bedea1002220939y393688d2sb25161c162a04795@mail.gmail.com>

yeah can somebody make one wiki for this alert??

im make down my link page now to prevent thes problems !!

OK

On Mon, Feb 22, 2010 at 2:26 PM, Giovanni Maruzzelli <gmaruzz at celliax.org>wrote:

> Eder,
>
> If you fear people can do such *really stupid* things, and this is
> nice from you, please add something to the wiki, for example a
> paragraph in the dialplan page, or whatever, explaining why this is a
> stupid thing.
>
> If you publish a page in your blog, that look like a security alert,
> or that you found a security flaw in FS, people will rightly think
> that you are just looking for some attention in the search engines,
> and to bring viewers to your page.
>
> Also, in doing so, you push non technical people to think there is a
> security problem in FS, and this is really a big damage to the
> project. Because it is not true, it is just how it look like in your
> page.
>
> So, delete that page, and add something to the wiki, if you care about
> telling people not to do stupid things.
>
> But please, be aware that your page, the page you published, is really
> something that do a damage and put a bad light on a project, and there
> is no one reason for doing this.
>
> -giovanni
>
>
>
> On Mon, Feb 22, 2010 at 6:09 PM, Eder Souza <ederwander at gmail.com> wrote:
> > i prefer FreeSwitch im left Asterisk
> >
> > FreeSwitch is Very Very betther then Asterisk in my option !!
> >
> >
> > my intention is just say dont use (.*), (.+)  or combinations of this
> > regular expressions, for me FreeSwitch is the betther  !!
> >
> >
> >
> > On Mon, Feb 22, 2010 at 1:47 PM, Anthony Minessale
> > <anthony.minessale at gmail.com> wrote:
> >>
> >> To me it sounds like a way to sound the alarms and bring negative
> >> attention.
> >>
> >> For instance, if you were sincerely concerned, you could have told us
> >> about your discovery privately first, and we could feature a story on
> our
> >> own site warning people of this danger and reminding them how to compose
> >> extension properly.
> >>
> >> The posting was instead made like a big public announcement calling our
> >> software "imperfect".
> >> Yes it is imperfect, It can't properly detect someone being a moron 100%
> >> of the time but it sure tries it's darndest.
> >>
> >>
> >>
> >>
> >> On Mon, Feb 22, 2010 at 10:33 AM, Eder Souza <ederwander at gmail.com>
> wrote:
> >>>
> >>> Antony i dont see why ??
> >>>
> >>>
> >>> this is just one alert for all comunity of danger in the use of regular
> >>> expression (.*) or (.*) ...
> >>>
> >>> many peoples can make dialplans witch use of this expressions ...
> >>>
> >>>
> >>>
> >>>
> >>>
> >>> On Mon, Feb 22, 2010 at 1:19 PM, Anthony Minessale
> >>> <anthony.minessale at gmail.com> wrote:
> >>>>
> >>>> Please do not use our project to try to make your blog more popular.
> >>>>
> >>>> Your example requires you to prepare an intentional specific extension
> >>>> on the FreeSWITCH custom made for your attack. It?s like saying if you
> leave
> >>>> your door wide open at your house and call and tell someone, they can
> come
> >>>> and rob you at 8:30.
> >>>>
> >>>> This extension is also vulnerable ?by virtue of the stupidity of the
> >>>> composer?
> >>>>
> >>>> <extension name=?please-hack-me?/>
> >>>>   <condition>
> >>>>    <action application=?system? data=?${destination_number}?/>
> >>>>   </condition>
> >>>> </extension>
> >>>>
> >>>> You should not allow tainted data from outside system to be fed
> directly
> >>>> into your code. There is a regex system in place to extract legitimate
> data
> >>>> from the user tainted input and safeguard against this.
> >>>>
> >>>>
> >>>>
> >>>> On Mon, Feb 22, 2010 at 9:58 AM, Eder Souza <ederwander at gmail.com>
> >>>> wrote:
> >>>>>
> >>>>>
> >>>>>
> http://ederwander.wordpress.com/2010/02/22/dial-string-inject-in-freeswitch/
> >>>>>
> >>>>> just for yours informations i write this article my test for
> injections
> >>>>> in freesitch
> >>>>>
> >>>>> version of my tests
> >>>>>
> >>>>> freeswitch at internal> version
> >>>>> FreeSWITCH Version 1.0.5-20100218-0400 (hacked)
> >>>>> freeswitch at internal>
> >>>>>
> >>>>>
> >>>>>
> >>>>>
> >>>>> _______________________________________________
> >>>>> FreeSWITCH-users mailing list
> >>>>> FreeSWITCH-users at lists.freeswitch.org
> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>>
> >>>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>> http://www.freeswitch.org
> >>>>>
> >>>>
> >>>>
> >>>>
> >>>> --
> >>>> Anthony Minessale II
> >>>>
> >>>> FreeSWITCH http://www.freeswitch.org/
> >>>> ClueCon http://www.cluecon.com/
> >>>> Twitter: http://twitter.com/FreeSWITCH_wire
> >>>>
> >>>> AIM: anthm
> >>>> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
> >>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >>>> IRC: irc.freenode.net #freeswitch
> >>>>
> >>>> FreeSWITCH Developer Conference
> >>>> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> >>>> iax:guest at conference.freeswitch.org/888
> >>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >>>> pstn:+19193869900
> >>>>
> >>>> _______________________________________________
> >>>> FreeSWITCH-users mailing list
> >>>> FreeSWITCH-users at lists.freeswitch.org
> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>> http://www.freeswitch.org
> >>>>
> >>>
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>>
> >>
> >>
> >>
> >> --
> >> Anthony Minessale II
> >>
> >> FreeSWITCH http://www.freeswitch.org/
> >> ClueCon http://www.cluecon.com/
> >> Twitter: http://twitter.com/FreeSWITCH_wire
> >>
> >> AIM: anthm
> >> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >> IRC: irc.freenode.net #freeswitch
> >>
> >> FreeSWITCH Developer Conference
> >> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> >> iax:guest at conference.freeswitch.org/888
> >> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >> pstn:+19193869900
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100222/9d2fd1e4/attachment-0002.html 

From ederwander at gmail.com  Mon Feb 22 09:45:49 2010
From: ederwander at gmail.com (Eder Souza)
Date: Mon, 22 Feb 2010 14:45:49 -0300
Subject: [Freeswitch-users] Dial String Inject in FreeSwitch
In-Reply-To: <622bedea1002220939y393688d2sb25161c162a04795@mail.gmail.com>
References: <622bedea1002220758r3fb59040g4a12af7996d289b5@mail.gmail.com>
	<191c3a031002220819j462ee4c1v3e573124bf161d5b@mail.gmail.com>
	<622bedea1002220833q2a14a9fbh280d0944f5b86fee@mail.gmail.com>
	<191c3a031002220847pa6ca94ey94fa526dad3583d5@mail.gmail.com>
	<622bedea1002220909x6694fdfdr6dd43f30b056edb2@mail.gmail.com>
	<7b197bef1002220926k5c659d67y1b2e6b843a97afee@mail.gmail.com>
	<622bedea1002220939y393688d2sb25161c162a04795@mail.gmail.com>
Message-ID: <622bedea1002220945r5fd10a4cp19ed1244a67374d9@mail.gmail.com>

Link Down :-)

i thaks if somebody create one wiki witch this alert


Eng Eder de Souza

On Mon, Feb 22, 2010 at 2:39 PM, Eder Souza <ederwander at gmail.com> wrote:

> yeah can somebody make one wiki for this alert??
>
> im make down my link page now to prevent thes problems !!
>
> OK
>
>   On Mon, Feb 22, 2010 at 2:26 PM, Giovanni Maruzzelli <
> gmaruzz at celliax.org> wrote:
>
>> Eder,
>>
>> If you fear people can do such *really stupid* things, and this is
>> nice from you, please add something to the wiki, for example a
>> paragraph in the dialplan page, or whatever, explaining why this is a
>> stupid thing.
>>
>> If you publish a page in your blog, that look like a security alert,
>> or that you found a security flaw in FS, people will rightly think
>> that you are just looking for some attention in the search engines,
>> and to bring viewers to your page.
>>
>> Also, in doing so, you push non technical people to think there is a
>> security problem in FS, and this is really a big damage to the
>> project. Because it is not true, it is just how it look like in your
>> page.
>>
>> So, delete that page, and add something to the wiki, if you care about
>> telling people not to do stupid things.
>>
>> But please, be aware that your page, the page you published, is really
>> something that do a damage and put a bad light on a project, and there
>> is no one reason for doing this.
>>
>> -giovanni
>>
>>
>>
>> On Mon, Feb 22, 2010 at 6:09 PM, Eder Souza <ederwander at gmail.com> wrote:
>> > i prefer FreeSwitch im left Asterisk
>> >
>> > FreeSwitch is Very Very betther then Asterisk in my option !!
>> >
>> >
>> > my intention is just say dont use (.*), (.+)  or combinations of this
>> > regular expressions, for me FreeSwitch is the betther  !!
>> >
>> >
>> >
>> > On Mon, Feb 22, 2010 at 1:47 PM, Anthony Minessale
>> > <anthony.minessale at gmail.com> wrote:
>> >>
>> >> To me it sounds like a way to sound the alarms and bring negative
>> >> attention.
>> >>
>> >> For instance, if you were sincerely concerned, you could have told us
>> >> about your discovery privately first, and we could feature a story on
>> our
>> >> own site warning people of this danger and reminding them how to
>> compose
>> >> extension properly.
>> >>
>> >> The posting was instead made like a big public announcement calling our
>> >> software "imperfect".
>> >> Yes it is imperfect, It can't properly detect someone being a moron
>> 100%
>> >> of the time but it sure tries it's darndest.
>> >>
>> >>
>> >>
>> >>
>> >> On Mon, Feb 22, 2010 at 10:33 AM, Eder Souza <ederwander at gmail.com>
>> wrote:
>> >>>
>> >>> Antony i dont see why ??
>> >>>
>> >>>
>> >>> this is just one alert for all comunity of danger in the use of
>> regular
>> >>> expression (.*) or (.*) ...
>> >>>
>> >>> many peoples can make dialplans witch use of this expressions ...
>> >>>
>> >>>
>> >>>
>> >>>
>> >>>
>> >>> On Mon, Feb 22, 2010 at 1:19 PM, Anthony Minessale
>> >>> <anthony.minessale at gmail.com> wrote:
>> >>>>
>> >>>> Please do not use our project to try to make your blog more popular.
>> >>>>
>> >>>> Your example requires you to prepare an intentional specific
>> extension
>> >>>> on the FreeSWITCH custom made for your attack. It?s like saying if
>> you leave
>> >>>> your door wide open at your house and call and tell someone, they can
>> come
>> >>>> and rob you at 8:30.
>> >>>>
>> >>>> This extension is also vulnerable ?by virtue of the stupidity of the
>> >>>> composer?
>> >>>>
>> >>>> <extension name=?please-hack-me?/>
>> >>>>   <condition>
>> >>>>    <action application=?system? data=?${destination_number}?/>
>> >>>>   </condition>
>> >>>> </extension>
>> >>>>
>> >>>> You should not allow tainted data from outside system to be fed
>> directly
>> >>>> into your code. There is a regex system in place to extract
>> legitimate data
>> >>>> from the user tainted input and safeguard against this.
>> >>>>
>> >>>>
>> >>>>
>> >>>> On Mon, Feb 22, 2010 at 9:58 AM, Eder Souza <ederwander at gmail.com>
>> >>>> wrote:
>> >>>>>
>> >>>>>
>> >>>>>
>> http://ederwander.wordpress.com/2010/02/22/dial-string-inject-in-freeswitch/
>> >>>>>
>> >>>>> just for yours informations i write this article my test for
>> injections
>> >>>>> in freesitch
>> >>>>>
>> >>>>> version of my tests
>> >>>>>
>> >>>>> freeswitch at internal> version
>> >>>>> FreeSWITCH Version 1.0.5-20100218-0400 (hacked)
>> >>>>> freeswitch at internal>
>> >>>>>
>> >>>>>
>> >>>>>
>> >>>>>
>> >>>>> _______________________________________________
>> >>>>> FreeSWITCH-users mailing list
>> >>>>> FreeSWITCH-users at lists.freeswitch.org
>> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>>>
>> >>>>> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>>>> http://www.freeswitch.org
>> >>>>>
>> >>>>
>> >>>>
>> >>>>
>> >>>> --
>> >>>> Anthony Minessale II
>> >>>>
>> >>>> FreeSWITCH http://www.freeswitch.org/
>> >>>> ClueCon http://www.cluecon.com/
>> >>>> Twitter: http://twitter.com/FreeSWITCH_wire
>> >>>>
>> >>>> AIM: anthm
>> >>>> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
>> >>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> >>>> IRC: irc.freenode.net #freeswitch
>> >>>>
>> >>>> FreeSWITCH Developer Conference
>> >>>> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>> >>>> iax:guest at conference.freeswitch.org/888
>> >>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> >>>> pstn:+19193869900
>> >>>>
>> >>>> _______________________________________________
>> >>>> FreeSWITCH-users mailing list
>> >>>> FreeSWITCH-users at lists.freeswitch.org
>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>>> http://www.freeswitch.org
>> >>>>
>> >>>
>> >>>
>> >>> _______________________________________________
>> >>> FreeSWITCH-users mailing list
>> >>> FreeSWITCH-users at lists.freeswitch.org
>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> http://www.freeswitch.org
>> >>>
>> >>
>> >>
>> >>
>> >> --
>> >> Anthony Minessale II
>> >>
>> >> FreeSWITCH http://www.freeswitch.org/
>> >> ClueCon http://www.cluecon.com/
>> >> Twitter: http://twitter.com/FreeSWITCH_wire
>> >>
>> >> AIM: anthm
>> >> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
>> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> >> IRC: irc.freenode.net #freeswitch
>> >>
>> >> FreeSWITCH Developer Conference
>> >> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>> >> iax:guest at conference.freeswitch.org/888
>> >> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> >> pstn:+19193869900
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>>
>>
>> --
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> Cell : +39-347-2665618
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100222/a44409e9/attachment-0002.html 

From niall.crosby at gmail.com  Mon Feb 22 10:07:42 2010
From: niall.crosby at gmail.com (Niall Crosby)
Date: Mon, 22 Feb 2010 18:07:42 +0000
Subject: [Freeswitch-users] sending custom events
Message-ID: <4aec92831002221007r3084f75l3db5c9d129625539@mail.gmail.com>

Dear List,

Can someone give an example of sending a custom event use the Event Socket
Layer directly, and not one of the wrapper APIs?

I'm writing in Java and have my own ESL wrapper implementation, but can't
work out how to send custom events.

Thanks in advance,
Niall.

-- 
-- 

The information transmitted is intended only for the person or entity to
which it is addressed and may contain confidential and/or privileged
material. Statements and opinions expressed in this e-mail may not represent
those of the sender. Any review, retransmission, dissemination or other use
of, or taking of any action in reliance upon, this information by persons or
entities other than the intended recipient is prohibited. If you received
this in error, please contact the sender immediately and delete the material
from any computer.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100222/075265fa/attachment-0002.html 

From gmaruzz at celliax.org  Mon Feb 22 10:20:37 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Mon, 22 Feb 2010 19:20:37 +0100
Subject: [Freeswitch-users] Dial String Inject in FreeSwitch
In-Reply-To: <622bedea1002220945r5fd10a4cp19ed1244a67374d9@mail.gmail.com>
References: <622bedea1002220758r3fb59040g4a12af7996d289b5@mail.gmail.com> 
	<191c3a031002220819j462ee4c1v3e573124bf161d5b@mail.gmail.com> 
	<622bedea1002220833q2a14a9fbh280d0944f5b86fee@mail.gmail.com> 
	<191c3a031002220847pa6ca94ey94fa526dad3583d5@mail.gmail.com> 
	<622bedea1002220909x6694fdfdr6dd43f30b056edb2@mail.gmail.com> 
	<7b197bef1002220926k5c659d67y1b2e6b843a97afee@mail.gmail.com> 
	<622bedea1002220939y393688d2sb25161c162a04795@mail.gmail.com> 
	<622bedea1002220945r5fd10a4cp19ed1244a67374d9@mail.gmail.com>
Message-ID: <7b197bef1002221020u7b6587f6l4c25b992f2ac01a3@mail.gmail.com>

Thanks a lot, Eder.

If you feel like, you can add a paragraph yourself, then we'll edit if
necessary.

Just let us know.

-giovanni

On Mon, Feb 22, 2010 at 6:45 PM, Eder Souza <ederwander at gmail.com> wrote:
> Link Down :-)
>
> i thaks if somebody create one wiki witch this alert
>
>
> Eng Eder de Souza
>
> On Mon, Feb 22, 2010 at 2:39 PM, Eder Souza <ederwander at gmail.com> wrote:
>>
>> yeah?can somebody make one wiki for this alert??
>>
>>
>> im make down my link page now?to prevent thes problems !!
>>
>> OK
>>
>> On Mon, Feb 22, 2010 at 2:26 PM, Giovanni Maruzzelli <gmaruzz at celliax.org>
>> wrote:
>>>
>>> Eder,
>>>
>>> If you fear people can do such *really stupid* things, and this is
>>> nice from you, please add something to the wiki, for example a
>>> paragraph in the dialplan page, or whatever, explaining why this is a
>>> stupid thing.
>>>
>>> If you publish a page in your blog, that look like a security alert,
>>> or that you found a security flaw in FS, people will rightly think
>>> that you are just looking for some attention in the search engines,
>>> and to bring viewers to your page.
>>>
>>> Also, in doing so, you push non technical people to think there is a
>>> security problem in FS, and this is really a big damage to the
>>> project. Because it is not true, it is just how it look like in your
>>> page.
>>>
>>> So, delete that page, and add something to the wiki, if you care about
>>> telling people not to do stupid things.
>>>
>>> But please, be aware that your page, the page you published, is really
>>> something that do a damage and put a bad light on a project, and there
>>> is no one reason for doing this.
>>>
>>> -giovanni
>>>
>>>
>>>
>>> On Mon, Feb 22, 2010 at 6:09 PM, Eder Souza <ederwander at gmail.com> wrote:
>>> > i prefer FreeSwitch im left Asterisk
>>> >
>>> > FreeSwitch is Very Very betther then Asterisk in my option !!
>>> >
>>> >
>>> > my intention is just say dont use (.*),?(.+)? or combinations of this
>>> > regular expressions, for me FreeSwitch?is the betther??!!
>>> >
>>> >
>>> >
>>> > On Mon, Feb 22, 2010 at 1:47 PM, Anthony Minessale
>>> > <anthony.minessale at gmail.com> wrote:
>>> >>
>>> >> To me it sounds like a way to sound the alarms and bring negative
>>> >> attention.
>>> >>
>>> >> For instance, if you were sincerely concerned, you could have told us
>>> >> about your discovery privately first, and we could feature a story on
>>> >> our
>>> >> own site warning people of this danger and reminding them how to
>>> >> compose
>>> >> extension properly.
>>> >>
>>> >> The posting was instead made like a big public announcement calling
>>> >> our
>>> >> software "imperfect".
>>> >> Yes it is imperfect, It can't properly detect someone being a moron
>>> >> 100%
>>> >> of the time but it sure tries it's darndest.
>>> >>
>>> >>
>>> >>
>>> >>
>>> >> On Mon, Feb 22, 2010 at 10:33 AM, Eder Souza <ederwander at gmail.com>
>>> >> wrote:
>>> >>>
>>> >>> Antony i dont see why ??
>>> >>>
>>> >>>
>>> >>> this is just one alert for all comunity of danger in the use of
>>> >>> regular
>>> >>> expression (.*) or (.*) ...
>>> >>>
>>> >>> many peoples can make dialplans?witch use of this expressions ...
>>> >>>
>>> >>>
>>> >>>
>>> >>>
>>> >>>
>>> >>> On Mon, Feb 22, 2010 at 1:19 PM, Anthony Minessale
>>> >>> <anthony.minessale at gmail.com> wrote:
>>> >>>>
>>> >>>> Please do not use our project to try to make your blog more popular.
>>> >>>>
>>> >>>> Your example requires you to prepare an intentional specific
>>> >>>> extension
>>> >>>> on the FreeSWITCH custom made for your attack. It?s like saying if
>>> >>>> you leave
>>> >>>> your door wide open at your house and call and tell someone, they
>>> >>>> can come
>>> >>>> and rob you at 8:30.
>>> >>>>
>>> >>>> This extension is also vulnerable ?by virtue of the stupidity of the
>>> >>>> composer?
>>> >>>>
>>> >>>> <extension name=?please-hack-me?/>
>>> >>>> ? <condition>
>>> >>>> ?? <action application=?system? data=?${destination_number}?/>
>>> >>>> ? </condition>
>>> >>>> </extension>
>>> >>>>
>>> >>>> You should not allow tainted data from outside system to be fed
>>> >>>> directly
>>> >>>> into your code. There is a regex system in place to extract
>>> >>>> legitimate data
>>> >>>> from the user tainted input and safeguard against this.
>>> >>>>
>>> >>>>
>>> >>>>
>>> >>>> On Mon, Feb 22, 2010 at 9:58 AM, Eder Souza <ederwander at gmail.com>
>>> >>>> wrote:
>>> >>>>>
>>> >>>>>
>>> >>>>>
>>> >>>>> http://ederwander.wordpress.com/2010/02/22/dial-string-inject-in-freeswitch/
>>> >>>>>
>>> >>>>> just for yours informations i?write this article my test for
>>> >>>>> injections
>>> >>>>> in freesitch
>>> >>>>>
>>> >>>>> version of my tests
>>> >>>>>
>>> >>>>> freeswitch at internal> version
>>> >>>>> FreeSWITCH Version 1.0.5-20100218-0400 (hacked)
>>> >>>>> freeswitch at internal>
>>> >>>>>
>>> >>>>>
>>> >>>>>
>>> >>>>>
>>> >>>>> _______________________________________________
>>> >>>>> FreeSWITCH-users mailing list
>>> >>>>> FreeSWITCH-users at lists.freeswitch.org
>>> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>>>>
>>> >>>>>
>>> >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >>>>> http://www.freeswitch.org
>>> >>>>>
>>> >>>>
>>> >>>>
>>> >>>>
>>> >>>> --
>>> >>>> Anthony Minessale II
>>> >>>>
>>> >>>> FreeSWITCH http://www.freeswitch.org/
>>> >>>> ClueCon http://www.cluecon.com/
>>> >>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>> >>>>
>>> >>>> AIM: anthm
>>> >>>> MSN:anthony_minessale at hotmail.com
>>> >>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>> >>>> IRC: irc.freenode.net #freeswitch
>>> >>>>
>>> >>>> FreeSWITCH Developer Conference
>>> >>>> sip:888 at conference.freeswitch.org
>>> >>>> iax:guest at conference.freeswitch.org/888
>>> >>>> googletalk:conf+888 at conference.freeswitch.org
>>> >>>> pstn:+19193869900
>>> >>>>
>>> >>>> _______________________________________________
>>> >>>> FreeSWITCH-users mailing list
>>> >>>> FreeSWITCH-users at lists.freeswitch.org
>>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>>>
>>> >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >>>> http://www.freeswitch.org
>>> >>>>
>>> >>>
>>> >>>
>>> >>> _______________________________________________
>>> >>> FreeSWITCH-users mailing list
>>> >>> FreeSWITCH-users at lists.freeswitch.org
>>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>>
>>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >>> http://www.freeswitch.org
>>> >>>
>>> >>
>>> >>
>>> >>
>>> >> --
>>> >> Anthony Minessale II
>>> >>
>>> >> FreeSWITCH http://www.freeswitch.org/
>>> >> ClueCon http://www.cluecon.com/
>>> >> Twitter: http://twitter.com/FreeSWITCH_wire
>>> >>
>>> >> AIM: anthm
>>> >> MSN:anthony_minessale at hotmail.com
>>> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>> >> IRC: irc.freenode.net #freeswitch
>>> >>
>>> >> FreeSWITCH Developer Conference
>>> >> sip:888 at conference.freeswitch.org
>>> >> iax:guest at conference.freeswitch.org/888
>>> >> googletalk:conf+888 at conference.freeswitch.org
>>> >> pstn:+19193869900
>>> >>
>>> >> _______________________________________________
>>> >> FreeSWITCH-users mailing list
>>> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>
>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> http://www.freeswitch.org
>>> >>
>>> >
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >
>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>> >
>>>
>>>
>>>
>>> --
>>> Sincerely,
>>>
>>> Giovanni Maruzzelli
>>> Cell : +39-347-2665618
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618



From Russell.Mosemann at cune.org  Mon Feb 22 10:27:40 2010
From: Russell.Mosemann at cune.org (Russell.Mosemann at cune.org)
Date: Mon, 22 Feb 2010 18:27:40 -0000
Subject: [Freeswitch-users] Dial String Inject in FreeSwitch
In-Reply-To: <622bedea1002220945r5fd10a4cp19ed1244a67374d9@mail.gmail.com>
Message-ID: <20100222182741.05F7829BF68@cuneorg-email.cune.pri>

> i thaks if somebody create one wiki witch this alert

A place to change is Example 1 of the dialplan XML examples. You can tell
people not to use the catchall expressions, because you cannot trust
information from the sender.

http://wiki.freeswitch.org/wiki/Dialplan_XML

A word of caution could also be added to

http://wiki.freeswitch.org/wiki/Regular_Expression

-- 
Russell Mosemann



________________________________________________________
Concordia University, Nebraska
See http://www.cune.edu/ for the latest news and events!




From dftoro at yahoo.com  Mon Feb 22 10:30:04 2010
From: dftoro at yahoo.com (Diego Toro)
Date: Mon, 22 Feb 2010 10:30:04 -0800 (PST)
Subject: [Freeswitch-users] sending a sip notify with sendevent
In-Reply-To: <4B82BBFB.1040804@xpirio.com>
Message-ID: <126320.178.qm@web33502.mail.mud.yahoo.com>

Read "case SWITCH_EVENT_NOTIFY:" in mod_sofia.c. I suggest you assign call-id in the header of the event.


Diego Toro
http://lacarretade.blogspot.com/


--- On Mon, 2/22/10, Christian L?schenkohl <christian.loeschenkohl at xpirio.com> wrote:

> From: Christian L?schenkohl <christian.loeschenkohl at xpirio.com>
> Subject: [Freeswitch-users] sending a sip notify with sendevent
> To: freeswitch-users at lists.freeswitch.org
> Date: Monday, February 22, 2010, 12:16 PM
> hi
> 
> i try to send a sip notify message to a registered sip
> device
> "sofia profile nat flush_inbound_reg 10 at vts.vie.xpirio.net
> reboot" works, but i need
> to send "check-sync;reboot=false" - so the device does a
> resync and don't do a reboot
> 
> my message looks like this
> 
> sendevent NOTIFY
> profile: nat
> event-string: check-sync;reboot=false
> user: 10
> host: vts.vie.xpirio.net
> content-type: application/simple-message-summary
> 
> if i listen on the loopback interface i do see
> 
> ##
> T 2010/02/22 18:11:59.083204 127.0.0.1:51840 ->
> 127.0.0.1:8021 [AP]
> sendevent NOTIFY
> profile: nat
> event-string: check-sync;reboot=false
> user: 10
> host: vts.vie.xpirio.net
> content-type: application/simple-message-summary
> 
> ##
> T 2010/02/22 18:11:59.084032 127.0.0.1:8021 ->
> 127.0.0.1:51840 [AP]
> Content-Type: command/reply
> Reply-Text: -ERR invalid
> 
> --------
> i don't get what it is wrong. i also rechecked the
> registered user in the sqlite database and this
> looks good to me.
> 
> no message is send to the user.
> 
> we do use multiple domains, so user could also be 10 at somedomain.com
> - or am i wrong on this?
> could somebody please bring some light in this.
> 
> we do use trunk rev. 16631
> 
> br
> 
> 
> 
> -- 
> Ing. Christian L?schenkohl
> Technische Leitung, Forschung & Entwicklung VoIP
> 
> xpirio
> Telekommunikation & Service GmbH
> Lakeside B04
> 9020 Klagenfurt
> Austria
> 
> T? +43 (0) 5 77 11 - 1000
> F? +43 (0) 5 77 11 - 1002
> E? christian.loeschenkohl at xpirio.com
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 


      



From lawwton at gmail.com  Mon Feb 22 10:31:21 2010
From: lawwton at gmail.com (Alfredo Quiroga-Villamil)
Date: Mon, 22 Feb 2010 13:31:21 -0500
Subject: [Freeswitch-users] sending custom events
In-Reply-To: <4aec92831002221007r3084f75l3db5c9d129625539@mail.gmail.com>
References: <4aec92831002221007r3084f75l3db5c9d129625539@mail.gmail.com>
Message-ID: <5fe6fa8f1002221031m79b6fefase512629c7e13fb8a@mail.gmail.com>

Niall:

I am really new to FS so I apologize before hand if my response is not
correct; but here is what I understand so far for Events. Some modules
have events associated with them. Some of the modules have Custom
Events.

a) All the way to the bottom of the link shown below you'll see the
modules that have custom events:

http://wiki.freeswitch.org/wiki/Event_list

b) All events listed shown below:

http://wiki.freeswitch.org/wiki/Event_list

c) In my case for example I am also using Java and opening a
connection using my own wrapper as well to ESL. For a conference, we
then send the following custom event via the socket.
An example of a custom event:

event plain CUSTOM conference::maintenance

Java Sample Code:

            Socket socket = new Socket("API URL", PORT);
            String cmd = String.format("event plain %s %s
conference::maintenance",
                                       FreeSwitchEvent.SHUTDOWN.toString(),
                                       FreeSwitchEvent.CUSTOM.toString());

So there I am registering to a couple of events. One of them being a
custom event for the conference in this case.

Like I said I am really new to FS, so I might not be 100% on the money here.

Regards,

Alfredo


On Mon, Feb 22, 2010 at 1:07 PM, Niall Crosby <niall.crosby at gmail.com> wrote:
>
> Dear List,
> Can someone give an example of sending a custom event use the Event Socket
> Layer directly, and not one of the wrapper APIs?
> I'm writing in Java and have my own ESL wrapper implementation, but can't
> work out how to send custom events.
> Thanks in advance,
> Niall.
>
> --
> --
>
> The information transmitted is intended only for the person or entity to
> which it is addressed and may contain confidential and/or privileged
> material. Statements and opinions expressed in this e-mail may not represent
> those of the sender. Any review, retransmission, dissemination or other use
> of, or taking of any action in reliance upon, this information by persons or
> entities other than the intended recipient is prohibited. If you received
> this in error, please contact the sender immediately and delete the material
> from any computer.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From brian at freeswitch.org  Mon Feb 22 10:35:51 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 22 Feb 2010 12:35:51 -0600
Subject: [Freeswitch-users] sending a sip notify with sendevent
In-Reply-To: <126320.178.qm@web33502.mail.mud.yahoo.com>
References: <126320.178.qm@web33502.mail.mud.yahoo.com>
Message-ID: <3959C202-34BE-4DDF-B387-45C1F702377D@freeswitch.org>

Is this not documented on the wiki yet?

/b

On Feb 22, 2010, at 12:30 PM, Diego Toro wrote:

> Read "case SWITCH_EVENT_NOTIFY:" in mod_sofia.c. I suggest you assign call-id in the header of the event.
> 
> 
> Diego Toro
> http://lacarretade.blogspot.com/

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100222/4eb1eacc/attachment-0002.html 

From ederwander at gmail.com  Mon Feb 22 10:38:53 2010
From: ederwander at gmail.com (Eder Souza)
Date: Mon, 22 Feb 2010 15:38:53 -0300
Subject: [Freeswitch-users] Dial String Inject in FreeSwitch
In-Reply-To: <20100222182741.05F7829BF68@cuneorg-email.cune.pri>
References: <622bedea1002220945r5fd10a4cp19ed1244a67374d9@mail.gmail.com>
	<20100222182741.05F7829BF68@cuneorg-email.cune.pri>
Message-ID: <622bedea1002221038u2a2e4232qd1d15cf006abb7db@mail.gmail.com>

Perfect place lol


On Mon, Feb 22, 2010 at 3:27 PM, <Russell.Mosemann at cune.org> wrote:

> > i thaks if somebody create one wiki witch this alert
>
> A place to change is Example 1 of the dialplan XML examples. You can tell
> people not to use the catchall expressions, because you cannot trust
> information from the sender.
>
> http://wiki.freeswitch.org/wiki/Dialplan_XML
>
> A word of caution could also be added to
>
> http://wiki.freeswitch.org/wiki/Regular_Expression
>
> --
> Russell Mosemann
>
>
>
> ________________________________________________________
> Concordia University, Nebraska
> See http://www.cune.edu/ for the latest news and events!
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100222/6ed6336b/attachment-0002.html 

From anthony.minessale at gmail.com  Mon Feb 22 11:05:24 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 22 Feb 2010 13:05:24 -0600
Subject: [Freeswitch-users] sending a sip notify with sendevent
In-Reply-To: <3959C202-34BE-4DDF-B387-45C1F702377D@freeswitch.org>
References: <126320.178.qm@web33502.mail.mud.yahoo.com>
	<3959C202-34BE-4DDF-B387-45C1F702377D@freeswitch.org>
Message-ID: <191c3a031002221105wa86cdfeq845a2d7cb00a4931@mail.gmail.com>

its at the very least missing the profile name

On Mon, Feb 22, 2010 at 12:35 PM, Brian West <brian at freeswitch.org> wrote:

> Is this not documented on the wiki yet?
>
> /b
>
> On Feb 22, 2010, at 12:30 PM, Diego Toro wrote:
>
> Read "case SWITCH_EVENT_NOTIFY:" in mod_sofia.c. I suggest you assign
> call-id in the header of the event.
>
>
> Diego Toro
> http://lacarretade.blogspot.com/
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100222/86bd786e/attachment-0002.html 

From ederwander at gmail.com  Mon Feb 22 11:13:04 2010
From: ederwander at gmail.com (Eder Souza)
Date: Mon, 22 Feb 2010 16:13:04 -0300
Subject: [Freeswitch-users] Dial String Inject in FreeSwitch
In-Reply-To: <622bedea1002221038u2a2e4232qd1d15cf006abb7db@mail.gmail.com>
References: <622bedea1002220945r5fd10a4cp19ed1244a67374d9@mail.gmail.com>
	<20100222182741.05F7829BF68@cuneorg-email.cune.pri>
	<622bedea1002221038u2a2e4232qd1d15cf006abb7db@mail.gmail.com>
Message-ID: <622bedea1002221113j5ac06477jb24ac51eedcd8d8f@mail.gmail.com>

http://wiki.freeswitch.org/wiki/User:Ederwander




On Mon, Feb 22, 2010 at 3:38 PM, Eder Souza <ederwander at gmail.com> wrote:

> Perfect place lol
>
>
> On Mon, Feb 22, 2010 at 3:27 PM, <Russell.Mosemann at cune.org> wrote:
>
>> > i thaks if somebody create one wiki witch this alert
>>
>> A place to change is Example 1 of the dialplan XML examples. You can tell
>> people not to use the catchall expressions, because you cannot trust
>> information from the sender.
>>
>> http://wiki.freeswitch.org/wiki/Dialplan_XML
>>
>> A word of caution could also be added to
>>
>> http://wiki.freeswitch.org/wiki/Regular_Expression
>>
>> --
>> Russell Mosemann
>>
>>
>>
>> ________________________________________________________
>> Concordia University, Nebraska
>> See http://www.cune.edu/ for the latest news and events!
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100222/302a85da/attachment-0002.html 

From andrew at hijacked.us  Mon Feb 22 11:20:42 2010
From: andrew at hijacked.us (Andrew Thompson)
Date: Mon, 22 Feb 2010 14:20:42 -0500
Subject: [Freeswitch-users] 8000 rate .wav files
In-Reply-To: <4B82BD24.2030108@marketelsystems.com>
References: <4B82BD24.2030108@marketelsystems.com>
Message-ID: <20100222192042.GI8518@hijacked.us>

On Mon, Feb 22, 2010 at 11:21:40AM -0600, Mark Sobkow wrote:
> I've got the 8000 sample rate .wav files installed for Freeswitch.  
> According to the logs, my SIP phone is connecting with an 8000 rate.  
> However, when I try to play_and_get_digits using those sound files, I 
> get errors:
> 
> 2010-02-22 11:15:22.029347 [ERR] switch_core_file.c:122 Invalid file 
> format [wav] for 
> [/opt/freeswitch/sounds/en/us/callie/conference/8000/conf-pin.wav]!
> 2010-02-22 11:15:22.029347 [ERR] switch_core_file.c:122 Invalid file 
> format [wav] for 
> [/opt/freeswitch/sounds/en/us/callie/conference/8000/conf-bad-pin.wav]!
> 
> Aren't .wav files supposed to be compatible with all codecs for 
> playback?  If not, what do I have to do to convert them to the proper 
> formats?  How do I find out what the proper formats are?
> 

Do you have mod_sndfile loaded?

Andrew



From m.sobkow at marketelsystems.com  Mon Feb 22 11:45:19 2010
From: m.sobkow at marketelsystems.com (Mark Sobkow)
Date: Mon, 22 Feb 2010 13:45:19 -0600
Subject: [Freeswitch-users] 8000 rate .wav files
In-Reply-To: <20100222192042.GI8518@hijacked.us>
References: <4B82BD24.2030108@marketelsystems.com>
	<20100222192042.GI8518@hijacked.us>
Message-ID: <4B82DECF.5000300@marketelsystems.com>

An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100222/3bbe847c/attachment-0002.html 

From m.sobkow at marketelsystems.com  Mon Feb 22 12:20:10 2010
From: m.sobkow at marketelsystems.com (Mark Sobkow)
Date: Mon, 22 Feb 2010 14:20:10 -0600
Subject: [Freeswitch-users] Is there any way to loop a dialplan?
Message-ID: <4B82E6FA.3090008@marketelsystems.com>

Let me explain what it is I'm trying to do.  Maybe there's another way 
to achieve it.

When an operator dials in to the log-in line (e.g. Extension 6000), I 
use play_and_get_digits to collect the operator's PIN.  I then need to 
be able to fire up some Erlang (or Javascript) to verify the PIN, and 
after verification, put the call into a park state, collecting the 
UUID.  I then need to fire an event to Erlang passing along the parked 
UUID and the operator's PIN so that Erlang can direct received customer 
calls to the operators based on relatively complex criteria that won't 
fit in a dialplan.

The catch is that when I get a customer call, I collect their info via 
IVR menus, park the call, and fire an event to Erlang with the UUID of 
the parked call and info collected from the IVR.  Erlang analyses the 
info, selects an operator who is free, and bridges the calls.

The problem I'm having is figuring out a way to get the operator leg to 
go back into a park state after handling the customer's call.  Ideally I 
want the operator to be presented with a short IVR to collect info about 
how the call was handled, but I can do that through a custom application 
GUI if I need to.  Regardless, once an operator logs in, they need to 
_stay_ logged in until they explicityly log out, but I can't figure out 
how I'm supposed to do that without some sort of looping capability.

One thing I was thinking that might work is to set up a set of "dummy" 
extensions that I can have Erlang dial and bridge which contain a 
dialplan fragment to collect the IVR call result, park the call, and 
issue the operator PIN and parked UUID again to Erlang.  That way 
between Erlang events and the dialplan fragment I end up with an 
effective "loop".  (Though I've yet to figure out how I can break out of 
that loop.  Maybe it'll have to be an IVR option for logging out.)

Sample code/dialplans would be good, but for now I'll settle for knowing 
whether I'm at least on the right track for how to implement this beast.

Note that I only want to drop into Javascript if I can't figure out how 
to do it with dialplans and Erlang.

Thanks for any ideas and suggestions.

-- 
Mark Sobkow
Senior Developer
MarkeTel Multi-Line Dialing Systems LTD.
428 Victoria Ave
Regina, SK S4N-0P6
Toll-Free: 800-289-8616-X533
Local: 306-359-6893-X533
Fax: 306-359-6879
Email: m.sobkow at marketelsystems.com
Web: http://www.marketelsystems.com 




From ivdreg at gmail.com  Mon Feb 22 12:23:00 2010
From: ivdreg at gmail.com (ivdreg ivdreg)
Date: Mon, 22 Feb 2010 22:23:00 +0200
Subject: [Freeswitch-users] SDP With T.38 in INVITE Problem
In-Reply-To: <93769c21002220929t1dbba5bcm3d9200f68a9e1800@mail.gmail.com>
References: <b437a0e1002190242y2aa64696j91e8bc76f7ae2a92@mail.gmail.com>
	<b437a0e1002220448g3e3bda9fpa283e93a11d319f1@mail.gmail.com>
	<24736FFE-BEEF-4D34-BBA7-77B6B6AE0EBD@jerris.com>
	<b437a0e1002220711u1bfd138l8a67b4e0b024d856@mail.gmail.com>
	<93769c21002220729q5bf36eadk80726aa8c00e3bd@mail.gmail.com>
	<b437a0e1002220849k16fd87ffl5a7257d80e43b036@mail.gmail.com>
	<4B82BA90.10709@coppice.org>
	<93769c21002220929t1dbba5bcm3d9200f68a9e1800@mail.gmail.com>
Message-ID: <b437a0e1002221223w53ef522asc807e56a13d8bc70@mail.gmail.com>

Thanks Michael,

I will try to find some solution/workaround.

2010/2/22 Michael Jerris <mike at jerris.com>

> the port 0 with PT of 19 is sofia rejecting the sdp becuase we don't
> support it.
>
>
> On Mon, Feb 22, 2010 at 12:10 PM, Steve Underwood <steveu at coppice.org>wrote:
>
>> Hi Michael,
>>
>> On 02/23/2010 12:49 AM, ivdreg ivdreg wrote:
>> > Hi Michael,
>> >
>> > As I said in a previous mails I know exactly what is happening.
>> > In working setup:
>> >
>> > ITSP ---> YATE (GW - Frontend) ---> FreeSwitch (routing
>> > server/xml_curl) ---> YATE (GW - SIP Interop) ---> OpenSIPS --->
>> > Subscriber.
>> >
>> > I prefer to change YATE (GW - Frontend) and YATE (GW - SIP Interop)
>> > with FreeSwitch for some reasons. The problem is:
>> >
>> > INVITE comes from ITSP and goes to subscriber via OpenSIPS. INIVITE
>> > between FreeSwitch (routing server) and YATE (GW - SIP Interop)
>> > contains SDP:
>> > m=audio 21108 RTP/AVP 18 4 8 0
>> > c=IN IP4 10.10.1.110
>> > a=rtpmap:18 G729/8000
>> > a=rtpmap:4 G723/8000
>> > a=rtpmap:8 PCMA/8000
>> > a=rtpmap:0 PCMU/8000
>> > m=image 21108 udptl t38
>> > c=IN IP4 10.10.1.110
>> > a=T38FaxVersion:0
>> > a=T38MaxBitRate:14400
>> > a=T38FaxUdpEC:t38UDPRedundancy
>> > a=T38FaxRateManagement:
>> > transferredTCF
>> >
>> > And reply 200 OK contains in SDP:
>> > *m=audio 34788 RTP/AVP 8*
>> > a=rtpmap:8 PCMA/8000
>> > a=silenceSupp:off - - - -
>> > a=ptime:20
>> >
>> > Reply 200 OK SDP between YATE (GW - SIP Interop) and OpenSIPS contains
>> > in SDP:
>> > *m=audio 16330 RTP/AVP 8*
>> > a=rtpmap:8 PCMA/8000
>> > a=silenceSupp:off - - - -
>> > a=ptime:20
>> > *m=image 0 udptl 19*
>> >
>> > In this case everything works fine. Line *m=image 0 udptl 19 *is
>> > removed by YATE.
>> > But same test with FreeSWITCH on the place of YATE (GW - SIP Interop)
>> > *"m=image 0 udptl 19" *call brakes as you can see in my first mail.
>> >
>> > I don't want to compare or discus YATE and FS functionality or
>> > something else. I just see difference in behavior and because I'm not
>> > a big expert don't know witch implementation is more accurate
>> > according standards. And second: Is it impossible for me to upgrade
>> > all CPE so only thing I can do is to fix it on server side. That is
>> > because I ask for a help.
>> You said the the broken line "m=image 0 udptl 19" goes from OpenSIPS to
>> YATE. Do you know if it originates from the OpenSIPS box or the
>> subscriber? If it originates from the OpenSIPS box it should be reported
>> to them. If its from the subscriber, well...... your chances of getting
>> anything fixed are usually small.
>>
>> Steve
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100222/5cd510a7/attachment-0002.html 

From lfurrea at gmail.com  Mon Feb 22 13:05:11 2010
From: lfurrea at gmail.com (Luis F Urrea)
Date: Mon, 22 Feb 2010 15:05:11 -0600
Subject: [Freeswitch-users] ESF_PAGE_GROUP in a box with 2 interfaces
Message-ID: <f120f4c81002221305r7c9b4dedw13ccf5f72eb75d05@mail.gmail.com>

 Hi all,
I have a FS process running on a fw with 2 ethernet
interfaces, the FS process is bound to the internal iface but
when I use esf_page_group it tries to forward multicast
packets through the external iface, is there a config
parameter maybe to be able to control this?

TIA
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100222/ead3f499/attachment-0002.html 

From anthony.minessale at gmail.com  Mon Feb 22 13:37:43 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 22 Feb 2010 15:37:43 -0600
Subject: [Freeswitch-users] ESF_PAGE_GROUP in a box with 2 interfaces
In-Reply-To: <f120f4c81002221305r7c9b4dedw13ccf5f72eb75d05@mail.gmail.com>
References: <f120f4c81002221305r7c9b4dedw13ccf5f72eb75d05@mail.gmail.com>
Message-ID: <191c3a031002221337t3c49cda7r14d80c71d510a8e9@mail.gmail.com>

yes you can control the IP and by virtue of your routing table which
interface.


On Mon, Feb 22, 2010 at 3:05 PM, Luis F Urrea <lfurrea at gmail.com> wrote:

>  Hi all,
> I have a FS process running on a fw with 2 ethernet
> interfaces, the FS process is bound to the internal iface but
> when I use esf_page_group it tries to forward multicast
> packets through the external iface, is there a config
> parameter maybe to be able to control this?
>
> TIA
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100222/e75a6eeb/attachment-0002.html 

From christian.loeschenkohl at xpirio.com  Mon Feb 22 13:40:18 2010
From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=)
Date: Mon, 22 Feb 2010 22:40:18 +0100
Subject: [Freeswitch-users] sending a sip notify with sendevent
In-Reply-To: <126320.178.qm@web33502.mail.mud.yahoo.com>
References: <126320.178.qm@web33502.mail.mud.yahoo.com>
Message-ID: <4B82F9C2.2040002@xpirio.com>

thank you for this advise

i read the section "case SWITCH_EVENT_NOTIFY" carefully, debugged my script (i messed something
up, with telnet the command works - returns Reply-Text: +OK)

sql is executed and returns 1 row
select sip_user,sip_host,contact,profile_name,'application/simple-message-summary','check-sync;reboot=true',''
from sip_registrations where sip_user='10' and sip_host='vts.vie.xpirio.net'

however no notify message is send to the device

i can't use call-id because i simply don't know it

br


Diego Toro wrote:

> Read "case SWITCH_EVENT_NOTIFY:" in mod_sofia.c. I suggest you assign call-id in the header of the event.
> 
> 
> Diego Toro
> http://lacarretade.blogspot.com/
> 
> 
> --- On Mon, 2/22/10, Christian L?schenkohl <christian.loeschenkohl at xpirio.com> wrote:
> 
>> From: Christian L?schenkohl <christian.loeschenkohl at xpirio.com>
>> Subject: [Freeswitch-users] sending a sip notify with sendevent
>> To: freeswitch-users at lists.freeswitch.org
>> Date: Monday, February 22, 2010, 12:16 PM
>> hi
>>
>> i try to send a sip notify message to a registered sip
>> device
>> "sofia profile nat flush_inbound_reg 10 at vts.vie.xpirio.net
>> reboot" works, but i need
>> to send "check-sync;reboot=false" - so the device does a
>> resync and don't do a reboot
>>
>> my message looks like this
>>
>> sendevent NOTIFY
>> profile: nat
>> event-string: check-sync;reboot=false
>> user: 10
>> host: vts.vie.xpirio.net
>> content-type: application/simple-message-summary
>>
>> if i listen on the loopback interface i do see
>>
>> ##
>> T 2010/02/22 18:11:59.083204 127.0.0.1:51840 ->
>> 127.0.0.1:8021 [AP]
>> sendevent NOTIFY
>> profile: nat
>> event-string: check-sync;reboot=false
>> user: 10
>> host: vts.vie.xpirio.net
>> content-type: application/simple-message-summary
>>
>> ##
>> T 2010/02/22 18:11:59.084032 127.0.0.1:8021 ->
>> 127.0.0.1:51840 [AP]
>> Content-Type: command/reply
>> Reply-Text: -ERR invalid
>>
>> --------
>> i don't get what it is wrong. i also rechecked the
>> registered user in the sqlite database and this
>> looks good to me.
>>
>> no message is send to the user.
>>
>> we do use multiple domains, so user could also be 10 at somedomain.com
>> - or am i wrong on this?
>> could somebody please bring some light in this.
>>
>> we do use trunk rev. 16631
>>
>> br
>>
>>
>>
>> -- 
>> Ing. Christian L?schenkohl
>> Technische Leitung, Forschung & Entwicklung VoIP
>>
>> xpirio
>> Telekommunikation & Service GmbH
>> Lakeside B04
>> 9020 Klagenfurt
>> Austria
>>
>> T  +43 (0) 5 77 11 - 1000
>> F  +43 (0) 5 77 11 - 1002
>> E  christian.loeschenkohl at xpirio.com
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
> 
> 
>       
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


-- 
Ing. Christian L?schenkohl
Technische Leitung, Forschung & Entwicklung VoIP

xpirio
Telekommunikation & Service GmbH
Lakeside B04
9020 Klagenfurt
Austria

T  +43 (0) 5 77 11 - 1000
F  +43 (0) 5 77 11 - 1002
E  christian.loeschenkohl at xpirio.com



From anthony.minessale at gmail.com  Mon Feb 22 13:48:45 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 22 Feb 2010 15:48:45 -0600
Subject: [Freeswitch-users] sending a sip notify with sendevent
In-Reply-To: <4B82F9C2.2040002@xpirio.com>
References: <126320.178.qm@web33502.mail.mud.yahoo.com>
	<4B82F9C2.2040002@xpirio.com>
Message-ID: <191c3a031002221348m7b474499rea4d8ab6e806d05a@mail.gmail.com>

compare that sql stmt to your db manually with the sqlite3 app

sqlite3 /usr/local/freeswitch/db/sofia_reg_internal.db


2010/2/22 Christian L?schenkohl <christian.loeschenkohl at xpirio.com>

> thank you for this advise
>
> i read the section "case SWITCH_EVENT_NOTIFY" carefully, debugged my script
> (i messed something
> up, with telnet the command works - returns Reply-Text: +OK)
>
> sql is executed and returns 1 row
> select
> sip_user,sip_host,contact,profile_name,'application/simple-message-summary','check-sync;reboot=true',''
> from sip_registrations where sip_user='10' and sip_host='
> vts.vie.xpirio.net'
>
> however no notify message is send to the device
>
> i can't use call-id because i simply don't know it
>
> br
>
>
> Diego Toro wrote:
>
> > Read "case SWITCH_EVENT_NOTIFY:" in mod_sofia.c. I suggest you assign
> call-id in the header of the event.
> >
> >
> > Diego Toro
> > http://lacarretade.blogspot.com/
> >
> >
> > --- On Mon, 2/22/10, Christian L?schenkohl <
> christian.loeschenkohl at xpirio.com> wrote:
> >
> >> From: Christian L?schenkohl <christian.loeschenkohl at xpirio.com>
> >> Subject: [Freeswitch-users] sending a sip notify with sendevent
> >> To: freeswitch-users at lists.freeswitch.org
> >> Date: Monday, February 22, 2010, 12:16 PM
> >> hi
> >>
> >> i try to send a sip notify message to a registered sip
> >> device
> >> "sofia profile nat flush_inbound_reg 10 at vts.vie.xpirio.net
> >> reboot" works, but i need
> >> to send "check-sync;reboot=false" - so the device does a
> >> resync and don't do a reboot
> >>
> >> my message looks like this
> >>
> >> sendevent NOTIFY
> >> profile: nat
> >> event-string: check-sync;reboot=false
> >> user: 10
> >> host: vts.vie.xpirio.net
> >> content-type: application/simple-message-summary
> >>
> >> if i listen on the loopback interface i do see
> >>
> >> ##
> >> T 2010/02/22 18:11:59.083204 127.0.0.1:51840 ->
> >> 127.0.0.1:8021 [AP]
> >> sendevent NOTIFY
> >> profile: nat
> >> event-string: check-sync;reboot=false
> >> user: 10
> >> host: vts.vie.xpirio.net
> >> content-type: application/simple-message-summary
> >>
> >> ##
> >> T 2010/02/22 18:11:59.084032 127.0.0.1:8021 ->
> >> 127.0.0.1:51840 [AP]
> >> Content-Type: command/reply
> >> Reply-Text: -ERR invalid
> >>
> >> --------
> >> i don't get what it is wrong. i also rechecked the
> >> registered user in the sqlite database and this
> >> looks good to me.
> >>
> >> no message is send to the user.
> >>
> >> we do use multiple domains, so user could also be 10 at somedomain.com
> >> - or am i wrong on this?
> >> could somebody please bring some light in this.
> >>
> >> we do use trunk rev. 16631
> >>
> >> br
> >>
> >>
> >>
> >> --
> >> Ing. Christian L?schenkohl
> >> Technische Leitung, Forschung & Entwicklung VoIP
> >>
> >> xpirio
> >> Telekommunikation & Service GmbH
> >> Lakeside B04
> >> 9020 Klagenfurt
> >> Austria
> >>
> >> T  +43 (0) 5 77 11 - 1000
> >> F  +43 (0) 5 77 11 - 1002
> >> E  christian.loeschenkohl at xpirio.com
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> --
> Ing. Christian L?schenkohl
> Technische Leitung, Forschung & Entwicklung VoIP
>
> xpirio
> Telekommunikation & Service GmbH
> Lakeside B04
> 9020 Klagenfurt
> Austria
>
> T  +43 (0) 5 77 11 - 1000
> F  +43 (0) 5 77 11 - 1002
> E  christian.loeschenkohl at xpirio.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100222/a4526a0e/attachment-0002.html 

From m.sobkow at marketelsystems.com  Mon Feb 22 14:06:12 2010
From: m.sobkow at marketelsystems.com (Mark Sobkow)
Date: Mon, 22 Feb 2010 16:06:12 -0600
Subject: [Freeswitch-users] Is there any way to loop a dialplan?
Message-ID: <4B82FFD4.40309@marketelsystems.com>

Apparently you can't have that first call waiting for the FIFO to be 
picked up.  It'll bridge a FIFO, but the remainder of the dial plan 
never executes. *sigh*

What is INTENDED to happen is that the operator dials in to extension 
6000 and enters their PIN, then gets put in the FIFO.  The idea is that 
after they've handled a call from the FIFO, the dialplan for extension 
6000 resumes, collects the call result, and then enters the looping 
dialplan for extension 6001, which just keeps putting them back in the 
queue, collecting a result code, and repeating ad-nauseum.

I've also tried calling a loopback extension so I could properly bridge 
the call, thinking that would respect the set hangup_after_bridge=false, 
but that doesn't work either.

I may be frustrated, but I'm having fun...

<extension name=\"TestBridging\">
\t<condition field=\"destination_number\" expression=\"^6000$\">
\t\t<action application=\"answer\" />
\t\t<action application=\"play_and_get_digits\" data=\"4 4 1 5000 # 
/opt/freeswitch/sounds/en/us/callie/conference/8000/conf-pin.wav 
/opt/freeswitch/sounds/en/us/callie/conference/8000/conf-bad-pin.wav 
operator_pin \\d+\" />
\t\t<action application=\"log\" data=\"INFO operator_pin is 
${operator_pin}\" />
\t\t<action application=\"log\" data=\"INFO Queing operator to fifo\" />
\t\t<action application=\"set\" data=\"hangup_after_bridge=false\" />
\t\t<action application=\"bridge\" data=\"loopback/6003\" />
\t\t<action application=\"log\" data=\"INFO Handled first call\" />
\t\t<action application=\"play_and_get_digits\" data=\"1 2 1 5000 # 
/opt/freeswitch/sounds/en/us/callie/conference/8000/conf-pin.wav 
/opt/freeswitch/sounds/en/us/callie/conference/8000/conf-bad-pin.wav 
call_result \\d+\" />
\t\t<action application=\"log\" data=\"INFO call_result is 
${call_result}\" />
\t\t<action application=\"set\" data=\"hangup_after_bridge=true\" />
\t\t<action application=\"bridge\" data=\"loopback/6001\" />
\t</condition>
</extension>
<extension name=\"TestLoop\">
\t<condition field=\"destination_number\" expression=\"^6001$\">
\t\t<action application=\"answer\" />
\t\t<action application=\"log\" data=\"INFO Requeing operator to fifo\" />
\t\t<action application=\"set\" data=\"hangup_after_bridge=false\" />
\t\t<action application=\"bridge\" data=\"loopback/6003\" />
\t\t<action application=\"log\" data=\"INFO Handled another call\" />
\t\t<action application=\"play_and_get_digits\" data=\"1 2 1 5000 # 
/opt/freeswitch/sounds/en/us/callie/conference/8000/conf-pin.wav 
/opt/freeswitch/sounds/en/us/callie/conference/8000/conf-bad-pin.wav 
call_result \\d+\" />
\t\t<action application=\"log\" data=\"INFO call_result is 
${call_result}\" />
\t\t<action application=\"set\" data=\"hangup_after_bridge=true\" />
\t\t<action application=\"bridge\" data=\"loopback/6001\" />
\t</condition>
</extension>
<extension name=\"TestCallIn\">
\t<condition field=\"destination_number\" expression=\"^6002$\">
\t\t<action application=\"answer\" />
\t\t<action application=\"log\" data=\"INFO Pulling operator\" />
\t\t<action application=\"set\" data=\"hangup_after_bridge=true\" />
\t\t<action application=\"fifo\" data=\"6003@${domain_name} out nowait\" />
\t\t<action application=\"log\" data=\"INFO Done call\" />
\t</condition>
</extension>
<extension name=\"park\">
\t<condition field=\"destination_number\" expression=\"^6003$\">
\t\t<action application=\"fifo\" data=\"6003@${domain_name} in\" />
\t</condition>
</extension>
<extension name=\"unpark\">
\t<condition field=\"destination_number\" expression=\"^6004$\">
\t\t<action application=\"answer\" />
\t\t<action application=\"fifo\" data=\"6003@${domain_name} out nowait\" />
\t</condition>
</extension>

-- 
Mark Sobkow
Senior Developer
MarkeTel Multi-Line Dialing Systems LTD.
428 Victoria Ave
Regina, SK S4N-0P6
Toll-Free: 800-289-8616-X533
Local: 306-359-6893-X533
Fax: 306-359-6879
Email: m.sobkow at marketelsystems.com
Web: http://www.marketelsystems.com 




From lfurrea at gmail.com  Mon Feb 22 14:27:19 2010
From: lfurrea at gmail.com (Luis F Urrea)
Date: Mon, 22 Feb 2010 16:27:19 -0600
Subject: [Freeswitch-users] ESF_PAGE_GROUP in a box with 2 interfaces
In-Reply-To: <191c3a031002221337t3c49cda7r14d80c71d510a8e9@mail.gmail.com>
References: <f120f4c81002221305r7c9b4dedw13ccf5f72eb75d05@mail.gmail.com>
	<191c3a031002221337t3c49cda7r14d80c71d510a8e9@mail.gmail.com>
Message-ID: <f120f4c81002221427x78956438oc8b54cd1b85cd76d@mail.gmail.com>

Well the rest of the story is that I'm running a FreeBSD box and when the
box has only one interface then it's not necessary to setup multicast
routing which by the way is not built in a FreeBSD generic kernel.

FreebsD docs state:

"FreeBSD supports multicast host operations by default"

Since this would be in the category of a host operation and there would be
no need to forward multicast traffic between interfaces I thought that maybe
there could be code in the esf application that would choose the IP it would
bound to.

Just wondering if that is the case but if you Anthony can confirm that it is
totally left out to the OS routing rules I'll take my inquiry somewhere else
to clarify further.

Thx


On Mon, Feb 22, 2010 at 3:37 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> yes you can control the IP and by virtue of your routing table which
> interface.
>
>
> On Mon, Feb 22, 2010 at 3:05 PM, Luis F Urrea <lfurrea at gmail.com> wrote:
>
>>  Hi all,
>> I have a FS process running on a fw with 2 ethernet
>> interfaces, the FS process is bound to the internal iface but
>> when I use esf_page_group it tries to forward multicast
>> packets through the external iface, is there a config
>> parameter maybe to be able to control this?
>>
>> TIA
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
<img width="300" height="160" border="0" _moz_dirty="" src="
http://www.simplecs.net/firmas/calufa.jpg" alt="firma"
_moz_resizing="true"/>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100222/84e8160a/attachment-0002.html 

From christian.loeschenkohl at xpirio.com  Mon Feb 22 14:28:39 2010
From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=)
Date: Mon, 22 Feb 2010 23:28:39 +0100
Subject: [Freeswitch-users] sending a sip notify with sendevent
In-Reply-To: <191c3a031002221348m7b474499rea4d8ab6e806d05a@mail.gmail.com>
References: <126320.178.qm@web33502.mail.mud.yahoo.com>	<4B82F9C2.2040002@xpirio.com>
	<191c3a031002221348m7b474499rea4d8ab6e806d05a@mail.gmail.com>
Message-ID: <4B830517.2070402@xpirio.com>

hi anthony

i did it

my profile is actually named nat, "sofia status profile nat" shows me presence_nat as the db name
so i had a look in the file presence_nat.db

i execute
select sip_user,sip_host,contact,profile_name,'application/simple-message-summary','check-sync;reboot=false','' from sip_registrations where sip_user='10' and sip_host='vts.vie.xpirio.net';

and it returns
10|vts.vie.xpirio.net|"10" <sip:10 at 81.94.53.5:1032>|nat|application/simple-message-summary|check-sync;reboot=false|

looks good to me so far, but as i said no sip notify message is send to the client

br

Anthony Minessale wrote:

> compare that sql stmt to your db manually with the sqlite3 app
> 
> sqlite3 /usr/local/freeswitch/db/sofia_reg_internal.db
> 
> 
> 2010/2/22 Christian L?schenkohl <christian.loeschenkohl at xpirio.com 
> <mailto:christian.loeschenkohl at xpirio.com>>
> 
>     thank you for this advise
> 
>     i read the section "case SWITCH_EVENT_NOTIFY" carefully, debugged my
>     script (i messed something
>     up, with telnet the command works - returns Reply-Text: +OK)
> 
>     sql is executed and returns 1 row
>     select
>     sip_user,sip_host,contact,profile_name,'application/simple-message-summary','check-sync;reboot=true',''
>     from sip_registrations where sip_user='10' and
>     sip_host='vts.vie.xpirio.net <http://vts.vie.xpirio.net>'
> 
>     however no notify message is send to the device
> 
>     i can't use call-id because i simply don't know it
> 
>     br
> 
> 
>     Diego Toro wrote:
> 
>      > Read "case SWITCH_EVENT_NOTIFY:" in mod_sofia.c. I suggest you
>     assign call-id in the header of the event.
>      >
>      >
>      > Diego Toro
>      > http://lacarretade.blogspot.com/
>      >
>      >
>      > --- On Mon, 2/22/10, Christian L?schenkohl
>     <christian.loeschenkohl at xpirio.com
>     <mailto:christian.loeschenkohl at xpirio.com>> wrote:
>      >
>      >> From: Christian L?schenkohl <christian.loeschenkohl at xpirio.com
>     <mailto:christian.loeschenkohl at xpirio.com>>
>      >> Subject: [Freeswitch-users] sending a sip notify with sendevent
>      >> To: freeswitch-users at lists.freeswitch.org
>     <mailto:freeswitch-users at lists.freeswitch.org>
>      >> Date: Monday, February 22, 2010, 12:16 PM
>      >> hi
>      >>
>      >> i try to send a sip notify message to a registered sip
>      >> device
>      >> "sofia profile nat flush_inbound_reg 10 at vts.vie.xpirio.net
>     <mailto:10 at vts.vie.xpirio.net>
>      >> reboot" works, but i need
>      >> to send "check-sync;reboot=false" - so the device does a
>      >> resync and don't do a reboot
>      >>
>      >> my message looks like this
>      >>
>      >> sendevent NOTIFY
>      >> profile: nat
>      >> event-string: check-sync;reboot=false
>      >> user: 10
>      >> host: vts.vie.xpirio.net <http://vts.vie.xpirio.net>
>      >> content-type: application/simple-message-summary
>      >>
>      >> if i listen on the loopback interface i do see
>      >>
>      >> ##
>      >> T 2010/02/22 18:11:59.083204 127.0.0.1:51840
>     <http://127.0.0.1:51840> ->
>      >> 127.0.0.1:8021 <http://127.0.0.1:8021> [AP]
>      >> sendevent NOTIFY
>      >> profile: nat
>      >> event-string: check-sync;reboot=false
>      >> user: 10
>      >> host: vts.vie.xpirio.net <http://vts.vie.xpirio.net>
>      >> content-type: application/simple-message-summary
>      >>
>      >> ##
>      >> T 2010/02/22 18:11:59.084032 127.0.0.1:8021
>     <http://127.0.0.1:8021> ->
>      >> 127.0.0.1:51840 <http://127.0.0.1:51840> [AP]
>      >> Content-Type: command/reply
>      >> Reply-Text: -ERR invalid
>      >>
>      >> --------
>      >> i don't get what it is wrong. i also rechecked the
>      >> registered user in the sqlite database and this
>      >> looks good to me.
>      >>
>      >> no message is send to the user.
>      >>
>      >> we do use multiple domains, so user could also be
>     10 at somedomain.com <mailto:10 at somedomain.com>
>      >> - or am i wrong on this?
>      >> could somebody please bring some light in this.
>      >>
>      >> we do use trunk rev. 16631
>      >>
>      >> br
>      >>
>      >>
>      >>
>      >> --
>      >> Ing. Christian L?schenkohl
>      >> Technische Leitung, Forschung & Entwicklung VoIP
>      >>
>      >> xpirio
>      >> Telekommunikation & Service GmbH
>      >> Lakeside B04
>      >> 9020 Klagenfurt
>      >> Austria
>      >>
>      >> T  +43 (0) 5 77 11 - 1000
>      >> F  +43 (0) 5 77 11 - 1002
>      >> E  christian.loeschenkohl at xpirio.com
>     <mailto:christian.loeschenkohl at xpirio.com>
>      >>
>      >> _______________________________________________
>      >> FreeSWITCH-users mailing list
>      >> FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>      >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>      >>
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>      >> http://www.freeswitch.org
>      >>
>      >
>      >
>      >
>      >
>      > _______________________________________________
>      > FreeSWITCH-users mailing list
>      > FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>      > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>      >
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>      > http://www.freeswitch.org
> 
> 
>     --
>     Ing. Christian L?schenkohl
>     Technische Leitung, Forschung & Entwicklung VoIP
> 
>     xpirio
>     Telekommunikation & Service GmbH
>     Lakeside B04
>     9020 Klagenfurt
>     Austria
> 
>     T  +43 (0) 5 77 11 - 1000
>     F  +43 (0) 5 77 11 - 1002
>     E  christian.loeschenkohl at xpirio.com
>     <mailto:christian.loeschenkohl at xpirio.com>
> 
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
> 
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com 
> <mailto:MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com 
> <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org 
> <mailto:sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888 
> <http://iax:guest at conference.freeswitch.org/888>
> googletalk:conf+888 at conference.freeswitch.org 
> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


-- 
Ing. Christian L?schenkohl
Technische Leitung, Forschung & Entwicklung VoIP

xpirio
Telekommunikation & Service GmbH
Lakeside B04
9020 Klagenfurt
Austria

T  +43 (0) 5 77 11 - 1000
F  +43 (0) 5 77 11 - 1002
E  christian.loeschenkohl at xpirio.com



From brian at freeswitch.org  Mon Feb 22 14:35:36 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 22 Feb 2010 16:35:36 -0600
Subject: [Freeswitch-users] ESF_PAGE_GROUP in a box with 2 interfaces
In-Reply-To: <f120f4c81002221427x78956438oc8b54cd1b85cd76d@mail.gmail.com>
References: <f120f4c81002221305r7c9b4dedw13ccf5f72eb75d05@mail.gmail.com>
	<191c3a031002221337t3c49cda7r14d80c71d510a8e9@mail.gmail.com>
	<f120f4c81002221427x78956438oc8b54cd1b85cd76d@mail.gmail.com>
Message-ID: <328BF8BA-3BBB-4C57-8C79-73BC681E8C87@freeswitch.org>

The args to the app are the ip and port to send on... 

/b

On Feb 22, 2010, at 4:27 PM, Luis F Urrea wrote:

> Since this would be in the category of a host operation and there would be no need to forward multicast traffic between interfaces I thought that maybe there could be code in the esf application that would choose the IP it would bound to.




From dftoro at yahoo.com  Mon Feb 22 14:47:25 2010
From: dftoro at yahoo.com (Diego Toro)
Date: Mon, 22 Feb 2010 14:47:25 -0800 (PST)
Subject: [Freeswitch-users] sending a sip notify with sendevent
In-Reply-To: <4B830517.2070402@xpirio.com>
Message-ID: <959367.79243.qm@web33508.mail.mud.yahoo.com>

Hi, Read "case SWITCH_EVENT_NOTIFY:" in mod_sofia.c. I suggest you assign call-id in the header of the event.


Diego Toro
http://lacarretade.blogspot.com/



--- On Mon, 2/22/10, Christian L?schenkohl <christian.loeschenkohl at xpirio.com> wrote:

> From: Christian L?schenkohl <christian.loeschenkohl at xpirio.com>
> Subject: Re: [Freeswitch-users] sending a sip notify with sendevent
> To: freeswitch-users at lists.freeswitch.org
> Date: Monday, February 22, 2010, 5:28 PM
> hi anthony
> 
> i did it
> 
> my profile is actually named nat, "sofia status profile
> nat" shows me presence_nat as the db name
> so i had a look in the file presence_nat.db
> 
> i execute
> select
> sip_user,sip_host,contact,profile_name,'application/simple-message-summary','check-sync;reboot=false',''
> from sip_registrations where sip_user='10' and
> sip_host='vts.vie.xpirio.net';
> 
> and it returns
> 10|vts.vie.xpirio.net|"10"
> <sip:10 at 81.94.53.5:1032>|nat|application/simple-message-summary|check-sync;reboot=false|
> 
> looks good to me so far, but as i said no sip notify
> message is send to the client
> 
> br
> 
> Anthony Minessale wrote:
> 
> > compare that sql stmt to your db manually with the
> sqlite3 app
> > 
> > sqlite3
> /usr/local/freeswitch/db/sofia_reg_internal.db
> > 
> > 
> > 2010/2/22 Christian L?schenkohl <christian.loeschenkohl at xpirio.com
> 
> > <mailto:christian.loeschenkohl at xpirio.com>>
> > 
> >? ???thank you for this advise
> > 
> >? ???i read the section "case
> SWITCH_EVENT_NOTIFY" carefully, debugged my
> >? ???script (i messed something
> >? ???up, with telnet the command
> works - returns Reply-Text: +OK)
> > 
> >? ???sql is executed and returns 1
> row
> >? ???select
> >?
> ???sip_user,sip_host,contact,profile_name,'application/simple-message-summary','check-sync;reboot=true',''
> >? ???from sip_registrations where
> sip_user='10' and
> >? ???sip_host='vts.vie.xpirio.net
> <http://vts.vie.xpirio.net>'
> > 
> >? ???however no notify message is
> send to the device
> > 
> >? ???i can't use call-id because i
> simply don't know it
> > 
> >? ???br
> > 
> > 
> >? ???Diego Toro wrote:
> > 
> >? ? ? > Read "case
> SWITCH_EVENT_NOTIFY:" in mod_sofia.c. I suggest you
> >? ???assign call-id in the header
> of the event.
> >? ? ? >
> >? ? ? >
> >? ? ? > Diego Toro
> >? ? ? > http://lacarretade.blogspot.com/
> >? ? ? >
> >? ? ? >
> >? ? ? > --- On Mon, 2/22/10,
> Christian L?schenkohl
> >? ???<christian.loeschenkohl at xpirio.com
> >? ???<mailto:christian.loeschenkohl at xpirio.com>>
> wrote:
> >? ? ? >
> >? ? ? >> From: Christian
> L?schenkohl <christian.loeschenkohl at xpirio.com
> >? ???<mailto:christian.loeschenkohl at xpirio.com>>
> >? ? ? >> Subject:
> [Freeswitch-users] sending a sip notify with sendevent
> >? ? ? >> To: freeswitch-users at lists.freeswitch.org
> >? ???<mailto:freeswitch-users at lists.freeswitch.org>
> >? ? ? >> Date: Monday, February
> 22, 2010, 12:16 PM
> >? ? ? >> hi
> >? ? ? >>
> >? ? ? >> i try to send a sip
> notify message to a registered sip
> >? ? ? >> device
> >? ? ? >> "sofia profile nat
> flush_inbound_reg 10 at vts.vie.xpirio.net
> >? ???<mailto:10 at vts.vie.xpirio.net>
> >? ? ? >> reboot" works, but i
> need
> >? ? ? >> to send
> "check-sync;reboot=false" - so the device does a
> >? ? ? >> resync and don't do a
> reboot
> >? ? ? >>
> >? ? ? >> my message looks like
> this
> >? ? ? >>
> >? ? ? >> sendevent NOTIFY
> >? ? ? >> profile: nat
> >? ? ? >> event-string:
> check-sync;reboot=false
> >? ? ? >> user: 10
> >? ? ? >> host: vts.vie.xpirio.net
> <http://vts.vie.xpirio.net>
> >? ? ? >> content-type:
> application/simple-message-summary
> >? ? ? >>
> >? ? ? >> if i listen on the
> loopback interface i do see
> >? ? ? >>
> >? ? ? >> ##
> >? ? ? >> T 2010/02/22
> 18:11:59.083204 127.0.0.1:51840
> >? ???<http://127.0.0.1:51840> ->
> >? ? ? >> 127.0.0.1:8021 <http://127.0.0.1:8021> [AP]
> >? ? ? >> sendevent NOTIFY
> >? ? ? >> profile: nat
> >? ? ? >> event-string:
> check-sync;reboot=false
> >? ? ? >> user: 10
> >? ? ? >> host: vts.vie.xpirio.net
> <http://vts.vie.xpirio.net>
> >? ? ? >> content-type:
> application/simple-message-summary
> >? ? ? >>
> >? ? ? >> ##
> >? ? ? >> T 2010/02/22
> 18:11:59.084032 127.0.0.1:8021
> >? ???<http://127.0.0.1:8021> ->
> >? ? ? >> 127.0.0.1:51840 <http://127.0.0.1:51840> [AP]
> >? ? ? >> Content-Type:
> command/reply
> >? ? ? >> Reply-Text: -ERR invalid
> >? ? ? >>
> >? ? ? >> --------
> >? ? ? >> i don't get what it is
> wrong. i also rechecked the
> >? ? ? >> registered user in the
> sqlite database and this
> >? ? ? >> looks good to me.
> >? ? ? >>
> >? ? ? >> no message is send to the
> user.
> >? ? ? >>
> >? ? ? >> we do use multiple
> domains, so user could also be
> >? ???10 at somedomain.com
> <mailto:10 at somedomain.com>
> >? ? ? >> - or am i wrong on this?
> >? ? ? >> could somebody please
> bring some light in this.
> >? ? ? >>
> >? ? ? >> we do use trunk rev.
> 16631
> >? ? ? >>
> >? ? ? >> br
> >? ? ? >>
> >? ? ? >>
> >? ? ? >>
> >? ? ? >> --
> >? ? ? >> Ing. Christian
> L?schenkohl
> >? ? ? >> Technische Leitung,
> Forschung & Entwicklung VoIP
> >? ? ? >>
> >? ? ? >> xpirio
> >? ? ? >> Telekommunikation &
> Service GmbH
> >? ? ? >> Lakeside B04
> >? ? ? >> 9020 Klagenfurt
> >? ? ? >> Austria
> >? ? ? >>
> >? ? ? >> T? +43 (0) 5 77 11 -
> 1000
> >? ? ? >> F? +43 (0) 5 77 11 -
> 1002
> >? ? ? >> E? christian.loeschenkohl at xpirio.com
> >? ???<mailto:christian.loeschenkohl at xpirio.com>
> >? ? ? >>
> >? ? ? >>
> _______________________________________________
> >? ? ? >> FreeSWITCH-users mailing
> list
> >? ? ? >> FreeSWITCH-users at lists.freeswitch.org
> >? ???<mailto:FreeSWITCH-users at lists.freeswitch.org>
> >? ? ? >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >? ? ? >>
> >? ???UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >? ? ? >> http://www.freeswitch.org
> >? ? ? >>
> >? ? ? >
> >? ? ? >
> >? ? ? >
> >? ? ? >
> >? ? ? >
> _______________________________________________
> >? ? ? > FreeSWITCH-users mailing
> list
> >? ? ? > FreeSWITCH-users at lists.freeswitch.org
> >? ???<mailto:FreeSWITCH-users at lists.freeswitch.org>
> >? ? ? > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >? ? ? >
> >? ???UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >? ? ? > http://www.freeswitch.org
> > 
> > 
> >? ???--
> >? ???Ing. Christian L?schenkohl
> >? ???Technische Leitung, Forschung
> & Entwicklung VoIP
> > 
> >? ???xpirio
> >? ???Telekommunikation &
> Service GmbH
> >? ???Lakeside B04
> >? ???9020 Klagenfurt
> >? ???Austria
> > 
> >? ???T? +43 (0) 5 77 11 -
> 1000
> >? ???F? +43 (0) 5 77 11 -
> 1002
> >? ???E? christian.loeschenkohl at xpirio.com
> >? ???<mailto:christian.loeschenkohl at xpirio.com>
> > 
> >?
> ???_______________________________________________
> >? ???FreeSWITCH-users mailing list
> >? ???FreeSWITCH-users at lists.freeswitch.org
> >? ???<mailto:FreeSWITCH-users at lists.freeswitch.org>
> >? ???http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >? ???UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >? ???http://www.freeswitch.org
> > 
> > 
> > 
> > 
> > -- 
> > Anthony Minessale II
> > 
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> > Twitter: http://twitter.com/FreeSWITCH_wire
> > 
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com
> 
> > <mailto:MSN%3Aanthony_minessale at hotmail.com>
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> 
> > <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
> > IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
> > 
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org
> 
> > <mailto:sip%3A888 at conference.freeswitch.org>
> > iax:guest at conference.freeswitch.org/888
> 
> > <http://iax:guest at conference.freeswitch.org/888>
> > googletalk:conf+888 at conference.freeswitch.org
> 
> > <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > pstn:+19193869900
> > 
> > 
> >
> ------------------------------------------------------------------------
> > 
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 
> 
> -- 
> Ing. Christian L?schenkohl
> Technische Leitung, Forschung & Entwicklung VoIP
> 
> xpirio
> Telekommunikation & Service GmbH
> Lakeside B04
> 9020 Klagenfurt
> Austria
> 
> T? +43 (0) 5 77 11 - 1000
> F? +43 (0) 5 77 11 - 1002
> E? christian.loeschenkohl at xpirio.com
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 


      



From lfurrea at gmail.com  Mon Feb 22 16:12:16 2010
From: lfurrea at gmail.com (Luis F Urrea)
Date: Mon, 22 Feb 2010 18:12:16 -0600
Subject: [Freeswitch-users] ESF_PAGE_GROUP in a box with 2 interfaces
In-Reply-To: <328BF8BA-3BBB-4C57-8C79-73BC681E8C87@freeswitch.org>
References: <f120f4c81002221305r7c9b4dedw13ccf5f72eb75d05@mail.gmail.com>
	<191c3a031002221337t3c49cda7r14d80c71d510a8e9@mail.gmail.com>
	<f120f4c81002221427x78956438oc8b54cd1b85cd76d@mail.gmail.com>
	<328BF8BA-3BBB-4C57-8C79-73BC681E8C87@freeswitch.org>
Message-ID: <f120f4c81002221612q6313de01h2bd4c6a53022e41f@mail.gmail.com>

Thx Brian,

I understand that you can set the *destination* IP:Port via variables, but I
was concerned with the source interface of the multicast traffic.

And I just confirmed that on FreeBSD you do not need to specify a route in
the single interface case because:

"the default multicast route is via the interface
with the default route; setting a route isn't necessary unless you need to
force multicast to go via a particular interface by default, this is done
by longest-prefix matching like all other IPv4 routing activities."

They also state that:

"An unprivileged userland application is also able to control where it is
sending its multicast traffic (without mucking with the routing table)
by using the sockopt IP_MULTICAST_IF.  It can specify the address of
any interface on the
machine"

But this would really be a hassle for the programmer :)

Thx for your help!


On Mon, Feb 22, 2010 at 4:35 PM, Brian West <brian at freeswitch.org> wrote:

> The args to the app are the ip and port to send on...
>
> /b
>
> On Feb 22, 2010, at 4:27 PM, Luis F Urrea wrote:
>
> > Since this would be in the category of a host operation and there would
> be no need to forward multicast traffic between interfaces I thought that
> maybe there could be code in the esf application that would choose the IP it
> would bound to.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100222/14f37b15/attachment-0002.html 

From lfurrea at gmail.com  Mon Feb 22 16:22:27 2010
From: lfurrea at gmail.com (Luis F Urrea)
Date: Mon, 22 Feb 2010 18:22:27 -0600
Subject: [Freeswitch-users] ESF_PAGE_GROUP in a box with 2 interfaces
In-Reply-To: <f120f4c81002221612q6313de01h2bd4c6a53022e41f@mail.gmail.com>
References: <f120f4c81002221305r7c9b4dedw13ccf5f72eb75d05@mail.gmail.com>
	<191c3a031002221337t3c49cda7r14d80c71d510a8e9@mail.gmail.com>
	<f120f4c81002221427x78956438oc8b54cd1b85cd76d@mail.gmail.com>
	<328BF8BA-3BBB-4C57-8C79-73BC681E8C87@freeswitch.org>
	<f120f4c81002221612q6313de01h2bd4c6a53022e41f@mail.gmail.com>
Message-ID: <f120f4c81002221622j6f77473asd1279adfbb7c97f8@mail.gmail.com>

wiki updated

On Mon, Feb 22, 2010 at 6:12 PM, Luis F Urrea <lfurrea at gmail.com> wrote:

> Thx Brian,
>
> I understand that you can set the *destination* IP:Port via variables, but
> I was concerned with the source interface of the multicast traffic.
>
> And I just confirmed that on FreeBSD you do not need to specify a route in
> the single interface case because:
>
> "the default multicast route is via the interface
> with the default route; setting a route isn't necessary unless you need to
> force multicast to go via a particular interface by default, this is done
>
> by longest-prefix matching like all other IPv4 routing activities."
>
> They also state that:
>
> "An unprivileged userland application is also able to control where it is
> sending its multicast traffic (without mucking with the routing table)
>
> by using the sockopt IP_MULTICAST_IF.  It can specify the address of any interface on the
> machine"
>
> But this would really be a hassle for the programmer :)
>
> Thx for your help!
>
>
> On Mon, Feb 22, 2010 at 4:35 PM, Brian West <brian at freeswitch.org> wrote:
>
>> The args to the app are the ip and port to send on...
>>
>> /b
>>
>> On Feb 22, 2010, at 4:27 PM, Luis F Urrea wrote:
>>
>> > Since this would be in the category of a host operation and there would
>> be no need to forward multicast traffic between interfaces I thought that
>> maybe there could be code in the esf application that would choose the IP it
>> would bound to.
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100222/0174a341/attachment-0002.html 

From joseph.puchalski at personalcyberspace.com  Mon Feb 22 16:24:41 2010
From: joseph.puchalski at personalcyberspace.com (Joseph Puchalski)
Date: Tue, 23 Feb 2010 00:24:41 +0000
Subject: [Freeswitch-users] How to set outbound caller id info
	for	multiple users/extensions
In-Reply-To: <3FF62C88-2423-43F0-B8A3-C64EF4BC80AC@freeswitch.org>
References: <093DD565390C1E4FB15D7B383E86BB05AF05AB@Goose.personalcyberspace.net>
	<191c3a031002130707v5f1d5ee0lb9e791e9a0aab956@mail.gmail.com>
	<3FF62C88-2423-43F0-B8A3-C64EF4BC80AC@freeswitch.org>
Message-ID: <093DD565390C1E4FB15D7B383E86BB05AF15CD@Goose.personalcyberspace.net>

Thanks for the replies. Since then I've poked around in the wiki and experimented with my config files.

It seems as if I was setting the caller ID for all traffic outbound through my trunk provider:

>From default.xml in /opt/freeswitch/conf/dialplan/

  <extension name="vitel-outbound">
   <condition field="destination_number" expression="^(1{0,1}\d{10})$">
   <action application="set" data="effective_caller_id_number=1234567890"/>
     <action application="bridge" data="sofia/gateway/vitelity-outbound/$1"/>
    </condition>
  </extension>

This worked, insofar as outbound calls carried the caller ID as configured.

I removed the line:

   <action application="set" data="effective_caller_id_number=1234567890"/>

I had hoped that this would allow the value set in 5859.xml to take effect. It didn't. Instead my calls go out without any Caller ID.

I'm obviously missing something.

Is it possible to explicitly set the outbound caller ID for an extension when configuring it? I've tried to do so as follows:

>From 5859.xml

<include>
  <user id="5859">
    <params>
      <param name="password" value="XXX"/>
      <param name="vm-password" value="XXX"/>
    </params>
    <variables>
..
      <variable name="user_context" value="default"/>
      <variable name="effective_caller_id_name" value="2034135859"/>
      <variable name="effective_caller_id_number" value="2034135859"/>
..
    </variables>
  </user>
</include>

Or should I be doing this via <action application="set" data="effective_caller_id_number=${outbound_caller_id_number}"/>
Somewhere else?

I did try to add a <condition> line in my vitelity config area to set "effective_caller_id"  based on originating number. I had no success, possibly because I was checking the  wrong variable, possibly because that's the wrong place to do it.

I've gone back and gotten a much better grounding in XML, but there are still more than enough simultaneously moving parts in FreeSWITCH to make me feel pretty clueless at the moment . I apologize if I'm totally missing something obvious.

Thanks again for any help. The capabilities of  FreeSWITCH continue to amaze me, sufficiently so that I won't be happy until I've got my head wrapped at least part way around it.

Joe P.


From: Brian West [mailto:brian at freeswitch.org]
Sent: Saturday, February 13, 2010 10:17 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] How to set outbound caller id info for multiple users/extensions

I also have to point out their is no such official variable for "outbound_caller_id_name" or "outbound_caller_id_number", Those are just made up variables I used in the default config.

01_example.com.xml

You'll notice I use these lines.

      <action application="set" data="effective_caller_id_number=${outbound_caller_id_number}"/>
      <action application="set" data="effective_caller_id_name=${outbound_caller_id_name}"/>

Its just a way to set the users default outbound caller ID .

/b


On Feb 13, 2010, at 9:07 AM, Anthony Minessale wrote:



It should be covered on the wiki http://wiki.freeswitch.org<http://wiki.freeswitch.org/>

On Feb 12, 2010 6:23 PM, "Joseph Puchalski" <joseph.puchalski at personalcyberspace.com<mailto:joseph.puchalski at personalcyberspace.com>> wrote:
I'm having problems setting different outbound caller id info for different extensions/users.

I've set up a small system with two active users. I set up my users by copying and modifying existing entries from the dialplan files that come with freeSWITCH

Here are my two extensions:

<include>
  <user id="5859">
    <params>
      <param name="password" value="xxxxxx"/>
      <param name="vm-password" value="xxxxxx"/>
    </params>
    <variables>
      <variable name="toll_allow" value="domestic,international,local"/>
      <variable name="accountcode" value="5859"/>
      <variable name="user_context" value="default"/>
      <variable name="effective_caller_id_name" value="5859"/>
      <variable name="effective_caller_id_number" value="5859"/>
      <variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/>
      <variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/>
      <variable name="callgroup" value="techsupport"/>
    </variables>
  </user>
</include>

<include>
  <user id="5515">
    <params>
      <param name="password" value="xxxxxxx"/>
      <param name="vm-password" value="xxxxxxx"/>
    </params>
    <variables>
      <variable name="toll_allow" value="domestic,international,local"/>
      <variable name="accountcode" value="5515"/>
      <variable name="user_context" value="default"/>
      <variable name="effective_caller_id_name" value="5515"/>
      <variable name="effective_caller_id_number" value="5515"/>
      <variable name="outbound_caller_id_name" value="Jane Doe"/>
      <variable name="outbound_caller_id_number" value="5558395515"/>
      <variable name="callgroup" value="techsupport"/>
    </variables>
  </user>
</include>

These extensions are in files named 5859.xml and 5515.xml respectively.

I'm using a SIP trunk from Vitelity (in and out) with two DIDs corresponding to the extensions above.

Inbound and outbound calling work as needed with one exception: Calls originating from user/extension 5515 go out with the caller ID of extension/user 5859.

Extension 5859 was the first that I created.

Where should I be setting the outbound caller id number for my second extension? I've been trying to track this down in the available documentation but have been unable to do so.

I apologize ahead of time if this is answered somewhere obvious that I've missed.

Thanks for any help.

Joe (FreeSWITCH newbie)

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org<http://www.freeswitch.org/>

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100223/b7d83763/attachment-0002.html 

From matt at webcontracts.co.uk  Mon Feb 22 16:55:12 2010
From: matt at webcontracts.co.uk (Matthew Law)
Date: Tue, 23 Feb 2010 00:55:12 -0000
Subject: [Freeswitch-users] How to debug time-based routing?
Message-ID: <eb2506f03e9045c509a6974bcc4a3a49.squirrel@www.webcontracts.co.uk>


After some playing around I now have a working config but it appears to be
routing calls straight to voicemail based on time of day.  I did see the
example of this in the default config, commented it out and reloaded but I
cannot see anything in the log output to verify this.

Looking at the log output, I can see it hit the regex for the outside
number, match that and then get sent to extension 1000 (which is correct)
but at that point it is sent straight to voicemail and I'm stumped.

Can someone please explain how to go about debugging this?  I have the log
level set to 7, if that helps.


Many thanks,

Matt.




From brian at freeswitch.org  Mon Feb 22 17:00:45 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 22 Feb 2010 19:00:45 -0600
Subject: [Freeswitch-users] How to debug time-based routing?
In-Reply-To: <eb2506f03e9045c509a6974bcc4a3a49.squirrel@www.webcontracts.co.uk>
References: <eb2506f03e9045c509a6974bcc4a3a49.squirrel@www.webcontracts.co.uk>
Message-ID: <74B270D5-E134-4221-A0FE-8275B05826A5@freeswitch.org>

Lets start with how about you pastebin your extension and logs... or better join #freeswitch on irc.freenode.net?  ;)

/b

On Feb 22, 2010, at 6:55 PM, Matthew Law wrote:

> Can someone please explain how to go about debugging this?  I have the log
> level set to 7, if that helps.




From andrewkt at aktzero.com  Mon Feb 22 19:10:17 2010
From: andrewkt at aktzero.com (Andrew Thompson)
Date: Mon, 22 Feb 2010 22:10:17 -0500
Subject: [Freeswitch-users] How to set outbound caller id info for
 multiple users/extensions
In-Reply-To: <093DD565390C1E4FB15D7B383E86BB05AF15CD@Goose.personalcyberspace.net>
References: <093DD565390C1E4FB15D7B383E86BB05AF05AB@Goose.personalcyberspace.net>	<191c3a031002130707v5f1d5ee0lb9e791e9a0aab956@mail.gmail.com>	<3FF62C88-2423-43F0-B8A3-C64EF4BC80AC@freeswitch.org>
	<093DD565390C1E4FB15D7B383E86BB05AF15CD@Goose.personalcyberspace.net>
Message-ID: <4B834719.3000505@aktzero.com>

On 2/22/2010 7:24 PM, Joseph Puchalski wrote:
>
> Or should I be doing this via <action application="set" 
> data="effective_caller_id_number=${outbound_caller_id_number}"/>
>
> Somewhere else?
>

I have the following set on my own extension, in 1000.xml:

<variable name="effective_caller_id_name" value="Andrew Thompson"/>
<variable name="effective_caller_id_number" value="1000"/>
<variable name="outbound_caller_id_name" value="MyCompany"/>
<variable name="outbound_caller_id_number" value="9109991234"/>

When I dial extensions internally, the effective_* name/number show up.

When I dial outbound via my SIP provider, I set the following before the 
bridge so that it passes externally valid info:

<action application="set" 
data="effective_caller_id_number=${outbound_caller_id_number}"/>
<action application="set" 
data="effective_caller_id_name=${outbound_caller_id_name}"/>


In my setup, if I don't explicitly overide the effective_* with 
outbound_*, I actually see 1000 as my callerid when I call my cell from 
my extension, so if you're not getting at least that much, something 
else might be wrong. (I have used vitelity, and they do pass callerid 
properly most of the time.)

-- 
Andrew Thompson

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100222/650f5e1b/attachment-0002.html 

From dftoro at yahoo.com  Mon Feb 22 19:18:03 2010
From: dftoro at yahoo.com (Diego Toro)
Date: Mon, 22 Feb 2010 19:18:03 -0800 (PST)
Subject: [Freeswitch-users] sending a sip notify with sendevent
In-Reply-To: <4B830517.2070402@xpirio.com>
Message-ID: <12033.93456.qm@web33507.mail.mud.yahoo.com>

hi, I'm using 

sendevent NOTIFY
profile: internal
event-string: check-sync;reboot=false
user: 1001
content-type: application/simple-message-sumary
profile: internal
event-string: check-sync;reboot=false
host: 192.168.7.3


where:
user 1001 is a registered user
host: IP of FreeSwitch

I see a sip notify message sent to the client.

Diego Toro 
http://lacarretade.blogspot.com/


--- On Mon, 2/22/10, Christian L?schenkohl <christian.loeschenkohl at xpirio.com> wrote:

> From: Christian L?schenkohl <christian.loeschenkohl at xpirio.com>
> Subject: Re: [Freeswitch-users] sending a sip notify with sendevent
> To: freeswitch-users at lists.freeswitch.org
> Date: Monday, February 22, 2010, 5:28 PM
> hi anthony
> 
> i did it
> 
> my profile is actually named nat, "sofia status profile
> nat" shows me presence_nat as the db name
> so i had a look in the file presence_nat.db
> 
> i execute
> select
> sip_user,sip_host,contact,profile_name,'application/simple-message-summary','check-sync;reboot=false',''
> from sip_registrations where sip_user='10' and
> sip_host='vts.vie.xpirio.net';
> 
> and it returns
> 10|vts.vie.xpirio.net|"10"
> <sip:10 at 81.94.53.5:1032>|nat|application/simple-message-summary|check-sync;reboot=false|
> 
> looks good to me so far, but as i said no sip notify
> message is send to the client
> 
> br
> 
> Anthony Minessale wrote:
> 
> > compare that sql stmt to your db manually with the
> sqlite3 app
> > 
> > sqlite3
> /usr/local/freeswitch/db/sofia_reg_internal.db
> > 
> > 
> > 2010/2/22 Christian L?schenkohl <christian.loeschenkohl at xpirio.com
> 
> > <mailto:christian.loeschenkohl at xpirio.com>>
> > 
> >? ???thank you for this advise
> > 
> >? ???i read the section "case
> SWITCH_EVENT_NOTIFY" carefully, debugged my
> >? ???script (i messed something
> >? ???up, with telnet the command
> works - returns Reply-Text: +OK)
> > 
> >? ???sql is executed and returns 1
> row
> >? ???select
> >?
> ???sip_user,sip_host,contact,profile_name,'application/simple-message-summary','check-sync;reboot=true',''
> >? ???from sip_registrations where
> sip_user='10' and
> >? ???sip_host='vts.vie.xpirio.net
> <http://vts.vie.xpirio.net>'
> > 
> >? ???however no notify message is
> send to the device
> > 
> >? ???i can't use call-id because i
> simply don't know it
> > 
> >? ???br
> > 
> > 
> >? ???Diego Toro wrote:
> > 
> >? ? ? > Read "case
> SWITCH_EVENT_NOTIFY:" in mod_sofia.c. I suggest you
> >? ???assign call-id in the header
> of the event.
> >? ? ? >
> >? ? ? >
> >? ? ? > Diego Toro
> >? ? ? > http://lacarretade.blogspot.com/
> >? ? ? >
> >? ? ? >
> >? ? ? > --- On Mon, 2/22/10,
> Christian L?schenkohl
> >? ???<christian.loeschenkohl at xpirio.com
> >? ???<mailto:christian.loeschenkohl at xpirio.com>>
> wrote:
> >? ? ? >
> >? ? ? >> From: Christian
> L?schenkohl <christian.loeschenkohl at xpirio.com
> >? ???<mailto:christian.loeschenkohl at xpirio.com>>
> >? ? ? >> Subject:
> [Freeswitch-users] sending a sip notify with sendevent
> >? ? ? >> To: freeswitch-users at lists.freeswitch.org
> >? ???<mailto:freeswitch-users at lists.freeswitch.org>
> >? ? ? >> Date: Monday, February
> 22, 2010, 12:16 PM
> >? ? ? >> hi
> >? ? ? >>
> >? ? ? >> i try to send a sip
> notify message to a registered sip
> >? ? ? >> device
> >? ? ? >> "sofia profile nat
> flush_inbound_reg 10 at vts.vie.xpirio.net
> >? ???<mailto:10 at vts.vie.xpirio.net>
> >? ? ? >> reboot" works, but i
> need
> >? ? ? >> to send
> "check-sync;reboot=false" - so the device does a
> >? ? ? >> resync and don't do a
> reboot
> >? ? ? >>
> >? ? ? >> my message looks like
> this
> >? ? ? >>
> >? ? ? >> sendevent NOTIFY
> >? ? ? >> profile: nat
> >? ? ? >> event-string:
> check-sync;reboot=false
> >? ? ? >> user: 10
> >? ? ? >> host: vts.vie.xpirio.net
> <http://vts.vie.xpirio.net>
> >? ? ? >> content-type:
> application/simple-message-summary
> >? ? ? >>
> >? ? ? >> if i listen on the
> loopback interface i do see
> >? ? ? >>
> >? ? ? >> ##
> >? ? ? >> T 2010/02/22
> 18:11:59.083204 127.0.0.1:51840
> >? ???<http://127.0.0.1:51840> ->
> >? ? ? >> 127.0.0.1:8021 <http://127.0.0.1:8021> [AP]
> >? ? ? >> sendevent NOTIFY
> >? ? ? >> profile: nat
> >? ? ? >> event-string:
> check-sync;reboot=false
> >? ? ? >> user: 10
> >? ? ? >> host: vts.vie.xpirio.net
> <http://vts.vie.xpirio.net>
> >? ? ? >> content-type:
> application/simple-message-summary
> >? ? ? >>
> >? ? ? >> ##
> >? ? ? >> T 2010/02/22
> 18:11:59.084032 127.0.0.1:8021
> >? ???<http://127.0.0.1:8021> ->
> >? ? ? >> 127.0.0.1:51840 <http://127.0.0.1:51840> [AP]
> >? ? ? >> Content-Type:
> command/reply
> >? ? ? >> Reply-Text: -ERR invalid
> >? ? ? >>
> >? ? ? >> --------
> >? ? ? >> i don't get what it is
> wrong. i also rechecked the
> >? ? ? >> registered user in the
> sqlite database and this
> >? ? ? >> looks good to me.
> >? ? ? >>
> >? ? ? >> no message is send to the
> user.
> >? ? ? >>
> >? ? ? >> we do use multiple
> domains, so user could also be
> >? ???10 at somedomain.com
> <mailto:10 at somedomain.com>
> >? ? ? >> - or am i wrong on this?
> >? ? ? >> could somebody please
> bring some light in this.
> >? ? ? >>
> >? ? ? >> we do use trunk rev.
> 16631
> >? ? ? >>
> >? ? ? >> br
> >? ? ? >>
> >? ? ? >>
> >? ? ? >>
> >? ? ? >> --
> >? ? ? >> Ing. Christian
> L?schenkohl
> >? ? ? >> Technische Leitung,
> Forschung & Entwicklung VoIP
> >? ? ? >>
> >? ? ? >> xpirio
> >? ? ? >> Telekommunikation &
> Service GmbH
> >? ? ? >> Lakeside B04
> >? ? ? >> 9020 Klagenfurt
> >? ? ? >> Austria
> >? ? ? >>
> >? ? ? >> T? +43 (0) 5 77 11 -
> 1000
> >? ? ? >> F? +43 (0) 5 77 11 -
> 1002
> >? ? ? >> E? christian.loeschenkohl at xpirio.com
> >? ???<mailto:christian.loeschenkohl at xpirio.com>
> >? ? ? >>
> >? ? ? >>
> _______________________________________________
> >? ? ? >> FreeSWITCH-users mailing
> list
> >? ? ? >> FreeSWITCH-users at lists.freeswitch.org
> >? ???<mailto:FreeSWITCH-users at lists.freeswitch.org>
> >? ? ? >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >? ? ? >>
> >? ???UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >? ? ? >> http://www.freeswitch.org
> >? ? ? >>
> >? ? ? >
> >? ? ? >
> >? ? ? >
> >? ? ? >
> >? ? ? >
> _______________________________________________
> >? ? ? > FreeSWITCH-users mailing
> list
> >? ? ? > FreeSWITCH-users at lists.freeswitch.org
> >? ???<mailto:FreeSWITCH-users at lists.freeswitch.org>
> >? ? ? > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >? ? ? >
> >? ???UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >? ? ? > http://www.freeswitch.org
> > 
> > 
> >? ???--
> >? ???Ing. Christian L?schenkohl
> >? ???Technische Leitung, Forschung
> & Entwicklung VoIP
> > 
> >? ???xpirio
> >? ???Telekommunikation &
> Service GmbH
> >? ???Lakeside B04
> >? ???9020 Klagenfurt
> >? ???Austria
> > 
> >? ???T? +43 (0) 5 77 11 -
> 1000
> >? ???F? +43 (0) 5 77 11 -
> 1002
> >? ???E? christian.loeschenkohl at xpirio.com
> >? ???<mailto:christian.loeschenkohl at xpirio.com>
> > 
> >?
> ???_______________________________________________
> >? ???FreeSWITCH-users mailing list
> >? ???FreeSWITCH-users at lists.freeswitch.org
> >? ???<mailto:FreeSWITCH-users at lists.freeswitch.org>
> >? ???http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >? ???UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >? ???http://www.freeswitch.org
> > 
> > 
> > 
> > 
> > -- 
> > Anthony Minessale II
> > 
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> > Twitter: http://twitter.com/FreeSWITCH_wire
> > 
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com
> 
> > <mailto:MSN%3Aanthony_minessale at hotmail.com>
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> 
> > <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
> > IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
> > 
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org
> 
> > <mailto:sip%3A888 at conference.freeswitch.org>
> > iax:guest at conference.freeswitch.org/888
> 
> > <http://iax:guest at conference.freeswitch.org/888>
> > googletalk:conf+888 at conference.freeswitch.org
> 
> > <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > pstn:+19193869900
> > 
> > 
> >
> ------------------------------------------------------------------------
> > 
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 
> 
> -- 
> Ing. Christian L?schenkohl
> Technische Leitung, Forschung & Entwicklung VoIP
> 
> xpirio
> Telekommunikation & Service GmbH
> Lakeside B04
> 9020 Klagenfurt
> Austria
> 
> T? +43 (0) 5 77 11 - 1000
> F? +43 (0) 5 77 11 - 1002
> E? christian.loeschenkohl at xpirio.com
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 


      



From infos at madovsky.org  Mon Feb 22 19:29:01 2010
From: infos at madovsky.org (Madovsky)
Date: Mon, 22 Feb 2010 22:29:01 -0500
Subject: [Freeswitch-users] call from an internal extension to external
	number
Message-ID: <D616CD3E2BFC4FDDA7ABEE7A3F268A4B@MOBILEE1705>

Hi,

day after I undertand a littlee more all these xml hell files (not friendly to read ;)),
but to be a PERl developer since 1999 understand regex and PERL language make life more easy...
However, I don't understand yet the concept of internal exterenal.
is it for phone registration AND outbound calls ?
for now I try to make an external call from 1000 ext (registered on port 5060)
so I added an extension in dialplan/default.xml 

    <extension name="outbound_europe">
        <condition field="destination_number" expression="^00?(\d+)$">
              <action application="bridge" data="sofia/gateway/myprovider_europe/00$1"/>
        </condition>
    </extension>

so if call starts with "00" it redirects to my provider that manage outbound calls, is it correct ?
I put the myprovider.xml account into sip_profiles/external/myprovider.xml.

Thanks for your help

Franck
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100222/166da759/attachment-0002.html 

From tim at communicatefreely.net  Mon Feb 22 20:06:15 2010
From: tim at communicatefreely.net (Tim St. Pierre)
Date: Mon, 22 Feb 2010 23:06:15 -0500
Subject: [Freeswitch-users] Is there any way to loop a dialplan?
In-Reply-To: <4B82E6FA.3090008@marketelsystems.com>
References: <4B82E6FA.3090008@marketelsystems.com>
Message-ID: <4B835437.2070803@communicatefreely.net>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Why not just use transfer?

Break your dialplan up into segments - one that does the authentication, and another that has the
call flow to the parking pool, and the post call work.  You can use the transfer application to
connect these segments together.  Variables are preserved across transfers, so things like the agent
ID and their authenticated status can be set in a variable.  You can also make routing decisions
based on the value of a variable.  In your condition statement, use the variable name
"${my_variable}" as the field, and then a very simple pattern match that decides if it's valid or
not.  You can also use the execute_extension application as a sort of "macro", to execute another
dialplan block, but return when it completes.  At the bottom of your dialplan, transfer back to the top.

Hope that's helpful.  Sorry I don't have any example code.  I'm generating XML dynamically in PHP,
but the above concept seems to work well.

- -Tim

Mark Sobkow wrote:
> Let me explain what it is I'm trying to do.  Maybe there's another way 
> to achieve it.
> 
> When an operator dials in to the log-in line (e.g. Extension 6000), I 
> use play_and_get_digits to collect the operator's PIN.  I then need to 
> be able to fire up some Erlang (or Javascript) to verify the PIN, and 
> after verification, put the call into a park state, collecting the 
> UUID.  I then need to fire an event to Erlang passing along the parked 
> UUID and the operator's PIN so that Erlang can direct received customer 
> calls to the operators based on relatively complex criteria that won't 
> fit in a dialplan.
> 
> The catch is that when I get a customer call, I collect their info via 
> IVR menus, park the call, and fire an event to Erlang with the UUID of 
> the parked call and info collected from the IVR.  Erlang analyses the 
> info, selects an operator who is free, and bridges the calls.
> 
> The problem I'm having is figuring out a way to get the operator leg to 
> go back into a park state after handling the customer's call.  Ideally I 
> want the operator to be presented with a short IVR to collect info about 
> how the call was handled, but I can do that through a custom application 
> GUI if I need to.  Regardless, once an operator logs in, they need to 
> _stay_ logged in until they explicityly log out, but I can't figure out 
> how I'm supposed to do that without some sort of looping capability.
> 
> One thing I was thinking that might work is to set up a set of "dummy" 
> extensions that I can have Erlang dial and bridge which contain a 
> dialplan fragment to collect the IVR call result, park the call, and 
> issue the operator PIN and parked UUID again to Erlang.  That way 
> between Erlang events and the dialplan fragment I end up with an 
> effective "loop".  (Though I've yet to figure out how I can break out of 
> that loop.  Maybe it'll have to be an IVR option for logging out.)
> 
> Sample code/dialplans would be good, but for now I'll settle for knowing 
> whether I'm at least on the right track for how to implement this beast.
> 
> Note that I only want to drop into Javascript if I can't figure out how 
> to do it with dialplans and Erlang.
> 
> Thanks for any ideas and suggestions.
> 

- --
Tim St. Pierre
IP Voice technician
Communicate Freely
1-877-291-8647 x5101
sip:5101 at communicatefreely.net
tim at communicatefreely.net
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v2.0.4 (FreeBSD)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org

iQCVAwUBS4NUN4qVcvNCnHOrAQIDcgP/SpzpLUpsnFjGaamy4EbUw95l2mDHrEYa
ay1cbciSV5qICRLoDvTrleqYkrMhgRlvzxvkLRRzFIOPjm4+cFQMojmMS5HZZQiJ
TWndXAiZGgtlKqEDfgqr1ea2BcXi/oozsJIk0iePgPLIGlMUa/O2p3kaizQzPMc7
fbMNwcSYSc8=
=KtCw
-----END PGP SIGNATURE-----



From troy at tlainvestments.com  Mon Feb 22 20:30:14 2010
From: troy at tlainvestments.com (Troy Anderson)
Date: Mon, 22 Feb 2010 21:30:14 -0700
Subject: [Freeswitch-users] Hook Flash
Message-ID: <915F9D0D-FB13-4D02-995B-FB7F5EB488D2@tlainvestments.com>

Is there a way to cause a hook-flash on a zap channel via the dial plan?  e.g. VoIP phone <=> FS <=> POTS line, and I want to flash via some star code on the phone.

I'm happy to document on the openzap page on the wiki if so.

Thanks,
Troy




From infos at madovsky.org  Mon Feb 22 22:41:04 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 23 Feb 2010 01:41:04 -0500
Subject: [Freeswitch-users] FScomm
Message-ID: <157FDB96FB88476FBAEC69C1309A0A5F@MOBILEE1705>

Hi,

is http://wiki.freeswitch.org/wiki/FSComm
available yet ? because QT framework link has broken
and some instruction don't work with last svn trunk (for example to compile FScomm in FS svn root)

Thanks

Franck
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100223/81c1d0f7/attachment-0002.html 

From infos at madovsky.org  Mon Feb 22 23:06:12 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 23 Feb 2010 02:06:12 -0500
Subject: [Freeswitch-users] rtp timeout and call hangs up
Message-ID: <3F9354A277B544A98E35F8C3F5900496@MOBILEE1705>

Hello,

A leg local extension (codec GSM)  -----> B leg local extension (codec PCMU)
rtp timeout after 300 sec the call hangs up,
I can ear audio and no problem of configuration.
did I forget to set anything ?

below the sip trace :

ACK sip:1001 at 67.xx.xx.138:5060;transport=udp SIP/2.0
Content-Length: 0
Via: SIP/2.0/UDP 67.xx.xx.138:62690;rport;branch=z9hG4bK4t7xGwrintJDYr4HPfz0UQ..
From: "1000" <sip:1000 at 67.xx.xx.138:5080>;tag=52834814554
To: "Extension 1001" <sip:1001 at 67.xx.xx.138>;tag=3SKarSKZB63DF
Contact: <sip:1000 at 67.xx.xx.138:62690>
CSeq: 1 ACK
Max-Forwards: 70
Call-ID: 2053375473 at 67.xx.xx.138

------------------------------------------------------------------------
2010-02-23 02:02:19.649316 [NOTICE] mod_sofia.c:853 Hangup sofia/external/1000 at 67.xx.xx.138:5080 [CS_EXECUTE] [MEDIA_TIMEOUT]
2010-02-23 02:02:19.649316 [NOTICE] switch_ivr_bridge.c:634 Hangup sofia/internal/sip:1001 at 70.81.84.218:2249 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
send 622 bytes to udp/[67.xx.xx.138]:62690 at 07:02:19.653239:
------------------------------------------------------------------------
BYE sip:1000 at 67.xx.xx.138:62690 SIP/2.0
Via: SIP/2.0/UDP 67.xx.xx.138;rport;branch=z9hG4bK790g187j50eNF
Max-Forwards: 70
From: "Extension 1001" <sip:1001 at 67.xx.xx.138>;tag=3SKarSKZB63DF
To: "1000" <sip:1000 at 67.xx.xx.138:5080>;tag=52834814554
Call-ID: 2053375473 at 67.xx.xx.138
CSeq: 127335933 BYE
Contact: <sip:1001 at 67.xx.xx.138:5060;transport=udp>
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16676M
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY
Supported: timer, precondition, path, replaces
Reason: FreeSWITCH;cause=604;text="MEDIA_TIMEOUT"
Content-Length: 0

------------------------------------------------------------------------
2010-02-23 02:02:19.649316 [NOTICE] switch_core_session.c:1179 Session 5 (sofia/external/1000 at 67.xx.xx.138:5080) Ended
2010-02-23 02:02:19.649316 [NOTICE] switch_core_session.c:1181 Close Channel sofia/external/1000 at 67.xx.xx.138:5080 [CS_DESTROY]
recv 289 bytes from udp/[67.xx.xx.138]:62690 at 07:02:19.656927:
------------------------------------------------------------------------
SIP/2.0 200 OK
Content-Length: 0
Via: SIP/2.0/UDP 67.xx.xx.138;rport=5060;branch=z9hG4bK790g187j50eNF
From: "Extension 1001" <sip:1001 at 67.xx.xx.138>;tag=3SKarSKZB63DF
To: "1000" <sip:1000 at 67.xx.xx.138:5080>;tag=52834814554
CSeq: 127335933 BYE
Call-ID: 2053375473 at 67.xx.xx.138

------------------------------------------------------------------------
recv 391 bytes from udp/[67.xx.xx.138]:62690 at 07:02:19.660173:
------------------------------------------------------------------------
BYE sip:1001 at 67.xx.xx.138:5060;transport=udp SIP/2.0
Content-Length: 0
Via: SIP/2.0/UDP 67.xx.xx.138:62690;rport;branch=z9hG4bKGTIQmsc5dUvSdP0ZtJFKcA..
From: "1000" <sip:1000 at 67.xx.xx.138:5080>;tag=52834814554
To: "Extension 1001" <sip:1001 at 67.xx.xx.138>;tag=3SKarSKZB63DF
Contact: <sip:1000 at 67.xx.xx.138:62690>
CSeq: 2 BYE
Max-Forwards: 70
Call-ID: 2053375473 at 67.xx.xx.138

------------------------------------------------------------------------
send 505 bytes to udp/[67.xx.xx.138]:62690 at 07:02:19.660607:
------------------------------------------------------------------------
SIP/2.0 481 Call Does Not Exist
Via: SIP/2.0/UDP 67.xx.xx.138:62690;rport=62690;branch=z9hG4bKGTIQmsc5dUvSdP0ZtJFKcA..
From: "1000" <sip:1000 at 67.xx.xx.138:5080>;tag=52834814554
To: "Extension 1001" <sip:1001 at 67.xx.xx.138>;tag=3SKarSKZB63DF
Call-ID: 2053375473 at 67.xx.xx.138
CSeq: 2 BYE
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16676M
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY
Supported: timer, precondition, path, replaces
Content-Length: 0

------------------------------------------------------------------------
send 669 bytes to udp/[70.81.84.218]:2249 at 07:02:19.869656:
------------------------------------------------------------------------
BYE sip:1001 at 70.81.84.218:2249;rinstance=f17f8a2dd028c23d SIP/2.0
Via: SIP/2.0/UDP 67.xx.xx.138:5080;rport;branch=z9hG4bKQ1y6QNQ56aDDS
Max-Forwards: 70
From: "1000" <sip:1000 at 67.xx.xx.138>;tag=SK3p6Frt5pa7D
To: <sip:1001 at 70.81.84.218:2249;rinstance=f17f8a2dd028c23d>;tag=df099549
Call-ID: 263d4030-9aec-122d-6d87-00e0ed0b00c2
CSeq: 127335918 BYE
Contact: <sip:mod_sofia at 67.xx.xx.138:5080>
User-Agent: FreeSWITCH and Rock!
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, precondition, path, replaces
Reason: Q.850;cause=16;text="NORMAL_CLEARING"
Content-Length: 0

------------------------------------------------------------------------
2010-02-23 02:02:19.869666 [NOTICE] switch_core_session.c:1179 Session 6 (sofia/internal/sip:1001 at 70.81.84.218:2249) Ended
2010-02-23 02:02:19.869666 [NOTICE] switch_core_session.c:1181 Close Channel sofia/internal/sip:1001 at 70.81.84.218:2249 [CS_DESTROY]
recv 411 bytes from udp/[70.81.84.218]:2249 at 07:02:19.986415:
------------------------------------------------------------------------
SIP/2.0 200 OK
Via: SIP/2.0/UDP 67.xx.xx.138:5080;rport=5080;branch=z9hG4bKQ1y6QNQ56aDDS
Contact: <sip:1001 at 70.81.84.218:2249;rinstance=f17f8a2dd028c23d>
To: <sip:1001 at 70.81.84.218:2249;rinstance=f17f8a2dd028c23d>;tag=df099549
From: "1000"<sip:1000 at 67.xx.xx.138>;tag=SK3p6Frt5pa7D
Call-ID: 263d4030-9aec-122d-6d87-00e0ed0b00c2
CSeq: 127335918 BYE
User-Agent: 3CXPhone 4.0.10858.0
Content-Length: 0

------------------------------------------------------------------------





Thanks


Franck
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100223/578bfa97/attachment-0002.html 

From moizchinoy at gmail.com  Mon Feb 22 23:59:04 2010
From: moizchinoy at gmail.com (Moiz Chinoy)
Date: Tue, 23 Feb 2010 11:59:04 +0400
Subject: [Freeswitch-users] Dingaling - external rtp supported?
In-Reply-To: <29b888f81002210800v617bca9dk4bf1abf1bd791ee0@mail.gmail.com>
References: <29b888f81002160835y6e290319ub0d04c088caa1b79@mail.gmail.com>
	<DEA39AD1-0EE9-473A-B78C-CDDF3F100578@freeswitch.org>
	<29b888f81002170318p63622f3fhc8d26ad0e85602d2@mail.gmail.com>
	<191c3a031002170538r3b029515i8a000812a70ad558@mail.gmail.com>
	<29b888f81002170741y3d13cb57ycb4436a372c4eded@mail.gmail.com>
	<191c3a031002171136q7424e9bai34379abb542daf93@mail.gmail.com>
	<29b888f81002210800v617bca9dk4bf1abf1bd791ee0@mail.gmail.com>
Message-ID: <29b888f81002222359w6ae4cd8ej944d453a5b099af4@mail.gmail.com>

Moreover, if I gtalk client is on the same machine as FS and i have
following settings, FS crashes with the same mutex error.

External Sip Profile has following lines:
---------------------------------------------------------
   <param name="rtp-ip" value="$${local_ip_v4}"/>
   <param name="sip-ip" value="$${local_ip_v4}"/>
   <param name="ext-rtp-ip" value="auto-nat"/>
   <param name="ext-sip-ip" value="auto-nat"/>

Jingle Client.xml has following lines:
-----------------------------------------------------
   <param name="rtp-ip" value="$${bind_server_ip}"/>
   <!--<param name="ext-rtp-ip" value="$${external_rtp_ip}"/>-->
   <!--<param name="disable-rtp-auto-adjust" value="true"/>-->
   <!--<param name="avatar" value="/path/to/tiny.jpg"/>-->
   <!--<param name="candidate-acl" value="wan.auto"/>-->
   <!--<param name="local-network-acl" value="localnet.auto"/>-->

If I uncomment the following line in client.xml (Jingle profile)
   <!--<param name="local-network-acl" value="localnet.auto"/>-->
then exception does not happen.

Is this a known issue or do I need to post it in JIRA?

Tell me if more logs are needed...


On Sun, Feb 21, 2010 at 8:00 PM, Moiz Chinoy <moizchinoy at gmail.com> wrote:
> Guys,
>
> To make things simple gtalk client is entirely on different network.
>
> Call comes from outside through external Sip profile.
>
> If gtalk answers the call after 3-4 rings both parties can hear each other.
> If gtalk answers the call after 2 rings both parties no one can hear each other.
> If gtalk answers the call immediately FS crashes.
>
> Attached is the screen shot of the error...
>
> Here is the FS log...
> --------------------------------
> http://pastebin.freeswitch.org/12197
>
> External Sip Profile has following lines:
> ---------------------------------------------------------
> ? ?<param name="rtp-ip" value="$${local_ip_v4}"/>
> ? ?<param name="sip-ip" value="$${local_ip_v4}"/>
> ? ?<param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
> ? ?<param name="ext-sip-ip" value="$${external_sip_ip}"/>
>
> Jingle Client.xml has following lines:
> -----------------------------------------------------
> ? ?<param name="rtp-ip" value="$${bind_server_ip}"/>
> ? ?<param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
> ? ?<param name="disable-rtp-auto-adjust" value="true"/>
> ? ?<!--<param name="avatar" value="/path/to/tiny.jpg"/>-->
> ? ?<!--<param name="candidate-acl" value="wan.auto"/>-->
> ? ?<!--<param name="local-network-acl" value="localnet.auto"/>-->
>
> Vars.xml has following lines:
> -------------------------------------------
> <X-PRE-PROCESS cmd="set" data="external_rtp_ip=stun:stun.freeswitch.org"/>
> <X-PRE-PROCESS cmd="set" data="external_sip_ip=stun:stun.freeswitch.org"/>
>
>
> Please advise me how can I provide more of the required data.
>
> On Wed, Feb 17, 2010 at 11:36 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>> you cant combine stun and gtalk and boxes in the same lan very easily if you
>> do need to do that you will need to mess with
>>
>> <param name="candidate-acl" value="wan.auto"/>
>> <param name="local-network-acl" value="localnet.auto"/>
>>
>>
>>
>>
>>
>> On Wed, Feb 17, 2010 at 9:41 AM, Moiz Chinoy <moizchinoy at gmail.com> wrote:
>>>
>>> Guys I am unable to produce the crash but now both parties cannot hear
>>> each other!
>>>
>>> Vars.xml has following lines:
>>> ?<X-PRE-PROCESS cmd="set"
>>> data="external_rtp_ip=stun:stun.freeswitch.org"/>
>>> ?<X-PRE-PROCESS cmd="set"
>>> data="external_sip_ip=stun:stun.freeswitch.org"/>
>>>
>>> Jingle Client.xml has following lines:
>>> ? ?<param name="rtp-ip" value="$${bind_server_ip}"/>
>>> ? ?<!-- <param name="ext-rtp-ip" value="auto-nat"/> -->
>>> ? ?<param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
>>> ? ?<param name="disable-rtp-auto-adjust" value="true"/>
>>> ? ?<param name="local-network-acl" value="localnet.auto"/>
>>>
>>> On Wed, Feb 17, 2010 at 5:38 PM, Anthony Minessale
>>> <anthony.minessale at gmail.com> wrote:
>>> > Obtain a stack trace from the crash.
>>> >
>>> > On Feb 17, 2010 5:26 AM, "Moiz Chinoy" <moizchinoy at gmail.com> wrote:
>>> >
>>> > Hi,
>>> >
>>> > FS rev: 16673
>>> > Platform: Windows
>>> >
>>> > More details:
>>> >
>>> > FS is behind NAT and machine is running a VPN connection.
>>> >
>>> > FS and GTalk client on the same machine:
>>> >
>>> > --------------------------------------------------------------------------------------------------
>>> > jingle profile client.xml has following line:
>>> > <param name="rtp-ip" value="$${bind_server_ip}"/>
>>> >
>>> > External SIP call is successfully bridged to GTalk client.
>>> >
>>> >
>>> > FS and GTalk client on the different machine:
>>> >
>>> > --------------------------------------------------------------------------------------------------
>>> > jingle profile client.xml has following lines:
>>> > <param name="rtp-ip" value="$${bind_server_ip}"/>
>>> >
>>> > <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
>>> >
>>> > As soon as external SIP call land and I try to bridge the call to
>>> > GTalk client, FS crashes.
>>> >
>>> >
>>> > NAT Details:
>>> > ---------------------------
>>> > I think my NAT does not support UpNP or PMP. The reason I say it
>>> > because when FS starts following message is displayed:
>>> >
>>> > 2010-02-17 09:15:42.809674 [INFO] switch_nat.c:409 Scanning for NAT
>>> > 2010-02-17 09:15:42.809674 [ERR] switch_nat.c:197 Error checking for
>>> > PMP [init failed]
>>> > 2010-02-17 09:15:42.809674 [DEBUG] switch_nat.c:414 Checking for UPnP
>>> > 2010-02-17 09:15:56.825388 [DEBUG] switch_nat.c:114 No
>>> > InternetGatewayDevice, using first entry as default
>>> > (http://192.168.16.17:50144/).
>>> > 2010-02-17 09:15:56.887889 [INFO] switch_nat.c:429 No PMP or UPnP NAT
>>> > devices detected!
>>> >
>>> >
>>> >
>>> > On Tue, Feb 16, 2010 at 8:41 PM, Brian West <brian at freeswitch.org>
>>> > wrote:
>>> >> can you please update...
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>> >
>>>
>>>
>>>
>>> --
>>> Regards,
>>> Moiz Chinoy.
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
>
> --
> Regards,
> Moiz Chinoy.
>



-- 
Regards,
Moiz Chinoy.



From helmut.kuper at ewetel.de  Tue Feb 23 01:41:59 2010
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Tue, 23 Feb 2010 10:41:59 +0100
Subject: [Freeswitch-users] Question about sofia_contact
In-Reply-To: <191c3a031002220842j5bec442an5f1ea89cb0e8a6ff@mail.gmail.com>
References: <4B7EA954.30402@ewetel.de>
	<4B7EADBC.1040001@ewetel.de>	<4B8249BF.3090708@ewetel.de>
	<191c3a031002220842j5bec442an5f1ea89cb0e8a6ff@mail.gmail.com>
Message-ID: <4B83A2E7.1060905@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hello Anthony,


you are right, I'm quite unpatient, sorry 4 that. Your solution works
fine. I thought the sip domain could be any string and must not be a
valid domain format.

Thanks to you, board and community for this fantastic project!

regards from rainy germany
Helmut



On 22.02.2010 17:42, Anthony Minessale wrote:
> it's mad at you for asking twice before waiting for a reply, so it's not
> working on purpose.
> 
> Actually it's mad at you because your domain does not contain a . so it
> is assuming you are specifying a profile name as the domain.  if your
> domain was mydomain.com <http://mydomain.com> instead it would work.
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.4.7 (MingW32)

iD8DBQFLg6Ln4tZeNddg3dwRAvj6AJ9ruybNpbL8mdUlx1jVtLPYVbCSDACfQJLo
zfieJnHZdp2Xv3OS6HTZE/k=
=ESgY
-----END PGP SIGNATURE-----



From nagalenoj at gmail.com  Tue Feb 23 02:02:46 2010
From: nagalenoj at gmail.com (Nagalenoj H.)
Date: Tue, 23 Feb 2010 15:32:46 +0530
Subject: [Freeswitch-users] Error loading
	/usr/local/freeswitch/mod/ozmod_sangoma_boost.so
Message-ID: <fe67a3341002230202y1d45a8f3t3a2cda07de2240a2@mail.gmail.com>

Dear friends,
    I've installed freeswitch trunk - 16729 and tried to configure with
wanpipe for sangoma A102 pri card.

Followed the steps given in
http://wiki.sangoma.com/wanpipe-freeswitch-install

When loading the freeswitch, I've got the following error.

2010-02-23 14:49:58.545726 [INFO] ozmod_wanpipe.c:335 configuring device
s2c15 as OpenZAP device 1:30 fd:57 DTMF: software
2010-02-23 14:49:58.546004 [ERR] ozmod_wanpipe.c:436 No valid wanpipe span
and channel was specified
2010-02-23 14:49:58.546275 [INFO] zap_io.c:2500 Configured 30 channel(s)
2010-02-23 14:49:58.560724 [ERR] zap_io.c:2560 Error loading
/usr/local/freeswitch/mod/ozmod_sangoma_boost.so
[/usr/local/freeswitch/mod/ozmod_sangoma_boost.so: cannot open shared object
file: No such file or directory]
2010-02-23 14:49:58.561048 [ERR] zap_io.c:2720 can't find 'sangoma_boost'
2010-02-23 14:49:58.561266 [ERR] mod_openzap.c:2458 Error starting OpenZAP
span 1 error:
2010-02-23 14:49:58.561712 [NOTICE] switch_loadable_module.c:144 Adding
Endpoint 'openzap'

Configuration and log files are pasted to pastebin. Kindly someone help me
to solve this issue.

openzap.conf and openzap.conf.xml
   http://pastebin.freeswitch.org/12214

freeswitch log
   http://pastebin.freeswitch.org/12216

smg_pri.conf
   http://pastebin.freeswitch.org/12217

-- 
Regards,
Nagalenoj H.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100223/35cf0dcb/attachment-0002.html 

From steveayre at gmail.com  Tue Feb 23 02:54:52 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 23 Feb 2010 10:54:52 +0000
Subject: [Freeswitch-users] rtp timeout and call hangs up
In-Reply-To: <3F9354A277B544A98E35F8C3F5900496@MOBILEE1705>
References: <3F9354A277B544A98E35F8C3F5900496@MOBILEE1705>
Message-ID: <eafa772f1002230254v7fcc8ae8mce35a19339a36170@mail.gmail.com>

Hi Franck,

Sorry, you haven't provided enough of the trace or logs to see the
reason for the timeout.

To hazard a guess though... are you using RTP in bypass media mode? If
so RTP won't go via the switch, so it won't see any RTP for the call
even though media's working - FS will then end the call if
rtp-timeout-sec is set in sofia.conf.xml.

Regards,
-Steve



On 23 February 2010 07:06, Madovsky <infos at madovsky.org> wrote:
> Hello,
>
> A leg local extension (codec GSM) ?-----> B leg local extension (codec PCMU)
> rtp timeout after 300 sec the call hangs up,
> I can ear audio and no problem of configuration.
> did I forget to set anything ?
>
> below the sip trace :
>
>
> ACK sip:1001 at 67.xx.xx.138:5060;transport=udp SIP/2.0
> Content-Length: 0
> Via: SIP/2.0/UDP
> 67.xx.xx.138:62690;rport;branch=z9hG4bK4t7xGwrintJDYr4HPfz0UQ..
> From: "1000" <sip:1000 at 67.xx.xx.138:5080>;tag=52834814554
> To: "Extension 1001" <sip:1001 at 67.xx.xx.138>;tag=3SKarSKZB63DF
> Contact: <sip:1000 at 67.xx.xx.138:62690>
> CSeq: 1 ACK
> Max-Forwards: 70
> Call-ID: 2053375473 at 67.xx.xx.138
>
> ------------------------------------------------------------------------
> 2010-02-23 02:02:19.649316 [NOTICE] mod_sofia.c:853 Hangup
> sofia/external/1000 at 67.xx.xx.138:5080 [CS_EXECUTE] [MEDIA_TIMEOUT]
> 2010-02-23 02:02:19.649316 [NOTICE] switch_ivr_bridge.c:634 Hangup
> sofia/internal/sip:1001 at 70.81.84.218:2249 [CS_EXCHANGE_MEDIA]
> [NORMAL_CLEARING]
> send 622 bytes to udp/[67.xx.xx.138]:62690 at 07:02:19.653239:
> ------------------------------------------------------------------------
> BYE sip:1000 at 67.xx.xx.138:62690 SIP/2.0
> Via: SIP/2.0/UDP 67.xx.xx.138;rport;branch=z9hG4bK790g187j50eNF
> Max-Forwards: 70
> From: "Extension 1001" <sip:1001 at 67.xx.xx.138>;tag=3SKarSKZB63DF
> To: "1000" <sip:1000 at 67.xx.xx.138:5080>;tag=52834814554
> Call-ID: 2053375473 at 67.xx.xx.138
> CSeq: 127335933 BYE
> Contact: <sip:1001 at 67.xx.xx.138:5060;transport=udp>
> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16676M
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER,
> REFER, NOTIFY
> Supported: timer, precondition, path, replaces
> Reason: FreeSWITCH;cause=604;text="MEDIA_TIMEOUT"
> Content-Length: 0
>
> ------------------------------------------------------------------------
> 2010-02-23 02:02:19.649316 [NOTICE] switch_core_session.c:1179 Session 5
> (sofia/external/1000 at 67.xx.xx.138:5080) Ended
> 2010-02-23 02:02:19.649316 [NOTICE] switch_core_session.c:1181 Close Channel
> sofia/external/1000 at 67.xx.xx.138:5080 [CS_DESTROY]
> recv 289 bytes from udp/[67.xx.xx.138]:62690 at 07:02:19.656927:
> ------------------------------------------------------------------------
> SIP/2.0 200 OK
> Content-Length: 0
> Via: SIP/2.0/UDP 67.xx.xx.138;rport=5060;branch=z9hG4bK790g187j50eNF
> From: "Extension 1001" <sip:1001 at 67.xx.xx.138>;tag=3SKarSKZB63DF
> To: "1000" <sip:1000 at 67.xx.xx.138:5080>;tag=52834814554
> CSeq: 127335933 BYE
> Call-ID: 2053375473 at 67.xx.xx.138
>
> ------------------------------------------------------------------------
> recv 391 bytes from udp/[67.xx.xx.138]:62690 at 07:02:19.660173:
> ------------------------------------------------------------------------
> BYE sip:1001 at 67.xx.xx.138:5060;transport=udp SIP/2.0
> Content-Length: 0
> Via: SIP/2.0/UDP
> 67.xx.xx.138:62690;rport;branch=z9hG4bKGTIQmsc5dUvSdP0ZtJFKcA..
> From: "1000" <sip:1000 at 67.xx.xx.138:5080>;tag=52834814554
> To: "Extension 1001" <sip:1001 at 67.xx.xx.138>;tag=3SKarSKZB63DF
> Contact: <sip:1000 at 67.xx.xx.138:62690>
> CSeq: 2 BYE
> Max-Forwards: 70
> Call-ID: 2053375473 at 67.xx.xx.138
>
> ------------------------------------------------------------------------
> send 505 bytes to udp/[67.xx.xx.138]:62690 at 07:02:19.660607:
> ------------------------------------------------------------------------
> SIP/2.0 481 Call Does Not Exist
> Via: SIP/2.0/UDP
> 67.xx.xx.138:62690;rport=62690;branch=z9hG4bKGTIQmsc5dUvSdP0ZtJFKcA..
> From: "1000" <sip:1000 at 67.xx.xx.138:5080>;tag=52834814554
> To: "Extension 1001" <sip:1001 at 67.xx.xx.138>;tag=3SKarSKZB63DF
> Call-ID: 2053375473 at 67.xx.xx.138
> CSeq: 2 BYE
> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16676M
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER,
> REFER, NOTIFY
> Supported: timer, precondition, path, replaces
> Content-Length: 0
>
> ------------------------------------------------------------------------
> send 669 bytes to udp/[70.81.84.218]:2249 at 07:02:19.869656:
> ------------------------------------------------------------------------
> BYE sip:1001 at 70.81.84.218:2249;rinstance=f17f8a2dd028c23d SIP/2.0
> Via: SIP/2.0/UDP 67.xx.xx.138:5080;rport;branch=z9hG4bKQ1y6QNQ56aDDS
> Max-Forwards: 70
> From: "1000" <sip:1000 at 67.xx.xx.138>;tag=SK3p6Frt5pa7D
> To: <sip:1001 at 70.81.84.218:2249;rinstance=f17f8a2dd028c23d>;tag=df099549
> Call-ID: 263d4030-9aec-122d-6d87-00e0ed0b00c2
> CSeq: 127335918 BYE
> Contact: <sip:mod_sofia at 67.xx.xx.138:5080>
> User-Agent: FreeSWITCH and Rock!
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER,
> REFER, NOTIFY, PUBLISH, SUBSCRIBE
> Supported: timer, precondition, path, replaces
> Reason: Q.850;cause=16;text="NORMAL_CLEARING"
> Content-Length: 0
>
> ------------------------------------------------------------------------
> 2010-02-23 02:02:19.869666 [NOTICE] switch_core_session.c:1179 Session 6
> (sofia/internal/sip:1001 at 70.81.84.218:2249) Ended
> 2010-02-23 02:02:19.869666 [NOTICE] switch_core_session.c:1181 Close Channel
> sofia/internal/sip:1001 at 70.81.84.218:2249 [CS_DESTROY]
> recv 411 bytes from udp/[70.81.84.218]:2249 at 07:02:19.986415:
> ------------------------------------------------------------------------
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 67.xx.xx.138:5080;rport=5080;branch=z9hG4bKQ1y6QNQ56aDDS
> Contact: <sip:1001 at 70.81.84.218:2249;rinstance=f17f8a2dd028c23d>
> To: <sip:1001 at 70.81.84.218:2249;rinstance=f17f8a2dd028c23d>;tag=df099549
> From: "1000"<sip:1000 at 67.xx.xx.138>;tag=SK3p6Frt5pa7D
> Call-ID: 263d4030-9aec-122d-6d87-00e0ed0b00c2
> CSeq: 127335918 BYE
> User-Agent: 3CXPhone 4.0.10858.0
> Content-Length: 0
>
> ------------------------------------------------------------------------
>
>
>
>
>
> Thanks
>
> Franck
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From technical at ttnc.co.uk  Tue Feb 23 04:21:02 2010
From: technical at ttnc.co.uk (TTNC - Technical)
Date: Tue, 23 Feb 2010 12:21:02 +0000
Subject: [Freeswitch-users] leg_timeout with ignore_early_media false
Message-ID: <F86E6E00-B774-487F-9F62-2B1BB00382FC@ttnc.co.uk>

Hi Guys

I'm trying to create a hunt group where ignore_early_media = false is set, so that the international ring tone is passed through to the caller. Setting ignore_early_media = false on the channel does what I want, but with this set leg_timeout is not honoured.

I've switched to use bridge_answer_timeout which is honoured if ignore_early_media = false and the call progresses through the different legs, but bridge_answer_timeout times out the call after the set period, even if the call has been successfully answered and bridged. I've tried all different combinations of group_confirm_cancel_timeout [1|2|3], none of them seem to affect bridge_answer_timeout.

Does anyone have a solution for timing out legs of a hunt group with ignore_early_media = false set?

This is my dial string:

'{caller-id-in-from=true,origination_caller_id_name=012345123123,origination_caller_id_number= 012345123123}[bridge_answer_timeout=20]sofia/internal/4412345123123 at sipipgw.siphost.net |sofia/internal/4412345123123 at sipipgw.siphost.net'

And in my lua application, I'm setting the following:

session:setVariable("group_confirm_cancel_timeout", "1"); -- substitute 1 for 1, 2 or 3, none work.
session:setVariable("ignore_early_media", "false");

I'm using the latest trunk revision - it's still happening.

Any suggestions welcome.

Thanks

Russ




From dftoro at yahoo.com  Tue Feb 23 05:33:12 2010
From: dftoro at yahoo.com (Diego Toro)
Date: Tue, 23 Feb 2010 05:33:12 -0800 (PST)
Subject: [Freeswitch-users] Hook Flash
In-Reply-To: <915F9D0D-FB13-4D02-995B-FB7F5EB488D2@tlainvestments.com>
Message-ID: <742756.71167.qm@web33501.mail.mud.yahoo.com>

hi, read http://jira.freeswitch.org/browse/OPENZAP-30

  <action application="send_dtmf" data="F"/>

Diego Toro
http://lacarretade.blogspot.com/


--- On Mon, 2/22/10, Troy Anderson <troy at tlainvestments.com> wrote:

> From: Troy Anderson <troy at tlainvestments.com>
> Subject: [Freeswitch-users] Hook Flash
> To: freeswitch-users at lists.freeswitch.org
> Date: Monday, February 22, 2010, 11:30 PM
> Is there a way to cause a hook-flash
> on a zap channel via the dial plan?? e.g. VoIP phone
> <=> FS <=> POTS line, and I want to flash via
> some star code on the phone.
> 
> I'm happy to document on the openzap page on the wiki if
> so.
> 
> Thanks,
> Troy
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 


      



From jeff at jefflenk.com  Tue Feb 23 06:23:54 2010
From: jeff at jefflenk.com (Jeff Lenk)
Date: Tue, 23 Feb 2010 08:23:54 -0600
Subject: [Freeswitch-users] FScomm
In-Reply-To: <157FDB96FB88476FBAEC69C1309A0A5F@MOBILEE1705>
References: <157FDB96FB88476FBAEC69C1309A0A5F@MOBILEE1705>
Message-ID: <COL117-W5563292C886257AFC56548B9420@phx.gbl>


What platform are you trying to build?
 


From: infos at madovsky.org
To: freeswitch-users at lists.freeswitch.org
Date: Tue, 23 Feb 2010 01:41:04 -0500
Subject: [Freeswitch-users] FScomm




Hi,
 
is http://wiki.freeswitch.org/wiki/FSComm
available yet ? because QT framework link has broken
and some instruction don't work with last svn trunk (for example to compile FScomm in FS svn root)
 
Thanks

Franck 		 	   		  
_________________________________________________________________
Your E-mail and More On-the-Go. Get Windows Live Hotmail Free.
http://clk.atdmt.com/GBL/go/201469229/direct/01/
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100223/8cc247d7/attachment-0002.html 

From christian.loeschenkohl at xpirio.com  Tue Feb 23 06:56:28 2010
From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=)
Date: Tue, 23 Feb 2010 15:56:28 +0100
Subject: [Freeswitch-users] sending a sip notify with sendevent
In-Reply-To: <12033.93456.qm@web33507.mail.mud.yahoo.com>
References: <12033.93456.qm@web33507.mail.mud.yahoo.com>
Message-ID: <4B83EC9C.8030501@xpirio.com>

i think it was my fault
i'm not 100% sure why - but i works now as expected

thank you very much for your suggestions

br


On 2010-02-23 04:18, Diego Toro wrote:

> hi, I'm using
>
> sendevent NOTIFY
> profile: internal
> event-string: check-sync;reboot=false
> user: 1001
> content-type: application/simple-message-sumary
> profile: internal
> event-string: check-sync;reboot=false
> host: 192.168.7.3
>
>
> where:
> user 1001 is a registered user
> host: IP of FreeSwitch
>
> I see a sip notify message sent to the client.
>
> Diego Toro
> http://lacarretade.blogspot.com/
>
>
> --- On Mon, 2/22/10, Christian L?schenkohl<christian.loeschenkohl at xpirio.com>  wrote:
>
>> From: Christian L?schenkohl<christian.loeschenkohl at xpirio.com>
>> Subject: Re: [Freeswitch-users] sending a sip notify with sendevent
>> To: freeswitch-users at lists.freeswitch.org
>> Date: Monday, February 22, 2010, 5:28 PM
>> hi anthony
>>
>> i did it
>>
>> my profile is actually named nat, "sofia status profile
>> nat" shows me presence_nat as the db name
>> so i had a look in the file presence_nat.db
>>
>> i execute
>> select
>> sip_user,sip_host,contact,profile_name,'application/simple-message-summary','check-sync;reboot=false',''
>> from sip_registrations where sip_user='10' and
>> sip_host='vts.vie.xpirio.net';
>>
>> and it returns
>> 10|vts.vie.xpirio.net|"10"
>> <sip:10 at 81.94.53.5:1032>|nat|application/simple-message-summary|check-sync;reboot=false|
>>
>> looks good to me so far, but as i said no sip notify
>> message is send to the client
>>
>> br
>>
>> Anthony Minessale wrote:
>>
>>> compare that sql stmt to your db manually with the
>> sqlite3 app
>>>
>>> sqlite3
>> /usr/local/freeswitch/db/sofia_reg_internal.db
>>>
>>>
>>> 2010/2/22 Christian L?schenkohl<christian.loeschenkohl at xpirio.com
>>
>>> <mailto:christian.loeschenkohl at xpirio.com>>
>>>
>>>       thank you for this advise
>>>
>>>       i read the section "case
>> SWITCH_EVENT_NOTIFY" carefully, debugged my
>>>       script (i messed something
>>>       up, with telnet the command
>> works - returns Reply-Text: +OK)
>>>
>>>       sql is executed and returns 1
>> row
>>>       select
>>>
>>     sip_user,sip_host,contact,profile_name,'application/simple-message-summary','check-sync;reboot=true',''
>>>       from sip_registrations where
>> sip_user='10' and
>>>       sip_host='vts.vie.xpirio.net
>> <http://vts.vie.xpirio.net>'
>>>
>>>       however no notify message is
>> send to the device
>>>
>>>       i can't use call-id because i
>> simply don't know it
>>>
>>>       br
>>>
>>>
>>>       Diego Toro wrote:
>>>
>>>        >  Read "case
>> SWITCH_EVENT_NOTIFY:" in mod_sofia.c. I suggest you
>>>       assign call-id in the header
>> of the event.
>>>        >
>>>        >
>>>        >  Diego Toro
>>>        >  http://lacarretade.blogspot.com/
>>>        >
>>>        >
>>>        >  --- On Mon, 2/22/10,
>> Christian L?schenkohl
>>>       <christian.loeschenkohl at xpirio.com
>>>       <mailto:christian.loeschenkohl at xpirio.com>>
>> wrote:
>>>        >
>>>        >>  From: Christian
>> L?schenkohl<christian.loeschenkohl at xpirio.com
>>>       <mailto:christian.loeschenkohl at xpirio.com>>
>>>        >>  Subject:
>> [Freeswitch-users] sending a sip notify with sendevent
>>>        >>  To: freeswitch-users at lists.freeswitch.org
>>>       <mailto:freeswitch-users at lists.freeswitch.org>
>>>        >>  Date: Monday, February
>> 22, 2010, 12:16 PM
>>>        >>  hi
>>>        >>
>>>        >>  i try to send a sip
>> notify message to a registered sip
>>>        >>  device
>>>        >>  "sofia profile nat
>> flush_inbound_reg 10 at vts.vie.xpirio.net
>>>       <mailto:10 at vts.vie.xpirio.net>
>>>        >>  reboot" works, but i
>> need
>>>        >>  to send
>> "check-sync;reboot=false" - so the device does a
>>>        >>  resync and don't do a
>> reboot
>>>        >>
>>>        >>  my message looks like
>> this
>>>        >>
>>>        >>  sendevent NOTIFY
>>>        >>  profile: nat
>>>        >>  event-string:
>> check-sync;reboot=false
>>>        >>  user: 10
>>>        >>  host: vts.vie.xpirio.net
>> <http://vts.vie.xpirio.net>
>>>        >>  content-type:
>> application/simple-message-summary
>>>        >>
>>>        >>  if i listen on the
>> loopback interface i do see
>>>        >>
>>>        >>  ##
>>>        >>  T 2010/02/22
>> 18:11:59.083204 127.0.0.1:51840
>>>       <http://127.0.0.1:51840>  ->
>>>        >>  127.0.0.1:8021<http://127.0.0.1:8021>  [AP]
>>>        >>  sendevent NOTIFY
>>>        >>  profile: nat
>>>        >>  event-string:
>> check-sync;reboot=false
>>>        >>  user: 10
>>>        >>  host: vts.vie.xpirio.net
>> <http://vts.vie.xpirio.net>
>>>        >>  content-type:
>> application/simple-message-summary
>>>        >>
>>>        >>  ##
>>>        >>  T 2010/02/22
>> 18:11:59.084032 127.0.0.1:8021
>>>       <http://127.0.0.1:8021>  ->
>>>        >>  127.0.0.1:51840<http://127.0.0.1:51840>  [AP]
>>>        >>  Content-Type:
>> command/reply
>>>        >>  Reply-Text: -ERR invalid
>>>        >>
>>>        >>  --------
>>>        >>  i don't get what it is
>> wrong. i also rechecked the
>>>        >>  registered user in the
>> sqlite database and this
>>>        >>  looks good to me.
>>>        >>
>>>        >>  no message is send to the
>> user.
>>>        >>
>>>        >>  we do use multiple
>> domains, so user could also be
>>>       10 at somedomain.com
>> <mailto:10 at somedomain.com>
>>>        >>  - or am i wrong on this?
>>>        >>  could somebody please
>> bring some light in this.
>>>        >>
>>>        >>  we do use trunk rev.
>> 16631
>>>        >>
>>>        >>  br
>>>        >>
>>>        >>
>>>        >>
>>>        >>  --
>>>        >>  Ing. Christian
>> L?schenkohl
>>>        >>  Technische Leitung,
>> Forschung&  Entwicklung VoIP
>>>        >>
>>>        >>  xpirio
>>>        >>  Telekommunikation&
>> Service GmbH
>>>        >>  Lakeside B04
>>>        >>  9020 Klagenfurt
>>>        >>  Austria
>>>        >>
>>>        >>  T  +43 (0) 5 77 11 -
>> 1000
>>>        >>  F  +43 (0) 5 77 11 -
>> 1002
>>>        >>  E  christian.loeschenkohl at xpirio.com
>>>       <mailto:christian.loeschenkohl at xpirio.com>
>>>        >>
>>>        >>
>> _______________________________________________
>>>        >>  FreeSWITCH-users mailing
>> list
>>>        >>  FreeSWITCH-users at lists.freeswitch.org
>>>       <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>        >>  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>        >>
>>>       UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>        >>  http://www.freeswitch.org
>>>        >>
>>>        >
>>>        >
>>>        >
>>>        >
>>>        >
>> _______________________________________________
>>>        >  FreeSWITCH-users mailing
>> list
>>>        >  FreeSWITCH-users at lists.freeswitch.org
>>>       <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>        >  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>        >
>>>       UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>        >  http://www.freeswitch.org
>>>
>>>
>>>       --
>>>       Ing. Christian L?schenkohl
>>>       Technische Leitung, Forschung
>> &  Entwicklung VoIP
>>>
>>>       xpirio
>>>       Telekommunikation&
>> Service GmbH
>>>       Lakeside B04
>>>       9020 Klagenfurt
>>>       Austria
>>>
>>>       T  +43 (0) 5 77 11 -
>> 1000
>>>       F  +43 (0) 5 77 11 -
>> 1002
>>>       E  christian.loeschenkohl at xpirio.com
>>>       <mailto:christian.loeschenkohl at xpirio.com>
>>>
>>>
>>     _______________________________________________
>>>       FreeSWITCH-users mailing list
>>>       FreeSWITCH-users at lists.freeswitch.org
>>>       <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>       http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>       UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>       http://www.freeswitch.org
>>>
>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com
>>
>>> <mailto:MSN%3Aanthony_minessale at hotmail.com>
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>
>>> <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
>>> IRC: irc.freenode.net<http://irc.freenode.net>  #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org
>>
>>> <mailto:sip%3A888 at conference.freeswitch.org>
>>> iax:guest at conference.freeswitch.org/888
>>
>>> <http://iax:guest at conference.freeswitch.org/888>
>>> googletalk:conf+888 at conference.freeswitch.org
>>
>>> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> pstn:+19193869900
>>>
>>>
>>>
>> ------------------------------------------------------------------------
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> --
>> Ing. Christian L?schenkohl
>> Technische Leitung, Forschung&  Entwicklung VoIP
>>
>> xpirio
>> Telekommunikation&  Service GmbH
>> Lakeside B04
>> 9020 Klagenfurt
>> Austria
>>
>> T  +43 (0) 5 77 11 - 1000
>> F  +43 (0) 5 77 11 - 1002
>> E  christian.loeschenkohl at xpirio.com
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


-- 
Ing. Christian L?schenkohl
Technische Leitung, Forschung & Entwicklung VoIP

xpirio
Telekommunikation & Service GmbH
Lakeside B04
9020 Klagenfurt
Austria

T  +43 (0) 5 77 11 - 1000
F  +43 (0) 5 77 11 - 1002
E  christian.loeschenkohl at xpirio.com



From christian.loeschenkohl at xpirio.com  Tue Feb 23 07:12:00 2010
From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=)
Date: Tue, 23 Feb 2010 16:12:00 +0100
Subject: [Freeswitch-users] big thanks to all freeswitch developers and
	contributing users
Message-ID: <4B83F040.7040005@xpirio.com>

i want to say a big THANKY YOU to all contributing freeswitch community members.

over one year has passed since i did fall in love with this project.
it is getting better every day, one get's help and advices if needed.
the admins do care about nearly every problem - no matter if it's big or small.
i also did manage an opensource project and i wish i had done it with that much
heart and intense power that i see here.

i also hope that i can contribute back enough (questions, bug reports, wiki enhancements).

wishing you all the best
br

-- 
Ing. Christian L?schenkohl
Technische Leitung, Forschung & Entwicklung VoIP

xpirio
Telekommunikation & Service GmbH
Lakeside B04
9020 Klagenfurt
Austria

T  +43 (0) 5 77 11 - 1000
F  +43 (0) 5 77 11 - 1002
E  christian.loeschenkohl at xpirio.com



From technical at ttnc.co.uk  Tue Feb 23 07:31:55 2010
From: technical at ttnc.co.uk (TTNC - Technical)
Date: Tue, 23 Feb 2010 15:31:55 +0000
Subject: [Freeswitch-users] leg_timeout with ignore_early_media false
In-Reply-To: <F86E6E00-B774-487F-9F62-2B1BB00382FC@ttnc.co.uk>
References: <F86E6E00-B774-487F-9F62-2B1BB00382FC@ttnc.co.uk>
Message-ID: <F25F531B-C93C-4293-8400-BB82B3B3A2FC@ttnc.co.uk>

I've opened a Jira for this issue as I believe it's a bug, there's an example LUA script attached to the bug to replicate the issue.

http://jira.freeswitch.org/browse/FSCORE-556

Russ

On 23 Feb 2010, at 12:21, TTNC - Technical wrote:

> Hi Guys
> 
> I'm trying to create a hunt group where ignore_early_media = false is set, so that the international ring tone is passed through to the caller. Setting ignore_early_media = false on the channel does what I want, but with this set leg_timeout is not honoured.
> 
> I've switched to use bridge_answer_timeout which is honoured if ignore_early_media = false and the call progresses through the different legs, but bridge_answer_timeout times out the call after the set period, even if the call has been successfully answered and bridged. I've tried all different combinations of group_confirm_cancel_timeout [1|2|3], none of them seem to affect bridge_answer_timeout.
> 
> Does anyone have a solution for timing out legs of a hunt group with ignore_early_media = false set?
> 
> This is my dial string:
> 
> '{caller-id-in-from=true,origination_caller_id_name=012345123123,origination_caller_id_number= 012345123123}[bridge_answer_timeout=20]sofia/internal/4412345123123 at sipipgw.siphost.net |sofia/internal/4412345123123 at sipipgw.siphost.net'
> 
> And in my lua application, I'm setting the following:
> 
> session:setVariable("group_confirm_cancel_timeout", "1"); -- substitute 1 for 1, 2 or 3, none work.
> session:setVariable("ignore_early_media", "false");
> 
> I'm using the latest trunk revision - it's still happening.
> 
> Any suggestions welcome.
> 
> Thanks
> 
> Russ
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From infos at madovsky.org  Tue Feb 23 08:51:51 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 23 Feb 2010 11:51:51 -0500
Subject: [Freeswitch-users] FScomm
References: <157FDB96FB88476FBAEC69C1309A0A5F@MOBILEE1705>
	<COL117-W5563292C886257AFC56548B9420@phx.gbl>
Message-ID: <D64783281177440B9094B8181E55EFA7@MOBILEE1705>


  ----- Original Message ----- 
  From: Jeff Lenk 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Tuesday, February 23, 2010 9:23 AM
  Subject: Re: [Freeswitch-users] FScomm


  What platform are you trying to build?
   

------------------------------------------------------------------------------
  From: infos at madovsky.org
  To: freeswitch-users at lists.freeswitch.org
  Date: Tue, 23 Feb 2010 01:41:04 -0500
  Subject: [Freeswitch-users] FScomm


  Hi,

  is http://wiki.freeswitch.org/wiki/FSComm
  available yet ? because QT framework link has broken
  and some instruction don't work with last svn trunk (for example to compile FScomm in FS svn root)

  Thanks

  Franck


------------------------------------------------------------------------------
  Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. Sign up now. 


------------------------------------------------------------------------------


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  http://www.freeswitch.org


  FSComm on Linux fedora 10 64 bits
  It says FSComm can be built inside FS svn folder typing gmake make
  but there is no Makefile inside

  Thanks

  Franck
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100223/22489c2b/attachment-0002.html 

From anthony.minessale at gmail.com  Tue Feb 23 09:44:03 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 23 Feb 2010 11:44:03 -0600
Subject: [Freeswitch-users] leg_timeout with ignore_early_media false
In-Reply-To: <F25F531B-C93C-4293-8400-BB82B3B3A2FC@ttnc.co.uk>
References: <F86E6E00-B774-487F-9F62-2B1BB00382FC@ttnc.co.uk>
	<F25F531B-C93C-4293-8400-BB82B3B3A2FC@ttnc.co.uk>
Message-ID: <191c3a031002230944h29741d21p58db0f9bfff57787@mail.gmail.com>

Bug is fixed and there is a note on your other bug awaiting your response.


On Tue, Feb 23, 2010 at 9:31 AM, TTNC - Technical <technical at ttnc.co.uk>wrote:

> I've opened a Jira for this issue as I believe it's a bug, there's an
> example LUA script attached to the bug to replicate the issue.
>
> http://jira.freeswitch.org/browse/FSCORE-556
>
> Russ
>
> On 23 Feb 2010, at 12:21, TTNC - Technical wrote:
>
> > Hi Guys
> >
> > I'm trying to create a hunt group where ignore_early_media = false is
> set, so that the international ring tone is passed through to the caller.
> Setting ignore_early_media = false on the channel does what I want, but with
> this set leg_timeout is not honoured.
> >
> > I've switched to use bridge_answer_timeout which is honoured if
> ignore_early_media = false and the call progresses through the different
> legs, but bridge_answer_timeout times out the call after the set period,
> even if the call has been successfully answered and bridged. I've tried all
> different combinations of group_confirm_cancel_timeout [1|2|3], none of them
> seem to affect bridge_answer_timeout.
> >
> > Does anyone have a solution for timing out legs of a hunt group with
> ignore_early_media = false set?
> >
> > This is my dial string:
> >
> >
> '{caller-id-in-from=true,origination_caller_id_name=012345123123,origination_caller_id_number=
> 012345123123}[bridge_answer_timeout=20]sofia/internal/
> 4412345123123 at sipipgw.siphost.net |sofia/internal/
> 4412345123123 at sipipgw.siphost.net'
> >
> > And in my lua application, I'm setting the following:
> >
> > session:setVariable("group_confirm_cancel_timeout", "1"); -- substitute 1
> for 1, 2 or 3, none work.
> > session:setVariable("ignore_early_media", "false");
> >
> > I'm using the latest trunk revision - it's still happening.
> >
> > Any suggestions welcome.
> >
> > Thanks
> >
> > Russ
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100223/7fd8c5b5/attachment-0002.html 

From robert.hadley at teotech.com  Tue Feb 23 09:48:54 2010
From: robert.hadley at teotech.com (Robert Hadley)
Date: Tue, 23 Feb 2010 09:48:54 -0800
Subject: [Freeswitch-users] Possible typo on Openzap.conf wiki page
	specifying fxs/fxo-channels
Message-ID: <2C33D9D6C1444AAC80950CDF8D38A428@greyhawk.tonecommander.com>


On the http://wiki.freeswitch.org/wiki/Openzap.conf_Examples wiki page, is
there a typo in the wanpipe /usr/local/freeswitch/conf/openzap.conf example
concerning specifying the fxo-channel vs. fxs-channel?


In the [span wanpipe FXS] section the channels are shown on wiki page as
fxo-channels => 1:1 and 1:2


In the [span wanpipe FXO] section the channels are shown on wiki page as
fxs-channels => 1:3 and 1:4


Are specifying the fxs-channels and fxo-channels shown in the wrong
sections?  I had to specify fxs-channels in the FXS span and fxo-channels in
the FXO span to get it working on my hardware.


Except from wiki page:


Sangoma A200/A400


*	A200, A200D, A400, A400D series and variants 

The configuration depends on whether wanpipe is configured to use Zaptel TDM
Voice, or the Sangoma standalone TDM Voice API. This is determined in the
installation and configuration of the Sangoma wanpipe software. 

If wanpipe is using Zaptel, you need to configure openzap.conf with [span
zt] entries. For example, if ports 1 and 2 are FXS (e.g. configured to
accept analog phone connections), and ports 3 and 4 are FXO (e.g. configured
to accept PSTN analog lines) you will need: 

/usr/local/freeswitch/conf/openzap.conf: 

[span zt FXO]
name => OpenZAP
number => 3001
fxo-channel => 1
number => 3002
fxo-channel => 2
 
[span zt FXS]
name => OpenZAP
number => 4165551111
fxs-channel => 3
number => 4165552222
fxs-channel => 4

If wanpipe is standalone, you need to configure openzap.conf with [span
wanpipe] entries. For the same example as above this would be: 

/usr/local/freeswitch/conf/openzap.conf: 

[span wanpipe FXS]
# This is the value of the callerid_name variable that is raised in the
dialplan
name => Analog Phone 1
# This is the value of the callerid_number variable that is raised in the
dialplan
number => 3001
fxo-channel => 1:1
name => Analog Phone 2
number => 3002
fxo-channel => 1:2
 
[span wanpipe FXO]
# the chan_name variable will raised as "OpenZAP/2:1/4165551111" in dialplan
when an incoming call arrives on this port
name => PSTN line 1
number => 4165551111
fxs-channel => 1:3
name => PSTN line 2
number => 4165552222
fxs-channel => 1:4

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100223/a54600fc/attachment-0002.html 

From brian at freeswitch.org  Tue Feb 23 09:53:46 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 23 Feb 2010 11:53:46 -0600
Subject: [Freeswitch-users] Possible typo on Openzap.conf wiki page
	specifying fxs/fxo-channels
In-Reply-To: <2C33D9D6C1444AAC80950CDF8D38A428@greyhawk.tonecommander.com>
References: <2C33D9D6C1444AAC80950CDF8D38A428@greyhawk.tonecommander.com>
Message-ID: <71D261B7-ECBA-4582-8BBA-CC34258970D7@freeswitch.org>

You can edit the examples on the wiki and it should be good.

/b

On Feb 23, 2010, at 11:48 AM, Robert Hadley wrote:

> On the http://wiki.freeswitch.org/wiki/Openzap.conf_Examples wiki page, is there a typo in the wanpipe/usr/local/freeswitch/conf/openzap.conf example concerning specifying the fxo-channel vs. fxs-channel?
> 
> In the [span wanpipe FXS] section the channels are shown on wiki page as fxo-channels => 1:1 and 1:2
> 
> In the [span wanpipe FXO] section the channels are shown on wiki page as fxs-channels => 1:3 and 1:4
> 
> Are specifying the fxs-channels and fxo-channels shown in the wrong sections?  I had to specify fxs-channels in the FXS span and fxo-channels in the FXO span to get it working on my hardware.
> 
> Except from wiki page:
> Sangoma A200/A400
> A200, A200D, A400, A400D series and variants
> The configuration depends on whether wanpipe is configured to use Zaptel TDM Voice, or the Sangoma standalone TDM Voice API. This is determined in the installation and configuration of the Sangoma wanpipe software.
> 
> If wanpipe is using Zaptel, you need to configure openzap.conf with [span zt] entries. For example, if ports 1 and 2 are FXS (e.g. configured to accept analog phone connections), and ports 3 and 4 are FXO (e.g. configured to accept PSTN analog lines) you will need:
> 
> /usr/local/freeswitch/conf/openzap.conf:
> 
> [span zt FXO]
> name => OpenZAP
> number => 3001
> fxo-channel => 1
> number => 3002
> fxo-channel => 2
>  
> [span zt FXS]
> name => OpenZAP
> number => 4165551111
> fxs-channel => 3
> number => 4165552222
> fxs-channel => 4
> If wanpipe is standalone, you need to configure openzap.conf with [span wanpipe] entries. For the same example as above this would be:
> 
> /usr/local/freeswitch/conf/openzap.conf:
> 
> [span wanpipe FXS]
> # This is the value of the callerid_name variable that is raised in the dialplan
> name => Analog Phone 1
> # This is the value of the callerid_number variable that is raised in the dialplan
> number => 3001
> fxo-channel => 1:1
> name => Analog Phone 2
> number => 3002
> fxo-channel => 1:2
>  
> [span wanpipe FXO]
> # the chan_name variable will raised as "OpenZAP/2:1/4165551111" in dialplan when an incoming call arrives on this port
> name => PSTN line 1
> number => 4165551111
> fxs-channel => 1:3
> name => PSTN line 2
> number => 4165552222
> fxs-channel => 1:4
>  
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100223/891675db/attachment-0002.html 

From srinivas.ksvreddy at gmail.com  Mon Feb 22 22:02:11 2010
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Tue, 23 Feb 2010 11:32:11 +0530
Subject: [Freeswitch-users] Freeswitch to another Freeswitch(or gateway)
Message-ID: <f8af5741002222202p3457e8efl2e683082dc597b47@mail.gmail.com>

Hi,
i want divert calls from my sipserver to another sipserver or third party
gateway, is there any way to achive this.

Regards
Srinivasula Reddy K
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100223/1660851f/attachment-0002.html 

From srinivas.ksvreddy at gmail.com  Tue Feb 23 06:00:07 2010
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Tue, 23 Feb 2010 19:30:07 +0530
Subject: [Freeswitch-users] Fwd: Freeswitch to another Freeswitch(or gateway)
In-Reply-To: <f8af5741002222202p3457e8efl2e683082dc597b47@mail.gmail.com>
References: <f8af5741002222202p3457e8efl2e683082dc597b47@mail.gmail.com>
Message-ID: <f8af5741002230600h6911e676iae6fc48375b31b1@mail.gmail.com>

 Hi,
i want divert calls from my sipserver to another sipserver or third party
gateway based on the host name, is there any way to achive this.

Regards
Srinivasula Reddy K



-- 
Srinivasula Reddy K
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100223/e25ca1ae/attachment-0002.html 

From phunk0000 at hotmail.com  Tue Feb 23 07:42:28 2010
From: phunk0000 at hotmail.com (Meg Stroodle)
Date: Tue, 23 Feb 2010 10:42:28 -0500
Subject: [Freeswitch-users] mod_nibblebill
Message-ID: <BLU110-W28EA8BD06A601D5D85268BA7420@phx.gbl>




Hello List!  I am trying to install mod_nibblebill on
my FS installation.  I get the following log entry form FS &
nibblebill, but the database table I setup remains unchanged.  Any help in
this matter would be greatly appreciated.  Following is an excerpt from
the FS log:

 

2010-02-23 10:32:52.869149 [DEBUG] switch_rtp.c:2004 Correct
ip/port confirmed.

2010-02-23 10:32:53.077162 [DEBUG] sofia.c:3727 Channel sofia/internal/3007 at 192.168.15.177
entering state [ready][200]

2010-02-23 10:32:53.149166 [DEBUG] switch_core_session.c:706
Send signal sofia/internal/sip:3008 at 192.168.15.176:21828 [BREAK]

2010-02-23 10:32:53.149166 [DEBUG] switch_core_session.c:706
Send signal sofia/internal/3007 at 192.168.15.177
[BREAK]

2010-02-23 10:32:53.149166 [DEBUG] switch_ivr_bridge.c:131
sofia/internal/sip:3008 at 192.168.15.176:21828 receive message [DISPLAY]

2010-02-23 10:32:53.169168 [DEBUG] switch_ivr_bridge.c:131 sofia/internal/3007 at 192.168.15.177
receive message [DISPLAY]

2010-02-23 10:33:21.388949 [DEBUG] mod_nibblebill.c:529
Received request via SESSION_HEARTBEAT!

2010-02-23 10:33:21.388949 [DEBUG] mod_nibblebill.c:397
Attempting to bill at $1.0 per minute to account 3007

2010-02-23 10:33:21.388949 [INFO] mod_nibblebill.c:447
Beginning new billing on 3ec17ffc-c94e-46ff-bb0d-7ad0ef27bdab

2010-02-23 10:33:21.388949 [DEBUG] mod_nibblebill.c:455 28
seconds passed since last bill time of 2010-02-23 10:32:52

2010-02-23 10:33:21.388949 [DEBUG] mod_nibblebill.c:461
Billing $0.475997 to 3007 (Call: 3ec17ffc-c94e-46ff-bb0d-7ad0ef27bdab /
0.000000 so far)

2010-02-23 10:33:51.508822 [DEBUG] mod_nibblebill.c:529
Received request via SESSION_HEARTBEAT!

2010-02-23 10:33:51.508822 [DEBUG] mod_nibblebill.c:397 Attempting
to bill at $1.0 per minute to account 3007

2010-02-23 10:33:51.508822 [DEBUG] mod_nibblebill.c:449 Last
successful billing time was 

2010-02-23 10:33:51.508822 [DEBUG] mod_nibblebill.c:455 30
seconds passed since last bill time of 2010-02-23 10:33:21

2010-02-23 10:33:51.508822 [DEBUG] mod_nibblebill.c:461
Billing $0.501998 to 3007 (Call: 3ec17ffc-c94e-46ff-bb0d-7ad0ef27bdab /
0.475997 so far)

2010-02-23 10:34:21.649708 [DEBUG] mod_nibblebill.c:529
Received request via SESSION_HEARTBEAT!

2010-02-23 10:34:21.649708 [DEBUG] mod_nibblebill.c:397
Attempting to bill at $1.0 per minute to account 3007

2010-02-23 10:34:21.649708 [DEBUG] mod_nibblebill.c:449 Last
successful billing time was 

2010-02-23 10:34:21.649708 [DEBUG] mod_nibblebill.c:455 30 seconds
passed since last bill time of 2010-02-23 10:33:51

2010-02-23 10:34:21.649708 [DEBUG] mod_nibblebill.c:461
Billing $0.502348 to 3007 (Call: 3ec17ffc-c94e-46ff-bb0d-7ad0ef27bdab /
0.977995 so far)

2010-02-23 10:34:26.254308 [NOTICE] sofia.c:329 Hangup sofia/internal/3007 at 192.168.15.177
[CS_EXECUTE] [NORMAL_CLEARING]

2010-02-23 10:34:26.254308 [DEBUG] switch_channel.c:1912
Send signal sofia/internal/3007 at 192.168.15.177
[KILL]

2010-02-23 10:34:26.254308 [DEBUG] switch_core_session.c:999
Send signal sofia/internal/3007 at 192.168.15.177
[BREAK]

2010-02-23 10:34:26.254308 [DEBUG]
switch_core_state_machine.c:459 sofia/internal/3007 at 192.168.15.177
thread mismatch skipping state handler.

2010-02-23 10:34:26.269008 [DEBUG] switch_ivr_bridge.c:470 sofia/internal/3007 at 192.168.15.177
ending bridge by request from read function

2010-02-23 10:34:26.269008 [DEBUG] switch_ivr_bridge.c:563
BRIDGE THREAD DONE [sofia/internal/3007 at 192.168.15.177]

2010-02-23 10:34:26.269008 [DEBUG] switch_ivr_bridge.c:565
Send signal sofia/internal/sip:3008 at 192.168.15.176:21828 [BREAK]

2010-02-23 10:34:26.269008 [DEBUG] switch_ivr_bridge.c:520
sofia/internal/sip:3008 at 192.168.15.176:21828 receive message [UNBRIDGE]

2010-02-23 10:34:26.269008 [DEBUG] switch_core_session.c:645
Send signal sofia/internal/sip:3008 at 192.168.15.176:21828 [BREAK]

2010-02-23 10:34:26.269008 [DEBUG] switch_ivr_bridge.c:563
BRIDGE THREAD DONE [sofia/internal/sip:3008 at 192.168.15.176:21828]

2010-02-23 10:34:26.269008 [DEBUG] switch_ivr_bridge.c:565
Send signal sofia/internal/3007 at 192.168.15.177
[BREAK]

2010-02-23 10:34:26.269008 [NOTICE] switch_ivr_bridge.c:617
Hangup sofia/internal/sip:3008 at 192.168.15.176:21828 [CS_EXCHANGE_MEDIA]
[NORMAL_CLEARING]

2010-02-23 10:34:26.269008 [DEBUG] switch_channel.c:1912
Send signal sofia/internal/sip:3008 at 192.168.15.176:21828 [KILL]

2010-02-23 10:34:26.269008 [DEBUG] switch_core_session.c:999
Send signal sofia/internal/sip:3008 at 192.168.15.176:21828 [BREAK]

2010-02-23 10:34:26.269008 [DEBUG]
switch_core_state_machine.c:488 (sofia/internal/sip:3008 at 192.168.15.176:21828)
State HANGUP

2010-02-23 10:34:26.269008 [DEBUG] mod_sofia.c:358 Channel
sofia/internal/sip:3008 at 192.168.15.176:21828 hanging up, cause: NORMAL_CLEARING

2010-02-23 10:34:26.269008 [DEBUG] mod_sofia.c:400 Sending
BYE to sofia/internal/sip:3008 at 192.168.15.176:21828

2010-02-23 10:34:26.269008 [DEBUG] mod_nibblebill.c:397
Attempting to bill at $2.30 per minute to account 3008

2010-02-23 10:34:26.269008 [INFO] mod_nibblebill.c:447
Beginning new billing on c1c42712-89e9-44c5-914f-11ed8f6aebb1

2010-02-23 10:34:26.269008 [DEBUG] mod_nibblebill.c:455 93
seconds passed since last bill time of 2010-02-23 10:32:52

2010-02-23 10:34:26.269008 [DEBUG] mod_nibblebill.c:461
Billing $3.582436 to 3008 (Call: c1c42712-89e9-44c5-914f-11ed8f6aebb1 / 0.000000
so far)

2010-02-23 10:34:26.269008 [DEBUG]
switch_core_state_machine.c:46 sofia/internal/sip:3008 at 192.168.15.176:21828
Standard HANGUP, cause: NORMAL_CLEARING

2010-02-23 10:34:26.269008 [DEBUG]
switch_core_state_machine.c:488 (sofia/internal/sip:3008 at 192.168.15.176:21828)
State HANGUP going to sleep

2010-02-23 10:34:26.269008 [INFO] switch_core_session.c:1108
sofia/internal/sip:3008 at 192.168.15.176:21828 setting session heartbeat to 30
second(s).

2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:351
(sofia/internal/sip:3008 at 192.168.15.176:21828) State EXCHANGE_MEDIA going to
sleep

2010-02-23 10:34:26.269008 [DEBUG]
switch_core_state_machine.c:314 (sofia/internal/sip:3008 at 192.168.15.176:21828)
Running State Change CS_HANGUP

2010-02-23 10:34:26.269008 [DEBUG]
switch_core_state_machine.c:465 sofia/internal/sip:3008 at 192.168.15.176:21828
handler already called, skipping state handler.

2010-02-23 10:34:26.269008 [DEBUG]
switch_core_state_machine.c:333 (sofia/internal/sip:3008 at 192.168.15.176:21828)
State Change CS_HANGUP -> CS_REPORTING

2010-02-23 10:34:26.269008 [DEBUG] switch_core_session.c:999
Send signal sofia/internal/sip:3008 at 192.168.15.176:21828 [BREAK]

2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/sip:3008 at 192.168.15.176:21828) Running State Change
CS_REPORTING

2010-02-23 10:34:26.269008 [DEBUG]
switch_core_state_machine.c:579 (sofia/internal/sip:3008 at 192.168.15.176:21828)
State REPORTING

2010-02-23 10:34:26.269008 [DEBUG]
switch_core_state_machine.c:53 sofia/internal/sip:3008 at 192.168.15.176:21828
Standard REPORTING, cause: NORMAL_CLEARING

2010-02-23 10:34:26.269008 [DEBUG]
switch_core_state_machine.c:579 (sofia/internal/sip:3008 at 192.168.15.176:21828)
State REPORTING going to sleep

2010-02-23 10:34:26.269008 [DEBUG]
switch_core_state_machine.c:327 (sofia/internal/sip:3008 at 192.168.15.176:21828)
State Change CS_REPORTING -> CS_DESTROY

2010-02-23 10:34:26.269008 [DEBUG] switch_core_session.c:999
Send signal sofia/internal/sip:3008 at 192.168.15.176:21828 [BREAK]

2010-02-23 10:34:26.269008 [DEBUG]
switch_core_session.c:1136 Session 6
(sofia/internal/sip:3008 at 192.168.15.176:21828) Locked, Waiting on external
entities

2010-02-23 10:34:26.272437 [DEBUG] switch_core_state_machine.c:348
(sofia/internal/3007 at 192.168.15.177)
State EXECUTE going to sleep

2010-02-23 10:34:26.272437 [DEBUG]
switch_core_state_machine.c:314 (sofia/internal/3007 at 192.168.15.177)
Running State Change CS_HANGUP

2010-02-23 10:34:26.272437 [DEBUG] switch_core_state_machine.c:488
(sofia/internal/3007 at 192.168.15.177)
State HANGUP

2010-02-23 10:34:26.272437 [DEBUG] mod_sofia.c:358 Channel sofia/internal/3007 at 192.168.15.177
hanging up, cause: NORMAL_CLEARING

2010-02-23 10:34:26.272437 [DEBUG] mod_nibblebill.c:397
Attempting to bill at $1.0 per minute to account 3007

2010-02-23 10:34:26.272437 [DEBUG] mod_nibblebill.c:449 Last
successful billing time was 

2010-02-23 10:34:26.272437 [DEBUG] mod_nibblebill.c:455 4
seconds passed since last bill time of 2010-02-23 10:34:21

2010-02-23 10:34:26.272437 [DEBUG] mod_nibblebill.c:461
Billing $0.077045 to 3007 (Call: 3ec17ffc-c94e-46ff-bb0d-7ad0ef27bdab /
1.480343 so far)

2010-02-23 10:34:26.272437 [DEBUG]
switch_core_state_machine.c:46 sofia/internal/3007 at 192.168.15.177
Standard HANGUP, cause: NORMAL_CLEARING

 

Anyhelp getting nibblebill to connect to the database would
be greatly appreciated.  Thanks

 

 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
http://clk.atdmt.com/GBL/go/201469226/direct/01/
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100223/12a2dc15/attachment-0002.html 

From infos at madovsky.org  Tue Feb 23 10:09:55 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 23 Feb 2010 13:09:55 -0500
Subject: [Freeswitch-users] RTP timeout
Message-ID: <A68022CC2AFE40E29FE73385397A665B@MOBILEE1705>

Hi,

thanks to answer me if I misunderstood something,
but if I run a softphone on the same IP as FS, there
is an RTP timeout.

Any idea ?

Thanks

Franck
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100223/de7f5a9c/attachment-0002.html 

From anthony.minessale at gmail.com  Tue Feb 23 11:13:45 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 23 Feb 2010 13:13:45 -0600
Subject: [Freeswitch-users] Question about sofia_contact
In-Reply-To: <4B83A2E7.1060905@ewetel.de>
References: <4B7EA954.30402@ewetel.de> <4B7EADBC.1040001@ewetel.de>
	<4B8249BF.3090708@ewetel.de>
	<191c3a031002220842j5bec442an5f1ea89cb0e8a6ff@mail.gmail.com>
	<4B83A2E7.1060905@ewetel.de>
Message-ID: <191c3a031002231113i5e838cf8g6354977d3b361e09@mail.gmail.com>

I added a patch that I think will allow what you want by being more strict
in the code about deciding if a string was meant to be a domain or profile
name.

On Tue, Feb 23, 2010 at 3:41 AM, Helmut Kuper <helmut.kuper at ewetel.de>wrote:

> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> Hello Anthony,
>
>
> you are right, I'm quite unpatient, sorry 4 that. Your solution works
> fine. I thought the sip domain could be any string and must not be a
> valid domain format.
>
> Thanks to you, board and community for this fantastic project!
>
> regards from rainy germany
> Helmut
>
>
>
> On 22.02.2010 17:42, Anthony Minessale wrote:
> > it's mad at you for asking twice before waiting for a reply, so it's not
> > working on purpose.
> >
> > Actually it's mad at you because your domain does not contain a . so it
> > is assuming you are specifying a profile name as the domain.  if your
> > domain was mydomain.com <http://mydomain.com> instead it would work.
> -----BEGIN PGP SIGNATURE-----
> Version: GnuPG v1.4.7 (MingW32)
>
> iD8DBQFLg6Ln4tZeNddg3dwRAvj6AJ9ruybNpbL8mdUlx1jVtLPYVbCSDACfQJLo
> zfieJnHZdp2Xv3OS6HTZE/k=
> =ESgY
> -----END PGP SIGNATURE-----
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100223/6c487fbd/attachment-0002.html 

From anthony.minessale at gmail.com  Tue Feb 23 11:31:49 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 23 Feb 2010 13:31:49 -0600
Subject: [Freeswitch-users] Dingaling - external rtp supported?
In-Reply-To: <29b888f81002222359w6ae4cd8ej944d453a5b099af4@mail.gmail.com>
References: <29b888f81002160835y6e290319ub0d04c088caa1b79@mail.gmail.com>
	<DEA39AD1-0EE9-473A-B78C-CDDF3F100578@freeswitch.org>
	<29b888f81002170318p63622f3fhc8d26ad0e85602d2@mail.gmail.com>
	<191c3a031002170538r3b029515i8a000812a70ad558@mail.gmail.com>
	<29b888f81002170741y3d13cb57ycb4436a372c4eded@mail.gmail.com>
	<191c3a031002171136q7424e9bai34379abb542daf93@mail.gmail.com>
	<29b888f81002210800v617bca9dk4bf1abf1bd791ee0@mail.gmail.com>
	<29b888f81002222359w6ae4cd8ej944d453a5b099af4@mail.gmail.com>
Message-ID: <191c3a031002231131x19c9791cgd986a6c1be18e09a@mail.gmail.com>

If you are modifying your build to add libgcrypt / libgnutls to win32, you
have chosen an incompatible version of one of these libs.  We do not support
manually adding this modification to the code, you will need to find someone

else who has done it successfully to help you.




On Tue, Feb 23, 2010 at 1:59 AM, Moiz Chinoy <moizchinoy at gmail.com> wrote:

> Moreover, if I gtalk client is on the same machine as FS and i have
> following settings, FS crashes with the same mutex error.
>
> External Sip Profile has following lines:
> ---------------------------------------------------------
>   <param name="rtp-ip" value="$${local_ip_v4}"/>
>   <param name="sip-ip" value="$${local_ip_v4}"/>
>    <param name="ext-rtp-ip" value="auto-nat"/>
>    <param name="ext-sip-ip" value="auto-nat"/>
>
> Jingle Client.xml has following lines:
> -----------------------------------------------------
>   <param name="rtp-ip" value="$${bind_server_ip}"/>
>   <!--<param name="ext-rtp-ip" value="$${external_rtp_ip}"/>-->
>   <!--<param name="disable-rtp-auto-adjust" value="true"/>-->
>   <!--<param name="avatar" value="/path/to/tiny.jpg"/>-->
>   <!--<param name="candidate-acl" value="wan.auto"/>-->
>   <!--<param name="local-network-acl" value="localnet.auto"/>-->
>
> If I uncomment the following line in client.xml (Jingle profile)
>    <!--<param name="local-network-acl" value="localnet.auto"/>-->
> then exception does not happen.
>
> Is this a known issue or do I need to post it in JIRA?
>
> Tell me if more logs are needed...
>
>
> On Sun, Feb 21, 2010 at 8:00 PM, Moiz Chinoy <moizchinoy at gmail.com> wrote:
> > Guys,
> >
> > To make things simple gtalk client is entirely on different network.
> >
> > Call comes from outside through external Sip profile.
> >
> > If gtalk answers the call after 3-4 rings both parties can hear each
> other.
> > If gtalk answers the call after 2 rings both parties no one can hear each
> other.
> > If gtalk answers the call immediately FS crashes.
> >
> > Attached is the screen shot of the error...
> >
> > Here is the FS log...
> > --------------------------------
> > http://pastebin.freeswitch.org/12197
> >
> > External Sip Profile has following lines:
> > ---------------------------------------------------------
> >    <param name="rtp-ip" value="$${local_ip_v4}"/>
> >    <param name="sip-ip" value="$${local_ip_v4}"/>
> >    <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
> >    <param name="ext-sip-ip" value="$${external_sip_ip}"/>
> >
> > Jingle Client.xml has following lines:
> > -----------------------------------------------------
> >    <param name="rtp-ip" value="$${bind_server_ip}"/>
> >    <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
> >    <param name="disable-rtp-auto-adjust" value="true"/>
> >    <!--<param name="avatar" value="/path/to/tiny.jpg"/>-->
> >    <!--<param name="candidate-acl" value="wan.auto"/>-->
> >    <!--<param name="local-network-acl" value="localnet.auto"/>-->
> >
> > Vars.xml has following lines:
> > -------------------------------------------
> > <X-PRE-PROCESS cmd="set" data="external_rtp_ip=stun:stun.freeswitch.org
> "/>
> > <X-PRE-PROCESS cmd="set" data="external_sip_ip=stun:stun.freeswitch.org
> "/>
> >
> >
> > Please advise me how can I provide more of the required data.
> >
> > On Wed, Feb 17, 2010 at 11:36 PM, Anthony Minessale
> > <anthony.minessale at gmail.com> wrote:
> >> you cant combine stun and gtalk and boxes in the same lan very easily if
> you
> >> do need to do that you will need to mess with
> >>
> >> <param name="candidate-acl" value="wan.auto"/>
> >> <param name="local-network-acl" value="localnet.auto"/>
> >>
> >>
> >>
> >>
> >>
> >> On Wed, Feb 17, 2010 at 9:41 AM, Moiz Chinoy <moizchinoy at gmail.com>
> wrote:
> >>>
> >>> Guys I am unable to produce the crash but now both parties cannot hear
> >>> each other!
> >>>
> >>> Vars.xml has following lines:
> >>>  <X-PRE-PROCESS cmd="set"
> >>> data="external_rtp_ip=stun:stun.freeswitch.org"/>
> >>>  <X-PRE-PROCESS cmd="set"
> >>> data="external_sip_ip=stun:stun.freeswitch.org"/>
> >>>
> >>> Jingle Client.xml has following lines:
> >>>    <param name="rtp-ip" value="$${bind_server_ip}"/>
> >>>    <!-- <param name="ext-rtp-ip" value="auto-nat"/> -->
> >>>    <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
> >>>    <param name="disable-rtp-auto-adjust" value="true"/>
> >>>    <param name="local-network-acl" value="localnet.auto"/>
> >>>
> >>> On Wed, Feb 17, 2010 at 5:38 PM, Anthony Minessale
> >>> <anthony.minessale at gmail.com> wrote:
> >>> > Obtain a stack trace from the crash.
> >>> >
> >>> > On Feb 17, 2010 5:26 AM, "Moiz Chinoy" <moizchinoy at gmail.com> wrote:
> >>> >
> >>> > Hi,
> >>> >
> >>> > FS rev: 16673
> >>> > Platform: Windows
> >>> >
> >>> > More details:
> >>> >
> >>> > FS is behind NAT and machine is running a VPN connection.
> >>> >
> >>> > FS and GTalk client on the same machine:
> >>> >
> >>> >
> --------------------------------------------------------------------------------------------------
> >>> > jingle profile client.xml has following line:
> >>> > <param name="rtp-ip" value="$${bind_server_ip}"/>
> >>> >
> >>> > External SIP call is successfully bridged to GTalk client.
> >>> >
> >>> >
> >>> > FS and GTalk client on the different machine:
> >>> >
> >>> >
> --------------------------------------------------------------------------------------------------
> >>> > jingle profile client.xml has following lines:
> >>> > <param name="rtp-ip" value="$${bind_server_ip}"/>
> >>> >
> >>> > <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
> >>> >
> >>> > As soon as external SIP call land and I try to bridge the call to
> >>> > GTalk client, FS crashes.
> >>> >
> >>> >
> >>> > NAT Details:
> >>> > ---------------------------
> >>> > I think my NAT does not support UpNP or PMP. The reason I say it
> >>> > because when FS starts following message is displayed:
> >>> >
> >>> > 2010-02-17 09:15:42.809674 [INFO] switch_nat.c:409 Scanning for NAT
> >>> > 2010-02-17 09:15:42.809674 [ERR] switch_nat.c:197 Error checking for
> >>> > PMP [init failed]
> >>> > 2010-02-17 09:15:42.809674 [DEBUG] switch_nat.c:414 Checking for UPnP
> >>> > 2010-02-17 09:15:56.825388 [DEBUG] switch_nat.c:114 No
> >>> > InternetGatewayDevice, using first entry as default
> >>> > (http://192.168.16.17:50144/).
> >>> > 2010-02-17 09:15:56.887889 [INFO] switch_nat.c:429 No PMP or UPnP NAT
> >>> > devices detected!
> >>> >
> >>> >
> >>> >
> >>> > On Tue, Feb 16, 2010 at 8:41 PM, Brian West <brian at freeswitch.org>
> >>> > wrote:
> >>> >> can you please update...
> >>> >
> >>> > _______________________________________________
> >>> > FreeSWITCH-users mailing list
> >>> > FreeSWITCH-users at lists.freeswitch.org
> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> > http://www.freeswitch.org
> >>> >
> >>> >
> >>>
> >>>
> >>>
> >>> --
> >>> Regards,
> >>> Moiz Chinoy.
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>
> >>
> >>
> >> --
> >> Anthony Minessale II
> >>
> >> FreeSWITCH http://www.freeswitch.org/
> >> ClueCon http://www.cluecon.com/
> >> Twitter: http://twitter.com/FreeSWITCH_wire
> >>
> >> AIM: anthm
> >> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >> IRC: irc.freenode.net #freeswitch
> >>
> >> FreeSWITCH Developer Conference
> >> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> >> iax:guest at conference.freeswitch.org/888
> >> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >> pstn:+19193869900
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >
> >
> >
> > --
> > Regards,
> > Moiz Chinoy.
> >
>
>
>
> --
> Regards,
> Moiz Chinoy.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100223/5c7e9c74/attachment-0002.html 

From rob4manhere at gmail.com  Tue Feb 23 11:51:22 2010
From: rob4manhere at gmail.com (Rob Forman)
Date: Tue, 23 Feb 2010 13:51:22 -0600
Subject: [Freeswitch-users] SIP provider recommendation for US termination
Message-ID: <F56BB50F-6584-4235-91C4-54E4C6EF4867@gmail.com>

Hey all,

I'm having on-going sporadic issues with one of my SIP providers (call  
quality, delayed or lost DTMFs, high random PPD).  Does anyone have  
some good experiences (for US termination) in terms of both quality  
and support?  There are so many bad ones out there; I don't want to  
switch blindly.

I don't know if we're supposed to share commercial endorsements on  
here.  If you have advice, would you mind dropping me a note off-list  
at rob4manhere (at) gmail.com.

Many thanks,
Rob



From joseph.puchalski at personalcyberspace.com  Tue Feb 23 12:12:41 2010
From: joseph.puchalski at personalcyberspace.com (Joseph Puchalski)
Date: Tue, 23 Feb 2010 20:12:41 +0000
Subject: [Freeswitch-users] How to set outbound caller id info
	for	multiple users/extensions
In-Reply-To: <4B834719.3000505@aktzero.com>
References: <093DD565390C1E4FB15D7B383E86BB05AF05AB@Goose.personalcyberspace.net>
	<191c3a031002130707v5f1d5ee0lb9e791e9a0aab956@mail.gmail.com>
	<3FF62C88-2423-43F0-B8A3-C64EF4BC80AC@freeswitch.org>
	<093DD565390C1E4FB15D7B383E86BB05AF15CD@Goose.personalcyberspace.net>
	<4B834719.3000505@aktzero.com>
Message-ID: <093DD565390C1E4FB15D7B383E86BB05AF1857@Goose.personalcyberspace.net>

Thanks, this helps. Setting "effective_caller_id*" in my extension xml file doesn't work for me at all.

Something must be wrong somewhere else in my config.

I think I'll probably go back and reinstall from the beginning. When I did this initially I made some xml changes late at night that seemed logical at the time.

Thanks again,

Joe

From: Andrew Thompson [mailto:andrewkt at aktzero.com]
Sent: Monday, February 22, 2010 10:10 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] How to set outbound caller id info for multiple users/extensions

On 2/22/2010 7:24 PM, Joseph Puchalski wrote:
Or should I be doing this via <action application="set" data="effective_caller_id_number=${outbound_caller_id_number}"/>
Somewhere else?

I have the following set on my own extension, in 1000.xml:

      <variable name="effective_caller_id_name" value="Andrew Thompson"/>
      <variable name="effective_caller_id_number" value="1000"/>
      <variable name="outbound_caller_id_name" value="MyCompany"/>
      <variable name="outbound_caller_id_number" value="9109991234"/>

When I dial extensions internally, the effective_* name/number show up.

When I dial outbound via my SIP provider, I set the following before the bridge so that it passes externally valid info:

      <action application="set" data="effective_caller_id_number=${outbound_caller_id_number}"/>
      <action application="set" data="effective_caller_id_name=${outbound_caller_id_name}"/>


In my setup, if I don't explicitly overide the effective_* with outbound_*, I actually see 1000 as my callerid when I call my cell from my extension, so if you're not getting at least that much, something else might be wrong. (I have used vitelity, and they do pass callerid properly most of the time.)



--

Andrew Thompson
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100223/b9450fcf/attachment-0002.html 

From william.suffill at gmail.com  Tue Feb 23 12:43:59 2010
From: william.suffill at gmail.com (William Suffill)
Date: Tue, 23 Feb 2010 15:43:59 -0500
Subject: [Freeswitch-users] SIP provider recommendation for US
	termination
In-Reply-To: <F56BB50F-6584-4235-91C4-54E4C6EF4867@gmail.com>
References: <F56BB50F-6584-4235-91C4-54E4C6EF4867@gmail.com>
Message-ID: <6b65470d1002231243x4268de5di655831071c9a28ab@mail.gmail.com>

There is a freeswitch-biz list too. I'm sure more people are faced with this
issue as well so it might be a good topic for the biz list.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100223/2487f618/attachment-0002.html 

From m.sobkow at marketelsystems.com  Tue Feb 23 13:00:46 2010
From: m.sobkow at marketelsystems.com (Mark Sobkow)
Date: Tue, 23 Feb 2010 15:00:46 -0600
Subject: [Freeswitch-users] mod_erlang_event
Message-ID: <4B8441FE.80506@marketelsystems.com>

It's become clear that I need to use Erlang event processing to do what 
I need to do with Freeswitch, but I can't even get the most basic of 
tasks working yet.  (i.e. Answer the call and collect the PIN code from 
the operator.)

The dialplan version of what I'm trying to do is:


<extension name=\"OpCallsIn\">
    <condition field=\"destination_number\" expression=\"^6100$\">
        <action application=\"answer\" />
        <action application=\"play_and_get_digits\" data=\"4 4 1 5000 # 
/opt/freeswitch/sounds/en/us/callie/conference/8000/conf-pin.wav 
/opt/freeswitch/sounds/en/us/callie/conference/8000/conf-bad-pin.wav 
operator_pin \\d+\" />
        <action application=\"log\" data=\"INFO operator_pin is 
${operator_pin}\" />
        <action application=\"log\" data=\"INFO Queueing operator to 
fifo\" />
        <action application=\"fifo\" data=\"6100@${domain_name} in\" />
    </condition>
</extension>

Attached is the Erlang that's attempting to do the same thing.  The 
Erlang is invoked by the following dialplan fragment:

<extension name=\"ErlangCallback\">
    <condition field=\"destination_number\" expression=\"^6200$\">
        <action application=\"erlang\" data=\"pbx_callback 
pursuit at testsrv\" />
    </condition>
</extension>

Any suggestions?

-- 
Mark Sobkow
Senior Developer
MarkeTel Multi-Line Dialing Systems LTD.
428 Victoria Ave
Regina, SK S4N-0P6
Toll-Free: 800-289-8616-X533
Local: 306-359-6893-X533
Fax: 306-359-6879
Email: m.sobkow at marketelsystems.com
Web: http://www.marketelsystems.com 

-------------- next part --------------
An embedded and charset-unspecified text was scrubbed...
Name: pbx_callback.erl
Url: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100223/55402511/attachment-0002.pl 

From msc at freeswitch.org  Tue Feb 23 13:35:24 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 23 Feb 2010 13:35:24 -0800
Subject: [Freeswitch-users] Hook Flash
In-Reply-To: <742756.71167.qm@web33501.mail.mud.yahoo.com>
References: <915F9D0D-FB13-4D02-995B-FB7F5EB488D2@tlainvestments.com>
	<742756.71167.qm@web33501.mail.mud.yahoo.com>
Message-ID: <87f2f3b91002231335o8c0b6f4vee98dbcd4d2b994d@mail.gmail.com>

On Tue, Feb 23, 2010 at 5:33 AM, Diego Toro <dftoro at yahoo.com> wrote:

> hi, read http://jira.freeswitch.org/browse/OPENZAP-30
>
>  <action application="send_dtmf" data="F"/>
>
> Diego Toro
> http://lacarretade.blogspot.com/
>

FYI, I didn't see this on the wiki so I added it to the OpenZAP FAQ:
http://wiki.freeswitch.org/wiki/OpenZAP#FAQ

Thanks Diego,
MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100223/d66f268e/attachment-0002.html 

From msc at freeswitch.org  Tue Feb 23 13:47:04 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 23 Feb 2010 13:47:04 -0800
Subject: [Freeswitch-users] call from an internal extension to external
	number
In-Reply-To: <D616CD3E2BFC4FDDA7ABEE7A3F268A4B@MOBILEE1705>
References: <D616CD3E2BFC4FDDA7ABEE7A3F268A4B@MOBILEE1705>
Message-ID: <87f2f3b91002231347k6fd1e5b8wc6543ba2696b1629@mail.gmail.com>

On Mon, Feb 22, 2010 at 7:29 PM, Madovsky <infos at madovsky.org> wrote:

>  Hi,
>
> day after I undertand a littlee more all these xml hell files (not friendly
> to read ;)),
>
Use a text editor that does syntax highlighting. :)


> but to be a PERl developer since 1999 understand regex and PERL language
> make life more easy...
>
Hint: don't say PERL, say Perl or perl instead. People who say PERL are
considered uneducated.


> However, I don't understand yet the concept of internal exterenal.
> is it for phone registration AND outbound calls ?
>
Internal and external are SIP profiles. Each SIP profile is a SIP user
agent, or UA. For a more complete discussion on this topic check out
http://en.wikipedia.org/wiki/User_agent

In short, the internal profile is listening on a particular IP and port and
usually it's to listen for registrations and calls from your telephones, as
well as to send calls out to your telephones. The external profile is
generally used just for outbound gateway registrations.


> for now I try to make an external call from 1000 ext (registered on port
> 5060)
> so I added an extension in dialplan/default.xml
>
>     <extension name="outbound_europe">
>         <condition field="destination_number" expression="^00?(\d+)$">
>               <action application="bridge"
> data="sofia/gateway/myprovider_europe/00$1"/>
>         </condition>
>     </extension>
> so if call starts with "00" it redirects to my provider that manage
> outbound calls, is it correct ?
> I put the myprovider.xml account into sip_profiles/external/myprovider.xml.
>
At first look this appears correct. You can make sure that the gateway is up
by typing "sofia status" at the fs_cli prompt. If you are having trouble
with making calls it is best to watch the debug output very carefully. It is
a lot of information to look at but eventually you will learn to focus on
the information that you need.

You're doing well! Just keep plugging away at it and you will figure it all
out and soon you will be helping others. :)
-MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100223/4a85626a/attachment-0002.html 

From msc at freeswitch.org  Tue Feb 23 13:57:51 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 23 Feb 2010 13:57:51 -0800
Subject: [Freeswitch-users] Error loading
	/usr/local/freeswitch/mod/ozmod_sangoma_boost.so
In-Reply-To: <fe67a3341002230202y1d45a8f3t3a2cda07de2240a2@mail.gmail.com>
References: <fe67a3341002230202y1d45a8f3t3a2cda07de2240a2@mail.gmail.com>
Message-ID: <87f2f3b91002231357t6c3e40cdld6a0a9861a9ec5a5@mail.gmail.com>

Does the file ozmod_sangoma_boost.so exist somewhere in your openzap build
tree? If not then something went wrong while compiling or you have an old
version. If it does exist, do a quick test: cp the file into
/usr/local/freeswitch/mod and restart FreeSWITCH and see if it finds the
file and loads OpenZAP properly. Let us know the results so we can determine
if it's a bug in the build system or not.

-MC

On Tue, Feb 23, 2010 at 2:02 AM, Nagalenoj H. <nagalenoj at gmail.com> wrote:

> Dear friends,
>     I've installed freeswitch trunk - 16729 and tried to configure with
> wanpipe for sangoma A102 pri card.
>
> Followed the steps given in
> http://wiki.sangoma.com/wanpipe-freeswitch-install
>
> When loading the freeswitch, I've got the following error.
>
> 2010-02-23 14:49:58.545726 [INFO] ozmod_wanpipe.c:335 configuring device
> s2c15 as OpenZAP device 1:30 fd:57 DTMF: software
> 2010-02-23 14:49:58.546004 [ERR] ozmod_wanpipe.c:436 No valid wanpipe span
> and channel was specified
> 2010-02-23 14:49:58.546275 [INFO] zap_io.c:2500 Configured 30 channel(s)
> 2010-02-23 14:49:58.560724 [ERR] zap_io.c:2560 Error loading
> /usr/local/freeswitch/mod/ozmod_sangoma_boost.so
> [/usr/local/freeswitch/mod/ozmod_sangoma_boost.so: cannot open shared object
> file: No such file or directory]
> 2010-02-23 14:49:58.561048 [ERR] zap_io.c:2720 can't find 'sangoma_boost'
> 2010-02-23 14:49:58.561266 [ERR] mod_openzap.c:2458 Error starting OpenZAP
> span 1 error:
> 2010-02-23 14:49:58.561712 [NOTICE] switch_loadable_module.c:144 Adding
> Endpoint 'openzap'
>
> Configuration and log files are pasted to pastebin. Kindly someone help me
> to solve this issue.
>
> openzap.conf and openzap.conf.xml
>    http://pastebin.freeswitch.org/12214
>
> freeswitch log
>    http://pastebin.freeswitch.org/12216
>
> smg_pri.conf
>    http://pastebin.freeswitch.org/12217
>
> --
> Regards,
> Nagalenoj H.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100223/10258572/attachment-0002.html 

From msc at freeswitch.org  Tue Feb 23 14:00:09 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 23 Feb 2010 14:00:09 -0800
Subject: [Freeswitch-users] big thanks to all freeswitch developers and
	contributing users
In-Reply-To: <4B83F040.7040005@xpirio.com>
References: <4B83F040.7040005@xpirio.com>
Message-ID: <87f2f3b91002231400h146b48ckb26aa407945b5979@mail.gmail.com>

2010/2/23 Christian L?schenkohl <christian.loeschenkohl at xpirio.com>

> i want to say a big THANKY YOU to all contributing freeswitch community
> members.
>
> over one year has passed since i did fall in love with this project.
> it is getting better every day, one get's help and advices if needed.
> the admins do care about nearly every problem - no matter if it's big or
> small.
> i also did manage an opensource project and i wish i had done it with that
> much
> heart and intense power that i see here.
>
> i also hope that i can contribute back enough (questions, bug reports, wiki
> enhancements).
>
> Don't forget "sitting on IRC all day long helping newcomers!" :)
-MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100223/1d11fbfa/attachment-0002.html 

From jeff at jefflenk.com  Tue Feb 23 14:05:33 2010
From: jeff at jefflenk.com (Jeff Lenk)
Date: Tue, 23 Feb 2010 16:05:33 -0600
Subject: [Freeswitch-users] FScomm
In-Reply-To: <D64783281177440B9094B8181E55EFA7@MOBILEE1705>
References: <157FDB96FB88476FBAEC69C1309A0A5F@MOBILEE1705>,
	<COL117-W5563292C886257AFC56548B9420@phx.gbl>,
	<D64783281177440B9094B8181E55EFA7@MOBILEE1705>
Message-ID: <COL117-W429F3BED9F781A58264238B9420@phx.gbl>


http://wiki.freeswitch.org/wiki/FSComm#Linux

 

you must run those from the FSComm directory
 


From: infos at madovsky.org
To: freeswitch-users at lists.freeswitch.org
Date: Tue, 23 Feb 2010 11:51:51 -0500
Subject: Re: [Freeswitch-users] FScomm




 

----- Original Message ----- 
From: Jeff Lenk 
To: freeswitch-users at lists.freeswitch.org 
Sent: Tuesday, February 23, 2010 9:23 AM
Subject: Re: [Freeswitch-users] FScomm

What platform are you trying to build?
 


From: infos at madovsky.org
To: freeswitch-users at lists.freeswitch.org
Date: Tue, 23 Feb 2010 01:41:04 -0500
Subject: [Freeswitch-users] FScomm




Hi,
 
is http://wiki.freeswitch.org/wiki/FSComm
available yet ? because QT framework link has broken
and some instruction don't work with last svn trunk (for example to compile FScomm in FS svn root)
 
Thanks

Franck


Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. Sign up now. 





_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

 
FSComm on Linux fedora 10 64 bits
It says FSComm can be built inside FS svn folder typing gmake make
but there is no Makefile inside
 
Thanks

Franck 		 	   		  
_________________________________________________________________
Hotmail: Powerful Free email with security by Microsoft.
http://clk.atdmt.com/GBL/go/201469230/direct/01/
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100223/842c5a3d/attachment-0002.html 

From andrew at hijacked.us  Tue Feb 23 14:49:02 2010
From: andrew at hijacked.us (Andrew Thompson)
Date: Tue, 23 Feb 2010 17:49:02 -0500
Subject: [Freeswitch-users] mod_erlang_event
In-Reply-To: <4B8441FE.80506@marketelsystems.com>
References: <4B8441FE.80506@marketelsystems.com>
Message-ID: <20100223224902.GB1751@hijacked.us>

Comments inline.

On Tue, Feb 23, 2010 at 03:00:46PM -0600, Mark Sobkow wrote:
> It's become clear that I need to use Erlang event processing to do what 
> I need to do with Freeswitch, but I can't even get the most basic of 
> tasks working yet.  (i.e. Answer the call and collect the PIN code from 
> the operator.)
> 
> The dialplan version of what I'm trying to do is:
> 
> 
> <extension name=\"OpCallsIn\">
>    <condition field=\"destination_number\" expression=\"^6100$\">
>        <action application=\"answer\" />
>        <action application=\"play_and_get_digits\" data=\"4 4 1 5000 # 
> /opt/freeswitch/sounds/en/us/callie/conference/8000/conf-pin.wav 
> /opt/freeswitch/sounds/en/us/callie/conference/8000/conf-bad-pin.wav 
> operator_pin \\d+\" />
>        <action application=\"log\" data=\"INFO operator_pin is 
> ${operator_pin}\" />
>        <action application=\"log\" data=\"INFO Queueing operator to 
> fifo\" />
>        <action application=\"fifo\" data=\"6100@${domain_name} in\" />
>    </condition>
> </extension>
> 
> Attached is the Erlang that's attempting to do the same thing.  The 
> Erlang is invoked by the following dialplan fragment:
> 
> <extension name=\"ErlangCallback\">
>    <condition field=\"destination_number\" expression=\"^6200$\">
>        <action application=\"erlang\" data=\"pbx_callback 
> pursuit at testsrv\" />
>    </condition>
> </extension>
> 
> Any suggestions?

Why not request the pin in the dialplan and then yield call control to
erlang? That's what I do most of the time.
 

> %% Author: mark
> %% Created: Feb 23, 2010
> %% Description: TODO: Add description to pbx_callback
> -module(pbx_callback).
> 
> %%
> %% Include files
> %%
> 
> %%
> %% Exported Functions
> %%
> -export([start/0, run/0, launch/1]).
> 
> start() ->
> 	Pid = spawn( ?MODULE, run, [] ),
> 	register( ?MODULE, Pid ),
> 	{ ok, Pid }.
> 
> run() ->
> 	receive
> 		{ call, Data } ->
> 			{ event, [UUID | Rest]} = Data,
> 			syslog:debug( "pbx_callback:run() New call received, UUID=~p, Rest=~p~n", [UUID, Rest] ),
> 			AnswerResults = pbx:api( eval, "uuid:" ++ UUID ++ " answer" ),
> 			syslog:debug( "pbx_callback:run() AnswerResults=~p~n", [AnswerResults] ),
> 			GetPinResults = pbx:api( eval, "uuid:" ++ UUID ++ " play_and_get_digits 4 4 1 5000 # /opt/freeswitch/sounds/en/us/callie/conference/8000/conf-pin.wav /opt/freeswitch/sounds/en/us/callie/conference/8000/conf-bad-pin.wav operator_pin \\d+" ),
> 			syslog:debug( "pbx_callback:run() GetPinResults=~p~n", [GetPinResults] ),
> 			GetPinVarResults = pbx:api( uuid_getvar, UUID ++ " operator_pin" ),
> 			syslog:debug( "pbx_callback:run() GetPinVarResults=~p~n", [GetPinVarResults] ),
> 			run();
> 		{call_event, Data} ->
> 			{ event, [UUID | Rest]} = Data,
> 			Name = proplists:get_value( "Event-Name", Rest ),
> 			syslog:debug( "pbx_callback:run() call_event UUID=~p, Name=~p, Rest=~p~n", [UUID, Name, Rest] ),
> 			run();
> 		{get_pid, UUID, Ref, Pid} ->
> 			NewPid = spawn( ?MODULE, run, [] ),
> 			syslog:debug( "pbx_callback:run() Request to spawn new handler process, returning PID ~p~n", [NewPid] ),
> 			Pid ! { Ref, NewPid },
> 			run()
> 	end.
> 
> launch( Ref ) ->
> 	NewPid = spawn( ?MODULE, run, [] ),
> 	syslog:debug( "pbx_callback:launch() Returning new PID ~p~n", [NewPid] ),
> 	{Ref, NewPid}.

I don't know what your 'pbx' module is doing so I can't really help you
there. Are you doing a sendmsg for play_and_get_digits or what? You
should be using a uuid_getvar to get the result of the
play_and_get_digits in any case. How far does this code get before
failing?

Andrew



From infos at madovsky.org  Tue Feb 23 15:18:58 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 23 Feb 2010 18:18:58 -0500
Subject: [Freeswitch-users] FScomm
References: <157FDB96FB88476FBAEC69C1309A0A5F@MOBILEE1705>,
	<COL117-W5563292C886257AFC56548B9420@phx.gbl>,
	<D64783281177440B9094B8181E55EFA7@MOBILEE1705>
	<COL117-W429F3BED9F781A58264238B9420@phx.gbl>
Message-ID: <11F8492DFC2546598E6FDC3E538301CD@MOBILEE1705>


----- Original Message ----- 
From: Jeff Lenk
To: freeswitch-users at lists.freeswitch.org
Sent: Tuesday, February 23, 2010 5:05 PM
Subject: Re: [Freeswitch-users] FScomm


http://wiki.freeswitch.org/wiki/FSComm#Linux

you must run those from the FSComm directory



From: infos at madovsky.org
To: freeswitch-users at lists.freeswitch.org
Date: Tue, 23 Feb 2010 11:51:51 -0500
Subject: Re: [Freeswitch-users] FScomm



----- Original Message ----- 
From: Jeff Lenk
To: freeswitch-users at lists.freeswitch.org
Sent: Tuesday, February 23, 2010 9:23 AM
Subject: Re: [Freeswitch-users] FScomm


What platform are you trying to build?



From: infos at madovsky.org
To: freeswitch-users at lists.freeswitch.org
Date: Tue, 23 Feb 2010 01:41:04 -0500
Subject: [Freeswitch-users] FScomm


Hi,

is http://wiki.freeswitch.org/wiki/FSComm
available yet ? because QT framework link has broken
and some instruction don't work with last svn trunk (for example to compile 
FScomm in FS svn root)

Thanks

Franck



Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. Sign up now.




_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


FSComm on Linux fedora 10 64 bits
It says FSComm can be built inside FS svn folder typing gmake make
but there is no Makefile inside

Thanks

Franck



Hotmail: Powerful Free email with security by Microsoft. Get it now.



_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



It's what I did,
but from FS trunk, inside fscomm directory,
there s only

account.cpp  conf                fshost.h        mainwindow.ui 
resources.qrc
account.h    FSComm.2008.vcproj  main.cpp        mod_qsettings
call.cpp     FSComm.pro          mainwindow.cpp  preferences
call.h       fshost.cpp          mainwindow.h    resources




From brian at microcomaustralia.com.au  Tue Feb 23 15:24:31 2010
From: brian at microcomaustralia.com.au (Brian May)
Date: Wed, 24 Feb 2010 10:24:31 +1100
Subject: [Freeswitch-users] Possible typo on Openzap.conf wiki page
	specifying fxs/fxo-channels
In-Reply-To: <2C33D9D6C1444AAC80950CDF8D38A428@greyhawk.tonecommander.com>
References: <2C33D9D6C1444AAC80950CDF8D38A428@greyhawk.tonecommander.com>
Message-ID: <3c5cf5261002231524n4e9af307y80b3accb1ee4ff26@mail.gmail.com>

On 24 February 2010 04:48, Robert Hadley <robert.hadley at teotech.com> wrote:
> On the http://wiki.freeswitch.org/wiki/Openzap.conf_Examples wiki page, is
> there a typo in the wanpipe /usr/local/freeswitch/conf/openzap.conf example
> concerning specifying the fxo-channel vs. fxs-channel?

I agree. The first example looks correct to me; the 2nd example looks wrong.

See the table I created to try and explain what term to use where:

http://wiki.freeswitch.org/wiki/OpenZAP#FXO.2FFXS_Terminology

In the examples you quoted, ports 1 and 2 are extension ports, so are
FXS ports, but should be defined as FXO ports in openzap.conf.

Ports 3 and 4 are telephone line ports, so are FXO ports, but should
be defined as FXS ports in openzap.conf.

I have only used zaptel myself, however I suspect the same applies to wanpipe.
-- 
Brian May <brian at microcomaustralia.com.au>



From infos at madovsky.org  Tue Feb 23 15:29:29 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 23 Feb 2010 18:29:29 -0500
Subject: [Freeswitch-users] call from an internal extension to
	externalnumber
References: <D616CD3E2BFC4FDDA7ABEE7A3F268A4B@MOBILEE1705>
	<87f2f3b91002231347k6fd1e5b8wc6543ba2696b1629@mail.gmail.com>
Message-ID: <9130B3FED335446C85DB26E71489FB91@MOBILEE1705>


  ----- Original Message ----- 
  From: Michael Collins 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Tuesday, February 23, 2010 4:47 PM
  Subject: Re: [Freeswitch-users] call from an internal extension to externalnumber





  On Mon, Feb 22, 2010 at 7:29 PM, Madovsky <infos at madovsky.org> wrote:

    Hi,

    day after I undertand a littlee more all these xml hell files (not friendly to read ;)),
  Use a text editor that does syntax highlighting. :)
   

    but to be a PERl developer since 1999 understand regex and PERL language make life more easy...
  Hint: don't say PERL, say Perl or perl instead. People who say PERL are considered uneducated.
   

    However, I don't understand yet the concept of internal exterenal.
    is it for phone registration AND outbound calls ?
  Internal and external are SIP profiles. Each SIP profile is a SIP user agent, or UA. For a more complete discussion on this topic check out http://en.wikipedia.org/wiki/User_agent

  In short, the internal profile is listening on a particular IP and port and usually it's to listen for registrations and calls from your telephones, as well as to send calls out to your telephones. The external profile is generally used just for outbound gateway registrations.
   
    for now I try to make an external call from 1000 ext (registered on port 5060)
    so I added an extension in dialplan/default.xml 

        <extension name="outbound_europe">
            <condition field="destination_number" expression="^00?(\d+)$">
                  <action application="bridge" data="sofia/gateway/myprovider_europe/00$1"/>
            </condition>
        </extension>

    so if call starts with "00" it redirects to my provider that manage outbound calls, is it correct ?
    I put the myprovider.xml account into sip_profiles/external/myprovider.xml.
  At first look this appears correct. You can make sure that the gateway is up by typing "sofia status" at the fs_cli prompt. If you are having trouble with making calls it is best to watch the debug output very carefully. It is a lot of information to look at but eventually you will learn to focus on the information that you need.

  You're doing well! Just keep plugging away at it and you will figure it all out and soon you will be helping others. :)
  -MC





------------------------------------------------------------------------------


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  http://www.freeswitch.org


  Use a text editor that does syntax highlighting. :)

  black an white on my putty ssh ;)


  Thanks for your help !

  Franck
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100223/4903abf4/attachment-0002.html 

From infos at madovsky.org  Tue Feb 23 15:37:02 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 23 Feb 2010 18:37:02 -0500
Subject: [Freeswitch-users] FreeSWITCH manual
Message-ID: <8E2F03C27AD4415BBDF1E9FF16F3DA98@MOBILEE1705>

Hi dev friends !

Is this manual yet available for the last trunk version ?

http://www.scribd.com/doc/17425068/Free-Switch-in-Real-Life

Thanks

Franck
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100223/99101e64/attachment-0002.html 

From rupa at rupa.com  Tue Feb 23 15:37:41 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Tue, 23 Feb 2010 17:37:41 -0600
Subject: [Freeswitch-users] mod_nibblebill
In-Reply-To: <BLU110-W28EA8BD06A601D5D85268BA7420@phx.gbl>
References: <BLU110-W28EA8BD06A601D5D85268BA7420@phx.gbl>
Message-ID: <b39273531002231537k626d1698vd88549a7f59e73a@mail.gmail.com>

what database backend are you using?  Have you verified the SQL is going to
the right database backend?  I use mod_nibblebill against postgresql w/out
problems.

On Tue, Feb 23, 2010 at 9:42 AM, Meg Stroodle <phunk0000 at hotmail.com> wrote:

>  Hello List!  I am trying to install mod_nibblebill on my FS
> installation.  I get the following log entry form FS & nibblebill, but the
> database table I setup remains unchanged.  Any help in this matter would be
> greatly appreciated.  Following is an excerpt from the FS log:
>
>
>
> 2010-02-23 10:32:52.869149 [DEBUG] switch_rtp.c:2004 Correct ip/port
> confirmed.
>
> 2010-02-23 10:32:53.077162 [DEBUG] sofia.c:3727 Channel
> sofia/internal/3007 at 192.168.15.177 entering state [ready][200]
>
> 2010-02-23 10:32:53.149166 [DEBUG] switch_core_session.c:706 Send signal
> sofia/internal/sip:3008 at 192.168.15.176:21828 [BREAK]
>
> 2010-02-23 10:32:53.149166 [DEBUG] switch_core_session.c:706 Send signal
> sofia/internal/3007 at 192.168.15.177 [BREAK]
>
> 2010-02-23 10:32:53.149166 [DEBUG] switch_ivr_bridge.c:131 sofia/internal/
> sip:3008 at 192.168.15.176:21828 receive message [DISPLAY]
>
> 2010-02-23 10:32:53.169168 [DEBUG] switch_ivr_bridge.c:131
> sofia/internal/3007 at 192.168.15.177 receive message [DISPLAY]
>
> 2010-02-23 10:33:21.388949 [DEBUG] mod_nibblebill.c:529 Received request
> via SESSION_HEARTBEAT!
>
> 2010-02-23 10:33:21.388949 [DEBUG] mod_nibblebill.c:397 Attempting to bill
> at $1.0 per minute to account 3007
>
> 2010-02-23 10:33:21.388949 [INFO] mod_nibblebill.c:447 Beginning new
> billing on 3ec17ffc-c94e-46ff-bb0d-7ad0ef27bdab
>
> 2010-02-23 10:33:21.388949 [DEBUG] mod_nibblebill.c:455 28 seconds passed
> since last bill time of 2010-02-23 10:32:52
>
> 2010-02-23 10:33:21.388949 [DEBUG] mod_nibblebill.c:461 Billing $0.475997
> to 3007 (Call: 3ec17ffc-c94e-46ff-bb0d-7ad0ef27bdab / 0.000000 so far)
>
> 2010-02-23 10:33:51.508822 [DEBUG] mod_nibblebill.c:529 Received request
> via SESSION_HEARTBEAT!
>
> 2010-02-23 10:33:51.508822 [DEBUG] mod_nibblebill.c:397 Attempting to bill
> at $1.0 per minute to account 3007
>
> 2010-02-23 10:33:51.508822 [DEBUG] mod_nibblebill.c:449 Last successful
> billing time was
>
> 2010-02-23 10:33:51.508822 [DEBUG] mod_nibblebill.c:455 30 seconds passed
> since last bill time of 2010-02-23 10:33:21
>
> 2010-02-23 10:33:51.508822 [DEBUG] mod_nibblebill.c:461 Billing $0.501998
> to 3007 (Call: 3ec17ffc-c94e-46ff-bb0d-7ad0ef27bdab / 0.475997 so far)
>
> 2010-02-23 10:34:21.649708 [DEBUG] mod_nibblebill.c:529 Received request
> via SESSION_HEARTBEAT!
>
> 2010-02-23 10:34:21.649708 [DEBUG] mod_nibblebill.c:397 Attempting to bill
> at $1.0 per minute to account 3007
>
> 2010-02-23 10:34:21.649708 [DEBUG] mod_nibblebill.c:449 Last successful
> billing time was
>
> 2010-02-23 10:34:21.649708 [DEBUG] mod_nibblebill.c:455 30 seconds passed
> since last bill time of 2010-02-23 10:33:51
>
> 2010-02-23 10:34:21.649708 [DEBUG] mod_nibblebill.c:461 Billing $0.502348
> to 3007 (Call: 3ec17ffc-c94e-46ff-bb0d-7ad0ef27bdab / 0.977995 so far)
>
> 2010-02-23 10:34:26.254308 [NOTICE] sofia.c:329 Hangup
> sofia/internal/3007 at 192.168.15.177 [CS_EXECUTE] [NORMAL_CLEARING]
>
> 2010-02-23 10:34:26.254308 [DEBUG] switch_channel.c:1912 Send signal
> sofia/internal/3007 at 192.168.15.177 [KILL]
>
> 2010-02-23 10:34:26.254308 [DEBUG] switch_core_session.c:999 Send signal
> sofia/internal/3007 at 192.168.15.177 [BREAK]
>
> 2010-02-23 10:34:26.254308 [DEBUG] switch_core_state_machine.c:459
> sofia/internal/3007 at 192.168.15.177 thread mismatch skipping state handler.
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_ivr_bridge.c:470
> sofia/internal/3007 at 192.168.15.177 ending bridge by request from read
> function
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_ivr_bridge.c:563 BRIDGE THREAD
> DONE [sofia/internal/3007 at 192.168.15.177]
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_ivr_bridge.c:565 Send signal
> sofia/internal/sip:3008 at 192.168.15.176:21828 [BREAK]
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_ivr_bridge.c:520 sofia/internal/
> sip:3008 at 192.168.15.176:21828 receive message [UNBRIDGE]
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_session.c:645 Send signal
> sofia/internal/sip:3008 at 192.168.15.176:21828 [BREAK]
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_ivr_bridge.c:563 BRIDGE THREAD
> DONE [sofia/internal/sip:3008 at 192.168.15.176:21828]
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_ivr_bridge.c:565 Send signal
> sofia/internal/3007 at 192.168.15.177 [BREAK]
>
> 2010-02-23 10:34:26.269008 [NOTICE] switch_ivr_bridge.c:617 Hangup
> sofia/internal/sip:3008 at 192.168.15.176:21828 [CS_EXCHANGE_MEDIA]
> [NORMAL_CLEARING]
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_channel.c:1912 Send signal
> sofia/internal/sip:3008 at 192.168.15.176:21828 [KILL]
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_session.c:999 Send signal
> sofia/internal/sip:3008 at 192.168.15.176:21828 [BREAK]
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:488
> (sofia/internal/sip:3008 at 192.168.15.176:21828) State HANGUP
>
> 2010-02-23 10:34:26.269008 [DEBUG] mod_sofia.c:358 Channel sofia/internal/
> sip:3008 at 192.168.15.176:21828 hanging up, cause: NORMAL_CLEARING
>
> 2010-02-23 10:34:26.269008 [DEBUG] mod_sofia.c:400 Sending BYE to
> sofia/internal/sip:3008 at 192.168.15.176:21828
>
> 2010-02-23 10:34:26.269008 [DEBUG] mod_nibblebill.c:397 Attempting to bill
> at $2.30 per minute to account 3008
>
> 2010-02-23 10:34:26.269008 [INFO] mod_nibblebill.c:447 Beginning new
> billing on c1c42712-89e9-44c5-914f-11ed8f6aebb1
>
> 2010-02-23 10:34:26.269008 [DEBUG] mod_nibblebill.c:455 93 seconds passed
> since last bill time of 2010-02-23 10:32:52
>
> 2010-02-23 10:34:26.269008 [DEBUG] mod_nibblebill.c:461 Billing $3.582436
> to 3008 (Call: c1c42712-89e9-44c5-914f-11ed8f6aebb1 / 0.000000 so far)
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:46
> sofia/internal/sip:3008 at 192.168.15.176:21828 Standard HANGUP, cause:
> NORMAL_CLEARING
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:488
> (sofia/internal/sip:3008 at 192.168.15.176:21828) State HANGUP going to sleep
>
> 2010-02-23 10:34:26.269008 [INFO] switch_core_session.c:1108
> sofia/internal/sip:3008 at 192.168.15.176:21828 setting session heartbeat to
> 30 second(s).
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:351
> (sofia/internal/sip:3008 at 192.168.15.176:21828) State EXCHANGE_MEDIA going
> to sleep
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:314
> (sofia/internal/sip:3008 at 192.168.15.176:21828) Running State Change
> CS_HANGUP
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:465
> sofia/internal/sip:3008 at 192.168.15.176:21828 handler already called,
> skipping state handler.
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:333
> (sofia/internal/sip:3008 at 192.168.15.176:21828) State Change CS_HANGUP ->
> CS_REPORTING
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_session.c:999 Send signal
> sofia/internal/sip:3008 at 192.168.15.176:21828 [BREAK]
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:314
> (sofia/internal/sip:3008 at 192.168.15.176:21828) Running State Change
> CS_REPORTING
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:579
> (sofia/internal/sip:3008 at 192.168.15.176:21828) State REPORTING
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:53
> sofia/internal/sip:3008 at 192.168.15.176:21828 Standard REPORTING, cause:
> NORMAL_CLEARING
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:579
> (sofia/internal/sip:3008 at 192.168.15.176:21828) State REPORTING going to
> sleep
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:327
> (sofia/internal/sip:3008 at 192.168.15.176:21828) State Change CS_REPORTING
> -> CS_DESTROY
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_session.c:999 Send signal
> sofia/internal/sip:3008 at 192.168.15.176:21828 [BREAK]
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_session.c:1136 Session 6
> (sofia/internal/sip:3008 at 192.168.15.176:21828) Locked, Waiting on external
> entities
>
> 2010-02-23 10:34:26.272437 [DEBUG] switch_core_state_machine.c:348 (
> sofia/internal/3007 at 192.168.15.177) State EXECUTE going to sleep
>
> 2010-02-23 10:34:26.272437 [DEBUG] switch_core_state_machine.c:314 (
> sofia/internal/3007 at 192.168.15.177) Running State Change CS_HANGUP
>
> 2010-02-23 10:34:26.272437 [DEBUG] switch_core_state_machine.c:488 (
> sofia/internal/3007 at 192.168.15.177) State HANGUP
>
> 2010-02-23 10:34:26.272437 [DEBUG] mod_sofia.c:358 Channel
> sofia/internal/3007 at 192.168.15.177 hanging up, cause: NORMAL_CLEARING
>
> 2010-02-23 10:34:26.272437 [DEBUG] mod_nibblebill.c:397 Attempting to bill
> at $1.0 per minute to account 3007
>
> 2010-02-23 10:34:26.272437 [DEBUG] mod_nibblebill.c:449 Last successful
> billing time was
>
> 2010-02-23 10:34:26.272437 [DEBUG] mod_nibblebill.c:455 4 seconds passed
> since last bill time of 2010-02-23 10:34:21
>
> 2010-02-23 10:34:26.272437 [DEBUG] mod_nibblebill.c:461 Billing $0.077045
> to 3007 (Call: 3ec17ffc-c94e-46ff-bb0d-7ad0ef27bdab / 1.480343 so far)
>
> 2010-02-23 10:34:26.272437 [DEBUG] switch_core_state_machine.c:46
> sofia/internal/3007 at 192.168.15.177 Standard HANGUP, cause: NORMAL_CLEARING
>
>
>
> Anyhelp getting nibblebill to connect to the database would be greatly
> appreciated.  Thanks
>
>
>
> ------------------------------
> Hotmail: Trusted email with Microsoft?s powerful SPAM protection. Sign up
> now. <http://clk.atdmt.com/GBL/go/201469226/direct/01/>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100223/1e887229/attachment-0002.html 

From brian at microcomaustralia.com.au  Tue Feb 23 15:49:15 2010
From: brian at microcomaustralia.com.au (Brian May)
Date: Wed, 24 Feb 2010 10:49:15 +1100
Subject: [Freeswitch-users] Possible typo on Openzap.conf wiki page
	specifying fxs/fxo-channels
In-Reply-To: <2C33D9D6C1444AAC80950CDF8D38A428@greyhawk.tonecommander.com>
References: <2C33D9D6C1444AAC80950CDF8D38A428@greyhawk.tonecommander.com>
Message-ID: <3c5cf5261002231549s7e847c91l98529a95432b7175@mail.gmail.com>

On 24 February 2010 04:48, Robert Hadley <robert.hadley at teotech.com> wrote:
> [span zt FXS]
> name => OpenZAP
> number => 4165551111
>
> fxs-channel => 3
> number => 4165552222
> fxs-channel => 4

Two other details really confused me at first, and I don't think are
addressed in the documentation.

1. What is this "number" setting?  Some of the examples make it look
like it is a channel number:

=== cut ===
[span zt FXS1]
name => OpenZAP-FXS
number => 1
fxs-channel => 1

[span zt FXO1]
name => OpenZAP-FXO1
number => 2
fxo-channel => 3

[span zt FXO2]
name => OpenZAP-FXO2
number => 3
fxo-channel => 4
=== cut ===

It is not, it looks like on FXO ports it is the telephone number used
when looking up the dialplan for incoming calls; for FXS ports it is
the telephone number used for the callerid.


2. I have seen examples that use different formats for <span ...>:

<span id="FXO">
<span name="SPAN1" id="1">
<span name="PRI_1">
<span id="1">

Which syntax is correct? Which one should we be trying to use? If both
name= and  id= are specified, which one is used?
-- 
Brian May <brian at microcomaustralia.com.au>



From brian at microcomaustralia.com.au  Tue Feb 23 16:27:38 2010
From: brian at microcomaustralia.com.au (Brian May)
Date: Wed, 24 Feb 2010 11:27:38 +1100
Subject: [Freeswitch-users] internal/external profiles
Message-ID: <3c5cf5261002231627s1dfb04b2m64bf9bfeb3ff171d@mail.gmail.com>

Hello,

Why is it recommended to use separate profiles for internal and external SIP?

This page:
<http://wiki.freeswitch.org/wiki/External_profile>
suggests it is because of NAT.

However this page recommends using separate profiles even if NAT is
not an issue:
<http://wiki.freeswitch.org/wiki/Auto_NAT>: "NOTE: It is still
recommended that you use a second profile for your SIP providers. The
default conf/sip_profiles/external.xml is set up specifically for use
with providers."

However I am still left uncertain what this means.

Not trying to criticize here, just trying to learn.

Thanks.
-- 
Brian May <brian at microcomaustralia.com.au>



From msc at freeswitch.org  Tue Feb 23 16:47:18 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 23 Feb 2010 16:47:18 -0800
Subject: [Freeswitch-users] FreeSWITCH manual
In-Reply-To: <8E2F03C27AD4415BBDF1E9FF16F3DA98@MOBILEE1705>
References: <8E2F03C27AD4415BBDF1E9FF16F3DA98@MOBILEE1705>
Message-ID: <87f2f3b91002231647g4c216c9av8c9417c5ff37281f@mail.gmail.com>

On Tue, Feb 23, 2010 at 3:37 PM, Madovsky <infos at madovsky.org> wrote:

>  Hi dev friends !
>
> Is this manual yet available for the last trunk version ?
>
> http://www.scribd.com/doc/17425068/Free-Switch-in-Real-Life
>
> Thanks
>
> Franck
>
No, that is a very old document. There is, however, a FreeSWITCH book in the
works. It's almost drafted and still has to go through the editing process
before it will be published.
-MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100223/61657f5c/attachment-0002.html 

From msc at freeswitch.org  Tue Feb 23 16:50:59 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 23 Feb 2010 16:50:59 -0800
Subject: [Freeswitch-users] REMINDER: FreeSWITCH Conf Call Moved to
	Wednesday!
Message-ID: <87f2f3b91002231650q6388c789v5f1287c7d2703204@mail.gmail.com>

Hi all,

Just a reminder that we are meeting up on Wednesay morning. Here's the
agenda:
http://wiki.freeswitch.org/wiki/FS_weekly_2010_02_24

It is light since we only had a few days since our meeting last Friday.
Remember that I won't be in right at the start of the meeting because I will
be taking my kids to school. Please feel free to use that time to mingle...

Thanks,
Michael
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100223/7ba1bb66/attachment-0002.html 

From infos at madovsky.org  Tue Feb 23 17:04:42 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 23 Feb 2010 20:04:42 -0500
Subject: [Freeswitch-users] FS directories explaination
Message-ID: <4D4DE3DF3BD344809C43D68B3E410CF3@MOBILEE1705>

Hi,

Maybe a wiki page of directories description of conf directory would be great..

Thanks

Franck
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100223/893ff61f/attachment-0002.html 

From brian at microcomaustralia.com.au  Tue Feb 23 17:17:24 2010
From: brian at microcomaustralia.com.au (Brian May)
Date: Wed, 24 Feb 2010 12:17:24 +1100
Subject: [Freeswitch-users] FS directories explaination
In-Reply-To: <4D4DE3DF3BD344809C43D68B3E410CF3@MOBILEE1705>
References: <4D4DE3DF3BD344809C43D68B3E410CF3@MOBILEE1705>
Message-ID: <3c5cf5261002231717k64418e34yf78fa24315652dbe@mail.gmail.com>

On 24 February 2010 12:04, Madovsky <infos at madovsky.org> wrote:
> Maybe a wiki page of directories description of conf
> directory would be great..

Not sure if I understand your question (do you want documentation on
/conf/directory/ or all of /conf/?), did you see this page?

http://wiki.freeswitch.org/wiki/XML_User_Directory_Guide
-- 
Brian May <brian at microcomaustralia.com.au>



From infos at madovsky.org  Tue Feb 23 18:44:08 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 23 Feb 2010 21:44:08 -0500
Subject: [Freeswitch-users] freeswitch minimum install
Message-ID: <219A4D36FE4A42A3B40A594F960461B9@MOBILEE1705>

Hi,

Is there a way to install freeswitch from source with the
strict minimum xml necassary in the conf dir to run freeswitch ?

Thanks

Franck
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100223/6546a80b/attachment-0002.html 

From infos at madovsky.org  Tue Feb 23 18:53:45 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 23 Feb 2010 21:53:45 -0500
Subject: [Freeswitch-users] call from an internal extension to
	externalnumber
References: <D616CD3E2BFC4FDDA7ABEE7A3F268A4B@MOBILEE1705>
	<87f2f3b91002231347k6fd1e5b8wc6543ba2696b1629@mail.gmail.com>
Message-ID: <7AB1FDA0C14343DFB0FB30D32A0ED33C@MOBILEE1705>


    day after I undertand a littlee more all these xml hell files (not friendly to read ;)),
  Use a text editor that does syntax highlighting. :)
   

    but to be a PERl developer since 1999 understand regex and PERL language make life more easy...
  Hint: don't say PERL, say Perl or perl instead. People who say PERL are considered uneducated.
   

    mmhmm,  I said PERL as you say SIP, ti's an acronysm....
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100223/61d9573c/attachment-0002.html 

From xanlich at gmail.com  Tue Feb 23 18:54:53 2010
From: xanlich at gmail.com (Chia-Yen Wu)
Date: Wed, 24 Feb 2010 10:54:53 +0800
Subject: [Freeswitch-users] Time condition in Lua
Message-ID: <314dc3f81002231854s7f20447brc48a1051bb83edbc@mail.gmail.com>

hello

is there anyway to do the time condition in lua script?
the only way i know is get the infomation by strftime() and compare it
but not all of them, like "*month of week" *doesnt support by strftime()
which dialplan XML does.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100224/8521bd6c/attachment-0002.html 

From Russell.Mosemann at cune.org  Tue Feb 23 19:07:33 2010
From: Russell.Mosemann at cune.org (Russell Mosemann)
Date: Tue, 23 Feb 2010 21:07:33 -0600
Subject: [Freeswitch-users] call from an internal extension
	toexternalnumber
In-Reply-To: <7AB1FDA0C14343DFB0FB30D32A0ED33C@MOBILEE1705>
References: <D616CD3E2BFC4FDDA7ABEE7A3F268A4B@MOBILEE1705><87f2f3b91002231347k6fd1e5b8wc6543ba2696b1629@mail.gmail.com>
	<7AB1FDA0C14343DFB0FB30D32A0ED33C@MOBILEE1705>
Message-ID: <E5386A8959B24CE59F401F9ECEFF201C@cune.pri>

Madovsky said:

> 		mmhmm,  I said PERL as you say SIP, ti's an acronysm....

No, it is not an acronym. See the following.

http://en.wikipedia.org/wiki/Perl

" When referring to the language, the name is normally capitalized (Perl) as a proper noun, as you would a spoken language (e.g. English or French). When referring to the interpreter program itself, the name is often uncapitalized (perl) because most Unix-like file systems are case-sensitive. Before the release of the first edition of Programming Perl, it was common to refer to the language as perl; Randal L. Schwartz, however, capitalized the language's name in the book to make it stand out better when typeset. This case distinction was subsequently documented as canonical."

" There is some contention about the all-caps spelling "PERL," which the documentation declares incorrect and which some core community members consider a sign of outsiders. Although the name is occasionally taken as an acronym for Practical Extraction and Report Language (which appears at the top of the documentation and in some printed literature), this expansion actually came after the name; several others have been suggested as equally canonical, including Wall's own humorous Pathologically Eclectic Rubbish Lister. Indeed, Wall claims that the name was intended to inspire many different expansions."

--
Russell Mosemann




From infos at madovsky.org  Tue Feb 23 19:57:50 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 23 Feb 2010 22:57:50 -0500
Subject: [Freeswitch-users] call from an internal
	extensiontoexternalnumber
References: <D616CD3E2BFC4FDDA7ABEE7A3F268A4B@MOBILEE1705><87f2f3b91002231347k6fd1e5b8wc6543ba2696b1629@mail.gmail.com><7AB1FDA0C14343DFB0FB30D32A0ED33C@MOBILEE1705>
	<E5386A8959B24CE59F401F9ECEFF201C@cune.pri>
Message-ID: <E510B0579DFB435FB3C5E99FD3451AC1@MOBILEE1705>


----- Original Message ----- 
From: "Russell Mosemann" <Russell.Mosemann at cune.org>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Tuesday, February 23, 2010 10:07 PM
Subject: Re: [Freeswitch-users] call from an internal 
extensiontoexternalnumber


> Madovsky said:
>
>> mmhmm,  I said PERL as you say SIP, ti's an acronysm....
>
> No, it is not an acronym. See the following.
>
> http://en.wikipedia.org/wiki/Perl
>
> " When referring to the language, the name is normally capitalized (Perl) 
> as a proper noun, as you would a spoken language (e.g. English or French). 
> When referring to the interpreter program itself, the name is often 
> uncapitalized (perl) because most Unix-like file systems are 
> case-sensitive. Before the release of the first edition of Programming 
> Perl, it was common to refer to the language as perl; Randal L. Schwartz, 
> however, capitalized the language's name in the book to make it stand out 
> better when typeset. This case distinction was subsequently documented as 
> canonical."
>
> " There is some contention about the all-caps spelling "PERL," which the 
> documentation declares incorrect and which some core community members 
> consider a sign of outsiders. Although the name is occasionally taken as 
> an acronym for Practical Extraction and Report Language (which appears at 
> the top of the documentation and in some printed literature), this 
> expansion actually came after the name; several others have been suggested 
> as equally canonical, including Wall's own humorous Pathologically 
> Eclectic Rubbish Lister. Indeed, Wall claims that the name was intended to 
> inspire many different expansions."
>
> --
> Russell Mosemann
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

They Changed definition since they changed thei website...
old website it said
PERL - Practical Extraction Report Language

Thanks

Franck 




From gorand at donevtechconsulting.com  Tue Feb 23 20:23:58 2010
From: gorand at donevtechconsulting.com (Goran Donev)
Date: Tue, 23 Feb 2010 22:23:58 -0600
Subject: [Freeswitch-users] Are we close to final version 1.05
In-Reply-To: <mailman.6775.1266983891.2495.freeswitch-users@lists.freeswitch.org>
References: <mailman.6775.1266983891.2495.freeswitch-users@lists.freeswitch.org>
Message-ID: <12c101cab509$31893850$949ba8f0$@com>

Just checking if all you wonderful developers of this great project have an
ETA as to the final stable code of 1.05 to set into production environments.


It has been two weeks since the Feb 8th dinner and announcement of release
of 1.05. 

Thanks
Goran




From talk2ram at gmail.com  Tue Feb 23 20:34:02 2010
From: talk2ram at gmail.com (ram)
Date: Wed, 24 Feb 2010 10:04:02 +0530
Subject: [Freeswitch-users] Fwd: Freeswitch to another Freeswitch(or
	gateway)
In-Reply-To: <f8af5741002230600h6911e676iae6fc48375b31b1@mail.gmail.com>
References: <f8af5741002222202p3457e8efl2e683082dc597b47@mail.gmail.com>
	<f8af5741002230600h6911e676iae6fc48375b31b1@mail.gmail.com>
Message-ID: <b74751491002232034s7b32c9f1q96f1ade7c88ecbf8@mail.gmail.com>

On Tue, Feb 23, 2010 at 7:30 PM, srinivasula reddy <
srinivas.ksvreddy at gmail.com> wrote:

>
>  Hi,
> i want divert calls from my sipserver to another sipserver or third party
> gateway based on the host name, is there any way to achive this.
>

how about this examples

http://wiki.freeswitch.org/wiki/Dialplan_XML#Example_1

Ram
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100224/5b9aa9d7/attachment-0002.html 

From troy at tlainvestments.com  Tue Feb 23 20:38:17 2010
From: troy at tlainvestments.com (Troy Anderson)
Date: Tue, 23 Feb 2010 21:38:17 -0700
Subject: [Freeswitch-users] Hook Flash
In-Reply-To: <87f2f3b91002231335o8c0b6f4vee98dbcd4d2b994d@mail.gmail.com>
References: <915F9D0D-FB13-4D02-995B-FB7F5EB488D2@tlainvestments.com>
	<742756.71167.qm@web33501.mail.mud.yahoo.com>
	<87f2f3b91002231335o8c0b6f4vee98dbcd4d2b994d@mail.gmail.com>
Message-ID: <84CAA6FA-F4C6-4BD5-83BB-9A19D782103F@tlainvestments.com>

Thanks, Diego!

-Troy


On Feb 23, 2010, at 2:35 PM, Michael Collins wrote:

> 
> 
> On Tue, Feb 23, 2010 at 5:33 AM, Diego Toro <dftoro at yahoo.com> wrote:
> hi, read http://jira.freeswitch.org/browse/OPENZAP-30
> 
>  <action application="send_dtmf" data="F"/>
> 
> Diego Toro
> http://lacarretade.blogspot.com/
> 
> FYI, I didn't see this on the wiki so I added it to the OpenZAP FAQ:
> http://wiki.freeswitch.org/wiki/OpenZAP#FAQ
> 
> Thanks Diego,
> MC
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100223/9accac2a/attachment-0002.html 

From nagalenoj at gmail.com  Tue Feb 23 20:43:45 2010
From: nagalenoj at gmail.com (Nagalenoj H.)
Date: Wed, 24 Feb 2010 10:13:45 +0530
Subject: [Freeswitch-users] Error loading
	/usr/local/freeswitch/mod/ozmod_sangoma_boost.so
In-Reply-To: <87f2f3b91002231357t6c3e40cdld6a0a9861a9ec5a5@mail.gmail.com>
References: <fe67a3341002230202y1d45a8f3t3a2cda07de2240a2@mail.gmail.com>
	<87f2f3b91002231357t6c3e40cdld6a0a9861a9ec5a5@mail.gmail.com>
Message-ID: <fe67a3341002232043w23e5c04bh2598bdd9641d9c23@mail.gmail.com>

ozmod_sangoma_boost.so doesn't exist anywhere. It may not be a old version,
since I've checked out the source yesterday.

I've a doubt in the installation steps given. It is given to edit the
modules.conf after executing ./configure. Is it right? Do I need to edit the
modules.conf before ./configure??

On Wed, Feb 24, 2010 at 3:27 AM, Michael Collins <msc at freeswitch.org> wrote:

> Does the file ozmod_sangoma_boost.so exist somewhere in your openzap build
> tree? If not then something went wrong while compiling or you have an old
> version. If it does exist, do a quick test: cp the file into
> /usr/local/freeswitch/mod and restart FreeSWITCH and see if it finds the
> file and loads OpenZAP properly. Let us know the results so we can determine
> if it's a bug in the build system or not.
>
> -MC
>
> On Tue, Feb 23, 2010 at 2:02 AM, Nagalenoj H. <nagalenoj at gmail.com> wrote:
>
>> Dear friends,
>>     I've installed freeswitch trunk - 16729 and tried to configure with
>> wanpipe for sangoma A102 pri card.
>>
>> Followed the steps given in
>> http://wiki.sangoma.com/wanpipe-freeswitch-install
>>
>> When loading the freeswitch, I've got the following error.
>>
>> 2010-02-23 14:49:58.545726 [INFO] ozmod_wanpipe.c:335 configuring device
>> s2c15 as OpenZAP device 1:30 fd:57 DTMF: software
>> 2010-02-23 14:49:58.546004 [ERR] ozmod_wanpipe.c:436 No valid wanpipe span
>> and channel was specified
>> 2010-02-23 14:49:58.546275 [INFO] zap_io.c:2500 Configured 30 channel(s)
>> 2010-02-23 14:49:58.560724 [ERR] zap_io.c:2560 Error loading
>> /usr/local/freeswitch/mod/ozmod_sangoma_boost.so
>> [/usr/local/freeswitch/mod/ozmod_sangoma_boost.so: cannot open shared object
>> file: No such file or directory]
>> 2010-02-23 14:49:58.561048 [ERR] zap_io.c:2720 can't find 'sangoma_boost'
>> 2010-02-23 14:49:58.561266 [ERR] mod_openzap.c:2458 Error starting OpenZAP
>> span 1 error:
>> 2010-02-23 14:49:58.561712 [NOTICE] switch_loadable_module.c:144 Adding
>> Endpoint 'openzap'
>>
>> Configuration and log files are pasted to pastebin. Kindly someone help me
>> to solve this issue.
>>
>> openzap.conf and openzap.conf.xml
>>    http://pastebin.freeswitch.org/12214
>>
>> freeswitch log
>>    http://pastebin.freeswitch.org/12216
>>
>> smg_pri.conf
>>    http://pastebin.freeswitch.org/12217
>>
>> --
>> Regards,
>> Nagalenoj H.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Regards,
Nagalenoj H.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100224/1b8c3402/attachment-0002.html 

From brian at freeswitch.org  Tue Feb 23 21:04:04 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 23 Feb 2010 23:04:04 -0600
Subject: [Freeswitch-users] Are we close to final version 1.05
In-Reply-To: <12c101cab509$31893850$949ba8f0$@com>
References: <mailman.6775.1266983891.2495.freeswitch-users@lists.freeswitch.org>
	<12c101cab509$31893850$949ba8f0$@com>
Message-ID: <3B5CB034-7DD5-406D-BC06-DA1B7A10F19F@freeswitch.org>

Its coming... we had a flood of issues we wanted to resolve before 1.0.5... I thank everyone that did donate for the dinner.

/b

On Feb 23, 2010, at 10:23 PM, Goran Donev wrote:

> It has been two weeks since the Feb 8th dinner and announcement of release
> of 1.05. 




From nazim.agabekov at gmail.com  Tue Feb 23 21:07:17 2010
From: nazim.agabekov at gmail.com (Nazim Agabekov)
Date: Wed, 24 Feb 2010 09:07:17 +0400
Subject: [Freeswitch-users] Time condition in Lua
In-Reply-To: <314dc3f81002231854s7f20447brc48a1051bb83edbc@mail.gmail.com>
References: <314dc3f81002231854s7f20447brc48a1051bb83edbc@mail.gmail.com>
Message-ID: <4B84B405.9020609@gmail.com>

Hello Wu

That's what I've found searching Lua Wiki:

http://lua-users.org/wiki/DayOfWeekAndDaysInMonthExample

There are a lot of helpful examples which could be used to create a
powerful calendar application.
Did you mean to say "week of month"?

Nazim.

On 02/24/2010 06:54 AM, Chia-Yen Wu wrote:
> hello
>
> is there anyway to do the time condition in lua script?
> the only way i know is get the infomation by strftime() and compare it
> but not all of them, like "*month of week" *doesnt support by
> strftime() which dialplan XML does.
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100224/34d81043/attachment-0002.html 

From anthony.minessale at gmail.com  Tue Feb 23 21:36:51 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 23 Feb 2010 23:36:51 -0600
Subject: [Freeswitch-users] Are we close to final version 1.05
In-Reply-To: <191c3a031002232135s1e12d42evbbb87fd7ecfde505@mail.gmail.com>
References: <mailman.6775.1266983891.2495.freeswitch-users@lists.freeswitch.org>
	<12c101cab509$31893850$949ba8f0$@com>
	<3B5CB034-7DD5-406D-BC06-DA1B7A10F19F@freeswitch.org>
	<191c3a031002232135s1e12d42evbbb87fd7ecfde505@mail.gmail.com>
Message-ID: <191c3a031002232136t43fe155fs64eb5a4da4c69e4f@mail.gmail.com>

Just an fyi, there will probably be less bugs in trunk the day after we
release it than there was in that release.  =P

On Feb 23, 2010 11:10 PM, "Brian West" <brian at freeswitch.org> wrote:

Its coming... we had a flood of issues we wanted to resolve before 1.0.5...
I thank everyone that did donate for the dinner.

/b


On Feb 23, 2010, at 10:23 PM, Goran Donev wrote:

> It has been two weeks since the Feb 8th dinner ...

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists...
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100223/6fe457db/attachment-0002.html 

From dome at tel.co.th  Tue Feb 23 21:54:00 2010
From: dome at tel.co.th (Dome Charoenyost)
Date: Wed, 24 Feb 2010 12:54:00 +0700
Subject: [Freeswitch-users] Mod_h323 On openvz (64 bit)
Message-ID: <8ccbff061002232154u38bee079y3226a09d0f36bce6@mail.gmail.com>

Dear All,
I'm testing mod_h323 follow
http://wiki.freeswitch.org/wiki/Mod_h323

FS running on openvz 64 bit (kernel 2.6.18 pve 1.5 from proxmox)
After build mod_h323 sucessful  i found problem when load mod_h323 in FS CLI

2010-02-24 11:59:01.517104 [CRIT] switch_loadable_module.c:882 Error
Loading module /opt/freeswitch/mod/mod_h323.so
**/opt/freeswitch/mod/mod_h323.so: undefined symbol:
_ZN14H323Connection24OnUserInputInlineRFC2833ER15OpalRFC2833Infoi**

How to fix this problem ?


BG
Dome C.
2010/2/24 Chia-Yen Wu <xanlich at gmail.com>:
> hello
> is there anyway to do the time condition in lua script?
> the only way i know is get the infomation by?strftime() and compare it
> but not all of them, like "month of week"?doesnt support by strftime() which
> dialplan XML does.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From bekelemartins at gmail.com  Tue Feb 23 19:33:41 2010
From: bekelemartins at gmail.com (Bekele Martins)
Date: Tue, 23 Feb 2010 22:33:41 -0500
Subject: [Freeswitch-users] Streaming conference audio to a website
Message-ID: <5cc9e8f31002231933na50bd6o90e0c23a6667175a@mail.gmail.com>

Hello. I'm new to Freeswitch, but I have a question.
Is it possible to have a conference call and then stream the audio of the
conference to my website so people can listen to it over the Internet?
I read about mod_conference, but I couldn't find the answer to my question.
Thank you.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100223/eefb8a17/attachment-0002.html 

From mike at jerris.com  Wed Feb 24 00:14:15 2010
From: mike at jerris.com (Michael Jerris)
Date: Wed, 24 Feb 2010 03:14:15 -0500
Subject: [Freeswitch-users] freeswitch minimum install
In-Reply-To: <219A4D36FE4A42A3B40A594F960461B9@MOBILEE1705>
References: <219A4D36FE4A42A3B40A594F960461B9@MOBILEE1705>
Message-ID: <8171AC69-88E7-4793-8C24-2594387C15E5@jerris.com>

no, other than manually creating that minimum conf

Mike
On Feb 23, 2010, at 9:44 PM, Madovsky wrote:

> Is there a way to install freeswitch from source with the
> strict minimum xml necassary in the conf dir to run FreeSWITCH ?

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100224/e368db19/attachment-0002.html 

From mike at jerris.com  Wed Feb 24 00:23:54 2010
From: mike at jerris.com (Michael Jerris)
Date: Wed, 24 Feb 2010 03:23:54 -0500
Subject: [Freeswitch-users] Error loading
	/usr/local/freeswitch/mod/ozmod_sangoma_boost.so
In-Reply-To: <fe67a3341002232043w23e5c04bh2598bdd9641d9c23@mail.gmail.com>
References: <fe67a3341002230202y1d45a8f3t3a2cda07de2240a2@mail.gmail.com>
	<87f2f3b91002231357t6c3e40cdld6a0a9861a9ec5a5@mail.gmail.com>
	<fe67a3341002232043w23e5c04bh2598bdd9641d9c23@mail.gmail.com>
Message-ID: <D62532C0-53A1-4327-9BAA-DB2FB0C4BEF8@jerris.com>

you missed the second 1/2 of step 3 of Wanpipe TDM Installation


On Feb 23, 2010, at 11:43 PM, Nagalenoj H. wrote:

> ozmod_sangoma_boost.so doesn't exist anywhere. It may not be a old version, since I've checked out the source yesterday.
> 
> I've a doubt in the installation steps given. It is given to edit the modules.conf after executing ./configure. Is it right? Do I need to edit the modules.conf before ./configure??
> 
> On Wed, Feb 24, 2010 at 3:27 AM, Michael Collins <msc at freeswitch.org> wrote:
> Does the file ozmod_sangoma_boost.so exist somewhere in your openzap build tree? If not then something went wrong while compiling or you have an old version. If it does exist, do a quick test: cp the file into /usr/local/freeswitch/mod and restart FreeSWITCH and see if it finds the file and loads OpenZAP properly. Let us know the results so we can determine if it's a bug in the build system or not.
> 
> -MC
> 
> On Tue, Feb 23, 2010 at 2:02 AM, Nagalenoj H. <nagalenoj at gmail.com> wrote:
> Dear friends,
>     I've installed freeswitch trunk - 16729 and tried to configure with wanpipe for sangoma A102 pri card.
> 
> Followed the steps given in http://wiki.sangoma.com/wanpipe-freeswitch-install
> 
> When loading the freeswitch, I've got the following error.
> 
> 2010-02-23 14:49:58.545726 [INFO] ozmod_wanpipe.c:335 configuring device s2c15 as OpenZAP device 1:30 fd:57 DTMF: software
> 2010-02-23 14:49:58.546004 [ERR] ozmod_wanpipe.c:436 No valid wanpipe span and channel was specified
> 2010-02-23 14:49:58.546275 [INFO] zap_io.c:2500 Configured 30 channel(s)
> 2010-02-23 14:49:58.560724 [ERR] zap_io.c:2560 Error loading /usr/local/freeswitch/mod/ozmod_sangoma_boost.so [/usr/local/freeswitch/mod/ozmod_sangoma_boost.so: cannot open shared object file: No such file or directory]
> 2010-02-23 14:49:58.561048 [ERR] zap_io.c:2720 can't find 'sangoma_boost'
> 2010-02-23 14:49:58.561266 [ERR] mod_openzap.c:2458 Error starting OpenZAP span 1 error:
> 2010-02-23 14:49:58.561712 [NOTICE] switch_loadable_module.c:144 Adding Endpoint 'openzap'
> 
> Configuration and log files are pasted to pastebin. Kindly someone help me to solve this issue.
> 
> openzap.conf and openzap.conf.xml
>    http://pastebin.freeswitch.org/12214
> 
> freeswitch log
>    http://pastebin.freeswitch.org/12216
> 
> smg_pri.conf
>    http://pastebin.freeswitch.org/12217
> 
> -- 
> Regards,
> Nagalenoj H.
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> 
> -- 
> Regards,
> Nagalenoj H.
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100224/2d9ab798/attachment-0002.html 

From mike at jerris.com  Wed Feb 24 00:28:25 2010
From: mike at jerris.com (Michael Jerris)
Date: Wed, 24 Feb 2010 03:28:25 -0500
Subject: [Freeswitch-users] Dial String Inject in FreeSwitch
In-Reply-To: <622bedea1002221113j5ac06477jb24ac51eedcd8d8f@mail.gmail.com>
References: <622bedea1002220945r5fd10a4cp19ed1244a67374d9@mail.gmail.com>
	<20100222182741.05F7829BF68@cuneorg-email.cune.pri>
	<622bedea1002221038u2a2e4232qd1d15cf006abb7db@mail.gmail.com>
	<622bedea1002221113j5ac06477jb24ac51eedcd8d8f@mail.gmail.com>
Message-ID: <FC38F850-32AF-4FBF-9BBE-626CBA2E48B3@jerris.com>

Why not actually edit the real wiki pages instead of hiding this information on your user page?

On Feb 22, 2010, at 2:13 PM, Eder Souza wrote:

> http://wiki.freeswitch.org/wiki/User:Ederwander
>  
> On Mon, Feb 22, 2010 at 3:38 PM, Eder Souza <ederwander at gmail.com> wrote:
> Perfect place lol
> 
>  
> On Mon, Feb 22, 2010 at 3:27 PM, <Russell.Mosemann at cune.org> wrote:
> > i thaks if somebody create one wiki witch this alert
> 
> A place to change is Example 1 of the dialplan XML examples. You can tell
> people not to use the catchall expressions, because you cannot trust
> information from the sender.
> 
> http://wiki.freeswitch.org/wiki/Dialplan_XML
> 
> A word of caution could also be added to
> 
> http://wiki.freeswitch.org/wiki/Regular_Expression
> 
> --
> Russell Mosemann
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100224/80680633/attachment-0002.html 

From jason at jasonjgw.net  Wed Feb 24 00:31:36 2010
From: jason at jasonjgw.net (Jason White)
Date: Wed, 24 Feb 2010 19:31:36 +1100
Subject: [Freeswitch-users] Streaming conference audio to a website
In-Reply-To: <5cc9e8f31002231933na50bd6o90e0c23a6667175a@mail.gmail.com>
References: <5cc9e8f31002231933na50bd6o90e0c23a6667175a@mail.gmail.com>
Message-ID: <20100224083136.GA2549@jdc.jasonjgw.net>

Bekele Martins <bekelemartins at gmail.com> wrote:
> Is it possible to have a conference call and then stream the audio of the
> conference to my website so people can listen to it over the Internet?

You could turn on recording in the conference and have it written to a file
which is accessible to your Web server.
conference confname record filename




From mike at jerris.com  Wed Feb 24 00:32:15 2010
From: mike at jerris.com (Michael Jerris)
Date: Wed, 24 Feb 2010 03:32:15 -0500
Subject: [Freeswitch-users] SDP With T.38 in INVITE Problem
In-Reply-To: <b437a0e1002220849k16fd87ffl5a7257d80e43b036@mail.gmail.com>
References: <b437a0e1002190242y2aa64696j91e8bc76f7ae2a92@mail.gmail.com>
	<b437a0e1002220448g3e3bda9fpa283e93a11d319f1@mail.gmail.com>
	<24736FFE-BEEF-4D34-BBA7-77B6B6AE0EBD@jerris.com>
	<b437a0e1002220711u1bfd138l8a67b4e0b024d856@mail.gmail.com>
	<93769c21002220729q5bf36eadk80726aa8c00e3bd@mail.gmail.com>
	<b437a0e1002220849k16fd87ffl5a7257d80e43b036@mail.gmail.com>
Message-ID: <5ED9E41E-DFF1-4E03-B0F8-032309EA9A61@jerris.com>

if you want clarity on this, read the rfc for sdp offer answer.  You are not supposed to remove an m= line in an answer, if something is doing that, it is incorrect.

Mike

On Feb 22, 2010, at 11:49 AM, ivdreg ivdreg wrote:

> Hi Michael,
> 
> As I said in a previous mails I know exactly what is happening.
> In working setup:
> 
> ITSP ---> YATE (GW - Frontend) ---> FreeSwitch (routing server/xml_curl) ---> YATE (GW - SIP Interop) ---> OpenSIPS ---> Subscriber.
> 
> I prefer to change YATE (GW - Frontend) and YATE (GW - SIP Interop) with FreeSwitch for some reasons. The problem is:
> 
> INVITE comes from ITSP and goes to subscriber via OpenSIPS. INIVITE between FreeSwitch (routing server) and YATE (GW - SIP Interop) contains SDP: 
> m=audio 21108 RTP/AVP 18 4 8 0
> c=IN IP4 10.10.1.110
> a=rtpmap:18 G729/8000
> a=rtpmap:4 G723/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:0 PCMU/8000
> m=image 21108 udptl t38
> c=IN IP4 10.10.1.110
> a=T38FaxVersion:0
> a=T38MaxBitRate:14400
> a=T38FaxUdpEC:t38UDPRedundancy
> a=T38FaxRateManagement:
> transferredTCF
> 
> And reply 200 OK contains in SDP:
> m=audio 34788 RTP/AVP 8
> a=rtpmap:8 PCMA/8000
> a=silenceSupp:off - - - -
> a=ptime:20
> 
> Reply 200 OK SDP between YATE (GW - SIP Interop) and OpenSIPS contains in SDP:
> m=audio 16330 RTP/AVP 8
> a=rtpmap:8 PCMA/8000
> a=silenceSupp:off - - - -
> a=ptime:20
> m=image 0 udptl 19
> 
> In this case everything works fine. Line m=image 0 udptl 19 is removed by YATE. 
> But same test with FreeSWITCH on the place of YATE (GW - SIP Interop) "m=image 0 udptl 19" call brakes as you can see in my first mail.
> 
> I don't want to compare or discus YATE and FS functionality or something else. I just see difference in behavior and because I'm not a big expert don't know witch implementation is more accurate according standards. And second: Is it impossible for me to upgrade all CPE so only thing I can do is to fix it on server side. That is because I ask for a help.
> 
> 
> Thanks to all.
> 
> 
> 2010/2/22 Michael Jerris <mike at jerris.com>
> if you want to see what is going on, crank up the debug in freeswitch and sofia and you should see exactly what is going on.
> 
> Mike
> 
> 
> On Mon, Feb 22, 2010 at 10:11 AM, ivdreg ivdreg <ivdreg at gmail.com> wrote:
> Hi Michael,
> 
> This happens when ONLY IF initial INVITE is coming with T.38 from a GW (this is ITSP equipment and I don't know vendor) to our SIP subscribers with ATCOM ATA and IP Phone. We use now in production YATE for terminating and originating GWs to ITSPs and FS as main routing logic (backend). We want to switch YATE to FS for a GW also but we faced this problem. This not happens if initial INVITE nave no T.38 offered and later re-INVITE with T.38 with valid SDP port.
> 
> Thanks
> 
> 2010/2/22 Michael Jerris <mike at jerris.com>
> If the endpoint does not correctly follow the sdp o/a model its not going to work.  This is not a "problem" with the sofia library, this is intended behavior and what we are supposed to do.  What is the device?
> 
> Mike
> 
> On Feb 22, 2010, at 7:48 AM, ivdreg ivdreg wrote:
> 
>> Hi All,
>> 
>> Actually while seeking the solution in internet I see some people having this problem with sofia library. I'm not sure that SIP reply in this case contains a valid SDP (I think that teminating endpoint is broken) but in my opinion if we have at least one valid media type in SDP (video, audio, image ...) call must be established. Can someone comment and/or help me with this issue.
>> 
>> Regards.
>> 
>> 2010/2/19 ivdreg ivdreg <ivdreg at gmail.com>
>> Hi all,
>> 
>> Dose someone have a problem that if there T.38 in coming from gateway FreeSwitch drops the call because of media error ? As I see from log only T.38 port is zero and SDP has also media port. Is it possible to configure FS to do not break a call but if media is OK.
>> 
>> 2010-02-19 12:26:14.305244 [NOTICE] switch_channel.c:666 New Channel sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [6cd9f634-411d-df11-99ca-003048bb99cc]
>> 2010-02-19 12:26:14.305244 [DEBUG] mod_sofia.c:3369 (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State Change CS_NEW -> CS_INIT
>> 2010-02-19 12:26:14.305244 [DEBUG] switch_core_session.c:1018 Send signal sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [BREAK]
>> 2010-02-19 12:26:14.305244 [DEBUG] switch_core_state_machine.c:314 (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) Running State Change CS_INIT
>> 2010-02-19 12:26:14.305244 [DEBUG] switch_core_state_machine.c:338 (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State INIT
>> 2010-02-19 12:26:14.305244 [DEBUG] mod_sofia.c:83 sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 SOFIA INIT
>> 2010-02-19 12:26:14.305244 [DEBUG] sofia_glue.c:1377 sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 Patched SDP
>> ---
>> v=0
>> o=- 3779949069 5166785777234550622 IN IP4 206.132.232.212
>> s=session
>> t=0 0
>> m=audio 21108 RTP/AVP 18 4 8 0
>> c=IN IP4 10.10.1.110
>> a=rtpmap:18 G729/8000
>> a=rtpmap:4 G723/8000
>> a=rtpmap:8 PCMA/8000
>> a=rtpmap:0 PCMU/8000
>> m=image 21108 udptl t38
>> c=IN IP4 10.10.1.110
>> a=T38FaxVersion:0
>> a=T38MaxBitRate:14400
>> a=T38FaxUdpEC:t38UDPRedundancy
>> a=T38FaxRateManagement:transferredTCF
>> 
>> +++
>> v=0
>> o=- 3779949069 5166785777234550622 IN IP4 206.132.232.212
>> s=session
>> t=0 0
>> m=audio 17058 RTP/AVP 18 4 8 0
>> c=IN IP4 10.10.1.100
>> a=rtpmap:18 G729/8000
>> a=rtpmap:4 G723/8000
>> a=rtpmap:8 PCMA/8000
>> a=rtpmap:0 PCMU/8000
>> m=image 17058 udptl t38
>> c=IN IP4 10.10.1.100
>> a=T38FaxVersion:0
>> a=T38MaxBitRate:14400
>> a=T38FaxUdpEC:t38UDPRedundancy
>> a=T38FaxRateManagement:transferredTCF
>> 
>> 2010-02-19 12:26:14.305244 [DEBUG] mod_sofia.c:117 (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State Change CS_INIT -> CS_ROUTING
>> ......
>> 2010-02-19 12:26:21.255201 [DEBUG] sofia.c:4135 Remote SDP:
>> v=0
>> o=FreeSWITCH 1266548331 1266548332 IN IP4 10.10.1.110
>> s=FreeSWITCH
>> c=IN IP4 10.10.1.110
>> t=0 0
>> m=audio 26850 RTP/AVP 8
>> a=rtpmap:8 PCMA/8000
>> a=silenceSupp:off - - - -
>> a=ptime:20
>> m=image 0 udptl 19
>> 
>> 2010-02-19 12:26:21.255201 [DEBUG] sofia.c:4124 Channel sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 entering state [ready][200]
>> 2010-02-19 12:26:21.255201 [DEBUG] switch_channel.c:2285 Send signal sofia/backend/YYYYYYYYYY at 10.10.1.110 [BREAK]
>> 2010-02-19 12:26:21.255201 [NOTICE] sofia.c:4668 Channel [sofia/backend/XXXXXXXXXX at 10.10.1.110:7065] has been answered
>> 2010-02-19 12:26:21.255201 [DEBUG] sofia_glue.c:2288 Set Codec sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 PROXY/8000 20 ms 160 samples
>> 2010-02-19 12:26:21.255201 [DEBUG] sofia_glue.c:2571 PROXY AUDIO RTP [sofia/backend/XXXXXXXXXX at 10.10.1.110:7065] 10.10.1.100:17058->10.10.1.110:0 codec: 0 ms: 20
>> 2010-02-19 12:26:21.255201 [ERR] sofia_glue.c:2879 AUDIO RTP REPORTS ERROR: [Missing remote port]
>> 2010-02-19 12:26:21.255201 [NOTICE] sofia_glue.c:2880 Hangup sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [CS_CONSUME_MEDIA] [DESTINATION_OUT_OF_ORDER]
>> 2010-02-19 12:26:21.255201 [DEBUG] switch_channel.c:2063 Send signal sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [KILL]
>> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_session.c:1018 Send signal sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [BREAK]
>> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:314 (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) Running State Change CS_HANGUP
>> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:499 (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State HANGUP
>> 2010-02-19 12:26:21.255201 [DEBUG] mod_sofia.c:411 Channel sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 hanging up, cause: DESTINATION_OUT_OF_ORDER
>> 2010-02-19 12:26:21.255201 [DEBUG] mod_sofia.c:454 Sending BYE to sofia/backend/XXXXXXXXXX at 10.10.1.110:7065
>> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:46 sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 Standard HANGUP, cause: DESTINATION_OUT_OF_ORDER
>> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:499 (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State HANGUP going to sleep
>> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:333 (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State Change CS_HANGUP -> CS_REPORTING
>> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_session.c:1018 Send signal sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [BREAK]
>> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:314 (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) Running State Change CS_REPORTING
>> 2010-02-19 12:26:21.255201 [DEBUG] switch_ivr_originate.c:3185 Originate Resulted in Error Cause: 27 [DESTINATION_OUT_OF_ORDER]
>> 2010-02-19 12:26:21.255201 [INFO] mod_dptools.c:2353 Originate Failed.  Cause: DESTINATION_OUT_OF_ORDER
>> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:590 (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State REPORTING
>> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100224/df6bbcb5/attachment-0002.html 

From mike at jerris.com  Wed Feb 24 00:36:37 2010
From: mike at jerris.com (Michael Jerris)
Date: Wed, 24 Feb 2010 03:36:37 -0500
Subject: [Freeswitch-users] RTP timeout
In-Reply-To: <A68022CC2AFE40E29FE73385397A665B@MOBILEE1705>
References: <A68022CC2AFE40E29FE73385397A665B@MOBILEE1705>
Message-ID: <FD559959-D7F2-4BE7-B713-234478FFA72D@jerris.com>

If it hurts ? 

On Feb 23, 2010, at 1:09 PM, Madovsky wrote:

> Hi,
>  
> thanks to answer me if I misunderstood something,
> but if I run a softphone on the same IP as FS, there
> is an RTP timeout.
>  
> Any idea ?

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100224/12352730/attachment-0002.html 

From mike at jerris.com  Wed Feb 24 00:40:32 2010
From: mike at jerris.com (Michael Jerris)
Date: Wed, 24 Feb 2010 03:40:32 -0500
Subject: [Freeswitch-users] FScomm
In-Reply-To: <11F8492DFC2546598E6FDC3E538301CD@MOBILEE1705>
References: <157FDB96FB88476FBAEC69C1309A0A5F@MOBILEE1705>,
	<COL117-W5563292C886257AFC56548B9420@phx.gbl>,
	<D64783281177440B9094B8181E55EFA7@MOBILEE1705>
	<COL117-W429F3BED9F781A58264238B9420@phx.gbl>
	<11F8492DFC2546598E6FDC3E538301CD@MOBILEE1705>
Message-ID: <BAA7D201-6E09-4EDE-BBA0-FDFA64E38391@jerris.com>


On Feb 23, 2010, at 6:18 PM, Madovsky wrote:

> 
> ----- Original Message ----- 
> From: Jeff Lenk
> To: freeswitch-users at lists.freeswitch.org
> Sent: Tuesday, February 23, 2010 5:05 PM
> Subject: Re: [Freeswitch-users] FScomm
> 
> 
> http://wiki.freeswitch.org/wiki/FSComm#Linux
> 
> you must run those from the FSComm directory
> 

?.

> 
> It's what I did,
> but from FS trunk, inside fscomm directory,
> there s only
> 
> account.cpp  conf                fshost.h        mainwindow.ui 
> resources.qrc
> account.h    FSComm.2008.vcproj  main.cpp        mod_qsettings
> call.cpp     FSComm.pro          mainwindow.cpp  preferences
> call.h       fshost.cpp          mainwindow.h    resources
> 


Read those installation instructions again and do them step by step, you skipped one.


From mike at jerris.com  Wed Feb 24 00:44:10 2010
From: mike at jerris.com (Michael Jerris)
Date: Wed, 24 Feb 2010 03:44:10 -0500
Subject: [Freeswitch-users] Streaming conference audio to a website
In-Reply-To: <5cc9e8f31002231933na50bd6o90e0c23a6667175a@mail.gmail.com>
References: <5cc9e8f31002231933na50bd6o90e0c23a6667175a@mail.gmail.com>
Message-ID: <F373C26A-1A75-45F0-9AFB-42269F64E9F0@jerris.com>

http://wiki.freeswitch.org/wiki/Mod_conference#.3Cprofiles.3E

see auto-record, note the example with mod_shout

http://wiki.freeswitch.org/wiki/Mod_shout

On Feb 23, 2010, at 10:33 PM, Bekele Martins wrote:

> Hello. I'm new to Freeswitch, but I have a question. 
> Is it possible to have a conference call and then stream the audio of the conference to my website so people can listen to it over the Internet?
> I read about mod_conference, but I couldn't find the answer to my question.

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100224/6568fcb7/attachment-0002.html 

From xanlich at gmail.com  Wed Feb 24 01:14:45 2010
From: xanlich at gmail.com (Chia-Yen Wu)
Date: Wed, 24 Feb 2010 17:14:45 +0800
Subject: [Freeswitch-users] Time condition in Lua
In-Reply-To: <4B84B405.9020609@gmail.com>
References: <314dc3f81002231854s7f20447brc48a1051bb83edbc@mail.gmail.com>
	<4B84B405.9020609@gmail.com>
Message-ID: <314dc3f81002240114v30099971gdd64ecfc8d202c3a@mail.gmail.com>

yep , sorry i mean to say "week of month"

thank for help! that's what i needed!

2010/2/24 Nazim Agabekov <nazim.agabekov at gmail.com>

>  Hello Wu
>
> That's what I've found searching Lua Wiki:
>
> http://lua-users.org/wiki/DayOfWeekAndDaysInMonthExample
>
> There are a lot of helpful examples which could be used to create a
> powerful calendar application.
> Did you mean to say "week of month"?
>
> Nazim.
>
>
> On 02/24/2010 06:54 AM, Chia-Yen Wu wrote:
>
> hello
>
>  is there anyway to do the time condition in lua script?
> the only way i know is get the infomation by strftime() and compare it
> but not all of them, like "*month of week" *doesnt support by strftime()
> which dialplan XML does.
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100224/4d95aad9/attachment-0002.html 

From devel at thom.fr.eu.org  Wed Feb 24 01:25:20 2010
From: devel at thom.fr.eu.org (=?UTF-8?Q?Fran=C3=A7ois_Legal?=)
Date: Wed, 24 Feb 2010 10:25:20 +0100
Subject: [Freeswitch-users] Possible typo on Openzap.conf wiki page
 specifying fxs/fxo-channels
In-Reply-To: <3c5cf5261002231524n4e9af307y80b3accb1ee4ff26@mail.gmail.com>
References: <2C33D9D6C1444AAC80950CDF8D38A428@greyhawk.tonecommander.com>
	<3c5cf5261002231524n4e9af307y80b3accb1ee4ff26@mail.gmail.com>
Message-ID: <a689f57fe216d671d7d1ff132c6b636e@thom.fr.eu.org>

Well, I am the guy who made the modification in the wiki.

I use sangoma card and the openzap file is generated by the Setup script
from sangoma driver.
It seems that the terminology used by zaptel is not used in wanpipe
configuration.

I have an A400 card with an FXO module (providing ports 11 and 12) and an
FXS module (providing ports 9 and 10)

My openzap.conf is like this :

[span wanpipe FXS]
name => Analog phone 1
number => 9000
fxs-channel => 1:9
name => Analog phone 2
number => 9001
fxs-channel => 1:10

[span wanpipe FXO]
name => POTS line 1
number => 1234567890
fxo-channel => 1:11
name => POTS line 2
number => 0987654321
fxo-channel => 1:12

About the name and number, this is what I get here by observation :
If I make a call from channel 1:9 and the diaplan do not modify the CID
variables, the called person see "Analog Phone 1" as CID name and 9000 as
CID number.
On the other hand, if I receive a call on channel 1:11, the "channel_name"
variable raised in the diaplan would be openzap/2/1/1234567890 (I guess
here that if number is not specified, I would get openzap/2/1). Moreover
(still this is by observation) if the carrier does not send CID (to be
precise, I mean no modulation is received) the CID name and number raised
in diaplan on incoming calls are set to name and number from openzap.conf

I hope this is clear enought

Fran?ois

On Wed, 24 Feb 2010 10:24:31 +1100, Brian May
<brian at microcomaustralia.com.au> wrote:
> On 24 February 2010 04:48, Robert Hadley <robert.hadley at teotech.com>
wrote:
>> On the http://wiki.freeswitch.org/wiki/Openzap.conf_Examples wiki page,
>> is
>> there a typo in the wanpipe /usr/local/freeswitch/conf/openzap.conf
>> example
>> concerning specifying the fxo-channel vs. fxs-channel?
> 
> I agree. The first example looks correct to me; the 2nd example looks
> wrong.
> 
> See the table I created to try and explain what term to use where:
> 
> http://wiki.freeswitch.org/wiki/OpenZAP#FXO.2FFXS_Terminology
> 
> In the examples you quoted, ports 1 and 2 are extension ports, so are
> FXS ports, but should be defined as FXO ports in openzap.conf.
> 
> Ports 3 and 4 are telephone line ports, so are FXO ports, but should
> be defined as FXS ports in openzap.conf.
> 
> I have only used zaptel myself, however I suspect the same applies to
> wanpipe.



From tculjaga at gmail.com  Wed Feb 24 01:53:49 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Wed, 24 Feb 2010 10:53:49 +0100
Subject: [Freeswitch-users] Mod_h323 On openvz (64 bit)
In-Reply-To: <8ccbff061002232154u38bee079y3226a09d0f36bce6@mail.gmail.com>
References: <8ccbff061002232154u38bee079y3226a09d0f36bce6@mail.gmail.com>
Message-ID: <65d96fc81002240153h213371dbpec400d651c6550@mail.gmail.com>

On Wed, Feb 24, 2010 at 6:54 AM, Dome Charoenyost <dome at tel.co.th> wrote:

> Dear All,
> I'm testing mod_h323 follow
> http://wiki.freeswitch.org/wiki/Mod_h323
>
> FS running on openvz 64 bit (kernel 2.6.18 pve 1.5 from proxmox)
> After build mod_h323 sucessful  i found problem when load mod_h323 in FS
> CLI
>
> 2010-02-24 11:59:01.517104 [CRIT] switch_loadable_module.c:882 Error
> Loading module /opt/freeswitch/mod/mod_h323.so
> **/opt/freeswitch/mod/mod_h323.so: undefined symbol:
> _ZN14H323Connection24OnUserInputInlineRFC2833ER15OpalRFC2833Infoi**
>
> How to fix this problem ?
>


can you start FS in debug mode (set debug level into switch.conf.xml)
without mod_h323 automatic load. Than, on console run "load mod_h323"
command and paste the output here ... just want to be sure of something.

If it is a larger log ... use pastebin with the url reference here.

T.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100224/3780d703/attachment-0002.html 

From phunk0000 at hotmail.com  Wed Feb 24 05:03:42 2010
From: phunk0000 at hotmail.com (Todd)
Date: Wed, 24 Feb 2010 08:03:42 -0500
Subject: [Freeswitch-users] mod_nibblebill
In-Reply-To: <b39273531002231537k626d1698vd88549a7f59e73a@mail.gmail.com>
References: <BLU110-W28EA8BD06A601D5D85268BA7420@phx.gbl>
	<b39273531002231537k626d1698vd88549a7f59e73a@mail.gmail.com>
Message-ID: <BLU110-DS68941CBE9213A6DA338ACA7410@phx.gbl>

I am attempting to us MySQL.  I installed the spidermonkey mod, newest ODBC,
compiled FS with ODBC, configured xml's in FS.  not 100% sure I did it right
though..followed wiki directions as close as possible.  What is the best way
to verify the SQL is talking to MySQL. or perhaps the easiest way to switch
to postgresql?  Still kinda new to DB admin.  Thanks a ton.

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rupa
Schomaker
Sent: Tuesday, February 23, 2010 6:38 PM
To: freeswitch-users
Subject: Re: [Freeswitch-users] mod_nibblebill

 

what database backend are you using?  Have you verified the SQL is going to
the right database backend?  I use mod_nibblebill against postgresql w/out
problems.

On Tue, Feb 23, 2010 at 9:42 AM, Meg Stroodle <phunk0000 at hotmail.com> wrote:

Hello List!  I am trying to install mod_nibblebill on my FS installation.  I
get the following log entry form FS & nibblebill, but the database table I
setup remains unchanged.  Any help in this matter would be greatly
appreciated.  Following is an excerpt from the FS log:

 

2010-02-23 10:32:52.869149 [DEBUG] switch_rtp.c:2004 Correct ip/port
confirmed.

2010-02-23 10:32:53.077162 [DEBUG] sofia.c:3727 Channel
sofia/internal/3007 at 192.168.15.177 entering state [ready][200]

2010-02-23 10:32:53.149166 [DEBUG] switch_core_session.c:706 Send signal
sofia/internal/sip:3008 at 192.168.15.176:21828 [BREAK]

2010-02-23 10:32:53.149166 [DEBUG] switch_core_session.c:706 Send signal
sofia/internal/3007 at 192.168.15.177 [BREAK]

2010-02-23 10:32:53.149166 [DEBUG] switch_ivr_bridge.c:131
sofia/internal/sip:3008 at 192.168.15.176:21828 receive message [DISPLAY]

2010-02-23 10:32:53.169168 [DEBUG] switch_ivr_bridge.c:131
sofia/internal/3007 at 192.168.15.177 receive message [DISPLAY]

2010-02-23 10:33:21.388949 [DEBUG] mod_nibblebill.c:529 Received request via
SESSION_HEARTBEAT!

2010-02-23 10:33:21.388949 [DEBUG] mod_nibblebill.c:397 Attempting to bill
at $1.0 per minute to account 3007

2010-02-23 10:33:21.388949 [INFO] mod_nibblebill.c:447 Beginning new billing
on 3ec17ffc-c94e-46ff-bb0d-7ad0ef27bdab

2010-02-23 10:33:21.388949 [DEBUG] mod_nibblebill.c:455 28 seconds passed
since last bill time of 2010-02-23 10:32:52

2010-02-23 10:33:21.388949 [DEBUG] mod_nibblebill.c:461 Billing $0.475997 to
3007 (Call: 3ec17ffc-c94e-46ff-bb0d-7ad0ef27bdab / 0.000000 so far)

2010-02-23 10:33:51.508822 [DEBUG] mod_nibblebill.c:529 Received request via
SESSION_HEARTBEAT!

2010-02-23 10:33:51.508822 [DEBUG] mod_nibblebill.c:397 Attempting to bill
at $1.0 per minute to account 3007

2010-02-23 10:33:51.508822 [DEBUG] mod_nibblebill.c:449 Last successful
billing time was 

2010-02-23 10:33:51.508822 [DEBUG] mod_nibblebill.c:455 30 seconds passed
since last bill time of 2010-02-23 10:33:21

2010-02-23 10:33:51.508822 [DEBUG] mod_nibblebill.c:461 Billing $0.501998 to
3007 (Call: 3ec17ffc-c94e-46ff-bb0d-7ad0ef27bdab / 0.475997 so far)

2010-02-23 10:34:21.649708 [DEBUG] mod_nibblebill.c:529 Received request via
SESSION_HEARTBEAT!

2010-02-23 10:34:21.649708 [DEBUG] mod_nibblebill.c:397 Attempting to bill
at $1.0 per minute to account 3007

2010-02-23 10:34:21.649708 [DEBUG] mod_nibblebill.c:449 Last successful
billing time was 

2010-02-23 10:34:21.649708 [DEBUG] mod_nibblebill.c:455 30 seconds passed
since last bill time of 2010-02-23 10:33:51

2010-02-23 10:34:21.649708 [DEBUG] mod_nibblebill.c:461 Billing $0.502348 to
3007 (Call: 3ec17ffc-c94e-46ff-bb0d-7ad0ef27bdab / 0.977995 so far)

2010-02-23 10:34:26.254308 [NOTICE] sofia.c:329 Hangup
sofia/internal/3007 at 192.168.15.177 [CS_EXECUTE] [NORMAL_CLEARING]

2010-02-23 10:34:26.254308 [DEBUG] switch_channel.c:1912 Send signal
sofia/internal/3007 at 192.168.15.177 [KILL]

2010-02-23 10:34:26.254308 [DEBUG] switch_core_session.c:999 Send signal
sofia/internal/3007 at 192.168.15.177 [BREAK]

2010-02-23 10:34:26.254308 [DEBUG] switch_core_state_machine.c:459
sofia/internal/3007 at 192.168.15.177 thread mismatch skipping state handler.

2010-02-23 10:34:26.269008 [DEBUG] switch_ivr_bridge.c:470
sofia/internal/3007 at 192.168.15.177 ending bridge by request from read
function

2010-02-23 10:34:26.269008 [DEBUG] switch_ivr_bridge.c:563 BRIDGE THREAD
DONE [sofia/internal/3007 at 192.168.15.177]

2010-02-23 10:34:26.269008 [DEBUG] switch_ivr_bridge.c:565 Send signal
sofia/internal/sip:3008 at 192.168.15.176:21828 [BREAK]

2010-02-23 10:34:26.269008 [DEBUG] switch_ivr_bridge.c:520
sofia/internal/sip:3008 at 192.168.15.176:21828 receive message [UNBRIDGE]

2010-02-23 10:34:26.269008 [DEBUG] switch_core_session.c:645 Send signal
sofia/internal/sip:3008 at 192.168.15.176:21828 [BREAK]

2010-02-23 10:34:26.269008 [DEBUG] switch_ivr_bridge.c:563 BRIDGE THREAD
DONE [sofia/internal/sip:3008 at 192.168.15.176:21828]

2010-02-23 10:34:26.269008 [DEBUG] switch_ivr_bridge.c:565 Send signal
sofia/internal/3007 at 192.168.15.177 [BREAK]

2010-02-23 10:34:26.269008 [NOTICE] switch_ivr_bridge.c:617 Hangup
sofia/internal/sip:3008 at 192.168.15.176:21828 [CS_EXCHANGE_MEDIA]
[NORMAL_CLEARING]

2010-02-23 10:34:26.269008 [DEBUG] switch_channel.c:1912 Send signal
sofia/internal/sip:3008 at 192.168.15.176:21828 [KILL]

2010-02-23 10:34:26.269008 [DEBUG] switch_core_session.c:999 Send signal
sofia/internal/sip:3008 at 192.168.15.176:21828 [BREAK]

2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:488
(sofia/internal/sip:3008 at 192.168.15.176:21828) State HANGUP

2010-02-23 10:34:26.269008 [DEBUG] mod_sofia.c:358 Channel
sofia/internal/sip:3008 at 192.168.15.176:21828 hanging up, cause:
NORMAL_CLEARING

2010-02-23 10:34:26.269008 [DEBUG] mod_sofia.c:400 Sending BYE to
sofia/internal/sip:3008 at 192.168.15.176:21828

2010-02-23 10:34:26.269008 [DEBUG] mod_nibblebill.c:397 Attempting to bill
at $2.30 per minute to account 3008

2010-02-23 10:34:26.269008 [INFO] mod_nibblebill.c:447 Beginning new billing
on c1c42712-89e9-44c5-914f-11ed8f6aebb1

2010-02-23 10:34:26.269008 [DEBUG] mod_nibblebill.c:455 93 seconds passed
since last bill time of 2010-02-23 10:32:52

2010-02-23 10:34:26.269008 [DEBUG] mod_nibblebill.c:461 Billing $3.582436 to
3008 (Call: c1c42712-89e9-44c5-914f-11ed8f6aebb1 / 0.000000 so far)

2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:46
sofia/internal/sip:3008 at 192.168.15.176:21828 Standard HANGUP, cause:
NORMAL_CLEARING

2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:488
(sofia/internal/sip:3008 at 192.168.15.176:21828) State HANGUP going to sleep

2010-02-23 10:34:26.269008 [INFO] switch_core_session.c:1108
sofia/internal/sip:3008 at 192.168.15.176:21828 setting session heartbeat to 30
second(s).

2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:351
(sofia/internal/sip:3008 at 192.168.15.176:21828) State EXCHANGE_MEDIA going to
sleep

2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/sip:3008 at 192.168.15.176:21828) Running State Change
CS_HANGUP

2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:465
sofia/internal/sip:3008 at 192.168.15.176:21828 handler already called,
skipping state handler.

2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:333
(sofia/internal/sip:3008 at 192.168.15.176:21828) State Change CS_HANGUP ->
CS_REPORTING

2010-02-23 10:34:26.269008 [DEBUG] switch_core_session.c:999 Send signal
sofia/internal/sip:3008 at 192.168.15.176:21828 [BREAK]

2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/sip:3008 at 192.168.15.176:21828) Running State Change
CS_REPORTING

2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:579
(sofia/internal/sip:3008 at 192.168.15.176:21828) State REPORTING

2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:53
sofia/internal/sip:3008 at 192.168.15.176:21828 Standard REPORTING, cause:
NORMAL_CLEARING

2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:579
(sofia/internal/sip:3008 at 192.168.15.176:21828) State REPORTING going to
sleep

2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:327
(sofia/internal/sip:3008 at 192.168.15.176:21828) State Change CS_REPORTING ->
CS_DESTROY

2010-02-23 10:34:26.269008 [DEBUG] switch_core_session.c:999 Send signal
sofia/internal/sip:3008 at 192.168.15.176:21828 [BREAK]

2010-02-23 10:34:26.269008 [DEBUG] switch_core_session.c:1136 Session 6
(sofia/internal/sip:3008 at 192.168.15.176:21828) Locked, Waiting on external
entities

2010-02-23 10:34:26.272437 [DEBUG] switch_core_state_machine.c:348
(sofia/internal/3007 at 192.168.15.177) State EXECUTE going to sleep

2010-02-23 10:34:26.272437 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/3007 at 192.168.15.177) Running State Change CS_HANGUP

2010-02-23 10:34:26.272437 [DEBUG] switch_core_state_machine.c:488
(sofia/internal/3007 at 192.168.15.177) State HANGUP

2010-02-23 10:34:26.272437 [DEBUG] mod_sofia.c:358 Channel
sofia/internal/3007 at 192.168.15.177 hanging up, cause: NORMAL_CLEARING

2010-02-23 10:34:26.272437 [DEBUG] mod_nibblebill.c:397 Attempting to bill
at $1.0 per minute to account 3007

2010-02-23 10:34:26.272437 [DEBUG] mod_nibblebill.c:449 Last successful
billing time was 

2010-02-23 10:34:26.272437 [DEBUG] mod_nibblebill.c:455 4 seconds passed
since last bill time of 2010-02-23 10:34:21

2010-02-23 10:34:26.272437 [DEBUG] mod_nibblebill.c:461 Billing $0.077045 to
3007 (Call: 3ec17ffc-c94e-46ff-bb0d-7ad0ef27bdab / 1.480343 so far)

2010-02-23 10:34:26.272437 [DEBUG] switch_core_state_machine.c:46
sofia/internal/3007 at 192.168.15.177 Standard HANGUP, cause: NORMAL_CLEARING

 

Anyhelp getting nibblebill to connect to the database would be greatly
appreciated.  Thanks

 

 

  _____  

Hotmail: Trusted email with Microsoft's powerful SPAM protection. Sign up
now. <http://clk.atdmt.com/GBL/go/201469226/direct/01/> 


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




-- 
-Rupa

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100224/d3687737/attachment-0002.html 

From mayamatakeshi at gmail.com  Wed Feb 24 05:12:33 2010
From: mayamatakeshi at gmail.com (mayamatakeshi)
Date: Wed, 24 Feb 2010 22:12:33 +0900
Subject: [Freeswitch-users] Setting username in the header To
Message-ID: <15b9404e1002240512y47beb0afha555c24e24d9cc2d@mail.gmail.com>

Hello,
while doing a bridge or originate,
is it possible to send a username in the header To that is different from
the one in the Request-URI?
This is to interoperate with a GW that understands this as a request for
redirection (it will send a call to the PSTN with a parameter ISUP
RedirectingNumber).

br,
Takeshi
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100224/3a668e82/attachment-0002.html 

From bekelemartins at gmail.com  Wed Feb 24 05:47:03 2010
From: bekelemartins at gmail.com (Bekele Martins)
Date: Wed, 24 Feb 2010 08:47:03 -0500
Subject: [Freeswitch-users] Streaming conference audio to a website
In-Reply-To: <F373C26A-1A75-45F0-9AFB-42269F64E9F0@jerris.com>
References: <5cc9e8f31002231933na50bd6o90e0c23a6667175a@mail.gmail.com>
	<F373C26A-1A75-45F0-9AFB-42269F64E9F0@jerris.com>
Message-ID: <5cc9e8f31002240547x6c3eab8lcf53486572b16d9d@mail.gmail.com>

I'm sorry, I meant how can I stream it live, so if someone joins the stream
from the website in the middle of the conference they will not hear the
beginning of the conversation, they will hear what's being said in real
time. Is this possible?


On Wed, Feb 24, 2010 at 3:44 AM, Michael Jerris <mike at jerris.com> wrote:

> http://wiki.freeswitch.org/wiki/Mod_conference#.3Cprofiles.3E
>
> see auto-record, note the example with mod_shout
>
> http://wiki.freeswitch.org/wiki/Mod_shout
>
> On Feb 23, 2010, at 10:33 PM, Bekele Martins wrote:
>
> Hello. I'm new to Freeswitch, but I have a question.
> Is it possible to have a conference call and then stream the audio of the
> conference to my website so people can listen to it over the Internet?
> I read about mod_conference, but I couldn't find the answer to my question.
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100224/a26ace5b/attachment-0002.html 

From max.bridgewater at gmail.com  Wed Feb 24 05:53:12 2010
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Wed, 24 Feb 2010 05:53:12 -0800
Subject: [Freeswitch-users] Increasing call Volume
Message-ID: <c55d46951002240553k9bf14bak1b684096e2a09fff@mail.gmail.com>

Hi,

Is there a way to increase the call volume with FS? I'm getting a call
from Portech but with an echo. their suggestion to resolve the echo
issue is to reduce the RX Gain. But then Rx Gain  also impacts the
volume in the call received from portech. So I was wondering if there
could be way to "correct" the stream at FS level.

Thanks,
Max.



From rupa at rupa.com  Wed Feb 24 06:03:39 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Wed, 24 Feb 2010 08:03:39 -0600
Subject: [Freeswitch-users] Streaming conference audio to a website
In-Reply-To: <5cc9e8f31002240547x6c3eab8lcf53486572b16d9d@mail.gmail.com>
References: <5cc9e8f31002231933na50bd6o90e0c23a6667175a@mail.gmail.com>
	<F373C26A-1A75-45F0-9AFB-42269F64E9F0@jerris.com>
	<5cc9e8f31002240547x6c3eab8lcf53486572b16d9d@mail.gmail.com>
Message-ID: <b39273531002240603y4440307bv78ff5e49d93fdac0@mail.gmail.com>

That is exactly what Mike's suggestion would do -- live streaming.

On Wed, Feb 24, 2010 at 7:47 AM, Bekele Martins <bekelemartins at gmail.com>wrote:

> I'm sorry, I meant how can I stream it live, so if someone joins the stream
> from the website in the middle of the conference they will not hear the
> beginning of the conversation, they will hear what's being said in real
> time. Is this possible?
>
>
> On Wed, Feb 24, 2010 at 3:44 AM, Michael Jerris <mike at jerris.com> wrote:
>
>> http://wiki.freeswitch.org/wiki/Mod_conference#.3Cprofiles.3E
>>
>> see auto-record, note the example with mod_shout
>>
>> http://wiki.freeswitch.org/wiki/Mod_shout
>>
>> On Feb 23, 2010, at 10:33 PM, Bekele Martins wrote:
>>
>> Hello. I'm new to Freeswitch, but I have a question.
>> Is it possible to have a conference call and then stream the audio of the
>> conference to my website so people can listen to it over the Internet?
>> I read about mod_conference, but I couldn't find the answer to my
>> question.
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100224/6e237f42/attachment-0002.html 

From rupa at rupa.com  Wed Feb 24 06:06:24 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Wed, 24 Feb 2010 08:06:24 -0600
Subject: [Freeswitch-users] mod_nibblebill
In-Reply-To: <BLU110-DS68941CBE9213A6DA338ACA7410@phx.gbl>
References: <BLU110-W28EA8BD06A601D5D85268BA7420@phx.gbl>
	<b39273531002231537k626d1698vd88549a7f59e73a@mail.gmail.com>
	<BLU110-DS68941CBE9213A6DA338ACA7410@phx.gbl>
Message-ID: <b39273531002240606p5bc03f34vbc8e934fedfe65a3@mail.gmail.com>

I am only passingly familiar with MySQL.   There must be a way for it to log
all sql statements sent to it?

Setting up postgresql would be the same (in broad terms) as mysql.  Install
packages, create database/user/tables, populate data, configure odbc dsn,
test.

On Wed, Feb 24, 2010 at 7:03 AM, Todd <phunk0000 at hotmail.com> wrote:

>  I am attempting to us MySQL.  I installed the spidermonkey mod, newest
> ODBC, compiled FS with ODBC, configured xml?s in FS?  not 100% sure I did it
> right though..followed wiki directions as close as possible.  What is the
> best way to verify the SQL is talking to MySQL? or perhaps the easiest way
> to switch  to postgresql?  Still kinda new to DB admin.  Thanks a ton.
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Rupa
> Schomaker
> *Sent:* Tuesday, February 23, 2010 6:38 PM
> *To:* freeswitch-users
> *Subject:* Re: [Freeswitch-users] mod_nibblebill
>
>
>
> what database backend are you using?  Have you verified the SQL is going to
> the right database backend?  I use mod_nibblebill against postgresql w/out
> problems.
>
> On Tue, Feb 23, 2010 at 9:42 AM, Meg Stroodle <phunk0000 at hotmail.com>
> wrote:
>
> Hello List!  I am trying to install mod_nibblebill on my FS installation.
> I get the following log entry form FS & nibblebill, but the database table I
> setup remains unchanged.  Any help in this matter would be greatly
> appreciated.  Following is an excerpt from the FS log:
>
>
>
> 2010-02-23 10:32:52.869149 [DEBUG] switch_rtp.c:2004 Correct ip/port
> confirmed.
>
> 2010-02-23 10:32:53.077162 [DEBUG] sofia.c:3727 Channel
> sofia/internal/3007 at 192.168.15.177 entering state [ready][200]
>
> 2010-02-23 10:32:53.149166 [DEBUG] switch_core_session.c:706 Send signal
> sofia/internal/sip:3008 at 192.168.15.176:21828 [BREAK]
>
> 2010-02-23 10:32:53.149166 [DEBUG] switch_core_session.c:706 Send signal
> sofia/internal/3007 at 192.168.15.177 [BREAK]
>
> 2010-02-23 10:32:53.149166 [DEBUG] switch_ivr_bridge.c:131 sofia/internal/
> sip:3008 at 192.168.15.176:21828 receive message [DISPLAY]
>
> 2010-02-23 10:32:53.169168 [DEBUG] switch_ivr_bridge.c:131
> sofia/internal/3007 at 192.168.15.177 receive message [DISPLAY]
>
> 2010-02-23 10:33:21.388949 [DEBUG] mod_nibblebill.c:529 Received request
> via SESSION_HEARTBEAT!
>
> 2010-02-23 10:33:21.388949 [DEBUG] mod_nibblebill.c:397 Attempting to bill
> at $1.0 per minute to account 3007
>
> 2010-02-23 10:33:21.388949 [INFO] mod_nibblebill.c:447 Beginning new
> billing on 3ec17ffc-c94e-46ff-bb0d-7ad0ef27bdab
>
> 2010-02-23 10:33:21.388949 [DEBUG] mod_nibblebill.c:455 28 seconds passed
> since last bill time of 2010-02-23 10:32:52
>
> 2010-02-23 10:33:21.388949 [DEBUG] mod_nibblebill.c:461 Billing $0.475997
> to 3007 (Call: 3ec17ffc-c94e-46ff-bb0d-7ad0ef27bdab / 0.000000 so far)
>
> 2010-02-23 10:33:51.508822 [DEBUG] mod_nibblebill.c:529 Received request
> via SESSION_HEARTBEAT!
>
> 2010-02-23 10:33:51.508822 [DEBUG] mod_nibblebill.c:397 Attempting to bill
> at $1.0 per minute to account 3007
>
> 2010-02-23 10:33:51.508822 [DEBUG] mod_nibblebill.c:449 Last successful
> billing time was
>
> 2010-02-23 10:33:51.508822 [DEBUG] mod_nibblebill.c:455 30 seconds passed
> since last bill time of 2010-02-23 10:33:21
>
> 2010-02-23 10:33:51.508822 [DEBUG] mod_nibblebill.c:461 Billing $0.501998
> to 3007 (Call: 3ec17ffc-c94e-46ff-bb0d-7ad0ef27bdab / 0.475997 so far)
>
> 2010-02-23 10:34:21.649708 [DEBUG] mod_nibblebill.c:529 Received request
> via SESSION_HEARTBEAT!
>
> 2010-02-23 10:34:21.649708 [DEBUG] mod_nibblebill.c:397 Attempting to bill
> at $1.0 per minute to account 3007
>
> 2010-02-23 10:34:21.649708 [DEBUG] mod_nibblebill.c:449 Last successful
> billing time was
>
> 2010-02-23 10:34:21.649708 [DEBUG] mod_nibblebill.c:455 30 seconds passed
> since last bill time of 2010-02-23 10:33:51
>
> 2010-02-23 10:34:21.649708 [DEBUG] mod_nibblebill.c:461 Billing $0.502348
> to 3007 (Call: 3ec17ffc-c94e-46ff-bb0d-7ad0ef27bdab / 0.977995 so far)
>
> 2010-02-23 10:34:26.254308 [NOTICE] sofia.c:329 Hangup
> sofia/internal/3007 at 192.168.15.177 [CS_EXECUTE] [NORMAL_CLEARING]
>
> 2010-02-23 10:34:26.254308 [DEBUG] switch_channel.c:1912 Send signal
> sofia/internal/3007 at 192.168.15.177 [KILL]
>
> 2010-02-23 10:34:26.254308 [DEBUG] switch_core_session.c:999 Send signal
> sofia/internal/3007 at 192.168.15.177 [BREAK]
>
> 2010-02-23 10:34:26.254308 [DEBUG] switch_core_state_machine.c:459
> sofia/internal/3007 at 192.168.15.177 thread mismatch skipping state handler.
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_ivr_bridge.c:470
> sofia/internal/3007 at 192.168.15.177 ending bridge by request from read
> function
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_ivr_bridge.c:563 BRIDGE THREAD
> DONE [sofia/internal/3007 at 192.168.15.177]
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_ivr_bridge.c:565 Send signal
> sofia/internal/sip:3008 at 192.168.15.176:21828 [BREAK]
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_ivr_bridge.c:520 sofia/internal/
> sip:3008 at 192.168.15.176:21828 receive message [UNBRIDGE]
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_session.c:645 Send signal
> sofia/internal/sip:3008 at 192.168.15.176:21828 [BREAK]
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_ivr_bridge.c:563 BRIDGE THREAD
> DONE [sofia/internal/sip:3008 at 192.168.15.176:21828]
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_ivr_bridge.c:565 Send signal
> sofia/internal/3007 at 192.168.15.177 [BREAK]
>
> 2010-02-23 10:34:26.269008 [NOTICE] switch_ivr_bridge.c:617 Hangup
> sofia/internal/sip:3008 at 192.168.15.176:21828 [CS_EXCHANGE_MEDIA]
> [NORMAL_CLEARING]
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_channel.c:1912 Send signal
> sofia/internal/sip:3008 at 192.168.15.176:21828 [KILL]
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_session.c:999 Send signal
> sofia/internal/sip:3008 at 192.168.15.176:21828 [BREAK]
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:488
> (sofia/internal/sip:3008 at 192.168.15.176:21828) State HANGUP
>
> 2010-02-23 10:34:26.269008 [DEBUG] mod_sofia.c:358 Channel sofia/internal/
> sip:3008 at 192.168.15.176:21828 hanging up, cause: NORMAL_CLEARING
>
> 2010-02-23 10:34:26.269008 [DEBUG] mod_sofia.c:400 Sending BYE to
> sofia/internal/sip:3008 at 192.168.15.176:21828
>
> 2010-02-23 10:34:26.269008 [DEBUG] mod_nibblebill.c:397 Attempting to bill
> at $2.30 per minute to account 3008
>
> 2010-02-23 10:34:26.269008 [INFO] mod_nibblebill.c:447 Beginning new
> billing on c1c42712-89e9-44c5-914f-11ed8f6aebb1
>
> 2010-02-23 10:34:26.269008 [DEBUG] mod_nibblebill.c:455 93 seconds passed
> since last bill time of 2010-02-23 10:32:52
>
> 2010-02-23 10:34:26.269008 [DEBUG] mod_nibblebill.c:461 Billing $3.582436
> to 3008 (Call: c1c42712-89e9-44c5-914f-11ed8f6aebb1 / 0.000000 so far)
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:46
> sofia/internal/sip:3008 at 192.168.15.176:21828 Standard HANGUP, cause:
> NORMAL_CLEARING
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:488
> (sofia/internal/sip:3008 at 192.168.15.176:21828) State HANGUP going to sleep
>
> 2010-02-23 10:34:26.269008 [INFO] switch_core_session.c:1108
> sofia/internal/sip:3008 at 192.168.15.176:21828 setting session heartbeat to
> 30 second(s).
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:351
> (sofia/internal/sip:3008 at 192.168.15.176:21828) State EXCHANGE_MEDIA going
> to sleep
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:314
> (sofia/internal/sip:3008 at 192.168.15.176:21828) Running State Change
> CS_HANGUP
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:465
> sofia/internal/sip:3008 at 192.168.15.176:21828 handler already called,
> skipping state handler.
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:333
> (sofia/internal/sip:3008 at 192.168.15.176:21828) State Change CS_HANGUP ->
> CS_REPORTING
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_session.c:999 Send signal
> sofia/internal/sip:3008 at 192.168.15.176:21828 [BREAK]
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:314
> (sofia/internal/sip:3008 at 192.168.15.176:21828) Running State Change
> CS_REPORTING
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:579
> (sofia/internal/sip:3008 at 192.168.15.176:21828) State REPORTING
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:53
> sofia/internal/sip:3008 at 192.168.15.176:21828 Standard REPORTING, cause:
> NORMAL_CLEARING
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:579
> (sofia/internal/sip:3008 at 192.168.15.176:21828) State REPORTING going to
> sleep
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:327
> (sofia/internal/sip:3008 at 192.168.15.176:21828) State Change CS_REPORTING
> -> CS_DESTROY
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_session.c:999 Send signal
> sofia/internal/sip:3008 at 192.168.15.176:21828 [BREAK]
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_session.c:1136 Session 6
> (sofia/internal/sip:3008 at 192.168.15.176:21828) Locked, Waiting on external
> entities
>
> 2010-02-23 10:34:26.272437 [DEBUG] switch_core_state_machine.c:348 (
> sofia/internal/3007 at 192.168.15.177) State EXECUTE going to sleep
>
> 2010-02-23 10:34:26.272437 [DEBUG] switch_core_state_machine.c:314 (
> sofia/internal/3007 at 192.168.15.177) Running State Change CS_HANGUP
>
> 2010-02-23 10:34:26.272437 [DEBUG] switch_core_state_machine.c:488 (
> sofia/internal/3007 at 192.168.15.177) State HANGUP
>
> 2010-02-23 10:34:26.272437 [DEBUG] mod_sofia.c:358 Channel
> sofia/internal/3007 at 192.168.15.177 hanging up, cause: NORMAL_CLEARING
>
> 2010-02-23 10:34:26.272437 [DEBUG] mod_nibblebill.c:397 Attempting to bill
> at $1.0 per minute to account 3007
>
> 2010-02-23 10:34:26.272437 [DEBUG] mod_nibblebill.c:449 Last successful
> billing time was
>
> 2010-02-23 10:34:26.272437 [DEBUG] mod_nibblebill.c:455 4 seconds passed
> since last bill time of 2010-02-23 10:34:21
>
> 2010-02-23 10:34:26.272437 [DEBUG] mod_nibblebill.c:461 Billing $0.077045
> to 3007 (Call: 3ec17ffc-c94e-46ff-bb0d-7ad0ef27bdab / 1.480343 so far)
>
> 2010-02-23 10:34:26.272437 [DEBUG] switch_core_state_machine.c:46
> sofia/internal/3007 at 192.168.15.177 Standard HANGUP, cause: NORMAL_CLEARING
>
>
>
> Anyhelp getting nibblebill to connect to the database would be greatly
> appreciated.  Thanks
>
>
>
>
>  ------------------------------
>
> Hotmail: Trusted email with Microsoft?s powerful SPAM protection. Sign up
> now. <http://clk.atdmt.com/GBL/go/201469226/direct/01/>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100224/ab8b4cca/attachment-0002.html 

From jeff at jefflenk.com  Wed Feb 24 06:07:19 2010
From: jeff at jefflenk.com (Jeff Lenk)
Date: Wed, 24 Feb 2010 08:07:19 -0600
Subject: [Freeswitch-users] Increasing call Volume
In-Reply-To: <c55d46951002240553k9bf14bak1b684096e2a09fff@mail.gmail.com>
References: <c55d46951002240553k9bf14bak1b684096e2a09fff@mail.gmail.com>
Message-ID: <COL117-W7D89C1D1F7F15CDFBF1D3B9410@phx.gbl>


You can use: this is not the recommended way to solve this problem!

 

http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_set_audio_level

 

 

 


 
> Date: Wed, 24 Feb 2010 05:53:12 -0800
> From: max.bridgewater at gmail.com
> To: freeswitch-users at lists.freeswitch.org
> Subject: [Freeswitch-users] Increasing call Volume
> 
> Hi,
> 
> Is there a way to increase the call volume with FS? I'm getting a call
> from Portech but with an echo. their suggestion to resolve the echo
> issue is to reduce the RX Gain. But then Rx Gain also impacts the
> volume in the call received from portech. So I was wondering if there
> could be way to "correct" the stream at FS level.
> 
> Thanks,
> Max.
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
 		 	   		  
_________________________________________________________________
Hotmail: Free, trusted and rich email service.
http://clk.atdmt.com/GBL/go/201469228/direct/01/
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100224/6130ec74/attachment-0002.html 

From andrew at hijacked.us  Wed Feb 24 06:19:55 2010
From: andrew at hijacked.us (Andrew Thompson)
Date: Wed, 24 Feb 2010 09:19:55 -0500
Subject: [Freeswitch-users] Increasing call Volume
In-Reply-To: <COL117-W7D89C1D1F7F15CDFBF1D3B9410@phx.gbl>
References: <c55d46951002240553k9bf14bak1b684096e2a09fff@mail.gmail.com>
	<COL117-W7D89C1D1F7F15CDFBF1D3B9410@phx.gbl>
Message-ID: <20100224141955.GE1751@hijacked.us>

On Wed, Feb 24, 2010 at 08:07:19AM -0600, Jeff Lenk wrote:
> 
> You can use: this is not the recommended way to solve this problem!
> 
>  
> 
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_set_audio_level
>

There's also the uuid_audio API command (again, maybe not the best tool
to solve the problem).

Andrew



From max.bridgewater at gmail.com  Wed Feb 24 06:20:11 2010
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Wed, 24 Feb 2010 06:20:11 -0800
Subject: [Freeswitch-users] Increasing call Volume
In-Reply-To: <COL117-W7D89C1D1F7F15CDFBF1D3B9410@phx.gbl>
References: <c55d46951002240553k9bf14bak1b684096e2a09fff@mail.gmail.com>
	<COL117-W7D89C1D1F7F15CDFBF1D3B9410@phx.gbl>
Message-ID: <c55d46951002240620s7d95789bidc564488d580ec0e@mail.gmail.com>

Thanks jeff. so what would be the recommended way for solving this problem?

Max.

On Wed, Feb 24, 2010 at 6:07 AM, Jeff Lenk <jeff at jefflenk.com> wrote:
> You can use: this is not the recommended way to solve this problem!
>
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_set_audio_level
>
>
>
>
>
>> Date: Wed, 24 Feb 2010 05:53:12 -0800
>> From: max.bridgewater at gmail.com
>> To: freeswitch-users at lists.freeswitch.org
>> Subject: [Freeswitch-users] Increasing call Volume
>>
>> Hi,
>>
>> Is there a way to increase the call volume with FS? I'm getting a call
>> from Portech but with an echo. their suggestion to resolve the echo
>> issue is to reduce the RX Gain. But then Rx Gain also impacts the
>> volume in the call received from portech. So I was wondering if there
>> could be way to "correct" the stream at FS level.
>>
>> Thanks,
>> Max.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> ________________________________
> Hotmail: Free, trusted and rich email service. Get it now.
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From matt at webcontracts.co.uk  Wed Feb 24 06:20:29 2010
From: matt at webcontracts.co.uk (Matthew Law)
Date: Wed, 24 Feb 2010 14:20:29 -0000
Subject: [Freeswitch-users] How to debug time-based routing?
In-Reply-To: <74B270D5-E134-4221-A0FE-8275B05826A5@freeswitch.org>
References: <eb2506f03e9045c509a6974bcc4a3a49.squirrel@www.webcontracts.co.uk>
	<74B270D5-E134-4221-A0FE-8275B05826A5@freeswitch.org>
Message-ID: <a797462218a492c72e11a575534c18e0.squirrel@www.webcontracts.co.uk>


On Tue, February 23, 2010 1:00 am, Brian West wrote:
> Lets start with how about you pastebin your extension and logs... or
> better join #freeswitch on irc.freenode.net?  ;)
>
> /b

Thanks, Brian.

It turns out that Zoiper, the softphone I was using on my Mac seems to be
a little hit and miss.  If I switch to windows and use X-lite or the 3CX
softphone it works perfectly.

Matt.




From Suneel.Papineni at mettoni.com  Wed Feb 24 06:35:49 2010
From: Suneel.Papineni at mettoni.com (Suneel Papineni)
Date: Wed, 24 Feb 2010 14:35:49 -0000
Subject: [Freeswitch-users] Issue with making calls through FSComm and Build
	issues with Freeswitch 1.0.5 latest updated on 24-Feb-2010 04:05
Message-ID: <3181A30B8C35AB4AA8577B78DDF4613806810AEF@nickel.mettonigroup.com>

Hi,

 

I am trying to make FSComm work with Freeswitch (1.0.5 latest), but
failed to make any calls. FSComm is getting registered properly and I
can see SIP messages at Freeswitch. When a call is tried to make from
one FSComm to another, there is no INVITE message seen in Freeswitch and
also in wireshark traces as well. (OS environment is Windows XP 32-bit
). Could someone let me know if there is any specific config changes to
be made to FSComm to work. (I am trying initially with FSComm pre-build
binary version. Results are same).

 

 

As per previous suggestions I tried to work with the latest version,
downloaded Freeswitch 1.0.5 Latest updated on 24th February 2010 at
4:00am from "http://latest.freeswitch.org/". When I tried to build this,
it is giving 4 errors. So build is failing. Errors are as follows:

 

Error      201         error C2491: 'spandsp_stop_inband_dtmf_session' :
definition of dllimport function not allowed
d:\FS\freeswitch-1.0.5-latest24022010\freeswitch-1.0.5-20100224-0400\src
\mod\applications\mod_fax\mod_fax.c             812         mod_fax

Error      202         error C2491: 'spandsp_inband_dtmf_session' :
definition of dllimport function not allowed
d:\FS\freeswitch-1.0.5-latest24022010\freeswitch-1.0.5-20100224-0400\src
\mod\applications\mod_fax\mod_fax.c             825         mod_fax

 

 

Error      213         error LNK2019: unresolved external symbol
_sip_dig_function referenced in function _mod_sofia_load
mod_sofia.obj  mod_sofia

Error      214         fatal error LNK1120: 1 unresolved externals
D:\FS\freeswitch-1.0.5-latest24022010\freeswitch-1.0.5-20100224-0400\Deb
ug\mod\mod_sofia.dll         mod_sofia

 

I am trying to build Freeswitch for a windows 32-bit system. Tried to
build both Debug and Release versions but failed with the above errors.

 

Could someone let me know from where I need to download the latest
version (possibly without errors).

 

Thanks & Regards

Suneel


*************************************************************************
Please consider the environment before printing this e-mail
*************************************************************************
This email and any files transmitted with it are confidential and
intended solely for the use of the individual or entity to whom they
are addressed. If you have received this email in error please notify
the system manager.  http://www.mettoni.com

Mettoni Ltd
Registered in England and Wales: 4485956
9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN
*************************************************************************

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100224/e62f441c/attachment-0002.html 

From ivdreg at gmail.com  Wed Feb 24 06:52:21 2010
From: ivdreg at gmail.com (ivdreg ivdreg)
Date: Wed, 24 Feb 2010 16:52:21 +0200
Subject: [Freeswitch-users] SDP With T.38 in INVITE Problem
In-Reply-To: <5ED9E41E-DFF1-4E03-B0F8-032309EA9A61@jerris.com>
References: <b437a0e1002190242y2aa64696j91e8bc76f7ae2a92@mail.gmail.com>
	<b437a0e1002220448g3e3bda9fpa283e93a11d319f1@mail.gmail.com>
	<24736FFE-BEEF-4D34-BBA7-77B6B6AE0EBD@jerris.com>
	<b437a0e1002220711u1bfd138l8a67b4e0b024d856@mail.gmail.com>
	<93769c21002220729q5bf36eadk80726aa8c00e3bd@mail.gmail.com>
	<b437a0e1002220849k16fd87ffl5a7257d80e43b036@mail.gmail.com>
	<5ED9E41E-DFF1-4E03-B0F8-032309EA9A61@jerris.com>
Message-ID: <b437a0e1002240652m70497461g165eb8e7d4a47bca@mail.gmail.com>

Hi Michael,

Thanks for clarifying. Unfortunately we don't live in prefect world. I was
fixed that by disabling T.38 in codec negotiation and everything works fine.

Thanks Again.

2010/2/24 Michael Jerris <mike at jerris.com>

> if you want clarity on this, read the rfc for sdp offer answer.  You are
> not supposed to remove an m= line in an answer, if something is doing that,
> it is incorrect.
>
> Mike
>
> On Feb 22, 2010, at 11:49 AM, ivdreg ivdreg wrote:
>
> Hi Michael,
>
> As I said in a previous mails I know exactly what is happening.
> In working setup:
>
> ITSP ---> YATE (GW - Frontend) ---> FreeSwitch (routing server/xml_curl)
> ---> YATE (GW - SIP Interop) ---> OpenSIPS ---> Subscriber.
>
> I prefer to change YATE (GW - Frontend) and YATE (GW - SIP Interop) with
> FreeSwitch for some reasons. The problem is:
>
> INVITE comes from ITSP and goes to subscriber via OpenSIPS. INIVITE between
> FreeSwitch (routing server) and YATE (GW - SIP Interop) contains SDP:
> m=audio 21108 RTP/AVP 18 4 8 0
> c=IN IP4 10.10.1.110
> a=rtpmap:18 G729/8000
> a=rtpmap:4 G723/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:0 PCMU/8000
> m=image 21108 udptl t38
> c=IN IP4 10.10.1.110
> a=T38FaxVersion:0
> a=T38MaxBitRate:14400
> a=T38FaxUdpEC:t38UDPRedundancy
> a=T38FaxRateManagement:
> transferredTCF
>
> And reply 200 OK contains in SDP:
> *m=audio 34788 RTP/AVP 8*
> a=rtpmap:8 PCMA/8000
> a=silenceSupp:off - - - -
> a=ptime:20
>
> Reply 200 OK SDP between YATE (GW - SIP Interop) and OpenSIPS contains in
> SDP:
> *m=audio 16330 RTP/AVP 8*
> a=rtpmap:8 PCMA/8000
> a=silenceSupp:off - - - -
> a=ptime:20
> *m=image 0 udptl 19*
>
> In this case everything works fine. Line *m=image 0 udptl 19 *is removed
> by YATE.
> But same test with FreeSWITCH on the place of YATE (GW - SIP Interop) *"m=image
> 0 udptl 19" *call brakes as you can see in my first mail.
>
> I don't want to compare or discus YATE and FS functionality or something
> else. I just see difference in behavior and because I'm not a big expert
> don't know witch implementation is more accurate according standards. And
> second: Is it impossible for me to upgrade all CPE so only thing I can do is
> to fix it on server side. That is because I ask for a help.
>
>
> Thanks to all.
>
>
> 2010/2/22 Michael Jerris <mike at jerris.com>
>
>> if you want to see what is going on, crank up the debug in freeswitch and
>> sofia and you should see exactly what is going on.
>>
>> Mike
>>
>>
>> On Mon, Feb 22, 2010 at 10:11 AM, ivdreg ivdreg <ivdreg at gmail.com> wrote:
>>
>>> Hi Michael,
>>>
>>> This happens when ONLY IF initial INVITE is coming with T.38 from a GW
>>> (this is ITSP equipment and I don't know vendor) to our SIP subscribers with
>>> ATCOM ATA and IP Phone. We use now in production YATE for terminating and
>>> originating GWs to ITSPs and FS as main routing logic (backend). We want to
>>> switch YATE to FS for a GW also but we faced this problem. This not happens
>>> if initial INVITE nave no T.38 offered and later re-INVITE with T.38 with
>>> valid SDP port.
>>>
>>> Thanks
>>>
>>> 2010/2/22 Michael Jerris <mike at jerris.com>
>>>
>>>>  If the endpoint does not correctly follow the sdp o/a model its not
>>>> going to work.  This is not a "problem" with the sofia library, this is
>>>> intended behavior and what we are supposed to do.  What is the device?
>>>>
>>>> Mike
>>>>
>>>> On Feb 22, 2010, at 7:48 AM, ivdreg ivdreg wrote:
>>>>
>>>> Hi All,
>>>>
>>>> Actually while seeking the solution in internet I see some people having
>>>> this problem with sofia library. I'm not sure that SIP reply in this case
>>>> contains a valid SDP (I think that teminating endpoint is broken) but in my
>>>> opinion if we have at least one valid media type in SDP (video, audio, image
>>>> ...) call must be established. Can someone comment and/or help me with this
>>>> issue.
>>>>
>>>> Regards.
>>>>
>>>> 2010/2/19 ivdreg ivdreg <ivdreg at gmail.com>
>>>>
>>>>> Hi all,
>>>>>
>>>>> Dose someone have a problem that if there T.38 in coming from gateway
>>>>> FreeSwitch drops the call because of media error ? As I see from log only
>>>>> T.38 port is zero and SDP has also media port. Is it possible to configure
>>>>> FS to do not break a call but if media is OK.
>>>>>
>>>>> 2010-02-19 12:26:14.305244 [NOTICE] switch_channel.c:666 New Channel
>>>>> sofia/backend/XXXXXXXXXX at 10.10.1.110:7065[6cd9f634-411d-df11-99ca-003048bb99cc]
>>>>> 2010-02-19 12:26:14.305244 [DEBUG] mod_sofia.c:3369 (sofia/backend/
>>>>> XXXXXXXXXX at 10.10.1.110:7065) State Change CS_NEW -> CS_INIT
>>>>> 2010-02-19 12:26:14.305244 [DEBUG] switch_core_session.c:1018 Send
>>>>> signal sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [BREAK]
>>>>> 2010-02-19 12:26:14.305244 [DEBUG] switch_core_state_machine.c:314
>>>>> (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) Running State Change
>>>>> CS_INIT
>>>>> 2010-02-19 12:26:14.305244 [DEBUG] switch_core_state_machine.c:338
>>>>> (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State INIT
>>>>> 2010-02-19 12:26:14.305244 [DEBUG] mod_sofia.c:83 sofia/backend/
>>>>> XXXXXXXXXX at 10.10.1.110:7065 SOFIA INIT
>>>>> 2010-02-19 12:26:14.305244 [DEBUG] sofia_glue.c:1377 sofia/backend/
>>>>> XXXXXXXXXX at 10.10.1.110:7065 Patched SDP
>>>>> ---
>>>>> v=0
>>>>> o=- 3779949069 5166785777234550622 IN IP4 206.132.232.212
>>>>> s=session
>>>>> t=0 0
>>>>> m=audio 21108 RTP/AVP 18 4 8 0
>>>>> c=IN IP4 10.10.1.110
>>>>> a=rtpmap:18 G729/8000
>>>>> a=rtpmap:4 G723/8000
>>>>> a=rtpmap:8 PCMA/8000
>>>>> a=rtpmap:0 PCMU/8000
>>>>> m=image 21108 udptl t38
>>>>> c=IN IP4 10.10.1.110
>>>>> a=T38FaxVersion:0
>>>>> a=T38MaxBitRate:14400
>>>>> a=T38FaxUdpEC:t38UDPRedundancy
>>>>> a=T38FaxRateManagement:transferredTCF
>>>>>
>>>>> +++
>>>>> v=0
>>>>> o=- 3779949069 5166785777234550622 IN IP4 206.132.232.212
>>>>> s=session
>>>>> t=0 0
>>>>> m=audio 17058 RTP/AVP 18 4 8 0
>>>>> c=IN IP4 10.10.1.100
>>>>> a=rtpmap:18 G729/8000
>>>>> a=rtpmap:4 G723/8000
>>>>> a=rtpmap:8 PCMA/8000
>>>>> a=rtpmap:0 PCMU/8000
>>>>> m=image 17058 udptl t38
>>>>> c=IN IP4 10.10.1.100
>>>>> a=T38FaxVersion:0
>>>>> a=T38MaxBitRate:14400
>>>>> a=T38FaxUdpEC:t38UDPRedundancy
>>>>> a=T38FaxRateManagement:transferredTCF
>>>>>
>>>>> 2010-02-19 12:26:14.305244 [DEBUG] mod_sofia.c:117 (sofia/backend/
>>>>> XXXXXXXXXX at 10.10.1.110:7065) State Change CS_INIT -> CS_ROUTING
>>>>> ......
>>>>> 2010-02-19 12:26:21.255201 [DEBUG] sofia.c:4135 Remote SDP:
>>>>> v=0
>>>>> o=FreeSWITCH 1266548331 1266548332 IN IP4 10.10.1.110
>>>>> s=FreeSWITCH
>>>>> c=IN IP4 10.10.1.110
>>>>> t=0 0
>>>>> *m=audio 26850 RTP/AVP 8*
>>>>> a=rtpmap:8 PCMA/8000
>>>>> a=silenceSupp:off - - - -
>>>>> a=ptime:20
>>>>> *m=image 0 udptl 19*
>>>>>
>>>>> 2010-02-19 12:26:21.255201 [DEBUG] sofia.c:4124 Channel sofia/backend/
>>>>> XXXXXXXXXX at 10.10.1.110:7065 entering state [ready][200]
>>>>> 2010-02-19 12:26:21.255201 [DEBUG] switch_channel.c:2285 Send signal
>>>>> sofia/backend/YYYYYYYYYY at 10.10.1.110 [BREAK]
>>>>> 2010-02-19 12:26:21.255201 [NOTICE] sofia.c:4668 Channel
>>>>> [sofia/backend/XXXXXXXXXX at 10.10.1.110:7065] has been answered
>>>>> 2010-02-19 12:26:21.255201 [DEBUG] sofia_glue.c:2288 Set Codec
>>>>> sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 PROXY/8000 20 ms 160 samples
>>>>> *2010-02-19 12:26:21.255201 [DEBUG] sofia_glue.c:2571 PROXY AUDIO RTP
>>>>> [sofia/backend/XXXXXXXXXX at 10.10.1.110:7065] 10.10.1.100:17058->
>>>>> 10.10.1.110:0 codec: 0 ms: 20
>>>>> 2010-02-19 12:26:21.255201 [ERR] sofia_glue.c:2879 AUDIO RTP REPORTS
>>>>> ERROR: [Missing remote port]
>>>>> 2010-02-19 12:26:21.255201 [NOTICE] sofia_glue.c:2880 Hangup
>>>>> sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [CS_CONSUME_MEDIA]
>>>>> [DESTINATION_OUT_OF_ORDER]*
>>>>> 2010-02-19 12:26:21.255201 [DEBUG] switch_channel.c:2063 Send signal
>>>>> sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [KILL]
>>>>> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_session.c:1018 Send
>>>>> signal sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [BREAK]
>>>>> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:314
>>>>> (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) Running State Change
>>>>> CS_HANGUP
>>>>> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:499
>>>>> (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State HANGUP
>>>>> 2010-02-19 12:26:21.255201 [DEBUG] mod_sofia.c:411 Channel
>>>>> sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 hanging up, cause:
>>>>> DESTINATION_OUT_OF_ORDER
>>>>> 2010-02-19 12:26:21.255201 [DEBUG] mod_sofia.c:454 Sending BYE to
>>>>> sofia/backend/XXXXXXXXXX at 10.10.1.110:7065
>>>>> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:46
>>>>> sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 Standard HANGUP, cause:
>>>>> DESTINATION_OUT_OF_ORDER
>>>>> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:499
>>>>> (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State HANGUP going to
>>>>> sleep
>>>>> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:333
>>>>> (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State Change CS_HANGUP ->
>>>>> CS_REPORTING
>>>>> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_session.c:1018 Send
>>>>> signal sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [BREAK]
>>>>> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:314
>>>>> (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) Running State Change
>>>>> CS_REPORTING
>>>>> 2010-02-19 12:26:21.255201 [DEBUG] switch_ivr_originate.c:3185
>>>>> Originate Resulted in Error Cause: 27 [DESTINATION_OUT_OF_ORDER]
>>>>> 2010-02-19 12:26:21.255201 [INFO] mod_dptools.c:2353 Originate Failed.
>>>>> Cause: DESTINATION_OUT_OF_ORDER
>>>>> 2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:590
>>>>> (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State REPORTING
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100224/4555411c/attachment-0002.html 

From intralanman at freeswitch.org  Wed Feb 24 07:06:31 2010
From: intralanman at freeswitch.org (Raymond Chandler)
Date: Wed, 24 Feb 2010 10:06:31 -0500
Subject: [Freeswitch-users] SIP provider recommendation for US
	termination
In-Reply-To: <6b65470d1002231243x4268de5di655831071c9a28ab@mail.gmail.com>
References: <F56BB50F-6584-4235-91C4-54E4C6EF4867@gmail.com>
	<6b65470d1002231243x4268de5di655831071c9a28ab@mail.gmail.com>
Message-ID: <4B854077.5070600@freeswitch.org>

You might also check the freeswitch.org front page for "friends of 
freeswitch"... These are companies that help to support the FreeSWITCH 
community, so they would probably be recommended first.

-Ray

On 2/23/10 3:43 PM, William Suffill wrote:
> There is a freeswitch-biz list too. I'm sure more people are faced 
> with this issue as well so it might be a good topic for the biz list.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>    
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100224/609ba4ef/attachment-0002.html 

From msc at freeswitch.org  Wed Feb 24 08:11:59 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 24 Feb 2010 08:11:59 -0800
Subject: [Freeswitch-users] freeswitch minimum install
In-Reply-To: <8171AC69-88E7-4793-8C24-2594387C15E5@jerris.com>
References: <219A4D36FE4A42A3B40A594F960461B9@MOBILEE1705>
	<8171AC69-88E7-4793-8C24-2594387C15E5@jerris.com>
Message-ID: <87f2f3b91002240811j1c01ad31t9fe5a87b3d83794e@mail.gmail.com>

On Wed, Feb 24, 2010 at 12:14 AM, Michael Jerris <mike at jerris.com> wrote:

> no, other than manually creating that minimum conf
>
> Mike
>

Although if you want an example of a small configuration look at bkw's FS
softphone configuration:
http://svn.freeswitch.org/svn/configs/softphone/

-MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100224/f0183f6e/attachment-0002.html 

From ivanov.maxim at gmail.com  Wed Feb 24 08:13:53 2010
From: ivanov.maxim at gmail.com (Max Ivanov)
Date: Wed, 24 Feb 2010 16:13:53 +0000
Subject: [Freeswitch-users] Multiple gateways dial string and user busy
Message-ID: <fbfa7b0c1002240813x58ba8541h7fef5b94f045cbda@mail.gmail.com>

Hi all!

when I do test call from fs_cli:
originate sofia/gateway/panas110/223|sofia/gateway/panas111/223
&playaback(local_stream://moh)

If firest attempt returns USER_BUSY it tries to call via second one.
Is it normal? How can I stop calling attempts after first USER_BUSY?



From msc at freeswitch.org  Wed Feb 24 08:18:00 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 24 Feb 2010 08:18:00 -0800
Subject: [Freeswitch-users] internal/external profiles
In-Reply-To: <3c5cf5261002231627s1dfb04b2m64bf9bfeb3ff171d@mail.gmail.com>
References: <3c5cf5261002231627s1dfb04b2m64bf9bfeb3ff171d@mail.gmail.com>
Message-ID: <87f2f3b91002240818q9f95269lc2cee35e8ac60498@mail.gmail.com>

On Tue, Feb 23, 2010 at 4:27 PM, Brian May
<brian at microcomaustralia.com.au>wrote:

> Hello,
>
> Why is it recommended to use separate profiles for internal and external
> SIP?
>
> This page:
> <http://wiki.freeswitch.org/wiki/External_profile>
> suggests it is because of NAT.
>
> However this page recommends using separate profiles even if NAT is
> not an issue:
> <http://wiki.freeswitch.org/wiki/Auto_NAT>: "NOTE: It is still
> recommended that you use a second profile for your SIP providers. The
> default conf/sip_profiles/external.xml is set up specifically for use
> with providers."
>
> However I am still left uncertain what this means.
>
> Not trying to criticize here, just trying to learn.
>

No problem. Separating them has a few advantages: security, scalability, and
readability. The first one on the list is definitely the most important. If
you stuff everything in the internal profile it's easier to open yourself up
to toll fraud.
-MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100224/1dbfc9ed/attachment-0002.html 

From matt at webcontracts.co.uk  Wed Feb 24 08:24:16 2010
From: matt at webcontracts.co.uk (Matthew Law)
Date: Wed, 24 Feb 2010 16:24:16 -0000
Subject: [Freeswitch-users] Snom 300. Any good?
Message-ID: <ce2fe116d187873bee34e78b01169589.squirrel@www.webcontracts.co.uk>


I am very new to VOIP in general and after spending some time getting a
simple FS installation running on a small Xen instance, I am looking to
buy my first VOIP phone.

I don't need anything too fancy.  I have looked at quite a few and the
Snom 300 looks the most favourable so far.  I need something which has a
reasonably priced headset option and will allow me to make and answer
calls 'as' my two businesses from my home office to the FS VM which is out
on the internet.  Do people on the list have experience of this handset or
could you recommend another with similar features and headset available?

Many thanks,

Matt.




From phunk0000 at hotmail.com  Wed Feb 24 08:35:22 2010
From: phunk0000 at hotmail.com (Todd)
Date: Wed, 24 Feb 2010 11:35:22 -0500
Subject: [Freeswitch-users] mod_nibblebill
In-Reply-To: <b39273531002240606p5bc03f34vbc8e934fedfe65a3@mail.gmail.com>
References: <BLU110-W28EA8BD06A601D5D85268BA7420@phx.gbl>	<b39273531002231537k626d1698vd88549a7f59e73a@mail.gmail.com>	<BLU110-DS68941CBE9213A6DA338ACA7410@phx.gbl>
	<b39273531002240606p5bc03f34vbc8e934fedfe65a3@mail.gmail.com>
Message-ID: <BLU110-DS17A81B781D7FAF7A1D188CA7410@phx.gbl>

Yeah,  think I'm going to give it a shot from the beginning again and be
very careful about install and config.  Thanks, I will keep you posted. 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rupa
Schomaker
Sent: Wednesday, February 24, 2010 9:06 AM
To: freeswitch-users
Subject: Re: [Freeswitch-users] mod_nibblebill

 

I am only passingly familiar with MySQL.   There must be a way for it to log
all sql statements sent to it?  

 

Setting up postgresql would be the same (in broad terms) as mysql.  Install
packages, create database/user/tables, populate data, configure odbc dsn,
test.

On Wed, Feb 24, 2010 at 7:03 AM, Todd <phunk0000 at hotmail.com> wrote:

I am attempting to us MySQL.  I installed the spidermonkey mod, newest ODBC,
compiled FS with ODBC, configured xml's in FS.  not 100% sure I did it right
though..followed wiki directions as close as possible.  What is the best way
to verify the SQL is talking to MySQL. or perhaps the easiest way to switch
to postgresql?  Still kinda new to DB admin.  Thanks a ton.

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rupa
Schomaker
Sent: Tuesday, February 23, 2010 6:38 PM
To: freeswitch-users
Subject: Re: [Freeswitch-users] mod_nibblebill

 

what database backend are you using?  Have you verified the SQL is going to
the right database backend?  I use mod_nibblebill against postgresql w/out
problems.

On Tue, Feb 23, 2010 at 9:42 AM, Meg Stroodle <phunk0000 at hotmail.com> wrote:

Hello List!  I am trying to install mod_nibblebill on my FS installation.  I
get the following log entry form FS & nibblebill, but the database table I
setup remains unchanged.  Any help in this matter would be greatly
appreciated.  Following is an excerpt from the FS log:

 

2010-02-23 10:32:52.869149 [DEBUG] switch_rtp.c:2004 Correct ip/port
confirmed.

2010-02-23 10:32:53.077162 [DEBUG] sofia.c:3727 Channel
sofia/internal/3007 at 192.168.15.177 entering state [ready][200]

2010-02-23 10:32:53.149166 [DEBUG] switch_core_session.c:706 Send signal
sofia/internal/sip:3008 at 192.168.15.176:21828 [BREAK]

2010-02-23 10:32:53.149166 [DEBUG] switch_core_session.c:706 Send signal
sofia/internal/3007 at 192.168.15.177 [BREAK]

2010-02-23 10:32:53.149166 [DEBUG] switch_ivr_bridge.c:131
sofia/internal/sip:3008 at 192.168.15.176:21828 receive message [DISPLAY]

2010-02-23 10:32:53.169168 [DEBUG] switch_ivr_bridge.c:131
sofia/internal/3007 at 192.168.15.177 receive message [DISPLAY]

2010-02-23 10:33:21.388949 [DEBUG] mod_nibblebill.c:529 Received request via
SESSION_HEARTBEAT!

2010-02-23 10:33:21.388949 [DEBUG] mod_nibblebill.c:397 Attempting to bill
at $1.0 per minute to account 3007

2010-02-23 10:33:21.388949 [INFO] mod_nibblebill.c:447 Beginning new billing
on 3ec17ffc-c94e-46ff-bb0d-7ad0ef27bdab

2010-02-23 10:33:21.388949 [DEBUG] mod_nibblebill.c:455 28 seconds passed
since last bill time of 2010-02-23 10:32:52

2010-02-23 10:33:21.388949 [DEBUG] mod_nibblebill.c:461 Billing $0.475997 to
3007 (Call: 3ec17ffc-c94e-46ff-bb0d-7ad0ef27bdab / 0.000000 so far)

2010-02-23 10:33:51.508822 [DEBUG] mod_nibblebill.c:529 Received request via
SESSION_HEARTBEAT!

2010-02-23 10:33:51.508822 [DEBUG] mod_nibblebill.c:397 Attempting to bill
at $1.0 per minute to account 3007

2010-02-23 10:33:51.508822 [DEBUG] mod_nibblebill.c:449 Last successful
billing time was 

2010-02-23 10:33:51.508822 [DEBUG] mod_nibblebill.c:455 30 seconds passed
since last bill time of 2010-02-23 10:33:21

2010-02-23 10:33:51.508822 [DEBUG] mod_nibblebill.c:461 Billing $0.501998 to
3007 (Call: 3ec17ffc-c94e-46ff-bb0d-7ad0ef27bdab / 0.475997 so far)

2010-02-23 10:34:21.649708 [DEBUG] mod_nibblebill.c:529 Received request via
SESSION_HEARTBEAT!

2010-02-23 10:34:21.649708 [DEBUG] mod_nibblebill.c:397 Attempting to bill
at $1.0 per minute to account 3007

2010-02-23 10:34:21.649708 [DEBUG] mod_nibblebill.c:449 Last successful
billing time was 

2010-02-23 10:34:21.649708 [DEBUG] mod_nibblebill.c:455 30 seconds passed
since last bill time of 2010-02-23 10:33:51

2010-02-23 10:34:21.649708 [DEBUG] mod_nibblebill.c:461 Billing $0.502348 to
3007 (Call: 3ec17ffc-c94e-46ff-bb0d-7ad0ef27bdab / 0.977995 so far)

2010-02-23 10:34:26.254308 [NOTICE] sofia.c:329 Hangup
sofia/internal/3007 at 192.168.15.177 [CS_EXECUTE] [NORMAL_CLEARING]

2010-02-23 10:34:26.254308 [DEBUG] switch_channel.c:1912 Send signal
sofia/internal/3007 at 192.168.15.177 [KILL]

2010-02-23 10:34:26.254308 [DEBUG] switch_core_session.c:999 Send signal
sofia/internal/3007 at 192.168.15.177 [BREAK]

2010-02-23 10:34:26.254308 [DEBUG] switch_core_state_machine.c:459
sofia/internal/3007 at 192.168.15.177 thread mismatch skipping state handler.

2010-02-23 10:34:26.269008 [DEBUG] switch_ivr_bridge.c:470
sofia/internal/3007 at 192.168.15.177 ending bridge by request from read
function

2010-02-23 10:34:26.269008 [DEBUG] switch_ivr_bridge.c:563 BRIDGE THREAD
DONE [sofia/internal/3007 at 192.168.15.177]

2010-02-23 10:34:26.269008 [DEBUG] switch_ivr_bridge.c:565 Send signal
sofia/internal/sip:3008 at 192.168.15.176:21828 [BREAK]

2010-02-23 10:34:26.269008 [DEBUG] switch_ivr_bridge.c:520
sofia/internal/sip:3008 at 192.168.15.176:21828 receive message [UNBRIDGE]

2010-02-23 10:34:26.269008 [DEBUG] switch_core_session.c:645 Send signal
sofia/internal/sip:3008 at 192.168.15.176:21828 [BREAK]

2010-02-23 10:34:26.269008 [DEBUG] switch_ivr_bridge.c:563 BRIDGE THREAD
DONE [sofia/internal/sip:3008 at 192.168.15.176:21828]

2010-02-23 10:34:26.269008 [DEBUG] switch_ivr_bridge.c:565 Send signal
sofia/internal/3007 at 192.168.15.177 [BREAK]

2010-02-23 10:34:26.269008 [NOTICE] switch_ivr_bridge.c:617 Hangup
sofia/internal/sip:3008 at 192.168.15.176:21828 [CS_EXCHANGE_MEDIA]
[NORMAL_CLEARING]

2010-02-23 10:34:26.269008 [DEBUG] switch_channel.c:1912 Send signal
sofia/internal/sip:3008 at 192.168.15.176:21828 [KILL]

2010-02-23 10:34:26.269008 [DEBUG] switch_core_session.c:999 Send signal
sofia/internal/sip:3008 at 192.168.15.176:21828 [BREAK]

2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:488
(sofia/internal/sip:3008 at 192.168.15.176:21828) State HANGUP

2010-02-23 10:34:26.269008 [DEBUG] mod_sofia.c:358 Channel
sofia/internal/sip:3008 at 192.168.15.176:21828 hanging up, cause:
NORMAL_CLEARING

2010-02-23 10:34:26.269008 [DEBUG] mod_sofia.c:400 Sending BYE to
sofia/internal/sip:3008 at 192.168.15.176:21828

2010-02-23 10:34:26.269008 [DEBUG] mod_nibblebill.c:397 Attempting to bill
at $2.30 per minute to account 3008

2010-02-23 10:34:26.269008 [INFO] mod_nibblebill.c:447 Beginning new billing
on c1c42712-89e9-44c5-914f-11ed8f6aebb1

2010-02-23 10:34:26.269008 [DEBUG] mod_nibblebill.c:455 93 seconds passed
since last bill time of 2010-02-23 10:32:52

2010-02-23 10:34:26.269008 [DEBUG] mod_nibblebill.c:461 Billing $3.582436 to
3008 (Call: c1c42712-89e9-44c5-914f-11ed8f6aebb1 / 0.000000 so far)

2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:46
sofia/internal/sip:3008 at 192.168.15.176:21828 Standard HANGUP, cause:
NORMAL_CLEARING

2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:488
(sofia/internal/sip:3008 at 192.168.15.176:21828) State HANGUP going to sleep

2010-02-23 10:34:26.269008 [INFO] switch_core_session.c:1108
sofia/internal/sip:3008 at 192.168.15.176:21828 setting session heartbeat to 30
second(s).

2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:351
(sofia/internal/sip:3008 at 192.168.15.176:21828) State EXCHANGE_MEDIA going to
sleep

2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/sip:3008 at 192.168.15.176:21828) Running State Change
CS_HANGUP

2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:465
sofia/internal/sip:3008 at 192.168.15.176:21828 handler already called,
skipping state handler.

2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:333
(sofia/internal/sip:3008 at 192.168.15.176:21828) State Change CS_HANGUP ->
CS_REPORTING

2010-02-23 10:34:26.269008 [DEBUG] switch_core_session.c:999 Send signal
sofia/internal/sip:3008 at 192.168.15.176:21828 [BREAK]

2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/sip:3008 at 192.168.15.176:21828) Running State Change
CS_REPORTING

2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:579
(sofia/internal/sip:3008 at 192.168.15.176:21828) State REPORTING

2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:53
sofia/internal/sip:3008 at 192.168.15.176:21828 Standard REPORTING, cause:
NORMAL_CLEARING

2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:579
(sofia/internal/sip:3008 at 192.168.15.176:21828) State REPORTING going to
sleep

2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:327
(sofia/internal/sip:3008 at 192.168.15.176:21828) State Change CS_REPORTING ->
CS_DESTROY

2010-02-23 10:34:26.269008 [DEBUG] switch_core_session.c:999 Send signal
sofia/internal/sip:3008 at 192.168.15.176:21828 [BREAK]

2010-02-23 10:34:26.269008 [DEBUG] switch_core_session.c:1136 Session 6
(sofia/internal/sip:3008 at 192.168.15.176:21828) Locked, Waiting on external
entities

2010-02-23 10:34:26.272437 [DEBUG] switch_core_state_machine.c:348
(sofia/internal/3007 at 192.168.15.177) State EXECUTE going to sleep

2010-02-23 10:34:26.272437 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/3007 at 192.168.15.177) Running State Change CS_HANGUP

2010-02-23 10:34:26.272437 [DEBUG] switch_core_state_machine.c:488
(sofia/internal/3007 at 192.168.15.177) State HANGUP

2010-02-23 10:34:26.272437 [DEBUG] mod_sofia.c:358 Channel
sofia/internal/3007 at 192.168.15.177 hanging up, cause: NORMAL_CLEARING

2010-02-23 10:34:26.272437 [DEBUG] mod_nibblebill.c:397 Attempting to bill
at $1.0 per minute to account 3007

2010-02-23 10:34:26.272437 [DEBUG] mod_nibblebill.c:449 Last successful
billing time was 

2010-02-23 10:34:26.272437 [DEBUG] mod_nibblebill.c:455 4 seconds passed
since last bill time of 2010-02-23 10:34:21

2010-02-23 10:34:26.272437 [DEBUG] mod_nibblebill.c:461 Billing $0.077045 to
3007 (Call: 3ec17ffc-c94e-46ff-bb0d-7ad0ef27bdab / 1.480343 so far)

2010-02-23 10:34:26.272437 [DEBUG] switch_core_state_machine.c:46
sofia/internal/3007 at 192.168.15.177 Standard HANGUP, cause: NORMAL_CLEARING

 

Anyhelp getting nibblebill to connect to the database would be greatly
appreciated.  Thanks

 

 

  _____  

Hotmail: Trusted email with Microsoft's powerful SPAM protection. Sign up
<http://clk.atdmt.com/GBL/go/201469226/direct/01/>  now.


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




-- 
-Rupa


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




-- 
-Rupa

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100224/e99d37d3/attachment-0002.html 

From infos at madovsky.org  Wed Feb 24 09:36:39 2010
From: infos at madovsky.org (Madovsky)
Date: Wed, 24 Feb 2010 12:36:39 -0500
Subject: [Freeswitch-users] Setting username in the header To
References: <15b9404e1002240512y47beb0afha555c24e24d9cc2d@mail.gmail.com>
Message-ID: <B95749F8EE194B39A0220C8D31065444@MOBILEE1705>


  ----- Original Message ----- 
  From: mayamatakeshi 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Wednesday, February 24, 2010 8:12 AM
  Subject: [Freeswitch-users] Setting username in the header To


  Hello,
  while doing a bridge or originate,
  is it possible to send a username in the header To that is different from the one in the Request-URI?
  This is to interoperate with a GW that understands this as a request for redirection (it will send a call to the PSTN with a parameter ISUP RedirectingNumber).

  br,
  Takeshi




------------------------------------------------------------------------------


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  http://www.freeswitch.org



  Maybe with chanel variables
  on wiki
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100224/1a1cf523/attachment-0002.html 

From infos at madovsky.org  Wed Feb 24 09:40:15 2010
From: infos at madovsky.org (Madovsky)
Date: Wed, 24 Feb 2010 12:40:15 -0500
Subject: [Freeswitch-users] FScomm
References: <157FDB96FB88476FBAEC69C1309A0A5F@MOBILEE1705>,
	<COL117-W5563292C886257AFC56548B9420@phx.gbl>,
	<D64783281177440B9094B8181E55EFA7@MOBILEE1705><COL117-W429F3BED9F781A58264238B9420@phx.gbl><11F8492DFC2546598E6FDC3E538301CD@MOBILEE1705>
	<BAA7D201-6E09-4EDE-BBA0-FDFA64E38391@jerris.com>
Message-ID: <8F829C80D359455095B38C0F848E9780@MOBILEE1705>


----- Original Message ----- 
From: "Michael Jerris" <mike at jerris.com>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Wednesday, February 24, 2010 3:40 AM
Subject: Re: [Freeswitch-users] FScomm



On Feb 23, 2010, at 6:18 PM, Madovsky wrote:

>
> ----- Original Message ----- 
> From: Jeff Lenk
> To: freeswitch-users at lists.freeswitch.org
> Sent: Tuesday, February 23, 2010 5:05 PM
> Subject: Re: [Freeswitch-users] FScomm
>
>
> http://wiki.freeswitch.org/wiki/FSComm#Linux
>
> you must run those from the FSComm directory
>

?.

>
> It's what I did,
> but from FS trunk, inside fscomm directory,
> there s only
>
> account.cpp  conf                fshost.h        mainwindow.ui
> resources.qrc
> account.h    FSComm.2008.vcproj  main.cpp        mod_qsettings
> call.cpp     FSComm.pro          mainwindow.cpp  preferences
> call.h       fshost.cpp          mainwindow.h    resources
>


Read those installation instructions again and do them step by step, you 
skipped one.
_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



OOoops, I did gmake rather than qmake... not good to become old.... ;) 




From infos at madovsky.org  Wed Feb 24 09:42:38 2010
From: infos at madovsky.org (Madovsky)
Date: Wed, 24 Feb 2010 12:42:38 -0500
Subject: [Freeswitch-users] FScomm
References: <157FDB96FB88476FBAEC69C1309A0A5F@MOBILEE1705>,
	<COL117-W5563292C886257AFC56548B9420@phx.gbl>,
	<D64783281177440B9094B8181E55EFA7@MOBILEE1705><COL117-W429F3BED9F781A58264238B9420@phx.gbl><11F8492DFC2546598E6FDC3E538301CD@MOBILEE1705>
	<BAA7D201-6E09-4EDE-BBA0-FDFA64E38391@jerris.com>
Message-ID: <D262917F3F4943D49370D0CC015A7582@MOBILEE1705>


----- Original Message ----- 
From: "Michael Jerris" <mike at jerris.com>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Wednesday, February 24, 2010 3:40 AM
Subject: Re: [Freeswitch-users] FScomm



On Feb 23, 2010, at 6:18 PM, Madovsky wrote:

>
> ----- Original Message ----- 
> From: Jeff Lenk
> To: freeswitch-users at lists.freeswitch.org
> Sent: Tuesday, February 23, 2010 5:05 PM
> Subject: Re: [Freeswitch-users] FScomm
>
>
> http://wiki.freeswitch.org/wiki/FSComm#Linux
>
> you must run those from the FSComm directory
>

?.

>
> It's what I did,
> but from FS trunk, inside fscomm directory,
> there s only
>
> account.cpp  conf                fshost.h        mainwindow.ui
> resources.qrc
> account.h    FSComm.2008.vcproj  main.cpp        mod_qsettings
> call.cpp     FSComm.pro          mainwindow.cpp  preferences
> call.h       fshost.cpp          mainwindow.h    resources
>


Read those installation instructions again and do them step by step, you 
skipped one.
_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


Ok now I have

[root at node250 fscomm]# qmake
WARNING: Found potential symbol conflict of mainwindow.cpp (mainwindow.cpp) 
in SOURCES
WARNING: Found potential symbol conflict of mainwindow.h (mainwindow.h) in 
HEADERS
WARNING: Found potential symbol conflict of prefdialog.cpp 
(preferences/prefdialog.cpp) in SOURCES
WARNING: Found potential symbol conflict of prefdialog.h 
(preferences/prefdialog.h) in HEADERS
WARNING: Found potential symbol conflict of accountdialog.cpp 
(preferences/accountdialog.cpp) in SOURCES
WARNING: Found potential symbol conflict of accountdialog.h 
(preferences/accountdialog.h) in HEADERS

[root at node250 fscomm]# qmake
WARNING: Found potential symbol conflict of mainwindow.cpp (mainwindow.cpp) 
in SOURCES
WARNING: Found potential symbol conflict of mainwindow.h (mainwindow.h) in 
HEADERS
WARNING: Found potential symbol conflict of prefdialog.cpp 
(preferences/prefdialog.cpp) in SOURCES
WARNING: Found potential symbol conflict of prefdialog.h 
(preferences/prefdialog.h) in HEADERS
WARNING: Found potential symbol conflict of accountdialog.cpp 
(preferences/accountdialog.cpp) in SOURCES
WARNING: Found potential symbol conflict of accountdialog.h 
(preferences/accountdialog.h) in HEADERS
[root at node250 fscomm]# make
Makefile:278: warning: overriding commands for target `prefdialog.o'
Makefile:215: warning: ignoring old commands for target `prefdialog.o'
Makefile:285: warning: overriding commands for target `accountdialog.o'
Makefile:234: warning: ignoring old commands for target `accountdialog.o'
Makefile:320: warning: overriding commands for target `moc_prefdialog.o'
Makefile:298: warning: ignoring old commands for target `moc_prefdialog.o'
Makefile:323: warning: overriding commands for target `moc_accountdialog.o'
Makefile:307: warning: ignoring old commands for target 
`moc_accountdialog.o'
Makefile:347: warning: overriding commands for target `moc_mainwindow.cpp'
Makefile:326: warning: ignoring old commands for target `moc_mainwindow.cpp'
Makefile:350: warning: overriding commands for target 
`preferences/moc_prefdialog.cpp'
Makefile:332: warning: ignoring old commands for target 
`preferences/moc_prefdialog.cpp'
Makefile:353: warning: overriding commands for target 
`preferences/moc_accountdialog.cpp'
Makefile:341: warning: ignoring old commands for target 
`preferences/moc_accountdialog.cpp'
g++ -c -pipe -Wall -W -O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions 
 -fstack-protector --param=ssp-buffer-size=4 -m64 -mtune=generic  -DQT_NO_DEBUG 
 -DQT_SHARED -DQT_TABLET_SUPPORT -DQT_THREAD_SUPPORT -I/usr/lib64/qt-3.3/mkspecs/default 
 -I. -I../src/include -I../libs/apr/include -I../libs/libteletone/src -I/usr/lib64/qt-3.3/include 
 -o main.o main.cpp
main.cpp:31:25: error: QSplashScreen: No such file or directory
In file included from main.cpp:32:
mainwindow.h:34:23: error: QMainWindow: No such file or directory
mainwindow.h:35:28: error: QTableWidgetItem: No such file or directory
mainwindow.h:36:25: error: QSignalMapper: No such file or directory
mainwindow.h:37:27: error: QSystemTrayIcon: No such file or directory
In file included from mainwindow.h:39,
                 from main.cpp:32:
./fshost.h:32:19: error: QThread: No such file or directory
./fshost.h:33:17: error: QHash: No such file or directory
./fshost.h:34:26: error: QSharedPointer: No such file or directory
In file included from ./fshost.h:36,
                 from mainwindow.h:39,
                 from main.cpp:32:
./call.h:32:18: error: QtCore: No such file or directory
./call.h:33:19: error: QString: No such file or directory
In file included from mainwindow.h:42,
                 from main.cpp:32:
preferences/prefdialog.h:4:19: error: QDialog: No such file or directory
preferences/prefdialog.h:5:24: error: QDomDocument: No such file or 
directory
preferences/prefdialog.h:6:21: error: QSettings: No such file or directory
In file included from ./fshost.h:37,
                 from mainwindow.h:39,
                 from main.cpp:32:
./account.h:18: error: expected constructor, destructor, or type conversion 
before ?static?
In file included from mainwindow.h:39,
                 from main.cpp:32:
./fshost.h:40: error: invalid use of incomplete type ?struct QThread?
/usr/include/QtCore/qobject.h:68: error: forward declaration of ?struct 
QThread?
./fshost.h:46: error: ISO C++ forbids declaration of ?QSharedPointer? with 
no type
./fshost.h:46: error: expected ?;? before ?<? token
./fshost.h:47: error: expected `;' before ?QSharedPointer?
./fshost.h:47: error: ISO C++ forbids declaration of ?QSharedPointer? with 
no type
./fshost.h:47: error: expected ?;? before ?<? token
./fshost.h:48: error: ?QSharedPointer? was not declared in this scope
./fshost.h:48: error: template argument 1 is invalid
./fshost.h:48: error: expected unqualified-id before ?>? token
./fshost.h:49: error: ISO C++ forbids declaration of ?QSharedPointer? with 
no type
./fshost.h:49: error: expected ?;? before ?<? token
./fshost.h:50: error: ISO C++ forbids declaration of ?QSharedPointer? with 
no type
./fshost.h:50: error: expected ?;? before ?<? token
./fshost.h:51: error: ISO C++ forbids declaration of ?QSharedPointer? with 
no type
./fshost.h:51: error: expected ?;? before ?<? token
./fshost.h:52: error: ?QSharedPointer? has not been declared
./fshost.h:52: error: expected ?,? or ?...? before ?<? token
./fshost.h:60: error: ?QSharedPointer? has not been declared
./fshost.h:60: error: expected ?,? or ?...? before ?<? token
./fshost.h:61: error: ?QSharedPointer? has not been declared
./fshost.h:61: error: expected ?,? or ?...? before ?<? token
./fshost.h:62: error: ?QSharedPointer? has not been declared
./fshost.h:62: error: expected ?,? or ?...? before ?<? token
./fshost.h:63: error: ?QSharedPointer? has not been declared
./fshost.h:63: error: expected ?,? or ?...? before ?<? token
./fshost.h:64: error: ?QSharedPointer? has not been declared
./fshost.h:64: error: expected ?,? or ?...? before ?<? token
./fshost.h:65: error: ?QSharedPointer? has not been declared
./fshost.h:65: error: expected ?,? or ?...? before ?<? token
./fshost.h:66: error: ?QSharedPointer? has not been declared
./fshost.h:66: error: expected ?,? or ?...? before ?<? token
./fshost.h:67: error: ?QSharedPointer? has not been declared
./fshost.h:67: error: expected ?,? or ?...? before ?<? token
./fshost.h:71: error: ?QSharedPointer? has not been declared
./fshost.h:71: error: expected ?,? or ?...? before ?<? token
./fshost.h:78: error: ?QSharedPointer? was not declared in this scope
./fshost.h:78: error: template argument 2 is invalid
./fshost.h:78: error: expected unqualified-id before ?>? token
./fshost.h:79: error: ?QSharedPointer? was not declared in this scope
./fshost.h:79: error: template argument 2 is invalid
./fshost.h:79: error: expected unqualified-id before ?>? token
./fshost.h:80: error: field ?_bleg_uuids? has incomplete type
In file included from mainwindow.h:42,
                 from main.cpp:32:
preferences/prefdialog.h:17: error: invalid use of incomplete type ?struct 
QDialog?
/usr/include/QtGui/qwindowdefs.h:57: error: forward declaration of ?struct 
QDialog?
preferences/prefdialog.h:31: error: ISO C++ forbids declaration of 
?QSettings? with no type
preferences/prefdialog.h:31: error: expected ?;? before ?*? token
In file included from main.cpp:32:
mainwindow.h:48: error: expected class-name before ?{? token
mainwindow.h:65: error: ?QTableWidgetItem? has not been declared
mainwindow.h:71: error: ?QSharedPointer? has not been declared
mainwindow.h:71: error: expected ?,? or ?...? before ?<? token
mainwindow.h:72: error: ?QSharedPointer? has not been declared
mainwindow.h:72: error: expected ?,? or ?...? before ?<? token
mainwindow.h:73: error: ?QSharedPointer? has not been declared
mainwindow.h:73: error: expected ?,? or ?...? before ?<? token
mainwindow.h:74: error: ?QSharedPointer? has not been declared
mainwindow.h:74: error: expected ?,? or ?...? before ?<? token
mainwindow.h:75: error: ?QSharedPointer? has not been declared
mainwindow.h:75: error: expected ?,? or ?...? before ?<? token
mainwindow.h:78: error: ?QSharedPointer? has not been declared
mainwindow.h:78: error: expected ?,? or ?...? before ?<? token
mainwindow.h:79: error: ?QSharedPointer? has not been declared
mainwindow.h:79: error: expected ?,? or ?...? before ?<? token
mainwindow.h:80: error: ?QSharedPointer? has not been declared
mainwindow.h:80: error: expected ?,? or ?...? before ?<? token
mainwindow.h:81: error: ?QSystemTrayIcon? has not been declared
mainwindow.h:81: error: expected ?,? or ?...? before ?reason?
mainwindow.h:86: error: ISO C++ forbids declaration of ?QSignalMapper? with 
no type
mainwindow.h:86: error: expected ?;? before ?*? token
mainwindow.h:88: error: ISO C++ forbids declaration of ?QSystemTrayIcon? 
with no type
mainwindow.h:88: error: expected ?;? before ?*? token
main.cpp: In function ?int main(int, char**)?:
main.cpp:41: error: variable ?QPixmap image? has initializer but incomplete 
type
main.cpp:42: error: ?QSplashScreen? was not declared in this scope
main.cpp:42: error: ?splash? was not declared in this scope
main.cpp:42: error: expected type-specifier before ?QSplashScreen?
main.cpp:42: error: expected `;' before ?QSplashScreen?
main.cpp:48: error: no matching function for call to 
?QObject::connect(FSHost*, const char [9], MainWindow*, const char [8])?
/usr/include/QtCore/qobject.h:202: note: candidates are: static bool 
QObject::connect(const QObject*, const char*, const QObject*, const char*, 
Qt::ConnectionType)
/usr/include/QtCore/qobject.h:307: note:                 bool 
QObject::connect(const QObject*, const char*, const char*, 
Qt::ConnectionType) const
main.cpp:49: error: ?class FSHost? has no member named ?start?
main.cpp:41: warning: unused variable ?image?
./fshost.h: At global scope:
./fshost.h:90: warning: ?void eventHandlerCallback(switch_event_t*)? defined 
but not used
make: *** [main.o] Error 1


Any idea ?

Thx





From jerry.richards at teotech.com  Wed Feb 24 09:56:22 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Wed, 24 Feb 2010 09:56:22 -0800
Subject: [Freeswitch-users] Two Ethernet Ports, One FS Instance?
Message-ID: <EDBC42167D634005BF79FC52D8D6CB56@greyhawk.tonecommander.com>


If my server has two ethernet ports, do I need two FS instances?  Or can a
single FS instance send/receive messages through both ports using two
different sip_profiles?

Best Regards,
Jerry




From mrene_lists at avgs.ca  Wed Feb 24 09:58:29 2010
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Wed, 24 Feb 2010 12:58:29 -0500
Subject: [Freeswitch-users] Two Ethernet Ports, One FS Instance?
In-Reply-To: <EDBC42167D634005BF79FC52D8D6CB56@greyhawk.tonecommander.com>
References: <EDBC42167D634005BF79FC52D8D6CB56@greyhawk.tonecommander.com>
Message-ID: <98EA0C25-9D5A-46A4-9DF9-FDA77BA8EFFB@avgs.ca>

You can use two different sip profiles. Each profile has its own  
binding parameters for sip and rtp.

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 24-Feb-10, at 12:56 PM, Jerry Richards wrote:

>
> If my server has two ethernet ports, do I need two FS instances?  Or  
> can a
> single FS instance send/receive messages through both ports using two
> different sip_profiles?
>
> Best Regards,
> Jerry
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From ledoktre at meanie.us  Wed Feb 24 10:02:51 2010
From: ledoktre at meanie.us (Tim Streit)
Date: Wed, 24 Feb 2010 12:02:51 -0600
Subject: [Freeswitch-users] Skypiax- how to know which card to use?
In-Reply-To: <7b197bef1002191441u455e8d8an4eb03c460254c6d3@mail.gmail.com>
References: <4B7F0302.3060303@meanie.us>
	<7b197bef1002191441u455e8d8an4eb03c460254c6d3@mail.gmail.com>
Message-ID: <4B8569CB.6070804@meanie.us>

Hello,

I was writing to inquire how this skypiax update was coming along.  I 
didn't see it in the mailing list, but also since it had been nearly 1 
week, I wanted to be sure if I didn't miss the announcement.  I am very 
anxious to try this new update of the module.. It should be awesome!

Thanks,

Tim

Giovanni Maruzzelli wrote:
> before to delve in the troubleshooting, I have to say that I'm
> modifying the audio skypiax code in svn, so maybe it's just my fault
> ;).
>
> please be patient for a little while, I hope to have done with it in a
> couple days.
>
> I'll announce to the mailing list when done.
>
> In the mean time, at least one good news for you user of SkypeIn
> service: a new feature of mod_skypiax is intended to recognize the
> DTMFs coming from SkypeIn, so the incoming calls will be able to use
> ivr, voicemail, etc.



From errotan at gmail.com  Wed Feb 24 10:35:14 2010
From: errotan at gmail.com (=?iso-8859-1?q?Pusk=E1s_Zsolt?=)
Date: Wed, 24 Feb 2010 19:35:14 +0100
Subject: [Freeswitch-users] mod_fax undefined symbol
In-Reply-To: <41F9D585-2116-4219-9AF7-E8E944D43362@ttnc.co.uk>
References: <6FCE44FF-70C1-412D-B722-4C8D359381BA@ttnc.co.uk>
	<201002191904.39081.errotan@gmail.com>
	<41F9D585-2116-4219-9AF7-E8E944D43362@ttnc.co.uk>
Message-ID: <201002241935.14921.errotan@gmail.com>

2010. febru?r 21. 22.25.05 TTNC - Technical d?tummal ezt ?rta:
> Out of interest, I downgraded my versions of libtiff and libjpeg to the
>  versions shipped with Lenny:
> 
> voipin1:/opt# dpkg -l | egrep 'libtiff|libjpeg'
> ii  libjpeg62                            6b-14                    The
>  Independent JPEG Group's JPEG runtime library ii  libjpeg62-dev           
>              6b-14                    Development files for the IJG JPEG
>  library ii  libtiff4                             3.8.2-11.2              
>  Tag Image File Format (TIFF) library ii  libtiff4-dev                     
>     3.8.2-11.2               Tag Image File Format library (TIFF),
>  development files ii  libtiffxx0c2                         3.8.2-11.2     
>           Tag Image File Format (TIFF) library -- C++ interface
> 
> Everything else stayed at the 'squeeze' version.
> 
> Still didn't make any different, **/opt/freeswitch/mod/mod_fax.so:
>  undefined symbol: TIFFDefaultStripSize**
> 
> I'm guessing that points to it being a problem outside of these packages
>  and somewhere else in Debian?
> 
> Russ
> 
> On 19 Feb 2010, at 18:04, Pusk?s Zsolt wrote:
> > 2010. febru?r 19. 18.44.32 TTNC - Technical d?tummal ezt ?rta:
> >> On 19 Feb 2010, at 17:25, Pusk?s Zsolt wrote:
> >>> Just compiled svn16700 on Debian "lenny" x86, mod_fax loads and works
> >>> perfectly. I have an ongoing compile on another machine (amd64) if It
> >>> don't works i will send a mail (in 1 hour) otherwise consider it
> >>> working.
> >>
> >> How did you compile it? Using dpkg-buildpackage or via make/make
> >> install?
> >>
> >> Do you have any debian versions of libtiff4(-dev) installed?
> >
> > Okay i can confirm that mod_fax works on Debian "lenny" amd64 but it
> > don't work on Debian "testing,squeeze" amd64.
> >
> > 2010-02-19 18:49:14.610297 [CRIT] switch_loadable_module.c:882 Error
> > Loading module /usr/local/freeswitch/mod/mod_fax.so
> > **/usr/local/freeswitch/mod/mod_fax.so: undefined symbol:
> > TIFFDefaultStripSize**
> >
> > I haven't tried to compile mod_fax on testing before so i don't know what
> > is causeing the problem :(
> >
> > # ldd mod_fax.so
> >        linux-vdso.so.1 =>  (0x00007fff106f6000)
> >        libm.so.6 => /lib/libm.so.6 (0x00007f506b711000)
> >        libfreeswitch.so.1 => /usr/local/freeswitch/lib/libfreeswitch.so.1
> > (0x00007f506b345000)
> >        libpthread.so.0 => /lib/libpthread.so.0 (0x00007f506b128000)
> >        libc.so.6 => /lib/libc.so.6 (0x00007f506add4000)
> >        libssl.so.0.9.8 => /usr/lib/libssl.so.0.9.8 (0x00007f506ab82000)
> >        libcrypto.so.0.9.8 => /usr/lib/libcrypto.so.0.9.8
> > (0x00007f506a7e2000) libncurses.so.5 => /lib/libncurses.so.5
> > (0x00007f506a59d000) libstdc++.so.6 => /usr/lib/libstdc++.so.6
> > (0x00007f506a28d000) libgcc_s.so.1 => /lib/libgcc_s.so.1
> > (0x00007f506a076000)
> >        libodbc.so.1 => /usr/lib/libodbc.so.1 (0x00007f5069e17000)
> >        /lib64/ld-linux-x86-64.so.2 (0x00007f506bc31000)
> >        libdl.so.2 => /lib/libdl.so.2 (0x00007f5069c13000)
> >        libz.so.1 => /usr/lib/libz.so.1 (0x00007f50699fb000)
> >        libltdl.so.7 => /usr/lib/libltdl.so.7 (0x00007f50697f2000)
> >
> > Recently in debian "testing" libtiff4 and libjpeg is upgraded:
> > libtiff 3.9.2-3+b1
> > libjpeg62 6b-16.1
> > libjeg8 8-2.1
> >
> > Q&A:
> > Q: How did you compile it? Using dpkg-buildpackage or via make/make
> > install? A: svn-clean ./bootsrap ./configure make etc.
> >
> > Q: Do you have any debian versions of libtiff4(-dev) installed?
> > A: Yes:3.8.2-11.2
> >
> > I open a jira for this.
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 

Could you retest if it works for you now ? It seems after the update from 
debootstrap 1.0.21 to 1.0.22 it works. As ldd shows libjpeg.so.62 was not 
linked to mod_fax. (maybe that was the problem)

If you can upgrade all your packages to the latest and report back that it 
works we can close this issue.

Current ldd for me:

# ldd /usr/local/freeswitch/mod/mod_fax.so
        linux-vdso.so.1 =>  (0x00007fff96794000)
        libm.so.6 => /lib/libm.so.6 (0x00007f57921b9000)
        libfreeswitch.so.1 => /usr/local/freeswitch/lib/libfreeswitch.so.1 
(0x00007f5791ded000)
        libjpeg.so.62 => /usr/lib/libjpeg.so.62 (0x00007f5791bc9000)
        libpthread.so.0 => /lib/libpthread.so.0 (0x00007f57919ad000)
        libc.so.6 => /lib/libc.so.6 (0x00007f5791659000)
        libssl.so.0.9.8 => /usr/lib/libssl.so.0.9.8 (0x00007f5791406000)
        libcrypto.so.0.9.8 => /usr/lib/libcrypto.so.0.9.8 (0x00007f5791067000)
        libncurses.so.5 => /lib/libncurses.so.5 (0x00007f5790e22000)
        libstdc++.so.6 => /usr/lib/libstdc++.so.6 (0x00007f5790b11000)
        libgcc_s.so.1 => /lib/libgcc_s.so.1 (0x00007f57908fb000)
        libodbc.so.1 => /usr/lib/libodbc.so.1 (0x00007f579069c000)
        /lib64/ld-linux-x86-64.so.2 (0x00007f579272a000)
        libdl.so.2 => /lib/libdl.so.2 (0x00007f5790497000)
        libz.so.1 => /usr/lib/libz.so.1 (0x00007f5790280000)
        libltdl.so.7 => /usr/lib/libltdl.so.7 (0x00007f5790077000)



From christian.loeschenkohl at xpirio.com  Wed Feb 24 10:38:30 2010
From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=)
Date: Wed, 24 Feb 2010 19:38:30 +0100
Subject: [Freeswitch-users] conferences lead to high server load
Message-ID: <4B857226.10308@xpirio.com>

hi

we do experience a unusual high server load with the latest freeswitch versions.
about 50 conference users lead to a server load of over 10 - reproducible by the way.
this wans't the case until my latest trunk update.

fs version: 16714
os: debian lenny x86_64

has something substantially changed in mod_conference recently?

br

-- 
Ing. Christian L?schenkohl
Technische Leitung, Forschung & Entwicklung VoIP

xpirio
Telekommunikation & Service GmbH
Lakeside B04
9020 Klagenfurt
Austria

T  +43 (0) 5 77 11 - 1000
F  +43 (0) 5 77 11 - 1002
E  christian.loeschenkohl at xpirio.com



From anthony.minessale at gmail.com  Wed Feb 24 10:56:19 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 24 Feb 2010 12:56:19 -0600
Subject: [Freeswitch-users] mod_fax undefined symbol
In-Reply-To: <201002241935.14921.errotan@gmail.com>
References: <6FCE44FF-70C1-412D-B722-4C8D359381BA@ttnc.co.uk>
	<201002191904.39081.errotan@gmail.com>
	<41F9D585-2116-4219-9AF7-E8E944D43362@ttnc.co.uk>
	<201002241935.14921.errotan@gmail.com>
Message-ID: <191c3a031002241056k34e2a743t5427f29aa765b526@mail.gmail.com>

This issue was already fixed yesterday afternoon.
It was a libtool2 issue.


On Wed, Feb 24, 2010 at 12:35 PM, Pusk?s Zsolt <errotan at gmail.com> wrote:

> 2010. febru?r 21. 22.25.05 TTNC - Technical d?tummal ezt ?rta:
> > Out of interest, I downgraded my versions of libtiff and libjpeg to the
> >  versions shipped with Lenny:
> >
> > voipin1:/opt# dpkg -l | egrep 'libtiff|libjpeg'
> > ii  libjpeg62                            6b-14                    The
> >  Independent JPEG Group's JPEG runtime library ii  libjpeg62-dev
> >              6b-14                    Development files for the IJG JPEG
> >  library ii  libtiff4                             3.8.2-11.2
> >  Tag Image File Format (TIFF) library ii  libtiff4-dev
> >     3.8.2-11.2               Tag Image File Format library (TIFF),
> >  development files ii  libtiffxx0c2                         3.8.2-11.2
> >           Tag Image File Format (TIFF) library -- C++ interface
> >
> > Everything else stayed at the 'squeeze' version.
> >
> > Still didn't make any different, **/opt/freeswitch/mod/mod_fax.so:
> >  undefined symbol: TIFFDefaultStripSize**
> >
> > I'm guessing that points to it being a problem outside of these packages
> >  and somewhere else in Debian?
> >
> > Russ
> >
> > On 19 Feb 2010, at 18:04, Pusk?s Zsolt wrote:
> > > 2010. febru?r 19. 18.44.32 TTNC - Technical d?tummal ezt ?rta:
> > >> On 19 Feb 2010, at 17:25, Pusk?s Zsolt wrote:
> > >>> Just compiled svn16700 on Debian "lenny" x86, mod_fax loads and works
> > >>> perfectly. I have an ongoing compile on another machine (amd64) if It
> > >>> don't works i will send a mail (in 1 hour) otherwise consider it
> > >>> working.
> > >>
> > >> How did you compile it? Using dpkg-buildpackage or via make/make
> > >> install?
> > >>
> > >> Do you have any debian versions of libtiff4(-dev) installed?
> > >
> > > Okay i can confirm that mod_fax works on Debian "lenny" amd64 but it
> > > don't work on Debian "testing,squeeze" amd64.
> > >
> > > 2010-02-19 18:49:14.610297 [CRIT] switch_loadable_module.c:882 Error
> > > Loading module /usr/local/freeswitch/mod/mod_fax.so
> > > **/usr/local/freeswitch/mod/mod_fax.so: undefined symbol:
> > > TIFFDefaultStripSize**
> > >
> > > I haven't tried to compile mod_fax on testing before so i don't know
> what
> > > is causeing the problem :(
> > >
> > > # ldd mod_fax.so
> > >        linux-vdso.so.1 =>  (0x00007fff106f6000)
> > >        libm.so.6 => /lib/libm.so.6 (0x00007f506b711000)
> > >        libfreeswitch.so.1 =>
> /usr/local/freeswitch/lib/libfreeswitch.so.1
> > > (0x00007f506b345000)
> > >        libpthread.so.0 => /lib/libpthread.so.0 (0x00007f506b128000)
> > >        libc.so.6 => /lib/libc.so.6 (0x00007f506add4000)
> > >        libssl.so.0.9.8 => /usr/lib/libssl.so.0.9.8 (0x00007f506ab82000)
> > >        libcrypto.so.0.9.8 => /usr/lib/libcrypto.so.0.9.8
> > > (0x00007f506a7e2000) libncurses.so.5 => /lib/libncurses.so.5
> > > (0x00007f506a59d000) libstdc++.so.6 => /usr/lib/libstdc++.so.6
> > > (0x00007f506a28d000) libgcc_s.so.1 => /lib/libgcc_s.so.1
> > > (0x00007f506a076000)
> > >        libodbc.so.1 => /usr/lib/libodbc.so.1 (0x00007f5069e17000)
> > >        /lib64/ld-linux-x86-64.so.2 (0x00007f506bc31000)
> > >        libdl.so.2 => /lib/libdl.so.2 (0x00007f5069c13000)
> > >        libz.so.1 => /usr/lib/libz.so.1 (0x00007f50699fb000)
> > >        libltdl.so.7 => /usr/lib/libltdl.so.7 (0x00007f50697f2000)
> > >
> > > Recently in debian "testing" libtiff4 and libjpeg is upgraded:
> > > libtiff 3.9.2-3+b1
> > > libjpeg62 6b-16.1
> > > libjeg8 8-2.1
> > >
> > > Q&A:
> > > Q: How did you compile it? Using dpkg-buildpackage or via make/make
> > > install? A: svn-clean ./bootsrap ./configure make etc.
> > >
> > > Q: Do you have any debian versions of libtiff4(-dev) installed?
> > > A: Yes:3.8.2-11.2
> > >
> > > I open a jira for this.
> > >
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> Could you retest if it works for you now ? It seems after the update from
> debootstrap 1.0.21 to 1.0.22 it works. As ldd shows libjpeg.so.62 was not
> linked to mod_fax. (maybe that was the problem)
>
> If you can upgrade all your packages to the latest and report back that it
> works we can close this issue.
>
> Current ldd for me:
>
> # ldd /usr/local/freeswitch/mod/mod_fax.so
>        linux-vdso.so.1 =>  (0x00007fff96794000)
>        libm.so.6 => /lib/libm.so.6 (0x00007f57921b9000)
>         libfreeswitch.so.1 => /usr/local/freeswitch/lib/libfreeswitch.so.1
> (0x00007f5791ded000)
>        libjpeg.so.62 => /usr/lib/libjpeg.so.62 (0x00007f5791bc9000)
>        libpthread.so.0 => /lib/libpthread.so.0 (0x00007f57919ad000)
>        libc.so.6 => /lib/libc.so.6 (0x00007f5791659000)
>        libssl.so.0.9.8 => /usr/lib/libssl.so.0.9.8 (0x00007f5791406000)
>        libcrypto.so.0.9.8 => /usr/lib/libcrypto.so.0.9.8
> (0x00007f5791067000)
>        libncurses.so.5 => /lib/libncurses.so.5 (0x00007f5790e22000)
>        libstdc++.so.6 => /usr/lib/libstdc++.so.6 (0x00007f5790b11000)
>        libgcc_s.so.1 => /lib/libgcc_s.so.1 (0x00007f57908fb000)
>        libodbc.so.1 => /usr/lib/libodbc.so.1 (0x00007f579069c000)
>        /lib64/ld-linux-x86-64.so.2 (0x00007f579272a000)
>        libdl.so.2 => /lib/libdl.so.2 (0x00007f5790497000)
>        libz.so.1 => /usr/lib/libz.so.1 (0x00007f5790280000)
>        libltdl.so.7 => /usr/lib/libltdl.so.7 (0x00007f5790077000)
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100224/e6f88f89/attachment-0002.html 

From anthony.minessale at gmail.com  Wed Feb 24 10:58:38 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 24 Feb 2010 12:58:38 -0600
Subject: [Freeswitch-users] conferences lead to high server load
In-Reply-To: <4B857226.10308@xpirio.com>
References: <4B857226.10308@xpirio.com>
Message-ID: <191c3a031002241058y517e63fdte0d929b97a33f9ac@mail.gmail.com>

load average has no meaning with FS, you have to look at the CPU usage per
CPU and thread.
Are you experiencing any audio problems or are you just concerned about that
load number?

If you have a box that has trouble with timing it could cost more resources.
you can always run freeswitch -vm to use an alternate form of timing that
may not manifest into the load average.


2010/2/24 Christian L?schenkohl <christian.loeschenkohl at xpirio.com>

> hi
>
> we do experience a unusual high server load with the latest freeswitch
> versions.
> about 50 conference users lead to a server load of over 10 - reproducible
> by the way.
> this wans't the case until my latest trunk update.
>
> fs version: 16714
> os: debian lenny x86_64
>
> has something substantially changed in mod_conference recently?
>
> br
>
> --
> Ing. Christian L?schenkohl
> Technische Leitung, Forschung & Entwicklung VoIP
>
> xpirio
> Telekommunikation & Service GmbH
> Lakeside B04
> 9020 Klagenfurt
> Austria
>
> T  +43 (0) 5 77 11 - 1000
> F  +43 (0) 5 77 11 - 1002
> E  christian.loeschenkohl at xpirio.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100224/0d6fe45e/attachment-0002.html 

From anthony.minessale at gmail.com  Wed Feb 24 11:08:51 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 24 Feb 2010 13:08:51 -0600
Subject: [Freeswitch-users] Setting username in the header To
In-Reply-To: <B95749F8EE194B39A0220C8D31065444@MOBILEE1705>
References: <15b9404e1002240512y47beb0afha555c24e24d9cc2d@mail.gmail.com>
	<B95749F8EE194B39A0220C8D31065444@MOBILEE1705>
Message-ID: <191c3a031002241108q3e268e64m983005b60c196f8c@mail.gmail.com>

use the variable
{sip_invite_to_uri=<desired number>}
at the beginning of your dial string
you can either supply a full sup uri or just then number
alternatively, you can terminate your dial string with ^<desired number>


On Wed, Feb 24, 2010 at 11:36 AM, Madovsky <infos at madovsky.org> wrote:

>
>
> ----- Original Message -----
> *From:* mayamatakeshi <mayamatakeshi at gmail.com>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Wednesday, February 24, 2010 8:12 AM
> *Subject:* [Freeswitch-users] Setting username in the header To
>
> Hello,
> while doing a bridge or originate,
> is it possible to send a username in the header To that is different from
> the one in the Request-URI?
> This is to interoperate with a GW that understands this as a request for
> redirection (it will send a call to the PSTN with a parameter ISUP
> RedirectingNumber).
>
> br,
> Takeshi
>
>  ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> Maybe with chanel variables
> on wiki
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100224/c7c5a59a/attachment-0002.html 

From brian at freeswitch.org  Wed Feb 24 11:10:40 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 24 Feb 2010 13:10:40 -0600
Subject: [Freeswitch-users] Setting username in the header To
In-Reply-To: <B95749F8EE194B39A0220C8D31065444@MOBILEE1705>
References: <15b9404e1002240512y47beb0afha555c24e24d9cc2d@mail.gmail.com>
	<B95749F8EE194B39A0220C8D31065444@MOBILEE1705>
Message-ID: <237546EF-35DF-4826-A330-C7F4D3190BDF@freeswitch.org>

Example when replying to the list... Please do not echo back the full headers of possible LIKE below.

To answer your question this is possible if you set the sip_invite_to_uri (needs the full URI)

/b

On Feb 24, 2010, at 11:36 AM, Madovsky wrote:

>  
> ----- Original Message -----
> From: mayamatakeshi
> To: freeswitch-users at lists.freeswitch.org
> Sent: Wednesday, February 24, 2010 8:12 AM
> Subject: [Freeswitch-users] Setting username in the header To
> 
> Hello,
> while doing a bridge or originate,
> is it possible to send a username in the header To that is different from the one in the Request-URI?
> This is to interoperate with a GW that understands this as a request for redirection (it will send a call to the PSTN with a parameter ISUP RedirectingNumber).
> 
> br,
> Takeshi
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>  
>  
> Maybe with chanel variables
> on wiki
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100224/8f0d0669/attachment-0002.html 

From msc at freeswitch.org  Wed Feb 24 11:14:48 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 24 Feb 2010 11:14:48 -0800
Subject: [Freeswitch-users] Snom 300. Any good?
In-Reply-To: <ce2fe116d187873bee34e78b01169589.squirrel@www.webcontracts.co.uk>
References: <ce2fe116d187873bee34e78b01169589.squirrel@www.webcontracts.co.uk>
Message-ID: <87f2f3b91002241114q25ca604es6f6863fc714006d2@mail.gmail.com>

On Wed, Feb 24, 2010 at 8:24 AM, Matthew Law <matt at webcontracts.co.uk>wrote:

>
> I am very new to VOIP in general and after spending some time getting a
> simple FS installation running on a small Xen instance, I am looking to
> buy my first VOIP phone.
>
> I don't need anything too fancy.  I have looked at quite a few and the
> Snom 300 looks the most favourable so far.  I need something which has a
> reasonably priced headset option and will allow me to make and answer
> calls 'as' my two businesses from my home office to the FS VM which is out
> on the internet.  Do people on the list have experience of this handset or
> could you recommend another with similar features and headset available?
>
> Many thanks,
>
> Matt.
>
I have used this phone quite a bit. It is nothing fancy but it works. The
only complaint I've heard is that the handset volume doesn't go too high so
you might want to consider a headset.
-MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100224/ef8ca4c3/attachment-0002.html 

From anthony.minessale at gmail.com  Wed Feb 24 11:17:13 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 24 Feb 2010 13:17:13 -0600
Subject: [Freeswitch-users] big thanks to all freeswitch developers and
	contributing users
In-Reply-To: <4B83F040.7040005@xpirio.com>
References: <4B83F040.7040005@xpirio.com>
Message-ID: <191c3a031002241117r3d630ccdo82e5d2a3cb24efa9@mail.gmail.com>

Thank you,

I should frame this email =p

2010/2/23 Christian L?schenkohl <christian.loeschenkohl at xpirio.com>

> i want to say a big THANKY YOU to all contributing freeswitch community
> members.
>
> over one year has passed since i did fall in love with this project.
> it is getting better every day, one get's help and advices if needed.
> the admins do care about nearly every problem - no matter if it's big or
> small.
> i also did manage an opensource project and i wish i had done it with that
> much
> heart and intense power that i see here.
>
> i also hope that i can contribute back enough (questions, bug reports, wiki
> enhancements).
>
> wishing you all the best
> br
>
> --
> Ing. Christian L?schenkohl
> Technische Leitung, Forschung & Entwicklung VoIP
>
> xpirio
> Telekommunikation & Service GmbH
> Lakeside B04
> 9020 Klagenfurt
> Austria
>
> T  +43 (0) 5 77 11 - 1000
> F  +43 (0) 5 77 11 - 1002
> E  christian.loeschenkohl at xpirio.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100224/9ad61d47/attachment-0002.html 

From christian.loeschenkohl at xpirio.com  Wed Feb 24 11:25:49 2010
From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=)
Date: Wed, 24 Feb 2010 20:25:49 +0100
Subject: [Freeswitch-users] conferences lead to high server load
In-Reply-To: <191c3a031002241058y517e63fdte0d929b97a33f9ac@mail.gmail.com>
References: <4B857226.10308@xpirio.com>
	<191c3a031002241058y517e63fdte0d929b97a33f9ac@mail.gmail.com>
Message-ID: <4B857D3D.5080000@xpirio.com>

yes, only the high load number concerned me.
i tested participating in one of the conferences, there is no audio problem.

i'll try -vm and give feedback on this.

br



Anthony Minessale wrote:

> load average has no meaning with FS, you have to look at the CPU usage 
> per CPU and thread.
> Are you experiencing any audio problems or are you just concerned about 
> that load number?
> 
> If you have a box that has trouble with timing it could cost more resources.
> you can always run freeswitch -vm to use an alternate form of timing 
> that may not manifest into the load average.
> 
> 
> 2010/2/24 Christian L?schenkohl <christian.loeschenkohl at xpirio.com 
> <mailto:christian.loeschenkohl at xpirio.com>>
> 
>     hi
> 
>     we do experience a unusual high server load with the latest
>     freeswitch versions.
>     about 50 conference users lead to a server load of over 10 -
>     reproducible by the way.
>     this wans't the case until my latest trunk update.
> 
>     fs version: 16714
>     os: debian lenny x86_64
> 
>     has something substantially changed in mod_conference recently?
> 
>     br
> 
>     --
>     Ing. Christian L?schenkohl
>     Technische Leitung, Forschung & Entwicklung VoIP
> 
>     xpirio
>     Telekommunikation & Service GmbH
>     Lakeside B04
>     9020 Klagenfurt
>     Austria
> 
>     T  +43 (0) 5 77 11 - 1000
>     F  +43 (0) 5 77 11 - 1002
>     E  christian.loeschenkohl at xpirio.com
>     <mailto:christian.loeschenkohl at xpirio.com>
> 
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
> 
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com 
> <mailto:MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com 
> <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org 
> <mailto:sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888 
> <http://iax:guest at conference.freeswitch.org/888>
> googletalk:conf+888 at conference.freeswitch.org 
> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


-- 
Ing. Christian L?schenkohl
Technische Leitung, Forschung & Entwicklung VoIP

xpirio
Telekommunikation & Service GmbH
Lakeside B04
9020 Klagenfurt
Austria

T  +43 (0) 5 77 11 - 1000
F  +43 (0) 5 77 11 - 1002
E  christian.loeschenkohl at xpirio.com



From msc at freeswitch.org  Wed Feb 24 11:42:17 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 24 Feb 2010 11:42:17 -0800
Subject: [Freeswitch-users] Increasing call Volume
In-Reply-To: <c55d46951002240620s7d95789bidc564488d580ec0e@mail.gmail.com>
References: <c55d46951002240553k9bf14bak1b684096e2a09fff@mail.gmail.com>
	<COL117-W7D89C1D1F7F15CDFBF1D3B9410@phx.gbl>
	<c55d46951002240620s7d95789bidc564488d580ec0e@mail.gmail.com>
Message-ID: <87f2f3b91002241142k1fc8a9c9ge573b8d913d29e80@mail.gmail.com>

On Wed, Feb 24, 2010 at 6:20 AM, Max Bridgewater
<max.bridgewater at gmail.com>wrote:

> Thanks jeff. so what would be the recommended way for solving this problem?
>
> You need to know why there's echo. Just tinkering with the audio levels
might lessen the symptoms for a while but if you don't know the underlying
cause then your solution may just be like putting a bandage on a gunshot
wound. I'm not familiar with Portech... are these VoIP calls? Get a pcap and
analyze with Wireshark. Also, do you experience echo on all calls to/from
Portech? Is it only with them? Do you have multiple devices on your end to
test with?  See if you can narrow the scope a little as that might help you
figure it out.
-MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100224/53158c37/attachment-0002.html 

From infos at madovsky.org  Wed Feb 24 12:05:29 2010
From: infos at madovsky.org (Madovsky)
Date: Wed, 24 Feb 2010 15:05:29 -0500
Subject: [Freeswitch-users] qt framework link broken
Message-ID: <0CD7185862B54C1ABF67D77BA55664F7@MOBILEE1705>

Just to inform that
at the link
http://wiki.freeswitch.org/wiki/FSComm#Linux

the qt framework link is broken, so as I'm new to this emailist
I don't want to correct myself on wiki.

Regards

Franck
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100224/e5678561/attachment-0002.html 

From gmaruzz at celliax.org  Wed Feb 24 12:07:56 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Wed, 24 Feb 2010 21:07:56 +0100
Subject: [Freeswitch-users] Skypiax- how to know which card to use?
In-Reply-To: <4B8569CB.6070804@meanie.us>
References: <4B7F0302.3060303@meanie.us>
	<7b197bef1002191441u455e8d8an4eb03c460254c6d3@mail.gmail.com> 
	<4B8569CB.6070804@meanie.us>
Message-ID: <7b197bef1002241207k4da85b36l5e2e9c8e14f4fe32@mail.gmail.com>

On Wed, Feb 24, 2010 at 7:02 PM, Tim Streit <ledoktre at meanie.us> wrote:
> Hello,
>
> I was writing to inquire how this skypiax update was coming along. ?I
> didn't see it in the mailing list, but also since it had been nearly 1
> week, I wanted to be sure if I didn't miss the announcement. ?I am very
> anxious to try this new update of the module.. It should be awesome!

Hehehe, no, you've not missed the announcement.

Is taking me some more time than I was expecting.
But's arriving...
I'll post here the announcement ;)

-giovanni



>
> Thanks,
>
> Tim
>
> Giovanni Maruzzelli wrote:
>> before to delve in the troubleshooting, I have to say that I'm
>> modifying the audio skypiax code in svn, so maybe it's just my fault
>> ;).
>>
>> please be patient for a little while, I hope to have done with it in a
>> couple days.
>>
>> I'll announce to the mailing list when done.
>>
>> In the mean time, at least one good news for you user of SkypeIn
>> service: a new feature of mod_skypiax is intended to recognize the
>> DTMFs coming from SkypeIn, so the incoming calls will be able to use
>> ivr, voicemail, etc.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618



From feeswitch.ml at hez.ca  Wed Feb 24 12:17:11 2010
From: feeswitch.ml at hez.ca (Hez Ronningen)
Date: Wed, 24 Feb 2010 12:17:11 -0800
Subject: [Freeswitch-users] error loading module dingaling
References: <C1FC75D9-29E0-4233-8613-2E23551045B6@hez.ca>
Message-ID: <C622EDED-A96F-412F-B98B-021946A6347D@hez.ca>

Hello,

Sorry for reposting this, but I have dug in to this extensively and cannot find the problem.

I've dug in to the libraries to try and figure out what exact library they expect but cannot find the link with ldd. I've tried a couple other gnutls libraries with no success. I've searched the mailing lists, google, and hounded the irc channel with no results. This loading problem is happening with both the compiled from source ver and the deb installed version.

Begin forwarded message:
> Installed freeswitch on ubuntu and enabled the dingaling module but when it boots I get the following error
> 
> 2010-02-21 11:59:55.213568 [CRIT] switch_loadable_module.c:882 Error Loading module /opt/freeswitch/mod/mod_dingaling.so
> **/opt/freeswitch/mod/mod_dingaling.so: undefined symbol: gnutls_global_init**
> 
> I have the following libraries installed
> 
> ii  libgnutls-dev                        2.8.3-2                                                the GNU TLS library - development files
> ii  libgnutls26                          2.8.3-2                                                the GNU TLS library - runtime library
> 
> 
> Is there a library I am missing or an incompatibility?
> 
> Any help is much appreciated,
> Hez




From phunk0000 at hotmail.com  Wed Feb 24 13:19:59 2010
From: phunk0000 at hotmail.com (Todd)
Date: Wed, 24 Feb 2010 16:19:59 -0500
Subject: [Freeswitch-users] mod_nibblebill
In-Reply-To: <BLU110-DS17A81B781D7FAF7A1D188CA7410@phx.gbl>
References: <BLU110-W28EA8BD06A601D5D85268BA7420@phx.gbl>	<b39273531002231537k626d1698vd88549a7f59e73a@mail.gmail.com>	<BLU110-DS68941CBE9213A6DA338ACA7410@phx.gbl>	<b39273531002240606p5bc03f34vbc8e934fedfe65a3@mail.gmail.com>
	<BLU110-DS17A81B781D7FAF7A1D188CA7410@phx.gbl>
Message-ID: <BLU110-DS16A00D0320F590FC964F93A7410@phx.gbl>

Sweet, figured it out.

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Todd
Sent: Wednesday, February 24, 2010 11:35 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] mod_nibblebill

 

Yeah,  think I'm going to give it a shot from the beginning again and be
very careful about install and config.  Thanks, I will keep you posted. 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rupa
Schomaker
Sent: Wednesday, February 24, 2010 9:06 AM
To: freeswitch-users
Subject: Re: [Freeswitch-users] mod_nibblebill

 

I am only passingly familiar with MySQL.   There must be a way for it to log
all sql statements sent to it?  

 

Setting up postgresql would be the same (in broad terms) as mysql.  Install
packages, create database/user/tables, populate data, configure odbc dsn,
test.

On Wed, Feb 24, 2010 at 7:03 AM, Todd <phunk0000 at hotmail.com> wrote:

I am attempting to us MySQL.  I installed the spidermonkey mod, newest ODBC,
compiled FS with ODBC, configured xml's in FS.  not 100% sure I did it right
though..followed wiki directions as close as possible.  What is the best way
to verify the SQL is talking to MySQL. or perhaps the easiest way to switch
to postgresql?  Still kinda new to DB admin.  Thanks a ton.

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rupa
Schomaker
Sent: Tuesday, February 23, 2010 6:38 PM
To: freeswitch-users
Subject: Re: [Freeswitch-users] mod_nibblebill

 

what database backend are you using?  Have you verified the SQL is going to
the right database backend?  I use mod_nibblebill against postgresql w/out
problems.

On Tue, Feb 23, 2010 at 9:42 AM, Meg Stroodle <phunk0000 at hotmail.com> wrote:

Hello List!  I am trying to install mod_nibblebill on my FS installation.  I
get the following log entry form FS & nibblebill, but the database table I
setup remains unchanged.  Any help in this matter would be greatly
appreciated.  Following is an excerpt from the FS log:

 

2010-02-23 10:32:52.869149 [DEBUG] switch_rtp.c:2004 Correct ip/port
confirmed.

2010-02-23 10:32:53.077162 [DEBUG] sofia.c:3727 Channel
sofia/internal/3007 at 192.168.15.177 entering state [ready][200]

2010-02-23 10:32:53.149166 [DEBUG] switch_core_session.c:706 Send signal
sofia/internal/sip:3008 at 192.168.15.176:21828 [BREAK]

2010-02-23 10:32:53.149166 [DEBUG] switch_core_session.c:706 Send signal
sofia/internal/3007 at 192.168.15.177 [BREAK]

2010-02-23 10:32:53.149166 [DEBUG] switch_ivr_bridge.c:131
sofia/internal/sip:3008 at 192.168.15.176:21828 receive message [DISPLAY]

2010-02-23 10:32:53.169168 [DEBUG] switch_ivr_bridge.c:131
sofia/internal/3007 at 192.168.15.177 receive message [DISPLAY]

2010-02-23 10:33:21.388949 [DEBUG] mod_nibblebill.c:529 Received request via
SESSION_HEARTBEAT!

2010-02-23 10:33:21.388949 [DEBUG] mod_nibblebill.c:397 Attempting to bill
at $1.0 per minute to account 3007

2010-02-23 10:33:21.388949 [INFO] mod_nibblebill.c:447 Beginning new billing
on 3ec17ffc-c94e-46ff-bb0d-7ad0ef27bdab

2010-02-23 10:33:21.388949 [DEBUG] mod_nibblebill.c:455 28 seconds passed
since last bill time of 2010-02-23 10:32:52

2010-02-23 10:33:21.388949 [DEBUG] mod_nibblebill.c:461 Billing $0.475997 to
3007 (Call: 3ec17ffc-c94e-46ff-bb0d-7ad0ef27bdab / 0.000000 so far)

2010-02-23 10:33:51.508822 [DEBUG] mod_nibblebill.c:529 Received request via
SESSION_HEARTBEAT!

2010-02-23 10:33:51.508822 [DEBUG] mod_nibblebill.c:397 Attempting to bill
at $1.0 per minute to account 3007

2010-02-23 10:33:51.508822 [DEBUG] mod_nibblebill.c:449 Last successful
billing time was 

2010-02-23 10:33:51.508822 [DEBUG] mod_nibblebill.c:455 30 seconds passed
since last bill time of 2010-02-23 10:33:21

2010-02-23 10:33:51.508822 [DEBUG] mod_nibblebill.c:461 Billing $0.501998 to
3007 (Call: 3ec17ffc-c94e-46ff-bb0d-7ad0ef27bdab / 0.475997 so far)

2010-02-23 10:34:21.649708 [DEBUG] mod_nibblebill.c:529 Received request via
SESSION_HEARTBEAT!

2010-02-23 10:34:21.649708 [DEBUG] mod_nibblebill.c:397 Attempting to bill
at $1.0 per minute to account 3007

2010-02-23 10:34:21.649708 [DEBUG] mod_nibblebill.c:449 Last successful
billing time was 

2010-02-23 10:34:21.649708 [DEBUG] mod_nibblebill.c:455 30 seconds passed
since last bill time of 2010-02-23 10:33:51

2010-02-23 10:34:21.649708 [DEBUG] mod_nibblebill.c:461 Billing $0.502348 to
3007 (Call: 3ec17ffc-c94e-46ff-bb0d-7ad0ef27bdab / 0.977995 so far)

2010-02-23 10:34:26.254308 [NOTICE] sofia.c:329 Hangup
sofia/internal/3007 at 192.168.15.177 [CS_EXECUTE] [NORMAL_CLEARING]

2010-02-23 10:34:26.254308 [DEBUG] switch_channel.c:1912 Send signal
sofia/internal/3007 at 192.168.15.177 [KILL]

2010-02-23 10:34:26.254308 [DEBUG] switch_core_session.c:999 Send signal
sofia/internal/3007 at 192.168.15.177 [BREAK]

2010-02-23 10:34:26.254308 [DEBUG] switch_core_state_machine.c:459
sofia/internal/3007 at 192.168.15.177 thread mismatch skipping state handler.

2010-02-23 10:34:26.269008 [DEBUG] switch_ivr_bridge.c:470
sofia/internal/3007 at 192.168.15.177 ending bridge by request from read
function

2010-02-23 10:34:26.269008 [DEBUG] switch_ivr_bridge.c:563 BRIDGE THREAD
DONE [sofia/internal/3007 at 192.168.15.177]

2010-02-23 10:34:26.269008 [DEBUG] switch_ivr_bridge.c:565 Send signal
sofia/internal/sip:3008 at 192.168.15.176:21828 [BREAK]

2010-02-23 10:34:26.269008 [DEBUG] switch_ivr_bridge.c:520
sofia/internal/sip:3008 at 192.168.15.176:21828 receive message [UNBRIDGE]

2010-02-23 10:34:26.269008 [DEBUG] switch_core_session.c:645 Send signal
sofia/internal/sip:3008 at 192.168.15.176:21828 [BREAK]

2010-02-23 10:34:26.269008 [DEBUG] switch_ivr_bridge.c:563 BRIDGE THREAD
DONE [sofia/internal/sip:3008 at 192.168.15.176:21828]

2010-02-23 10:34:26.269008 [DEBUG] switch_ivr_bridge.c:565 Send signal
sofia/internal/3007 at 192.168.15.177 [BREAK]

2010-02-23 10:34:26.269008 [NOTICE] switch_ivr_bridge.c:617 Hangup
sofia/internal/sip:3008 at 192.168.15.176:21828 [CS_EXCHANGE_MEDIA]
[NORMAL_CLEARING]

2010-02-23 10:34:26.269008 [DEBUG] switch_channel.c:1912 Send signal
sofia/internal/sip:3008 at 192.168.15.176:21828 [KILL]

2010-02-23 10:34:26.269008 [DEBUG] switch_core_session.c:999 Send signal
sofia/internal/sip:3008 at 192.168.15.176:21828 [BREAK]

2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:488
(sofia/internal/sip:3008 at 192.168.15.176:21828) State HANGUP

2010-02-23 10:34:26.269008 [DEBUG] mod_sofia.c:358 Channel
sofia/internal/sip:3008 at 192.168.15.176:21828 hanging up, cause:
NORMAL_CLEARING

2010-02-23 10:34:26.269008 [DEBUG] mod_sofia.c:400 Sending BYE to
sofia/internal/sip:3008 at 192.168.15.176:21828

2010-02-23 10:34:26.269008 [DEBUG] mod_nibblebill.c:397 Attempting to bill
at $2.30 per minute to account 3008

2010-02-23 10:34:26.269008 [INFO] mod_nibblebill.c:447 Beginning new billing
on c1c42712-89e9-44c5-914f-11ed8f6aebb1

2010-02-23 10:34:26.269008 [DEBUG] mod_nibblebill.c:455 93 seconds passed
since last bill time of 2010-02-23 10:32:52

2010-02-23 10:34:26.269008 [DEBUG] mod_nibblebill.c:461 Billing $3.582436 to
3008 (Call: c1c42712-89e9-44c5-914f-11ed8f6aebb1 / 0.000000 so far)

2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:46
sofia/internal/sip:3008 at 192.168.15.176:21828 Standard HANGUP, cause:
NORMAL_CLEARING

2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:488
(sofia/internal/sip:3008 at 192.168.15.176:21828) State HANGUP going to sleep

2010-02-23 10:34:26.269008 [INFO] switch_core_session.c:1108
sofia/internal/sip:3008 at 192.168.15.176:21828 setting session heartbeat to 30
second(s).

2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:351
(sofia/internal/sip:3008 at 192.168.15.176:21828) State EXCHANGE_MEDIA going to
sleep

2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/sip:3008 at 192.168.15.176:21828) Running State Change
CS_HANGUP

2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:465
sofia/internal/sip:3008 at 192.168.15.176:21828 handler already called,
skipping state handler.

2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:333
(sofia/internal/sip:3008 at 192.168.15.176:21828) State Change CS_HANGUP ->
CS_REPORTING

2010-02-23 10:34:26.269008 [DEBUG] switch_core_session.c:999 Send signal
sofia/internal/sip:3008 at 192.168.15.176:21828 [BREAK]

2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/sip:3008 at 192.168.15.176:21828) Running State Change
CS_REPORTING

2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:579
(sofia/internal/sip:3008 at 192.168.15.176:21828) State REPORTING

2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:53
sofia/internal/sip:3008 at 192.168.15.176:21828 Standard REPORTING, cause:
NORMAL_CLEARING

2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:579
(sofia/internal/sip:3008 at 192.168.15.176:21828) State REPORTING going to
sleep

2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:327
(sofia/internal/sip:3008 at 192.168.15.176:21828) State Change CS_REPORTING ->
CS_DESTROY

2010-02-23 10:34:26.269008 [DEBUG] switch_core_session.c:999 Send signal
sofia/internal/sip:3008 at 192.168.15.176:21828 [BREAK]

2010-02-23 10:34:26.269008 [DEBUG] switch_core_session.c:1136 Session 6
(sofia/internal/sip:3008 at 192.168.15.176:21828) Locked, Waiting on external
entities

2010-02-23 10:34:26.272437 [DEBUG] switch_core_state_machine.c:348
(sofia/internal/3007 at 192.168.15.177) State EXECUTE going to sleep

2010-02-23 10:34:26.272437 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/3007 at 192.168.15.177) Running State Change CS_HANGUP

2010-02-23 10:34:26.272437 [DEBUG] switch_core_state_machine.c:488
(sofia/internal/3007 at 192.168.15.177) State HANGUP

2010-02-23 10:34:26.272437 [DEBUG] mod_sofia.c:358 Channel
sofia/internal/3007 at 192.168.15.177 hanging up, cause: NORMAL_CLEARING

2010-02-23 10:34:26.272437 [DEBUG] mod_nibblebill.c:397 Attempting to bill
at $1.0 per minute to account 3007

2010-02-23 10:34:26.272437 [DEBUG] mod_nibblebill.c:449 Last successful
billing time was 

2010-02-23 10:34:26.272437 [DEBUG] mod_nibblebill.c:455 4 seconds passed
since last bill time of 2010-02-23 10:34:21

2010-02-23 10:34:26.272437 [DEBUG] mod_nibblebill.c:461 Billing $0.077045 to
3007 (Call: 3ec17ffc-c94e-46ff-bb0d-7ad0ef27bdab / 1.480343 so far)

2010-02-23 10:34:26.272437 [DEBUG] switch_core_state_machine.c:46
sofia/internal/3007 at 192.168.15.177 Standard HANGUP, cause: NORMAL_CLEARING

 

Anyhelp getting nibblebill to connect to the database would be greatly
appreciated.  Thanks

 

 

  _____  

Hotmail: Trusted email with Microsoft's powerful SPAM protection. Sign up
<http://clk.atdmt.com/GBL/go/201469226/direct/01/>  now.


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




-- 
-Rupa


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




-- 
-Rupa

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100224/1122d508/attachment-0002.html 

From joseph.puchalski at personalcyberspace.com  Wed Feb 24 15:45:02 2010
From: joseph.puchalski at personalcyberspace.com (Joseph Puchalski)
Date: Wed, 24 Feb 2010 23:45:02 +0000
Subject: [Freeswitch-users] Retrieving voicemail without entering user ID
	(extension)
Message-ID: <093DD565390C1E4FB15D7B383E86BB05AFF6BC@Goose.personalcyberspace.net>

I'm trying to modify my dialplan so that I can press a single button on my phone, be connected to voicemail, and enter only a password to gain access.

Currently I use a programmable key to dial 4000. I am prompted for my ID, and then password.

I've poked around "mod voicemail" on the wiki and searched the mailing list and web, but haven't found enough info. I have discovered that this behavior seems to have been available in previous versions of the default dialplan.

Is it still possible? Is it advisable? Was this feature/behavior removed for security reasons?

I apologize ahead of time if the answer is somewhere in plain sight that I haven't looked yet. If so, I'd much appreciate being pointed in the right direction.

As always, thanks for any help,

Joe P.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100224/9ab0a2a9/attachment-0002.html 

From rupa at rupa.com  Wed Feb 24 16:01:15 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Wed, 24 Feb 2010 18:01:15 -0600
Subject: [Freeswitch-users] Retrieving voicemail without entering user
	ID (extension)
In-Reply-To: <093DD565390C1E4FB15D7B383E86BB05AFF6BC@Goose.personalcyberspace.net>
References: <093DD565390C1E4FB15D7B383E86BB05AFF6BC@Goose.personalcyberspace.net>
Message-ID: <b39273531002241601m273ff333y2261d930dbcb3fa7@mail.gmail.com>

Look at the end of:

http://wiki.freeswitch.org/wiki/Mod_voicemail#Check_Voice_Mail

<http://wiki.freeswitch.org/wiki/Mod_voicemail#Check_Voice_Mail>Advisable?
 With it enabled, I can walk up to anyone's phone and retrieve their VM
w/out authentication.  It was removed on purpose due to that reason as far
as I remember.

On Wed, Feb 24, 2010 at 5:45 PM, Joseph Puchalski <
joseph.puchalski at personalcyberspace.com> wrote:

>  I?m trying to modify my dialplan so that I can press a single button on
> my phone, be connected to voicemail, and enter only a password to gain
> access.
>
>
>
> Currently I use a programmable key to dial 4000. I am prompted for my ID,
> and then password.
>
>
>
> I?ve poked around ?mod voicemail? on the wiki and searched the mailing list
> and web, but haven?t found enough info. I have discovered that this behavior
> seems to have been available in previous versions of the default dialplan.
>
>
>
> Is it still possible? Is it advisable? Was this feature/behavior removed
> for security reasons?
>
>
>
> I apologize ahead of time if the answer is somewhere in plain sight that I
> haven?t looked yet. If so, I?d much appreciate being pointed in the right
> direction.
>
>
>
> As always, thanks for any help,
>
>
>
> Joe P.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100224/42eddbfa/attachment-0002.html 

From lists at redbonez.net  Wed Feb 24 17:10:23 2010
From: lists at redbonez.net (Adam Ford)
Date: Wed, 24 Feb 2010 18:10:23 -0700
Subject: [Freeswitch-users] Retrieving voicemail without entering
	user	ID (extension)
In-Reply-To: <b39273531002241601m273ff333y2261d930dbcb3fa7@mail.gmail.com>
References: <093DD565390C1E4FB15D7B383E86BB05AFF6BC@Goose.personalcyberspace.net>
	<b39273531002241601m273ff333y2261d930dbcb3fa7@mail.gmail.com>
Message-ID: <012c01cab5b7$5052d990$f0f88cb0$@net>

>From reading that wiki article it seems to me that the key to achieving the
functionality you are looking for would simply be a matter of adding the
desired extension to the end of the default action (where the $1 is):

 

<action application="voicemail" data="check $${voicemail_profile} $${domain}
$1"/>

 

If I am reading it correctly, this should bypass having to enter a mailbox
ID, but still require your voicemail password. Off the top of my head, you
could probably achieve this by replacing the $1 with a variable storing the
extension which called 4000.  I would have to look it up to see if there is
a system variable for that or if you would have to assign a custom one. I am
still relatively new to FreeSWITCH myself.

 

-Adam

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rupa
Schomaker
Sent: Wednesday, February 24, 2010 5:01 PM
To: freeswitch-users
Subject: Re: [Freeswitch-users] Retrieving voicemail without entering user
ID (extension)

 

Look at the end of:

 

http://wiki.freeswitch.org/wiki/Mod_voicemail#Check_Voice_Mail

 

Advisable?  With it enabled, I can walk up to anyone's phone and retrieve
their VM w/out authentication.  It was removed on purpose due to that reason
as far as I remember.

On Wed, Feb 24, 2010 at 5:45 PM, Joseph Puchalski
<joseph.puchalski at personalcyberspace.com> wrote:

I'm trying to modify my dialplan so that I can press a single button on my
phone, be connected to voicemail, and enter only a password to gain access. 

 

Currently I use a programmable key to dial 4000. I am prompted for my ID,
and then password.

 

I've poked around "mod voicemail" on the wiki and searched the mailing list
and web, but haven't found enough info. I have discovered that this behavior
seems to have been available in previous versions of the default dialplan.

 

Is it still possible? Is it advisable? Was this feature/behavior removed for
security reasons?

 

I apologize ahead of time if the answer is somewhere in plain sight that I
haven't looked yet. If so, I'd much appreciate being pointed in the right
direction.

 

As always, thanks for any help,

 

Joe P.


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




-- 
-Rupa

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100224/4fd70554/attachment-0002.html 

From mayamatakeshi at gmail.com  Wed Feb 24 17:14:14 2010
From: mayamatakeshi at gmail.com (mayamatakeshi)
Date: Thu, 25 Feb 2010 10:14:14 +0900
Subject: [Freeswitch-users] Setting username in the header To
In-Reply-To: <191c3a031002241108q3e268e64m983005b60c196f8c@mail.gmail.com>
References: <15b9404e1002240512y47beb0afha555c24e24d9cc2d@mail.gmail.com>
	<B95749F8EE194B39A0220C8D31065444@MOBILEE1705>
	<191c3a031002241108q3e268e64m983005b60c196f8c@mail.gmail.com>
Message-ID: <15b9404e1002241714w782cd54i10d3c4c74626c0cc@mail.gmail.com>

Thanks,
I have added an entry for it in the wiki:

http://wiki.freeswitch.org/wiki/Channel_Variables#sip_invite_to_uri


On Thu, Feb 25, 2010 at 4:08 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> use the variable
> {sip_invite_to_uri=<desired number>}
> at the beginning of your dial string
> you can either supply a full sup uri or just then number
> alternatively, you can terminate your dial string with ^<desired number>
>
>
>
> On Wed, Feb 24, 2010 at 11:36 AM, Madovsky <infos at madovsky.org> wrote:
>
>>
>>
>> ----- Original Message -----
>>  *From:* mayamatakeshi <mayamatakeshi at gmail.com>
>> *To:* freeswitch-users at lists.freeswitch.org
>> *Sent:* Wednesday, February 24, 2010 8:12 AM
>> *Subject:* [Freeswitch-users] Setting username in the header To
>>
>> Hello,
>> while doing a bridge or originate,
>> is it possible to send a username in the header To that is different from
>> the one in the Request-URI?
>> This is to interoperate with a GW that understands this as a request for
>> redirection (it will send a call to the PSTN with a parameter ISUP
>> RedirectingNumber).
>>
>> br,
>> Takeshi
>>
>>  ------------------------------
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> Maybe with chanel variables
>> on wiki
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100225/f2c8e09d/attachment-0002.html 

From brian at freeswitch.org  Wed Feb 24 17:20:57 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 24 Feb 2010 19:20:57 -0600
Subject: [Freeswitch-users] Retrieving voicemail without entering
	user	ID (extension)
In-Reply-To: <012c01cab5b7$5052d990$f0f88cb0$@net>
References: <093DD565390C1E4FB15D7B383E86BB05AFF6BC@Goose.personalcyberspace.net>
	<b39273531002241601m273ff333y2261d930dbcb3fa7@mail.gmail.com>
	<012c01cab5b7$5052d990$f0f88cb0$@net>
Message-ID: <0180F807-95D2-4F1A-9F3A-679795053EA2@freeswitch.org>

You are 100% correct.

/b

On Feb 24, 2010, at 7:10 PM, Adam Ford wrote:

> From reading that wiki article it seems to me that the key to achieving the functionality you are looking for would simply be a matter of adding the desired extension to the end of the default action (where the $1 is):
>  
> <action application="voicemail" data="check $${voicemail_profile} $${domain} $1"/>
>  
> If I am reading it correctly, this should bypass having to enter a mailbox ID, but still require your voicemail password. Off the top of my head, you could probably achieve this by replacing the $1 with a variable storing the extension which called 4000.  I would have to look it up to see if there is a system variable for that or if you would have to assign a custom one. I am still relatively new to FreeSWITCH myself.
>  
> -Adam

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100224/7a7be836/attachment-0002.html 

From larclap at yahoo.com  Wed Feb 24 19:08:45 2010
From: larclap at yahoo.com (Lars Zeb)
Date: Wed, 24 Feb 2010 19:08:45 -0800
Subject: [Freeswitch-users] Retrieving voicemail without entering
	user	ID (extension)
In-Reply-To: <b39273531002241601m273ff333y2261d930dbcb3fa7@mail.gmail.com>
References: <093DD565390C1E4FB15D7B383E86BB05AFF6BC@Goose.personalcyberspace.net>
	<b39273531002241601m273ff333y2261d930dbcb3fa7@mail.gmail.com>
Message-ID: <000f01cab5c7$d7e292f0$87a7b8d0$@com>

Joe,

 

I used the extension below, but I think that Brian said it was too insecure.
Being a total beginner, I removed the condition. 

 

    <extension name="Local_Extension_Abe">

      <condition field="destination_number" expression="^(101[0-9])$">

        <action application="set" data="dialed_ext=$1"/>

        <action application="export" data="dialed_ext=$1"/>

      </condition>

      <condition field="destination_number"
expression="^${caller_id_number}$">

        <action application="set"
data="voicemail_authorized=${sip_authorized}"/>

        <action application="answer"/>

        <action application="sleep" data="1000"/>

        <action application="voicemail" data="check default $${domain}
${dialed_ext}"/>

        <anti-action application="bind_meta_app" data="1 a a
execute_extension::dx XML features"/>

        <anti-action application="bind_meta_app" data="2 a a
record_session::$${base_dir}/recordings/${caller_id_number}.${strftime(%Y-%m
-%d-%H-%M-%S)}.wav"/>

        <anti-action application="bind_meta_app" data="3 a a
execute_extension::cf XML features"/>

        <anti-action application="set" data="transfer_ringback=${us-ring}"/>

        <anti-action application="set" data="call_timeout=15"/>

        <anti-action application="set"
data="sip_exclude_contact=${network_addr}"/>

        <anti-action application="set" data="hangup_after_bridge=true"/>

        <!--<anti-action application="set"
data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,BUSY,USER_BUSY,NO_ANSWER,TIM
EOUT,NO_ROUTE_DESTINATION"/> -->

        <anti-action application="set" data="continue_on_fail=true"/>

        <anti-action application="db"
data="insert/call_return/${dialed_ext}/${caller_id_number}"/>

        <anti-action application="db"
data="insert/last_dial_ext/${dialed_ext}/${uuid}"/>

        <anti-action application="bridge"
data="user/${dialed_ext}@$${domain}"/>

        <anti-action application="answer"/>

        <anti-action application="sleep" data="1000"/>

        <anti-action application="voicemail" data="default $${domain}
${dialed_ext}"/>

      </condition>

    </extension>

 

Lars

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rupa
Schomaker
Sent: Wednesday, February 24, 2010 4:01 PM
To: freeswitch-users
Subject: Re: [Freeswitch-users] Retrieving voicemail without entering user
ID (extension)

 

Look at the end of:

 

http://wiki.freeswitch.org/wiki/Mod_voicemail#Check_Voice_Mail

 

Advisable?  With it enabled, I can walk up to anyone's phone and retrieve
their VM w/out authentication.  It was removed on purpose due to that reason
as far as I remember.

On Wed, Feb 24, 2010 at 5:45 PM, Joseph Puchalski
<joseph.puchalski at personalcyberspace.com> wrote:

I'm trying to modify my dialplan so that I can press a single button on my
phone, be connected to voicemail, and enter only a password to gain access. 

 

Currently I use a programmable key to dial 4000. I am prompted for my ID,
and then password.

 

I've poked around "mod voicemail" on the wiki and searched the mailing list
and web, but haven't found enough info. I have discovered that this behavior
seems to have been available in previous versions of the default dialplan.

 

Is it still possible? Is it advisable? Was this feature/behavior removed for
security reasons?

 

I apologize ahead of time if the answer is somewhere in plain sight that I
haven't looked yet. If so, I'd much appreciate being pointed in the right
direction.

 

As always, thanks for any help,

 

Joe P.


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




-- 
-Rupa

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100224/a70f78ec/attachment-0002.html 

From msc at freeswitch.org  Wed Feb 24 20:08:00 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 24 Feb 2010 20:08:00 -0800
Subject: [Freeswitch-users] Call for help - adding information to the wiki:
	SIP ALG's
Message-ID: <87f2f3b91002242008y599f86f0t86f467f8c16b4a20@mail.gmail.com>

Hi all,

I've just completed a new wiki page:

http://wiki.freeswitch.org/wiki/ALG

I would like all of you who have dealt with routers with SIP ALG's to submit
your input. I would like to see this page have a list of how-to's for all of
the popular routers. If we can make it easy for people to disable SIP ALG's
then I think we can all save ourselves time and energy answering questions
in IRC and the mailing lists. Please by all means add your knowledge here. I
started with the 2wire 3800HGV that I got for my ATT Uverse service.

If you have knowledge that you like to add to the wiki (on this subject or
any other) but are not confident in your wiki editing skills then contact me
off list and I will be happy to help you get up to speed. Editing your first
wiki page is always the hardest... :)

Thanks again for all of your help! By the way, today's community conference
call was great. Please plan on attending next week and we'll talk about more
great FreeSWITCH stuff. I will have the recording of Rupa discussing
mod_limit up on line as soon as I can.

Take care,
Michael
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100224/568aca63/attachment-0002.html 

From jason at jasonjgw.net  Wed Feb 24 20:27:30 2010
From: jason at jasonjgw.net (Jason White)
Date: Thu, 25 Feb 2010 15:27:30 +1100
Subject: [Freeswitch-users] Call for help - adding information to the
 wiki: SIP ALG's
In-Reply-To: <87f2f3b91002242008y599f86f0t86f467f8c16b4a20@mail.gmail.com>
References: <87f2f3b91002242008y599f86f0t86f467f8c16b4a20@mail.gmail.com>
Message-ID: <20100225042730.GA19249@jdc.jasonjgw.net>

Michael Collins <msc at freeswitch.org> wrote:
> I've just completed a new wiki page:
> 
> http://wiki.freeswitch.org/wiki/ALG
> 
> I would like all of you who have dealt with routers with SIP ALG's to submit
> your input. I would like to see this page have a list of how-to's for all of
> the popular routers. If we can make it easy for people to disable SIP ALG's
> then I think we can all save ourselves time and energy answering questions
> in IRC and the mailing lists. Please by all means add your knowledge here. 

For Cisco IOS, the following commands do it:
no ip nat service sip tcp port 5060
no ip nat service sip udp port 5060

Regrettably I'm not in a position to edit the wiki, but anyone is welcome to
add the above.




From mouncifbb at gmail.com  Wed Feb 24 20:48:19 2010
From: mouncifbb at gmail.com (Mouncif Benniane)
Date: Wed, 24 Feb 2010 23:48:19 -0500
Subject: [Freeswitch-users] compilation error of skypiax_protocol.c
In-Reply-To: <13529f9d0912022243y700728d4l30c7eb4e3152d1c9@mail.gmail.com>
References: <13529f9d0912022202g3254517dscc635bfb9c8b4439@mail.gmail.com>
	<9387ED56-4CDE-4685-AFF7-98E930BBA1F9@avgs.ca>
	<13529f9d0912022233i368b84c7s9bd960eb1380b53e@mail.gmail.com>
	<13529f9d0912022243y700728d4l30c7eb4e3152d1c9@mail.gmail.com>
Message-ID: <ade741d61002242048v150cb025q23c4463d38c9bbb9@mail.gmail.com>

you probably need: libjpeg-devel instead. just a thought.



On Thu, Dec 3, 2009 at 1:43 AM, Jingwei Yang <jingwei.yang at gmail.com> wrote:

> Not sure whether this error is due to the lack of libjpeg. I just double
> checked, this library had been installed.
>
> Package libjpeg-6b-37.i386 already installed and latest version
>
>
>
> On Thu, Dec 3, 2009 at 2:33 PM, Jingwei Yang <jingwei.yang at gmail.com>wrote:
>
>> Hi Mathieu, thanks for the promptly reply. The error has been fixed.
>> However, I encounter another one.
>>
>> gcc -I/usr/src/freeswitch/libs/tiff-3.8.2/libtiff -DNDEBUG -std=gnu99
>> -ffast-math -Wall -Wunused-variable -Wwrite-strings -Wstrict-prototypes
>> -Wmissing-prototypes -fvisibility=hidden -DSWITCH_API_VISIBILITY=1
>> -DHAVE_VISIBILITY=1 -g -O2 -o make_at_dictionary make_at_dictionary.o
>> -L/usr/src/freeswitch/libs/tiff-3.8.2/libtiff
>> /usr/src/freeswitch/libs/tiff-3.8.2/libtiff/.libs/libtiff.a -ljpeg -lz -lm
>> -lc
>> ./make_at_dictionary: error while loading shared libraries: libjpeg.so.7:
>> cannot open shared object file: No such file or directory
>> make[8]: *** [at_interpreter_dictionary.h] Error 127
>> make[7]: *** [all] Error 2
>> make[6]: *** [all-recursive] Error 1
>> make[5]: *** [../../../../libs/spandsp/src/libspandsp.la] Error 2
>>
>> make[4]: *** [install] Error 1
>> make[3]: *** [mod_voipcodecs-install] Error 1
>>
>> make[2]: *** [install-recursive] Error 1
>>
>> Do you have idea about this one?
>>
>> Thanks!
>>
>>
>> On Thu, Dec 3, 2009 at 2:09 PM, Mathieu Rene <mrene_lists at avgs.ca> wrote:
>>
>>> Consider it fixed.
>>> Committed revision 15765.
>>>
>>> Mathieu Rene
>>> Avant-Garde Solutions Inc
>>> Office: + 1 (514) 664-1044 x100
>>> Cell: +1 (514) 664-1044 x200
>>> mrene at avgs.ca
>>>
>>>
>>>
>>>
>>> On 3-Dec-09, at 1:02 AM, Jingwei Yang wrote:
>>>
>>> Hi Guys,
>>>
>>> I got a compilation error of skypiax_protocol.c with the latest version
>>> r15764.
>>>
>>> Compiling skypiax_protocol.c...
>>> *cc1: warnings being treated as errors*
>>> skypiax_protocol.c: In function ???X11_errors_handler???:
>>> skypiax_protocol.c:1548: warning: ISO C90 forbids mixed declarations and
>>> code
>>> skypiax_protocol.c: In function ???skypiax_send_message???:
>>> skypiax_protocol.c:1582: warning: ISO C90 forbids mixed declarations and
>>> code
>>> skypiax_protocol.c: In function ???skypiax_do_skypeapi_thread_func???:
>>> skypiax_protocol.c:1726: warning: ISO C90 forbids mixed declarations and
>>> code
>>> skypiax_protocol.c:1758: warning: ISO C90 forbids mixed declarations and
>>> code
>>> make[5]: *** [skypiax_protocol.o] Error 1
>>> make[4]: *** [install] Error 1
>>> make[3]: *** [mod_skypiax-install] Error 1
>>> make[2]: *** [install-recursive] Error 1
>>>
>>> I personally checked the file and it shouldn't be a merge problem. Does
>>> anyone encounter this as well?
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100224/deafda57/attachment-0002.html 

From ahmedmunir007 at gmail.com  Wed Feb 24 21:31:41 2010
From: ahmedmunir007 at gmail.com (Ahmed Munir)
Date: Thu, 25 Feb 2010 10:31:41 +0500
Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 44, Issue 217
In-Reply-To: <mailman.6837.1267029336.2495.freeswitch-users@lists.freeswitch.org>
References: <mailman.6837.1267029336.2495.freeswitch-users@lists.freeswitch.org>
Message-ID: <a60fed5e1002242131w43f622e1r3dd46d541868b1b6@mail.gmail.com>

Hi Tod,

After configuring ODBC connection using MySQL database issue isql *ODBC
Connection Name *on cli. If it connects successfully you can see the
database's tables what you've mentioned in odbc.ini file i.e.

isql mysql_fs (ODBC connection name)



---------- Forwarded message ----------
> From: "Todd" <phunk0000 at hotmail.com>
> To: <freeswitch-users at lists.freeswitch.org>
> Date: Wed, 24 Feb 2010 11:35:22 -0500
> Subject: Re: [Freeswitch-users] mod_nibblebill
>
> Yeah,  think I?m going to give it a shot from the beginning again and be
> very careful about install and config.  Thanks, I will keep you posted.
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Rupa
> Schomaker
> *Sent:* Wednesday, February 24, 2010 9:06 AM
> *To:* freeswitch-users
> *Subject:* Re: [Freeswitch-users] mod_nibblebill
>
>
>
> I am only passingly familiar with MySQL.   There must be a way for it to
> log all sql statements sent to it?
>
>
>
> Setting up postgresql would be the same (in broad terms) as mysql.  Install
> packages, create database/user/tables, populate data, configure odbc dsn,
> test.
>
> On Wed, Feb 24, 2010 at 7:03 AM, Todd <phunk0000 at hotmail.com> wrote:
>
> I am attempting to us MySQL.  I installed the spidermonkey mod, newest
> ODBC, compiled FS with ODBC, configured xml?s in FS?  not 100% sure I did it
> right though..followed wiki directions as close as possible.  What is the
> best way to verify the SQL is talking to MySQL? or perhaps the easiest way
> to switch  to postgresql?  Still kinda new to DB admin.  Thanks a ton.
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Rupa
> Schomaker
> *Sent:* Tuesday, February 23, 2010 6:38 PM
> *To:* freeswitch-users
> *Subject:* Re: [Freeswitch-users] mod_nibblebill
>
>
>
> what database backend are you using?  Have you verified the SQL is going to
> the right database backend?  I use mod_nibblebill against postgresql w/out
> problems.
>
> On Tue, Feb 23, 2010 at 9:42 AM, Meg Stroodle <phunk0000 at hotmail.com>
> wrote:
>
> Hello List!  I am trying to install mod_nibblebill on my FS installation.
> I get the following log entry form FS & nibblebill, but the database table I
> setup remains unchanged.  Any help in this matter would be greatly
> appreciated.  Following is an excerpt from the FS log:
>
>
>
> 2010-02-23 10:32:52.869149 [DEBUG] switch_rtp.c:2004 Correct ip/port
> confirmed.
>
> 2010-02-23 10:32:53.077162 [DEBUG] sofia.c:3727 Channel
> sofia/internal/3007 at 192.168.15.177 entering state [ready][200]
>
> 2010-02-23 10:32:53.149166 [DEBUG] switch_core_session.c:706 Send signal
> sofia/internal/sip:3008 at 192.168.15.176:21828 [BREAK]
>
> 2010-02-23 10:32:53.149166 [DEBUG] switch_core_session.c:706 Send signal
> sofia/internal/3007 at 192.168.15.177 [BREAK]
>
> 2010-02-23 10:32:53.149166 [DEBUG] switch_ivr_bridge.c:131 sofia/internal/
> sip:3008 at 192.168.15.176:21828 receive message [DISPLAY]
>
> 2010-02-23 10:32:53.169168 [DEBUG] switch_ivr_bridge.c:131
> sofia/internal/3007 at 192.168.15.177 receive message [DISPLAY]
>
> 2010-02-23 10:33:21.388949 [DEBUG] mod_nibblebill.c:529 Received request
> via SESSION_HEARTBEAT!
>
> 2010-02-23 10:33:21.388949 [DEBUG] mod_nibblebill.c:397 Attempting to bill
> at $1.0 per minute to account 3007
>
> 2010-02-23 10:33:21.388949 [INFO] mod_nibblebill.c:447 Beginning new
> billing on 3ec17ffc-c94e-46ff-bb0d-7ad0ef27bdab
>
> 2010-02-23 10:33:21.388949 [DEBUG] mod_nibblebill.c:455 28 seconds passed
> since last bill time of 2010-02-23 10:32:52
>
> 2010-02-23 10:33:21.388949 [DEBUG] mod_nibblebill.c:461 Billing $0.475997
> to 3007 (Call: 3ec17ffc-c94e-46ff-bb0d-7ad0ef27bdab / 0.000000 so far)
>
> 2010-02-23 10:33:51.508822 [DEBUG] mod_nibblebill.c:529 Received request
> via SESSION_HEARTBEAT!
>
> 2010-02-23 10:33:51.508822 [DEBUG] mod_nibblebill.c:397 Attempting to bill
> at $1.0 per minute to account 3007
>
> 2010-02-23 10:33:51.508822 [DEBUG] mod_nibblebill.c:449 Last successful
> billing time was
>
> 2010-02-23 10:33:51.508822 [DEBUG] mod_nibblebill.c:455 30 seconds passed
> since last bill time of 2010-02-23 10:33:21
>
> 2010-02-23 10:33:51.508822 [DEBUG] mod_nibblebill.c:461 Billing $0.501998
> to 3007 (Call: 3ec17ffc-c94e-46ff-bb0d-7ad0ef27bdab / 0.475997 so far)
>
> 2010-02-23 10:34:21.649708 [DEBUG] mod_nibblebill.c:529 Received request
> via SESSION_HEARTBEAT!
>
> 2010-02-23 10:34:21.649708 [DEBUG] mod_nibblebill.c:397 Attempting to bill
> at $1.0 per minute to account 3007
>
> 2010-02-23 10:34:21.649708 [DEBUG] mod_nibblebill.c:449 Last successful
> billing time was
>
> 2010-02-23 10:34:21.649708 [DEBUG] mod_nibblebill.c:455 30 seconds passed
> since last bill time of 2010-02-23 10:33:51
>
> 2010-02-23 10:34:21.649708 [DEBUG] mod_nibblebill.c:461 Billing $0.502348
> to 3007 (Call: 3ec17ffc-c94e-46ff-bb0d-7ad0ef27bdab / 0.977995 so far)
>
> 2010-02-23 10:34:26.254308 [NOTICE] sofia.c:329 Hangup
> sofia/internal/3007 at 192.168.15.177 [CS_EXECUTE] [NORMAL_CLEARING]
>
> 2010-02-23 10:34:26.254308 [DEBUG] switch_channel.c:1912 Send signal
> sofia/internal/3007 at 192.168.15.177 [KILL]
>
> 2010-02-23 10:34:26.254308 [DEBUG] switch_core_session.c:999 Send signal
> sofia/internal/3007 at 192.168.15.177 [BREAK]
>
> 2010-02-23 10:34:26.254308 [DEBUG] switch_core_state_machine.c:459
> sofia/internal/3007 at 192.168.15.177 thread mismatch skipping state handler.
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_ivr_bridge.c:470
> sofia/internal/3007 at 192.168.15.177 ending bridge by request from read
> function
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_ivr_bridge.c:563 BRIDGE THREAD
> DONE [sofia/internal/3007 at 192.168.15.177]
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_ivr_bridge.c:565 Send signal
> sofia/internal/sip:3008 at 192.168.15.176:21828 [BREAK]
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_ivr_bridge.c:520 sofia/internal/
> sip:3008 at 192.168.15.176:21828 receive message [UNBRIDGE]
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_session.c:645 Send signal
> sofia/internal/sip:3008 at 192.168.15.176:21828 [BREAK]
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_ivr_bridge.c:563 BRIDGE THREAD
> DONE [sofia/internal/sip:3008 at 192.168.15.176:21828]
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_ivr_bridge.c:565 Send signal
> sofia/internal/3007 at 192.168.15.177 [BREAK]
>
> 2010-02-23 10:34:26.269008 [NOTICE] switch_ivr_bridge.c:617 Hangup
> sofia/internal/sip:3008 at 192.168.15.176:21828 [CS_EXCHANGE_MEDIA]
> [NORMAL_CLEARING]
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_channel.c:1912 Send signal
> sofia/internal/sip:3008 at 192.168.15.176:21828 [KILL]
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_session.c:999 Send signal
> sofia/internal/sip:3008 at 192.168.15.176:21828 [BREAK]
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:488
> (sofia/internal/sip:3008 at 192.168.15.176:21828) State HANGUP
>
> 2010-02-23 10:34:26.269008 [DEBUG] mod_sofia.c:358 Channel sofia/internal/
> sip:3008 at 192.168.15.176:21828 hanging up, cause: NORMAL_CLEARING
>
> 2010-02-23 10:34:26.269008 [DEBUG] mod_sofia.c:400 Sending BYE to
> sofia/internal/sip:3008 at 192.168.15.176:21828
>
> 2010-02-23 10:34:26.269008 [DEBUG] mod_nibblebill.c:397 Attempting to bill
> at $2.30 per minute to account 3008
>
> 2010-02-23 10:34:26.269008 [INFO] mod_nibblebill.c:447 Beginning new
> billing on c1c42712-89e9-44c5-914f-11ed8f6aebb1
>
> 2010-02-23 10:34:26.269008 [DEBUG] mod_nibblebill.c:455 93 seconds passed
> since last bill time of 2010-02-23 10:32:52
>
> 2010-02-23 10:34:26.269008 [DEBUG] mod_nibblebill.c:461 Billing $3.582436
> to 3008 (Call: c1c42712-89e9-44c5-914f-11ed8f6aebb1 / 0.000000 so far)
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:46
> sofia/internal/sip:3008 at 192.168.15.176:21828 Standard HANGUP, cause:
> NORMAL_CLEARING
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:488
> (sofia/internal/sip:3008 at 192.168.15.176:21828) State HANGUP going to sleep
>
> 2010-02-23 10:34:26.269008 [INFO] switch_core_session.c:1108
> sofia/internal/sip:3008 at 192.168.15.176:21828 setting session heartbeat to
> 30 second(s).
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:351
> (sofia/internal/sip:3008 at 192.168.15.176:21828) State EXCHANGE_MEDIA going
> to sleep
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:314
> (sofia/internal/sip:3008 at 192.168.15.176:21828) Running State Change
> CS_HANGUP
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:465
> sofia/internal/sip:3008 at 192.168.15.176:21828 handler already called,
> skipping state handler.
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:333
> (sofia/internal/sip:3008 at 192.168.15.176:21828) State Change CS_HANGUP ->
> CS_REPORTING
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_session.c:999 Send signal
> sofia/internal/sip:3008 at 192.168.15.176:21828 [BREAK]
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:314
> (sofia/internal/sip:3008 at 192.168.15.176:21828) Running State Change
> CS_REPORTING
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:579
> (sofia/internal/sip:3008 at 192.168.15.176:21828) State REPORTING
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:53
> sofia/internal/sip:3008 at 192.168.15.176:21828 Standard REPORTING, cause:
> NORMAL_CLEARING
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:579
> (sofia/internal/sip:3008 at 192.168.15.176:21828) State REPORTING going to
> sleep
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_state_machine.c:327
> (sofia/internal/sip:3008 at 192.168.15.176:21828) State Change CS_REPORTING
> -> CS_DESTROY
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_session.c:999 Send signal
> sofia/internal/sip:3008 at 192.168.15.176:21828 [BREAK]
>
> 2010-02-23 10:34:26.269008 [DEBUG] switch_core_session.c:1136 Session 6
> (sofia/internal/sip:3008 at 192.168.15.176:21828) Locked, Waiting on external
> entities
>
> 2010-02-23 10:34:26.272437 [DEBUG] switch_core_state_machine.c:348 (
> sofia/internal/3007 at 192.168.15.177) State EXECUTE going to sleep
>
> 2010-02-23 10:34:26.272437 [DEBUG] switch_core_state_machine.c:314 (
> sofia/internal/3007 at 192.168.15.177) Running State Change CS_HANGUP
>
> 2010-02-23 10:34:26.272437 [DEBUG] switch_core_state_machine.c:488 (
> sofia/internal/3007 at 192.168.15.177) State HANGUP
>
> 2010-02-23 10:34:26.272437 [DEBUG] mod_sofia.c:358 Channel
> sofia/internal/3007 at 192.168.15.177 hanging up, cause: NORMAL_CLEARING
>
> 2010-02-23 10:34:26.272437 [DEBUG] mod_nibblebill.c:397 Attempting to bill
> at $1.0 per minute to account 3007
>
> 2010-02-23 10:34:26.272437 [DEBUG] mod_nibblebill.c:449 Last successful
> billing time was
>
> 2010-02-23 10:34:26.272437 [DEBUG] mod_nibblebill.c:455 4 seconds passed
> since last bill time of 2010-02-23 10:34:21
>
> 2010-02-23 10:34:26.272437 [DEBUG] mod_nibblebill.c:461 Billing $0.077045
> to 3007 (Call: 3ec17ffc-c94e-46ff-bb0d-7ad0ef27bdab / 1.480343 so far)
>
> 2010-02-23 10:34:26.272437 [DEBUG] switch_core_state_machine.c:46
> sofia/internal/3007 at 192.168.15.177 Standard HANGUP, cause: NORMAL_CLEARING
>
>
>
> Anyhelp getting nibblebill to connect to the database would be greatly
> appreciated.  Thanks
>
>
>
>
>  ------------------------------
>
> Hotmail: Trusted email with Microsoft?s powerful SPAM protection. Sign up
> now. <http://clk.atdmt.com/GBL/go/201469226/direct/01/>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> --
> -Rupa
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Regards,

Ahmed Munir
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100225/29cb30e4/attachment-0002.html 

From moizchinoy at gmail.com  Wed Feb 24 22:04:46 2010
From: moizchinoy at gmail.com (Moiz Chinoy)
Date: Thu, 25 Feb 2010 10:04:46 +0400
Subject: [Freeswitch-users] Dingaling - external rtp supported?
In-Reply-To: <191c3a031002231131x19c9791cgd986a6c1be18e09a@mail.gmail.com>
References: <29b888f81002160835y6e290319ub0d04c088caa1b79@mail.gmail.com>
	<DEA39AD1-0EE9-473A-B78C-CDDF3F100578@freeswitch.org>
	<29b888f81002170318p63622f3fhc8d26ad0e85602d2@mail.gmail.com>
	<191c3a031002170538r3b029515i8a000812a70ad558@mail.gmail.com>
	<29b888f81002170741y3d13cb57ycb4436a372c4eded@mail.gmail.com>
	<191c3a031002171136q7424e9bai34379abb542daf93@mail.gmail.com>
	<29b888f81002210800v617bca9dk4bf1abf1bd791ee0@mail.gmail.com>
	<29b888f81002222359w6ae4cd8ej944d453a5b099af4@mail.gmail.com>
	<191c3a031002231131x19c9791cgd986a6c1be18e09a@mail.gmail.com>
Message-ID: <29b888f81002242204x171f4ab3la34d4ace604cd50a@mail.gmail.com>

I was using GuntTls-2.7.3 for windows. Now I am using GuntTls-2.9.9. I have
modified only gnutls.h, added following line:

typedef long ssize_t;

because otherwise it was giving errors...

What is the recommended version of the TLS lib for windows?

After upgrading the the GnuTls and freeswitch to rev 16806, I ran the
freeswitch with mod_dingalilg enabled. Once started, I issued just the
'shutdown' command on the console, exception happened.

......................
2010-02-25 09:45:29.795285 [CONSOLE] switch_loadable_module.c:1277 Stopping:
CORE_SOFTTIMER_MODULE
2010-02-25 09:45:29.810910 [CONSOLE] switch_time.c:780 Soft timer thread
exiting.
2010-02-25 09:45:29.810910 [NOTICE] switch_loadable_module.c:98 Thread ended
for CORE_SOFTTIMER_MODULE
2010-02-25 09:45:29.826535 [DEBUG] switch_loadable_module.c:456 Write lock
interface 'dingaling' to wait for existing references.
2010-02-25 09:45:29.826535 [NOTICE] switch_loadable_module.c:464 Deleting
Endpoint 'dingaling'
2010-02-25 09:45:29.826535 [NOTICE] switch_loadable_module.c:572 Deleting
API Function 'dl_debug'
2010-02-25 09:45:29.826535 [DEBUG] switch_loadable_module.c:574 Write lock
interface 'dl_debug' to wait for existing references.
2010-02-25 09:45:29.826535 [NOTICE] switch_loadable_module.c:572 Deleting
API Function 'dl_pres'
2010-02-25 09:45:29.826535 [DEBUG] switch_loadable_module.c:574 Write lock
interface 'dl_pres' to wait for existing references.
2010-02-25 09:45:29.826535 [NOTICE] switch_loadable_module.c:572 Deleting
API Function 'dl_logout'
2010-02-25 09:45:29.826535 [DEBUG] switch_loadable_module.c:574 Write lock
interface 'dl_logout' to wait for existing references.
2010-02-25 09:45:29.826535 [NOTICE] switch_loadable_module.c:572 Deleting
API Function 'dl_login'
2010-02-25 09:45:29.826535 [DEBUG] switch_loadable_module.c:574 Write lock
interface 'dl_login' to wait for existing references.
2010-02-25 09:45:29.826535 [NOTICE] switch_loadable_module.c:572 Deleting
API Function 'dingaling'
2010-02-25 09:45:29.826535 [DEBUG] switch_loadable_module.c:574 Write lock
interface 'dingaling' to wait for existing references.
2010-02-25 09:45:29.826535 [DEBUG] switch_loadable_module.c:710 Write lock
interface 'jingle' to wait for existing references.
2010-02-25 09:45:29.826535 [NOTICE] switch_loadable_module.c:719 Deleting
Chat interface 'jingle'
2010-02-25 09:45:29.826535 [CONSOLE] switch_loadable_module.c:1277 Stopping:
mod_dingaling
2010-02-25 09:45:31.185910 [DEBUG] libdingaling.c:1546 io error 2 7 retry in
1 second(s)
........................

And the code went in the stream.c...

int
iks_fd (iksparser *prs)
{
struct stream_data *data;

if (prs) {
data = iks_user_data (prs);
if (data) {
return (int) data->sock;
}
}
return -1;
}



On Tue, Feb 23, 2010 at 11:31 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:
> If you are modifying your build to add libgcrypt / libgnutls to win32, you
> have chosen an incompatible version of one of these libs.  We do not
support
> manually adding this modification to the code, you will need to find
someone
> else who has done it successfully to help you.
>
>
>
>
> On Tue, Feb 23, 2010 at 1:59 AM, Moiz Chinoy <moizchinoy at gmail.com> wrote:
>>
>> Moreover, if I gtalk client is on the same machine as FS and i have
>> following settings, FS crashes with the same mutex error.
>>
>> External Sip Profile has following lines:
>> ---------------------------------------------------------
>>   <param name="rtp-ip" value="$${local_ip_v4}"/>
>>   <param name="sip-ip" value="$${local_ip_v4}"/>
>>   <param name="ext-rtp-ip" value="auto-nat"/>
>>   <param name="ext-sip-ip" value="auto-nat"/>
>>
>> Jingle Client.xml has following lines:
>> -----------------------------------------------------
>>   <param name="rtp-ip" value="$${bind_server_ip}"/>
>>   <!--<param name="ext-rtp-ip" value="$${external_rtp_ip}"/>-->
>>   <!--<param name="disable-rtp-auto-adjust" value="true"/>-->
>>   <!--<param name="avatar" value="/path/to/tiny.jpg"/>-->
>>   <!--<param name="candidate-acl" value="wan.auto"/>-->
>>   <!--<param name="local-network-acl" value="localnet.auto"/>-->
>>
>> If I uncomment the following line in client.xml (Jingle profile)
>>   <!--<param name="local-network-acl" value="localnet.auto"/>-->
>> then exception does not happen.
>>
>> Is this a known issue or do I need to post it in JIRA?
>>
>> Tell me if more logs are needed...
>>
>>
>> On Sun, Feb 21, 2010 at 8:00 PM, Moiz Chinoy <moizchinoy at gmail.com>
wrote:
>> > Guys,
>> >
>> > To make things simple gtalk client is entirely on different network.
>> >
>> > Call comes from outside through external Sip profile.
>> >
>> > If gtalk answers the call after 3-4 rings both parties can hear each
>> > other.
>> > If gtalk answers the call after 2 rings both parties no one can hear
>> > each other.
>> > If gtalk answers the call immediately FS crashes.
>> >
>> > Attached is the screen shot of the error...
>> >
>> > Here is the FS log...
>> > --------------------------------
>> > http://pastebin.freeswitch.org/12197
>> >
>> > External Sip Profile has following lines:
>> > ---------------------------------------------------------
>> >    <param name="rtp-ip" value="$${local_ip_v4}"/>
>> >    <param name="sip-ip" value="$${local_ip_v4}"/>
>> >    <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
>> >    <param name="ext-sip-ip" value="$${external_sip_ip}"/>
>> >
>> > Jingle Client.xml has following lines:
>> > -----------------------------------------------------
>> >    <param name="rtp-ip" value="$${bind_server_ip}"/>
>> >    <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
>> >    <param name="disable-rtp-auto-adjust" value="true"/>
>> >    <!--<param name="avatar" value="/path/to/tiny.jpg"/>-->
>> >    <!--<param name="candidate-acl" value="wan.auto"/>-->
>> >    <!--<param name="local-network-acl" value="localnet.auto"/>-->
>> >
>> > Vars.xml has following lines:
>> > -------------------------------------------
>> > <X-PRE-PROCESS cmd="set"
>> > data="external_rtp_ip=stun:stun.freeswitch.org"/>
>> > <X-PRE-PROCESS cmd="set"
>> > data="external_sip_ip=stun:stun.freeswitch.org"/>
>> >
>> >
>> > Please advise me how can I provide more of the required data.
>> >
>> > On Wed, Feb 17, 2010 at 11:36 PM, Anthony Minessale
>> > <anthony.minessale at gmail.com> wrote:
>> >> you cant combine stun and gtalk and boxes in the same lan very easily
>> >> if you
>> >> do need to do that you will need to mess with
>> >>
>> >> <param name="candidate-acl" value="wan.auto"/>
>> >> <param name="local-network-acl" value="localnet.auto"/>
>> >>
>> >>
>> >>
>> >>
>> >>
>> >> On Wed, Feb 17, 2010 at 9:41 AM, Moiz Chinoy <moizchinoy at gmail.com>
>> >> wrote:
>> >>>
>> >>> Guys I am unable to produce the crash but now both parties cannot
hear
>> >>> each other!
>> >>>
>> >>> Vars.xml has following lines:
>> >>>  <X-PRE-PROCESS cmd="set"
>> >>> data="external_rtp_ip=stun:stun.freeswitch.org"/>
>> >>>  <X-PRE-PROCESS cmd="set"
>> >>> data="external_sip_ip=stun:stun.freeswitch.org"/>
>> >>>
>> >>> Jingle Client.xml has following lines:
>> >>>    <param name="rtp-ip" value="$${bind_server_ip}"/>
>> >>>    <!-- <param name="ext-rtp-ip" value="auto-nat"/> -->
>> >>>    <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
>> >>>    <param name="disable-rtp-auto-adjust" value="true"/>
>> >>>    <param name="local-network-acl" value="localnet.auto"/>
>> >>>
>> >>> On Wed, Feb 17, 2010 at 5:38 PM, Anthony Minessale
>> >>> <anthony.minessale at gmail.com> wrote:
>> >>> > Obtain a stack trace from the crash.
>> >>> >
>> >>> > On Feb 17, 2010 5:26 AM, "Moiz Chinoy" <moizchinoy at gmail.com>
wrote:
>> >>> >
>> >>> > Hi,
>> >>> >
>> >>> > FS rev: 16673
>> >>> > Platform: Windows
>> >>> >
>> >>> > More details:
>> >>> >
>> >>> > FS is behind NAT and machine is running a VPN connection.
>> >>> >
>> >>> > FS and GTalk client on the same machine:
>> >>> >
>> >>> >
>> >>> >
--------------------------------------------------------------------------------------------------
>> >>> > jingle profile client.xml has following line:
>> >>> > <param name="rtp-ip" value="$${bind_server_ip}"/>
>> >>> >
>> >>> > External SIP call is successfully bridged to GTalk client.
>> >>> >
>> >>> >
>> >>> > FS and GTalk client on the different machine:
>> >>> >
>> >>> >
>> >>> >
--------------------------------------------------------------------------------------------------
>> >>> > jingle profile client.xml has following lines:
>> >>> > <param name="rtp-ip" value="$${bind_server_ip}"/>
>> >>> >
>> >>> > <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
>> >>> >
>> >>> > As soon as external SIP call land and I try to bridge the call to
>> >>> > GTalk client, FS crashes.
>> >>> >
>> >>> >
>> >>> > NAT Details:
>> >>> > ---------------------------
>> >>> > I think my NAT does not support UpNP or PMP. The reason I say it
>> >>> > because when FS starts following message is displayed:
>> >>> >
>> >>> > 2010-02-17 09:15:42.809674 [INFO] switch_nat.c:409 Scanning for NAT
>> >>> > 2010-02-17 09:15:42.809674 [ERR] switch_nat.c:197 Error checking
for
>> >>> > PMP [init failed]
>> >>> > 2010-02-17 09:15:42.809674 [DEBUG] switch_nat.c:414 Checking for
>> >>> > UPnP
>> >>> > 2010-02-17 09:15:56.825388 [DEBUG] switch_nat.c:114 No
>> >>> > InternetGatewayDevice, using first entry as default
>> >>> > (http://192.168.16.17:50144/).
>> >>> > 2010-02-17 09:15:56.887889 [INFO] switch_nat.c:429 No PMP or UPnP
>> >>> > NAT
>> >>> > devices detected!
>> >>> >
>> >>> >
>> >>> >
>> >>> > On Tue, Feb 16, 2010 at 8:41 PM, Brian West <brian at freeswitch.org>
>> >>> > wrote:
>> >>> >> can you please update...
>> >>> >
>> >>> > _______________________________________________
>> >>> > FreeSWITCH-users mailing list
>> >>> > FreeSWITCH-users at lists.freeswitch.org
>> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> >
>> >>> > UNSUBSCRIBE:
http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> > http://www.freeswitch.org
>> >>> >
>> >>> >
>> >>>
>> >>>
>> >>>
>> >>> --
>> >>> Regards,
>> >>> Moiz Chinoy.
>> >>>
>> >>> _______________________________________________
>> >>> FreeSWITCH-users mailing list
>> >>> FreeSWITCH-users at lists.freeswitch.org
>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>
>> >>> UNSUBSCRIBE:
http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> http://www.freeswitch.org
>> >>
>> >>
>> >>
>> >> --
>> >> Anthony Minessale II
>> >>
>> >> FreeSWITCH http://www.freeswitch.org/
>> >> ClueCon http://www.cluecon.com/
>> >> Twitter: http://twitter.com/FreeSWITCH_wire
>> >>
>> >> AIM: anthm
>> >> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
>> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> >> IRC: irc.freenode.net #freeswitch
>> >>
>> >> FreeSWITCH Developer Conference
>> >> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>> >> iax:guest at conference.freeswitch.org/888
>> >> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> >> pstn:+19193869900
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:
http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >>
>> >
>> >
>> >
>> > --
>> > Regards,
>> > Moiz Chinoy.
>> >
>>
>>
>>
>> --
>> Regards,
>> Moiz Chinoy.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Regards,
Moiz Chinoy.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100225/892e5636/attachment-0002.html 

From gkuri at ieee.org  Wed Feb 24 22:48:53 2010
From: gkuri at ieee.org (Gabriel Kuri)
Date: Wed, 24 Feb 2010 22:48:53 -0800
Subject: [Freeswitch-users] [Freeswitch-dev] Call for help - adding
	information to the wiki: SIP ALG's
In-Reply-To: <87f2f3b91002242008y599f86f0t86f467f8c16b4a20@mail.gmail.com>
References: <87f2f3b91002242008y599f86f0t86f467f8c16b4a20@mail.gmail.com>
Message-ID: <8b1c9cda1002242248u5e39a4ecvda6a3644527f59bf@mail.gmail.com>

I can think of several devices that have serious SIP ALG issues, I'll
spend the time and add some of the devices I know about.

Do you want to limit the page to SIP ALG problems or anything entirely
stupid that routers seem to do that could possibly break VoIP? For
example, we've found several routers with broken DNS
resolvers/forwarders that don't know how to deal with SRV records, in
particular routers that run VxWorks internally. People relying on
routers running VxWorks to resolve SRV records could bang their head
on the wall trying to figure out why nothing is working, unless they
manually configure their devices to use other DNS servers (Stinksys'
newer G and N routers running VxWorks comes to mind).

Cheers,
Gabe

On Wed, Feb 24, 2010 at 8:08 PM, Michael Collins <msc at freeswitch.org> wrote:
> Hi all,
>
> I've just completed a new wiki page:
>
> http://wiki.freeswitch.org/wiki/ALG
>
> I would like all of you who have dealt with routers with SIP ALG's to submit
> your input. I would like to see this page have a list of how-to's for all of
> the popular routers. If we can make it easy for people to disable SIP ALG's
> then I think we can all save ourselves time and energy answering questions
> in IRC and the mailing lists. Please by all means add your knowledge here. I
> started with the 2wire 3800HGV that I got for my ATT Uverse service.
>
> If you have knowledge that you like to add to the wiki (on this subject or
> any other) but are not confident in your wiki editing skills then contact me
> off list and I will be happy to help you get up to speed. Editing your first
> wiki page is always the hardest... :)
>
> Thanks again for all of your help! By the way, today's community conference
> call was great. Please plan on attending next week and we'll talk about more
> great FreeSWITCH stuff. I will have the recording of Rupa discussing
> mod_limit up on line as soon as I can.
>
> Take care,
> Michael
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>



From jingwei.yang at gmail.com  Wed Feb 24 23:14:18 2010
From: jingwei.yang at gmail.com (Jingwei Yang)
Date: Thu, 25 Feb 2010 15:14:18 +0800
Subject: [Freeswitch-users] compilation error of skypiax_protocol.c
In-Reply-To: <ade741d61002242048v150cb025q23c4463d38c9bbb9@mail.gmail.com>
References: <13529f9d0912022202g3254517dscc635bfb9c8b4439@mail.gmail.com>
	<9387ED56-4CDE-4685-AFF7-98E930BBA1F9@avgs.ca>
	<13529f9d0912022233i368b84c7s9bd960eb1380b53e@mail.gmail.com>
	<13529f9d0912022243y700728d4l30c7eb4e3152d1c9@mail.gmail.com>
	<ade741d61002242048v150cb025q23c4463d38c9bbb9@mail.gmail.com>
Message-ID: <13529f9d1002242314h4e36badbgeba7cf5500253235@mail.gmail.com>

Thanks Mouncif, the os was reinstalled and the problem disappeared.

On Thu, Feb 25, 2010 at 12:48 PM, Mouncif Benniane <mouncifbb at gmail.com>wrote:

> you probably need: libjpeg-devel instead. just a thought.
>
>
>
>
> On Thu, Dec 3, 2009 at 1:43 AM, Jingwei Yang <jingwei.yang at gmail.com>wrote:
>
>> Not sure whether this error is due to the lack of libjpeg. I just double
>> checked, this library had been installed.
>>
>> Package libjpeg-6b-37.i386 already installed and latest version
>>
>>
>>
>> On Thu, Dec 3, 2009 at 2:33 PM, Jingwei Yang <jingwei.yang at gmail.com>wrote:
>>
>>> Hi Mathieu, thanks for the promptly reply. The error has been fixed.
>>> However, I encounter another one.
>>>
>>> gcc -I/usr/src/freeswitch/libs/tiff-3.8.2/libtiff -DNDEBUG -std=gnu99
>>> -ffast-math -Wall -Wunused-variable -Wwrite-strings -Wstrict-prototypes
>>> -Wmissing-prototypes -fvisibility=hidden -DSWITCH_API_VISIBILITY=1
>>> -DHAVE_VISIBILITY=1 -g -O2 -o make_at_dictionary make_at_dictionary.o
>>> -L/usr/src/freeswitch/libs/tiff-3.8.2/libtiff
>>> /usr/src/freeswitch/libs/tiff-3.8.2/libtiff/.libs/libtiff.a -ljpeg -lz -lm
>>> -lc
>>> ./make_at_dictionary: error while loading shared libraries: libjpeg.so.7:
>>> cannot open shared object file: No such file or directory
>>> make[8]: *** [at_interpreter_dictionary.h] Error 127
>>> make[7]: *** [all] Error 2
>>> make[6]: *** [all-recursive] Error 1
>>> make[5]: *** [../../../../libs/spandsp/src/libspandsp.la] Error 2
>>>
>>> make[4]: *** [install] Error 1
>>> make[3]: *** [mod_voipcodecs-install] Error 1
>>>
>>> make[2]: *** [install-recursive] Error 1
>>>
>>> Do you have idea about this one?
>>>
>>> Thanks!
>>>
>>>
>>> On Thu, Dec 3, 2009 at 2:09 PM, Mathieu Rene <mrene_lists at avgs.ca>wrote:
>>>
>>>> Consider it fixed.
>>>> Committed revision 15765.
>>>>
>>>> Mathieu Rene
>>>> Avant-Garde Solutions Inc
>>>> Office: + 1 (514) 664-1044 x100
>>>> Cell: +1 (514) 664-1044 x200
>>>> mrene at avgs.ca
>>>>
>>>>
>>>>
>>>>
>>>> On 3-Dec-09, at 1:02 AM, Jingwei Yang wrote:
>>>>
>>>> Hi Guys,
>>>>
>>>> I got a compilation error of skypiax_protocol.c with the latest version
>>>> r15764.
>>>>
>>>> Compiling skypiax_protocol.c...
>>>> *cc1: warnings being treated as errors*
>>>> skypiax_protocol.c: In function ???X11_errors_handler???:
>>>> skypiax_protocol.c:1548: warning: ISO C90 forbids mixed declarations and
>>>> code
>>>> skypiax_protocol.c: In function ???skypiax_send_message???:
>>>> skypiax_protocol.c:1582: warning: ISO C90 forbids mixed declarations and
>>>> code
>>>> skypiax_protocol.c: In function ???skypiax_do_skypeapi_thread_func???:
>>>> skypiax_protocol.c:1726: warning: ISO C90 forbids mixed declarations and
>>>> code
>>>> skypiax_protocol.c:1758: warning: ISO C90 forbids mixed declarations and
>>>> code
>>>> make[5]: *** [skypiax_protocol.o] Error 1
>>>> make[4]: *** [install] Error 1
>>>> make[3]: *** [mod_skypiax-install] Error 1
>>>> make[2]: *** [install-recursive] Error 1
>>>>
>>>> I personally checked the file and it shouldn't be a merge problem. Does
>>>> anyone encounter this as well?
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100225/66831f31/attachment-0002.html 

From rm at callrica.co.za  Thu Feb 25 00:33:08 2010
From: rm at callrica.co.za (Roly Maz)
Date: Thu, 25 Feb 2010 10:33:08 +0200
Subject: [Freeswitch-users] SIP Trunk with Private Static IP?
Message-ID: <008401cab5f5$4795b910$d6c12b30$@co.za>

Hi Community 

 

 

My Provider provides the following info when they supply a SIP trunk:

 

.         A direct connection into their network. i.e. they provide private
IPs:

.         An IP address I must use for my FS box e.g. IP: 10. 42.0.66 MASK:
255.255.255.248 GW: 10. 42.0.68

.         An IP address for their SIP server 10.42.0.1

 

I have setup a dual homed FS box (Windows Server 2008, latest FS version)

 

NIC 1 - Internal LAN IP: 10.0.2.185 MASK: 255.255.255.0 GW: 10.0.2.253

NIC 2 - SIP Provider (WAN) IP: 10. 42.0.66 MASK: 255.255.255.248 GW: 10.
42.0.68

 

Windows complains about multiple gateways - which I ignore? I can ping
internal addresses  and the SIP Server 

 

When I fire up FS, I can register Xlite phones on my LAN. I can dial and
hear the test IVR (5000)

 

This means my Internal SIP Profile is ok. 

 

Now, how do i route a call out to the 10.42.01 SIP Server?

 

 Creating a gateway doesn't make sense, because I am not supplied a
username/password? 

 

Any pointers would be most appreciated, I am sure I am missing something
really simple.

 

Roland

 

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100225/d18b57dc/attachment-0002.html 

From mcampbellsmith at gmail.com  Thu Feb 25 00:43:14 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Thu, 25 Feb 2010 19:43:14 +1100
Subject: [Freeswitch-users] [Freeswitch-dev] Call for help - adding
	information to the wiki: SIP ALG's
In-Reply-To: <8b1c9cda1002242248u5e39a4ecvda6a3644527f59bf@mail.gmail.com>
References: <87f2f3b91002242008y599f86f0t86f467f8c16b4a20@mail.gmail.com>
	<8b1c9cda1002242248u5e39a4ecvda6a3644527f59bf@mail.gmail.com>
Message-ID: <33c87fa31002250043l1edbb87dg2e7a58e879656f06@mail.gmail.com>

I had an issue with a Thomson SpeedTouch 530 router that was causing
authentication to fail... its the thread titled 'Forbidden using UDP,
works with TCP/TLS' that everyone said was caused by a broken ATA ...
lucky no one chipped in so I could buy a new one (as suggested by
Anthony) - it wouldn't have helped because the router was broken and
modifying the authentication parameters!

I'll update the wiki with the info

On Thu, Feb 25, 2010 at 5:48 PM, Gabriel Kuri <gkuri at ieee.org> wrote:
> I can think of several devices that have serious SIP ALG issues, I'll
> spend the time and add some of the devices I know about.
>
> Do you want to limit the page to SIP ALG problems or anything entirely
> stupid that routers seem to do that could possibly break VoIP? For
> example, we've found several routers with broken DNS
> resolvers/forwarders that don't know how to deal with SRV records, in
> particular routers that run VxWorks internally. People relying on
> routers running VxWorks to resolve SRV records could bang their head
> on the wall trying to figure out why nothing is working, unless they
> manually configure their devices to use other DNS servers (Stinksys'
> newer G and N routers running VxWorks comes to mind).
>
> Cheers,
> Gabe
>
> On Wed, Feb 24, 2010 at 8:08 PM, Michael Collins <msc at freeswitch.org> wrote:
>> Hi all,
>>
>> I've just completed a new wiki page:
>>
>> http://wiki.freeswitch.org/wiki/ALG
>>
>> I would like all of you who have dealt with routers with SIP ALG's to submit
>> your input. I would like to see this page have a list of how-to's for all of
>> the popular routers. If we can make it easy for people to disable SIP ALG's
>> then I think we can all save ourselves time and energy answering questions
>> in IRC and the mailing lists. Please by all means add your knowledge here. I
>> started with the 2wire 3800HGV that I got for my ATT Uverse service.
>>
>> If you have knowledge that you like to add to the wiki (on this subject or
>> any other) but are not confident in your wiki editing skills then contact me
>> off list and I will be happy to help you get up to speed. Editing your first
>> wiki page is always the hardest... :)
>>
>> Thanks again for all of your help! By the way, today's community conference
>> call was great. Please plan on attending next week and we'll talk about more
>> great FreeSWITCH stuff. I will have the recording of Rupa discussing
>> mod_limit up on line as soon as I can.
>>
>> Take care,
>> Michael
>>
>> _______________________________________________
>> FreeSWITCH-dev mailing list
>> FreeSWITCH-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From steveayre at gmail.com  Thu Feb 25 00:57:39 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 25 Feb 2010 08:57:39 +0000
Subject: [Freeswitch-users] SIP Trunk with Private Static IP?
In-Reply-To: <008401cab5f5$4795b910$d6c12b30$@co.za>
References: <008401cab5f5$4795b910$d6c12b30$@co.za>
Message-ID: <eafa772f1002250057l3de5b2abu54da8725cbb77281@mail.gmail.com>

Create two SIP profiles, each bound to one of your local IPs.

You may create a gateway on the profile for the SIP trunk IP for the
10.42.0.1 server, but this is optional.

You can then bridge calls via the SIP server using one of:
<action application="bridge" data="sofia/gateway/gatewayname"/>
<action application="bridge" data="sofia/profilename/number at 10.42.0.1"/>

The advantages of using a gateway are:
- supports authentication
- will monitor the gateway to detect if it goes down (so calls fail
instantly rather than after a timeout)

As for the default gateway, it is the IP you send via to reach IPs
that are not on a network you are connected directly to - you should
probably only have one set, and it should be the one you go via to
reach the Internet.

-Steve


On 25 February 2010 08:33, Roly Maz <rm at callrica.co.za> wrote:
> Hi Community
>
>
>
>
>
> My Provider provides the following info when they supply a SIP trunk:
>
>
>
> ????????? A direct connection into their network. i.e. they provide private
> IPs:
>
> ????????? An IP address I must use for my FS box e.g. IP: 10. 42.0.66 MASK:
> 255.255.255.248 GW: 10. 42.0.68
>
> ????????? An IP address for their SIP server 10.42.0.1
>
>
>
> I have setup a dual homed FS box (Windows Server 2008, latest FS version)
>
>
>
> NIC 1 ? Internal LAN IP: 10.0.2.185 MASK: 255.255.255.0 GW: 10.0.2.253
>
> NIC 2 ? SIP Provider (WAN) IP: 10. 42.0.66 MASK: 255.255.255.248 GW: 10.
> 42.0.68
>
>
>
> Windows complains about multiple gateways ? which I ignore? I can ping
> internal addresses ?and the SIP Server
>
>
>
> When I fire up FS, I can register Xlite phones on my LAN. I can dial and
> hear the test IVR (5000)
>
>
>
> This means my Internal SIP Profile is ok.
>
>
>
> Now, how do i route a call out to the 10.42.01 SIP Server?
>
>
>
> ?Creating a gateway doesn?t make sense, because I am not supplied a
> username/password?
>
>
>
> Any pointers would be most appreciated, I am sure I am missing something
> really simple.
>
>
>
> Roland
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From steveayre at gmail.com  Thu Feb 25 00:58:59 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 25 Feb 2010 08:58:59 +0000
Subject: [Freeswitch-users] SIP Trunk with Private Static IP?
In-Reply-To: <eafa772f1002250057l3de5b2abu54da8725cbb77281@mail.gmail.com>
References: <008401cab5f5$4795b910$d6c12b30$@co.za>
	<eafa772f1002250057l3de5b2abu54da8725cbb77281@mail.gmail.com>
Message-ID: <eafa772f1002250058i2cd7b30ai42f44d0ea1d6eb88@mail.gmail.com>

Gateways do not require usernames and passwords. You are required to
set the parameter, but if no authentication is needed they are ignored
so you can put anything in the field, so that is not a reason to avoid
them.

-Steve


On 25 February 2010 08:57, Steven Ayre <steveayre at gmail.com> wrote:
> Create two SIP profiles, each bound to one of your local IPs.
>
> You may create a gateway on the profile for the SIP trunk IP for the
> 10.42.0.1 server, but this is optional.
>
> You can then bridge calls via the SIP server using one of:
> <action application="bridge" data="sofia/gateway/gatewayname"/>
> <action application="bridge" data="sofia/profilename/number at 10.42.0.1"/>
>
> The advantages of using a gateway are:
> - supports authentication
> - will monitor the gateway to detect if it goes down (so calls fail
> instantly rather than after a timeout)
>
> As for the default gateway, it is the IP you send via to reach IPs
> that are not on a network you are connected directly to - you should
> probably only have one set, and it should be the one you go via to
> reach the Internet.
>
> -Steve
>
>
> On 25 February 2010 08:33, Roly Maz <rm at callrica.co.za> wrote:
>> Hi Community
>>
>>
>>
>>
>>
>> My Provider provides the following info when they supply a SIP trunk:
>>
>>
>>
>> ????????? A direct connection into their network. i.e. they provide private
>> IPs:
>>
>> ????????? An IP address I must use for my FS box e.g. IP: 10. 42.0.66 MASK:
>> 255.255.255.248 GW: 10. 42.0.68
>>
>> ????????? An IP address for their SIP server 10.42.0.1
>>
>>
>>
>> I have setup a dual homed FS box (Windows Server 2008, latest FS version)
>>
>>
>>
>> NIC 1 ? Internal LAN IP: 10.0.2.185 MASK: 255.255.255.0 GW: 10.0.2.253
>>
>> NIC 2 ? SIP Provider (WAN) IP: 10. 42.0.66 MASK: 255.255.255.248 GW: 10.
>> 42.0.68
>>
>>
>>
>> Windows complains about multiple gateways ? which I ignore? I can ping
>> internal addresses ?and the SIP Server
>>
>>
>>
>> When I fire up FS, I can register Xlite phones on my LAN. I can dial and
>> hear the test IVR (5000)
>>
>>
>>
>> This means my Internal SIP Profile is ok.
>>
>>
>>
>> Now, how do i route a call out to the 10.42.01 SIP Server?
>>
>>
>>
>> ?Creating a gateway doesn?t make sense, because I am not supplied a
>> username/password?
>>
>>
>>
>> Any pointers would be most appreciated, I am sure I am missing something
>> really simple.
>>
>>
>>
>> Roland
>>
>>
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>



From brian at microcomaustralia.com.au  Thu Feb 25 01:07:27 2010
From: brian at microcomaustralia.com.au (Brian May)
Date: Thu, 25 Feb 2010 20:07:27 +1100
Subject: [Freeswitch-users] Retrieving voicemail without entering user
	ID (extension)
In-Reply-To: <000f01cab5c7$d7e292f0$87a7b8d0$@com>
References: <093DD565390C1E4FB15D7B383E86BB05AFF6BC@Goose.personalcyberspace.net>
	<b39273531002241601m273ff333y2261d930dbcb3fa7@mail.gmail.com>
	<000f01cab5c7$d7e292f0$87a7b8d0$@com>
Message-ID: <3c5cf5261002250107l3a4ff2fan8803c622ce59021e@mail.gmail.com>

On 25 February 2010 14:08, Lars Zeb <larclap at yahoo.com> wrote:
> I used the extension below, but I think that Brian said it was too insecure.
> Being a total beginner, I removed the condition.

Too insecure for what? I think it really depends on the installation,
what the phone are for, where the phones are positioned, who has
access, etc.

I could imagine scenarios where being able to "walk up to anyone's
phone and retrieve their VM w/out authentication" might be considered
a feature. e.g. home office. At my home, under my asterisk setup, it
always seems to be up to me to delete the messages, because others
consider it too complicated to log in. Generally people are use to
being able to walk up to an answering machine, push a button, and
retrieve messages without any authentication.

Then again, making this the default configuration would be a bad idea.
People need to understand the consequences first.

Oh, just a minor nitpick, or possibly an opportunity for me to learn
<grin>. I see near the top that there is an export immediately after
the set. Is the set really needed? I thought the export would override
this? Why is export needed?
-- 
Brian May <brian at microcomaustralia.com.au>



From srinivas.ksvreddy at gmail.com  Thu Feb 25 01:24:01 2010
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Thu, 25 Feb 2010 14:54:01 +0530
Subject: [Freeswitch-users] freeswitch to gateway
Message-ID: <f8af5741002250124s5f3a200fp178d83a90945d73b@mail.gmail.com>

Hi Good afternoon everybody,

my freeswitch domain name is gw.proxy.com, i have registered 1000 to
freeswitch,  i have configured a gateway(gateway.com) to my freeswitch, i
want to make a call to gateway from 1000 to 1003 registered in gateway.com ,
how can i make call.

Thanks--
Srinivasula Reddy K
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100225/27a1fc0f/attachment-0002.html 

From technical at ttnc.co.uk  Thu Feb 25 03:35:14 2010
From: technical at ttnc.co.uk (TTNC - Technical)
Date: Thu, 25 Feb 2010 11:35:14 +0000
Subject: [Freeswitch-users] When using bridge_answer_timeout,
	hangup_after_bridge isn't respected.
Message-ID: <88A1EC04-4060-442E-8DC3-9CD214D48C18@ttnc.co.uk>

I'm pretty sure this is a bug, I've already opened a jira: http://jira.freeswitch.org/browse/FSCORE-561

But I thought after I'd done it it'd probably be an idea to ask here first incase I'm missing something obvious... When using bridge_answer_timeout, hangup_after_bridge isn't respected. In as much as the aleg hangs up the call and it will go back and continue executing the dialplan, in my case - the lua script. 

Anyone got any ideas how else to force freeswitch to end the call after a hangup other than using hangup_after_bridge?

Thanks





From m.krivushin at imarto.net  Thu Feb 25 04:59:11 2010
From: m.krivushin at imarto.net (Mikhail Krivushin)
Date: Thu, 25 Feb 2010 18:59:11 +0600
Subject: [Freeswitch-users] Video pass problem
Message-ID: <5be734a51002250459wb974018ue1ffd7d7a88ace59@mail.gmail.com>

Hello! We have problem with pass video over FreeSWITCH. I tshark traf, and
see that we have 1280 video packets input, and only 560 passed to B leg.
Anyone can point me to right direction? I can send pcap file by request.

Most time I have black screen, and then one frame can appear, and I see
frozen picture pair minutes, and then other frozen picture. We have not
network issues, we have good video, when bypass FS. We have not perfomance
troubles to.

We have ubuntu 9.10 x64, and powerfull server board. uname:
Linux fs 2.6.31-17-generic #54-Ubuntu SMP Thu Dec 10 17:01:44 UTC 2009
x86_64 GNU/Linux

config:
  <X-PRE-PROCESS cmd="set" data="global_codec_prefs=BV32,BV16,speex at 8000h
@20i,speex at 16000h@20i,speex at 32000h
@20i,PCMU,PCMA,GSM,H264,H263,H263-1998,H263-2000,H261"/>
  <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=BV32,BV16,speex at 8000h
@20i,speex at 16000h@20i,speex at 32000h
@20i,PCMU,PCMA,GSM,H264,H263,H263-1998,H263-2000,H261"/>


-- 
? ?????????, ???????? ??????
?. ????? ???. +7 913 865 78 66
icq: 218 744 127
xmpp: KrivushinME at jabber.ru
skype: mkrivushin
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100225/f549fc79/attachment-0002.html 

From brian at freeswitch.org  Thu Feb 25 05:33:17 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 25 Feb 2010 07:33:17 -0600
Subject: [Freeswitch-users] [Freeswitch-dev] Call for help - adding
	information to the wiki: SIP ALG's
In-Reply-To: <33c87fa31002250043l1edbb87dg2e7a58e879656f06@mail.gmail.com>
References: <87f2f3b91002242008y599f86f0t86f467f8c16b4a20@mail.gmail.com>
	<8b1c9cda1002242248u5e39a4ecvda6a3644527f59bf@mail.gmail.com>
	<33c87fa31002250043l1edbb87dg2e7a58e879656f06@mail.gmail.com>
Message-ID: <EC776797-0320-4A5B-B959-2D3A3BF77314@freeswitch.org>

I believe all SIP ALG's are broken.  :P

/b

On Feb 25, 2010, at 2:43 AM, Mark Campbell-Smith wrote:

> I had an issue with a Thomson SpeedTouch 530 router that was causing
> authentication to fail... its the thread titled 'Forbidden using UDP,
> works with TCP/TLS' that everyone said was caused by a broken ATA ...
> lucky no one chipped in so I could buy a new one (as suggested by
> Anthony) - it wouldn't have helped because the router was broken and
> modifying the authentication parameters!
> 
> I'll update the wiki with the info




From rm at callrica.co.za  Thu Feb 25 07:02:56 2010
From: rm at callrica.co.za (Roly Maz)
Date: Thu, 25 Feb 2010 17:02:56 +0200
Subject: [Freeswitch-users] SIP Trunk with Private Static IP?
In-Reply-To: <eafa772f1002250058i2cd7b30ai42f44d0ea1d6eb88@mail.gmail.com>
References: <008401cab5f5$4795b910$d6c12b30$@co.za>	<eafa772f1002250057l3de5b2abu54da8725cbb77281@mail.gmail.com>
	<eafa772f1002250058i2cd7b30ai42f44d0ea1d6eb88@mail.gmail.com>
Message-ID: <009b01cab62b$b8889c10$2999d430$@co.za>

Many thanks for your prompt reply and the help 

I removed the LAN GW and kept the WAN GW. 

I have modified the standard internal and external sip profiles accordingly

What is odd is that if i run a ping from the windows command line, I get a
reply from the SIP Server. However, if I setup a ping within FS, it fails. 

I am investigating...




-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Steven
Ayre
Sent: 25 February 2010 10:59 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] SIP Trunk with Private Static IP?

Gateways do not require usernames and passwords. You are required to
set the parameter, but if no authentication is needed they are ignored
so you can put anything in the field, so that is not a reason to avoid
them.

-Steve


On 25 February 2010 08:57, Steven Ayre <steveayre at gmail.com> wrote:
> Create two SIP profiles, each bound to one of your local IPs.
>
> You may create a gateway on the profile for the SIP trunk IP for the
> 10.42.0.1 server, but this is optional.
>
> You can then bridge calls via the SIP server using one of:
> <action application="bridge" data="sofia/gateway/gatewayname"/>
> <action application="bridge" data="sofia/profilename/number at 10.42.0.1"/>
>
> The advantages of using a gateway are:
> - supports authentication
> - will monitor the gateway to detect if it goes down (so calls fail
> instantly rather than after a timeout)
>
> As for the default gateway, it is the IP you send via to reach IPs
> that are not on a network you are connected directly to - you should
> probably only have one set, and it should be the one you go via to
> reach the Internet.
>
> -Steve
>
>
> On 25 February 2010 08:33, Roly Maz <rm at callrica.co.za> wrote:
>> Hi Community
>>
>>
>>
>>
>>
>> My Provider provides the following info when they supply a SIP trunk:
>>
>>
>>
>> ????????? A direct connection into their network. i.e. they provide
private
>> IPs:
>>
>> ????????? An IP address I must use for my FS box e.g. IP: 10. 42.0.66
MASK:
>> 255.255.255.248 GW: 10. 42.0.68
>>
>> ????????? An IP address for their SIP server 10.42.0.1
>>
>>
>>
>> I have setup a dual homed FS box (Windows Server 2008, latest FS version)
>>
>>
>>
>> NIC 1 ? Internal LAN IP: 10.0.2.185 MASK: 255.255.255.0 GW: 10.0.2.253
>>
>> NIC 2 ? SIP Provider (WAN) IP: 10. 42.0.66 MASK: 255.255.255.248 GW: 10.
>> 42.0.68
>>
>>
>>
>> Windows complains about multiple gateways ? which I ignore? I can ping
>> internal addresses ?and the SIP Server
>>
>>
>>
>> When I fire up FS, I can register Xlite phones on my LAN. I can dial and
>> hear the test IVR (5000)
>>
>>
>>
>> This means my Internal SIP Profile is ok.
>>
>>
>>
>> Now, how do i route a call out to the 10.42.01 SIP Server?
>>
>>
>>
>> ?Creating a gateway doesn?t make sense, because I am not supplied a
>> username/password?
>>
>>
>>
>> Any pointers would be most appreciated, I am sure I am missing something
>> really simple.
>>
>>
>>
>> Roland
>>
>>
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




From peter.olsson at visionutveckling.se  Thu Feb 25 07:08:18 2010
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Thu, 25 Feb 2010 16:08:18 +0100
Subject: [Freeswitch-users] "hold" tone when dialing into FS
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C557729ECBB@cooper>

I guess this is a really stupid question, but I can't find anything about it in my config files...

When dialing into the dialplan, and I just execute "answer" and then "park", I get a ring tone played for me. But I just can't find where this ring tone can be specified, it also seems to play if I execute answer and then sleep. I want to replace this ringtone with another sound, so that's why I'm asking...

Regards,

Peter Olsson

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100225/59f5c362/attachment-0002.html 

From phunk0000 at hotmail.com  Thu Feb 25 07:54:05 2010
From: phunk0000 at hotmail.com (Todd)
Date: Thu, 25 Feb 2010 10:54:05 -0500
Subject: [Freeswitch-users] mod_shout
Message-ID: <BLU110-DS76D3E972959D2ADAB8540A7400@phx.gbl>

Hey everybody.. I followed all the directions on the wiki for mod_shout, but
I still get the following in the load up log of FS..  Any ideas on how to
get this module to work?  Thanks..

 

2010-02-25 10:39:02.473513 [CRIT] switch_loadable_module.c:872 Error Loading
module /usr/local/freeswitch/mod/mod_shout.so

**/usr/local/freeswitch/mod/mod_shout.so: cannot open shared object file: No
such file or directory**

 

.

 

2010-02-25 10:39:03.471572 [ERR] switch_core_file.c:122 Invalid file format
[mp3] for [/usr/local/freeswitch/sounds/music/8000/06 - Free The Robots -
Session Two.mp3]!

2010-02-25 10:39:03.471572 [ERR] mod_local_stream.c:210 Can't open
/usr/local/freeswitch/sounds/music/8000/06 - Free The Robots - Session
Two.mp3

2010-02-25 10:39:04.473632 [ERR] switch_core_file.c:122 Invalid file format
[mp3] for [/usr/local/freeswitch/sounds/music/8000/01 - Free The Robots -
Listen To The Future.mp3]!

2010-02-25 10:39:04.473632 [ERR] mod_local_stream.c:210 Can't open
/usr/local/freeswitch/sounds/music/8000/01 - Free The Robots - Listen To The
Future.mp3

2010-02-25 10:39:04.473632 [ERR] mod_local_stream.c:210 Can't open
/usr/local/freeswitch/sounds/music/8000/01 - Free The Robots - Listen To The
Future.mp3

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100225/c147a66c/attachment-0002.html 

From brian at freeswitch.org  Thu Feb 25 08:01:22 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 25 Feb 2010 10:01:22 -0600
Subject: [Freeswitch-users] mod_shout
In-Reply-To: <BLU110-DS76D3E972959D2ADAB8540A7400@phx.gbl>
References: <BLU110-DS76D3E972959D2ADAB8540A7400@phx.gbl>
Message-ID: <6A0E1E89-69A2-4E6F-959E-82A377D39107@freeswitch.org>

you need to build and load mod_shout... and you also need to stop hijacking threads... When you compose a message to the list Click NEW then input the list address then the subject and then type your body.  By all means DO NOT click reply, change the subject and delete the body.  That is HOW you hijack a thread... even the archives will list it as hijacked.

edit modules.conf in the src tree and uncomment the one about mod_shout... then make mod_shout-install

/b

On Feb 25, 2010, at 9:54 AM, Todd wrote:

> Hey everybody?. I followed all the directions on the wiki for mod_shout, but I still get the following in the load up log of FS?.  Any ideas on how to get this module to work?  Thanks..
>  
> 2010-02-25 10:39:02.473513 [CRIT] switch_loadable_module.c:872 Error Loading module /usr/local/freeswitch/mod/mod_shout.so
> **/usr/local/freeswitch/mod/mod_shout.so: cannot open shared object file: No such file or directory**
>  
> ?
>  
> 2010-02-25 10:39:03.471572 [ERR] switch_core_file.c:122 Invalid file format [mp3] for [/usr/local/freeswitch/sounds/music/8000/06 - Free The Robots - Session Two.mp3]!
> 2010-02-25 10:39:03.471572 [ERR] mod_local_stream.c:210 Can't open /usr/local/freeswitch/sounds/music/8000/06 - Free The Robots - Session Two.mp3
> 2010-02-25 10:39:04.473632 [ERR] switch_core_file.c:122 Invalid file format [mp3] for [/usr/local/freeswitch/sounds/music/8000/01 - Free The Robots - Listen To The Future.mp3]!
> 2010-02-25 10:39:04.473632 [ERR] mod_local_stream.c:210 Can't open /usr/local/freeswitch/sounds/music/8000/01 - Free The Robots - Listen To The Future.mp3
> 2010-02-25 10:39:04.473632 [ERR] mod_local_stream.c:210 Can't open /usr/local/freeswitch/sounds/music/8000/01 - Free The Robots - Listen To The Future.mp3
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100225/ab19953c/attachment-0002.html 

From m.krivushin at imarto.net  Thu Feb 25 08:03:35 2010
From: m.krivushin at imarto.net (Mikhail Krivushin)
Date: Thu, 25 Feb 2010 22:03:35 +0600
Subject: [Freeswitch-users] mod_shout
In-Reply-To: <BLU110-DS76D3E972959D2ADAB8540A7400@phx.gbl>
References: <BLU110-DS76D3E972959D2ADAB8540A7400@phx.gbl>
Message-ID: <5be734a51002250803h7e3863bg428c205cf300cc8a@mail.gmail.com>

check permissions on /usr/local/freeswitch/mod/mod_shout.so
It must be redeable by FS user.

2010/2/25 Todd <phunk0000 at hotmail.com>

>  Hey everybody?. I followed all the directions on the wiki for mod_shout,
> but I still get the following in the load up log of FS?.  Any ideas on how
> to get this module to work?  Thanks..
>
>
>
> 2010-02-25 10:39:02.473513 [CRIT] switch_loadable_module.c:872 Error
> Loading module /usr/local/freeswitch/mod/mod_shout.so
>
> **/usr/local/freeswitch/mod/mod_shout.so: cannot open shared object file:
> No such file or directory**
>
>
>
> ?
>
>
>
> 2010-02-25 10:39:03.471572 [ERR] switch_core_file.c:122 Invalid file format
> [mp3] for [/usr/local/freeswitch/sounds/music/8000/06 - Free The Robots -
> Session Two.mp3]!
>
> 2010-02-25 10:39:03.471572 [ERR] mod_local_stream.c:210 Can't open
> /usr/local/freeswitch/sounds/music/8000/06 - Free The Robots - Session
> Two.mp3
>
> 2010-02-25 10:39:04.473632 [ERR] switch_core_file.c:122 Invalid file format
> [mp3] for [/usr/local/freeswitch/sounds/music/8000/01 - Free The Robots -
> Listen To The Future.mp3]!
>
> 2010-02-25 10:39:04.473632 [ERR] mod_local_stream.c:210 Can't open
> /usr/local/freeswitch/sounds/music/8000/01 - Free The Robots - Listen To The
> Future.mp3
>
> 2010-02-25 10:39:04.473632 [ERR] mod_local_stream.c:210 Can't open
> /usr/local/freeswitch/sounds/music/8000/01 - Free The Robots - Listen To The
> Future.mp3
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
? ?????????, ???????? ??????
?. ????? ???. +7 913 865 78 66
icq: 218 744 127
xmpp: KrivushinME at jabber.ru
skype: mkrivushin
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100225/1ab2215d/attachment-0002.html 

From brian at freeswitch.org  Thu Feb 25 08:07:13 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 25 Feb 2010 10:07:13 -0600
Subject: [Freeswitch-users] mod_shout
In-Reply-To: <5be734a51002250803h7e3863bg428c205cf300cc8a@mail.gmail.com>
References: <BLU110-DS76D3E972959D2ADAB8540A7400@phx.gbl>
	<5be734a51002250803h7e3863bg428c205cf300cc8a@mail.gmail.com>
Message-ID: <E7216619-05EB-4285-AA8D-51326770F6D0@freeswitch.org>

I think it would have said something besides "No such file or directory" if that were the case.  Its not in the default compile.

/b

On Feb 25, 2010, at 10:03 AM, Mikhail Krivushin wrote:

> check permissions on /usr/local/freeswitch/mod/mod_shout.so
> It must be redeable by FS user.




From chris.chen2004 at gmail.com  Thu Feb 25 08:08:37 2010
From: chris.chen2004 at gmail.com (Chris Chen)
Date: Thu, 25 Feb 2010 11:08:37 -0500
Subject: [Freeswitch-users] mod_shout
In-Reply-To: <BLU110-DS76D3E972959D2ADAB8540A7400@phx.gbl>
References: <BLU110-DS76D3E972959D2ADAB8540A7400@phx.gbl>
Message-ID: <507898381002250808s75aca323yb43544513dcaa9f5@mail.gmail.com>

Your mod_shout is not loaded,
make sure you compiled the shout correctly, and load the shout in the
/usr/local/freeswitch/conf/auto_configs/modules.conf.xml

Chris


On Thu, Feb 25, 2010 at 10:54 AM, Todd <phunk0000 at hotmail.com> wrote:

>  Hey everybody?. I followed all the directions on the wiki for mod_shout,
> but I still get the following in the load up log of FS?.  Any ideas on how
> to get this module to work?  Thanks..
>
>
>
> 2010-02-25 10:39:02.473513 [CRIT] switch_loadable_module.c:872 Error
> Loading module /usr/local/freeswitch/mod/mod_shout.so
>
> **/usr/local/freeswitch/mod/mod_shout.so: cannot open shared object file:
> No such file or directory**
>
>
>
> ?
>
>
>
> 2010-02-25 10:39:03.471572 [ERR] switch_core_file.c:122 Invalid file format
> [mp3] for [/usr/local/freeswitch/sounds/music/8000/06 - Free The Robots -
> Session Two.mp3]!
>
> 2010-02-25 10:39:03.471572 [ERR] mod_local_stream.c:210 Can't open
> /usr/local/freeswitch/sounds/music/8000/06 - Free The Robots - Session
> Two.mp3
>
> 2010-02-25 10:39:04.473632 [ERR] switch_core_file.c:122 Invalid file format
> [mp3] for [/usr/local/freeswitch/sounds/music/8000/01 - Free The Robots -
> Listen To The Future.mp3]!
>
> 2010-02-25 10:39:04.473632 [ERR] mod_local_stream.c:210 Can't open
> /usr/local/freeswitch/sounds/music/8000/01 - Free The Robots - Listen To The
> Future.mp3
>
> 2010-02-25 10:39:04.473632 [ERR] mod_local_stream.c:210 Can't open
> /usr/local/freeswitch/sounds/music/8000/01 - Free The Robots - Listen To The
> Future.mp3
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100225/02d4013d/attachment-0002.html 

From phunk0000 at hotmail.com  Thu Feb 25 08:15:21 2010
From: phunk0000 at hotmail.com (Todd)
Date: Thu, 25 Feb 2010 11:15:21 -0500
Subject: [Freeswitch-users] mod_shout
In-Reply-To: <6A0E1E89-69A2-4E6F-959E-82A377D39107@freeswitch.org>
References: <BLU110-DS76D3E972959D2ADAB8540A7400@phx.gbl>
	<6A0E1E89-69A2-4E6F-959E-82A377D39107@freeswitch.org>
Message-ID: <BLU110-DS192C502EC1E6527CEC7657A7400@phx.gbl>

?  That's what I did thanks.  brand new email to
freeswitch-users at lists.freeswitch.org   didn't reply to anything.  I
followed all the directions in the mod_shout wiki as far as running
./configure and make; make install.. Are you talking about something other
than that?

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: Thursday, February 25, 2010 11:01 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] mod_shout

 

you need to build and load mod_shout... and you also need to stop hijacking
threads... When you compose a message to the list Click NEW then input the
list address then the subject and then type your body.  By all means DO NOT
click reply, change the subject and delete the body.  That is HOW you hijack
a thread... even the archives will list it as hijacked.

 

edit modules.conf in the src tree and uncomment the one about mod_shout...
then make mod_shout-install

 

/b

 

On Feb 25, 2010, at 9:54 AM, Todd wrote:





Hey everybody.. I followed all the directions on the wiki for mod_shout, but
I still get the following in the load up log of FS..  Any ideas on how to
get this module to work?  Thanks..

 

2010-02-25 10:39:02.473513 [CRIT] switch_loadable_module.c:872 Error Loading
module /usr/local/freeswitch/mod/mod_shout.so

**/usr/local/freeswitch/mod/mod_shout.so: cannot open shared object file: No
such file or directory**

 

.

 

2010-02-25 10:39:03.471572 [ERR] switch_core_file.c:122 Invalid file format
[mp3] for [/usr/local/freeswitch/sounds/music/8000/06 - Free The Robots -
Session Two.mp3]!

2010-02-25 10:39:03.471572 [ERR] mod_local_stream.c:210 Can't open
/usr/local/freeswitch/sounds/music/8000/06 - Free The Robots - Session
Two.mp3

2010-02-25 10:39:04.473632 [ERR] switch_core_file.c:122 Invalid file format
[mp3] for [/usr/local/freeswitch/sounds/music/8000/01 - Free The Robots -
Listen To The Future.mp3]!

2010-02-25 10:39:04.473632 [ERR] mod_local_stream.c:210 Can't open
/usr/local/freeswitch/sounds/music/8000/01 - Free The Robots - Listen To The
Future.mp3

2010-02-25 10:39:04.473632 [ERR] mod_local_stream.c:210 Can't open
/usr/local/freeswitch/sounds/music/8000/01 - Free The Robots - Listen To The
Future.mp3

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org <http://www.freeswitch.org/> 

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100225/5b94ab0d/attachment-0002.html 

From brian at freeswitch.org  Thu Feb 25 08:19:54 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 25 Feb 2010 10:19:54 -0600
Subject: [Freeswitch-users] mod_shout
In-Reply-To: <BLU110-DS192C502EC1E6527CEC7657A7400@phx.gbl>
References: <BLU110-DS76D3E972959D2ADAB8540A7400@phx.gbl>
	<6A0E1E89-69A2-4E6F-959E-82A377D39107@freeswitch.org>
	<BLU110-DS192C502EC1E6527CEC7657A7400@phx.gbl>
Message-ID: <A3BF8C8C-82CB-4099-B008-9E049566C236@freeswitch.org>

Seems you got mixed in with another naughty thread hijacker :)  Its ok... Sorry for pinning it on you... ;)

/b

On Feb 25, 2010, at 10:15 AM, Todd wrote:

> ?  That?s what I did thanks?  brand new email to    freeswitch-users at lists.freeswitch.org   didn?t reply to anything.  I followed all the directions in the mod_shout wiki as far as running ./configure and make; make install?. Are you talking about something other than that?
>  

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100225/564b2243/attachment-0002.html 

From phunk0000 at hotmail.com  Thu Feb 25 08:24:11 2010
From: phunk0000 at hotmail.com (Todd)
Date: Thu, 25 Feb 2010 11:24:11 -0500
Subject: [Freeswitch-users] mod_shout
In-Reply-To: <507898381002250808s75aca323yb43544513dcaa9f5@mail.gmail.com>
References: <BLU110-DS76D3E972959D2ADAB8540A7400@phx.gbl>
	<507898381002250808s75aca323yb43544513dcaa9f5@mail.gmail.com>
Message-ID: <BLU110-DS3D63E1802B946A5B42D11A7400@phx.gbl>

Followed the directions in the wiki to compile, already loaded in
auto_configs/modules.conf.xml and in freeswitch/modules.conf...   is there
some other way/directions  to compile?

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Chris
Chen
Sent: Thursday, February 25, 2010 11:09 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] mod_shout

 

Your mod_shout is not loaded,
make sure you compiled the shout correctly, and load the shout in the
/usr/local/freeswitch/conf/auto_configs/modules.conf.xml

Chris



On Thu, Feb 25, 2010 at 10:54 AM, Todd <phunk0000 at hotmail.com> wrote:

Hey everybody.. I followed all the directions on the wiki for mod_shout, but
I still get the following in the load up log of FS..  Any ideas on how to
get this module to work?  Thanks..

 

2010-02-25 10:39:02.473513 [CRIT] switch_loadable_module.c:872 Error Loading
module /usr/local/freeswitch/mod/mod_shout.so

**/usr/local/freeswitch/mod/mod_shout.so: cannot open shared object file: No
such file or directory**

 

.

 

2010-02-25 10:39:03.471572 [ERR] switch_core_file.c:122 Invalid file format
[mp3] for [/usr/local/freeswitch/sounds/music/8000/06 - Free The Robots -
Session Two.mp3]!

2010-02-25 10:39:03.471572 [ERR] mod_local_stream.c:210 Can't open
/usr/local/freeswitch/sounds/music/8000/06 - Free The Robots - Session
Two.mp3

2010-02-25 10:39:04.473632 [ERR] switch_core_file.c:122 Invalid file format
[mp3] for [/usr/local/freeswitch/sounds/music/8000/01 - Free The Robots -
Listen To The Future.mp3]!

2010-02-25 10:39:04.473632 [ERR] mod_local_stream.c:210 Can't open
/usr/local/freeswitch/sounds/music/8000/01 - Free The Robots - Listen To The
Future.mp3

2010-02-25 10:39:04.473632 [ERR] mod_local_stream.c:210 Can't open
/usr/local/freeswitch/sounds/music/8000/01 - Free The Robots - Listen To The
Future.mp3


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100225/93ee5b86/attachment-0002.html 

From brian at freeswitch.org  Thu Feb 25 08:31:28 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 25 Feb 2010 10:31:28 -0600
Subject: [Freeswitch-users] mod_shout
In-Reply-To: <BLU110-DS3D63E1802B946A5B42D11A7400@phx.gbl>
References: <BLU110-DS76D3E972959D2ADAB8540A7400@phx.gbl>
	<507898381002250808s75aca323yb43544513dcaa9f5@mail.gmail.com>
	<BLU110-DS3D63E1802B946A5B42D11A7400@phx.gbl>
Message-ID: <A81F3F31-7D03-4F4E-83CF-16F6238AA3D4@freeswitch.org>

While you're going down this road i'm going to highly recommend you turn around and go back to wav files.... MP3 is overly CPU hungry and if by chance you get some MP3 data thats invalid you have a chance of crashing the decoder...

/b

On Feb 25, 2010, at 10:24 AM, Todd wrote:

> Followed the directions in the wiki to compile, already loaded in auto_configs/modules.conf.xml and in freeswitch/modules.conf?..   is there some other way/directions  to compile?
>  

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100225/05d82da1/attachment-0002.html 

From jonas.gauffin at gmail.com  Thu Feb 25 08:36:00 2010
From: jonas.gauffin at gmail.com (Jonas Gauffin)
Date: Thu, 25 Feb 2010 17:36:00 +0100
Subject: [Freeswitch-users] bind_meta_app
Message-ID: <d021d3be1002250836p7010371epf943f5d68e77d5bf@mail.gmail.com>

Hello,

I'm trying to use bind_meta_app together with variables defined by me in the
original dial plan.
The problem is that they doesn't seem to follow the channel into the dial
plan when the meta application is running.

i.e.

Dialing from an external number:
<document type="freeswitch/xml">
        <section name="dialplan" description="ModCurlDialplan">
                <context name="public">
                        <extension name="dynamic">
                                <condition field="destination_number"
expression="0500650662">
                                        <action application="set"
data="hangup_after_bridge=true"/>
                                        <action application="set"
data="continue_on_fail=true"/>
                                        <action application="set"
data="gate_caller_site_id=3"/>
                                        <action application="bind_meta_app"
data="1 b s execute_extension::dx XML default"/>
                                        <action application="bind_meta_app"
data="3 b s execute_extension::cf XML default"/>
                                        <action application="set"
data="gate_bill_extension=95" />
                                        <action application="set"
data="gate_destination_user=6" />
                                        <action application="ring_ready"/>
                                        <action application="bridge"
data="[leg_timeout=5]sofia/internal/u1000006" />
                                        <action application="set"
data="gate_ivr=voicemail" />
                                        <action application="javascript"
data="voicemail.js customer 3 6" />
                                <action application="hangup"
data="NO_ANSWER"/>

                        </condition>
                </extension>
        </context>
</section>
</document>

Destination user pressed *1

Freeswitch sends second request through mod_curl to my server. My
variable_gate_XXXXX variables is not defined.

Regards,
  Jonas
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100225/8b21de39/attachment-0002.html 

From phunk0000 at hotmail.com  Thu Feb 25 08:40:13 2010
From: phunk0000 at hotmail.com (Todd)
Date: Thu, 25 Feb 2010 11:40:13 -0500
Subject: [Freeswitch-users] mod_shout
In-Reply-To: <A81F3F31-7D03-4F4E-83CF-16F6238AA3D4@freeswitch.org>
References: <BLU110-DS76D3E972959D2ADAB8540A7400@phx.gbl>	<507898381002250808s75aca323yb43544513dcaa9f5@mail.gmail.com>	<BLU110-DS3D63E1802B946A5B42D11A7400@phx.gbl>
	<A81F3F31-7D03-4F4E-83CF-16F6238AA3D4@freeswitch.org>
Message-ID: <BLU110-DS15B7839071CA1439C6ADF7A7400@phx.gbl>

Good to know.  I was really just trying to make it work.. Don't need it for
anything

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: Thursday, February 25, 2010 11:31 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] mod_shout

 

While you're going down this road i'm going to highly recommend you turn
around and go back to wav files.... MP3 is overly CPU hungry and if by
chance you get some MP3 data thats invalid you have a chance of crashing the
decoder...

 

/b

 

On Feb 25, 2010, at 10:24 AM, Todd wrote:





Followed the directions in the wiki to compile, already loaded in
auto_configs/modules.conf.xml and in freeswitch/modules.conf...   is there
some other way/directions  to compile?

 

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100225/525b67b3/attachment-0002.html 

From anthony.minessale at gmail.com  Thu Feb 25 08:43:38 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 25 Feb 2010 10:43:38 -0600
Subject: [Freeswitch-users] bind_meta_app
In-Reply-To: <d021d3be1002250836p7010371epf943f5d68e77d5bf@mail.gmail.com>
References: <d021d3be1002250836p7010371epf943f5d68e77d5bf@mail.gmail.com>
Message-ID: <191c3a031002250843l70761215q2cc98b4b2e2d3660@mail.gmail.com>

because you are executing the app on the B leg.
you need to set the vars on *that* channel if you want to see them.


On Thu, Feb 25, 2010 at 10:36 AM, Jonas Gauffin <jonas.gauffin at gmail.com>wrote:

> Hello,
>
> I'm trying to use bind_meta_app together with variables defined by me in
> the original dial plan.
> The problem is that they doesn't seem to follow the channel into the dial
> plan when the meta application is running.
>
> i.e.
>
> Dialing from an external number:
> <document type="freeswitch/xml">
>         <section name="dialplan" description="ModCurlDialplan">
>                 <context name="public">
>                         <extension name="dynamic">
>                                 <condition field="destination_number"
> expression="0500650662">
>                                         <action application="set"
> data="hangup_after_bridge=true"/>
>                                         <action application="set"
> data="continue_on_fail=true"/>
>                                         <action application="set"
> data="gate_caller_site_id=3"/>
>                                         <action application="bind_meta_app"
> data="1 b s execute_extension::dx XML default"/>
>                                         <action application="bind_meta_app"
> data="3 b s execute_extension::cf XML default"/>
>                                         <action application="set"
> data="gate_bill_extension=95" />
>                                         <action application="set"
> data="gate_destination_user=6" />
>                                         <action application="ring_ready"/>
>                                         <action application="bridge"
> data="[leg_timeout=5]sofia/internal/u1000006" />
>                                         <action application="set"
> data="gate_ivr=voicemail" />
>                                         <action application="javascript"
> data="voicemail.js customer 3 6" />
>                                 <action application="hangup"
> data="NO_ANSWER"/>
>
>                         </condition>
>                 </extension>
>         </context>
> </section>
> </document>
>
> Destination user pressed *1
>
> Freeswitch sends second request through mod_curl to my server. My
> variable_gate_XXXXX variables is not defined.
>
> Regards,
>   Jonas
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100225/fe4d1f87/attachment-0002.html 

From anthony.minessale at gmail.com  Thu Feb 25 08:44:21 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 25 Feb 2010 10:44:21 -0600
Subject: [Freeswitch-users] bind_meta_app
In-Reply-To: <191c3a031002250843l70761215q2cc98b4b2e2d3660@mail.gmail.com>
References: <d021d3be1002250836p7010371epf943f5d68e77d5bf@mail.gmail.com>
	<191c3a031002250843l70761215q2cc98b4b2e2d3660@mail.gmail.com>
Message-ID: <191c3a031002250844k328a8b1fh42237da31e53d466@mail.gmail.com>

btw,

you can set the variable export_vars=<comma sep list of var names>
to copy all the variables to any B leg that may be spawned.


On Thu, Feb 25, 2010 at 10:43 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> because you are executing the app on the B leg.
> you need to set the vars on *that* channel if you want to see them.
>
>
> On Thu, Feb 25, 2010 at 10:36 AM, Jonas Gauffin <jonas.gauffin at gmail.com>wrote:
>
>> Hello,
>>
>> I'm trying to use bind_meta_app together with variables defined by me in
>> the original dial plan.
>> The problem is that they doesn't seem to follow the channel into the dial
>> plan when the meta application is running.
>>
>> i.e.
>>
>> Dialing from an external number:
>> <document type="freeswitch/xml">
>>         <section name="dialplan" description="ModCurlDialplan">
>>                 <context name="public">
>>                         <extension name="dynamic">
>>                                 <condition field="destination_number"
>> expression="0500650662">
>>                                         <action application="set"
>> data="hangup_after_bridge=true"/>
>>                                         <action application="set"
>> data="continue_on_fail=true"/>
>>                                         <action application="set"
>> data="gate_caller_site_id=3"/>
>>                                         <action
>> application="bind_meta_app" data="1 b s execute_extension::dx XML default"/>
>>                                         <action
>> application="bind_meta_app" data="3 b s execute_extension::cf XML default"/>
>>                                         <action application="set"
>> data="gate_bill_extension=95" />
>>                                         <action application="set"
>> data="gate_destination_user=6" />
>>                                         <action application="ring_ready"/>
>>                                         <action application="bridge"
>> data="[leg_timeout=5]sofia/internal/u1000006" />
>>                                         <action application="set"
>> data="gate_ivr=voicemail" />
>>                                         <action application="javascript"
>> data="voicemail.js customer 3 6" />
>>                                 <action application="hangup"
>> data="NO_ANSWER"/>
>>
>>                         </condition>
>>                 </extension>
>>         </context>
>> </section>
>> </document>
>>
>> Destination user pressed *1
>>
>> Freeswitch sends second request through mod_curl to my server. My
>> variable_gate_XXXXX variables is not defined.
>>
>> Regards,
>>   Jonas
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100225/a2a3eddd/attachment-0002.html 

From jonas.gauffin at gmail.com  Thu Feb 25 08:56:05 2010
From: jonas.gauffin at gmail.com (Jonas Gauffin)
Date: Thu, 25 Feb 2010 17:56:05 +0100
Subject: [Freeswitch-users] bind_meta_app
In-Reply-To: <191c3a031002250844k328a8b1fh42237da31e53d466@mail.gmail.com>
References: <d021d3be1002250836p7010371epf943f5d68e77d5bf@mail.gmail.com>
	<191c3a031002250843l70761215q2cc98b4b2e2d3660@mail.gmail.com>
	<191c3a031002250844k328a8b1fh42237da31e53d466@mail.gmail.com>
Message-ID: <d021d3be1002250856t56e2df49j63733b4b8f3bf6e6@mail.gmail.com>

Thanks for the quick answer =)

On Thu, Feb 25, 2010 at 5:44 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> btw,
>
> you can set the variable export_vars=<comma sep list of var names>
> to copy all the variables to any B leg that may be spawned.
>
>
>
> On Thu, Feb 25, 2010 at 10:43 AM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> because you are executing the app on the B leg.
>> you need to set the vars on *that* channel if you want to see them.
>>
>>
>> On Thu, Feb 25, 2010 at 10:36 AM, Jonas Gauffin <jonas.gauffin at gmail.com>wrote:
>>
>>> Hello,
>>>
>>> I'm trying to use bind_meta_app together with variables defined by me in
>>> the original dial plan.
>>> The problem is that they doesn't seem to follow the channel into the dial
>>> plan when the meta application is running.
>>>
>>> i.e.
>>>
>>> Dialing from an external number:
>>> <document type="freeswitch/xml">
>>>         <section name="dialplan" description="ModCurlDialplan">
>>>                 <context name="public">
>>>                         <extension name="dynamic">
>>>                                 <condition field="destination_number"
>>> expression="0500650662">
>>>                                         <action application="set"
>>> data="hangup_after_bridge=true"/>
>>>                                         <action application="set"
>>> data="continue_on_fail=true"/>
>>>                                         <action application="set"
>>> data="gate_caller_site_id=3"/>
>>>                                         <action
>>> application="bind_meta_app" data="1 b s execute_extension::dx XML default"/>
>>>                                         <action
>>> application="bind_meta_app" data="3 b s execute_extension::cf XML default"/>
>>>                                         <action application="set"
>>> data="gate_bill_extension=95" />
>>>                                         <action application="set"
>>> data="gate_destination_user=6" />
>>>                                         <action
>>> application="ring_ready"/>
>>>                                         <action application="bridge"
>>> data="[leg_timeout=5]sofia/internal/u1000006" />
>>>                                         <action application="set"
>>> data="gate_ivr=voicemail" />
>>>                                         <action application="javascript"
>>> data="voicemail.js customer 3 6" />
>>>                                 <action application="hangup"
>>> data="NO_ANSWER"/>
>>>
>>>                         </condition>
>>>                 </extension>
>>>         </context>
>>> </section>
>>> </document>
>>>
>>> Destination user pressed *1
>>>
>>> Freeswitch sends second request through mod_curl to my server. My
>>> variable_gate_XXXXX variables is not defined.
>>>
>>> Regards,
>>>   Jonas
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100225/61619343/attachment-0002.html 

From frank at carmickle.com  Thu Feb 25 08:58:28 2010
From: frank at carmickle.com (Frank Carmickle)
Date: Thu, 25 Feb 2010 11:58:28 -0500
Subject: [Freeswitch-users] "hold" tone when dialing into FS
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C557729ECBB@cooper>
References: <549CFEF87AEDE841A38E9D15EAB4C04C557729ECBB@cooper>
Message-ID: <20100225165828.GF9832@base.carmickle.com>

On Thu, Feb 25, Peter Olsson wrote:
> I guess this is a really stupid question, but I can't find anything about it in my config files...
> 
> When dialing into the dialplan, and I just execute "answer" and then "park", I get a ring tone played for me. But I just can't find where this ring tone can be specified, it also seems to play if I execute answer and then sleep. I want to replace this ringtone with another sound, so that's why I'm asking...

<action application="set" data="transfer_ringback=$${hold_music}"/>

--FC



From frank at carmickle.com  Thu Feb 25 09:09:08 2010
From: frank at carmickle.com (Frank Carmickle)
Date: Thu, 25 Feb 2010 12:09:08 -0500
Subject: [Freeswitch-users] freeswitch to gateway
In-Reply-To: <f8af5741002250124s5f3a200fp178d83a90945d73b@mail.gmail.com>
References: <f8af5741002250124s5f3a200fp178d83a90945d73b@mail.gmail.com>
Message-ID: <20100225170907.GG9832@base.carmickle.com>


On Thu, Feb 25, srinivasula reddy wrote:
> Hi Good afternoon everybody,
> 
> my freeswitch domain name is gw.proxy.com, i have registered 1000 to
> freeswitch,  i have configured a gateway(gateway.com) to my freeswitch, i
> want to make a call to gateway from 1000 to 1003 registered in gateway.com ,
> how can i make call.

Add an extension that matches on 1003 and bridge to it
<action application="bridge" data="sofia/gateway/gatewayname/$1"/>

--FC



From peter.olsson at visionutveckling.se  Thu Feb 25 09:10:16 2010
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Thu, 25 Feb 2010 18:10:16 +0100
Subject: [Freeswitch-users] "hold" tone when dialing into FS
In-Reply-To: <20100225165828.GF9832@base.carmickle.com>
References: <549CFEF87AEDE841A38E9D15EAB4C04C557729ECBB@cooper>
	<20100225165828.GF9832@base.carmickle.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C557729ED17@cooper>

Thanks,

However, I figured out that it's probably caused by an Asterisk server that I dial through, when dialing this number. When dialing directly to FS from a SIP phone it's silent (as I expected it to be), but when dialing through a PRI connection on Asterisk, using SIP to FS, the Asterisk seems to generate this sound on the PRI card (while not receiving "real" RTP data from FS).

Sorry for bothering the wrong people :)

/Peter


-----Ursprungligt meddelande-----
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Frank Carmickle
Skickat: den 25 februari 2010 17:58
Till: freeswitch-users at lists.freeswitch.org
?mne: Re: [Freeswitch-users] "hold" tone when dialing into FS

On Thu, Feb 25, Peter Olsson wrote:
> I guess this is a really stupid question, but I can't find anything about it in my config files...
> 
> When dialing into the dialplan, and I just execute "answer" and then "park", I get a ring tone played for me. But I just can't find where this ring tone can be specified, it also seems to play if I execute answer and then sleep. I want to replace this ringtone with another sound, so that's why I'm asking...

<action application="set" data="transfer_ringback=$${hold_music}"/>

--FC

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

!DSPAM:4b86ae0132931189169766!




From lists at redbonez.net  Thu Feb 25 09:18:23 2010
From: lists at redbonez.net (Adam Ford)
Date: Thu, 25 Feb 2010 10:18:23 -0700
Subject: [Freeswitch-users] Retrieving voicemail without entering
	user	ID (extension)
In-Reply-To: <3c5cf5261002250107l3a4ff2fan8803c622ce59021e@mail.gmail.com>
References: <093DD565390C1E4FB15D7B383E86BB05AFF6BC@Goose.personalcyberspace.net>	<b39273531002241601m273ff333y2261d930dbcb3fa7@mail.gmail.com>	<000f01cab5c7$d7e292f0$87a7b8d0$@com>
	<3c5cf5261002250107l3a4ff2fan8803c622ce59021e@mail.gmail.com>
Message-ID: <016c01cab63e$8a4f97a0$9eeec6e0$@net>

I think many of you are missing the fact that he said he still wanted to
have to enter the mailbox password. He just didn't want people to have to
enter the mailbox ID AND the mailbox password.  Still plenty secure as long
as you don't have a default voicemail password for all extensions.

-Adam

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
May
Sent: Thursday, February 25, 2010 2:07 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Retrieving voicemail without entering user
ID (extension)

On 25 February 2010 14:08, Lars Zeb <larclap at yahoo.com> wrote:
> I used the extension below, but I think that Brian said it was too
insecure.
> Being a total beginner, I removed the condition.

Too insecure for what? I think it really depends on the installation,
what the phone are for, where the phones are positioned, who has
access, etc.

I could imagine scenarios where being able to "walk up to anyone's
phone and retrieve their VM w/out authentication" might be considered
a feature. e.g. home office. At my home, under my asterisk setup, it
always seems to be up to me to delete the messages, because others
consider it too complicated to log in. Generally people are use to
being able to walk up to an answering machine, push a button, and
retrieve messages without any authentication.

Then again, making this the default configuration would be a bad idea.
People need to understand the consequences first.

Oh, just a minor nitpick, or possibly an opportunity for me to learn
<grin>. I see near the top that there is an export immediately after
the set. Is the set really needed? I thought the export would override
this? Why is export needed?
-- 
Brian May <brian at microcomaustralia.com.au>

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




From m.sobkow at marketelsystems.com  Thu Feb 25 09:22:07 2010
From: m.sobkow at marketelsystems.com (Mark Sobkow)
Date: Thu, 25 Feb 2010 11:22:07 -0600
Subject: [Freeswitch-users] mod_erlang_event
In-Reply-To: <20100223224902.GB1751@hijacked.us>
References: <4B8441FE.80506@marketelsystems.com>
	<20100223224902.GB1751@hijacked.us>
Message-ID: <4B86B1BF.50809@marketelsystems.com>

An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100225/bce9e99e/attachment-0002.html 

From lists at redbonez.net  Thu Feb 25 09:26:18 2010
From: lists at redbonez.net (Adam Ford)
Date: Thu, 25 Feb 2010 10:26:18 -0700
Subject: [Freeswitch-users] [Freeswitch-dev] Call for help -
	adding	information to the wiki: SIP ALG's
In-Reply-To: <EC776797-0320-4A5B-B959-2D3A3BF77314@freeswitch.org>
References: <87f2f3b91002242008y599f86f0t86f467f8c16b4a20@mail.gmail.com>	<8b1c9cda1002242248u5e39a4ecvda6a3644527f59bf@mail.gmail.com>	<33c87fa31002250043l1edbb87dg2e7a58e879656f06@mail.gmail.com>
	<EC776797-0320-4A5B-B959-2D3A3BF77314@freeswitch.org>
Message-ID: <017001cab63f$a51397c0$ef3ac740$@net>

I wish you would have told me that back when I was trying to solve my
authentication issue that I thought was being caused by using rport. :P

You just told me to upgrade to 1.0.5, when a $%$#! Sonicwall SIP ALG would
have been more helpful, haha. As it turned out it was the Sonicwall SIP
transformations, and switching to TCP SIP resolved it. 

I have added my Sonicwall experience to the wiki page.

-Adam


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: Thursday, February 25, 2010 6:33 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] [Freeswitch-dev] Call for help - adding
information to the wiki: SIP ALG's

I believe all SIP ALG's are broken.  :P

/b

On Feb 25, 2010, at 2:43 AM, Mark Campbell-Smith wrote:

> I had an issue with a Thomson SpeedTouch 530 router that was causing
> authentication to fail... its the thread titled 'Forbidden using UDP,
> works with TCP/TLS' that everyone said was caused by a broken ATA ...
> lucky no one chipped in so I could buy a new one (as suggested by
> Anthony) - it wouldn't have helped because the router was broken and
> modifying the authentication parameters!
> 
> I'll update the wiki with the info


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




From infos at madovsky.org  Thu Feb 25 10:24:54 2010
From: infos at madovsky.org (Madovsky)
Date: Thu, 25 Feb 2010 13:24:54 -0500
Subject: [Freeswitch-users] freeswitch to gateway
References: <f8af5741002250124s5f3a200fp178d83a90945d73b@mail.gmail.com>
	<20100225170907.GG9832@base.carmickle.com>
Message-ID: <74713287E44D44D8BC02EB42F686076F@MOBILEE1705>


----- Original Message ----- 
From: "Frank Carmickle" <frank at carmickle.com>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Thursday, February 25, 2010 12:09 PM
Subject: Re: [Freeswitch-users] freeswitch to gateway


>
> On Thu, Feb 25, srinivasula reddy wrote:
>> Hi Good afternoon everybody,
>>
>> my freeswitch domain name is gw.proxy.com, i have registered 1000 to
>> freeswitch,  i have configured a gateway(gateway.com) to my freeswitch, i
>> want to make a call to gateway from 1000 to 1003 registered in 
>> gateway.com ,
>> how can i make call.
>
> Add an extension that matches on 1003 and bridge to it
> <action application="bridge" data="sofia/gateway/gatewayname/$1"/>
>
> --FC
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

Also look at the "Local_Extension" extension in dialplan/default.xml

F 




From msc at freeswitch.org  Thu Feb 25 11:49:03 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 25 Feb 2010 11:49:03 -0800
Subject: [Freeswitch-users] Multiple gateways dial string and user busy
In-Reply-To: <fbfa7b0c1002240813x58ba8541h7fef5b94f045cbda@mail.gmail.com>
References: <fbfa7b0c1002240813x58ba8541h7fef5b94f045cbda@mail.gmail.com>
Message-ID: <87f2f3b91002251149m13c0d961k722dd3375f46b6e8@mail.gmail.com>

On Wed, Feb 24, 2010 at 8:13 AM, Max Ivanov <ivanov.maxim at gmail.com> wrote:

> Hi all!
>
> when I do test call from fs_cli:
> originate sofia/gateway/panas110/223|sofia/gateway/panas111/223
> &playaback(local_stream://moh)
>
> If firest attempt returns USER_BUSY it tries to call via second one.
> Is it normal? How can I stop calling attempts after first USER_BUSY?
>

This is normal for the syntax you're using. You can try setting
ignore_early_media=true if you don't need call progress tones like ringing
and busy. It might help to know what the application is before answering
your question further. What solution are you building?
-MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100225/39d16a77/attachment-0002.html 

From msc at freeswitch.org  Thu Feb 25 11:54:42 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 25 Feb 2010 11:54:42 -0800
Subject: [Freeswitch-users] qt framework link broken
In-Reply-To: <0CD7185862B54C1ABF67D77BA55664F7@MOBILEE1705>
References: <0CD7185862B54C1ABF67D77BA55664F7@MOBILEE1705>
Message-ID: <87f2f3b91002251154t79f6f9e8q183f0281d4dd8126@mail.gmail.com>

On Wed, Feb 24, 2010 at 12:05 PM, Madovsky <infos at madovsky.org> wrote:

>  Just to inform that
> at the link
> http://wiki.freeswitch.org/wiki/FSComm#Linux
>
> the qt framework link is broken, so as I'm new to this emailist
> I don't want to correct myself on wiki.
>
Okay I fixed it.
-MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100225/e408cbb7/attachment-0002.html 

From msc at freeswitch.org  Thu Feb 25 11:58:14 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 25 Feb 2010 11:58:14 -0800
Subject: [Freeswitch-users] [Freeswitch-dev] Call for help - adding
	information to the wiki: SIP ALG's
In-Reply-To: <017001cab63f$a51397c0$ef3ac740$@net>
References: <87f2f3b91002242008y599f86f0t86f467f8c16b4a20@mail.gmail.com>
	<8b1c9cda1002242248u5e39a4ecvda6a3644527f59bf@mail.gmail.com>
	<33c87fa31002250043l1edbb87dg2e7a58e879656f06@mail.gmail.com>
	<EC776797-0320-4A5B-B959-2D3A3BF77314@freeswitch.org>
	<017001cab63f$a51397c0$ef3ac740$@net>
Message-ID: <87f2f3b91002251158o70b5ab25gbfbb960e31ccad5f@mail.gmail.com>

Gents,
Thanks for all of your input on this. I see a few more entries on the ALG
wiki page and that is exactly what I had hoped to see. Please keep up the
good work.

Thanks,
MC

On Thu, Feb 25, 2010 at 9:26 AM, Adam Ford <lists at redbonez.net> wrote:

> I wish you would have told me that back when I was trying to solve my
> authentication issue that I thought was being caused by using rport. :P
>
> You just told me to upgrade to 1.0.5, when a $%$#! Sonicwall SIP ALG would
> have been more helpful, haha. As it turned out it was the Sonicwall SIP
> transformations, and switching to TCP SIP resolved it.
>
> I have added my Sonicwall experience to the wiki page.
>
> -Adam
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
> West
> Sent: Thursday, February 25, 2010 6:33 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] [Freeswitch-dev] Call for help - adding
> information to the wiki: SIP ALG's
>
> I believe all SIP ALG's are broken.  :P
>
> /b
>
> On Feb 25, 2010, at 2:43 AM, Mark Campbell-Smith wrote:
>
> > I had an issue with a Thomson SpeedTouch 530 router that was causing
> > authentication to fail... its the thread titled 'Forbidden using UDP,
> > works with TCP/TLS' that everyone said was caused by a broken ATA ...
> > lucky no one chipped in so I could buy a new one (as suggested by
> > Anthony) - it wouldn't have helped because the router was broken and
> > modifying the authentication parameters!
> >
> > I'll update the wiki with the info
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100225/8b5be986/attachment-0002.html 

From msc at freeswitch.org  Thu Feb 25 13:10:50 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 25 Feb 2010 13:10:50 -0800
Subject: [Freeswitch-users] freeswitch to gateway
In-Reply-To: <74713287E44D44D8BC02EB42F686076F@MOBILEE1705>
References: <f8af5741002250124s5f3a200fp178d83a90945d73b@mail.gmail.com>
	<20100225170907.GG9832@base.carmickle.com>
	<74713287E44D44D8BC02EB42F686076F@MOBILEE1705>
Message-ID: <87f2f3b91002251310j10ade421m5473944256564afa@mail.gmail.com>

Keep in mind that "1003" is in the default "local extension" range, so if
you want to route 1003 differently than 1000 to 1019 then be sure to put
your specific extension in the dialplan before Local_Extension, otherwise
when you dial "1003" it will go out the Local_Extension and not your custom
extension... :)

-MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100225/e1b07945/attachment-0002.html 

From tculjaga at gmail.com  Thu Feb 25 14:34:55 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Thu, 25 Feb 2010 23:34:55 +0100
Subject: [Freeswitch-users] [Freeswitch-dev] Call for help - adding
	information to the wiki: SIP ALG's
In-Reply-To: <87f2f3b91002251158o70b5ab25gbfbb960e31ccad5f@mail.gmail.com>
References: <87f2f3b91002242008y599f86f0t86f467f8c16b4a20@mail.gmail.com>
	<8b1c9cda1002242248u5e39a4ecvda6a3644527f59bf@mail.gmail.com>
	<33c87fa31002250043l1edbb87dg2e7a58e879656f06@mail.gmail.com>
	<EC776797-0320-4A5B-B959-2D3A3BF77314@freeswitch.org>
	<017001cab63f$a51397c0$ef3ac740$@net>
	<87f2f3b91002251158o70b5ab25gbfbb960e31ccad5f@mail.gmail.com>
Message-ID: <65d96fc81002251434t5987cfcdmb7cf72a45948597e@mail.gmail.com>

well .. thats a know thing.. SIP ALG is to be disabled on every router as
this is not the way NAT traversal is donoe for SIP...

This is something a border element (SBC) on carrier side has to deal with.
If some "intelligent" router thinks it can "fix" the translation it actually
screws it from the border element perspective and you will end up with
either failed call establishment or without audio.


I had a device (zxyel) that was using SIP ALG (hiden command that was
available via telnet only) and everything was fine for a while ... but after
some time my single device behnd NAT was not able to register to the
proviser because of a broken branch string.


so, just disable ALG for SIP on ALL devices you have on customer side .. it
really doesn't have any sense to keep it on.

T.





On Thu, Feb 25, 2010 at 8:58 PM, Michael Collins <msc at freeswitch.org> wrote:

> Gents,
> Thanks for all of your input on this. I see a few more entries on the ALG
> wiki page and that is exactly what I had hoped to see. Please keep up the
> good work.
>
> Thanks,
> MC
>
>
> On Thu, Feb 25, 2010 at 9:26 AM, Adam Ford <lists at redbonez.net> wrote:
>
>> I wish you would have told me that back when I was trying to solve my
>> authentication issue that I thought was being caused by using rport. :P
>>
>> You just told me to upgrade to 1.0.5, when a $%$#! Sonicwall SIP ALG would
>> have been more helpful, haha. As it turned out it was the Sonicwall SIP
>> transformations, and switching to TCP SIP resolved it.
>>
>> I have added my Sonicwall experience to the wiki page.
>>
>> -Adam
>>
>>
>> -----Original Message-----
>> From: freeswitch-users-bounces at lists.freeswitch.org
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
>> West
>> Sent: Thursday, February 25, 2010 6:33 AM
>> To: freeswitch-users at lists.freeswitch.org
>> Subject: Re: [Freeswitch-users] [Freeswitch-dev] Call for help - adding
>> information to the wiki: SIP ALG's
>>
>> I believe all SIP ALG's are broken.  :P
>>
>> /b
>>
>> On Feb 25, 2010, at 2:43 AM, Mark Campbell-Smith wrote:
>>
>> > I had an issue with a Thomson SpeedTouch 530 router that was causing
>> > authentication to fail... its the thread titled 'Forbidden using UDP,
>> > works with TCP/TLS' that everyone said was caused by a broken ATA ...
>> > lucky no one chipped in so I could buy a new one (as suggested by
>> > Anthony) - it wouldn't have helped because the router was broken and
>> > modifying the authentication parameters!
>> >
>> > I'll update the wiki with the info
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100225/9bb90360/attachment-0002.html 

From joseph.puchalski at personalcyberspace.com  Thu Feb 25 15:29:17 2010
From: joseph.puchalski at personalcyberspace.com (Joseph Puchalski)
Date: Thu, 25 Feb 2010 23:29:17 +0000
Subject: [Freeswitch-users] Retrieving voicemail without
	entering	user	ID (extension)
In-Reply-To: <012c01cab5b7$5052d990$f0f88cb0$@net>
References: <093DD565390C1E4FB15D7B383E86BB05AFF6BC@Goose.personalcyberspace.net>
	<b39273531002241601m273ff333y2261d930dbcb3fa7@mail.gmail.com>
	<012c01cab5b7$5052d990$f0f88cb0$@net>
Message-ID: <093DD565390C1E4FB15D7B383E86BB05AFF90C@Goose.personalcyberspace.net>

Adam,

Thanks! I'll give this a try. I'm FreeSWITCH newbie myself, having a fun time figuring out everything in this amazingly rich (and challenging) environment :)

Joe

From: Adam Ford [mailto:lists at redbonez.net]
Sent: Wednesday, February 24, 2010 8:10 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Retrieving voicemail without entering user ID (extension)

>From reading that wiki article it seems to me that the key to achieving the functionality you are looking for would simply be a matter of adding the desired extension to the end of the default action (where the $1 is):

<action application="voicemail" data="check $${voicemail_profile} $${domain} $1"/>

If I am reading it correctly, this should bypass having to enter a mailbox ID, but still require your voicemail password. Off the top of my head, you could probably achieve this by replacing the $1 with a variable storing the extension which called 4000.  I would have to look it up to see if there is a system variable for that or if you would have to assign a custom one. I am still relatively new to FreeSWITCH myself.

-Adam


From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rupa Schomaker
Sent: Wednesday, February 24, 2010 5:01 PM
To: freeswitch-users
Subject: Re: [Freeswitch-users] Retrieving voicemail without entering user ID (extension)

Look at the end of:

http://wiki.freeswitch.org/wiki/Mod_voicemail#Check_Voice_Mail

Advisable?  With it enabled, I can walk up to anyone's phone and retrieve their VM w/out authentication.  It was removed on purpose due to that reason as far as I remember.
On Wed, Feb 24, 2010 at 5:45 PM, Joseph Puchalski <joseph.puchalski at personalcyberspace.com<mailto:joseph.puchalski at personalcyberspace.com>> wrote:
I'm trying to modify my dialplan so that I can press a single button on my phone, be connected to voicemail, and enter only a password to gain access.

Currently I use a programmable key to dial 4000. I am prompted for my ID, and then password.

I've poked around "mod voicemail" on the wiki and searched the mailing list and web, but haven't found enough info. I have discovered that this behavior seems to have been available in previous versions of the default dialplan.

Is it still possible? Is it advisable? Was this feature/behavior removed for security reasons?

I apologize ahead of time if the answer is somewhere in plain sight that I haven't looked yet. If so, I'd much appreciate being pointed in the right direction.

As always, thanks for any help,

Joe P.

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



--
-Rupa
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100225/2568fe08/attachment-0002.html 

From joseph.puchalski at personalcyberspace.com  Thu Feb 25 15:30:14 2010
From: joseph.puchalski at personalcyberspace.com (Joseph Puchalski)
Date: Thu, 25 Feb 2010 23:30:14 +0000
Subject: [Freeswitch-users] Retrieving voicemail without entering
	user	ID (extension)
In-Reply-To: <b39273531002241601m273ff333y2261d930dbcb3fa7@mail.gmail.com>
References: <093DD565390C1E4FB15D7B383E86BB05AFF6BC@Goose.personalcyberspace.net>
	<b39273531002241601m273ff333y2261d930dbcb3fa7@mail.gmail.com>
Message-ID: <093DD565390C1E4FB15D7B383E86BB05AFF919@Goose.personalcyberspace.net>

Thanks for the reply.

I'm not trying to remove the requirement for a password, just the need to enter an extension from a user's "home" phone. The current methods I'm familiar with require that the user enter both a user ID and a password. I'm hoping there's a way that the user ID can be defaulted to be the calling extension. This way I can set up a "voicemail" key on a user's desktop phone and allow them to access vmail by entering just a password.

Thanks again,

Joe

From: Rupa Schomaker [mailto:rupa at rupa.com]
Sent: Wednesday, February 24, 2010 7:01 PM
To: freeswitch-users
Subject: Re: [Freeswitch-users] Retrieving voicemail without entering user ID (extension)

Look at the end of:

http://wiki.freeswitch.org/wiki/Mod_voicemail#Check_Voice_Mail

Advisable?  With it enabled, I can walk up to anyone's phone and retrieve their VM w/out authentication.  It was removed on purpose due to that reason as far as I remember.
On Wed, Feb 24, 2010 at 5:45 PM, Joseph Puchalski <joseph.puchalski at personalcyberspace.com<mailto:joseph.puchalski at personalcyberspace.com>> wrote:
I'm trying to modify my dialplan so that I can press a single button on my phone, be connected to voicemail, and enter only a password to gain access.

Currently I use a programmable key to dial 4000. I am prompted for my ID, and then password.

I've poked around "mod voicemail" on the wiki and searched the mailing list and web, but haven't found enough info. I have discovered that this behavior seems to have been available in previous versions of the default dialplan.

Is it still possible? Is it advisable? Was this feature/behavior removed for security reasons?

I apologize ahead of time if the answer is somewhere in plain sight that I haven't looked yet. If so, I'd much appreciate being pointed in the right direction.

As always, thanks for any help,

Joe P.

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



--
-Rupa
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100225/90563d0e/attachment-0002.html 

From joseph.puchalski at personalcyberspace.com  Thu Feb 25 15:31:18 2010
From: joseph.puchalski at personalcyberspace.com (Joseph Puchalski)
Date: Thu, 25 Feb 2010 23:31:18 +0000
Subject: [Freeswitch-users] Retrieving voicemail without
	entering	user	ID (extension)
In-Reply-To: <000f01cab5c7$d7e292f0$87a7b8d0$@com>
References: <093DD565390C1E4FB15D7B383E86BB05AFF6BC@Goose.personalcyberspace.net>
	<b39273531002241601m273ff333y2261d930dbcb3fa7@mail.gmail.com>
	<000f01cab5c7$d7e292f0$87a7b8d0$@com>
Message-ID: <093DD565390C1E4FB15D7B383E86BB05AFF928@Goose.personalcyberspace.net>

Lars,

Thanks!

Joe

From: Lars Zeb [mailto:larclap at yahoo.com]
Sent: Wednesday, February 24, 2010 10:09 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Retrieving voicemail without entering user ID (extension)

Joe,

I used the extension below, but I think that Brian said it was too insecure. Being a total beginner, I removed the condition.

    <extension name="Local_Extension_Abe">
      <condition field="destination_number" expression="^(101[0-9])$">
        <action application="set" data="dialed_ext=$1"/>
        <action application="export" data="dialed_ext=$1"/>
      </condition>
      <condition field="destination_number" expression="^${caller_id_number}$">
        <action application="set" data="voicemail_authorized=${sip_authorized}"/>
        <action application="answer"/>
        <action application="sleep" data="1000"/>
        <action application="voicemail" data="check default $${domain} ${dialed_ext}"/>
        <anti-action application="bind_meta_app" data="1 a a execute_extension::dx XML features"/>
        <anti-action application="bind_meta_app" data="2 a a record_session::$${base_dir}/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
        <anti-action application="bind_meta_app" data="3 a a execute_extension::cf XML features"/>
        <anti-action application="set" data="transfer_ringback=${us-ring}"/>
        <anti-action application="set" data="call_timeout=15"/>
        <anti-action application="set" data="sip_exclude_contact=${network_addr}"/>
        <anti-action application="set" data="hangup_after_bridge=true"/>
        <!--<anti-action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,BUSY,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION"/> -->
        <anti-action application="set" data="continue_on_fail=true"/>
        <anti-action application="db" data="insert/call_return/${dialed_ext}/${caller_id_number}"/>
        <anti-action application="db" data="insert/last_dial_ext/${dialed_ext}/${uuid}"/>
        <anti-action application="bridge" data="user/${dialed_ext}@$${domain}"/>
        <anti-action application="answer"/>
        <anti-action application="sleep" data="1000"/>
        <anti-action application="voicemail" data="default $${domain} ${dialed_ext}"/>
      </condition>
    </extension>

Lars
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rupa Schomaker
Sent: Wednesday, February 24, 2010 4:01 PM
To: freeswitch-users
Subject: Re: [Freeswitch-users] Retrieving voicemail without entering user ID (extension)

Look at the end of:

http://wiki.freeswitch.org/wiki/Mod_voicemail#Check_Voice_Mail

Advisable?  With it enabled, I can walk up to anyone's phone and retrieve their VM w/out authentication.  It was removed on purpose due to that reason as far as I remember.
On Wed, Feb 24, 2010 at 5:45 PM, Joseph Puchalski <joseph.puchalski at personalcyberspace.com<mailto:joseph.puchalski at personalcyberspace.com>> wrote:
I'm trying to modify my dialplan so that I can press a single button on my phone, be connected to voicemail, and enter only a password to gain access.

Currently I use a programmable key to dial 4000. I am prompted for my ID, and then password.

I've poked around "mod voicemail" on the wiki and searched the mailing list and web, but haven't found enough info. I have discovered that this behavior seems to have been available in previous versions of the default dialplan.

Is it still possible? Is it advisable? Was this feature/behavior removed for security reasons?

I apologize ahead of time if the answer is somewhere in plain sight that I haven't looked yet. If so, I'd much appreciate being pointed in the right direction.

As always, thanks for any help,

Joe P.

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



--
-Rupa
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100225/2bfa1243/attachment-0002.html 

From robert.hadley at teotech.com  Thu Feb 25 16:13:10 2010
From: robert.hadley at teotech.com (Robert Hadley)
Date: Thu, 25 Feb 2010 16:13:10 -0800
Subject: [Freeswitch-users] Freeswitch/Openzap dials out on disconnected FXO
	line
Message-ID: <4DF42CB92831454193CEC0E375E06725@greyhawk.tonecommander.com>

When dialing out, Freeswitch/Openzap is not detecting that an analog FXO
channel is disconnected and tries dialing out on the channel anyway.  No
error is reported.  The call doesn't timeout until a minute later.
Shouldn't Freeswitch/Openzap skip over a disconnected channel to the next
connected channel?

 

I have configured a Sangoma A200 FXO card as a FXO span.

 

[span wanpipe FXO]

name => PSTN Line 1

number => 4253491059

fxo-channel => 2:3

name => PSTN Line 2

number => 4253491058

fxo-channel => 2:4

 

 

The wanpipe driver does detect and report when a CO line is connected or
disconnected (in /var/log/messages), and Freeswitch/Openzap gets an event as
reported in the log.

 

/var/log/messages: Feb 25 15:23:10 roberth-c53 kernel: wanpipe2: Module 3:
FXO Line is disconnected!

FS_CLI: 2010-02-25 15:23:10.711604 [DEBUG] ozmod_analog.c:788 EVENT
[ALARM_TRAP][3:1] STATE [DOWN]

 

/var/log/messages: Feb 25 15:23:44 roberth-c53 kernel: wanpipe2: Module 4:
FXO Line is connected!

FS_CLI: 2010-02-25 15:23:44.901979 [DEBUG] ozmod_analog.c:788 EVENT
[ALARM_CLEAR][3:2] STATE [DOWN]

 

 

I have the dialplan configured to use the next available port in the FXO
span (there will be more than 2 channels later).

 

<extension name="OutgoingFXO">

  <condition field="destination_number" expression="^7(\d+)$">

    <action application="bridge" data="openzap/FXO/a/$1"/>

  </condition>

</extension>

 

 

Here is a portion of the log when that shows dialing out on a disconnected
analog FXO channel.

 

EXECUTE sofia/internal/5410 at 192.168.72.45:5060
bridge(openzap/FXO/a/93491045)

2010-02-25 15:26:17.891443 [DEBUG] mod_openzap.c:366 Set codec PCMU 20ms

2010-02-25 15:26:17.891443 [DEBUG] mod_openzap.c:1257 Connect outbound
channel OpenZAP/3:1/93491045

2010-02-25 15:26:17.891443 [NOTICE] switch_channel.c:642 New Channel
OpenZAP/3:1/93491045 [3c8f46f5-77a8-498f-a51c-015837746cb7]

2010-02-25 15:26:17.891443 [DEBUG] mod_openzap.c:1269 (OpenZAP/3:1/93491045)
State Change CS_NEW -> CS_INIT

2010-02-25 15:26:17.891443 [DEBUG] switch_core_session.c:1019 Send signal
OpenZAP/3:1/93491045 [BREAK]

2010-02-25 15:26:17.891443 [DEBUG] ozmod_analog.c:59 Changing state on 3:1
from DOWN to DIALING

2010-02-25 15:26:17.891443 [WARNING] switch_core_session.c:486
OpenZAP/3:1/93491045 does not support the proxy feature, disabling.

2010-02-25 15:26:17.891443 [DEBUG] ozmod_analog.c:279 ANALOG CHANNEL thread
starting.

2010-02-25 15:26:17.891443 [DEBUG] ozmod_analog.c:450 Executing state
handler on 3:1 for DIALING

2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:314
(OpenZAP/3:1/93491045) Running State Change CS_INIT

2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:338
(OpenZAP/3:1/93491045) State INIT

2010-02-25 15:26:17.891443 [DEBUG] mod_openzap.c:394 (OpenZAP/3:1/93491045)
State Change CS_INIT -> CS_ROUTING

2010-02-25 15:26:17.891443 [DEBUG] switch_core_session.c:1019 Send signal
OpenZAP/3:1/93491045 [BREAK]

2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:338
(OpenZAP/3:1/93491045) State INIT going to sleep

2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:314
(OpenZAP/3:1/93491045) Running State Change CS_ROUTING

2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:341
(OpenZAP/3:1/93491045) State ROUTING

2010-02-25 15:26:17.891443 [DEBUG] mod_openzap.c:417 OpenZAP/3:1/93491045
CHANNEL ROUTING

2010-02-25 15:26:17.891443 [DEBUG] switch_ivr_originate.c:66
(OpenZAP/3:1/93491045) State Change CS_ROUTING -> CS_CONSUME_MEDIA

2010-02-25 15:26:17.891443 [DEBUG] switch_core_session.c:1019 Send signal
OpenZAP/3:1/93491045 [BREAK]

2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:341
(OpenZAP/3:1/93491045) State ROUTING going to sleep

2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:314
(OpenZAP/3:1/93491045) Running State Change CS_CONSUME_MEDIA

2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:360
(OpenZAP/3:1/93491045) State CONSUME_MEDIA

2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:360
(OpenZAP/3:1/93491045) State CONSUME_MEDIA going to sleep

 

 

Thanks,

Robert

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100225/b238b94e/attachment-0002.html 

From robert.hadley at teotech.com  Thu Feb 25 17:16:07 2010
From: robert.hadley at teotech.com (Robert Hadley)
Date: Thu, 25 Feb 2010 17:16:07 -0800
Subject: [Freeswitch-users] Cannot make calls through PRI via wanpipe
	running as daemon
Message-ID: <6FCED74CF54F466AA25CBDA7DFF42198@greyhawk.tonecommander.com>

When running Freeswitch as service called teoswitch as user teoswitch I
cannot make calls through the Sangoma PRI or analog cards using wanpipe
driver.  I have added a file called 30-wanpipe.rules to /etc/udev/rules.d
and rebooted the server.

 

cat 30-wanpipe.rules

# /etc/udev/rules.d/30-wanpipe.rules

SUBSYSTEM=="wptdm", OWNER="teoswitch", GROUP="teoswitch", MODE="0660"

SUBSYSTEM=="wanpipe", OWNER="teoswitch", GROUP="teoswitch", MODE="0660"

 

 

Freeswitch log:

Dialplan: sofia/internal/5410 at 192.168.72.45:5060 parsing
[default->SangomaPRI] continue=false

Dialplan: sofia/internal/5410 at 192.168.72.45:5060 Regex (PASS) [SangomaPRI]
destination_number(93491045) =~ /^9(\d+)$/ break=on-false

Dialplan: sofia/internal/5410 at 192.168.72.45:5060 Action
set(effective_caller_id_number=425740${caller_id_number})

Dialplan: sofia/internal/5410 at 192.168.72.45:5060 Action
bridge(openzap/smg_prid/a/3491045 at g1)

2010-02-25 16:51:11.328635 [DEBUG] switch_core_state_machine.c:122
(sofia/internal/5410 at 192.168.72.45:5060) State Change CS_ROUTING ->
CS_EXECUTE

2010-02-25 16:51:11.328635 [DEBUG] switch_core_session.c:1019 Send signal
sofia/internal/5410 at 192.168.72.45:5060 [BREAK]

2010-02-25 16:51:11.328635 [DEBUG] switch_core_state_machine.c:341
(sofia/internal/5410 at 192.168.72.45:5060) State ROUTING going to sleep

2010-02-25 16:51:11.328635 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/5410 at 192.168.72.45:5060) Running State Change CS_EXECUTE

2010-02-25 16:51:11.328635 [DEBUG] switch_core_state_machine.c:348
(sofia/internal/5410 at 192.168.72.45:5060) State EXECUTE

2010-02-25 16:51:11.328635 [DEBUG] mod_sofia.c:181
sofia/internal/5410 at 192.168.72.45:5060 SOFIA EXECUTE

2010-02-25 16:51:11.328635 [DEBUG] switch_core_state_machine.c:159
sofia/internal/5410 at 192.168.72.45:5060 Standard EXECUTE

EXECUTE sofia/internal/5410 at 192.168.72.45:5060
set(effective_caller_id_number=4257405410)

2010-02-25 16:51:11.328635 [DEBUG] mod_dptools.c:811
sofia/internal/5410 at 192.168.72.45:5060 SET
[effective_caller_id_number]=[4257405410]

EXECUTE sofia/internal/5410 at 192.168.72.45:5060
bridge(openzap/smg_prid/a/3491045 at g1)

2010-02-25 16:51:11.339637 [ERR] switch_core_session.c:357 Could not locate
channel type openzap

2010-02-25 16:51:11.339637 [ERR] switch_ivr_originate.c:2411 Cannot create
outgoing channel of type [openzap] cause: [CHAN_NOT_IMPLEMENTED]

2010-02-25 16:51:11.339637 [DEBUG] switch_ivr_originate.c:3209 Originate
Resulted in Error Cause: 66 [CHAN_NOT_IMPLEMENTED]

2010-02-25 16:51:11.339637 [INFO] mod_dptools.c:2346 Originate Failed.
Cause: CHAN_NOT_IMPLEMENTED

2010-02-25 16:51:11.339637 [NOTICE] mod_dptools.c:2409 Hangup
sofia/internal/5410 at 192.168.72.45:5060 [CS_EXECUTE] [CHAN_NOT_IMPLEMENTED]

2010-02-25 16:51:11.339637 [DEBUG] switch_channel.c:1976 Send signal
sofia/internal/5410 at 192.168.72.45:5060 [KILL]

2010-02-25 16:51:11.339637 [DEBUG] switch_core_session.c:1019 Send signal
sofia/internal/5410 at 192.168.72.45:5060 [BREAK]

 

 

Thanks,

Robert

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100225/bd95f30b/attachment-0002.html 

From brian at freeswitch.org  Thu Feb 25 17:22:51 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 25 Feb 2010 19:22:51 -0600
Subject: [Freeswitch-users] ASR Apps
Message-ID: <AC3A71D6-4EAB-4261-B14A-2869C0C05C91@freeswitch.org>

I'm looking for some enterprising community members to create some interesting voice apps using ASR.  Please email me off list and we'll get you what you need to do this.

Thanks,
Brian




From edpimentl at gmail.com  Thu Feb 25 18:55:12 2010
From: edpimentl at gmail.com (EdPimentl)
Date: Thu, 25 Feb 2010 21:55:12 -0500
Subject: [Freeswitch-users] ASR Apps
In-Reply-To: <AC3A71D6-4EAB-4261-B14A-2869C0C05C91@freeswitch.org>
References: <AC3A71D6-4EAB-4261-B14A-2869C0C05C91@freeswitch.org>
Message-ID: <9dc4a1671002251855y586218f6h2c281b4a5acb664e@mail.gmail.com>

Hello Bryon,

We looking to create a Twilio like service using FreeSwitch.

Sincerely,
-E
http://vCardCloud.com
GV:  678.685.9858
EdPimentl: Skype



On Thu, Feb 25, 2010 at 8:22 PM, Brian West <brian at freeswitch.org> wrote:

> I'm looking for some enterprising community members to create some
> interesting voice apps using ASR.  Please email me off list and we'll get
> you what you need to do this.
>
> Thanks,
> Brian
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100225/c7098d9b/attachment-0002.html 

From brian at freeswitch.org  Thu Feb 25 19:02:14 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 25 Feb 2010 21:02:14 -0600
Subject: [Freeswitch-users] ASR Apps
In-Reply-To: <9dc4a1671002251855y586218f6h2c281b4a5acb664e@mail.gmail.com>
References: <AC3A71D6-4EAB-4261-B14A-2869C0C05C91@freeswitch.org>
	<9dc4a1671002251855y586218f6h2c281b4a5acb664e@mail.gmail.com>
Message-ID: <16DBCD58-A962-4121-9899-F2BB56F13554@freeswitch.org>

I'm looking for someone to build some really nice apps like dial by name speech apps or other such apps or frameworks using ASR and possibly lua or js.  Anyone wanna do something.

/b

On Feb 25, 2010, at 8:55 PM, EdPimentl wrote:

> Hello Bryon,
> 
> We looking to create a Twilio like service using FreeSwitch.
> 
> Sincerely,
> -E
> http://vCardCloud.com
> GV:  678.685.9858
> EdPimentl: Skype

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100225/52881261/attachment-0002.html 

From srinivas.ksvreddy at gmail.com  Thu Feb 25 20:42:46 2010
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Fri, 26 Feb 2010 10:12:46 +0530
Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 44, Issue 234
In-Reply-To: <mailman.6938.1267143241.2495.freeswitch-users@lists.freeswitch.org>
References: <mailman.6938.1267143241.2495.freeswitch-users@lists.freeswitch.org>
Message-ID: <f8af5741002252042r51671ec9ia0ca7a2fbcd44e62@mail.gmail.com>

Hi,

thank you very munch for reply,
this is working fine, when we configure <condition
destination_number="^1003"
<action application="bridge" data="sofia/gateway/gatewayname/$1"/>

but in my scenario, i dont want to route call based on extensions( eg, here
1003) routing, i just want to route the calls when the destination domain is
defferan from local domain,

example: INVITE packet from registered extension to sipserver like this.

   From: 1000 at gw.proxy.com:5060
   To :  1003 at gateway.com:5060

here from uri and to uri is different. any help


Srinivas

On Fri, Feb 26, 2010 at 5:44 AM, <
freeswitch-users-request at lists.freeswitch.org> wrote:

> Send FreeSWITCH-users mailing list submissions to
>        freeswitch-users at lists.freeswitch.org
>
> To subscribe or unsubscribe via the World Wide Web, visit
>        http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> or, via email, send a message with subject or body 'help' to
>        freeswitch-users-request at lists.freeswitch.org
>
> You can reach the person managing the list at
>        freeswitch-users-owner at lists.freeswitch.org
>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of FreeSWITCH-users digest..."
>
> Today's Topics:
>
>   1. Re: Retrieving voicemail without  entering        user    ID
> (extension)
>      (Joseph Puchalski)
>   2. Freeswitch/Openzap dials out on disconnected FXO  line
>      (Robert Hadley)
>
>
> ---------- Forwarded message ----------
> From: Joseph Puchalski <joseph.puchalski at personalcyberspace.com>
> To: "freeswitch-users at lists.freeswitch.org" <
> freeswitch-users at lists.freeswitch.org>
> Date: Thu, 25 Feb 2010 23:31:18 +0000
> Subject: Re: [Freeswitch-users] Retrieving voicemail without entering user
> ID (extension)
>
> Lars,
>
>
>
> Thanks!
>
>
>
> Joe
>
>
>
> *From:* Lars Zeb [mailto:larclap at yahoo.com]
> *Sent:* Wednesday, February 24, 2010 10:09 PM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Retrieving voicemail without entering
> user ID (extension)
>
>
>
> Joe,
>
>
>
> I used the extension below, but I think that Brian said it was too
> insecure. Being a total beginner, I removed the condition.
>
>
>
>     <extension name="Local_Extension_Abe">
>
>       <condition field="destination_number" expression="^(101[0-9])$">
>
>         <action application="set" data="dialed_ext=$1"/>
>
>         <action application="export" data="dialed_ext=$1"/>
>
>       </condition>
>
>       <condition field="destination_number"
> expression="^${caller_id_number}$">
>
>         <action application="set"
> data="voicemail_authorized=${sip_authorized}"/>
>
>         <action application="answer"/>
>
>         <action application="sleep" data="1000"/>
>
>         <action application="voicemail" data="check default $${domain}
> ${dialed_ext}"/>
>
>         <anti-action application="bind_meta_app" data="1 a a
> execute_extension::dx XML features"/>
>
>         <anti-action application="bind_meta_app" data="2 a a
> record_session::$${base_dir}/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
>
>         <anti-action application="bind_meta_app" data="3 a a
> execute_extension::cf XML features"/>
>
>         <anti-action application="set"
> data="transfer_ringback=${us-ring}"/>
>
>         <anti-action application="set" data="call_timeout=15"/>
>
>         <anti-action application="set"
> data="sip_exclude_contact=${network_addr}"/>
>
>         <anti-action application="set" data="hangup_after_bridge=true"/>
>
>         <!--<anti-action application="set"
> data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,BUSY,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION"/>
> -->
>
>         <anti-action application="set" data="continue_on_fail=true"/>
>
>         <anti-action application="db"
> data="insert/call_return/${dialed_ext}/${caller_id_number}"/>
>
>         <anti-action application="db"
> data="insert/last_dial_ext/${dialed_ext}/${uuid}"/>
>
>         <anti-action application="bridge" data="user/${dialed_ext}@
> $${domain}"/>
>
>         <anti-action application="answer"/>
>
>         <anti-action application="sleep" data="1000"/>
>
>         <anti-action application="voicemail" data="default $${domain}
> ${dialed_ext}"/>
>
>       </condition>
>
>     </extension>
>
>
>
> Lars
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Rupa
> Schomaker
> *Sent:* Wednesday, February 24, 2010 4:01 PM
> *To:* freeswitch-users
> *Subject:* Re: [Freeswitch-users] Retrieving voicemail without entering
> user ID (extension)
>
>
>
> Look at the end of:
>
>
>
> http://wiki.freeswitch.org/wiki/Mod_voicemail#Check_Voice_Mail
>
>
>
> Advisable?  With it enabled, I can walk up to anyone's phone and retrieve
> their VM w/out authentication.  It was removed on purpose due to that reason
> as far as I remember.
>
> On Wed, Feb 24, 2010 at 5:45 PM, Joseph Puchalski <
> joseph.puchalski at personalcyberspace.com> wrote:
>
> I?m trying to modify my dialplan so that I can press a single button on my
> phone, be connected to voicemail, and enter only a password to gain access.
>
>
>
> Currently I use a programmable key to dial 4000. I am prompted for my ID,
> and then password.
>
>
>
> I?ve poked around ?mod voicemail? on the wiki and searched the mailing list
> and web, but haven?t found enough info. I have discovered that this behavior
> seems to have been available in previous versions of the default dialplan.
>
>
>
> Is it still possible? Is it advisable? Was this feature/behavior removed
> for security reasons?
>
>
>
> I apologize ahead of time if the answer is somewhere in plain sight that I
> haven?t looked yet. If so, I?d much appreciate being pointed in the right
> direction.
>
>
>
> As always, thanks for any help,
>
>
>
> Joe P.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> --
> -Rupa
>
>
> ---------- Forwarded message ----------
> From: "Robert Hadley" <robert.hadley at teotech.com>
> To: <freeswitch-users at lists.freeswitch.org>
> Date: Thu, 25 Feb 2010 16:13:10 -0800
> Subject: [Freeswitch-users] Freeswitch/Openzap dials out on disconnected
> FXO line
>
> When dialing out, Freeswitch/Openzap is not detecting that an analog FXO
> channel is disconnected and tries dialing out on the channel anyway.  No
> error is reported.  The call doesn?t timeout until a minute later.
> Shouldn?t Freeswitch/Openzap skip over a disconnected channel to the next
> connected channel?
>
>
>
> I have configured a Sangoma A200 FXO card as a FXO span.
>
>
>
> [span wanpipe FXO]
>
> name => PSTN Line 1
>
> number => 4253491059
>
> fxo-channel => 2:3
>
> name => PSTN Line 2
>
> number => 4253491058
>
> fxo-channel => 2:4
>
>
>
>
>
> The wanpipe driver does detect and report when a CO line is connected or
> disconnected (in /var/log/messages), and Freeswitch/Openzap gets an event as
> reported in the log.
>
>
>
> /var/log/messages: Feb 25 15:23:10 roberth-c53 kernel: wanpipe2: Module 3:
> FXO Line is disconnected!
>
> FS_CLI: 2010-02-25 15:23:10.711604 [DEBUG] ozmod_analog.c:788 EVENT
> [ALARM_TRAP][3:1] STATE [DOWN]
>
>
>
> /var/log/messages: Feb 25 15:23:44 roberth-c53 kernel: wanpipe2: Module 4:
> FXO Line is connected!
>
> FS_CLI: 2010-02-25 15:23:44.901979 [DEBUG] ozmod_analog.c:788 EVENT
> [ALARM_CLEAR][3:2] STATE [DOWN]
>
>
>
>
>
> I have the dialplan configured to use the next available port in the FXO
> span (there will be more than 2 channels later).
>
>
>
> <extension name="OutgoingFXO">
>
>   <condition field="destination_number" expression="^7(\d+)$">
>
>     <action application="bridge" data="openzap/FXO/a/$1"/>
>
>   </condition>
>
> </extension>
>
>
>
>
>
> Here is a portion of the log when that shows dialing out on a disconnected
> analog FXO channel.
>
>
>
> EXECUTE sofia/internal/5410 at 192.168.72.45:5060bridge(openzap/FXO/a/93491045)
>
> 2010-02-25 15:26:17.891443 [DEBUG] mod_openzap.c:366 Set codec PCMU 20ms
>
> 2010-02-25 15:26:17.891443 [DEBUG] mod_openzap.c:1257 Connect outbound
> channel OpenZAP/3:1/93491045
>
> 2010-02-25 15:26:17.891443 [NOTICE] switch_channel.c:642 New Channel
> OpenZAP/3:1/93491045 [3c8f46f5-77a8-498f-a51c-015837746cb7]
>
> 2010-02-25 15:26:17.891443 [DEBUG] mod_openzap.c:1269
> (OpenZAP/3:1/93491045) State Change CS_NEW -> CS_INIT
>
> 2010-02-25 15:26:17.891443 [DEBUG] switch_core_session.c:1019 Send signal
> OpenZAP/3:1/93491045 [BREAK]
>
> 2010-02-25 15:26:17.891443 [DEBUG] ozmod_analog.c:59 Changing state on 3:1
> from DOWN to DIALING
>
> 2010-02-25 15:26:17.891443 [WARNING] switch_core_session.c:486
> OpenZAP/3:1/93491045 does not support the proxy feature, disabling.
>
> 2010-02-25 15:26:17.891443 [DEBUG] ozmod_analog.c:279 ANALOG CHANNEL thread
> starting.
>
> 2010-02-25 15:26:17.891443 [DEBUG] ozmod_analog.c:450 Executing state
> handler on 3:1 for DIALING
>
> 2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:314
> (OpenZAP/3:1/93491045) Running State Change CS_INIT
>
> 2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:338
> (OpenZAP/3:1/93491045) State INIT
>
> 2010-02-25 15:26:17.891443 [DEBUG] mod_openzap.c:394 (OpenZAP/3:1/93491045)
> State Change CS_INIT -> CS_ROUTING
>
> 2010-02-25 15:26:17.891443 [DEBUG] switch_core_session.c:1019 Send signal
> OpenZAP/3:1/93491045 [BREAK]
>
> 2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:338
> (OpenZAP/3:1/93491045) State INIT going to sleep
>
> 2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:314
> (OpenZAP/3:1/93491045) Running State Change CS_ROUTING
>
> 2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:341
> (OpenZAP/3:1/93491045) State ROUTING
>
> 2010-02-25 15:26:17.891443 [DEBUG] mod_openzap.c:417 OpenZAP/3:1/93491045
> CHANNEL ROUTING
>
> 2010-02-25 15:26:17.891443 [DEBUG] switch_ivr_originate.c:66
> (OpenZAP/3:1/93491045) State Change CS_ROUTING -> CS_CONSUME_MEDIA
>
> 2010-02-25 15:26:17.891443 [DEBUG] switch_core_session.c:1019 Send signal
> OpenZAP/3:1/93491045 [BREAK]
>
> 2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:341
> (OpenZAP/3:1/93491045) State ROUTING going to sleep
>
> 2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:314
> (OpenZAP/3:1/93491045) Running State Change CS_CONSUME_MEDIA
>
> 2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:360
> (OpenZAP/3:1/93491045) State CONSUME_MEDIA
>
> 2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:360
> (OpenZAP/3:1/93491045) State CONSUME_MEDIA going to sleep
>
>
>
>
>
> Thanks,
>
> Robert
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Srinivasula Reddy K
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100226/637c285b/attachment-0002.html 

From srinivas.ksvreddy at gmail.com  Thu Feb 25 20:51:21 2010
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Fri, 26 Feb 2010 10:21:21 +0530
Subject: [Freeswitch-users] freeswitch to gateway
Message-ID: <f8af5741002252051id2e247al1138ea08f8a507ec@mail.gmail.com>

Hi,

thank you very munch for reply,
this is working fine, when we configure <condition
destination_number="^1003"
<action application="bridge" data="sofia/gateway/gatewayname/$1"/>

but in my scenario, i dont want to route call based on extensions( eg, here
1003) routing, i just want to route the calls when the destination host uri
is defferan from local domain,

example: INVITE packet from registered extension to sipserver like this.

   From: 1000 at gw.proxy.com:5060
   To :  1003 at gateway.com:5060

here from uri and to uri is different. any help


Srinivas
-- 
Srinivasula Reddy K
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100226/b701f674/attachment-0002.html 

From nagalenoj at gmail.com  Thu Feb 25 21:33:28 2010
From: nagalenoj at gmail.com (Nagalenoj H.)
Date: Fri, 26 Feb 2010 11:03:28 +0530
Subject: [Freeswitch-users] Error loading
	/usr/local/freeswitch/mod/ozmod_sangoma_boost.so
In-Reply-To: <D62532C0-53A1-4327-9BAA-DB2FB0C4BEF8@jerris.com>
References: <fe67a3341002230202y1d45a8f3t3a2cda07de2240a2@mail.gmail.com>
	<87f2f3b91002231357t6c3e40cdld6a0a9861a9ec5a5@mail.gmail.com>
	<fe67a3341002232043w23e5c04bh2598bdd9641d9c23@mail.gmail.com>
	<D62532C0-53A1-4327-9BAA-DB2FB0C4BEF8@jerris.com>
Message-ID: <fe67a3341002252133k6e2793a7t102a580ad1c879f5@mail.gmail.com>

I was missing lksctp-tools and libsctp-dev packages.
After installing these two, I started installing from first. The issue got
solved.!

On Wed, Feb 24, 2010 at 1:53 PM, Michael Jerris <mike at jerris.com> wrote:

> you missed the second 1/2 of step 3 of *Wan**pipe TDM Installation*
> *
> *
> *
> *
> On Feb 23, 2010, at 11:43 PM, Nagalenoj H. wrote:
>
> ozmod_sangoma_boost.so doesn't exist anywhere. It may not be a old version,
> since I've checked out the source yesterday.
>
> I've a doubt in the installation steps given. It is given to edit the
> modules.conf after executing ./configure. Is it right? Do I need to edit the
> modules.conf before ./configure??
>
> On Wed, Feb 24, 2010 at 3:27 AM, Michael Collins <msc at freeswitch.org>wrote:
>
>> Does the file ozmod_sangoma_boost.so exist somewhere in your openzap build
>> tree? If not then something went wrong while compiling or you have an old
>> version. If it does exist, do a quick test: cp the file into
>> /usr/local/freeswitch/mod and restart FreeSWITCH and see if it finds the
>> file and loads OpenZAP properly. Let us know the results so we can determine
>> if it's a bug in the build system or not.
>>
>> -MC
>>
>> On Tue, Feb 23, 2010 at 2:02 AM, Nagalenoj H. <nagalenoj at gmail.com>wrote:
>>
>>> Dear friends,
>>>     I've installed freeswitch trunk - 16729 and tried to configure with
>>> wanpipe for sangoma A102 pri card.
>>>
>>> Followed the steps given in
>>> http://wiki.sangoma.com/wanpipe-freeswitch-install
>>>
>>> When loading the freeswitch, I've got the following error.
>>>
>>> 2010-02-23 14:49:58.545726 [INFO] ozmod_wanpipe.c:335 configuring device
>>> s2c15 as OpenZAP device 1:30 fd:57 DTMF: software
>>> 2010-02-23 14:49:58.546004 [ERR] ozmod_wanpipe.c:436 No valid wanpipe
>>> span and channel was specified
>>> 2010-02-23 14:49:58.546275 [INFO] zap_io.c:2500 Configured 30 channel(s)
>>> 2010-02-23 14:49:58.560724 [ERR] zap_io.c:2560 Error loading
>>> /usr/local/freeswitch/mod/ozmod_sangoma_boost.so
>>> [/usr/local/freeswitch/mod/ozmod_sangoma_boost.so: cannot open shared object
>>> file: No such file or directory]
>>> 2010-02-23 14:49:58.561048 [ERR] zap_io.c:2720 can't find 'sangoma_boost'
>>> 2010-02-23 14:49:58.561266 [ERR] mod_openzap.c:2458 Error starting
>>> OpenZAP span 1 error:
>>> 2010-02-23 14:49:58.561712 [NOTICE] switch_loadable_module.c:144 Adding
>>> Endpoint 'openzap'
>>>
>>> Configuration and log files are pasted to pastebin. Kindly someone help
>>> me to solve this issue.
>>>
>>> openzap.conf and openzap.conf.xml
>>>    http://pastebin.freeswitch.org/12214
>>>
>>> freeswitch log
>>>    http://pastebin.freeswitch.org/12216
>>>
>>> smg_pri.conf
>>>    http://pastebin.freeswitch.org/12217
>>>
>>> --
>>> Regards,
>>> Nagalenoj H.
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Regards,
> Nagalenoj H.
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Regards,
Nagalenoj H.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100226/dec9909b/attachment-0002.html 

From lakindia89 at gmail.com  Thu Feb 25 21:55:29 2010
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Fri, 26 Feb 2010 11:25:29 +0530
Subject: [Freeswitch-users] smg_prid not bridging the call
Message-ID: <7d79b3931002252155t47c86968q9e451482926d93ac@mail.gmail.com>

Dear all,
I'm having a A102 Sangoma hardware. I configured it with freeswitch.
wanrouter status, says both the port as connected.
My smg_prid version is

Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: ================System
restart=============
Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: = Sangoma PRI Protocol Stack
Daemon       =
Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: = Version:
1.54                           =
Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: = Date: Feb 15
2010                       =
Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: = Wanpipe Release:
wanpipe-3.5.8.6       =
Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: = Revision:Revision:
15288                =
Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid:
===========================================

My freeswitch version is 16729.
I started freeswitch.

oz list
+OK
span: 1 (smg_prid)
type: Sangoma (boost)
chan_count: 60
dialplan: XML
context: default
dial_regex:
fail_dial_regex:
hold_music:
analog_options none

I originated a call as
originate openzap/smg_prid/a/9952248266 &park(), which hits my mobile.

But when I issued the following command:
originate openzap/smg_prid/a/9952248266
&bridge(openzap/smg_prid/a/8122133885)
It rings my mobile (9952248266) first, but after that the following error
was displayed

2010-02-26 16:20:51.736080 [ERR] switch_ivr_originate.c:2387 Cannot create
outgoing channel of type [openzap] cause: [NORMAL_CIRCUIT_CONGESTION]
The call got ended in my mobile.

Freeswitch log and smg_pri.conf
          http://pastebin.freeswitch.org/12248
openzap.conf:
[span wanpipe smg_prid]
name => smg_prid
trunk_type =>e1
b-channel => 1:1-15
b-channel => 1:17-31
trunk_type =>e1
b-channel => 2:1-15
b-channel => 2:17-31

openzap.conf.xml:
<configuration name="openzap.conf" description="OpenZAP Configuration">
  <settings>
    <param name="debug" value="0"/>
  </settings>
  <boost_spans>
        <span id="smg_prid">
        <param name="dialplan" value="XML"/>
        <param name="context" value="default"/>
        </span>
  </boost_spans>
</configuration>

Please guide me to setup this one!!.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100226/c62fd7ab/attachment-0002.html 

From david.varnes at gmail.com  Fri Feb 26 02:18:34 2010
From: david.varnes at gmail.com (david varnes)
Date: Fri, 26 Feb 2010 21:18:34 +1100
Subject: [Freeswitch-users] ASR Apps
In-Reply-To: <AC3A71D6-4EAB-4261-B14A-2869C0C05C91@freeswitch.org>
References: <AC3A71D6-4EAB-4261-B14A-2869C0C05C91@freeswitch.org>
Message-ID: <74a861001002260218i38bfbf72s5637ed20f684a40b@mail.gmail.com>

Hi Brian,

I have just started porting an ASR based framework from
a VXML engine to use FS.

It is java based, which I know is not a big focus for the
project ...

Do you have some ASR ports we could use for testing ?
I am very interested ..

davidv

On 26 February 2010 12:22, Brian West <brian at freeswitch.org> wrote:
> I'm looking for some enterprising community members to create some interesting voice apps using ASR. ?Please email me off list and we'll get you what you need to do this.
>
> Thanks,
> Brian
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
david varnes

e: david.varnes at gmail.com
p: +61 404 925 633



From infos at madovsky.org  Fri Feb 26 02:39:37 2010
From: infos at madovsky.org (Madovsky)
Date: Fri, 26 Feb 2010 05:39:37 -0500
Subject: [Freeswitch-users] about pizza demo
Message-ID: <1871231B5C344CB08333565ED2EC2260@MOBILEE1705>

Hi,

I'm trying to change the language of pizza demo script.
is it need to change only words inside addItemAlias() ?

Many thanks

Franck
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100226/b776d532/attachment-0002.html 

From steveayre at gmail.com  Fri Feb 26 04:52:36 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 26 Feb 2010 12:52:36 +0000
Subject: [Freeswitch-users] SIP Trunk with Private Static IP?
In-Reply-To: <009b01cab62b$b8889c10$2999d430$@co.za>
References: <008401cab5f5$4795b910$d6c12b30$@co.za>
	<eafa772f1002250057l3de5b2abu54da8725cbb77281@mail.gmail.com>
	<eafa772f1002250058i2cd7b30ai42f44d0ea1d6eb88@mail.gmail.com>
	<009b01cab62b$b8889c10$2999d430$@co.za>
Message-ID: <eafa772f1002260452h14dd6baavd2a5ac5fa012abd5@mail.gmail.com>

A SIP 'ping' is not a ICMP ping...

It works by sending a OPTIONS SIP request to the gateway, which then
responds with 200 OK. It has the advantage of working even if ICMP is
filtered by a firewall and testing whether the SIP server software is
running, not just whether the server is online.

Best Regards,
-Steve


On 25 February 2010 15:02, Roly Maz <rm at callrica.co.za> wrote:
> Many thanks for your prompt reply and the help
>
> I removed the LAN GW and kept the WAN GW.
>
> I have modified the standard internal and external sip profiles accordingly
>
> What is odd is that if i run a ping from the windows command line, I get a
> reply from the SIP Server. However, if I setup a ping within FS, it fails.
>
> I am investigating...
>
>
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Steven
> Ayre
> Sent: 25 February 2010 10:59 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] SIP Trunk with Private Static IP?
>
> Gateways do not require usernames and passwords. You are required to
> set the parameter, but if no authentication is needed they are ignored
> so you can put anything in the field, so that is not a reason to avoid
> them.
>
> -Steve
>
>
> On 25 February 2010 08:57, Steven Ayre <steveayre at gmail.com> wrote:
>> Create two SIP profiles, each bound to one of your local IPs.
>>
>> You may create a gateway on the profile for the SIP trunk IP for the
>> 10.42.0.1 server, but this is optional.
>>
>> You can then bridge calls via the SIP server using one of:
>> <action application="bridge" data="sofia/gateway/gatewayname"/>
>> <action application="bridge" data="sofia/profilename/number at 10.42.0.1"/>
>>
>> The advantages of using a gateway are:
>> - supports authentication
>> - will monitor the gateway to detect if it goes down (so calls fail
>> instantly rather than after a timeout)
>>
>> As for the default gateway, it is the IP you send via to reach IPs
>> that are not on a network you are connected directly to - you should
>> probably only have one set, and it should be the one you go via to
>> reach the Internet.
>>
>> -Steve
>>
>>
>> On 25 February 2010 08:33, Roly Maz <rm at callrica.co.za> wrote:
>>> Hi Community
>>>
>>>
>>>
>>>
>>>
>>> My Provider provides the following info when they supply a SIP trunk:
>>>
>>>
>>>
>>> ????????? A direct connection into their network. i.e. they provide
> private
>>> IPs:
>>>
>>> ????????? An IP address I must use for my FS box e.g. IP: 10. 42.0.66
> MASK:
>>> 255.255.255.248 GW: 10. 42.0.68
>>>
>>> ????????? An IP address for their SIP server 10.42.0.1
>>>
>>>
>>>
>>> I have setup a dual homed FS box (Windows Server 2008, latest FS version)
>>>
>>>
>>>
>>> NIC 1 ? Internal LAN IP: 10.0.2.185 MASK: 255.255.255.0 GW: 10.0.2.253
>>>
>>> NIC 2 ? SIP Provider (WAN) IP: 10. 42.0.66 MASK: 255.255.255.248 GW: 10.
>>> 42.0.68
>>>
>>>
>>>
>>> Windows complains about multiple gateways ? which I ignore? I can ping
>>> internal addresses ?and the SIP Server
>>>
>>>
>>>
>>> When I fire up FS, I can register Xlite phones on my LAN. I can dial and
>>> hear the test IVR (5000)
>>>
>>>
>>>
>>> This means my Internal SIP Profile is ok.
>>>
>>>
>>>
>>> Now, how do i route a call out to the 10.42.01 SIP Server?
>>>
>>>
>>>
>>> ?Creating a gateway doesn?t make sense, because I am not supplied a
>>> username/password?
>>>
>>>
>>>
>>> Any pointers would be most appreciated, I am sure I am missing something
>>> really simple.
>>>
>>>
>>>
>>> Roland
>>>
>>>
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From rob4manhere at gmail.com  Fri Feb 26 05:10:49 2010
From: rob4manhere at gmail.com (Rob Forman)
Date: Fri, 26 Feb 2010 07:10:49 -0600
Subject: [Freeswitch-users] SIP Trunk with Private Static IP?
In-Reply-To: <eafa772f1002260452h14dd6baavd2a5ac5fa012abd5@mail.gmail.com>
References: <008401cab5f5$4795b910$d6c12b30$@co.za>
	<eafa772f1002250057l3de5b2abu54da8725cbb77281@mail.gmail.com>
	<eafa772f1002250058i2cd7b30ai42f44d0ea1d6eb88@mail.gmail.com>
	<009b01cab62b$b8889c10$2999d430$@co.za>
	<eafa772f1002260452h14dd6baavd2a5ac5fa012abd5@mail.gmail.com>
Message-ID: <AC723E3A-93AD-4FE5-A79D-C59C48197808@gmail.com>

Here's a simple sip ping script (sip_ping.pl) I found and like:

http://pastebin.freeswitch.org/12250

Good luck,
Rob

On Feb 26, 2010, at 6:52 AM, Steven Ayre wrote:

> A SIP 'ping' is not a ICMP ping...
>
> It works by sending a OPTIONS SIP request to the gateway, which then
> responds with 200 OK. It has the advantage of working even if ICMP is
> filtered by a firewall and testing whether the SIP server software is
> running, not just whether the server is online.
>
> Best Regards,
> -Steve
>
>
> On 25 February 2010 15:02, Roly Maz <rm at callrica.co.za> wrote:
>> Many thanks for your prompt reply and the help
>>
>> I removed the LAN GW and kept the WAN GW.
>>
>> I have modified the standard internal and external sip profiles  
>> accordingly
>>
>> What is odd is that if i run a ping from the windows command line,  
>> I get a
>> reply from the SIP Server. However, if I setup a ping within FS, it  
>> fails.
>>
>> I am investigating...
>>
>>
>>
>>
>> -----Original Message-----
>> From: freeswitch-users-bounces at lists.freeswitch.org
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of  
>> Steven
>> Ayre
>> Sent: 25 February 2010 10:59 AM
>> To: freeswitch-users at lists.freeswitch.org
>> Subject: Re: [Freeswitch-users] SIP Trunk with Private Static IP?
>>
>> Gateways do not require usernames and passwords. You are required to
>> set the parameter, but if no authentication is needed they are  
>> ignored
>> so you can put anything in the field, so that is not a reason to  
>> avoid
>> them.
>>
>> -Steve
>>
>>
>> On 25 February 2010 08:57, Steven Ayre <steveayre at gmail.com> wrote:
>>> Create two SIP profiles, each bound to one of your local IPs.
>>>
>>> You may create a gateway on the profile for the SIP trunk IP for the
>>> 10.42.0.1 server, but this is optional.
>>>
>>> You can then bridge calls via the SIP server using one of:
>>> <action application="bridge" data="sofia/gateway/gatewayname"/>
>>> <action application="bridge" data="sofia/profilename/number at 10.42.0.1 
>>> "/>
>>>
>>> The advantages of using a gateway are:
>>> - supports authentication
>>> - will monitor the gateway to detect if it goes down (so calls fail
>>> instantly rather than after a timeout)
>>>
>>> As for the default gateway, it is the IP you send via to reach IPs
>>> that are not on a network you are connected directly to - you should
>>> probably only have one set, and it should be the one you go via to
>>> reach the Internet.
>>>
>>> -Steve
>>>
>>>
>>> On 25 February 2010 08:33, Roly Maz <rm at callrica.co.za> wrote:
>>>> Hi Community
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> My Provider provides the following info when they supply a SIP  
>>>> trunk:
>>>>
>>>>
>>>>
>>>> ?         A direct connection into their network. i.e. they provide
>> private
>>>> IPs:
>>>>
>>>> ?         An IP address I must use for my FS box e.g. IP: 10.  
>>>> 42.0.66
>> MASK:
>>>> 255.255.255.248 GW: 10. 42.0.68
>>>>
>>>> ?         An IP address for their SIP server 10.42.0.1
>>>>
>>>>
>>>>
>>>> I have setup a dual homed FS box (Windows Server 2008, latest FS  
>>>> version)
>>>>
>>>>
>>>>
>>>> NIC 1 ? Internal LAN IP: 10.0.2.185 MASK: 255.255.255.0 GW:  
>>>> 10.0.2.253
>>>>
>>>> NIC 2 ? SIP Provider (WAN) IP: 10. 42.0.66 MASK: 255.255.255.248  
>>>> GW: 10.
>>>> 42.0.68
>>>>
>>>>
>>>>
>>>> Windows complains about multiple gateways ? which I ignore? I can  
>>>> ping
>>>> internal addresses  and the SIP Server
>>>>
>>>>
>>>>
>>>> When I fire up FS, I can register Xlite phones on my LAN. I can  
>>>> dial and
>>>> hear the test IVR (5000)
>>>>
>>>>
>>>>
>>>> This means my Internal SIP Profile is ok.
>>>>
>>>>
>>>>
>>>> Now, how do i route a call out to the 10.42.01 SIP Server?
>>>>
>>>>
>>>>
>>>>  Creating a gateway doesn?t make sense, because I am not supplied a
>>>> username/password?
>>>>
>>>>
>>>>
>>>> Any pointers would be most appreciated, I am sure I am missing  
>>>> something
>>>> really simple.
>>>>
>>>>
>>>>
>>>> Roland
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From javieraristizabal at gmail.com  Fri Feb 26 05:48:45 2010
From: javieraristizabal at gmail.com (=?ISO-8859-1?Q?Javier_Aristiz=E1bal?=)
Date: Fri, 26 Feb 2010 08:48:45 -0500
Subject: [Freeswitch-users] ASR Apps
In-Reply-To: <74a861001002260218i38bfbf72s5637ed20f684a40b@mail.gmail.com>
References: <AC3A71D6-4EAB-4261-B14A-2869C0C05C91@freeswitch.org>
	<74a861001002260218i38bfbf72s5637ed20f684a40b@mail.gmail.com>
Message-ID: <b6f11f301002260548r740ea3fdpd8102d4f2ce1f53@mail.gmail.com>

Hi Brain.. i Want!!! :D

/Javier

On Fri, Feb 26, 2010 at 5:18 AM, david varnes <david.varnes at gmail.com>wrote:

> Hi Brian,
>
> I have just started porting an ASR based framework from
> a VXML engine to use FS.
>
> It is java based, which I know is not a big focus for the
> project ...
>
> Do you have some ASR ports we could use for testing ?
> I am very interested ..
>
> davidv
>
> On 26 February 2010 12:22, Brian West <brian at freeswitch.org> wrote:
> > I'm looking for some enterprising community members to create some
> interesting voice apps using ASR.  Please email me off list and we'll get
> you what you need to do this.
> >
> > Thanks,
> > Brian
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
>
> --
> david varnes
>
> e: david.varnes at gmail.com
> p: +61 404 925 633
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100226/5bae7124/attachment-0002.html 

From moises.silva at gmail.com  Fri Feb 26 07:31:34 2010
From: moises.silva at gmail.com (Moises Silva)
Date: Fri, 26 Feb 2010 10:31:34 -0500
Subject: [Freeswitch-users] smg_prid not bridging the call
In-Reply-To: <7d79b3931002252155t47c86968q9e451482926d93ac@mail.gmail.com>
References: <7d79b3931002252155t47c86968q9e451482926d93ac@mail.gmail.com>
Message-ID: <c4d05cbe1002260731v29dbc428i1e72d24a220c62eb@mail.gmail.com>

Hello lakshmanan,

Please enable the dchan_log=q931 in /etc/wanpipe/smg_pri.conf and then
restart it (smg_ctrl restart), then pastebin the logs

 /var/log/sangoma_pri/dchan_<span-no>.log
/var/log/sangoma_mgd.log

That will contain the Q931 details (if any). Also pastebin your
smg_pri.conf.

Also enable the FreeSWITCH debug logging (see the FreeSWITCH wiki for
details about that) and paste them too.

-- 
Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com

On Fri, Feb 26, 2010 at 12:55 AM, lakshmanan ganapathy <lakindia89 at gmail.com
> wrote:

> Dear all,
> I'm having a A102 Sangoma hardware. I configured it with freeswitch.
> wanrouter status, says both the port as connected.
> My smg_prid version is
>
> Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: ================System
> restart=============
> Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: = Sangoma PRI Protocol Stack
> Daemon       =
> Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: = Version:
> 1.54                           =
> Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: = Date: Feb 15
> 2010                       =
> Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: = Wanpipe Release:
> wanpipe-3.5.8.6       =
> Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: = Revision:Revision:
> 15288                =
> Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid:
> ===========================================
>
> My freeswitch version is 16729.
> I started freeswitch.
>
> oz list
> +OK
> span: 1 (smg_prid)
> type: Sangoma (boost)
> chan_count: 60
> dialplan: XML
> context: default
> dial_regex:
> fail_dial_regex:
> hold_music:
> analog_options none
>
> I originated a call as
> originate openzap/smg_prid/a/9952248266 &park(), which hits my mobile.
>
> But when I issued the following command:
> originate openzap/smg_prid/a/9952248266
> &bridge(openzap/smg_prid/a/8122133885)
> It rings my mobile (9952248266) first, but after that the following error
> was displayed
>
> 2010-02-26 16:20:51.736080 [ERR] switch_ivr_originate.c:2387 Cannot create
> outgoing channel of type [openzap] cause: [NORMAL_CIRCUIT_CONGESTION]
> The call got ended in my mobile.
>
> Freeswitch log and smg_pri.conf
>           http://pastebin.freeswitch.org/12248
> openzap.conf:
> [span wanpipe smg_prid]
> name => smg_prid
> trunk_type =>e1
> b-channel => 1:1-15
> b-channel => 1:17-31
> trunk_type =>e1
> b-channel => 2:1-15
> b-channel => 2:17-31
>
> openzap.conf.xml:
> <configuration name="openzap.conf" description="OpenZAP Configuration">
>   <settings>
>     <param name="debug" value="0"/>
>   </settings>
>   <boost_spans>
>         <span id="smg_prid">
>         <param name="dialplan" value="XML"/>
>         <param name="context" value="default"/>
>         </span>
>   </boost_spans>
> </configuration>
>
> Please guide me to setup this one!!.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100226/0d5cda53/attachment-0002.html 

From christian.loeschenkohl at xpirio.com  Fri Feb 26 08:02:59 2010
From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=)
Date: Fri, 26 Feb 2010 17:02:59 +0100
Subject: [Freeswitch-users] conferences lead to high server load
In-Reply-To: <191c3a031002241058y517e63fdte0d929b97a33f9ac@mail.gmail.com>
References: <4B857226.10308@xpirio.com>
	<191c3a031002241058y517e63fdte0d929b97a33f9ac@mail.gmail.com>
Message-ID: <4B87F0B3.2090606@xpirio.com>

hello

yesterday we did experience high audio delays (2-3 sec) and drops every few seconds.
we had about 70 users in 4 conference rooms, the server had a load of about 40
and used all 4 cpu's (we had a load of 10 with 50 users)

i didn't have the chance to try out -vm so far, the next chance will be this evening
or tomorrow - but i think some change has hit the performance of conferencing very badly.

br

On 2010-02-24 19:58, Anthony Minessale wrote:

> load average has no meaning with FS, you have to look at the CPU usage
> per CPU and thread.
> Are you experiencing any audio problems or are you just concerned about
> that load number?
>
> If you have a box that has trouble with timing it could cost more resources.
> you can always run freeswitch -vm to use an alternate form of timing
> that may not manifest into the load average.
>
>
> 2010/2/24 Christian L?schenkohl <christian.loeschenkohl at xpirio.com
> <mailto:christian.loeschenkohl at xpirio.com>>
>
>     hi
>
>     we do experience a unusual high server load with the latest
>     freeswitch versions.
>     about 50 conference users lead to a server load of over 10 -
>     reproducible by the way.
>     this wans't the case until my latest trunk update.
>
>     fs version: 16714
>     os: debian lenny x86_64
>
>     has something substantially changed in mod_conference recently?
>
>     br
>
>     --
>     Ing. Christian L?schenkohl
>     Technische Leitung, Forschung & Entwicklung VoIP
>
>     xpirio
>     Telekommunikation & Service GmbH
>     Lakeside B04
>     9020 Klagenfurt
>     Austria
>
>     T  +43 (0) 5 77 11 - 1000
>     F  +43 (0) 5 77 11 - 1002
>     E christian.loeschenkohl at xpirio.com
>     <mailto:christian.loeschenkohl at xpirio.com>
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> <mailto:MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> <mailto:sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> <http://iax:guest at conference.freeswitch.org/888>
> googletalk:conf+888 at conference.freeswitch.org
> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


-- 
Ing. Christian L?schenkohl
Technische Leitung, Forschung & Entwicklung VoIP

xpirio
Telekommunikation & Service GmbH
Lakeside B04
9020 Klagenfurt
Austria

T  +43 (0) 5 77 11 - 1000
F  +43 (0) 5 77 11 - 1002
E  christian.loeschenkohl at xpirio.com



From ivanov.maxim at gmail.com  Fri Feb 26 08:23:52 2010
From: ivanov.maxim at gmail.com (Max Ivanov)
Date: Fri, 26 Feb 2010 16:23:52 +0000
Subject: [Freeswitch-users] Multiple gateways dial string and user busy
In-Reply-To: <87f2f3b91002251149m13c0d961k722dd3375f46b6e8@mail.gmail.com>
References: <fbfa7b0c1002240813x58ba8541h7fef5b94f045cbda@mail.gmail.com> 
	<87f2f3b91002251149m13c0d961k722dd3375f46b6e8@mail.gmail.com>
Message-ID: <fbfa7b0c1002260823l11b8d609lf4162eeb61ffc036@mail.gmail.com>

> This is normal for the syntax you're using. You can try setting
> ignore_early_media=true if you don't need call progress tones like ringing
> and busy. It might help to know what the application is before answering
> your question further. What solution are you building?

I use Panasonic station with multiple SIP extensions, each of them is
different gateway in FS. To call panasonics user I use dialstring
sofia/gateway/panas110/223|sofia/gateway/panas111/223|sofia/gateway/panasNNN/223
where 223 is panasonics extension number. Panasonics allows only 1
call per SIP extension, that's why I have to try all of them for each
call, to find first aviable. If SIP extension (but not destination
extension) is occupied by another call it Panasonic return
NO_USER_RESPONSE, if destination extension is busy it returns
USER_BUSY.

Also I use custom calls logging software and each attempt to call to
user appears in logs as tens of call attempts even if user was busy.
What I want to is to stop whole call attempt on USER_BUSY and try next
gateway on NO_USER_RESPONSE. Is it possible? How dialstring have to
look like to achieve that?



From kristian.kielhofner at gmail.com  Fri Feb 26 08:38:15 2010
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Fri, 26 Feb 2010 11:38:15 -0500
Subject: [Freeswitch-users] Multiple gateways dial string and user busy
In-Reply-To: <fbfa7b0c1002260823l11b8d609lf4162eeb61ffc036@mail.gmail.com>
References: <fbfa7b0c1002240813x58ba8541h7fef5b94f045cbda@mail.gmail.com>
	<87f2f3b91002251149m13c0d961k722dd3375f46b6e8@mail.gmail.com>
	<fbfa7b0c1002260823l11b8d609lf4162eeb61ffc036@mail.gmail.com>
Message-ID: <4d15ff861002260838v6aa624a1t398eb50bddc75ab1@mail.gmail.com>

http://wiki.freeswitch.org/wiki/Channel_Variables#failure_causes

On Fri, Feb 26, 2010 at 11:23 AM, Max Ivanov <ivanov.maxim at gmail.com> wrote:
>> This is normal for the syntax you're using. You can try setting
>> ignore_early_media=true if you don't need call progress tones like ringing
>> and busy. It might help to know what the application is before answering
>> your question further. What solution are you building?
>
> I use Panasonic station with multiple SIP extensions, each of them is
> different gateway in FS. To call panasonics user I use dialstring
> sofia/gateway/panas110/223|sofia/gateway/panas111/223|sofia/gateway/panasNNN/223
> where 223 is panasonics extension number. Panasonics allows only 1
> call per SIP extension, that's why I have to try all of them for each
> call, to find first aviable. If SIP extension (but not destination
> extension) is occupied by another call it Panasonic return
> NO_USER_RESPONSE, if destination extension is busy it returns
> USER_BUSY.
>
> Also I use custom calls logging software and each attempt to call to
> user appears in logs as tens of call attempts even if user was busy.
> What I want to is to stop whole call attempt on USER_BUSY and try next
> gateway on NO_USER_RESPONSE. Is it possible? How dialstring have to
> look like to achieve that?
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com



From kristian.kielhofner at gmail.com  Fri Feb 26 08:41:45 2010
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Fri, 26 Feb 2010 11:41:45 -0500
Subject: [Freeswitch-users] SIP Trunk with Private Static IP?
In-Reply-To: <eafa772f1002260452h14dd6baavd2a5ac5fa012abd5@mail.gmail.com>
References: <008401cab5f5$4795b910$d6c12b30$@co.za>
	<eafa772f1002250057l3de5b2abu54da8725cbb77281@mail.gmail.com>
	<eafa772f1002250058i2cd7b30ai42f44d0ea1d6eb88@mail.gmail.com>
	<009b01cab62b$b8889c10$2999d430$@co.za>
	<eafa772f1002260452h14dd6baavd2a5ac5fa012abd5@mail.gmail.com>
Message-ID: <4d15ff861002260841s3ae98464l64244285523bc1ab@mail.gmail.com>

Keep in mind the remote side might not always send a 200 OK.  Some
send 404, 501, 503, etc.  When pinging a gateway (I believe) FS treats
all of them the same.  As long as *something* comes back the gw is
marked "up".  On some software (OpenSIPS with various gw modules, etc)
this behavior is configurable.  Not sure about FS.

On Fri, Feb 26, 2010 at 7:52 AM, Steven Ayre <steveayre at gmail.com> wrote:
> A SIP 'ping' is not a ICMP ping...
>
> It works by sending a OPTIONS SIP request to the gateway, which then
> responds with 200 OK. It has the advantage of working even if ICMP is
> filtered by a firewall and testing whether the SIP server software is
> running, not just whether the server is online.
>
> Best Regards,
> -Steve


-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com



From mike at jerris.com  Fri Feb 26 08:45:22 2010
From: mike at jerris.com (Michael Jerris)
Date: Fri, 26 Feb 2010 11:45:22 -0500
Subject: [Freeswitch-users] Dingaling - external rtp supported?
In-Reply-To: <29b888f81002242204x171f4ab3la34d4ace604cd50a@mail.gmail.com>
References: <29b888f81002160835y6e290319ub0d04c088caa1b79@mail.gmail.com>
	<DEA39AD1-0EE9-473A-B78C-CDDF3F100578@freeswitch.org>
	<29b888f81002170318p63622f3fhc8d26ad0e85602d2@mail.gmail.com>
	<191c3a031002170538r3b029515i8a000812a70ad558@mail.gmail.com>
	<29b888f81002170741y3d13cb57ycb4436a372c4eded@mail.gmail.com>
	<191c3a031002171136q7424e9bai34379abb542daf93@mail.gmail.com>
	<29b888f81002210800v617bca9dk4bf1abf1bd791ee0@mail.gmail.com>
	<29b888f81002222359w6ae4cd8ej944d453a5b099af4@mail.gmail.com>
	<191c3a031002231131x19c9791cgd986a6c1be18e09a@mail.gmail.com>
	<29b888f81002242204x171f4ab3la34d4ace604cd50a@mail.gmail.com>
Message-ID: <67ABA2B8-8335-486A-A43F-0025ECD13D7E@jerris.com>

There is no recommendation because no one has ever contributed a working build for windows, if there was, it would just build and work.


On Feb 25, 2010, at 1:04 AM, Moiz Chinoy wrote:

>  
> I was using GuntTls-2.7.3 for windows. Now I am using GuntTls-2.9.9. I have modified only gnutls.h, added following line:
>  
> typedef long ssize_t;
>  
> because otherwise it was giving errors...
>  
> What is the recommended version of the TLS lib for windows?
>  
> After upgrading the the GnuTls and freeswitch to rev 16806, I ran the freeswitch with mod_dingalilg enabled. Once started, I issued just the 'shutdown' command on the console, exception happened.
>  
> ......................
> 2010-02-25 09:45:29.795285 [CONSOLE] switch_loadable_module.c:1277 Stopping: CORE_SOFTTIMER_MODULE
> 2010-02-25 09:45:29.810910 [CONSOLE] switch_time.c:780 Soft timer thread exiting.
> 2010-02-25 09:45:29.810910 [NOTICE] switch_loadable_module.c:98 Thread ended for CORE_SOFTTIMER_MODULE
> 2010-02-25 09:45:29.826535 [DEBUG] switch_loadable_module.c:456 Write lock interface 'dingaling' to wait for existing references.
> 2010-02-25 09:45:29.826535 [NOTICE] switch_loadable_module.c:464 Deleting Endpoint 'dingaling'
> 2010-02-25 09:45:29.826535 [NOTICE] switch_loadable_module.c:572 Deleting API Function 'dl_debug'
> 2010-02-25 09:45:29.826535 [DEBUG] switch_loadable_module.c:574 Write lock interface 'dl_debug' to wait for existing references.
> 2010-02-25 09:45:29.826535 [NOTICE] switch_loadable_module.c:572 Deleting API Function 'dl_pres'
> 2010-02-25 09:45:29.826535 [DEBUG] switch_loadable_module.c:574 Write lock interface 'dl_pres' to wait for existing references.
> 2010-02-25 09:45:29.826535 [NOTICE] switch_loadable_module.c:572 Deleting API Function 'dl_logout'
> 2010-02-25 09:45:29.826535 [DEBUG] switch_loadable_module.c:574 Write lock interface 'dl_logout' to wait for existing references.
> 2010-02-25 09:45:29.826535 [NOTICE] switch_loadable_module.c:572 Deleting API Function 'dl_login'
> 2010-02-25 09:45:29.826535 [DEBUG] switch_loadable_module.c:574 Write lock interface 'dl_login' to wait for existing references.
> 2010-02-25 09:45:29.826535 [NOTICE] switch_loadable_module.c:572 Deleting API Function 'dingaling'
> 2010-02-25 09:45:29.826535 [DEBUG] switch_loadable_module.c:574 Write lock interface 'dingaling' to wait for existing references.
> 2010-02-25 09:45:29.826535 [DEBUG] switch_loadable_module.c:710 Write lock interface 'jingle' to wait for existing references.
> 2010-02-25 09:45:29.826535 [NOTICE] switch_loadable_module.c:719 Deleting Chat interface 'jingle'
> 2010-02-25 09:45:29.826535 [CONSOLE] switch_loadable_module.c:1277 Stopping: mod_dingaling
> 2010-02-25 09:45:31.185910 [DEBUG] libdingaling.c:1546 io error 2 7 retry in 1 second(s)
> ........................
>  
> And the code went in the stream.c...
>  
> int
> iks_fd (iksparser *prs)
> {
> struct stream_data *data;
>  
> if (prs) {
> data = iks_user_data (prs);
> if (data) {
> return (int) data->sock;
> }
> }
> return -1;
> }
>  
>  
>  
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100226/45e519e0/attachment-0002.html 

From anthony.minessale at gmail.com  Fri Feb 26 08:54:08 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 26 Feb 2010 10:54:08 -0600
Subject: [Freeswitch-users] Dingaling - external rtp supported?
In-Reply-To: <67ABA2B8-8335-486A-A43F-0025ECD13D7E@jerris.com>
References: <29b888f81002160835y6e290319ub0d04c088caa1b79@mail.gmail.com>
	<29b888f81002170318p63622f3fhc8d26ad0e85602d2@mail.gmail.com>
	<191c3a031002170538r3b029515i8a000812a70ad558@mail.gmail.com>
	<29b888f81002170741y3d13cb57ycb4436a372c4eded@mail.gmail.com>
	<191c3a031002171136q7424e9bai34379abb542daf93@mail.gmail.com>
	<29b888f81002210800v617bca9dk4bf1abf1bd791ee0@mail.gmail.com>
	<29b888f81002222359w6ae4cd8ej944d453a5b099af4@mail.gmail.com>
	<191c3a031002231131x19c9791cgd986a6c1be18e09a@mail.gmail.com>
	<29b888f81002242204x171f4ab3la34d4ace604cd50a@mail.gmail.com>
	<67ABA2B8-8335-486A-A43F-0025ECD13D7E@jerris.com>
Message-ID: <191c3a031002260854q550618bbj8c0e54a6bb5ded54@mail.gmail.com>

its probably not an upgrade you need, more likely a downgrade (1 or 2 years
ago version), and we have no idea, as mike said, nobody has contributed it
back in working order so maybe you can ask whoever showed you how to add
your own gnutls how they did it.


On Fri, Feb 26, 2010 at 10:45 AM, Michael Jerris <mike at jerris.com> wrote:

> There is no recommendation because no one has ever contributed a working
> build for windows, if there was, it would just build and work.
>
>
> On Feb 25, 2010, at 1:04 AM, Moiz Chinoy wrote:
>
>
> I was using GuntTls-2.7.3 for windows. Now I am using GuntTls-2.9.9. I have
> modified only gnutls.h, added following line:
>
> typedef long ssize_t;
>
> because otherwise it was giving errors...
>
> What is the recommended version of the TLS lib for windows?
>
> After upgrading the the GnuTls and freeswitch to rev 16806, I ran the
> freeswitch with mod_dingalilg enabled. Once started, I issued just the
> 'shutdown' command on the console, exception happened.
>
> ......................
> 2010-02-25 09:45:29.795285 [CONSOLE] switch_loadable_module.c:1277
> Stopping: CORE_SOFTTIMER_MODULE
> 2010-02-25 09:45:29.810910 [CONSOLE] switch_time.c:780 Soft timer thread
> exiting.
> 2010-02-25 09:45:29.810910 [NOTICE] switch_loadable_module.c:98 Thread
> ended for CORE_SOFTTIMER_MODULE
> 2010-02-25 09:45:29.826535 [DEBUG] switch_loadable_module.c:456 Write lock
> interface 'dingaling' to wait for existing references.
> 2010-02-25 09:45:29.826535 [NOTICE] switch_loadable_module.c:464 Deleting
> Endpoint 'dingaling'
> 2010-02-25 09:45:29.826535 [NOTICE] switch_loadable_module.c:572 Deleting
> API Function 'dl_debug'
> 2010-02-25 09:45:29.826535 [DEBUG] switch_loadable_module.c:574 Write lock
> interface 'dl_debug' to wait for existing references.
> 2010-02-25 09:45:29.826535 [NOTICE] switch_loadable_module.c:572 Deleting
> API Function 'dl_pres'
> 2010-02-25 09:45:29.826535 [DEBUG] switch_loadable_module.c:574 Write lock
> interface 'dl_pres' to wait for existing references.
> 2010-02-25 09:45:29.826535 [NOTICE] switch_loadable_module.c:572 Deleting
> API Function 'dl_logout'
> 2010-02-25 09:45:29.826535 [DEBUG] switch_loadable_module.c:574 Write lock
> interface 'dl_logout' to wait for existing references.
> 2010-02-25 09:45:29.826535 [NOTICE] switch_loadable_module.c:572 Deleting
> API Function 'dl_login'
> 2010-02-25 09:45:29.826535 [DEBUG] switch_loadable_module.c:574 Write lock
> interface 'dl_login' to wait for existing references.
> 2010-02-25 09:45:29.826535 [NOTICE] switch_loadable_module.c:572 Deleting
> API Function 'dingaling'
> 2010-02-25 09:45:29.826535 [DEBUG] switch_loadable_module.c:574 Write lock
> interface 'dingaling' to wait for existing references.
> 2010-02-25 09:45:29.826535 [DEBUG] switch_loadable_module.c:710 Write lock
> interface 'jingle' to wait for existing references.
> 2010-02-25 09:45:29.826535 [NOTICE] switch_loadable_module.c:719 Deleting
> Chat interface 'jingle'
> 2010-02-25 09:45:29.826535 [CONSOLE] switch_loadable_module.c:1277
> Stopping: mod_dingaling
> 2010-02-25 09:45:31.185910 [DEBUG] libdingaling.c:1546 io error 2 7 retry
> in 1 second(s)
> ........................
>
> And the code went in the stream.c...
>
> int
> iks_fd (iksparser *prs)
> {
> struct stream_data *data;
>
> if (prs) {
> data = iks_user_data (prs);
> if (data) {
> return (int) data->sock;
> }
> }
> return -1;
> }
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100226/32e8a5a2/attachment-0002.html 

From Suneel.Papineni at mettoni.com  Fri Feb 26 08:59:36 2010
From: Suneel.Papineni at mettoni.com (Suneel Papineni)
Date: Fri, 26 Feb 2010 16:59:36 -0000
Subject: [Freeswitch-users] FSComm basic issue
Message-ID: <3181A30B8C35AB4AA8577B78DDF4613806886903@nickel.mettonigroup.com>

Hi,

 

I am trying to use FSComm with Freeswitch and facing following issues. 

 

1.       Using pre-build binary (windows), when the application is
started FSComm is getting Registered properly. When I tried to make a
call, UI displays Dialing... but unable to see any SIP (INVITE) messages
in wireshark traces. After sometime UI displays with message "Call with
(destination number) failed with reason DESTINATION_OUT_OF_ORDER though
destination number is registered with another FSComm"

2.       Also I am unable to see any logs generated in the log folder.

 

Downloaded the latest source code (Freeswitch 1.0.5 latest updated as on
26/02/10 at 4am) and tried to build FSComm. Build was succeeded.
Application (FSComm) also started and displayed with UI. When I try to
change the preferences, it has thrown Porta Audio Error saying "Error
Querying Audio Devices" even though proper audio devices are present.
Also it doesn't create folders like "conf", "mod". Even after copying
all the required dll's and mod files (as specified in FSComm wiki
pages), application is throwing the same error.

 

I am using Windows XP machine. Built a Debug & Release version with
32-bit option.

If someone has built FSComm for windows environment and is working fine,
could you please let me know if there are any additional things I need
to do to make it work.

 

Thanks & Regards

Suneel

 

 


*************************************************************************
Please consider the environment before printing this e-mail
*************************************************************************
This email and any files transmitted with it are confidential and
intended solely for the use of the individual or entity to whom they
are addressed. If you have received this email in error please notify
the system manager.  http://www.mettoni.com

Mettoni Ltd
Registered in England and Wales: 4485956
9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN
*************************************************************************

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100226/f188c6ac/attachment-0002.html 

From mbsip at gazeta.pl  Fri Feb 26 09:10:30 2010
From: mbsip at gazeta.pl (Maciej Bylica)
Date: Fri, 26 Feb 2010 18:10:30 +0100
Subject: [Freeswitch-users] Phrases - Can't find macro
Message-ID: <28f27f5d1002260910h7b3ce773l28294e7354078b05@mail.gmail.com>

Hello,

I am playing around with Phrases to use them with conference application.
But i've encountered rudimentary problem of how to use newly added macro.

What I already did is (according to wiki Speech Phrase Management)
http://wiki.freeswitch.org/wiki/Speech_Phrase_Management
- confirmed that there is "mod_say_en" loaded (<load module="mod_say_en"/>)
- confirmed that there are proper .wav files
- modified onf/lang/en/en.xml file:
 <section name="phrases" description="Speech Phrase Management">
   <macros>
     <language name="en" sound_path="$${sounds_dir}/en/us/callie"
tts_engine="cepstral" tts_voice="callie">
       <macro name="confwelcome">
         <input pattern="(.*)">
           <match>
              <action function="play-file" data="vm-youhave.wav"/>
<!-- just to test the phrase-->
           </match>
     </language>
   </macros>
 </section>
- modified a part of dialplan:
    <extension name="conference_48112223344">
      <condition field="destination_number" expression="^(\d{11})$">
         <action application="phrase" data="confwelcome"/>
      </condition>
     </extension>

I have following outcome:
2010-02-26 19:57:09.487245 [DEBUG] switch_core_session.c:638 Send
signal sofia/internal/1000 at 217.153.192.36 [BREAK]
EXECUTE sofia/internal/1000 at 217.153.192.36 phrase(confwelcome)
2010-02-26 19:57:09.487245 [DEBUG] mod_dptools.c:1850 Execute confwelcome() lang
2010-02-26 19:57:09.487245 [DEBUG] switch_ivr_play_say.c:118 No
language specified - Using [en]
2010-02-26 19:57:09.496322 [ERR] switch_ivr_play_say.c:202 Can't find
macro confwelcome.
2010-02-26 19:57:09.496322 [WARNING] switch_ivr_play_say.c:368 Macro
[confwelcome] did not match any patterns



Strange is that if i use a wiki example, it works. To be more precise:
- conf/lang/en/en.xml file was overwritten with an example macros
(directly from aforementioned wiki).
- dialplan was modified:
    <extension name="conference_48112223344">
      <condition field="destination_number" expression="^(\d{11})$">
         <action application="phrase" data="msgcount"/>
      </condition>
     </extension>


Am i doing something wrong?

Thx,
Maciej.



From lawwton at gmail.com  Fri Feb 26 10:19:00 2010
From: lawwton at gmail.com (Alfredo Quiroga-Villamil)
Date: Fri, 26 Feb 2010 13:19:00 -0500
Subject: [Freeswitch-users] Conference - Originate Question
Message-ID: <5fe6fa8f1002261019p3812ce5bq297e91f7ddeda1ab@mail.gmail.com>

All:

I am currently using the following cmd to dynamically create a conference:

originate sofia/gateway/url/1XXXXXXXXXX &conference(myconf)

I have noticed that when I send that cmd even if I specify:

originate sofia/gateway/url/1XXXXXXXXXX &conference(myconf) public

I am not hitting the dialplan. Is there a way to send the command and
force it to hit the dialplan?

Thanks in advance,

Alfredo



From rupa at rupa.com  Fri Feb 26 10:50:48 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Fri, 26 Feb 2010 12:50:48 -0600
Subject: [Freeswitch-users] Conference - Originate Question
In-Reply-To: <5fe6fa8f1002261019p3812ce5bq297e91f7ddeda1ab@mail.gmail.com>
References: <5fe6fa8f1002261019p3812ce5bq297e91f7ddeda1ab@mail.gmail.com>
Message-ID: <b39273531002261050m327590bfs637bd9fff1f40391@mail.gmail.com>

Use the loopback endpoint to have it go back through the dialplan.

http://wiki.freeswitch.org/wiki/Loopback

On Fri, Feb 26, 2010 at 12:19 PM, Alfredo Quiroga-Villamil <
lawwton at gmail.com> wrote:

> All:
>
> I am currently using the following cmd to dynamically create a conference:
>
> originate sofia/gateway/url/1XXXXXXXXXX &conference(myconf)
>
> I have noticed that when I send that cmd even if I specify:
>
> originate sofia/gateway/url/1XXXXXXXXXX &conference(myconf) public
>
> I am not hitting the dialplan. Is there a way to send the command and
> force it to hit the dialplan?
>
> Thanks in advance,
>
> Alfredo
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
-Rupa
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100226/3c68ca16/attachment-0002.html 

From phunk0000 at hotmail.com  Fri Feb 26 10:53:25 2010
From: phunk0000 at hotmail.com (Todd)
Date: Fri, 26 Feb 2010 13:53:25 -0500
Subject: [Freeswitch-users] actition after a set time during call
Message-ID: <BLU110-DS12F13B3418ABF2053C86AAA73F0@phx.gbl>

Hey List-

 

I want to have nibblebill pause after a certain time during a call.  I was
wondering what the best way to put this into the dialplan is?  Still kind of
new to this.. 
 
 <action application="nibblebill" data="pause"/>
 
Is the action I want to implement 2 minutes into a call. what is the best
way to do this?  I have the nibblerate set in the individual extension XMLs
and the nibblerate heartbeat set in the nibble.conf.xml  Thanks

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100226/3e630b68/attachment-0002.html 

From jerry.richards at teotech.com  Fri Feb 26 10:57:06 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Fri, 26 Feb 2010 10:57:06 -0800
Subject: [Freeswitch-users] 415 Unsupported Media Handling
Message-ID: <AE66B63BAF424BBCBE585E244E0C42B4@greyhawk.tonecommander.com>


I have two types of devices, one supports text/html MESSAGE content and one
that only supports text/plain MESSAGE content.  When I send an IM from the
first to the second, the second replies with 415 Unsupported Media Type (as
shown below).  Section 8.1.3.5 of RFC 3261 says the sender should retry
using the media type acceptable to the receiver (in this case: plain/text).

The problem I have is that Freeswitch doesn't pass the error back to the
sender (nor does it retry itself using plain/text).  So the IM is lost.
Does anyone see the reason why the error is not being handled correctly?

   ------------------------------------------------------------------------
send 668 bytes to udp/[192.168.72.141]:5062 at 18:43:24.720446:
   ------------------------------------------------------------------------
   MESSAGE sip:5398 at 192.168.72.141:5062 SIP/2.0
   Via: SIP/2.0/UDP 192.168.72.141;rport;branch=z9hG4bKXg6SNUcQa5Kre
   Max-Forwards: 70
   From: "5382 on 141" <sip:5382 at 192.168.72.141>;tag=66661130
   To: "5398" <sip:5398 at 192.168.72.141:5062>
   Call-ID: a9891a15-9da9-122d-b3ad-003048d7e9f0
   CSeq: 127444135 MESSAGE
   Contact: <sip:mod_sofia at 192.168.72.141:5060>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-32M
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: timer, precondition, path, replaces
   Content-Type: text/html
   Content-Length: 63

   <font size="2" face="Arial">hello this is Jerry from Teo</font>
   ------------------------------------------------------------------------
recv 459 bytes from udp/[192.168.72.141]:5062 at 18:43:24.728390:
   ------------------------------------------------------------------------
   SIP/2.0 415 Unsupported media type
   Via: SIP/2.0/UDP
192.168.72.141;rport=5060;branch=z9hG4bKXg6SNUcQa5Kre;received=192.168.72.14
1
   From: "5382 on 141" <sip:5382 at 192.168.72.141>;tag=66661130
   To: "5398" <sip:5398 at 192.168.72.141:5062>
   Call-ID: a9891a15-9da9-122d-b3ad-003048d7e9f0
   CSeq: 127444135 MESSAGE
   Date: Fri, 26 Feb 2010 18:43:24 GMT
   User-Agent: MobilityGateway-2.0.34078
   Server: MobilityGateway-2.0.34078
   Accept: text/plain
   Content-Length: 0

Best Regards,
Jerry




From brian at freeswitch.org  Fri Feb 26 11:01:04 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 26 Feb 2010 13:01:04 -0600
Subject: [Freeswitch-users] 415 Unsupported Media Handling
In-Reply-To: <AE66B63BAF424BBCBE585E244E0C42B4@greyhawk.tonecommander.com>
References: <AE66B63BAF424BBCBE585E244E0C42B4@greyhawk.tonecommander.com>
Message-ID: <835A36B2-1C04-42B0-86C1-36528E93589D@freeswitch.org>

Its really clear here you'll need to say text/plain in the content type their accept header says they only take text/plain.

/b

On Feb 26, 2010, at 12:57 PM, Jerry Richards wrote:

> 
> I have two types of devices, one supports text/html MESSAGE content and one
> that only supports text/plain MESSAGE content.  When I send an IM from the
> first to the second, the second replies with 415 Unsupported Media Type (as
> shown below).  Section 8.1.3.5 of RFC 3261 says the sender should retry
> using the media type acceptable to the receiver (in this case: plain/text).
> 
> The problem I have is that Freeswitch doesn't pass the error back to the
> sender (nor does it retry itself using plain/text).  So the IM is lost.
> Does anyone see the reason why the error is not being handled correctly?
> 
>   ------------------------------------------------------------------------
> send 668 bytes to udp/[192.168.72.141]:5062 at 18:43:24.720446:
>   ------------------------------------------------------------------------
>   MESSAGE sip:5398 at 192.168.72.141:5062 SIP/2.0
>   Via: SIP/2.0/UDP 192.168.72.141;rport;branch=z9hG4bKXg6SNUcQa5Kre
>   Max-Forwards: 70
>   From: "5382 on 141" <sip:5382 at 192.168.72.141>;tag=66661130
>   To: "5398" <sip:5398 at 192.168.72.141:5062>
>   Call-ID: a9891a15-9da9-122d-b3ad-003048d7e9f0
>   CSeq: 127444135 MESSAGE
>   Contact: <sip:mod_sofia at 192.168.72.141:5060>
>   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-32M
>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>   Supported: timer, precondition, path, replaces
>   Content-Type: text/html
>   Content-Length: 63
> 
>   <font size="2" face="Arial">hello this is Jerry from Teo</font>
>   ------------------------------------------------------------------------
> recv 459 bytes from udp/[192.168.72.141]:5062 at 18:43:24.728390:
>   ------------------------------------------------------------------------
>   SIP/2.0 415 Unsupported media type
>   Via: SIP/2.0/UDP
> 192.168.72.141;rport=5060;branch=z9hG4bKXg6SNUcQa5Kre;received=192.168.72.14
> 1
>   From: "5382 on 141" <sip:5382 at 192.168.72.141>;tag=66661130
>   To: "5398" <sip:5398 at 192.168.72.141:5062>
>   Call-ID: a9891a15-9da9-122d-b3ad-003048d7e9f0
>   CSeq: 127444135 MESSAGE
>   Date: Fri, 26 Feb 2010 18:43:24 GMT
>   User-Agent: MobilityGateway-2.0.34078
>   Server: MobilityGateway-2.0.34078
>   Accept: text/plain
>   Content-Length: 0
> 
> Best Regards,
> Jerry
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From lawwton at gmail.com  Fri Feb 26 11:24:03 2010
From: lawwton at gmail.com (Alfredo Quiroga-Villamil)
Date: Fri, 26 Feb 2010 14:24:03 -0500
Subject: [Freeswitch-users] Conference - Originate Question
In-Reply-To: <b39273531002261050m327590bfs637bd9fff1f40391@mail.gmail.com>
References: <5fe6fa8f1002261019p3812ce5bq297e91f7ddeda1ab@mail.gmail.com>
	<b39273531002261050m327590bfs637bd9fff1f40391@mail.gmail.com>
Message-ID: <5fe6fa8f1002261124o27b5921fl7ca64ba5b32e9c92@mail.gmail.com>

Thanks Rupa.

On Fri, Feb 26, 2010 at 1:50 PM, Rupa Schomaker <rupa at rupa.com> wrote:
> Use the loopback endpoint to have it go back through the dialplan.
> http://wiki.freeswitch.org/wiki/Loopback
>
> On Fri, Feb 26, 2010 at 12:19 PM, Alfredo Quiroga-Villamil
> <lawwton at gmail.com> wrote:
>>
>> All:
>>
>> I am currently using the following cmd to dynamically create a conference:
>>
>> originate sofia/gateway/url/1XXXXXXXXXX &conference(myconf)
>>
>> I have noticed that when I send that cmd even if I specify:
>>
>> originate sofia/gateway/url/1XXXXXXXXXX &conference(myconf) public
>>
>> I am not hitting the dialplan. Is there a way to send the command and
>> force it to hit the dialplan?
>>
>> Thanks in advance,
>>
>> Alfredo
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From msc at freeswitch.org  Fri Feb 26 11:28:28 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 26 Feb 2010 11:28:28 -0800
Subject: [Freeswitch-users] Cannot make calls through PRI via wanpipe
	running as daemon
In-Reply-To: <6FCED74CF54F466AA25CBDA7DFF42198@greyhawk.tonecommander.com>
References: <6FCED74CF54F466AA25CBDA7DFF42198@greyhawk.tonecommander.com>
Message-ID: <87f2f3b91002261128n17ea78acx160fcfdb04532004@mail.gmail.com>

On Thu, Feb 25, 2010 at 5:16 PM, Robert Hadley <robert.hadley at teotech.com>wrote:

>  When running Freeswitch as service called teoswitch as user teoswitch I
> cannot make calls through the Sangoma PRI or analog cards using wanpipe
> driver.  I have added a file called 30-wanpipe.rules to /etc/udev/rules.d
> and rebooted the server.
>
>
>
> cat 30-wanpipe.rules
>
> # /etc/udev/rules.d/30-wanpipe.rules
>
> SUBSYSTEM=="wptdm", OWNER="teoswitch", GROUP="teoswitch", MODE="0660"
>
> SUBSYSTEM=="wanpipe", OWNER="teoswitch", GROUP="teoswitch", MODE="0660"
>
>
>
>
>
> Freeswitch log:
>
> Dialplan: sofia/internal/5410 at 192.168.72.45:5060 parsing
> [default->SangomaPRI] continue=false
>
> Dialplan: sofia/internal/5410 at 192.168.72.45:5060 Regex (PASS) [SangomaPRI]
> destination_number(93491045) =~ /^9(\d+)$/ break=on-false
>
> Dialplan: sofia/internal/5410 at 192.168.72.45:5060 Action
> set(effective_caller_id_number=425740${caller_id_number})
>
> Dialplan: sofia/internal/5410 at 192.168.72.45:5060 Action
> bridge(openzap/smg_prid/a/3491045 at g1)
>
> 2010-02-25 16:51:11.328635 [DEBUG] switch_core_state_machine.c:122
> (sofia/internal/5410 at 192.168.72.45:5060) State Change CS_ROUTING ->
> CS_EXECUTE
>
> 2010-02-25 16:51:11.328635 [DEBUG] switch_core_session.c:1019 Send signal
> sofia/internal/5410 at 192.168.72.45:5060 [BREAK]
>
> 2010-02-25 16:51:11.328635 [DEBUG] switch_core_state_machine.c:341
> (sofia/internal/5410 at 192.168.72.45:5060) State ROUTING going to sleep
>
> 2010-02-25 16:51:11.328635 [DEBUG] switch_core_state_machine.c:314
> (sofia/internal/5410 at 192.168.72.45:5060) Running State Change CS_EXECUTE
>
> 2010-02-25 16:51:11.328635 [DEBUG] switch_core_state_machine.c:348
> (sofia/internal/5410 at 192.168.72.45:5060) State EXECUTE
>
> 2010-02-25 16:51:11.328635 [DEBUG] mod_sofia.c:181 sofia/internal/
> 5410 at 192.168.72.45:5060 SOFIA EXECUTE
>
> 2010-02-25 16:51:11.328635 [DEBUG] switch_core_state_machine.c:159
> sofia/internal/5410 at 192.168.72.45:5060 Standard EXECUTE
>
> EXECUTE sofia/internal/5410 at 192.168.72.45:5060set(effective_caller_id_number=4257405410)
>
> 2010-02-25 16:51:11.328635 [DEBUG] mod_dptools.c:811 sofia/internal/
> 5410 at 192.168.72.45:5060 SET [effective_caller_id_number]=[4257405410]
>
> EXECUTE sofia/internal/5410 at 192.168.72.45:5060bridge(openzap/smg_prid/a/3491045 at g1
> )
>
> 2010-02-25 16:51:11.339637 [ERR] switch_core_session.c:357 Could not locate
> channel type openzap
>
> 2010-02-25 16:51:11.339637 [ERR] switch_ivr_originate.c:2411 Cannot create
> outgoing channel of type [openzap] cause: [CHAN_NOT_IMPLEMENTED]
>
> 2010-02-25 16:51:11.339637 [DEBUG] switch_ivr_originate.c:3209 Originate
> Resulted in Error Cause: 66 [CHAN_NOT_IMPLEMENTED]
>
> 2010-02-25 16:51:11.339637 [INFO] mod_dptools.c:2346 Originate Failed.
> Cause: CHAN_NOT_IMPLEMENTED
>

CHAN_NOT_IMPLEMENTED implies that OpenZAP did not load. Capture the output
of "load mod_openzap" and look for the reason that it is failing to load.
-MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100226/c722137c/attachment-0002.html 

From jerry.richards at teotech.com  Fri Feb 26 11:43:48 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Fri, 26 Feb 2010 11:43:48 -0800
Subject: [Freeswitch-users] 415 Unsupported Media Handling
In-Reply-To: <835A36B2-1C04-42B0-86C1-36528E93589D@freeswitch.org>
References: <AE66B63BAF424BBCBE585E244E0C42B4@greyhawk.tonecommander.com>
	<835A36B2-1C04-42B0-86C1-36528E93589D@freeswitch.org>
Message-ID: <ADCCADE853D6427CBC7CD978E7E0A0ED@greyhawk.tonecommander.com>

I'm not sure I follow your comment.  The first device prefers text/html so
that's what it normally sets in the initial MESSAGE.  Devices that support
text/html will not generate this 415 error reply.  It's only devices that
don't support it that would send the 415 reply, so the issue is that the 415
is not getting passed back to the originator.

Best Regards,
Jerry


-----Original Message-----
From: Brian West [mailto:brian at freeswitch.org] 
Sent: Friday, February 26, 2010 11:01 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] 415 Unsupported Media Handling

Its really clear here you'll need to say text/plain in the content type
their accept header says they only take text/plain.

/b

On Feb 26, 2010, at 12:57 PM, Jerry Richards wrote:

> 
> I have two types of devices, one supports text/html MESSAGE content 
> and one that only supports text/plain MESSAGE content.  When I send an 
> IM from the first to the second, the second replies with 415 
> Unsupported Media Type (as shown below).  Section 8.1.3.5 of RFC 3261 
> says the sender should retry using the media type acceptable to the
receiver (in this case: plain/text).
> 
> The problem I have is that Freeswitch doesn't pass the error back to 
> the sender (nor does it retry itself using plain/text).  So the IM is
lost.
> Does anyone see the reason why the error is not being handled correctly?
> 
>   
> ----------------------------------------------------------------------
> -- send 668 bytes to udp/[192.168.72.141]:5062 at 18:43:24.720446:
>   ------------------------------------------------------------------------
>   MESSAGE sip:5398 at 192.168.72.141:5062 SIP/2.0
>   Via: SIP/2.0/UDP 192.168.72.141;rport;branch=z9hG4bKXg6SNUcQa5Kre
>   Max-Forwards: 70
>   From: "5382 on 141" <sip:5382 at 192.168.72.141>;tag=66661130
>   To: "5398" <sip:5398 at 192.168.72.141:5062>
>   Call-ID: a9891a15-9da9-122d-b3ad-003048d7e9f0
>   CSeq: 127444135 MESSAGE
>   Contact: <sip:mod_sofia at 192.168.72.141:5060>
>   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-32M
>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, 
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>   Supported: timer, precondition, path, replaces
>   Content-Type: text/html
>   Content-Length: 63
> 
>   <font size="2" face="Arial">hello this is Jerry from Teo</font>
>   
> ----------------------------------------------------------------------
> -- recv 459 bytes from udp/[192.168.72.141]:5062 at 18:43:24.728390:
>   ------------------------------------------------------------------------
>   SIP/2.0 415 Unsupported media type
>   Via: SIP/2.0/UDP
> 192.168.72.141;rport=5060;branch=z9hG4bKXg6SNUcQa5Kre;received=192.168
> .72.14
> 1
>   From: "5382 on 141" <sip:5382 at 192.168.72.141>;tag=66661130
>   To: "5398" <sip:5398 at 192.168.72.141:5062>
>   Call-ID: a9891a15-9da9-122d-b3ad-003048d7e9f0
>   CSeq: 127444135 MESSAGE
>   Date: Fri, 26 Feb 2010 18:43:24 GMT
>   User-Agent: MobilityGateway-2.0.34078
>   Server: MobilityGateway-2.0.34078
>   Accept: text/plain
>   Content-Length: 0
> 
> Best Regards,
> Jerry
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> rs
> http://www.freeswitch.org







From ben at langfeld.co.uk  Fri Feb 26 10:41:14 2010
From: ben at langfeld.co.uk (Ben Langfeld)
Date: Fri, 26 Feb 2010 18:41:14 +0000
Subject: [Freeswitch-users] Freeswitch SPA3000 HUP Not Sent
Message-ID: <fb4ab1e41002261041v381dd63ejafac36712f33cf96@mail.gmail.com>

Hey,

I have a small 7 seat mostly softphone based PBX installation in place using
freeswitch and a couple of SPA3000s for PSTN termination. Currently, aside
from PSTN HUP detection, outbound calls are fine setup to directly dial the
SPAs. When the internal VoIP side hangs up, freeswitch sends NORMAL_CLEARING
to the ATA and the ATA drops the call. Lovely.

Unfortunately inbound calls aren't so successful. Calls are sent using the
SPA dialplan to an internal freeswitch extension, and the calls connect
fine. This time, when the VoIP side hangs up, no NORMAL_CLEARING is sent to
the ATA. Is there a reason for this?

Can anyone give me any idea how I can get freeswitch to instruct the SPA to
drop the line?

Regards,
Ben Langfeld

Wave > Email
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100226/4ca31ca4/attachment-0002.html 

From anthony.minessale at gmail.com  Fri Feb 26 12:01:23 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 26 Feb 2010 14:01:23 -0600
Subject: [Freeswitch-users] Freeswitch SPA3000 HUP Not Sent
In-Reply-To: <fb4ab1e41002261041v381dd63ejafac36712f33cf96@mail.gmail.com>
References: <fb4ab1e41002261041v381dd63ejafac36712f33cf96@mail.gmail.com>
Message-ID: <191c3a031002261201s40a7a53cyfbe30529b77918eb@mail.gmail.com>

you would have to be more specific.
We would need to have you test this on latest SVN trunk with a full debug
log.

enter the following at your cli and reproduce the call saving all the
output.

console loglevel debug
sofia profile internal siptrace on



On Fri, Feb 26, 2010 at 12:41 PM, Ben Langfeld <ben at langfeld.co.uk> wrote:

> Hey,
>
> I have a small 7 seat mostly softphone based PBX installation in place
> using freeswitch and a couple of SPA3000s for PSTN termination. Currently,
> aside from PSTN HUP detection, outbound calls are fine setup to directly
> dial the SPAs. When the internal VoIP side hangs up, freeswitch sends
> NORMAL_CLEARING to the ATA and the ATA drops the call. Lovely.
>
> Unfortunately inbound calls aren't so successful. Calls are sent using the
> SPA dialplan to an internal freeswitch extension, and the calls connect
> fine. This time, when the VoIP side hangs up, no NORMAL_CLEARING is sent to
> the ATA. Is there a reason for this?
>
> Can anyone give me any idea how I can get freeswitch to instruct the SPA to
> drop the line?
>
> Regards,
> Ben Langfeld
>
> Wave > Email
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100226/a5da23c2/attachment-0002.html 

From robert.hadley at teotech.com  Fri Feb 26 12:36:47 2010
From: robert.hadley at teotech.com (Robert Hadley)
Date: Fri, 26 Feb 2010 12:36:47 -0800
Subject: [Freeswitch-users] Cannot make calls through PRI via
	wanpiperunning as daemon
In-Reply-To: <87f2f3b91002261128n17ea78acx160fcfdb04532004@mail.gmail.com>
References: <6FCED74CF54F466AA25CBDA7DFF42198@greyhawk.tonecommander.com>
	<87f2f3b91002261128n17ea78acx160fcfdb04532004@mail.gmail.com>
Message-ID: <D98570990911498788224B48CF455501@greyhawk.tonecommander.com>

Hi Mike,

 

Thanks for your help.

 

Manually loading mod_openzap fails.  It looks like there is something wrong
with my udev rules not changing permission of the /dev/wanpipe files.

 

freeswitch at internal> load mod_openzap

-ERR [module load file routine returned an error]

 

2010-02-26 12:24:26.807413 [DEBUG] zap_config.c:56 Configuration file is
/opt/teoswitch/conf/modules.conf.

freeswitch at internal> 2010-02-26 12:24:26.807413 [NOTICE] zap_io.c:2778
Modules configured: 1

2010-02-26 12:24:26.807413 [DEBUG] zap_config.c:56 Configuration file is
/opt/teoswitch/conf/openzap.conf.

2010-02-26 12:24:26.807413 [DEBUG] zap_io.c:2362 found config for span

2010-02-26 12:24:26.807413 [INFO] zap_io.c:2579 Loading IO from
/opt/teoswitch/mod/ozmod_wanpipe.so [wanpipe]

2010-02-26 12:24:26.807413 [DEBUG] zap_config.c:56 Configuration file is
/opt/teoswitch/conf/wanpipe.conf.

2010-02-26 12:24:26.807413 [INFO] zap_io.c:2379 auto-loaded 'wanpipe'

2010-02-26 12:24:26.807413 [DEBUG] zap_io.c:2400 created span 1 (smg_prid)
of type wanpipe

2010-02-26 12:24:26.807413 [DEBUG] zap_io.c:2413 span 1 [name]=[smg_prid]

2010-02-26 12:24:26.807413 [DEBUG] zap_io.c:2413 span 1 [trunk_type]=[t1]

2010-02-26 12:24:26.817864 [DEBUG] zap_io.c:2417 setting trunk type to 'T1'

2010-02-26 12:24:26.817864 [DEBUG] zap_io.c:2413 span 1 [b-channel]=[1:1-23]

2010-02-26 12:24:26.817864 [ERR] ozmod_wanpipe.c:233 Failed to open wanpipe
device span 1 channel 1

2010-02-26 12:24:26.817864 [ERR] ozmod_wanpipe.c:233 Failed to open wanpipe
device span 1 channel 2

2010-02-26 12:24:26.817864 [ERR] ozmod_wanpipe.c:233 Failed to open wanpipe
device span 1 channel 3

2010-02-26 12:24:26.817864 [ERR] ozmod_wanpipe.c:233 Failed to open wanpipe
device span 1 channel 4

2010-02-26 12:24:26.817864 [ERR] ozmod_wanpipe.c:233 Failed to open wanpipe
device span 1 channel 5

2010-02-26 12:24:26.817864 [ERR] ozmod_wanpipe.c:233 Failed to open wanpipe
device span 1 channel 6

2010-02-26 12:24:26.817864 [ERR] ozmod_wanpipe.c:233 Failed to open wanpipe
device span 1 channel 7

 

 

Here is what /dev/wan* looks like after udev changes:

[root at roberth-c53 bin]# ls -l /dev/wan*

crw------- 1 root root 242,    0 Feb 26 11:24 /dev/wanec

crw------- 1 root root 241, 2080 Feb 26 11:24 /dev/wanpipe

crw------- 1 root root 241,    1 Feb 26 11:24 /dev/wanpipe1_if1

crw------- 1 root root 241,   10 Feb 26 11:24 /dev/wanpipe1_if10

crw------- 1 root root 241,   11 Feb 26 11:24 /dev/wanpipe1_if11

crw------- 1 root root 241,   12 Feb 26 11:24 /dev/wanpipe1_if12

 

CUT A FEW LINES

crw------- 1 root root 241,   34 Feb 26 11:24 /dev/wanpipe2_if2

crw------- 1 root root 241,   35 Feb 26 11:24 /dev/wanpipe2_if3

crw------- 1 root root 241,   36 Feb 26 11:24 /dev/wanpipe2_if4

crw------- 1 root root 241, 2112 Feb 26 11:24 /dev/wanpipe_ctrl

crw------- 1 root root 241, 2144 Feb 26 11:24 /dev/wanpipe_logger

crw------- 1 root root 241, 2368 Feb 26 11:24 /dev/wanpipe_timer0

 

 

I made and installed the wanpipe driver as root, is that part of the
problem?

 

Thanks again,

Robert

 

 

  _____  

From: Michael Collins [mailto:msc at freeswitch.org] 
Sent: Friday, February 26, 2010 11:28 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Cannot make calls through PRI via
wanpiperunning as daemon

 

 

On Thu, Feb 25, 2010 at 5:16 PM, Robert Hadley <robert.hadley at teotech.com>
wrote:

When running Freeswitch as service called teoswitch as user teoswitch I
cannot make calls through the Sangoma PRI or analog cards using wanpipe
driver.  I have added a file called 30-wanpipe.rules to /etc/udev/rules.d
and rebooted the server.

 

cat 30-wanpipe.rules

# /etc/udev/rules.d/30-wanpipe.rules

SUBSYSTEM=="wptdm", OWNER="teoswitch", GROUP="teoswitch", MODE="0660"

SUBSYSTEM=="wanpipe", OWNER="teoswitch", GROUP="teoswitch", MODE="0660"

I also tried adding:

SUBSYSTEM=="wanec", OWNER="teoswitch", GROUP="teoswitch", MODE="0660"

 

Freeswitch log:

Dialplan: sofia/internal/5410 at 192.168.72.45:5060 parsing
[default->SangomaPRI] continue=false

Dialplan: sofia/internal/5410 at 192.168.72.45:5060 Regex (PASS) [SangomaPRI]
destination_number(93491045) =~ /^9(\d+)$/ break=on-false

Dialplan: sofia/internal/5410 at 192.168.72.45:5060 Action
set(effective_caller_id_number=425740${caller_id_number})

Dialplan: sofia/internal/5410 at 192.168.72.45:5060 Action
bridge(openzap/smg_prid/a/3491045 at g1)

2010-02-25 16:51:11.328635 [DEBUG] switch_core_state_machine.c:122
(sofia/internal/5410 at 192.168.72.45:5060) State Change CS_ROUTING ->
CS_EXECUTE

2010-02-25 16:51:11.328635 [DEBUG] switch_core_session.c:1019 Send signal
sofia/internal/5410 at 192.168.72.45:5060 [BREAK]

2010-02-25 16:51:11.328635 [DEBUG] switch_core_state_machine.c:341
(sofia/internal/5410 at 192.168.72.45:5060) State ROUTING going to sleep

2010-02-25 16:51:11.328635 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/5410 at 192.168.72.45:5060) Running State Change CS_EXECUTE

2010-02-25 16:51:11.328635 [DEBUG] switch_core_state_machine.c:348
(sofia/internal/5410 at 192.168.72.45:5060) State EXECUTE

2010-02-25 16:51:11.328635 [DEBUG] mod_sofia.c:181
sofia/internal/5410 at 192.168.72.45:5060 SOFIA EXECUTE

2010-02-25 16:51:11.328635 [DEBUG] switch_core_state_machine.c:159
sofia/internal/5410 at 192.168.72.45:5060 Standard EXECUTE

EXECUTE sofia/internal/5410 at 192.168.72.45:5060
set(effective_caller_id_number=4257405410)

2010-02-25 16:51:11.328635 [DEBUG] mod_dptools.c:811
sofia/internal/5410 at 192.168.72.45:5060 SET
[effective_caller_id_number]=[4257405410]

EXECUTE sofia/internal/5410 at 192.168.72.45:5060
bridge(openzap/smg_prid/a/3491045 at g1)

2010-02-25 16:51:11.339637 [ERR] switch_core_session.c:357 Could not locate
channel type openzap

2010-02-25 16:51:11.339637 [ERR] switch_ivr_originate.c:2411 Cannot create
outgoing channel of type [openzap] cause: [CHAN_NOT_IMPLEMENTED]

2010-02-25 16:51:11.339637 [DEBUG] switch_ivr_originate.c:3209 Originate
Resulted in Error Cause: 66 [CHAN_NOT_IMPLEMENTED]

2010-02-25 16:51:11.339637 [INFO] mod_dptools.c:2346 Originate Failed.
Cause: CHAN_NOT_IMPLEMENTED


CHAN_NOT_IMPLEMENTED implies that OpenZAP did not load. Capture the output
of "load mod_openzap" and look for the reason that it is failing to load.
-MC

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100226/aba32892/attachment-0002.html 

From msc at freeswitch.org  Fri Feb 26 12:45:22 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 26 Feb 2010 12:45:22 -0800
Subject: [Freeswitch-users] Phrases - Can't find macro
In-Reply-To: <28f27f5d1002260910h7b3ce773l28294e7354078b05@mail.gmail.com>
References: <28f27f5d1002260910h7b3ce773l28294e7354078b05@mail.gmail.com>
Message-ID: <87f2f3b91002261245l176c49f6j769bd673be0324fe@mail.gmail.com>

You might try this suggestion:
Create a new file for your custom macros:
/conf/lang/en/demo/custom-phrases.xml
<include>
  <macro name="confwelcome">
    <input pattern="(.*)">
      <action function="play-file" data="vm/vm-youhave.wav"/>
    </input>
  </macro>
</include>

Now you have a single place to put all of your custom macros. Be sure to
reloadxml!
-MC

On Fri, Feb 26, 2010 at 9:10 AM, Maciej Bylica <mbsip at gazeta.pl> wrote:

> Hello,
>
> I am playing around with Phrases to use them with conference application.
> But i've encountered rudimentary problem of how to use newly added macro.
>
> What I already did is (according to wiki Speech Phrase Management)
> http://wiki.freeswitch.org/wiki/Speech_Phrase_Management
> - confirmed that there is "mod_say_en" loaded (<load module="mod_say_en"/>)
> - confirmed that there are proper .wav files
> - modified onf/lang/en/en.xml file:
>  <section name="phrases" description="Speech Phrase Management">
>   <macros>
>     <language name="en" sound_path="$${sounds_dir}/en/us/callie"
> tts_engine="cepstral" tts_voice="callie">
>       <macro name="confwelcome">
>         <input pattern="(.*)">
>           <match>
>              <action function="play-file" data="vm-youhave.wav"/>
> <!-- just to test the phrase-->
>           </match>
>     </language>
>   </macros>
>  </section>
> - modified a part of dialplan:
>    <extension name="conference_48112223344">
>      <condition field="destination_number" expression="^(\d{11})$">
>         <action application="phrase" data="confwelcome"/>
>      </condition>
>     </extension>
>
> I have following outcome:
> 2010-02-26 19:57:09.487245 [DEBUG] switch_core_session.c:638 Send
> signal sofia/internal/1000 at 217.153.192.36 [BREAK]
> EXECUTE sofia/internal/1000 at 217.153.192.36 phrase(confwelcome)
> 2010-02-26 19:57:09.487245 [DEBUG] mod_dptools.c:1850 Execute confwelcome()
> lang
> 2010-02-26 19:57:09.487245 [DEBUG] switch_ivr_play_say.c:118 No
> language specified - Using [en]
> 2010-02-26 19:57:09.496322 [ERR] switch_ivr_play_say.c:202 Can't find
> macro confwelcome.
> 2010-02-26 19:57:09.496322 [WARNING] switch_ivr_play_say.c:368 Macro
> [confwelcome] did not match any patterns
>
>
>
> Strange is that if i use a wiki example, it works. To be more precise:
> - conf/lang/en/en.xml file was overwritten with an example macros
> (directly from aforementioned wiki).
> - dialplan was modified:
>    <extension name="conference_48112223344">
>      <condition field="destination_number" expression="^(\d{11})$">
>         <action application="phrase" data="msgcount"/>
>      </condition>
>     </extension>
>
>
> Am i doing something wrong?
>
> Thx,
> Maciej.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100226/0143ba00/attachment-0002.html 

From anthony.minessale at gmail.com  Fri Feb 26 12:55:32 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 26 Feb 2010 14:55:32 -0600
Subject: [Freeswitch-users] 415 Unsupported Media Handling
In-Reply-To: <ADCCADE853D6427CBC7CD978E7E0A0ED@greyhawk.tonecommander.com>
References: <AE66B63BAF424BBCBE585E244E0C42B4@greyhawk.tonecommander.com>
	<835A36B2-1C04-42B0-86C1-36528E93589D@freeswitch.org>
	<ADCCADE853D6427CBC7CD978E7E0A0ED@greyhawk.tonecommander.com>
Message-ID: <191c3a031002261255x7e037e6ftc119eab0ed21b69b@mail.gmail.com>

It says SHOULD, not MUST right?

The message passing in FS is abstracted and protocol agnostic and we are a
b2bua not a proxy in terms of SIP.
You are sending a message to 1 UA on FS who is accepting the message and
delivering it to the core who is happy to receive messages in any format.
Then it's routed back out another sip dialog where it's rejected.  It's too
late to go tell the sender that is unacceptable because we already happily
accepted it (messages are not always passed out to other phones they can
easily be directed at other internal resources).  We don't know what the
content-type means as we are a neutral party in the whole thing so there is
not much else we can do but violate this scope issue and break out of our
role as a neutral party and translate it to plain text and try again which
is not very elegant.

If FS was a proxy software, like openser and friends, we would be passing
the data between UA in the way you expect but we are not a proxy.


Based on the frequency and specific nature of your constant inquiries, and
the likelihood that you are offering commercial VoIP services to customers.
I suggest you contact us at consulting at freeswitch.org to investigate
commercial support options for FreeSWITCH.  Even then, I am not sure I could
help you besides maybe a param to convert all text/html messages to plain
text or some other sad hack.




On Fri, Feb 26, 2010 at 1:43 PM, Jerry Richards
<jerry.richards at teotech.com>wrote:

> I'm not sure I follow your comment.  The first device prefers text/html so
> that's what it normally sets in the initial MESSAGE.  Devices that support
> text/html will not generate this 415 error reply.  It's only devices that
> don't support it that would send the 415 reply, so the issue is that the
> 415
> is not getting passed back to the originator.
>
> Best Regards,
> Jerry
>
>
> -----Original Message-----
> From: Brian West [mailto:brian at freeswitch.org]
> Sent: Friday, February 26, 2010 11:01 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] 415 Unsupported Media Handling
>
> Its really clear here you'll need to say text/plain in the content type
> their accept header says they only take text/plain.
>
> /b
>
> On Feb 26, 2010, at 12:57 PM, Jerry Richards wrote:
>
> >
> > I have two types of devices, one supports text/html MESSAGE content
> > and one that only supports text/plain MESSAGE content.  When I send an
> > IM from the first to the second, the second replies with 415
> > Unsupported Media Type (as shown below).  Section 8.1.3.5 of RFC 3261
> > says the sender should retry using the media type acceptable to the
> receiver (in this case: plain/text).
> >
> > The problem I have is that Freeswitch doesn't pass the error back to
> > the sender (nor does it retry itself using plain/text).  So the IM is
> lost.
> > Does anyone see the reason why the error is not being handled correctly?
> >
> >
> > ----------------------------------------------------------------------
> > -- send 668 bytes to udp/[192.168.72.141]:5062 at 18:43:24.720446:
> >
> ------------------------------------------------------------------------
> >   MESSAGE sip:5398 at 192.168.72.141:5062 SIP/2.0
> >   Via: SIP/2.0/UDP 192.168.72.141;rport;branch=z9hG4bKXg6SNUcQa5Kre
> >   Max-Forwards: 70
> >   From: "5382 on 141" <sip:5382 at 192.168.72.141<sip%3A5382 at 192.168.72.141>
> >;tag=66661130
> >   To: "5398" <sip:5398 at 192.168.72.141:5062>
> >   Call-ID: a9891a15-9da9-122d-b3ad-003048d7e9f0
> >   CSeq: 127444135 MESSAGE
> >   Contact: <sip:mod_sofia at 192.168.72.141:5060>
> >   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-32M
> >   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
> >   Supported: timer, precondition, path, replaces
> >   Content-Type: text/html
> >   Content-Length: 63
> >
> >   <font size="2" face="Arial">hello this is Jerry from Teo</font>
> >
> > ----------------------------------------------------------------------
> > -- recv 459 bytes from udp/[192.168.72.141]:5062 at 18:43:24.728390:
> >
> ------------------------------------------------------------------------
> >   SIP/2.0 415 Unsupported media type
> >   Via: SIP/2.0/UDP
> > 192.168.72.141;rport=5060;branch=z9hG4bKXg6SNUcQa5Kre;received=192.168
> > .72.14
> > 1
> >   From: "5382 on 141" <sip:5382 at 192.168.72.141<sip%3A5382 at 192.168.72.141>
> >;tag=66661130
> >   To: "5398" <sip:5398 at 192.168.72.141:5062>
> >   Call-ID: a9891a15-9da9-122d-b3ad-003048d7e9f0
> >   CSeq: 127444135 MESSAGE
> >   Date: Fri, 26 Feb 2010 18:43:24 GMT
> >   User-Agent: MobilityGateway-2.0.34078
> >   Server: MobilityGateway-2.0.34078
> >   Accept: text/plain
> >   Content-Length: 0
> >
> > Best Regards,
> > Jerry
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> > rs
> > http://www.freeswitch.org
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100226/9ae01931/attachment-0002.html 

From anthony.minessale at gmail.com  Fri Feb 26 13:01:20 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 26 Feb 2010 15:01:20 -0600
Subject: [Freeswitch-users] Cannot make calls through PRI via
	wanpiperunning as daemon
In-Reply-To: <D98570990911498788224B48CF455501@greyhawk.tonecommander.com>
References: <6FCED74CF54F466AA25CBDA7DFF42198@greyhawk.tonecommander.com>
	<87f2f3b91002261128n17ea78acx160fcfdb04532004@mail.gmail.com>
	<D98570990911498788224B48CF455501@greyhawk.tonecommander.com>
Message-ID: <191c3a031002261301m53b47e84hbcf87dbfb29a0be9@mail.gmail.com>

yes you will need to give your user access to a group who can read and write
/dev


On Fri, Feb 26, 2010 at 2:36 PM, Robert Hadley <robert.hadley at teotech.com>wrote:

>  Hi Mike,
>
>
>
> Thanks for your help.
>
>
>
> Manually loading mod_openzap fails.  It looks like there is something wrong
> with my udev rules not changing permission of the /dev/wanpipe files.
>
>
>
> freeswitch at internal> load mod_openzap
>
> -ERR [module load file routine returned an error]
>
>
>
> 2010-02-26 12:24:26.807413 [DEBUG] zap_config.c:56 Configuration file is
> /opt/teoswitch/conf/modules.conf.
>
> freeswitch at internal> 2010-02-26 12:24:26.807413 [NOTICE] zap_io.c:2778
> Modules configured: 1
>
> 2010-02-26 12:24:26.807413 [DEBUG] zap_config.c:56 Configuration file is
> /opt/teoswitch/conf/openzap.conf.
>
> 2010-02-26 12:24:26.807413 [DEBUG] zap_io.c:2362 found config for span
>
> 2010-02-26 12:24:26.807413 [INFO] zap_io.c:2579 Loading IO from
> /opt/teoswitch/mod/ozmod_wanpipe.so [wanpipe]
>
> 2010-02-26 12:24:26.807413 [DEBUG] zap_config.c:56 Configuration file is
> /opt/teoswitch/conf/wanpipe.conf.
>
> 2010-02-26 12:24:26.807413 [INFO] zap_io.c:2379 auto-loaded 'wanpipe'
>
> 2010-02-26 12:24:26.807413 [DEBUG] zap_io.c:2400 created span 1 (smg_prid)
> of type wanpipe
>
> 2010-02-26 12:24:26.807413 [DEBUG] zap_io.c:2413 span 1 [name]=[smg_prid]
>
> 2010-02-26 12:24:26.807413 [DEBUG] zap_io.c:2413 span 1 [trunk_type]=[t1]
>
> 2010-02-26 12:24:26.817864 [DEBUG] zap_io.c:2417 setting trunk type to 'T1'
>
> 2010-02-26 12:24:26.817864 [DEBUG] zap_io.c:2413 span 1
> [b-channel]=[1:1-23]
>
> 2010-02-26 12:24:26.817864 [ERR] ozmod_wanpipe.c:233 Failed to open wanpipe
> device span 1 channel 1
>
> 2010-02-26 12:24:26.817864 [ERR] ozmod_wanpipe.c:233 Failed to open wanpipe
> device span 1 channel 2
>
> 2010-02-26 12:24:26.817864 [ERR] ozmod_wanpipe.c:233 Failed to open wanpipe
> device span 1 channel 3
>
> 2010-02-26 12:24:26.817864 [ERR] ozmod_wanpipe.c:233 Failed to open wanpipe
> device span 1 channel 4
>
> 2010-02-26 12:24:26.817864 [ERR] ozmod_wanpipe.c:233 Failed to open wanpipe
> device span 1 channel 5
>
> 2010-02-26 12:24:26.817864 [ERR] ozmod_wanpipe.c:233 Failed to open wanpipe
> device span 1 channel 6
>
> 2010-02-26 12:24:26.817864 [ERR] ozmod_wanpipe.c:233 Failed to open wanpipe
> device span 1 channel 7
>
>
>
>
>
> Here is what /dev/wan* looks like after udev changes:
>
> [root at roberth-c53 bin]# ls -l /dev/wan*
>
> crw------- 1 root root 242,    0 Feb 26 11:24 /dev/wanec
>
> crw------- 1 root root 241, 2080 Feb 26 11:24 /dev/wanpipe
>
> crw------- 1 root root 241,    1 Feb 26 11:24 /dev/wanpipe1_if1
>
> crw------- 1 root root 241,   10 Feb 26 11:24 /dev/wanpipe1_if10
>
> crw------- 1 root root 241,   11 Feb 26 11:24 /dev/wanpipe1_if11
>
> crw------- 1 root root 241,   12 Feb 26 11:24 /dev/wanpipe1_if12
>
>
>
> CUT A FEW LINES
>
> crw------- 1 root root 241,   34 Feb 26 11:24 /dev/wanpipe2_if2
>
> crw------- 1 root root 241,   35 Feb 26 11:24 /dev/wanpipe2_if3
>
> crw------- 1 root root 241,   36 Feb 26 11:24 /dev/wanpipe2_if4
>
> crw------- 1 root root 241, 2112 Feb 26 11:24 /dev/wanpipe_ctrl
>
> crw------- 1 root root 241, 2144 Feb 26 11:24 /dev/wanpipe_logger
>
> crw------- 1 root root 241, 2368 Feb 26 11:24 /dev/wanpipe_timer0
>
>
>
>
>
> I made and installed the wanpipe driver as root, is that part of the
> problem?
>
>
>
> Thanks again,
>
> Robert
>
>
>
>
>  ------------------------------
>
> *From:* Michael Collins [mailto:msc at freeswitch.org]
> *Sent:* Friday, February 26, 2010 11:28 AM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Cannot make calls through PRI via
> wanpiperunning as daemon
>
>
>
>
>
> On Thu, Feb 25, 2010 at 5:16 PM, Robert Hadley <robert.hadley at teotech.com>
> wrote:
>
> When running Freeswitch as service called teoswitch as user teoswitch I
> cannot make calls through the Sangoma PRI or analog cards using wanpipe
> driver.  I have added a file called 30-wanpipe.rules to /etc/udev/rules.d
> and rebooted the server.
>
>
>
> cat 30-wanpipe.rules
>
> # /etc/udev/rules.d/30-wanpipe.rules
>
> SUBSYSTEM=="wptdm", OWNER="teoswitch", GROUP="teoswitch", MODE="0660"
>
> SUBSYSTEM=="wanpipe", OWNER="teoswitch", GROUP="teoswitch", MODE="0660"
>
> I also tried adding:
>
> SUBSYSTEM=="wanec", OWNER="teoswitch", GROUP="teoswitch", MODE="0660"
>
>
>
> Freeswitch log:
>
> Dialplan: sofia/internal/5410 at 192.168.72.45:5060 parsing
> [default->SangomaPRI] continue=false
>
> Dialplan: sofia/internal/5410 at 192.168.72.45:5060 Regex (PASS) [SangomaPRI]
> destination_number(93491045) =~ /^9(\d+)$/ break=on-false
>
> Dialplan: sofia/internal/5410 at 192.168.72.45:5060 Action
> set(effective_caller_id_number=425740${caller_id_number})
>
> Dialplan: sofia/internal/5410 at 192.168.72.45:5060 Action
> bridge(openzap/smg_prid/a/3491045 at g1)
>
> 2010-02-25 16:51:11.328635 [DEBUG] switch_core_state_machine.c:122
> (sofia/internal/5410 at 192.168.72.45:5060) State Change CS_ROUTING ->
> CS_EXECUTE
>
> 2010-02-25 16:51:11.328635 [DEBUG] switch_core_session.c:1019 Send signal
> sofia/internal/5410 at 192.168.72.45:5060 [BREAK]
>
> 2010-02-25 16:51:11.328635 [DEBUG] switch_core_state_machine.c:341
> (sofia/internal/5410 at 192.168.72.45:5060) State ROUTING going to sleep
>
> 2010-02-25 16:51:11.328635 [DEBUG] switch_core_state_machine.c:314
> (sofia/internal/5410 at 192.168.72.45:5060) Running State Change CS_EXECUTE
>
> 2010-02-25 16:51:11.328635 [DEBUG] switch_core_state_machine.c:348
> (sofia/internal/5410 at 192.168.72.45:5060) State EXECUTE
>
> 2010-02-25 16:51:11.328635 [DEBUG] mod_sofia.c:181 sofia/internal/
> 5410 at 192.168.72.45:5060 SOFIA EXECUTE
>
> 2010-02-25 16:51:11.328635 [DEBUG] switch_core_state_machine.c:159
> sofia/internal/5410 at 192.168.72.45:5060 Standard EXECUTE
>
> EXECUTE sofia/internal/5410 at 192.168.72.45:5060set(effective_caller_id_number=4257405410)
>
> 2010-02-25 16:51:11.328635 [DEBUG] mod_dptools.c:811 sofia/internal/
> 5410 at 192.168.72.45:5060 SET [effective_caller_id_number]=[4257405410]
>
> EXECUTE sofia/internal/5410 at 192.168.72.45:5060bridge(openzap/smg_prid/a/3491045 at g1
> )
>
> 2010-02-25 16:51:11.339637 [ERR] switch_core_session.c:357 Could not locate
> channel type openzap
>
> 2010-02-25 16:51:11.339637 [ERR] switch_ivr_originate.c:2411 Cannot create
> outgoing channel of type [openzap] cause: [CHAN_NOT_IMPLEMENTED]
>
> 2010-02-25 16:51:11.339637 [DEBUG] switch_ivr_originate.c:3209 Originate
> Resulted in Error Cause: 66 [CHAN_NOT_IMPLEMENTED]
>
> 2010-02-25 16:51:11.339637 [INFO] mod_dptools.c:2346 Originate Failed.
> Cause: CHAN_NOT_IMPLEMENTED
>
>
> CHAN_NOT_IMPLEMENTED implies that OpenZAP did not load. Capture the output
> of "load mod_openzap" and look for the reason that it is failing to load.
> -MC
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100226/8a286def/attachment-0002.html 

From dave at 3c.co.uk  Fri Feb 26 13:59:12 2010
From: dave at 3c.co.uk (David Knell)
Date: Fri, 26 Feb 2010 14:59:12 -0700
Subject: [Freeswitch-users] ASR Apps
References: <AC3A71D6-4EAB-4261-B14A-2869C0C05C91@freeswitch.org><9dc4a1671002251855y586218f6h2c281b4a5acb664e@mail.gmail.com>
	<16DBCD58-A962-4121-9899-F2BB56F13554@freeswitch.org>
Message-ID: <8AE0BC1F0D764475ADF09B938F15E432@DELL9>

Hi Brian,

Here's a starting point for someone wanting to do voice dial from a Google address book:
http://www.softivr.com/wiki/index.php/Voice_dial

Cheers --

Dave
  ----- Original Message ----- 
  From: Brian West 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Thursday, February 25, 2010 8:02 PM
  Subject: Re: [Freeswitch-users] ASR Apps


  I'm looking for someone to build some really nice apps like dial by name speech apps or other such apps or frameworks using ASR and possibly lua or js.  Anyone wanna do something.


  /b


  On Feb 25, 2010, at 8:55 PM, EdPimentl wrote:


    Hello Bryon,

    We looking to create a Twilio like service using FreeSwitch.

    Sincerely,
    -E
    http://vCardCloud.com
    GV:  678.685.9858
    EdPimentl: Skype




------------------------------------------------------------------------------


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100226/74eb7da5/attachment-0002.html 

From brian at microcomaustralia.com.au  Fri Feb 26 16:23:05 2010
From: brian at microcomaustralia.com.au (Brian May)
Date: Sat, 27 Feb 2010 11:23:05 +1100
Subject: [Freeswitch-users] end call detect on FXO port
Message-ID: <3c5cf5261002261623tca4e78em6800859844465d35@mail.gmail.com>

Hello,

I need to get Freeswitch to detect when the caller has hang up, so it
will hang up the ilne also. Especially important for when the caller
has left a message, although ideally it should work for all calls.

With Asterisk this required the use of automatically detecting the
busy signal, at the driver level. How can I do something similar with
Freeswitch?

Some web pages suggest I use this:

<action application="tone_detect" data="busy 425 r 0 hangup
'normal_clearing' 3"/>

Unfortunately this has the side affect that it answers the call, I
don't want to change the answer behaviour,
only the hang up behaviour.

2010-02-27 14:27:52.655081 [DEBUG] switch_core_session.c:1728
Application tone_detect Requires media! pre_answering channel
OpenZAP/3:1/0397551926

I also see this message, but so far no solution:

http://lists.freeswitch.org/pipermail/freeswitch-users/2008-July/004567.html

How do I do this?

Thanks
-- 
Brian May <brian at microcomaustralia.com.au>



From brian at freeswitch.org  Fri Feb 26 16:30:53 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 26 Feb 2010 18:30:53 -0600
Subject: [Freeswitch-users] end call detect on FXO port
In-Reply-To: <3c5cf5261002261623tca4e78em6800859844465d35@mail.gmail.com>
References: <3c5cf5261002261623tca4e78em6800859844465d35@mail.gmail.com>
Message-ID: <11E1DA2C-0A14-4421-8E6A-994346BE941D@freeswitch.org>

What are you using for your PSTN interface?

/b

On Feb 26, 2010, at 6:23 PM, Brian May wrote:

> Hello,
> 
> I need to get Freeswitch to detect when the caller has hang up, so it
> will hang up the ilne also. Especially important for when the caller
> has left a message, although ideally it should work for all calls.
> 
> With Asterisk this required the use of automatically detecting the
> busy signal, at the driver level. How can I do something similar with
> Freeswitch?
> 
> Some web pages suggest I use this:
> 
> <action application="tone_detect" data="busy 425 r 0 hangup
> 'normal_clearing' 3"/>
> 
> Unfortunately this has the side affect that it answers the call, I
> don't want to change the answer behaviour,
> only the hang up behaviour.
> 
> 2010-02-27 14:27:52.655081 [DEBUG] switch_core_session.c:1728
> Application tone_detect Requires media! pre_answering channel
> OpenZAP/3:1/0397551926
> 
> I also see this message, but so far no solution:
> 
> http://lists.freeswitch.org/pipermail/freeswitch-users/2008-July/004567.html
> 
> How do I do this?
> 
> Thanks
> -- 
> Brian May <brian at microcomaustralia.com.au>
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From brian at microcomaustralia.com.au  Fri Feb 26 18:00:43 2010
From: brian at microcomaustralia.com.au (Brian May)
Date: Sat, 27 Feb 2010 13:00:43 +1100
Subject: [Freeswitch-users] end call detect on FXO port
In-Reply-To: <11E1DA2C-0A14-4421-8E6A-994346BE941D@freeswitch.org>
References: <3c5cf5261002261623tca4e78em6800859844465d35@mail.gmail.com>
	<11E1DA2C-0A14-4421-8E6A-994346BE941D@freeswitch.org>
Message-ID: <3c5cf5261002261800j21248353u46e5279a12ed7b65@mail.gmail.com>

On 27 February 2010 11:30, Brian West <brian at freeswitch.org> wrote:
> What are you using for your PSTN interface?

Not quite sure if this is what you are asking; however I am using a TDM400 card.
-- 
Brian May <brian at microcomaustralia.com.au>



From brian at freeswitch.org  Fri Feb 26 18:09:42 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 26 Feb 2010 20:09:42 -0600
Subject: [Freeswitch-users] end call detect on FXO port
In-Reply-To: <3c5cf5261002261800j21248353u46e5279a12ed7b65@mail.gmail.com>
References: <3c5cf5261002261623tca4e78em6800859844465d35@mail.gmail.com>
	<11E1DA2C-0A14-4421-8E6A-994346BE941D@freeswitch.org>
	<3c5cf5261002261800j21248353u46e5279a12ed7b65@mail.gmail.com>
Message-ID: <DA13207F-7196-4177-8C6E-D2CECA5D3E83@freeswitch.org>

I think the hangup detection should work exactly the same then.

/b

On Feb 26, 2010, at 8:00 PM, Brian May wrote:

> Not quite sure if this is what you are asking; however I am using a TDM400 card.




From brian at microcomaustralia.com.au  Fri Feb 26 18:24:24 2010
From: brian at microcomaustralia.com.au (Brian May)
Date: Sat, 27 Feb 2010 13:24:24 +1100
Subject: [Freeswitch-users] end call detect on FXO port
In-Reply-To: <DA13207F-7196-4177-8C6E-D2CECA5D3E83@freeswitch.org>
References: <3c5cf5261002261623tca4e78em6800859844465d35@mail.gmail.com>
	<11E1DA2C-0A14-4421-8E6A-994346BE941D@freeswitch.org>
	<3c5cf5261002261800j21248353u46e5279a12ed7b65@mail.gmail.com>
	<DA13207F-7196-4177-8C6E-D2CECA5D3E83@freeswitch.org>
Message-ID: <3c5cf5261002261824j68b9f74ckc7f5b5c9b7ae12bb@mail.gmail.com>

On 27 February 2010 13:09, Brian West <brian at freeswitch.org> wrote:
> I think the hangup detection should work exactly the same then.

Exactly the same as what?
-- 
Brian May <brian at microcomaustralia.com.au>



From lakindia89 at gmail.com  Fri Feb 26 20:57:07 2010
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Sat, 27 Feb 2010 10:27:07 +0530
Subject: [Freeswitch-users] smg_prid not bridging the call
In-Reply-To: <c4d05cbe1002260731v29dbc428i1e72d24a220c62eb@mail.gmail.com>
References: <7d79b3931002252155t47c86968q9e451482926d93ac@mail.gmail.com>
	<c4d05cbe1002260731v29dbc428i1e72d24a220c62eb@mail.gmail.com>
Message-ID: <7d79b3931002262057g3b325bctcdbcfb833d4aaf55@mail.gmail.com>

Dear Moy,
Here are the details:

FreeSwitch Log:
http://pastebin.freeswitch.org/12256

 /var/log/sangoma_pri/dchan_<span-no>.log:
 http://pastebin.freeswitch.org/12257

/var/log/sangoma_mgd.log:
http://pastebin.freeswitch.org/12258

smg_pri.conf
http://pastebin.freeswitch.org/12259


On Fri, Feb 26, 2010 at 9:01 PM, Moises Silva <moises.silva at gmail.com>wrote:

> Hello lakshmanan,
>
> Please enable the dchan_log=q931 in /etc/wanpipe/smg_pri.conf and then
> restart it (smg_ctrl restart), then pastebin the logs
>
>  /var/log/sangoma_pri/dchan_<span-no>.log
> /var/log/sangoma_mgd.log
>
> That will contain the Q931 details (if any). Also pastebin your
> smg_pri.conf.
>
> Also enable the FreeSWITCH debug logging (see the FreeSWITCH wiki for
> details about that) and paste them too.
>
> --
> Moises Silva
> Senior Software Engineer
> Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R
> 9T3 Canada
> t. 1 905 474 1990 x 128 | e. moy at sangoma.com
>
> On Fri, Feb 26, 2010 at 12:55 AM, lakshmanan ganapathy <
> lakindia89 at gmail.com> wrote:
>
>> Dear all,
>> I'm having a A102 Sangoma hardware. I configured it with freeswitch.
>> wanrouter status, says both the port as connected.
>> My smg_prid version is
>>
>> Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: ================System
>> restart=============
>> Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: = Sangoma PRI Protocol Stack
>> Daemon       =
>> Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: = Version:
>> 1.54                           =
>> Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: = Date: Feb 15
>> 2010                       =
>> Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: = Wanpipe Release:
>> wanpipe-3.5.8.6       =
>> Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: = Revision:Revision:
>> 15288                =
>> Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid:
>> ===========================================
>>
>> My freeswitch version is 16729.
>> I started freeswitch.
>>
>> oz list
>> +OK
>> span: 1 (smg_prid)
>> type: Sangoma (boost)
>> chan_count: 60
>> dialplan: XML
>> context: default
>> dial_regex:
>> fail_dial_regex:
>> hold_music:
>> analog_options none
>>
>> I originated a call as
>> originate openzap/smg_prid/a/9952248266 &park(), which hits my mobile.
>>
>> But when I issued the following command:
>> originate openzap/smg_prid/a/9952248266
>> &bridge(openzap/smg_prid/a/8122133885)
>> It rings my mobile (9952248266) first, but after that the following error
>> was displayed
>>
>> 2010-02-26 16:20:51.736080 [ERR] switch_ivr_originate.c:2387 Cannot create
>> outgoing channel of type [openzap] cause: [NORMAL_CIRCUIT_CONGESTION]
>> The call got ended in my mobile.
>>
>> Freeswitch log and smg_pri.conf
>>           http://pastebin.freeswitch.org/12248
>> openzap.conf:
>> [span wanpipe smg_prid]
>> name => smg_prid
>> trunk_type =>e1
>> b-channel => 1:1-15
>> b-channel => 1:17-31
>> trunk_type =>e1
>> b-channel => 2:1-15
>> b-channel => 2:17-31
>>
>> openzap.conf.xml:
>> <configuration name="openzap.conf" description="OpenZAP Configuration">
>>   <settings>
>>     <param name="debug" value="0"/>
>>   </settings>
>>   <boost_spans>
>>         <span id="smg_prid">
>>         <param name="dialplan" value="XML"/>
>>         <param name="context" value="default"/>
>>         </span>
>>   </boost_spans>
>> </configuration>
>>
>> Please guide me to setup this one!!.
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100227/9b76b747/attachment-0002.html 

From lakindia89 at gmail.com  Fri Feb 26 21:02:08 2010
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Sat, 27 Feb 2010 10:32:08 +0530
Subject: [Freeswitch-users] smg_prid not bridging the call
In-Reply-To: <7d79b3931002262057g3b325bctcdbcfb833d4aaf55@mail.gmail.com>
References: <7d79b3931002252155t47c86968q9e451482926d93ac@mail.gmail.com>
	<c4d05cbe1002260731v29dbc428i1e72d24a220c62eb@mail.gmail.com>
	<7d79b3931002262057g3b325bctcdbcfb833d4aaf55@mail.gmail.com>
Message-ID: <7d79b3931002262102w79e16ae1nd786600b44dfb758@mail.gmail.com>

In the Dchan log it is saying Invalid Information Elements. That might be a
problem??? But I even don't know why it is saying Invalid Information
Element??
Please guide me!!!


On Sat, Feb 27, 2010 at 10:27 AM, lakshmanan ganapathy <lakindia89 at gmail.com
> wrote:

> Dear Moy,
> Here are the details:
>
> FreeSwitch Log:
> http://pastebin.freeswitch.org/12256
>
>  /var/log/sangoma_pri/dchan_<span-no>.log:
>  http://pastebin.freeswitch.org/12257
>
> /var/log/sangoma_mgd.log:
> http://pastebin.freeswitch.org/12258
>
> smg_pri.conf
> http://pastebin.freeswitch.org/12259
>
>
>
> On Fri, Feb 26, 2010 at 9:01 PM, Moises Silva <moises.silva at gmail.com>wrote:
>
>> Hello lakshmanan,
>>
>> Please enable the dchan_log=q931 in /etc/wanpipe/smg_pri.conf and then
>> restart it (smg_ctrl restart), then pastebin the logs
>>
>>  /var/log/sangoma_pri/dchan_<span-no>.log
>> /var/log/sangoma_mgd.log
>>
>> That will contain the Q931 details (if any). Also pastebin your
>> smg_pri.conf.
>>
>> Also enable the FreeSWITCH debug logging (see the FreeSWITCH wiki for
>> details about that) and paste them too.
>>
>> --
>> Moises Silva
>> Senior Software Engineer
>> Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R
>> 9T3 Canada
>> t. 1 905 474 1990 x 128 | e. moy at sangoma.com
>>
>>  On Fri, Feb 26, 2010 at 12:55 AM, lakshmanan ganapathy <
>> lakindia89 at gmail.com> wrote:
>>
>>>  Dear all,
>>> I'm having a A102 Sangoma hardware. I configured it with freeswitch.
>>> wanrouter status, says both the port as connected.
>>> My smg_prid version is
>>>
>>> Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: ================System
>>> restart=============
>>> Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: = Sangoma PRI Protocol Stack
>>> Daemon       =
>>> Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: = Version:
>>> 1.54                           =
>>> Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: = Date: Feb 15
>>> 2010                       =
>>> Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: = Wanpipe Release:
>>> wanpipe-3.5.8.6       =
>>> Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: = Revision:Revision:
>>> 15288                =
>>> Feb 26 16:08:14 FMS-FreeSwitch
>>> sangoma_prid:
>>> ===========================================
>>>
>>> My freeswitch version is 16729.
>>> I started freeswitch.
>>>
>>> oz list
>>> +OK
>>> span: 1 (smg_prid)
>>> type: Sangoma (boost)
>>> chan_count: 60
>>> dialplan: XML
>>> context: default
>>> dial_regex:
>>> fail_dial_regex:
>>> hold_music:
>>> analog_options none
>>>
>>> I originated a call as
>>> originate openzap/smg_prid/a/9952248266 &park(), which hits my mobile.
>>>
>>> But when I issued the following command:
>>> originate openzap/smg_prid/a/9952248266
>>> &bridge(openzap/smg_prid/a/8122133885)
>>> It rings my mobile (9952248266) first, but after that the following error
>>> was displayed
>>>
>>> 2010-02-26 16:20:51.736080 [ERR] switch_ivr_originate.c:2387 Cannot
>>> create outgoing channel of type [openzap] cause: [NORMAL_CIRCUIT_CONGESTION]
>>> The call got ended in my mobile.
>>>
>>> Freeswitch log and smg_pri.conf
>>>           http://pastebin.freeswitch.org/12248
>>> openzap.conf:
>>> [span wanpipe smg_prid]
>>> name => smg_prid
>>> trunk_type =>e1
>>> b-channel => 1:1-15
>>> b-channel => 1:17-31
>>> trunk_type =>e1
>>> b-channel => 2:1-15
>>> b-channel => 2:17-31
>>>
>>> openzap.conf.xml:
>>> <configuration name="openzap.conf" description="OpenZAP Configuration">
>>>   <settings>
>>>     <param name="debug" value="0"/>
>>>   </settings>
>>>   <boost_spans>
>>>         <span id="smg_prid">
>>>         <param name="dialplan" value="XML"/>
>>>         <param name="context" value="default"/>
>>>         </span>
>>>   </boost_spans>
>>> </configuration>
>>>
>>> Please guide me to setup this one!!.
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100227/03b7567c/attachment-0002.html 

From lakindia89 at gmail.com  Fri Feb 26 21:13:45 2010
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Sat, 27 Feb 2010 10:43:45 +0530
Subject: [Freeswitch-users] smg_prid not bridging the call
In-Reply-To: <7d79b3931002262102w79e16ae1nd786600b44dfb758@mail.gmail.com>
References: <7d79b3931002252155t47c86968q9e451482926d93ac@mail.gmail.com>
	<c4d05cbe1002260731v29dbc428i1e72d24a220c62eb@mail.gmail.com>
	<7d79b3931002262057g3b325bctcdbcfb833d4aaf55@mail.gmail.com>
	<7d79b3931002262102w79e16ae1nd786600b44dfb758@mail.gmail.com>
Message-ID: <7d79b3931002262113s49495e7cga59715f41fdbb35d@mail.gmail.com>

I think it says Invalid Information Element for the DISPLAY
smg_prid/a/8122133885!!!
correct?? If so, can you please help me to solve this?

On Sat, Feb 27, 2010 at 10:32 AM, lakshmanan ganapathy <lakindia89 at gmail.com
> wrote:

> In the Dchan log it is saying Invalid Information Elements. That might be a
> problem??? But I even don't know why it is saying Invalid Information
> Element??
> Please guide me!!!
>
>
>
> On Sat, Feb 27, 2010 at 10:27 AM, lakshmanan ganapathy <
> lakindia89 at gmail.com> wrote:
>
>> Dear Moy,
>> Here are the details:
>>
>> FreeSwitch Log:
>> http://pastebin.freeswitch.org/12256
>>
>>  /var/log/sangoma_pri/dchan_<span-no>.log:
>>  http://pastebin.freeswitch.org/12257
>>
>> /var/log/sangoma_mgd.log:
>> http://pastebin.freeswitch.org/12258
>>
>> smg_pri.conf
>> http://pastebin.freeswitch.org/12259
>>
>>
>>
>> On Fri, Feb 26, 2010 at 9:01 PM, Moises Silva <moises.silva at gmail.com>wrote:
>>
>>> Hello lakshmanan,
>>>
>>> Please enable the dchan_log=q931 in /etc/wanpipe/smg_pri.conf and then
>>> restart it (smg_ctrl restart), then pastebin the logs
>>>
>>>  /var/log/sangoma_pri/dchan_<span-no>.log
>>> /var/log/sangoma_mgd.log
>>>
>>> That will contain the Q931 details (if any). Also pastebin your
>>> smg_pri.conf.
>>>
>>> Also enable the FreeSWITCH debug logging (see the FreeSWITCH wiki for
>>> details about that) and paste them too.
>>>
>>> --
>>> Moises Silva
>>> Senior Software Engineer
>>> Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R
>>> 9T3 Canada
>>> t. 1 905 474 1990 x 128 | e. moy at sangoma.com
>>>
>>>  On Fri, Feb 26, 2010 at 12:55 AM, lakshmanan ganapathy <
>>> lakindia89 at gmail.com> wrote:
>>>
>>>>  Dear all,
>>>> I'm having a A102 Sangoma hardware. I configured it with freeswitch.
>>>> wanrouter status, says both the port as connected.
>>>> My smg_prid version is
>>>>
>>>> Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: ================System
>>>> restart=============
>>>> Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: = Sangoma PRI Protocol
>>>> Stack Daemon       =
>>>> Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: = Version:
>>>> 1.54                           =
>>>> Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: = Date: Feb 15
>>>> 2010                       =
>>>> Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: = Wanpipe Release:
>>>> wanpipe-3.5.8.6       =
>>>> Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: = Revision:Revision:
>>>> 15288                =
>>>> Feb 26 16:08:14 FMS-FreeSwitch
>>>> sangoma_prid:
>>>> ===========================================
>>>>
>>>> My freeswitch version is 16729.
>>>> I started freeswitch.
>>>>
>>>> oz list
>>>> +OK
>>>> span: 1 (smg_prid)
>>>> type: Sangoma (boost)
>>>> chan_count: 60
>>>> dialplan: XML
>>>> context: default
>>>> dial_regex:
>>>> fail_dial_regex:
>>>> hold_music:
>>>> analog_options none
>>>>
>>>> I originated a call as
>>>> originate openzap/smg_prid/a/9952248266 &park(), which hits my mobile.
>>>>
>>>> But when I issued the following command:
>>>> originate openzap/smg_prid/a/9952248266
>>>> &bridge(openzap/smg_prid/a/8122133885)
>>>> It rings my mobile (9952248266) first, but after that the following
>>>> error was displayed
>>>>
>>>> 2010-02-26 16:20:51.736080 [ERR] switch_ivr_originate.c:2387 Cannot
>>>> create outgoing channel of type [openzap] cause: [NORMAL_CIRCUIT_CONGESTION]
>>>> The call got ended in my mobile.
>>>>
>>>> Freeswitch log and smg_pri.conf
>>>>           http://pastebin.freeswitch.org/12248
>>>> openzap.conf:
>>>> [span wanpipe smg_prid]
>>>> name => smg_prid
>>>> trunk_type =>e1
>>>> b-channel => 1:1-15
>>>> b-channel => 1:17-31
>>>> trunk_type =>e1
>>>> b-channel => 2:1-15
>>>> b-channel => 2:17-31
>>>>
>>>> openzap.conf.xml:
>>>> <configuration name="openzap.conf" description="OpenZAP Configuration">
>>>>   <settings>
>>>>     <param name="debug" value="0"/>
>>>>   </settings>
>>>>   <boost_spans>
>>>>         <span id="smg_prid">
>>>>         <param name="dialplan" value="XML"/>
>>>>         <param name="context" value="default"/>
>>>>         </span>
>>>>   </boost_spans>
>>>> </configuration>
>>>>
>>>> Please guide me to setup this one!!.
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100227/e7b3482b/attachment-0002.html 

From kond at nstel.ru  Fri Feb 26 23:15:22 2010
From: kond at nstel.ru (Nikolay Kondratyev)
Date: Sat, 27 Feb 2010 10:15:22 +0300
Subject: [Freeswitch-users] How to tie context to a gateway?
In-Reply-To: <20100216141634.75B3511FC6@mail.nstel.ru>
Message-ID: <20100227071523.10E2A1226D@mail.nstel.ru>

Hi all,

Raising my question again (see below).

I have some idea how to do it, but I'd like to know what experienced FS
users think.

Thanks in advance,

Nikolay.

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Nikolay
Kondratyev
Sent: Tuesday, February 16, 2010 5:17 PM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] How to tie context to a gateway?

 

Hi all,

I have several gateways in the external profile.

Let's say GW1 and GW2. I'd like to process calls from the GW1 in the context
C1 and calls from GW2 in the context C2.

Parameter "context", as far as I understand works for the whole profile, not
for individual gateways in the profile.

How do send calls from GW1 into context C1?

What will be a good practice to do that?

Thanks in advance,

Nikolay.

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100227/8f550720/attachment-0002.html 

From moises.silva at gmail.com  Sat Feb 27 10:35:33 2010
From: moises.silva at gmail.com (Moises Silva)
Date: Sat, 27 Feb 2010 13:35:33 -0500
Subject: [Freeswitch-users] smg_prid not bridging the call
In-Reply-To: <7d79b3931002262113s49495e7cga59715f41fdbb35d@mail.gmail.com>
References: <7d79b3931002252155t47c86968q9e451482926d93ac@mail.gmail.com>
	<c4d05cbe1002260731v29dbc428i1e72d24a220c62eb@mail.gmail.com>
	<7d79b3931002262057g3b325bctcdbcfb833d4aaf55@mail.gmail.com>
	<7d79b3931002262102w79e16ae1nd786600b44dfb758@mail.gmail.com>
	<7d79b3931002262113s49495e7cga59715f41fdbb35d@mail.gmail.com>
Message-ID: <c4d05cbe1002271035g6112b277hdf305b61233b94ad@mail.gmail.com>

I believe the problem FreeSWITCH is setting that as a default callerid name,
which your telco does not like.

Try setting the caller id name and number by yourself as explained in the
"originate" section here http://wiki.freeswitch.org/wiki/Mod_commands
<http://wiki.freeswitch.org/wiki/Mod_commands>

On Sat, Feb 27, 2010 at 12:13 AM, lakshmanan ganapathy <lakindia89 at gmail.com
> wrote:

> I think it says Invalid Information Element for the DISPLAY
> smg_prid/a/8122133885!!!
> correct?? If so, can you please help me to solve this?
>
>
> On Sat, Feb 27, 2010 at 10:32 AM, lakshmanan ganapathy <
> lakindia89 at gmail.com> wrote:
>
>> In the Dchan log it is saying Invalid Information Elements. That might be
>> a problem??? But I even don't know why it is saying Invalid Information
>> Element??
>> Please guide me!!!
>>
>>
>>
>> On Sat, Feb 27, 2010 at 10:27 AM, lakshmanan ganapathy <
>> lakindia89 at gmail.com> wrote:
>>
>>> Dear Moy,
>>> Here are the details:
>>>
>>> FreeSwitch Log:
>>> http://pastebin.freeswitch.org/12256
>>>
>>>  /var/log/sangoma_pri/dchan_<span-no>.log:
>>>  http://pastebin.freeswitch.org/12257
>>>
>>> /var/log/sangoma_mgd.log:
>>> http://pastebin.freeswitch.org/12258
>>>
>>> smg_pri.conf
>>> http://pastebin.freeswitch.org/12259
>>>
>>>
>>>
>>> On Fri, Feb 26, 2010 at 9:01 PM, Moises Silva <moises.silva at gmail.com>wrote:
>>>
>>>> Hello lakshmanan,
>>>>
>>>> Please enable the dchan_log=q931 in /etc/wanpipe/smg_pri.conf and then
>>>> restart it (smg_ctrl restart), then pastebin the logs
>>>>
>>>>  /var/log/sangoma_pri/dchan_<span-no>.log
>>>> /var/log/sangoma_mgd.log
>>>>
>>>> That will contain the Q931 details (if any). Also pastebin your
>>>> smg_pri.conf.
>>>>
>>>> Also enable the FreeSWITCH debug logging (see the FreeSWITCH wiki for
>>>> details about that) and paste them too.
>>>>
>>>> --
>>>> Moises Silva
>>>> Senior Software Engineer
>>>> Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R
>>>> 9T3 Canada
>>>> t. 1 905 474 1990 x 128 | e. moy at sangoma.com
>>>>
>>>>  On Fri, Feb 26, 2010 at 12:55 AM, lakshmanan ganapathy <
>>>> lakindia89 at gmail.com> wrote:
>>>>
>>>>>  Dear all,
>>>>> I'm having a A102 Sangoma hardware. I configured it with freeswitch.
>>>>> wanrouter status, says both the port as connected.
>>>>> My smg_prid version is
>>>>>
>>>>> Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: ================System
>>>>> restart=============
>>>>> Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: = Sangoma PRI Protocol
>>>>> Stack Daemon       =
>>>>> Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: = Version:
>>>>> 1.54                           =
>>>>> Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: = Date: Feb 15
>>>>> 2010                       =
>>>>> Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: = Wanpipe Release:
>>>>> wanpipe-3.5.8.6       =
>>>>> Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: = Revision:Revision:
>>>>> 15288                =
>>>>> Feb 26 16:08:14 FMS-FreeSwitch
>>>>> sangoma_prid:
>>>>> ===========================================
>>>>>
>>>>> My freeswitch version is 16729.
>>>>> I started freeswitch.
>>>>>
>>>>> oz list
>>>>> +OK
>>>>> span: 1 (smg_prid)
>>>>> type: Sangoma (boost)
>>>>> chan_count: 60
>>>>> dialplan: XML
>>>>> context: default
>>>>> dial_regex:
>>>>> fail_dial_regex:
>>>>> hold_music:
>>>>> analog_options none
>>>>>
>>>>> I originated a call as
>>>>> originate openzap/smg_prid/a/9952248266 &park(), which hits my mobile.
>>>>>
>>>>> But when I issued the following command:
>>>>> originate openzap/smg_prid/a/9952248266
>>>>> &bridge(openzap/smg_prid/a/8122133885)
>>>>> It rings my mobile (9952248266) first, but after that the following
>>>>> error was displayed
>>>>>
>>>>> 2010-02-26 16:20:51.736080 [ERR] switch_ivr_originate.c:2387 Cannot
>>>>> create outgoing channel of type [openzap] cause: [NORMAL_CIRCUIT_CONGESTION]
>>>>> The call got ended in my mobile.
>>>>>
>>>>> Freeswitch log and smg_pri.conf
>>>>>           http://pastebin.freeswitch.org/12248
>>>>> openzap.conf:
>>>>> [span wanpipe smg_prid]
>>>>> name => smg_prid
>>>>> trunk_type =>e1
>>>>> b-channel => 1:1-15
>>>>> b-channel => 1:17-31
>>>>> trunk_type =>e1
>>>>> b-channel => 2:1-15
>>>>> b-channel => 2:17-31
>>>>>
>>>>> openzap.conf.xml:
>>>>> <configuration name="openzap.conf" description="OpenZAP Configuration">
>>>>>   <settings>
>>>>>     <param name="debug" value="0"/>
>>>>>   </settings>
>>>>>   <boost_spans>
>>>>>         <span id="smg_prid">
>>>>>         <param name="dialplan" value="XML"/>
>>>>>         <param name="context" value="default"/>
>>>>>         </span>
>>>>>   </boost_spans>
>>>>> </configuration>
>>>>>
>>>>> Please guide me to setup this one!!.
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100227/4d266a48/attachment-0002.html 

From michal.kalinowski at interia.pl  Sat Feb 27 13:10:16 2010
From: michal.kalinowski at interia.pl (michal kalinowski)
Date: Sat, 27 Feb 2010 22:10:16 +0100
Subject: [Freeswitch-users] ivr from mysql
In-Reply-To: <E9EBF591-5DEC-4FF3-B2D8-20DDA54B4FF5@jerris.com>
References: <7c74f5761002150824i41a2ed5did008f36cbb4499a4@mail.gmail.com>
	<87f2f3b91002151002v6bbc3e7fgb67dd6d3e72a915@mail.gmail.com>
	<7c74f5761002151425g512fa685n13c58d8f4b894740@mail.gmail.com>
	<0FD30AAA-C2E2-4228-A09B-DB09F4E26811@avgs.ca>
	<7c74f5761002151525p7cc690ces632d46c6ab5e236c@mail.gmail.com>
	<E9EBF591-5DEC-4FF3-B2D8-20DDA54B4FF5@jerris.com>
Message-ID: <7c74f5761002271310h7e925f7cy25dd69ba60b896ef@mail.gmail.com>

Coming back to this case I create in lua some script with XML ivr.

#!/usr/local/bin/lua

mydialplan = [[
<?xml version="1.0" encoding="UTF-8" standalone="no"?>
<document type="freeswitch/xml">
<include>
  <!-- demo IVR setup -->
  <!-- demo IVR, Main Menu -->
  <menu name="demo_ivr"
      greet-long="phrase:demo_ivr_main_menu"
      greet-short="phrase:demo_ivr_main_menu_short"
      invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"
      exit-sound="voicemail/vm-goodbye.wav"
      confirm-macro=""
      confirm-key=""
      tts-engine="flite"
      tts-voice="rms"
      confirm-attempts="3"
      timeout="10000"
      inter-digit-timeout="2000"
      max-failures="3"
      max-timeouts="3"
      digit-len="4">

    <!-- The following are the definitions for the digits the user dials -->
    <!-- Digit 1 transfer caller to the public FreeSWITCH conference -->
    <entry action="menu-exec-app" digits="1" param="bridge
sofia/$${domain}/888 at conference.freeswitch.org"/>
    <entry action="menu-exec-app" digits="2" param="transfer 9996 XML
default"/>    <!-- FS echo -->
    <entry action="menu-exec-app" digits="3" param="transfer 9999 XML
default"/>    <!-- MOH -->
    <entry action="menu-exec-app" digits="4" param="transfer 9991 XML
default"/>    <!-- ClueCon -->
    <entry action="menu-exec-app" digits="5" param="transfer 1234*256
enum"/>       <!-- Screaming monkeys -->
    <entry action="menu-sub" digits="6" param="demo_ivr_submenu"/>
             <!-- demo sub menu -->
    <!-- Using a regex in the digits tag lets you define a dial
pattern for the caller
         You may define multiple regexes if you need a different
pattern for some reason -->
    <entry action="menu-exec-app" digits="/^(10[01][0-9])$/"
param="transfer $1 XML features"/>
    <entry action="menu-top" digits="9"/>          <!-- Repeat this menu -->
  </menu>

</include>
]]
XML_STRING = mydialplan

in dialplan I have context with this ivr

  <context name="ivr">
<extension name="ivr">
     <condition field="caller_id_number" expression="^[0-9]"/>
     <condition field="destination_number" expression="^(\d{11})$">
        <action application="answer"/>
        <action application="ivr" data="demo_ivr"/>
     </condition>
</extension>
  </context>

in ivr.conf i have this

<configuration name="ivr.conf" description="IVR menus">
  <menus>
        <X-PRE-PROCESS cmd="set" data="application=lua lua.ivr"/>
  </menus>
</configuration>

but for some reasons Freeswitch say "2010-02-27 22:27:48.380342 [ERR]
mod_dptools.c:1247 Unable to find menu"
what I do wrong ?


BR,
Micha?

2010/2/18 Michael Jerris <mike at jerris.com>:
> an example is available here : ? http://svn.freeswitch.org/svn/freeswitch/trunk/conf/ivr_menus/demo_ivr.xml
>
> Mike
>
> On Feb 15, 2010, at 6:25 PM, michal kalinowski wrote:
>> Could you insert several examples here?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From brian at freeswitch.org  Sat Feb 27 13:48:04 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 27 Feb 2010 15:48:04 -0600
Subject: [Freeswitch-users] ivr from mysql
In-Reply-To: <7c74f5761002271310h7e925f7cy25dd69ba60b896ef@mail.gmail.com>
References: <7c74f5761002150824i41a2ed5did008f36cbb4499a4@mail.gmail.com>
	<87f2f3b91002151002v6bbc3e7fgb67dd6d3e72a915@mail.gmail.com>
	<7c74f5761002151425g512fa685n13c58d8f4b894740@mail.gmail.com>
	<0FD30AAA-C2E2-4228-A09B-DB09F4E26811@avgs.ca>
	<7c74f5761002151525p7cc690ces632d46c6ab5e236c@mail.gmail.com>
	<E9EBF591-5DEC-4FF3-B2D8-20DDA54B4FF5@jerris.com>
	<7c74f5761002271310h7e925f7cy25dd69ba60b896ef@mail.gmail.com>
Message-ID: <23CC9D8A-65D5-438F-B117-00FEC087418D@freeswitch.org>

can you point out on the wiki that indicated you are able to dot his?  You're mixing two concepts incorrectly here.

For example you can't do "set" and exec the file to get the contents.. you can however on linux use "exec" instead of set.  But your script needs to print it to stdout.  Your example in your lua script is for the config engine in in lua thats like XML curl... in which case you're not building the full document like you should.  Read thru the XML CURL docs if you want to do "XML_STRING = mydialplan".  

/b


On Feb 27, 2010, at 3:10 PM, michal kalinowski wrote:

> Coming back to this case I create in lua some script with XML ivr.
> 
> #!/usr/local/bin/lua
> 
> mydialplan = [[
> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
> <document type="freeswitch/xml">
> <include>
>  <!-- demo IVR setup -->
>  <!-- demo IVR, Main Menu -->
>  <menu name="demo_ivr"
>      greet-long="phrase:demo_ivr_main_menu"
>      greet-short="phrase:demo_ivr_main_menu_short"
>      invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"
>      exit-sound="voicemail/vm-goodbye.wav"
>      confirm-macro=""
>      confirm-key=""
>      tts-engine="flite"
>      tts-voice="rms"
>      confirm-attempts="3"
>      timeout="10000"
>      inter-digit-timeout="2000"
>      max-failures="3"
>      max-timeouts="3"
>      digit-len="4">
> 
>    <!-- The following are the definitions for the digits the user dials -->
>    <!-- Digit 1 transfer caller to the public FreeSWITCH conference -->
>    <entry action="menu-exec-app" digits="1" param="bridge
> sofia/$${domain}/888 at conference.freeswitch.org"/>
>    <entry action="menu-exec-app" digits="2" param="transfer 9996 XML
> default"/>    <!-- FS echo -->
>    <entry action="menu-exec-app" digits="3" param="transfer 9999 XML
> default"/>    <!-- MOH -->
>    <entry action="menu-exec-app" digits="4" param="transfer 9991 XML
> default"/>    <!-- ClueCon -->
>    <entry action="menu-exec-app" digits="5" param="transfer 1234*256
> enum"/>       <!-- Screaming monkeys -->
>    <entry action="menu-sub" digits="6" param="demo_ivr_submenu"/>
>             <!-- demo sub menu -->
>    <!-- Using a regex in the digits tag lets you define a dial
> pattern for the caller
>         You may define multiple regexes if you need a different
> pattern for some reason -->
>    <entry action="menu-exec-app" digits="/^(10[01][0-9])$/"
> param="transfer $1 XML features"/>
>    <entry action="menu-top" digits="9"/>          <!-- Repeat this menu -->
>  </menu>
> 
> </include>
> ]]
> XML_STRING = mydialplan
> 
> in dialplan I have context with this ivr
> 
>  <context name="ivr">
> <extension name="ivr">
>     <condition field="caller_id_number" expression="^[0-9]"/>
>     <condition field="destination_number" expression="^(\d{11})$">
>        <action application="answer"/>
>        <action application="ivr" data="demo_ivr"/>
>     </condition>
> </extension>
>  </context>
> 
> in ivr.conf i have this
> 
> <configuration name="ivr.conf" description="IVR menus">
>  <menus>
>        <X-PRE-PROCESS cmd="set" data="application=lua lua.ivr"/>
>  </menus>
> </configuration>
> 
> but for some reasons Freeswitch say "2010-02-27 22:27:48.380342 [ERR]
> mod_dptools.c:1247 Unable to find menu"
> what I do wrong ?
> 
> 
> BR,
> Micha?




From mbsip at gazeta.pl  Sat Feb 27 14:35:40 2010
From: mbsip at gazeta.pl (Maciej Bylica)
Date: Sat, 27 Feb 2010 23:35:40 +0100
Subject: [Freeswitch-users] Phrases - Can't find macro
In-Reply-To: <87f2f3b91002261245l176c49f6j769bd673be0324fe@mail.gmail.com>
References: <28f27f5d1002260910h7b3ce773l28294e7354078b05@mail.gmail.com>
	<87f2f3b91002261245l176c49f6j769bd673be0324fe@mail.gmail.com>
Message-ID: <28f27f5d1002271435x20e068b1w6112c5f19ef6b146@mail.gmail.com>

Thx Michael for Your prompt answer.
I did exactly what you had said -- in addition there was a need to use
<match>, but result is possitive.

Thank You,
Maciej

> You might try this suggestion:
> Create a new file for your custom macros:
> /conf/lang/en/demo/custom-phrases.xml
> <include>
> ? <macro name="confwelcome">
> ??? <input pattern="(.*)">
> ????? <action function="play-file" data="vm/vm-youhave.wav"/>
> ??? </input>
> ? </macro>
> </include>
>
> Now you have a single place to put all of your custom macros. Be sure to
> reloadxml!
> -MC



From msc at freeswitch.org  Sat Feb 27 16:45:15 2010
From: msc at freeswitch.org (Michael Collins)
Date: Sat, 27 Feb 2010 16:45:15 -0800
Subject: [Freeswitch-users] Phrases - Can't find macro
In-Reply-To: <28f27f5d1002271435x20e068b1w6112c5f19ef6b146@mail.gmail.com>
References: <28f27f5d1002260910h7b3ce773l28294e7354078b05@mail.gmail.com>
	<87f2f3b91002261245l176c49f6j769bd673be0324fe@mail.gmail.com>
	<28f27f5d1002271435x20e068b1w6112c5f19ef6b146@mail.gmail.com>
Message-ID: <87f2f3b91002271645x54e2dd5eqc3a2169b9eca2e39@mail.gmail.com>

On Sat, Feb 27, 2010 at 2:35 PM, Maciej Bylica <mbsip at gazeta.pl> wrote:

> Thx Michael for Your prompt answer.
> I did exactly what you had said -- in addition there was a need to use
> <match>, but result is possitive.
>

You are quite correct - I left out the <match> and optional <nomatch> nodes.
-MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100227/04bf978f/attachment-0002.html 

From brian at microcomaustralia.com.au  Sat Feb 27 17:11:33 2010
From: brian at microcomaustralia.com.au (Brian May)
Date: Sun, 28 Feb 2010 12:11:33 +1100
Subject: [Freeswitch-users] Possible typo on Openzap.conf wiki page
	specifying fxs/fxo-channels
In-Reply-To: <a689f57fe216d671d7d1ff132c6b636e@thom.fr.eu.org>
References: <2C33D9D6C1444AAC80950CDF8D38A428@greyhawk.tonecommander.com>
	<3c5cf5261002231524n4e9af307y80b3accb1ee4ff26@mail.gmail.com>
	<a689f57fe216d671d7d1ff132c6b636e@thom.fr.eu.org>
Message-ID: <3c5cf5261002271711t59294b4fm9aebfdf5b48a6b31@mail.gmail.com>

On 24 February 2010 20:25, Fran?ois Legal <devel at thom.fr.eu.org> wrote:
> I use sangoma card and the openzap file is generated by the Setup script
> from sangoma driver.
> It seems that the terminology used by zaptel is not used in wanpipe
> configuration.

Yes, that is correct.

> I have an A400 card with an FXO module (providing ports 11 and 12) and an
> FXS module (providing ports 9 and 10)
>
> My openzap.conf is like this :
>
> [span wanpipe FXS]
> name => Analog phone 1
> number => 9000
> fxs-channel => 1:9
> name => Analog phone 2
> number => 9001
> fxs-channel => 1:10
>
> [span wanpipe FXO]
> name => POTS line 1
> number => 1234567890
> fxo-channel => 1:11
> name => POTS line 2
> number => 0987654321
> fxo-channel => 1:12

So ports 9 and 10 are actually FXO ports - extension ports; ports 11
and 12 are FXS ports, or telephone lines. This is what I have been
saying.

Oh, wait, no it isn't. Looks like I was confused. :-(

It matches my config however.

Hopefully this fixed the problems with the wiki:

http://wiki.freeswitch.org/index.php?title=Openzap.conf_Examples&diff=18693&oldid=18491

My guess is that this change is needed also (not absolutely sure here):

http://wiki.freeswitch.org/index.php?title=Openzap.conf_Examples&diff=18694&oldid=18693
-- 
Brian May <brian at microcomaustralia.com.au>



From christian.loeschenkohl at xpirio.com  Sun Feb 28 00:57:48 2010
From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=)
Date: Sun, 28 Feb 2010 09:57:48 +0100
Subject: [Freeswitch-users] conferences lead to high server load
In-Reply-To: <191c3a031002241058y517e63fdte0d929b97a33f9ac@mail.gmail.com>
References: <4B857226.10308@xpirio.com>
	<191c3a031002241058y517e63fdte0d929b97a33f9ac@mail.gmail.com>
Message-ID: <4B8A300C.4060805@xpirio.com>

hello

problem solved with -vm
with this option we now have the usual low load for 50-70 conference users

i think it would be good to explain this in the wiki, how can i get more information
on this to put it in there - "use possibly more vm-friendly timing code" wouldn't be enough :-)
i could create an article about the startup flags in general

br


Anthony Minessale wrote:

> load average has no meaning with FS, you have to look at the CPU usage 
> per CPU and thread.
> Are you experiencing any audio problems or are you just concerned about 
> that load number?
> 
> If you have a box that has trouble with timing it could cost more resources.
> you can always run freeswitch -vm to use an alternate form of timing 
> that may not manifest into the load average.
> 
> 
> 2010/2/24 Christian L?schenkohl <christian.loeschenkohl at xpirio.com 
> <mailto:christian.loeschenkohl at xpirio.com>>
> 
>     hi
> 
>     we do experience a unusual high server load with the latest
>     freeswitch versions.
>     about 50 conference users lead to a server load of over 10 -
>     reproducible by the way.
>     this wans't the case until my latest trunk update.
> 
>     fs version: 16714
>     os: debian lenny x86_64
> 
>     has something substantially changed in mod_conference recently?
> 
>     br
> 
>     --
>     Ing. Christian L?schenkohl
>     Technische Leitung, Forschung & Entwicklung VoIP
> 
>     xpirio
>     Telekommunikation & Service GmbH
>     Lakeside B04
>     9020 Klagenfurt
>     Austria
> 
>     T  +43 (0) 5 77 11 - 1000
>     F  +43 (0) 5 77 11 - 1002
>     E  christian.loeschenkohl at xpirio.com
>     <mailto:christian.loeschenkohl at xpirio.com>
> 
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
> 
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com 
> <mailto:MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com 
> <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org 
> <mailto:sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888 
> <http://iax:guest at conference.freeswitch.org/888>
> googletalk:conf+888 at conference.freeswitch.org 
> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


-- 
Ing. Christian L?schenkohl
Technische Leitung, Forschung & Entwicklung VoIP

xpirio
Telekommunikation & Service GmbH
Lakeside B04
9020 Klagenfurt
Austria

T  +43 (0) 5 77 11 - 1000
F  +43 (0) 5 77 11 - 1002
E  christian.loeschenkohl at xpirio.com



From yehavi.bourvine at gmail.com  Sun Feb 28 01:49:16 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Sun, 28 Feb 2010 11:49:16 +0200
Subject: [Freeswitch-users] Announcement: FreeSWITCH Add Broadsoft SCA
	Support
In-Reply-To: <191c3a031002190908r3922efccqc6d223a5fabd458b@mail.gmail.com>
References: <87f2f3b91001111742k691966cfs82515db77c13d4d4@mail.gmail.com>
	<191c3a031002170557g5d8c9df9xf40d1b74b9a6f19d@mail.gmail.com>
	<d361ca7b1002170700x15630ab2tabc6f3d91ebc3303@mail.gmail.com>
	<ADC8B813-4161-481C-877E-93FC012858B4@freeswitch.org>
	<d361ca7b1002182101p60b4e8f8hf4dc95e1d2251e30@mail.gmail.com>
	<8b1c9cda1002182144m65ccf979w71384a1e7a55e2ea@mail.gmail.com>
	<d361ca7b1002182154o52352c62h8c0ec02fa66e8444@mail.gmail.com>
	<2F87B95C-EA23-4072-B845-3F7AE0617BE3@jerris.com>
	<d361ca7b1002182346k71fd602fh6e6725834a6de528@mail.gmail.com>
	<191c3a031002190908r3922efccqc6d223a5fabd458b@mail.gmail.com>
Message-ID: <d361ca7b1002280149i18e66cbdo2031e7f385f23075@mail.gmail.com>

Hello all,

The problem is solved, at least for my Polycoms.

The solution is to have "domain" variable (in vars.xml) set to the IP
address of the profile that is used for phones registration (the default is
ok if you have only one interface). Define the phone's registration and
proxy servers to that address. You must use IP addresses and not DNS names.

                  Thanks to Anthony for his helpfull tips!  __Yehavi:

 2010/2/19 Anthony Minessale <anthony.minessale at gmail.com>

> go see my comments on that bug note.
> be prepared to give us ssh access and call or irc so we can can see you
> reproducing it.
>
> If you are not on the latest firmware on all the phones, we will not
> continue with this process.
>
>
>
>
> On Fri, Feb 19, 2010 at 1:46 AM, Yehavi Bourvine <
> yehavi.bourvine at gmail.com> wrote:
>
>>  A jira issue has been created: *MODSOFIA-61*<http://jira.freeswitch.org/browse/MODSOFIA-61>
>> .
>>
>>                   Thanks, __Yehavi:
>>
>>  2010/2/19 Michael Jerris <mike at jerris.com>
>>
>>>  If this issue is not already on jira could you please make sure it gets
>>> added?
>>>
>>> Mike
>>>
>>>
>>> On Feb 19, 2010, at 12:54 AM, Yehavi Bourvine <yehavi.bourvine at gmail.com>
>>> wrote:
>>>
>>>     Hello Gabe,
>>>
>>>   As you can see - Brian is actively investigating it, so you can expect
>>> for some fix soon...
>>>
>>>                        Regards, __Yehavi:
>>>
>>> 2010/2/19 Gabriel Kuri < <gkuri at ieee.org>gkuri at ieee.org>
>>>
>>>> >  When a call arrives, both ring; the one that did not answer gets only
>>>> a
>>>> > cancel mesage without any further notification that the extension is
>>>> in use
>>>> > by the other phone.
>>>>
>>>> These are the same exact symptoms I posted about earlier this week,
>>>> with the Cisco SPA-5xx series phones. I still have yet to figure out
>>>> why this is happening, if you find out what's going on, please post
>>>> back the solution, I'd like to know the resolution.
>>>>
>>>> Thanks,
>>>> Gabe
>>>>
>>>>
>>>>
>>>> On Thu, Feb 18, 2010 at 9:01 PM, Yehavi Bourvine
>>>> < <yehavi.bourvine at gmail.com>yehavi.bourvine at gmail.com> wrote:
>>>> > Thanks Brian. It now works better, but not fully (using 16659M).
>>>> >
>>>> > What happens is:
>>>> >
>>>> > When one of the Polycoms seize the line it is ok - the other phone
>>>> gets
>>>> > notification and the extension status is "in use".
>>>> > When one of the Polycom phones initiates a call - all is ok:
>>>> >
>>>> > The other side sees that the extension is in use.
>>>> > When it is put to hold all phones who share this extension see it and
>>>> can
>>>> > pick the call.
>>>> >
>>>>
>>>> >
>>>>  >                          Thanks! __Yehavi:
>>>> >
>>>> > 2010/2/17 Brian West < <brian at freeswitch.org>brian at freeswitch.org>
>>>> >>
>>>> >> Step 1. Enable manage-shared-appearance=true
>>>> >>
>>>> >> Step 2. Now in the phone's config Configure the phone as usually, set
>>>> the
>>>> >> line shared and DO NOT set the third party name.
>>>> >>
>>>> >> Step 3. Reboot
>>>> >>
>>>> >> It should work.
>>>> >>
>>>> >> I wish someone that has this working would write some wiki docs these
>>>> >> threads about it not working are getting rather old when I know for a
>>>> fact
>>>> >> they work fine.
>>>> >>
>>>> >> The gateway info missing is a gateway you have configured getting a
>>>> >> notify.  It has nothing to do with SCA.
>>>> >>
>>>> >> /b
>>>> >>
>>>> >> On Feb 17, 2010, at 9:00 AM, Yehavi Bourvine wrote:
>>>> >>
>>>> >> > .
>>>> >>
>>>> >>
>>>> >> _______________________________________________
>>>> >> FreeSWITCH-users mailing list
>>>> >> <FreeSWITCH-users at lists.freeswitch.org>
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> >> <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> >> <http://www.freeswitch.org/>http://www.freeswitch.org
>>>> >
>>>> > _______________________________________________
>>>> > FreeSWITCH-users mailing list
>>>> > <FreeSWITCH-users at lists.freeswitch.org>
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> > <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> > UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> > <http://www.freeswitch.org/>http://www.freeswitch.org
>>>> >
>>>> >
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> <FreeSWITCH-users at lists.freeswitch.org>
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> <http://www.freeswitch.org/>http://www.freeswitch.org
>>>>
>>>  _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100228/31dbad40/attachment-0002.html 

From mattdfong at gmail.com  Sun Feb 28 03:36:18 2010
From: mattdfong at gmail.com (Matthew Fong)
Date: Sun, 28 Feb 2010 18:36:18 +0700
Subject: [Freeswitch-users] Detecting Energy Levels on a Channel with mod_lua
Message-ID: <4256bf831002280336vb7268e2oe48b0f74bac335f8@mail.gmail.com>

Is there anyway to detect energy levels on a channel that is being
controlled by mod_lua? Please point me in the right direction if there is.
Thanks.

--matt
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100228/4e5c6183/attachment-0002.html 

From yehavi.bourvine at gmail.com  Sun Feb 28 05:14:44 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Sun, 28 Feb 2010 15:14:44 +0200
Subject: [Freeswitch-users] SQL ERR: library routine called out of
	sequence
In-Reply-To: <191c3a031002161230h2b6e6f9wf3f65ddb71f08297@mail.gmail.com>
References: <d361ca7b1002160656w7d616f6dn5e80087ebde94dbe@mail.gmail.com>
	<191c3a031002160705w4050c3cdk2bf2922e9aff5ba3@mail.gmail.com>
	<d361ca7b1002160727ucb6e0c8id62e803757333213@mail.gmail.com>
	<191c3a031002160755q569fe2ednfefb01b6d9cf7f0e@mail.gmail.com>
	<d361ca7b1002161059w1ef9484dr87e69a33aa2b62fc@mail.gmail.com>
	<191c3a031002161110s637a48a2r717ae94557fe9e9a@mail.gmail.com>
	<d361ca7b1002161130k3eb2d88by3100b30a9d29bd51@mail.gmail.com>
	<191c3a031002161230h2b6e6f9wf3f65ddb71f08297@mail.gmail.com>
Message-ID: <d361ca7b1002280514p71003588t8a0daebce5ba9e4d@mail.gmail.com>

Hello,

  I've installed a machine with CentOS-5.4 and the latest FreeSwitch
(16841M). The problem still happens, even if I set sql-in-transactions to
false. It happens also on a very light load.

I would like to remind that we query the SQLite core DB from a LUA script
which is called from the dial plan; might this be the cause of it?

I am willing to test whatever you like, and can do it now as this machine is
not in production yet.

            Thanks! __Yehavi:

 2010/2/16 Anthony Minessale <anthony.minessale at gmail.com>

> Strange, even on abusive testing we have not seen this problem.
>
> please update to latest trunk.
> There was only one change I can think of that may cause your issue and I
> added a patch for it.
> If it persists try setting the sql-in-transactions profile param to false.
>
>
>
>
>
> On Tue, Feb 16, 2010 at 1:30 PM, Yehavi Bourvine <
> yehavi.bourvine at gmail.com> wrote:
>
>>  Most of the queries are ok, only some fail, thus it doesn't look like
>> permission problem. Furthermore, under 1.0.5pre10 it works for months.
>>
>> Might it be thread unsafe function calls? I've found the following while
>> searching the WEB:
>>
>> *According to the MSDN docs, System.Timers.Timer operates in a thread
>> pool.  If that's the case, your code is breaking the "connections cannot be
>> shared across threads" rule for SQLit*
>>
>> Although it quotes MSDN, it might be related to Linux as well.
>>
>>                     Thanks, __Yehavi:
>>
>>  2010/2/16 Anthony Minessale <anthony.minessale at gmail.com>
>>
>>> That sounds about right.
>>>
>>> That error usually has something to do with using db calls on a closed
>>> file or something along those lines.
>>> Maybe you have a permission problem on the directory where the db files
>>> are?
>>>
>>>
>>>
>>> On Tue, Feb 16, 2010 at 12:59 PM, Yehavi Bourvine <
>>> yehavi.bourvine at gmail.com> wrote:
>>>
>>>>  The OS is Fedora-10 (soon to be upgraded to 12).
>>>>
>>>> What I do when I want to test a new version:
>>>>
>>>>    - Download the latest one into a fresh directory
>>>>    - bootstrap.sh, configure and make
>>>>    - stop Freeswitch, delete everything in lib, mod, bin ,db
>>>>    - make install and run it.
>>>>
>>>>
>>>> Is there additional place to clean?
>>>>
>>>>                         Thanks! __Yehavi:
>>>>
>>>>  2010/2/16 Anthony Minessale <anthony.minessale at gmail.com>
>>>>
>>>>> you may want to do a clean wipe of all files related to FS then.
>>>>> you clearly have some problem with legacy something or other because we
>>>>> don't see that on dozens of dev boxes.
>>>>>
>>>>> What os is it?
>>>>>
>>>>>
>>>>>
>>>>> On Tue, Feb 16, 2010 at 9:27 AM, Yehavi Bourvine <
>>>>> yehavi.bourvine at gmail.com> wrote:
>>>>>
>>>>>>  Tried this, but it didn't help. I delete these DB files before any
>>>>>> upgrade just to be sure.
>>>>>>
>>>>>>                   Thanks! __Yehavi:
>>>>>>
>>>>>>  2010/2/16 Anthony Minessale <anthony.minessale at gmail.com>
>>>>>>
>>>>>>> try removing all the .db files from /usr/local/freeswitch/db
>>>>>>>
>>>>>>>
>>>>>>>   On Tue, Feb 16, 2010 at 8:56 AM, Yehavi Bourvine <
>>>>>>> yehavi.bourvine at gmail.com> wrote:
>>>>>>>
>>>>>>>>   Hello,
>>>>>>>>
>>>>>>>>   After upgrading from 1.0.5pre10 to 1.0.5-20100216-0400 we started
>>>>>>>> getting the above errors (I append bellow two samples). It seems Freeswitch
>>>>>>>> fails to read a database using Sqlite.
>>>>>>>> Anyone have seen this?
>>>>>>>>
>>>>>>>> Other details: Fedora 10, SQlite 3.5.9.
>>>>>>>> We also do SQLite quesries during call setup via LUA from CoreDB. Is
>>>>>>>> it an SQLite problem?
>>>>>>>>
>>>>>>>>                          Thanks! __Yehavi:
>>>>>>>>
>>>>>>>> The samples:
>>>>>>>> 2010-02-16 16:22:13.762679 [ERR] switch_core_sqldb.c:404 SQL ERR
>>>>>>>> [library routin
>>>>>>>> e called out of sequence]
>>>>>>>> delete from sip_dialogs where call_id='
>>>>>>>> 8656841832142-120172129116107 at 10.64.1.2'
>>>>>>>>
>>>>>>>> 2010-02-16 16:42:00.802442 [ERR] switch_core_sqldb.c:722 SQL ERR:
>>>>>>>> [select call_i
>>>>>>>>
>>>>>>>> d,sip_user,sip_host,contact,status,rpid,expires,user_agent,server_user,server_ho
>>>>>>>>
>>>>>>>> st,profile_name,hostname,network_ip,network_port,sip_username,sip_realm,mwi_user
>>>>>>>> ,mwi_host from sip_registrations where profile_name='phones' and
>>>>>>>> contact like '%
>>>>>>>> 80635%'] library routine called out of sequence
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> --
>>>>>>> Anthony Minessale II
>>>>>>>
>>>>>>> FreeSWITCH http://www.freeswitch.org/
>>>>>>> ClueCon http://www.cluecon.com/
>>>>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>>>>
>>>>>>> AIM: anthm
>>>>>>> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
>>>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>>>>> IRC: irc.freenode.net #freeswitch
>>>>>>>
>>>>>>> FreeSWITCH Developer Conference
>>>>>>> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>>>>>>> iax:guest at conference.freeswitch.org/888
>>>>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>>>>> pstn:+19193869900
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> Anthony Minessale II
>>>>>
>>>>> FreeSWITCH http://www.freeswitch.org/
>>>>> ClueCon http://www.cluecon.com/
>>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>>
>>>>> AIM: anthm
>>>>> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>>> IRC: irc.freenode.net #freeswitch
>>>>>
>>>>> FreeSWITCH Developer Conference
>>>>> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>>>>> iax:guest at conference.freeswitch.org/888
>>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>>> pstn:+19193869900
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>> iax:guest at conference.freeswitch.org/888
>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> pstn:+19193869900
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100228/956a418b/attachment-0002.html 

From yehavi.bourvine at gmail.com  Sun Feb 28 06:47:56 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Sun, 28 Feb 2010 16:47:56 +0200
Subject: [Freeswitch-users] SQL ERR: library routine called out of
	sequence
In-Reply-To: <d361ca7b1002280514p71003588t8a0daebce5ba9e4d@mail.gmail.com>
References: <d361ca7b1002160656w7d616f6dn5e80087ebde94dbe@mail.gmail.com>
	<191c3a031002160705w4050c3cdk2bf2922e9aff5ba3@mail.gmail.com>
	<d361ca7b1002160727ucb6e0c8id62e803757333213@mail.gmail.com>
	<191c3a031002160755q569fe2ednfefb01b6d9cf7f0e@mail.gmail.com>
	<d361ca7b1002161059w1ef9484dr87e69a33aa2b62fc@mail.gmail.com>
	<191c3a031002161110s637a48a2r717ae94557fe9e9a@mail.gmail.com>
	<d361ca7b1002161130k3eb2d88by3100b30a9d29bd51@mail.gmail.com>
	<191c3a031002161230h2b6e6f9wf3f65ddb71f08297@mail.gmail.com>
	<d361ca7b1002280514p71003588t8a0daebce5ba9e4d@mail.gmail.com>
Message-ID: <d361ca7b1002280647l700206cfv6e130351d8ac6d5@mail.gmail.com>

After a few more tests I *think* it is related to SLA. Since it cannot be
reproduced consistenty I say "think".

What I have is:
- 80635 which is an SLA extension between two Polycoms.
- 80636 which is a private extension on Polycom
- 80632 which is a Cisco extension.
- 86111 which is connected behind a Cisco SIP<->PSTN gateway. Since this
gateway doesn't hav any login information it is not defined as a gateway but
accepted via ACL.

The tests that passed ok:
- From/to 80636/80632 to 80635 (i.e. - all internal).
- From/to 80635 to 80636/80632 (i.e. - all internal)
- From/to 80636/80632 to 86111 and vice versa (i.e. via the gateway, single
extensions).

The one that fails after a few attempts:
- From 86111 to 80635 (i.e. from the gateway to a shared extension).

I hope that this gives some more clues.

                          Thanks, __Yehavi:

2010/2/28 Yehavi Bourvine <yehavi.bourvine at gmail.com>

>  Hello,
>
>   I've installed a machine with CentOS-5.4 and the latest FreeSwitch
> (16841M). The problem still happens, even if I set sql-in-transactions to
> false. It happens also on a very light load.
>
> I would like to remind that we query the SQLite core DB from a LUA script
> which is called from the dial plan; might this be the cause of it?
>
> I am willing to test whatever you like, and can do it now as this machine
> is not in production yet.
>
>             Thanks! __Yehavi:
>
>  2010/2/16 Anthony Minessale <anthony.minessale at gmail.com>
>
>> Strange, even on abusive testing we have not seen this problem.
>>
>> please update to latest trunk.
>> There was only one change I can think of that may cause your issue and I
>> added a patch for it.
>> If it persists try setting the sql-in-transactions profile param to false.
>>
>>
>>
>>
>>
>>
>> On Tue, Feb 16, 2010 at 1:30 PM, Yehavi Bourvine <
>> yehavi.bourvine at gmail.com> wrote:
>>
>>>  Most of the queries are ok, only some fail, thus it doesn't look like
>>> permission problem. Furthermore, under 1.0.5pre10 it works for months.
>>>
>>> Might it be thread unsafe function calls? I've found the following while
>>> searching the WEB:
>>>
>>> *According to the MSDN docs, System.Timers.Timer operates in a thread
>>> pool.  If that's the case, your code is breaking the "connections cannot be
>>> shared across threads" rule for SQLit*
>>>
>>> Although it quotes MSDN, it might be related to Linux as well.
>>>
>>>                     Thanks, __Yehavi:
>>>
>>>  2010/2/16 Anthony Minessale <anthony.minessale at gmail.com>
>>>
>>>> That sounds about right.
>>>>
>>>> That error usually has something to do with using db calls on a closed
>>>> file or something along those lines.
>>>> Maybe you have a permission problem on the directory where the db files
>>>> are?
>>>>
>>>>
>>>>
>>>> On Tue, Feb 16, 2010 at 12:59 PM, Yehavi Bourvine <
>>>> yehavi.bourvine at gmail.com> wrote:
>>>>
>>>>>  The OS is Fedora-10 (soon to be upgraded to 12).
>>>>>
>>>>> What I do when I want to test a new version:
>>>>>
>>>>>    - Download the latest one into a fresh directory
>>>>>    - bootstrap.sh, configure and make
>>>>>    - stop Freeswitch, delete everything in lib, mod, bin ,db
>>>>>    - make install and run it.
>>>>>
>>>>>
>>>>> Is there additional place to clean?
>>>>>
>>>>>                         Thanks! __Yehavi:
>>>>>
>>>>>  2010/2/16 Anthony Minessale <anthony.minessale at gmail.com>
>>>>>
>>>>>> you may want to do a clean wipe of all files related to FS then.
>>>>>> you clearly have some problem with legacy something or other because
>>>>>> we don't see that on dozens of dev boxes.
>>>>>>
>>>>>> What os is it?
>>>>>>
>>>>>>
>>>>>>
>>>>>> On Tue, Feb 16, 2010 at 9:27 AM, Yehavi Bourvine <
>>>>>> yehavi.bourvine at gmail.com> wrote:
>>>>>>
>>>>>>>  Tried this, but it didn't help. I delete these DB files before any
>>>>>>> upgrade just to be sure.
>>>>>>>
>>>>>>>                   Thanks! __Yehavi:
>>>>>>>
>>>>>>>  2010/2/16 Anthony Minessale <anthony.minessale at gmail.com>
>>>>>>>
>>>>>>>> try removing all the .db files from /usr/local/freeswitch/db
>>>>>>>>
>>>>>>>>
>>>>>>>>   On Tue, Feb 16, 2010 at 8:56 AM, Yehavi Bourvine <
>>>>>>>> yehavi.bourvine at gmail.com> wrote:
>>>>>>>>
>>>>>>>>>   Hello,
>>>>>>>>>
>>>>>>>>>   After upgrading from 1.0.5pre10 to 1.0.5-20100216-0400 we started
>>>>>>>>> getting the above errors (I append bellow two samples). It seems Freeswitch
>>>>>>>>> fails to read a database using Sqlite.
>>>>>>>>> Anyone have seen this?
>>>>>>>>>
>>>>>>>>> Other details: Fedora 10, SQlite 3.5.9.
>>>>>>>>> We also do SQLite quesries during call setup via LUA from CoreDB.
>>>>>>>>> Is it an SQLite problem?
>>>>>>>>>
>>>>>>>>>                          Thanks! __Yehavi:
>>>>>>>>>
>>>>>>>>> The samples:
>>>>>>>>> 2010-02-16 16:22:13.762679 [ERR] switch_core_sqldb.c:404 SQL ERR
>>>>>>>>> [library routin
>>>>>>>>> e called out of sequence]
>>>>>>>>> delete from sip_dialogs where call_id='
>>>>>>>>> 8656841832142-120172129116107 at 10.64.1.2'
>>>>>>>>>
>>>>>>>>> 2010-02-16 16:42:00.802442 [ERR] switch_core_sqldb.c:722 SQL ERR:
>>>>>>>>> [select call_i
>>>>>>>>>
>>>>>>>>> d,sip_user,sip_host,contact,status,rpid,expires,user_agent,server_user,server_ho
>>>>>>>>>
>>>>>>>>> st,profile_name,hostname,network_ip,network_port,sip_username,sip_realm,mwi_user
>>>>>>>>> ,mwi_host from sip_registrations where profile_name='phones' and
>>>>>>>>> contact like '%
>>>>>>>>> 80635%'] library routine called out of sequence
>>>>>>>>>
>>>>>>>>> _______________________________________________
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>
>>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> --
>>>>>>>> Anthony Minessale II
>>>>>>>>
>>>>>>>> FreeSWITCH http://www.freeswitch.org/
>>>>>>>> ClueCon http://www.cluecon.com/
>>>>>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>>>>>
>>>>>>>> AIM: anthm
>>>>>>>> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
>>>>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>>>>>> IRC: irc.freenode.net #freeswitch
>>>>>>>>
>>>>>>>> FreeSWITCH Developer Conference
>>>>>>>> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>>>>>>>> iax:guest at conference.freeswitch.org/888
>>>>>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>>>>>> pstn:+19193869900
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>> Anthony Minessale II
>>>>>>
>>>>>> FreeSWITCH http://www.freeswitch.org/
>>>>>> ClueCon http://www.cluecon.com/
>>>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>>>
>>>>>> AIM: anthm
>>>>>> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
>>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>>>> IRC: irc.freenode.net #freeswitch
>>>>>>
>>>>>> FreeSWITCH Developer Conference
>>>>>> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>>>>>> iax:guest at conference.freeswitch.org/888
>>>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>>>> pstn:+19193869900
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>>
>>>> --
>>>> Anthony Minessale II
>>>>
>>>> FreeSWITCH http://www.freeswitch.org/
>>>> ClueCon http://www.cluecon.com/
>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>
>>>> AIM: anthm
>>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>> IRC: irc.freenode.net #freeswitch
>>>>
>>>> FreeSWITCH Developer Conference
>>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>>> iax:guest at conference.freeswitch.org/888
>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>> pstn:+19193869900
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100228/170a6557/attachment-0002.html 

From jbrucehopkins at gmail.com  Sun Feb 28 07:37:51 2010
From: jbrucehopkins at gmail.com (Bruce Hopkins)
Date: Sun, 28 Feb 2010 15:37:51 +0000
Subject: [Freeswitch-users] Kernel timer warning with CentOS 5.4
Message-ID: <d03a10e01002280737l61e1db84xa8388a73e932d460@mail.gmail.com>

Hi,

I wonder if anyone would be able to advise please:

When I build FreeSWITCH on yum updated CentOS 5.4, I get a warning when I
start FreeSWITCH that

"Abnormally large timer gap detected"
"Do you have your kernel timer set to greater than 1kHz?  You may experience
audio problems".

I get no such warning if I build on CentOS 5.3, and the test timings it
measures on starting FreeSWITCH do look lower.  All I was doing to upgrade
to Centos5.4 was a yum update on the 5.3 build.

I guess the warning comes from here:
http://fisheye.freeswitch.org/browse/~raw,r=16409/FreeSWITCH/src/switch_time.c

This is all on pretty low spec hardware - a couple of different Dell
optiplex p4's I use for testing.

Does anyone happen to know if I should just stick to Cent)S 5.3, or use 5.4
and not worry about the warnings, or if there is something I can do to fix
the problem it is warning about.  Perhaps it is just that I shouldn't use
such crummy hardware?!

Many thanks in advance
Bruce
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100228/aee29df4/attachment-0002.html 

From gkuri at ieee.org  Sun Feb 28 09:47:14 2010
From: gkuri at ieee.org (Gabriel Kuri)
Date: Sun, 28 Feb 2010 09:47:14 -0800
Subject: [Freeswitch-users] Announcement: FreeSWITCH Add Broadsoft SCA
	Support
In-Reply-To: <d361ca7b1002280149i18e66cbdo2031e7f385f23075@mail.gmail.com>
References: <87f2f3b91001111742k691966cfs82515db77c13d4d4@mail.gmail.com>
	<d361ca7b1002170700x15630ab2tabc6f3d91ebc3303@mail.gmail.com>
	<ADC8B813-4161-481C-877E-93FC012858B4@freeswitch.org>
	<d361ca7b1002182101p60b4e8f8hf4dc95e1d2251e30@mail.gmail.com>
	<8b1c9cda1002182144m65ccf979w71384a1e7a55e2ea@mail.gmail.com>
	<d361ca7b1002182154o52352c62h8c0ec02fa66e8444@mail.gmail.com>
	<2F87B95C-EA23-4072-B845-3F7AE0617BE3@jerris.com>
	<d361ca7b1002182346k71fd602fh6e6725834a6de528@mail.gmail.com>
	<191c3a031002190908r3922efccqc6d223a5fabd458b@mail.gmail.com>
	<d361ca7b1002280149i18e66cbdo2031e7f385f23075@mail.gmail.com>
Message-ID: <8b1c9cda1002280947qcb5ec5anfeef645b447b1866@mail.gmail.com>

That's great it's working, but doesn't that seems more like a
workaround than an actual solution? What if you want to avail of DNS
and use SRV records, do you really want to be hardcoding all your
phones with IPs?

Cheers,
Gabe

On Sun, Feb 28, 2010 at 1:49 AM, Yehavi Bourvine
<yehavi.bourvine at gmail.com> wrote:
> Hello all,
>
> The problem is solved, at least for my Polycoms.
>
> The solution is to have "domain" variable (in vars.xml) set to the IP
> address of the profile that is used for phones registration (the default is
> ok if you have only one interface). Define the phone's registration and
> proxy servers to that address. You must use IP addresses and not DNS names.
>
> ????????????????? Thanks to Anthony for his helpfull tips!? __Yehavi:
>
> 2010/2/19 Anthony Minessale <anthony.minessale at gmail.com>
>>
>> go see my comments on that bug note.
>> be prepared to give us ssh access and call or irc so we can can see you
>> reproducing it.
>>
>> If you are not on the latest firmware on all the phones, we will not
>> continue with this process.
>>
>>
>>
>> On Fri, Feb 19, 2010 at 1:46 AM, Yehavi Bourvine
>> <yehavi.bourvine at gmail.com> wrote:
>>>
>>> A jira issue has been created: MODSOFIA-61.
>>>
>>> ????????????????? Thanks, __Yehavi:
>>>
>>> 2010/2/19 Michael Jerris <mike at jerris.com>
>>>>
>>>> If this issue is not already on jira could you please make sure it gets
>>>> added?
>>>> Mike
>>>>
>>>> On Feb 19, 2010, at 12:54 AM, Yehavi Bourvine
>>>> <yehavi.bourvine at gmail.com> wrote:
>>>>
>>>> Hello Gabe,
>>>>
>>>> ? As you can see - Brian is actively investigating it, so?you can?expect
>>>> for some fix soon...
>>>>
>>>> ?????????????????????? Regards, __Yehavi:
>>>>
>>>> 2010/2/19 Gabriel Kuri <gkuri at ieee.org>
>>>>>
>>>>> > ?When a call arrives, both ring; the one that did not answer gets
>>>>> > only a
>>>>> > cancel mesage without any further notification that the extension is
>>>>> > in use
>>>>> > by the other phone.
>>>>>
>>>>> These are the same exact symptoms I posted about earlier this week,
>>>>> with the Cisco SPA-5xx series phones. I still have yet to figure out
>>>>> why this is happening, if you find out what's going on, please post
>>>>> back the solution, I'd like to know the resolution.
>>>>>
>>>>> Thanks,
>>>>> Gabe
>>>>>
>>>>>
>>>>>
>>>>> On Thu, Feb 18, 2010 at 9:01 PM, Yehavi Bourvine
>>>>> <yehavi.bourvine at gmail.com> wrote:
>>>>> > Thanks Brian. It now works better, but not fully (using 16659M).
>>>>> >
>>>>> > What happens is:
>>>>> >
>>>>> > When one of the Polycoms seize the line it is ok?- the other phone
>>>>> > gets
>>>>> > notification and the extension status is "in use".
>>>>> > When?one of the Polycom phones initiates a call - all is ok:
>>>>> >
>>>>> > The other side sees that the extension is in use.
>>>>> > When it is put to hold all phones?who share this extension see it and
>>>>> > can
>>>>> > pick the call.
>>>>> >
>>>>>
>>>>> >
>>>>> > ???????????????????????? Thanks! __Yehavi:
>>>>> >
>>>>> > 2010/2/17 Brian West <brian at freeswitch.org>
>>>>> >>
>>>>> >> Step 1. Enable manage-shared-appearance=true
>>>>> >>
>>>>> >> Step 2. Now in the phone's config Configure the phone as usually,
>>>>> >> set the
>>>>> >> line shared and DO NOT set the third party name.
>>>>> >>
>>>>> >> Step 3. Reboot
>>>>> >>
>>>>> >> It should work.
>>>>> >>
>>>>> >> I wish someone that has this working would write some wiki docs
>>>>> >> these
>>>>> >> threads about it not working are getting rather old when I know for
>>>>> >> a fact
>>>>> >> they work fine.
>>>>> >>
>>>>> >> The gateway info missing is a gateway you have configured getting a
>>>>> >> notify. ?It has nothing to do with SCA.
>>>>> >>
>>>>> >> /b
>>>>> >>
>>>>> >> On Feb 17, 2010, at 9:00 AM, Yehavi Bourvine wrote:
>>>>> >>
>>>>> >> > .
>>>>> >>
>>>>> >>
>>>>> >> _______________________________________________
>>>>> >> FreeSWITCH-users mailing list
>>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> >>
>>>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> >> http://www.freeswitch.org
>>>>> >
>>>>> > _______________________________________________
>>>>> > FreeSWITCH-users mailing list
>>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> >
>>>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> > http://www.freeswitch.org
>>>>> >
>>>>> >
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From yehavi.bourvine at gmail.com  Sun Feb 28 09:56:21 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Sun, 28 Feb 2010 19:56:21 +0200
Subject: [Freeswitch-users] Announcement: FreeSWITCH Add Broadsoft SCA
	Support
In-Reply-To: <8b1c9cda1002280947qcb5ec5anfeef645b447b1866@mail.gmail.com>
References: <87f2f3b91001111742k691966cfs82515db77c13d4d4@mail.gmail.com>
	<ADC8B813-4161-481C-877E-93FC012858B4@freeswitch.org>
	<d361ca7b1002182101p60b4e8f8hf4dc95e1d2251e30@mail.gmail.com>
	<8b1c9cda1002182144m65ccf979w71384a1e7a55e2ea@mail.gmail.com>
	<d361ca7b1002182154o52352c62h8c0ec02fa66e8444@mail.gmail.com>
	<2F87B95C-EA23-4072-B845-3F7AE0617BE3@jerris.com>
	<d361ca7b1002182346k71fd602fh6e6725834a6de528@mail.gmail.com>
	<191c3a031002190908r3922efccqc6d223a5fabd458b@mail.gmail.com>
	<d361ca7b1002280149i18e66cbdo2031e7f385f23075@mail.gmail.com>
	<8b1c9cda1002280947qcb5ec5anfeef645b447b1866@mail.gmail.com>
Message-ID: <d361ca7b1002280956g8ef520bw5f4fb891cfc2d473@mail.gmail.com>

Hello Gabe,

  I agree that this is somewhat limiting, but with Polycom's central
provisioning (via XML files) I don't see this as a major drawback.

                         __Yehavi:

2010/2/28 Gabriel Kuri <gkuri at ieee.org>

> That's great it's working, but doesn't that seems more like a
> workaround than an actual solution? What if you want to avail of DNS
> and use SRV records, do you really want to be hardcoding all your
> phones with IPs?
>
> Cheers,
> Gabe
>
> On Sun, Feb 28, 2010 at 1:49 AM, Yehavi Bourvine
>  <yehavi.bourvine at gmail.com> wrote:
> > Hello all,
> >
> > The problem is solved, at least for my Polycoms.
> >
> > The solution is to have "domain" variable (in vars.xml) set to the IP
> > address of the profile that is used for phones registration (the default
> is
> > ok if you have only one interface). Define the phone's registration and
> > proxy servers to that address. You must use IP addresses and not DNS
> names.
> >
> >                   Thanks to Anthony for his helpfull tips!  __Yehavi:
> >
> > 2010/2/19 Anthony Minessale <anthony.minessale at gmail.com>
> >>
> >> go see my comments on that bug note.
> >> be prepared to give us ssh access and call or irc so we can can see you
> >> reproducing it.
> >>
> >> If you are not on the latest firmware on all the phones, we will not
> >> continue with this process.
> >>
> >>
> >>
> >> On Fri, Feb 19, 2010 at 1:46 AM, Yehavi Bourvine
> >> <yehavi.bourvine at gmail.com> wrote:
> >>>
> >>> A jira issue has been created: MODSOFIA-61.
> >>>
> >>>                   Thanks, __Yehavi:
> >>>
> >>> 2010/2/19 Michael Jerris <mike at jerris.com>
> >>>>
> >>>> If this issue is not already on jira could you please make sure it
> gets
> >>>> added?
> >>>> Mike
> >>>>
> >>>> On Feb 19, 2010, at 12:54 AM, Yehavi Bourvine
> >>>> <yehavi.bourvine at gmail.com> wrote:
> >>>>
> >>>> Hello Gabe,
> >>>>
> >>>>   As you can see - Brian is actively investigating it, so you
> can expect
> >>>> for some fix soon...
> >>>>
> >>>>                        Regards, __Yehavi:
> >>>>
> >>>> 2010/2/19 Gabriel Kuri <gkuri at ieee.org>
> >>>>>
> >>>>> >  When a call arrives, both ring; the one that did not answer gets
> >>>>> > only a
> >>>>> > cancel mesage without any further notification that the extension
> is
> >>>>> > in use
> >>>>> > by the other phone.
> >>>>>
> >>>>> These are the same exact symptoms I posted about earlier this week,
> >>>>> with the Cisco SPA-5xx series phones. I still have yet to figure out
> >>>>> why this is happening, if you find out what's going on, please post
> >>>>> back the solution, I'd like to know the resolution.
> >>>>>
> >>>>> Thanks,
> >>>>> Gabe
> >>>>>
> >>>>>
> >>>>>
> >>>>> On Thu, Feb 18, 2010 at 9:01 PM, Yehavi Bourvine
> >>>>> <yehavi.bourvine at gmail.com> wrote:
> >>>>> > Thanks Brian. It now works better, but not fully (using 16659M).
> >>>>> >
> >>>>> > What happens is:
> >>>>> >
> >>>>> > When one of the Polycoms seize the line it is ok - the other phone
> >>>>> > gets
> >>>>> > notification and the extension status is "in use".
> >>>>> > When one of the Polycom phones initiates a call - all is ok:
> >>>>> >
> >>>>> > The other side sees that the extension is in use.
> >>>>> > When it is put to hold all phones who share this extension see it
> and
> >>>>> > can
> >>>>> > pick the call.
> >>>>> >
> >>>>>
> >>>>> >
> >>>>> >                          Thanks! __Yehavi:
> >>>>> >
> >>>>> > 2010/2/17 Brian West <brian at freeswitch.org>
> >>>>> >>
> >>>>> >> Step 1. Enable manage-shared-appearance=true
> >>>>> >>
> >>>>> >> Step 2. Now in the phone's config Configure the phone as usually,
> >>>>> >> set the
> >>>>> >> line shared and DO NOT set the third party name.
> >>>>> >>
> >>>>> >> Step 3. Reboot
> >>>>> >>
> >>>>> >> It should work.
> >>>>> >>
> >>>>> >> I wish someone that has this working would write some wiki docs
> >>>>> >> these
> >>>>> >> threads about it not working are getting rather old when I know
> for
> >>>>> >> a fact
> >>>>> >> they work fine.
> >>>>> >>
> >>>>> >> The gateway info missing is a gateway you have configured getting
> a
> >>>>> >> notify.  It has nothing to do with SCA.
> >>>>> >>
> >>>>> >> /b
> >>>>> >>
> >>>>> >> On Feb 17, 2010, at 9:00 AM, Yehavi Bourvine wrote:
> >>>>> >>
> >>>>> >> > .
> >>>>> >>
> >>>>> >>
> >>>>> >> _______________________________________________
> >>>>> >> FreeSWITCH-users mailing list
> >>>>> >> FreeSWITCH-users at lists.freeswitch.org
> >>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>> >>
> >>>>> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>> >> http://www.freeswitch.org
> >>>>> >
> >>>>> > _______________________________________________
> >>>>> > FreeSWITCH-users mailing list
> >>>>> > FreeSWITCH-users at lists.freeswitch.org
> >>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>> >
> >>>>> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>> > http://www.freeswitch.org
> >>>>> >
> >>>>> >
> >>>>>
> >>>>> _______________________________________________
> >>>>> FreeSWITCH-users mailing list
> >>>>> FreeSWITCH-users at lists.freeswitch.org
> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>>
> >>>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>> http://www.freeswitch.org
> >>>>
> >>>> _______________________________________________
> >>>> FreeSWITCH-users mailing list
> >>>> FreeSWITCH-users at lists.freeswitch.org
> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>> http://www.freeswitch.org
> >>>>
> >>>> _______________________________________________
> >>>> FreeSWITCH-users mailing list
> >>>> FreeSWITCH-users at lists.freeswitch.org
> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>> http://www.freeswitch.org
> >>>>
> >>>
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>>
> >>
> >>
> >>
> >> --
> >> Anthony Minessale II
> >>
> >> FreeSWITCH http://www.freeswitch.org/
> >> ClueCon http://www.cluecon.com/
> >> Twitter: http://twitter.com/FreeSWITCH_wire
> >>
> >> AIM: anthm
> >> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >> IRC: irc.freenode.net #freeswitch
> >>
> >> FreeSWITCH Developer Conference
> >> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> >> iax:guest at conference.freeswitch.org/888
> >> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >> pstn:+19193869900
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100228/763fa274/attachment-0002.html 

From jbrucehopkins at gmail.com  Sun Feb 28 10:03:03 2010
From: jbrucehopkins at gmail.com (Bruce Hopkins)
Date: Sun, 28 Feb 2010 18:03:03 +0000
Subject: [Freeswitch-users] Kernel timer warning with CentOS 5.4
In-Reply-To: <d03a10e01002280737l61e1db84xa8388a73e932d460@mail.gmail.com>
References: <d03a10e01002280737l61e1db84xa8388a73e932d460@mail.gmail.com>
Message-ID: <d03a10e01002281003j669b6e75t31db4dc351f38ece@mail.gmail.com>

OK - I've realised I do get the same warning with CentOS 5.3, it just goes
past more quickly so I didn't see it.  Maybe it is just the hardware ....

On 28 February 2010 15:37, Bruce Hopkins <jbrucehopkins at gmail.com> wrote:

> Hi,
>
> I wonder if anyone would be able to advise please:
>
> When I build FreeSWITCH on yum updated CentOS 5.4, I get a warning when I
> start FreeSWITCH that
>
> "Abnormally large timer gap detected"
> "Do you have your kernel timer set to greater than 1kHz?  You may
> experience audio problems".
>
> I get no such warning if I build on CentOS 5.3, and the test timings it
> measures on starting FreeSWITCH do look lower.  All I was doing to upgrade
> to Centos5.4 was a yum update on the 5.3 build.
>
> I guess the warning comes from here:
> http://fisheye.freeswitch.org/browse/~raw,r=16409/FreeSWITCH/src/switch_time.c<http://fisheye.freeswitch.org/browse/%7Eraw,r=16409/FreeSWITCH/src/switch_time.c>
>
> This is all on pretty low spec hardware - a couple of different Dell
> optiplex p4's I use for testing.
>
> Does anyone happen to know if I should just stick to Cent)S 5.3, or use 5.4
> and not worry about the warnings, or if there is something I can do to fix
> the problem it is warning about.  Perhaps it is just that I shouldn't use
> such crummy hardware?!
>
> Many thanks in advance
> Bruce
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100228/c4424531/attachment-0002.html 

From brian at freeswitch.org  Sun Feb 28 10:04:12 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 28 Feb 2010 12:04:12 -0600
Subject: [Freeswitch-users] Announcement: FreeSWITCH Add Broadsoft SCA
	Support
In-Reply-To: <d361ca7b1002280956g8ef520bw5f4fb891cfc2d473@mail.gmail.com>
References: <87f2f3b91001111742k691966cfs82515db77c13d4d4@mail.gmail.com>
	<ADC8B813-4161-481C-877E-93FC012858B4@freeswitch.org>
	<d361ca7b1002182101p60b4e8f8hf4dc95e1d2251e30@mail.gmail.com>
	<8b1c9cda1002182144m65ccf979w71384a1e7a55e2ea@mail.gmail.com>
	<d361ca7b1002182154o52352c62h8c0ec02fa66e8444@mail.gmail.com>
	<2F87B95C-EA23-4072-B845-3F7AE0617BE3@jerris.com>
	<d361ca7b1002182346k71fd602fh6e6725834a6de528@mail.gmail.com>
	<191c3a031002190908r3922efccqc6d223a5fabd458b@mail.gmail.com>
	<d361ca7b1002280149i18e66cbdo2031e7f385f23075@mail.gmail.com>
	<8b1c9cda1002280947qcb5ec5anfeef645b447b1866@mail.gmail.com>
	<d361ca7b1002280956g8ef520bw5f4fb891cfc2d473@mail.gmail.com>
Message-ID: <987F87B1-FA49-47B4-9823-5A7140459AFD@freeswitch.org>

Really?  Come on guys... the feature is something you can't find elsewhere without paying and you're all still not totally pleased with it?  

/me shakes his head.

We have already started talking about how to make the feature more robust. 

/b

On Feb 28, 2010, at 11:56 AM, Yehavi Bourvine wrote:

> Hello Gabe,
>  
>   I agree that this is somewhat limiting, but with Polycom's central provisioning (via XML files) I don't see this as a major drawback.
>  
>                          __Yehavi:




From yehavi.bourvine at gmail.com  Sun Feb 28 10:23:53 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Sun, 28 Feb 2010 20:23:53 +0200
Subject: [Freeswitch-users] Announcement: FreeSWITCH Add Broadsoft SCA
	Support
In-Reply-To: <987F87B1-FA49-47B4-9823-5A7140459AFD@freeswitch.org>
References: <87f2f3b91001111742k691966cfs82515db77c13d4d4@mail.gmail.com>
	<8b1c9cda1002182144m65ccf979w71384a1e7a55e2ea@mail.gmail.com>
	<d361ca7b1002182154o52352c62h8c0ec02fa66e8444@mail.gmail.com>
	<2F87B95C-EA23-4072-B845-3F7AE0617BE3@jerris.com>
	<d361ca7b1002182346k71fd602fh6e6725834a6de528@mail.gmail.com>
	<191c3a031002190908r3922efccqc6d223a5fabd458b@mail.gmail.com>
	<d361ca7b1002280149i18e66cbdo2031e7f385f23075@mail.gmail.com>
	<8b1c9cda1002280947qcb5ec5anfeef645b447b1866@mail.gmail.com>
	<d361ca7b1002280956g8ef520bw5f4fb891cfc2d473@mail.gmail.com>
	<987F87B1-FA49-47B4-9823-5A7140459AFD@freeswitch.org>
Message-ID: <d361ca7b1002281023s38d81815t48d985aa830a87c9@mail.gmail.com>

This is one of the most important feature my users want. They don't care how
I do it, they are just happy it works.

                        Thanks! __Yehavi:

2010/2/28 Brian West <brian at freeswitch.org>

> Really?  Come on guys... the feature is something you can't find elsewhere
> without paying and you're all still not totally pleased with it?
>
> /me shakes his head.
>
> We have already started talking about how to make the feature more robust.
>
> /b
>
> On Feb 28, 2010, at 11:56 AM, Yehavi Bourvine wrote:
>
> > Hello Gabe,
> >
> >   I agree that this is somewhat limiting, but with Polycom's central
> provisioning (via XML files) I don't see this as a major drawback.
> >
> >                          __Yehavi:
>
>
>  _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100228/0dcb42a2/attachment-0002.html 

From anthony.minessale at gmail.com  Sun Feb 28 21:06:38 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sun, 28 Feb 2010 23:06:38 -0600
Subject: [Freeswitch-users] SQL ERR: library routine called out of
	sequence
In-Reply-To: <191c3a031002282105p4ed80135jc115a007ec7e0d4a@mail.gmail.com>
References: <d361ca7b1002160656w7d616f6dn5e80087ebde94dbe@mail.gmail.com>
	<d361ca7b1002160727ucb6e0c8id62e803757333213@mail.gmail.com>
	<191c3a031002160755q569fe2ednfefb01b6d9cf7f0e@mail.gmail.com>
	<d361ca7b1002161059w1ef9484dr87e69a33aa2b62fc@mail.gmail.com>
	<191c3a031002161110s637a48a2r717ae94557fe9e9a@mail.gmail.com>
	<d361ca7b1002161130k3eb2d88by3100b30a9d29bd51@mail.gmail.com>
	<191c3a031002161230h2b6e6f9wf3f65ddb71f08297@mail.gmail.com>
	<d361ca7b1002280514p71003588t8a0daebce5ba9e4d@mail.gmail.com>
	<d361ca7b1002280647l700206cfv6e130351d8ac6d5@mail.gmail.com>
	<191c3a031002282105p4ed80135jc115a007ec7e0d4a@mail.gmail.com>
Message-ID: <191c3a031002282106t59801ddctb12c4fab160e9a07@mail.gmail.com>

You probably had it right in the last email.
Polling the core's db from lua is not recommended.
We use sla extensively and never once see your issue.

On Feb 28, 2010 8:55 AM, "Yehavi Bourvine" <yehavi.bourvine at gmail.com>
wrote:

After a few more tests I *think* it is related to SLA. Since it cannot be
reproduced consistenty I say "think".

What I have is:
- 80635 which is an SLA extension between two Polycoms.
- 80636 which is a private extension on Polycom
- 80632 which is a Cisco extension.
- 86111 which is connected behind a Cisco SIP<->PSTN gateway. Since this
gateway doesn't hav any login information it is not defined as a gateway but
accepted via ACL.

The tests that passed ok:
- From/to 80636/80632 to 80635 (i.e. - all internal).
- From/to 80635 to 80636/80632 (i.e. - all internal)
- From/to 80636/80632 to 86111 and vice versa (i.e. via the gateway, single
extensions).

The one that fails after a few attempts:
- From 86111 to 80635 (i.e. from the gateway to a shared extension).

I hope that this gives some more clues.

                          Thanks, __Yehavi:

2010/2/28 Yehavi Bourvine <yehavi.bourvine at gmail.com>


>
> Hello,
>
>   I've installed a machine with CentOS-5.4 and the latest FreeSwitch
(16841M). The...

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100228/bd2346ea/attachment-0002.html 

From lakindia89 at gmail.com  Sun Feb 28 22:16:20 2010
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Mon, 1 Mar 2010 11:46:20 +0530
Subject: [Freeswitch-users] smg_prid not bridging the call
In-Reply-To: <c4d05cbe1002271035g6112b277hdf305b61233b94ad@mail.gmail.com>
References: <7d79b3931002252155t47c86968q9e451482926d93ac@mail.gmail.com>
	<c4d05cbe1002260731v29dbc428i1e72d24a220c62eb@mail.gmail.com>
	<7d79b3931002262057g3b325bctcdbcfb833d4aaf55@mail.gmail.com>
	<7d79b3931002262102w79e16ae1nd786600b44dfb758@mail.gmail.com>
	<7d79b3931002262113s49495e7cga59715f41fdbb35d@mail.gmail.com>
	<c4d05cbe1002271035g6112b277hdf305b61233b94ad@mail.gmail.com>
Message-ID: <7d79b3931002282216w2e2ec844q36d28b3f50423ec4@mail.gmail.com>

Dear Moy,
That's didn't seem to solve the problem.
I gave the following command.
originate
{origination_caller_id_number=04439114600}openzap/smg_prid/a/9952248266
&bridge({origination_caller_id_number=04439114600}openzap/smg_prid/a/9976975781)

The D-Chan Log is
http://pastebin.freeswitch.org/12268

Kindly refer the attached pcap file that I captured with wanpipemon utility.
I think it might help.


On Sun, Feb 28, 2010 at 12:05 AM, Moises Silva <moises.silva at gmail.com>wrote:

> I believe the problem FreeSWITCH is setting that as a default callerid
> name, which your telco does not like.
>
> Try setting the caller id name and number by yourself as explained in the
> "originate" section here http://wiki.freeswitch.org/wiki/Mod_commands
>  <http://wiki.freeswitch.org/wiki/Mod_commands>
>
> On Sat, Feb 27, 2010 at 12:13 AM, lakshmanan ganapathy <
> lakindia89 at gmail.com> wrote:
>
>> I think it says Invalid Information Element for the DISPLAY
>> smg_prid/a/8122133885!!!
>> correct?? If so, can you please help me to solve this?
>>
>>
>> On Sat, Feb 27, 2010 at 10:32 AM, lakshmanan ganapathy <
>> lakindia89 at gmail.com> wrote:
>>
>>> In the Dchan log it is saying Invalid Information Elements. That might be
>>> a problem??? But I even don't know why it is saying Invalid Information
>>> Element??
>>> Please guide me!!!
>>>
>>>
>>>
>>> On Sat, Feb 27, 2010 at 10:27 AM, lakshmanan ganapathy <
>>> lakindia89 at gmail.com> wrote:
>>>
>>>> Dear Moy,
>>>> Here are the details:
>>>>
>>>> FreeSwitch Log:
>>>> http://pastebin.freeswitch.org/12256
>>>>
>>>>  /var/log/sangoma_pri/dchan_<span-no>.log:
>>>>  http://pastebin.freeswitch.org/12257
>>>>
>>>> /var/log/sangoma_mgd.log:
>>>> http://pastebin.freeswitch.org/12258
>>>>
>>>> smg_pri.conf
>>>> http://pastebin.freeswitch.org/12259
>>>>
>>>>
>>>>
>>>> On Fri, Feb 26, 2010 at 9:01 PM, Moises Silva <moises.silva at gmail.com>wrote:
>>>>
>>>>> Hello lakshmanan,
>>>>>
>>>>> Please enable the dchan_log=q931 in /etc/wanpipe/smg_pri.conf and then
>>>>> restart it (smg_ctrl restart), then pastebin the logs
>>>>>
>>>>>  /var/log/sangoma_pri/dchan_<span-no>.log
>>>>> /var/log/sangoma_mgd.log
>>>>>
>>>>> That will contain the Q931 details (if any). Also pastebin your
>>>>> smg_pri.conf.
>>>>>
>>>>> Also enable the FreeSWITCH debug logging (see the FreeSWITCH wiki for
>>>>> details about that) and paste them too.
>>>>>
>>>>> --
>>>>> Moises Silva
>>>>> Senior Software Engineer
>>>>> Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON
>>>>> L3R 9T3 Canada
>>>>> t. 1 905 474 1990 x 128 | e. moy at sangoma.com
>>>>>
>>>>>  On Fri, Feb 26, 2010 at 12:55 AM, lakshmanan ganapathy <
>>>>> lakindia89 at gmail.com> wrote:
>>>>>
>>>>>>  Dear all,
>>>>>> I'm having a A102 Sangoma hardware. I configured it with freeswitch.
>>>>>> wanrouter status, says both the port as connected.
>>>>>> My smg_prid version is
>>>>>>
>>>>>> Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: ================System
>>>>>> restart=============
>>>>>> Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: = Sangoma PRI Protocol
>>>>>> Stack Daemon       =
>>>>>> Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: = Version:
>>>>>> 1.54                           =
>>>>>> Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: = Date: Feb 15
>>>>>> 2010                       =
>>>>>> Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: = Wanpipe Release:
>>>>>> wanpipe-3.5.8.6       =
>>>>>> Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: = Revision:Revision:
>>>>>> 15288                =
>>>>>> Feb 26 16:08:14 FMS-FreeSwitch
>>>>>> sangoma_prid:
>>>>>> ===========================================
>>>>>>
>>>>>> My freeswitch version is 16729.
>>>>>> I started freeswitch.
>>>>>>
>>>>>> oz list
>>>>>> +OK
>>>>>> span: 1 (smg_prid)
>>>>>> type: Sangoma (boost)
>>>>>> chan_count: 60
>>>>>> dialplan: XML
>>>>>> context: default
>>>>>> dial_regex:
>>>>>> fail_dial_regex:
>>>>>> hold_music:
>>>>>> analog_options none
>>>>>>
>>>>>> I originated a call as
>>>>>> originate openzap/smg_prid/a/9952248266 &park(), which hits my mobile.
>>>>>>
>>>>>> But when I issued the following command:
>>>>>> originate openzap/smg_prid/a/9952248266
>>>>>> &bridge(openzap/smg_prid/a/8122133885)
>>>>>> It rings my mobile (9952248266) first, but after that the following
>>>>>> error was displayed
>>>>>>
>>>>>> 2010-02-26 16:20:51.736080 [ERR] switch_ivr_originate.c:2387 Cannot
>>>>>> create outgoing channel of type [openzap] cause: [NORMAL_CIRCUIT_CONGESTION]
>>>>>> The call got ended in my mobile.
>>>>>>
>>>>>> Freeswitch log and smg_pri.conf
>>>>>>           http://pastebin.freeswitch.org/12248
>>>>>> openzap.conf:
>>>>>> [span wanpipe smg_prid]
>>>>>> name => smg_prid
>>>>>> trunk_type =>e1
>>>>>> b-channel => 1:1-15
>>>>>> b-channel => 1:17-31
>>>>>> trunk_type =>e1
>>>>>> b-channel => 2:1-15
>>>>>> b-channel => 2:17-31
>>>>>>
>>>>>> openzap.conf.xml:
>>>>>> <configuration name="openzap.conf" description="OpenZAP
>>>>>> Configuration">
>>>>>>   <settings>
>>>>>>     <param name="debug" value="0"/>
>>>>>>   </settings>
>>>>>>   <boost_spans>
>>>>>>         <span id="smg_prid">
>>>>>>         <param name="dialplan" value="XML"/>
>>>>>>         <param name="context" value="default"/>
>>>>>>         </span>
>>>>>>   </boost_spans>
>>>>>> </configuration>
>>>>>>
>>>>>> Please guide me to setup this one!!.
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Moises Silva
> Senior Software Engineer
> Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R
> 9T3 Canada
> t. 1 905 474 1990 x 128 | e. moy at sangoma.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100301/5d790312/attachment-0002.html 
-------------- next part --------------
A non-text attachment was scrubbed...
Name: isdn.pcap
Type: application/cap
Size: 1231 bytes
Desc: not available
Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100301/5d790312/attachment-0002.bin 

From srinivas.ksvreddy at gmail.com  Sun Feb 28 23:39:56 2010
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Mon, 1 Mar 2010 13:09:56 +0530
Subject: [Freeswitch-users] call routing from freeswitch based on INVITE
Message-ID: <f8af5741002282339l52fe9ad7q44a7e4c65f82cd95@mail.gmail.com>

I have two sipservers like server1 and server2,  if sever1 receives invite
packet like

INVITE From:   1000 at server1.domain.com
           To:   1002 at server2.railvoice.com.

how can i route the invite packet to server2 from server1,

Thanks & Regards
Srinivasula Reddy K
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100301/59f89744/attachment-0002.html 

From tomek.augustyn at gmail.com  Sat Feb 27 00:33:44 2010
From: tomek.augustyn at gmail.com (Tomasz Augustyn)
Date: Sat, 27 Feb 2010 09:33:44 +0100
Subject: [Freeswitch-users] smg_prid not bridging the call
Message-ID: <6d15d07f1002270033n71d8ac85u1895b05f75540e63@mail.gmail.com>

Hello,

I had similar problem and I think it is more a problem between Sangoma card
and your E1 provider than with freeswitch.

In my case it was necessary to set "origination_caller_id_number" to one of
the telephone numbers linked to my E1 line. In other case the calls were
rejected with "invalid information element" error.

You can try Sangoma's support they are very helpful.

Tomasz Augustyn


---------- Forwarded message ----------
From: lakshmanan ganapathy <lakindia89 at gmail.com>
To: freeswitch-users at lists.freeswitch.org
Date: Sat, 27 Feb 2010 10:32:08 +0530
Subject: Re: [Freeswitch-users] smg_prid not bridging the call
In the Dchan log it is saying Invalid Information Elements. That might be a
problem??? But I even don't know why it is saying Invalid Information
Element??
Please guide me!!!
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100227/88e0b91e/attachment-0002.html 

From mmg at transtelco.net  Sun Feb 28 21:20:28 2010
From: mmg at transtelco.net (=?iso-8859-1?Q?Manuel_Mar=EDn?=)
Date: Mon, 1 Mar 2010 00:20:28 -0500
Subject: [Freeswitch-users] High CPU usage 1.0.5
Message-ID: <4502F03F8260234AB94179D6E1BDD0CF3F53555A73@VMBX113.ihostexchange.net>

Dear freeswitch group

I Just upgraded a system from 1.0.4 to 1.0.5 and we are seeing high CPU usage even if there are only a few calls on the system or no calls at all. We are running Debian with kernel 2.6.26-2-686

Anyone experimenting a similar issue?


Thanks in advance


freeswitch at internal> version
FreeSWITCH Version 1.0.5-20100225-0400 (16810M)






Manuel Mar?n
Transtelco
US 1.915.2172232
MX 52.656.6921109
FAX 1.915.2311214

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100301/22dc4991/attachment-0002.html